diff options
| author | Denys Dmytriyenko <denis@denix.org> | 2009-03-17 14:32:59 -0400 |
|---|---|---|
| committer | Denys Dmytriyenko <denis@denix.org> | 2009-03-17 14:32:59 -0400 |
| commit | 709c4d66e0b107ca606941b988bad717c0b45d9b (patch) | |
| tree | 37ee08b1eb308f3b2b6426d5793545c38396b838 /recipes/linux/logicpd-pxa270-2.6.19.2/asoc-v0.12.4.patch | |
| parent | fa6cd5a3b993f16c27de4ff82b42684516d433ba (diff) | |
rename packages/ to recipes/ per earlier agreement
See links below for more details:
http://thread.gmane.org/gmane.comp.handhelds.openembedded/21326
http://thread.gmane.org/gmane.comp.handhelds.openembedded/21816
Signed-off-by: Denys Dmytriyenko <denis@denix.org>
Acked-by: Mike Westerhof <mwester@dls.net>
Acked-by: Philip Balister <philip@balister.org>
Acked-by: Khem Raj <raj.khem@gmail.com>
Acked-by: Marcin Juszkiewicz <hrw@openembedded.org>
Acked-by: Koen Kooi <koen@openembedded.org>
Acked-by: Frans Meulenbroeks <fransmeulenbroeks@gmail.com>
Diffstat (limited to 'recipes/linux/logicpd-pxa270-2.6.19.2/asoc-v0.12.4.patch')
| -rw-r--r-- | recipes/linux/logicpd-pxa270-2.6.19.2/asoc-v0.12.4.patch | 31712 |
1 files changed, 31712 insertions, 0 deletions
diff --git a/recipes/linux/logicpd-pxa270-2.6.19.2/asoc-v0.12.4.patch b/recipes/linux/logicpd-pxa270-2.6.19.2/asoc-v0.12.4.patch new file mode 100644 index 0000000000..420f4aa2e2 --- /dev/null +++ b/recipes/linux/logicpd-pxa270-2.6.19.2/asoc-v0.12.4.patch @@ -0,0 +1,31712 @@ +Index: linux-2.6-pxa-new/Documentation/sound/alsa/soc/DAI.txt +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/Documentation/sound/alsa/soc/DAI.txt +@@ -0,0 +1,546 @@ ++ASoC currently supports the three main Digital Audio Interfaces (DAI) found on ++SoC controllers and portable audio CODECS today, namely AC97, I2S and PCM. ++ ++ ++AC97 ++==== ++ ++ AC97 is a five wire interface commonly found on many PC sound cards. It is ++now also popular in many portable devices. This DAI has a reset line and time ++multiplexes its data on its SDATA_OUT (playback) and SDATA_IN (capture) lines. ++The bit clock (BCLK) is always driven by the CODEC (usually 12.288MHz) and the ++frame (FRAME) (usually 48kHz) is always driven by the controller. Each AC97 ++frame is 21uS long and is divided into 13 time slots. ++ ++The AC97 specification can be found at :- ++http://www.intel.com/design/chipsets/audio/ac97_r23.pdf ++ ++ ++I2S ++=== ++ ++ I2S is a common 4 wire DAI used in HiFi, STB and portable devices. The Tx and ++Rx lines are used for audio transmision, whilst the bit clock (BCLK) and ++left/right clock (LRC) synchronise the link. I2S is flexible in that either the ++controller or CODEC can drive (master) the BCLK and LRC clock lines. Bit clock ++usually varies depending on the sample rate and the master system clock ++(SYSCLK). LRCLK is the same as the sample rate. A few devices support separate ++ADC and DAC LRCLK's, this allows for similtanious capture and playback at ++different sample rates. ++ ++I2S has several different operating modes:- ++ ++ o I2S - MSB is transmitted on the falling edge of the first BCLK after LRC ++ transition. ++ ++ o Left Justified - MSB is transmitted on transition of LRC. ++ ++ o Right Justified - MSB is transmitted sample size BCLK's before LRC ++ transition. ++ ++PCM ++=== ++ ++PCM is another 4 wire interface, very similar to I2S, that can support a more ++flexible protocol. It has bit clock (BCLK) and sync (SYNC) lines that are used ++to synchronise the link whilst the Tx and Rx lines are used to transmit and ++receive the audio data. Bit clock usually varies depending on sample rate ++whilst sync runs at the sample rate. PCM also supports Time Division ++Multiplexing (TDM) in that several devices can use the bus similtaniuosly (This ++is sometimes referred to as network mode). ++ ++Common PCM operating modes:- ++ ++ o Mode A - MSB is transmitted on falling edge of first BCLK after FRAME/SYNC. ++ ++ o Mode B - MSB is transmitted on rising edge of FRAME/SYNC. ++ ++ ++ASoC DAI Configuration ++====================== ++ ++Every CODEC DAI and SoC DAI must have their capabilities defined in order to ++be configured together at runtime when the audio and clocking parameters are ++known. This is achieved by creating an array of struct snd_soc_hw_mode in the ++the CODEC and SoC interface drivers. Each element in the array describes a DAI ++mode and each mode is usually based upon the DAI system clock to sample rate ++ratio (FS). ++ ++i.e. 48k sample rate @ 256 FS = sytem clock of 12.288 MHz ++ 48000 * 256 = 12288000 ++ ++The CPU and Codec DAI modes are then ANDed together at runtime to determine the ++rutime DAI configuration for both the Codec and CPU. ++ ++When creating a new codec or SoC DAI it's probably best to start of with a few ++sample rates first and then test your interface. ++ ++struct snd_soc_dai_mode is defined (in soc.h) as:- ++ ++/* SoC DAI mode */ ++struct snd_soc_dai_mode { ++ u16 fmt; /* SND_SOC_DAIFMT_* */ ++ u16 tdm; /* SND_SOC_HWTDM_* */ ++ u64 pcmfmt; /* SNDRV_PCM_FMTBIT_* */ ++ u16 pcmrate; /* SND_SOC_HWRATE_* */ ++ u16 pcmdir:2; /* SND_SOC_HWDIR_* */ ++ u16 flags:8; /* hw flags */ ++ u16 fs; /* mclk to rate divider */ ++ u64 bfs; /* mclk to bclk dividers */ ++ unsigned long priv; /* private mode data */ ++}; ++ ++fmt: ++---- ++This field defines the DAI mode hardware format (e.g. I2S settings) and ++supports the following settings:- ++ ++ 1) hardware DAI formats ++ ++#define SND_SOC_DAIFMT_I2S (1 << 0) /* I2S mode */ ++#define SND_SOC_DAIFMT_RIGHT_J (1 << 1) /* Right justified mode */ ++#define SND_SOC_DAIFMT_LEFT_J (1 << 2) /* Left Justified mode */ ++#define SND_SOC_DAIFMT_DSP_A (1 << 3) /* L data msb after FRM */ ++#define SND_SOC_DAIFMT_DSP_B (1 << 4) /* L data msb during FRM */ ++#define SND_SOC_DAIFMT_AC97 (1 << 5) /* AC97 */ ++ ++ 2) hw DAI signal inversions ++ ++#define SND_SOC_DAIFMT_NB_NF (1 << 8) /* normal bit clock + frame */ ++#define SND_SOC_DAIFMT_NB_IF (1 << 9) /* normal bclk + inv frm */ ++#define SND_SOC_DAIFMT_IB_NF (1 << 10) /* invert bclk + nor frm */ ++#define SND_SOC_DAIFMT_IB_IF (1 << 11) /* invert bclk + frm */ ++ ++ 3) hw clock masters ++ This is wrt the codec, the inverse is true for the interface ++ i.e. if the codec is clk and frm master then the interface is ++ clk and frame slave. ++ ++#define SND_SOC_DAIFMT_CBM_CFM (1 << 12) /* codec clk & frm master */ ++#define SND_SOC_DAIFMT_CBS_CFM (1 << 13) /* codec clk slave & frm master */ ++#define SND_SOC_DAIFMT_CBM_CFS (1 << 14) /* codec clk master & frame slave */ ++#define SND_SOC_DAIFMT_CBS_CFS (1 << 15) /* codec clk & frm slave */ ++ ++At least one option from each section must be selected. Multiple selections are ++also supported e.g. ++ ++ .fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_RIGHT_J | \ ++ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_NB_IF | SND_SOC_DAIFMT_IB_NF | \ ++ SND_SOC_DAIFMT_IB_IF ++ ++ ++tdm: ++------ ++This field defines the Time Division Multiplexing left and right word ++positions for the DAI mode if applicable. Set to SND_SOC_DAITDM_LRDW(0,0) for ++no TDM. ++ ++ ++pcmfmt: ++--------- ++The hardware PCM format. This describes the PCM formats supported by the DAI ++mode e.g. ++ ++ .pcmfmt = SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S20_3LE | \ ++ SNDRV_PCM_FORMAT_S24_3LE ++ ++pcmrate: ++---------- ++The PCM sample rates supported by the DAI mode. e.g. ++ ++ .pcmrate = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ ++ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ ++ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 ++ ++ ++pcmdir: ++--------- ++The stream directions supported by this mode. e.g. playback and capture ++ ++ ++flags: ++-------- ++The DAI hardware flags supported by the mode. ++ ++/* use bfs mclk divider mode (BCLK = MCLK / x) */ ++#define SND_SOC_DAI_BFS_DIV 0x1 ++/* use bfs rate mulitplier (BCLK = RATE * x)*/ ++#define SND_SOC_DAI_BFS_RATE 0x2 ++/* use bfs rcw multiplier (BCLK = RATE * CHN * WORD SIZE) */ ++#define SND_SOC_DAI_BFS_RCW 0x4 ++/* capture and playback can use different clocks */ ++#define SND_SOC_DAI_ASYNC 0x8 ++ ++NOTE: Bitclock division and mulitiplication modes can be safely matched by the ++core logic. ++ ++ ++fs: ++----- ++The FS supported by this DAI mode FS is the ratio between the system clock and ++the sample rate. See above ++ ++bfs: ++------ ++BFS is the ratio of BCLK to MCLK or the ratio of BCLK to sample rate (this ++depends on the codec or CPU DAI). ++ ++The BFS supported by the DAI mode. This can either be the ratio between the ++bitclock (BCLK) and the sample rate OR the ratio between the system clock and ++the sample rate. Depends on the flags above. ++ ++priv: ++----- ++private codec mode data. ++ ++ ++ ++Examples ++======== ++ ++Note that Codec DAI and CPU DAI examples are interchangeable in these examples ++as long as the bus master is reversed. i.e. ++ ++ SND_SOC_DAIFMT_CBM_CFM would become SND_SOC_DAIFMT_CBS_CFS ++ and vice versa. ++ ++This applies to all SND_SOC_DAIFMT_CB*_CF*. ++ ++Example 1 ++--------- ++ ++Simple codec that only runs at 8k & 48k @ 256FS in master mode, can generate a ++BCLK of either MCLK/2 or MCLK/4. ++ ++ /* codec master */ ++ { ++ .fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = SNDRV_PCM_FORMAT_S16_LE, ++ .pcmrate = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_48000, ++ .pcmdir = SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 256, ++ .bfs = SND_SOC_FSBD(2) | SND_SOC_FSBD(4), ++ } ++ ++ ++Example 2 ++--------- ++Simple codec that only runs at 8k & 48k @ 256FS in master mode, can generate a ++BCLK of either Rate * 32 or Rate * 64. ++ ++ /* codec master */ ++ { ++ .fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = SNDRV_PCM_FORMAT_S16_LE, ++ .pcmrate = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_48000, ++ .pcmdir = SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE, ++ .flags = SND_SOC_DAI_BFS_RATE, ++ .fs = 256, ++ .bfs = 32, ++ }, ++ { ++ .fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = SNDRV_PCM_FORMAT_S16_LE, ++ .pcmrate = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_48000, ++ .pcmdir = SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE, ++ .flags = SND_SOC_DAI_BFS_RATE, ++ .fs = 256, ++ .bfs = 64, ++ }, ++ ++ ++Example 3 ++--------- ++Codec that runs at 8k & 48k @ 256FS in master mode, can generate a BCLK that ++is a multiple of Rate * channels * word size. (RCW) i.e. ++ ++ BCLK = 8000 * 2 * 16 (8k, stereo, 16bit) ++ = 256kHz ++ ++This codecs supports a RCW multiple of 1,2 ++ ++ { ++ .fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = SNDRV_PCM_FORMAT_S16_LE, ++ .pcmrate = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_48000, ++ .pcmdir = SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE, ++ .flags = SND_SOC_DAI_BFS_RCW, ++ .fs = 256, ++ .bfs = SND_SOC_FSBW(1) | SND_SOC_FSBW(2), ++ } ++ ++ ++Example 4 ++--------- ++Codec that only runs at 8k & 48k @ 256FS in master mode, can generate a ++BCLK of either Rate * 32 or Rate * 64. Codec can also run in slave mode as long ++as BCLK is rate * 32 or rate * 64. ++ ++ /* codec master */ ++ { ++ .fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = SNDRV_PCM_FORMAT_S16_LE, ++ .pcmrate = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_48000, ++ .pcmdir = SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE, ++ .flags = SND_SOC_DAI_BFS_RATE, ++ .fs = 256, ++ .bfs = 32, ++ }, ++ { ++ .fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = SNDRV_PCM_FORMAT_S16_LE, ++ .pcmrate = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_48000, ++ .pcmdir = SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE, ++ .flags = SND_SOC_DAI_BFS_RATE, ++ .fs = 256, ++ .bfs = 64, ++ }, ++ ++ /* codec slave */ ++ { ++ .fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS, ++ .pcmfmt = SNDRV_PCM_FORMAT_S16_LE, ++ .pcmdir = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_48000, ++ .pcmdir = SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE, ++ .flags = SND_SOC_DAI_BFS_RATE, ++ .fs = SND_SOC_FS_ALL, ++ .bfs = 32, ++ }, ++ { ++ .fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS, ++ .pcmfmt = SNDRV_PCM_FORMAT_S16_LE, ++ .pcmdir = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_48000, ++ .pcmdir = SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE, ++ .flags = SND_SOC_DAI_BFS_RATE, ++ .fs = SND_SOC_FS_ALL, ++ .bfs = 64, ++ }, ++ ++ ++Example 5 ++--------- ++Codec that only runs at 8k, 16k, 32k, 48k, 96k @ 128FS, 192FS & 256FS in master ++mode and can generate a BCLK of MCLK / (1,2,4,8,16). Codec can also run in slave ++mode as and does not care about FS or BCLK (as long as there is enough bandwidth). ++ ++ #define CODEC_FSB \ ++ (SND_SOC_FSBD(1) | SND_SOC_FSBD(2) | SND_SOC_FSBD(4) | \ ++ SND_SOC_FSBD(8) | SND_SOC_FSBD(16)) ++ ++ #define CODEC_RATES \ ++ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_32000 |\ ++ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000) ++ ++ /* codec master @ 128, 192 & 256 FS */ ++ { ++ .fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = SNDRV_PCM_FORMAT_S16_LE, ++ .pcmrate = CODEC_RATES, ++ .pcmdir = SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 128, ++ .bfs = CODEC_FSB, ++ }, ++ ++ { ++ .fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = SNDRV_PCM_FORMAT_S16_LE, ++ .pcmrate = CODEC_RATES, ++ .pcmdir = SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 192, ++ .bfs = CODEC_FSB ++ }, ++ ++ { ++ .fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = SNDRV_PCM_FORMAT_S16_LE, ++ .pcmrate = CODEC_RATES, ++ .pcmdir = SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 256, ++ .bfs = CODEC_FSB, ++ }, ++ ++ /* codec slave */ ++ { ++ .fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS, ++ .pcmfmt = SNDRV_PCM_FORMAT_S16_LE, ++ .pcmrate = CODEC_RATES, ++ .pcmdir = SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE, ++ .fs = SND_SOC_FS_ALL, ++ .bfs = SND_SOC_FSB_ALL, ++ }, ++ ++ ++Example 6 ++--------- ++Codec that only runs at 8k, 44.1k, 48k @ different FS in master mode (for use ++with a fixed MCLK) and can generate a BCLK of MCLK / (1,2,4,8,16). ++Codec can also run in slave mode as and does not care about FS or BCLK (as long ++as there is enough bandwidth). Codec can support 16, 24 and 32 bit PCM sample ++sizes. ++ ++ #define CODEC_FSB \ ++ (SND_SOC_FSBD(1) | SND_SOC_FSBD(2) | SND_SOC_FSBD(4) | \ ++ SND_SOC_FSBD(8) | SND_SOC_FSBD(16)) ++ ++ #define CODEC_PCM_FORMATS \ ++ (SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S20_3LE | \ ++ SNDRV_PCM_FORMAT_S24_3LE | SNDRV_PCM_FORMAT_S24_LE | SNDRV_PCM_FORMAT_S32_LE) ++ ++ /* codec master */ ++ { ++ .fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = SNDRV_PCM_FORMAT_S16_LE, ++ .pcmrate = SNDRV_PCM_RATE_8000, ++ .pcmdir = SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 1536, ++ .bfs = CODEC_FSB, ++ }, ++ ++ { ++ .fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = SNDRV_PCM_FORMAT_S16_LE, ++ .pcmrate = SNDRV_PCM_RATE_44100, ++ .pcmdir = SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 272, ++ .bfs = CODEC_FSB, ++ }, ++ ++ { ++ .fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = SNDRV_PCM_FORMAT_S16_LE, ++ .pcmrate = SNDRV_PCM_RATE_48000, ++ .pcmdir = SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 256, ++ .bfs = CODEC_FSB, ++ }, ++ ++ /* codec slave */ ++ { ++ .fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS, ++ .pcmfmt = SNDRV_PCM_FORMAT_S16_LE, ++ .pcmrate = CODEC_RATES, ++ .pcmdir = SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE, ++ .fs = SND_SOC_FS_ALL, ++ .bfs = SND_SOC_FSB_ALL, ++ }, ++ ++ ++Example 7 ++--------- ++AC97 Codec that does not support VRA (i.e only runs at 48k). ++ ++ #define AC97_DIR \ ++ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE) ++ ++ #define AC97_PCM_FORMATS \ ++ (SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S18_3LE | \ ++ SNDRV_PCM_FORMAT_S20_3LE) ++ ++ /* AC97 with no VRA */ ++ { ++ .pcmfmt = AC97_PCM_FORMATS, ++ .pcmrate = SNDRV_PCM_RATE_48000, ++ } ++ ++ ++Example 8 ++--------- ++ ++CPU DAI that supports 8k - 48k @ 256FS and BCLK = MCLK / 4 in master mode. ++Slave mode (CPU DAI is FRAME master) supports 8k - 96k at any FS as long as ++BCLK = 64 * rate. (Intel XScale I2S controller). ++ ++ #define PXA_I2S_DAIFMT \ ++ (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF) ++ ++ #define PXA_I2S_DIR \ ++ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE) ++ ++ #define PXA_I2S_RATES \ ++ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ ++ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ ++ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) ++ ++ /* priv is divider */ ++ static struct snd_soc_dai_mode pxa2xx_i2s_modes[] = { ++ /* pxa2xx I2S frame and clock master modes */ ++ { ++ .fmt = PXA_I2S_DAIFMT | SND_SOC_DAIFMT_CBS_CFS, ++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, ++ .pcmrate = SNDRV_PCM_RATE_8000, ++ .pcmdir = PXA_I2S_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 256, ++ .bfs = SND_SOC_FSBD(4), ++ .priv = 0x48, ++ }, ++ { ++ .fmt = PXA_I2S_DAIFMT | SND_SOC_DAIFMT_CBS_CFS, ++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, ++ .pcmrate = SNDRV_PCM_RATE_11025, ++ .pcmdir = PXA_I2S_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 256, ++ .bfs = SND_SOC_FSBD(4), ++ .priv = 0x34, ++ }, ++ { ++ .fmt = PXA_I2S_DAIFMT | SND_SOC_DAIFMT_CBS_CFS, ++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, ++ .pcmrate = SNDRV_PCM_RATE_16000, ++ .pcmdir = PXA_I2S_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 256, ++ .bfs = SND_SOC_FSBD(4), ++ .priv = 0x24, ++ }, ++ { ++ .fmt = PXA_I2S_DAIFMT | SND_SOC_DAIFMT_CBS_CFS, ++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, ++ .pcmrate = SNDRV_PCM_RATE_22050, ++ .pcmdir = PXA_I2S_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 256, ++ .bfs = SND_SOC_FSBD(4), ++ .priv = 0x1a, ++ }, ++ { ++ .fmt = PXA_I2S_DAIFMT | SND_SOC_DAIFMT_CBS_CFS, ++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, ++ .pcmrate = SNDRV_PCM_RATE_44100, ++ .pcmdir = PXA_I2S_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 256, ++ .bfs = SND_SOC_FSBD(4), ++ .priv = 0xd, ++ }, ++ { ++ .fmt = PXA_I2S_DAIFMT | SND_SOC_DAIFMT_CBS_CFS, ++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, ++ .pcmrate = SNDRV_PCM_RATE_48000, ++ .pcmdir = PXA_I2S_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 256, ++ .bfs = SND_SOC_FSBD(4), ++ .priv = 0xc, ++ }, ++ ++ /* pxa2xx I2S frame master and clock slave mode */ ++ { ++ .fmt = PXA_I2S_DAIFMT | SND_SOC_DAIFMT_CBM_CFS, ++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, ++ .pcmrate = PXA_I2S_RATES, ++ .pcmdir = PXA_I2S_DIR, ++ .fs = SND_SOC_FS_ALL, ++ .flags = SND_SOC_DAI_BFS_RATE, ++ .bfs = 64, ++ .priv = 0x48, ++ }, ++}; +Index: linux-2.6-pxa-new/Documentation/sound/alsa/soc/clocking.txt +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/Documentation/sound/alsa/soc/clocking.txt +@@ -0,0 +1,314 @@ ++Audio Clocking ++============== ++ ++This text describes the audio clocking terms in ASoC and digital audio in ++general. Note: Audio clocking can be complex ! ++ ++ ++Master Clock ++------------ ++ ++Every audio subsystem is driven by a master clock (sometimes refered to as MCLK ++or SYSCLK). This audio master clock can be derived from a number of sources ++(e.g. crystal, PLL, CPU clock) and is responsible for producing the correct ++audio playback and capture sample rates. ++ ++Some master clocks (e.g. PLL's and CPU based clocks) are configuarble in that ++their speed can be altered by software (depending on the system use and to save ++power). Other master clocks are fixed at at set frequency (i.e. crystals). ++ ++ ++DAI Clocks ++---------- ++The Digital Audio Interface is usually driven by a Bit Clock (often referred to ++as BCLK). This clock is used to drive the digital audio data across the link ++between the codec and CPU. ++ ++The DAI also has a frame clock to signal the start of each audio frame. This ++clock is sometimes referred to as LRC (left right clock) or FRAME. This clock ++runs at exactly the sample rate (LRC = Rate). ++ ++Bit Clock can be generated as follows:- ++ ++BCLK = MCLK / x ++ ++ or ++ ++BCLK = LRC * x ++ ++ or ++ ++BCLK = LRC * Channels * Word Size ++ ++This relationship depends on the codec or SoC CPU in particular. ASoC can quite ++easily match BCLK generated by division (SND_SOC_DAI_BFS_DIV) with BCLK by ++multiplication (SND_SOC_DAI_BFS_RATE) or BCLK generated by ++Rate * Channels * Word size (RCW or SND_SOC_DAI_BFS_RCW). ++ ++ ++ASoC Clocking ++------------- ++ ++The ASoC core determines the clocking for each particular configuration at ++runtime. This is to allow for dynamic audio clocking wereby the audio clock is ++variable and depends on the system state or device usage scenario. i.e. a voice ++call requires slower clocks (and hence less power) than MP3 playback. ++ ++ASoC will call the config_sysclock() function for the target machine during the ++audio parameters configuration. The function is responsible for then clocking ++the machine audio subsytem and returning the audio clock speed to the core. ++This function should also call the codec and cpu DAI clock_config() functions ++to configure their respective internal clocking if required. ++ ++ ++ASoC Clocking Control Flow ++-------------------------- ++ ++The ASoC core will call the machine drivers config_sysclock() when most of the ++DAI capabilities are known. The machine driver is then responsible for calling ++the codec and/or CPU DAI drivers with the selected capabilities and the current ++MCLK. Note that the machine driver is also resonsible for setting the MCLK (and ++enabling it). ++ ++ (1) Match Codec and CPU DAI capabilities. At this point we have ++ matched the majority of the DAI fields and now need to make sure this ++ mode is currently clockable. ++ ++ (2) machine->config_sysclk() is now called with the matched DAI FS, sample ++ rate and BCLK master. This function then gets/sets the current audio ++ clock (depening on usage) and calls the codec and CPUI DAI drivers with ++ the FS, rate, BCLK master and MCLK. ++ ++ (3) Codec/CPU DAI config_sysclock(). This function checks that the FS, rate, ++ BCLK master and MCLK are acceptable for the codec or CPU DAI. It also ++ sets the DAI internal state to work with said clocks. ++ ++The config_sysclk() functions for CPU, codec and machine should return the MCLK ++on success and 0 on failure. ++ ++ ++Examples (b = BCLK, l = LRC) ++============================ ++ ++Example 1 ++--------- ++ ++Simple codec that only runs at 48k @ 256FS in master mode. ++ ++CPU only runs as slave DAI, however it generates a variable MCLK. ++ ++ -------- --------- ++ | | <----mclk--- | | ++ | Codec |b -----------> | CPU | ++ | |l -----------> | | ++ | | | | ++ -------- --------- ++ ++The codec driver has the following config_sysclock() ++ ++ static unsigned int config_sysclk(struct snd_soc_codec_dai *dai, ++ struct snd_soc_clock_info *info, unsigned int clk) ++ { ++ /* make sure clock is 256 * rate */ ++ if(info->rate << 8 == clk) { ++ dai->mclk = clk; ++ return clk; ++ } ++ ++ return 0; ++ } ++ ++The CPU I2S DAI driver has the following config_sysclk() ++ ++ static unsigned int config_sysclk(struct snd_soc_codec_dai *dai, ++ struct snd_soc_clock_info *info, unsigned int clk) ++ { ++ /* can we support this clk */ ++ if(set_audio_clk(clk) < 0) ++ return -EINVAL; ++ ++ dai->mclk = clk; ++ return dai->clk; ++ } ++ ++The machine driver config_sysclk() in this example is as follows:- ++ ++ unsigned int machine_config_sysclk(struct snd_soc_pcm_runtime *rtd, ++ struct snd_soc_clock_info *info) ++ { ++ int clk = info->rate * info->fs; ++ ++ /* check that CPU can deliver clock */ ++ if(rtd->cpu_dai->config_sysclk(rtd->cpu_dai, info, clk) < 0) ++ return -EINVAL; ++ ++ /* can codec work with this clock */ ++ return rtd->codec_dai->config_sysclk(rtd->codec_dai, info, clk); ++ } ++ ++ ++Example 2 ++--------- ++ ++Codec that can master at 8k and 48k at various FS (and hence supports a fixed ++set of input MCLK's) and can also be slave at various FS . ++ ++The CPU can master at 8k and 48k @256 FS and can be slave at any FS. ++ ++MCLK is a 12.288MHz crystal on this machine. ++ ++ -------- --------- ++ | | <---xtal---> | | ++ | Codec |b <----------> | CPU | ++ | |l <----------> | | ++ | | | | ++ -------- --------- ++ ++ ++The codec driver has the following config_sysclock() ++ ++ /* supported input clocks */ ++ const static int hifi_clks[] = {11289600, 12000000, 12288000, ++ 16934400, 18432000}; ++ ++ static unsigned int config_hsysclk(struct snd_soc_codec_dai *dai, ++ struct snd_soc_clock_info *info, unsigned int clk) ++ { ++ int i; ++ ++ /* is clk supported */ ++ for(i = 0; i < ARRAY_SIZE(hifi_clks); i++) { ++ if(clk == hifi_clks[i]) { ++ dai->mclk = clk; ++ return clk; ++ } ++ } ++ ++ /* this clk is not supported */ ++ return 0; ++ } ++ ++The CPU I2S DAI driver has the following config_sysclk() ++ ++ static unsigned int config_sysclk(struct snd_soc_codec_dai *dai, ++ struct snd_soc_clock_info *info, unsigned int clk) ++ { ++ /* are we master or slave */ ++ if (info->bclk_master & ++ (SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS)) { ++ ++ /* we can only master @ 256FS */ ++ if(info->rate << 8 == clk) { ++ dai->mclk = clk; ++ return dai->mclk; ++ } ++ } else { ++ /* slave we can run at any FS */ ++ dai->mclk = clk; ++ return dai->mclk; ++ } ++ ++ /* not supported */ ++ return dai->clk; ++ } ++ ++The machine driver config_sysclk() in this example is as follows:- ++ ++ unsigned int machine_config_sysclk(struct snd_soc_pcm_runtime *rtd, ++ struct snd_soc_clock_info *info) ++ { ++ int clk = 12288000; /* 12.288MHz */ ++ ++ /* who's driving the link */ ++ if (info->bclk_master & ++ (SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS)) { ++ /* codec master */ ++ ++ /* check that CPU can work with clock */ ++ if(rtd->cpu_dai->config_sysclk(rtd->cpu_dai, info, clk) < 0) ++ return -EINVAL; ++ ++ /* can codec work with this clock */ ++ return rtd->codec_dai->config_sysclk(rtd->codec_dai, info, clk); ++ } else { ++ /* cpu master */ ++ ++ /* check that codec can work with clock */ ++ if(rtd->codec_dai->config_sysclk(rtd->codec_dai, info, clk) < 0) ++ return -EINVAL; ++ ++ /* can CPU work with this clock */ ++ return rtd->cpu_dai->config_sysclk(rtd->cpu_dai, info, clk); ++ } ++ } ++ ++ ++ ++Example 3 ++--------- ++ ++Codec that masters at 8k ... 48k @256 FS. Codec can also be slave and ++doesn't care about FS. The codec has an internal PLL and dividers to generate ++the necessary internal clocks (for 256FS). ++ ++CPU can only be slave and doesn't care about FS. ++ ++MCLK is a non controllable 13MHz clock from the CPU. ++ ++ ++ -------- --------- ++ | | <----mclk--- | | ++ | Codec |b <----------> | CPU | ++ | |l <----------> | | ++ | | | | ++ -------- --------- ++ ++The codec driver has the following config_sysclock() ++ ++ /* valid PCM clock dividers * 2 */ ++ static int pcm_divs[] = {2, 6, 11, 4, 8, 12, 16}; ++ ++ static unsigned int config_vsysclk(struct snd_soc_codec_dai *dai, ++ struct snd_soc_clock_info *info, unsigned int clk) ++ { ++ int i, j, best_clk = info->fs * info->rate; ++ ++ /* can we run at this clk without the PLL ? */ ++ for (i = 0; i < ARRAY_SIZE(pcm_divs); i++) { ++ if ((best_clk >> 1) * pcm_divs[i] == clk) { ++ dai->pll_in = 0; ++ dai->clk_div = pcm_divs[i]; ++ dai->mclk = best_clk; ++ return dai->mclk; ++ } ++ } ++ ++ /* now check for PLL support */ ++ for (i = 0; i < ARRAY_SIZE(pll_div); i++) { ++ if (pll_div[i].pll_in == clk) { ++ for (j = 0; j < ARRAY_SIZE(pcm_divs); j++) { ++ if (pll_div[i].pll_out == pcm_divs[j] * (best_clk >> 1)) { ++ dai->pll_in = clk; ++ dai->pll_out = pll_div[i].pll_out; ++ dai->clk_div = pcm_divs[j]; ++ dai->mclk = best_clk; ++ return dai->mclk; ++ } ++ } ++ } ++ } ++ ++ /* this clk is not supported */ ++ return 0; ++ } ++ ++ ++The CPU I2S DAI driver has the does not need a config_sysclk() as it can slave ++at any FS. ++ ++ unsigned int config_sysclk(struct snd_soc_pcm_runtime *rtd, ++ struct snd_soc_clock_info *info) ++ { ++ /* codec has pll that generates mclk from 13MHz xtal */ ++ return rtd->codec_dai->config_sysclk(rtd->codec_dai, info, 13000000); ++ } +Index: linux-2.6-pxa-new/Documentation/sound/alsa/soc/codec.txt +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/Documentation/sound/alsa/soc/codec.txt +@@ -0,0 +1,232 @@ ++ASoC Codec Driver ++================= ++ ++The codec driver is generic and hardware independent code that configures the ++codec to provide audio capture and playback. It should contain no code that is ++specific to the target platform or machine. All platform and machine specific ++code should be added to the platform and machine drivers respectively. ++ ++Each codec driver must provide the following features:- ++ ++ 1) Digital audio interface (DAI) description ++ 2) Digital audio interface configuration ++ 3) PCM's description ++ 4) Codec control IO - using I2C, 3 Wire(SPI) or both API's ++ 5) Mixers and audio controls ++ 6) Sysclk configuration ++ 7) Codec audio operations ++ ++Optionally, codec drivers can also provide:- ++ ++ 8) DAPM description. ++ 9) DAPM event handler. ++10) DAC Digital mute control. ++ ++It's probably best to use this guide in conjuction with the existing codec ++driver code in sound/soc/codecs/ ++ ++ASoC Codec driver breakdown ++=========================== ++ ++1 - Digital Audio Interface (DAI) description ++--------------------------------------------- ++The DAI is a digital audio data transfer link between the codec and host SoC ++CPU. It typically has data transfer capabilities in both directions ++(playback and capture) and can run at a variety of different speeds. ++Supported interfaces currently include AC97, I2S and generic PCM style links. ++Please read DAI.txt for implementation information. ++ ++ ++2 - Digital Audio Interface (DAI) configuration ++----------------------------------------------- ++DAI configuration is handled by the codec_pcm_prepare function and is ++responsible for configuring and starting the DAI on the codec. This can be ++called multiple times and is atomic. It can access the runtime parameters. ++ ++This usually consists of a large function with numerous switch statements to ++set up each configuration option. These options are set by the core at runtime. ++ ++ ++3 - Codec PCM's ++--------------- ++Each codec must have it's PCM's defined. This defines the number of channels, ++stream names, callbacks and codec name. It is also used to register the DAI ++with the ASoC core. The PCM structure also associates the DAI capabilities with ++the ALSA PCM. ++ ++e.g. ++ ++static struct snd_soc_pcm_codec wm8731_pcm_client = { ++ .name = "WM8731", ++ .playback = { ++ .stream_name = "Playback", ++ .channels_min = 1, ++ .channels_max = 2, ++ }, ++ .capture = { ++ .stream_name = "Capture", ++ .channels_min = 1, ++ .channels_max = 2, ++ }, ++ .config_sysclk = wm8731_config_sysclk, ++ .ops = { ++ .prepare = wm8731_pcm_prepare, ++ }, ++ .caps = { ++ .num_modes = ARRAY_SIZE(wm8731_hwfmt), ++ .modes = &wm8731_hwfmt[0], ++ }, ++}; ++ ++ ++4 - Codec control IO ++-------------------- ++The codec can ususally be controlled via an I2C or SPI style interface (AC97 ++combines control with data in the DAI). The codec drivers will have to provide ++functions to read and write the codec registers along with supplying a register ++cache:- ++ ++ /* IO control data and register cache */ ++ void *control_data; /* codec control (i2c/3wire) data */ ++ void *reg_cache; ++ ++Codec read/write should do any data formatting and call the hardware read write ++below to perform the IO. These functions are called by the core and alsa when ++performing DAPM or changing the mixer:- ++ ++ unsigned int (*read)(struct snd_soc_codec *, unsigned int); ++ int (*write)(struct snd_soc_codec *, unsigned int, unsigned int); ++ ++Codec hardware IO functions - usually points to either the I2C, SPI or AC97 ++read/write:- ++ ++ hw_write_t hw_write; ++ hw_read_t hw_read; ++ ++ ++5 - Mixers and audio controls ++----------------------------- ++All the codec mixers and audio controls can be defined using the convenience ++macros defined in soc.h. ++ ++ #define SOC_SINGLE(xname, reg, shift, mask, invert) ++ ++Defines a single control as follows:- ++ ++ xname = Control name e.g. "Playback Volume" ++ reg = codec register ++ shift = control bit(s) offset in register ++ mask = control bit size(s) e.g. mask of 7 = 3 bits ++ invert = the control is inverted ++ ++Other macros include:- ++ ++ #define SOC_DOUBLE(xname, reg, shift_left, shift_right, mask, invert) ++ ++A stereo control ++ ++ #define SOC_DOUBLE_R(xname, reg_left, reg_right, shift, mask, invert) ++ ++A stereo control spanning 2 registers ++ ++ #define SOC_ENUM_SINGLE(xreg, xshift, xmask, xtexts) ++ ++Defines an single enumerated control as follows:- ++ ++ xreg = register ++ xshift = control bit(s) offset in register ++ xmask = control bit(s) size ++ xtexts = pointer to array of strings that describe each setting ++ ++ #define SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, xtexts) ++ ++Defines a stereo enumerated control ++ ++ ++6 - System clock configuration. ++------------------------------- ++The system clock that drives the audio subsystem can change depending on sample ++rate and the system power state. i.e. ++ ++o Higher sample rates sometimes need a higher system clock. ++o Low system power states can sometimes limit the available clocks. ++ ++This function is a callback that the machine driver can call to set and ++determine if the clock and sample rate combination is supported by the codec at ++the present time (and system state). ++ ++NOTE: If the codec has a PLL then it has a lot more flexability wrt clock and ++sample rate combinations. ++ ++Your config_sysclock function should return the MCLK if it's a valid ++combination for your codec else 0; ++ ++Please read clocking.txt now. ++ ++ ++7 - Codec Audio Operations ++-------------------------- ++The codec driver also supports the following alsa operations:- ++ ++/* SoC audio ops */ ++struct snd_soc_ops { ++ int (*startup)(snd_pcm_substream_t *); ++ void (*shutdown)(snd_pcm_substream_t *); ++ int (*hw_params)(snd_pcm_substream_t *, snd_pcm_hw_params_t *); ++ int (*hw_free)(snd_pcm_substream_t *); ++ int (*prepare)(snd_pcm_substream_t *); ++}; ++ ++Please refer to the alsa driver PCM documentation for details. ++http://www.alsa-project.org/~iwai/writing-an-alsa-driver/c436.htm ++ ++ ++8 - DAPM description. ++--------------------- ++The Dynamic Audio Power Management description describes the codec's power ++components, their relationships and registers to the ASoC core. Please read ++dapm.txt for details of building the description. ++ ++Please also see the examples in other codec drivers. ++ ++ ++9 - DAPM event handler ++---------------------- ++This function is a callback that handles codec domain PM calls and system ++domain PM calls (e.g. suspend and resume). It's used to put the codec to sleep ++when not in use. ++ ++Power states:- ++ ++ SNDRV_CTL_POWER_D0: /* full On */ ++ /* vref/mid, clk and osc on, active */ ++ ++ SNDRV_CTL_POWER_D1: /* partial On */ ++ SNDRV_CTL_POWER_D2: /* partial On */ ++ ++ SNDRV_CTL_POWER_D3hot: /* Off, with power */ ++ /* everything off except vref/vmid, inactive */ ++ ++ SNDRV_CTL_POWER_D3cold: /* Everything Off, without power */ ++ ++ ++10 - Codec DAC digital mute control. ++------------------------------------ ++Most codecs have a digital mute before the DAC's that can be used to minimise ++any system noise. The mute stops any digital data from entering the DAC. ++ ++A callback can be created that is called by the core for each codec DAI when the ++mute is applied or freed. ++ ++i.e. ++ ++static int wm8974_mute(struct snd_soc_codec *codec, ++ struct snd_soc_codec_dai *dai, int mute) ++{ ++ u16 mute_reg = wm8974_read_reg_cache(codec, WM8974_DAC) & 0xffbf; ++ if(mute) ++ wm8974_write(codec, WM8974_DAC, mute_reg | 0x40); ++ else ++ wm8974_write(codec, WM8974_DAC, mute_reg); ++ return 0; ++} +Index: linux-2.6-pxa-new/Documentation/sound/alsa/soc/dapm.txt +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/Documentation/sound/alsa/soc/dapm.txt +@@ -0,0 +1,297 @@ ++Dynamic Audio Power Management for Portable Devices ++=================================================== ++ ++1. Description ++============== ++ ++Dynamic Audio Power Management (DAPM) is designed to allow portable Linux devices ++to use the minimum amount of power within the audio subsystem at all times. It ++is independent of other kernel PM and as such, can easily co-exist with the ++other PM systems. ++ ++DAPM is also completely transparent to all user space applications as all power ++switching is done within the ASoC core. No code changes or recompiling are ++required for user space applications. DAPM makes power switching descisions based ++upon any audio stream (capture/playback) activity and audio mixer settings ++within the device. ++ ++DAPM spans the whole machine. It covers power control within the entire audio ++subsystem, this includes internal codec power blocks and machine level power ++systems. ++ ++There are 4 power domains within DAPM ++ ++ 1. Codec domain - VREF, VMID (core codec and audio power) ++ Usually controlled at codec probe/remove and suspend/resume, although ++ can be set at stream time if power is not needed for sidetone, etc. ++ ++ 2. Platform/Machine domain - physically connected inputs and outputs ++ Is platform/machine and user action specific, is configured by the ++ machine driver and responds to asynchronous events e.g when HP ++ are inserted ++ ++ 3. Path domain - audio susbsystem signal paths ++ Automatically set when mixer and mux settings are changed by the user. ++ e.g. alsamixer, amixer. ++ ++ 4. Stream domain - DAC's and ADC's. ++ Enabled and disabled when stream playback/capture is started and ++ stopped respectively. e.g. aplay, arecord. ++ ++All DAPM power switching descisons are made automatically by consulting an audio ++routing map of the whole machine. This map is specific to each machine and ++consists of the interconnections between every audio component (including ++internal codec components). All audio components that effect power are called ++widgets hereafter. ++ ++ ++2. DAPM Widgets ++=============== ++ ++Audio DAPM widgets fall into a number of types:- ++ ++ o Mixer - Mixes several analog signals into a single analog signal. ++ o Mux - An analog switch that outputs only 1 of it's inputs. ++ o PGA - A programmable gain amplifier or attenuation widget. ++ o ADC - Analog to Digital Converter ++ o DAC - Digital to Analog Converter ++ o Switch - An analog switch ++ o Input - A codec input pin ++ o Output - A codec output pin ++ o Headphone - Headphone (and optional Jack) ++ o Mic - Mic (and optional Jack) ++ o Line - Line Input/Output (and optional Jack) ++ o Speaker - Speaker ++ o Pre - Special PRE widget (exec before all others) ++ o Post - Special POST widget (exec after all others) ++ ++(Widgets are defined in include/sound/soc-dapm.h) ++ ++Widgets are usually added in the codec driver and the machine driver. There are ++convience macros defined in soc-dapm.h that can be used to quickly build a ++list of widgets of the codecs and machines DAPM widgets. ++ ++Most widgets have a name, register, shift and invert. Some widgets have extra ++parameters for stream name and kcontrols. ++ ++ ++2.1 Stream Domain Widgets ++------------------------- ++ ++Stream Widgets relate to the stream power domain and only consist of ADC's ++(analog to digital converters) and DAC's (digital to analog converters). ++ ++Stream widgets have the following format:- ++ ++SND_SOC_DAPM_DAC(name, stream name, reg, shift, invert), ++ ++NOTE: the stream name must match the corresponding stream name in your codecs ++snd_soc_codec_dai. ++ ++e.g. stream widgets for HiFi playback and capture ++ ++SND_SOC_DAPM_DAC("HiFi DAC", "HiFi Playback", REG, 3, 1), ++SND_SOC_DAPM_ADC("HiFi ADC", "HiFi Capture", REG, 2, 1), ++ ++ ++2.2 Path Domain Widgets ++----------------------- ++ ++Path domain widgets have a ability to control or effect the audio signal or ++audio paths within the audio subsystem. They have the following form:- ++ ++SND_SOC_DAPM_PGA(name, reg, shift, invert, controls, num_controls) ++ ++Any widget kcontrols can be set using the controls and num_controls members. ++ ++e.g. Mixer widget (the kcontrols are declared first) ++ ++/* Output Mixer */ ++static const snd_kcontrol_new_t wm8731_output_mixer_controls[] = { ++SOC_DAPM_SINGLE("Line Bypass Switch", WM8731_APANA, 3, 1, 0), ++SOC_DAPM_SINGLE("Mic Sidetone Switch", WM8731_APANA, 5, 1, 0), ++SOC_DAPM_SINGLE("HiFi Playback Switch", WM8731_APANA, 4, 1, 0), ++}; ++ ++SND_SOC_DAPM_MIXER("Output Mixer", WM8731_PWR, 4, 1, wm8731_output_mixer_controls, ++ ARRAY_SIZE(wm8731_output_mixer_controls)), ++ ++ ++2.3 Platform/Machine domain Widgets ++----------------------------------- ++ ++Machine widgets are different from codec widgets in that they don't have a ++codec register bit associated with them. A machine widget is assigned to each ++machine audio component (non codec) that can be independently powered. e.g. ++ ++ o Speaker Amp ++ o Microphone Bias ++ o Jack connectors ++ ++A machine widget can have an optional call back. ++ ++e.g. Jack connector widget for an external Mic that enables Mic Bias ++when the Mic is inserted:- ++ ++static int spitz_mic_bias(struct snd_soc_dapm_widget* w, int event) ++{ ++ if(SND_SOC_DAPM_EVENT_ON(event)) ++ set_scoop_gpio(&spitzscoop2_device.dev, SPITZ_SCP2_MIC_BIAS); ++ else ++ reset_scoop_gpio(&spitzscoop2_device.dev, SPITZ_SCP2_MIC_BIAS); ++ ++ return 0; ++} ++ ++SND_SOC_DAPM_MIC("Mic Jack", spitz_mic_bias), ++ ++ ++2.4 Codec Domain ++---------------- ++ ++The Codec power domain has no widgets and is handled by the codecs DAPM event ++handler. This handler is called when the codec powerstate is changed wrt to any ++stream event or by kernel PM events. ++ ++ ++2.5 Virtual Widgets ++------------------- ++ ++Sometimes widgets exist in the codec or machine audio map that don't have any ++corresponding register bit for power control. In this case it's necessary to ++create a virtual widget - a widget with no control bits e.g. ++ ++SND_SOC_DAPM_MIXER("AC97 Mixer", SND_SOC_DAPM_NOPM, 0, 0, NULL, 0), ++ ++This can be used to merge to signal paths together in software. ++ ++After all the widgets have been defined, they can then be added to the DAPM ++subsystem individually with a call to snd_soc_dapm_new_control(). ++ ++ ++3. Codec Widget Interconnections ++================================ ++ ++Widgets are connected to each other within the codec and machine by audio ++paths (called interconnections). Each interconnection must be defined in order ++to create a map of all audio paths between widgets. ++This is easiest with a diagram of the codec (and schematic of the machine audio ++system), as it requires joining widgets together via their audio signal paths. ++ ++i.e. from the WM8731 codec's output mixer (wm8731.c) ++ ++The WM8731 output mixer has 3 inputs (sources) ++ ++ 1. Line Bypass Input ++ 2. DAC (HiFi playback) ++ 3. Mic Sidetone Input ++ ++Each input in this example has a kcontrol associated with it (defined in example ++above) and is connected to the output mixer via it's kcontrol name. We can now ++connect the destination widget (wrt audio signal) with it's source widgets. ++ ++ /* output mixer */ ++ {"Output Mixer", "Line Bypass Switch", "Line Input"}, ++ {"Output Mixer", "HiFi Playback Switch", "DAC"}, ++ {"Output Mixer", "Mic Sidetone Switch", "Mic Bias"}, ++ ++So we have :- ++ ++ Destination Widget <=== Path Name <=== Source Widget ++ ++Or:- ++ ++ Sink, Path, Source ++ ++Or :- ++ ++ "Output Mixer" is connected to the "DAC" via the "HiFi Playback Switch". ++ ++When there is no path name connecting widgets (e.g. a direct connection) we ++pass NULL for the path name. ++ ++Interconnections are created with a call to:- ++ ++snd_soc_dapm_connect_input(codec, sink, path, source); ++ ++Finally, snd_soc_dapm_new_widgets(codec) must be called after all widgets and ++interconnections have been registered with the core. This causes the core to ++scan the codec and machine so that the internal DAPM state matches the ++physical state of the machine. ++ ++ ++3.1 Machine Widget Interconnections ++----------------------------------- ++Machine widget interconnections are created in the same way as codec ones and ++directly connect the codec pins to machine level widgets. ++ ++e.g. connects the speaker out codec pins to the internal speaker. ++ ++ /* ext speaker connected to codec pins LOUT2, ROUT2 */ ++ {"Ext Spk", NULL , "ROUT2"}, ++ {"Ext Spk", NULL , "LOUT2"}, ++ ++This allows the DAPM to power on and off pins that are connected (and in use) ++and pins that are NC respectively. ++ ++ ++4 Endpoint Widgets ++=================== ++An endpoint is a start or end point (widget) of an audio signal within the ++machine and includes the codec. e.g. ++ ++ o Headphone Jack ++ o Internal Speaker ++ o Internal Mic ++ o Mic Jack ++ o Codec Pins ++ ++When a codec pin is NC it can be marked as not used with a call to ++ ++snd_soc_dapm_set_endpoint(codec, "Widget Name", 0); ++ ++The last argument is 0 for inactive and 1 for active. This way the pin and its ++input widget will never be powered up and consume power. ++ ++This also applies to machine widgets. e.g. if a headphone is connected to a ++jack then the jack can be marked active. If the headphone is removed, then ++the headphone jack can be marked inactive. ++ ++ ++5 DAPM Widget Events ++==================== ++ ++Some widgets can register their interest with the DAPM core in PM events. ++e.g. A Speaker with an amplifier registers a widget so the amplifier can be ++powered only when the spk is in use. ++ ++/* turn speaker amplifier on/off depending on use */ ++static int corgi_amp_event(struct snd_soc_dapm_widget *w, int event) ++{ ++ if (SND_SOC_DAPM_EVENT_ON(event)) ++ set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_APM_ON); ++ else ++ reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_APM_ON); ++ ++ return 0; ++} ++ ++/* corgi machine dapm widgets */ ++static const struct snd_soc_dapm_widget wm8731_dapm_widgets = ++ SND_SOC_DAPM_SPK("Ext Spk", corgi_amp_event); ++ ++Please see soc-dapm.h for all other widgets that support events. ++ ++ ++5.1 Event types ++--------------- ++ ++The following event types are supported by event widgets. ++ ++/* dapm event types */ ++#define SND_SOC_DAPM_PRE_PMU 0x1 /* before widget power up */ ++#define SND_SOC_DAPM_POST_PMU 0x2 /* after widget power up */ ++#define SND_SOC_DAPM_PRE_PMD 0x4 /* before widget power down */ ++#define SND_SOC_DAPM_POST_PMD 0x8 /* after widget power down */ ++#define SND_SOC_DAPM_PRE_REG 0x10 /* before audio path setup */ ++#define SND_SOC_DAPM_POST_REG 0x20 /* after audio path setup */ +Index: linux-2.6-pxa-new/Documentation/sound/alsa/soc/machine.txt +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/Documentation/sound/alsa/soc/machine.txt +@@ -0,0 +1,114 @@ ++ASoC Machine Driver ++=================== ++ ++The ASoC machine (or board) driver is the code that glues together the platform ++and codec drivers. ++ ++The machine driver can contain codec and platform specific code. It registers ++the audio subsystem with the kernel as a platform device and is represented by ++the following struct:- ++ ++/* SoC machine */ ++struct snd_soc_machine { ++ char *name; ++ ++ int (*probe)(struct platform_device *pdev); ++ int (*remove)(struct platform_device *pdev); ++ ++ /* the pre and post PM functions are used to do any PM work before and ++ * after the codec and DAI's do any PM work. */ ++ int (*suspend_pre)(struct platform_device *pdev, pm_message_t state); ++ int (*suspend_post)(struct platform_device *pdev, pm_message_t state); ++ int (*resume_pre)(struct platform_device *pdev); ++ int (*resume_post)(struct platform_device *pdev); ++ ++ /* machine stream operations */ ++ struct snd_soc_ops *ops; ++ ++ /* CPU <--> Codec DAI links */ ++ struct snd_soc_dai_link *dai_link; ++ int num_links; ++}; ++ ++probe()/remove() ++---------------- ++probe/remove are optional. Do any machine specific probe here. ++ ++ ++suspend()/resume() ++------------------ ++The machine driver has pre and post versions of suspend and resume to take care ++of any machine audio tasks that have to be done before or after the codec, DAI's ++and DMA is suspended and resumed. Optional. ++ ++ ++Machine operations ++------------------ ++The machine specific audio operations can be set here. Again this is optional. ++ ++ ++Machine DAI Configuration ++------------------------- ++The machine DAI configuration glues all the codec and CPU DAI's together. It can ++also be used to set up the DAI system clock and for any machine related DAI ++initialisation e.g. the machine audio map can be connected to the codec audio ++map, unconnnected codec pins can be set as such. Please see corgi.c, spitz.c ++for examples. ++ ++struct snd_soc_dai_link is used to set up each DAI in your machine. e.g. ++ ++/* corgi digital audio interface glue - connects codec <--> CPU */ ++static struct snd_soc_dai_link corgi_dai = { ++ .name = "WM8731", ++ .stream_name = "WM8731", ++ .cpu_dai = &pxa_i2s_dai, ++ .codec_dai = &wm8731_dai, ++ .init = corgi_wm8731_init, ++ .config_sysclk = corgi_config_sysclk, ++}; ++ ++struct snd_soc_machine then sets up the machine with it's DAI's. e.g. ++ ++/* corgi audio machine driver */ ++static struct snd_soc_machine snd_soc_machine_corgi = { ++ .name = "Corgi", ++ .dai_link = &corgi_dai, ++ .num_links = 1, ++ .ops = &corgi_ops, ++}; ++ ++ ++Machine Audio Subsystem ++----------------------- ++ ++The machine soc device glues the platform, machine and codec driver together. ++Private data can also be set here. e.g. ++ ++/* corgi audio private data */ ++static struct wm8731_setup_data corgi_wm8731_setup = { ++ .i2c_address = 0x1b, ++}; ++ ++/* corgi audio subsystem */ ++static struct snd_soc_device corgi_snd_devdata = { ++ .machine = &snd_soc_machine_corgi, ++ .platform = &pxa2xx_soc_platform, ++ .codec_dev = &soc_codec_dev_wm8731, ++ .codec_data = &corgi_wm8731_setup, ++}; ++ ++ ++Machine Power Map ++----------------- ++ ++The machine driver can optionally extend the codec power map and to become an ++audio power map of the audio subsystem. This allows for automatic power up/down ++of speaker/HP amplifiers, etc. Codec pins can be connected to the machines jack ++sockets in the machine init function. See soc/pxa/spitz.c and dapm.txt for ++details. ++ ++ ++Machine Controls ++---------------- ++ ++Machine specific audio mixer controls can be added in the dai init function. +\ No newline at end of file +Index: linux-2.6-pxa-new/Documentation/sound/alsa/soc/overview.txt +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/Documentation/sound/alsa/soc/overview.txt +@@ -0,0 +1,83 @@ ++ALSA SoC Layer ++============== ++ ++The overall project goal of the ALSA System on Chip (ASoC) layer is to provide ++better ALSA support for embedded system on chip procesors (e.g. pxa2xx, au1x00, ++iMX, etc) and portable audio codecs. Currently there is some support in the ++kernel for SoC audio, however it has some limitations:- ++ ++ * Currently, codec drivers are often tightly coupled to the underlying SoC ++ cpu. This is not ideal and leads to code duplication i.e. Linux now has 4 ++ different wm8731 drivers for 4 different SoC platforms. ++ ++ * There is no standard method to signal user initiated audio events. ++ e.g. Headphone/Mic insertion, Headphone/Mic detection after an insertion ++ event. These are quite common events on portable devices and ofter require ++ machine specific code to re route audio, enable amps etc after such an event. ++ ++ * Current drivers tend to power up the entire codec when playing ++ (or recording) audio. This is fine for a PC, but tends to waste a lot of ++ power on portable devices. There is also no support for saving power via ++ changing codec oversampling rates, bias currents, etc. ++ ++ ++ASoC Design ++=========== ++ ++The ASoC layer is designed to address these issues and provide the following ++features :- ++ ++ * Codec independence. Allows reuse of codec drivers on other platforms ++ and machines. ++ ++ * Easy I2S/PCM audio interface setup between codec and SoC. Each SoC interface ++ and codec registers it's audio interface capabilities with the core and are ++ subsequently matched and configured when the application hw params are known. ++ ++ * Dynamic Audio Power Management (DAPM). DAPM automatically sets the codec to ++ it's minimum power state at all times. This includes powering up/down ++ internal power blocks depending on the internal codec audio routing and any ++ active streams. ++ ++ * Pop and click reduction. Pops and clicks can be reduced by powering the ++ codec up/down in the correct sequence (including using digital mute). ASoC ++ signals the codec when to change power states. ++ ++ * Machine specific controls: Allow machines to add controls to the sound card ++ e.g. volume control for speaker amp. ++ ++To achieve all this, ASoC basically splits an embedded audio system into 3 ++components :- ++ ++ * Codec driver: The codec driver is platform independent and contains audio ++ controls, audio interface capabilities, codec dapm definition and codec IO ++ functions. ++ ++ * Platform driver: The platform driver contains the audio dma engine and audio ++ interface drivers (e.g. I2S, AC97, PCM) for that platform. ++ ++ * Machine driver: The machine driver handles any machine specific controls and ++ audio events. i.e. turing on an amp at start of playback. ++ ++ ++Documentation ++============= ++ ++The documentation is spilt into the following sections:- ++ ++overview.txt: This file. ++ ++codec.txt: Codec driver internals. ++ ++DAI.txt: Description of Digital Audio Interface standards and how to configure ++a DAI within your codec and CPU DAI drivers. ++ ++dapm.txt: Dynamic Audio Power Management ++ ++platform.txt: Platform audio DMA and DAI. ++ ++machine.txt: Machine driver internals. ++ ++pop_clicks.txt: How to minimise audio artifacts. ++ ++clocking.txt: ASoC clocking for best power performance. +\ No newline at end of file +Index: linux-2.6-pxa-new/Documentation/sound/alsa/soc/platform.txt +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/Documentation/sound/alsa/soc/platform.txt +@@ -0,0 +1,58 @@ ++ASoC Platform Driver ++==================== ++ ++An ASoC platform driver can be divided into audio DMA and SoC DAI configuration ++and control. The platform drivers only target the SoC CPU and must have no board ++specific code. ++ ++Audio DMA ++========= ++ ++The platform DMA driver optionally supports the following alsa operations:- ++ ++/* SoC audio ops */ ++struct snd_soc_ops { ++ int (*startup)(snd_pcm_substream_t *); ++ void (*shutdown)(snd_pcm_substream_t *); ++ int (*hw_params)(snd_pcm_substream_t *, snd_pcm_hw_params_t *); ++ int (*hw_free)(snd_pcm_substream_t *); ++ int (*prepare)(snd_pcm_substream_t *); ++ int (*trigger)(snd_pcm_substream_t *, int); ++}; ++ ++The platform driver exports it's DMA functionailty via struct snd_soc_platform:- ++ ++struct snd_soc_platform { ++ char *name; ++ ++ int (*probe)(struct platform_device *pdev); ++ int (*remove)(struct platform_device *pdev); ++ int (*suspend)(struct platform_device *pdev, struct snd_soc_cpu_dai *cpu_dai); ++ int (*resume)(struct platform_device *pdev, struct snd_soc_cpu_dai *cpu_dai); ++ ++ /* pcm creation and destruction */ ++ int (*pcm_new)(snd_card_t *, struct snd_soc_codec_dai *, snd_pcm_t *); ++ void (*pcm_free)(snd_pcm_t *); ++ ++ /* platform stream ops */ ++ snd_pcm_ops_t *pcm_ops; ++}; ++ ++Please refer to the alsa driver documentation for details of audio DMA. ++http://www.alsa-project.org/~iwai/writing-an-alsa-driver/c436.htm ++ ++An example DMA driver is soc/pxa/pxa2xx-pcm.c ++ ++ ++SoC DAI Drivers ++=============== ++ ++Each SoC DAI driver must provide the following features:- ++ ++ 1) Digital audio interface (DAI) description ++ 2) Digital audio interface configuration ++ 3) PCM's description ++ 4) Sysclk configuration ++ 5) Suspend and resume (optional) ++ ++Please see codec.txt for a description of items 1 - 4. +Index: linux-2.6-pxa-new/Documentation/sound/alsa/soc/pops_clicks.txt +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/Documentation/sound/alsa/soc/pops_clicks.txt +@@ -0,0 +1,52 @@ ++Audio Pops and Clicks ++===================== ++ ++Pops and clicks are unwanted audio artifacts caused by the powering up and down ++of components within the audio subsystem. This is noticable on PC's when an audio ++module is either loaded or unloaded (at module load time the sound card is ++powered up and causes a popping noise on the speakers). ++ ++Pops and clicks can be more frequent on portable systems with DAPM. This is because ++the components within the subsystem are being dynamically powered depending on ++the audio usage and this can subsequently cause a small pop or click every time a ++component power state is changed. ++ ++ ++Minimising Playback Pops and Clicks ++=================================== ++ ++Playback pops in portable audio subsystems cannot be completely eliminated atm, ++however future audio codec hardware will have better pop and click supression. ++Pops can be reduced within playback by powering the audio components in a ++specific order. This order is different for startup and shutdown and follows ++some basic rules:- ++ ++ Startup Order :- DAC --> Mixers --> Output PGA --> Digital Unmute ++ ++ Shutdown Order :- Digital Mute --> Output PGA --> Mixers --> DAC ++ ++This assumes that the codec PCM output path from the DAC is via a mixer and then ++a PGA (programmable gain amplifier) before being output to the speakers. ++ ++ ++Minimising Capture Pops and Clicks ++================================== ++ ++Capture artifacts are somewhat easier to get rid as we can delay activating the ++ADC until all the pops have occured. This follows similar power rules to ++playback in that components are powered in a sequence depending upon stream ++startup or shutdown. ++ ++ Startup Order - Input PGA --> Mixers --> ADC ++ ++ Shutdown Order - ADC --> Mixers --> Input PGA ++ ++ ++Zipper Noise ++============ ++An unwanted zipper noise can occur within the audio playback or capture stream ++when a volume control is changed near its maximum gain value. The zipper noise ++is heard when the gain increase or decrease changes the mean audio signal ++amplitude too quickly. It can be minimised by enabling the zero cross setting ++for each volume control. The ZC forces the gain change to occur when the signal ++crosses the zero amplitude line. +Index: linux-2.6-pxa-new/include/sound/ac97_codec.h +=================================================================== +--- linux-2.6-pxa-new.orig/include/sound/ac97_codec.h ++++ linux-2.6-pxa-new/include/sound/ac97_codec.h +@@ -425,6 +425,7 @@ struct snd_ac97_build_ops { + + struct snd_ac97_bus_ops { + void (*reset) (struct snd_ac97 *ac97); ++ void (*warm_reset)(struct snd_ac97 *ac97); + void (*write) (struct snd_ac97 *ac97, unsigned short reg, unsigned short val); + unsigned short (*read) (struct snd_ac97 *ac97, unsigned short reg); + void (*wait) (struct snd_ac97 *ac97); +Index: linux-2.6-pxa-new/include/sound/soc-dapm.h +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/include/sound/soc-dapm.h +@@ -0,0 +1,286 @@ ++/* ++ * linux/sound/soc-dapm.h -- ALSA SoC Dynamic Audio Power Management ++ * ++ * Author: Liam Girdwood ++ * Created: Aug 11th 2005 ++ * Copyright: Wolfson Microelectronics. PLC. ++ * ++ * This program is free software; you can redistribute it and/or modify ++ * it under the terms of the GNU General Public License version 2 as ++ * published by the Free Software Foundation. ++ */ ++ ++#ifndef __LINUX_SND_SOC_DAPM_H ++#define __LINUX_SND_SOC_DAPM_H ++ ++#include <linux/device.h> ++#include <linux/types.h> ++#include <sound/control.h> ++#include <sound/soc.h> ++ ++/* widget has no PM register bit */ ++#define SND_SOC_NOPM -1 ++ ++/* ++ * SoC dynamic audio power managment ++ * ++ * We can have upto 4 power domains ++ * 1. Codec domain - VREF, VMID ++ * Usually controlled at codec probe/remove, although can be set ++ * at stream time if power is not needed for sidetone, etc. ++ * 2. Platform/Machine domain - physically connected inputs and outputs ++ * Is platform/machine and user action specific, is set in the machine ++ * driver and by userspace e.g when HP are inserted ++ * 3. Path domain - Internal codec path mixers ++ * Are automatically set when mixer and mux settings are ++ * changed by the user. ++ * 4. Stream domain - DAC's and ADC's. ++ * Enabled when stream playback/capture is started. ++ */ ++ ++/* codec domain */ ++#define SND_SOC_DAPM_VMID(wname) \ ++{ .id = snd_soc_dapm_vmid, .name = wname, .kcontrols = NULL, \ ++ .num_kcontrols = 0} ++ ++/* platform domain */ ++#define SND_SOC_DAPM_INPUT(wname) \ ++{ .id = snd_soc_dapm_input, .name = wname, .kcontrols = NULL, \ ++ .num_kcontrols = 0} ++#define SND_SOC_DAPM_OUTPUT(wname) \ ++{ .id = snd_soc_dapm_output, .name = wname, .kcontrols = NULL, \ ++ .num_kcontrols = 0} ++#define SND_SOC_DAPM_MIC(wname, wevent) \ ++{ .id = snd_soc_dapm_mic, .name = wname, .kcontrols = NULL, \ ++ .num_kcontrols = 0, .event = wevent, \ ++ .event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD} ++#define SND_SOC_DAPM_HP(wname, wevent) \ ++{ .id = snd_soc_dapm_hp, .name = wname, .kcontrols = NULL, \ ++ .num_kcontrols = 0, .event = wevent, \ ++ .event_flags = SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD} ++#define SND_SOC_DAPM_SPK(wname, wevent) \ ++{ .id = snd_soc_dapm_spk, .name = wname, .kcontrols = NULL, \ ++ .num_kcontrols = 0, .event = wevent, \ ++ .event_flags = SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD} ++#define SND_SOC_DAPM_LINE(wname, wevent) \ ++{ .id = snd_soc_dapm_line, .name = wname, .kcontrols = NULL, \ ++ .num_kcontrols = 0, .event = wevent, \ ++ .event_flags = SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD} ++ ++/* path domain */ ++#define SND_SOC_DAPM_PGA(wname, wreg, wshift, winvert,\ ++ wcontrols, wncontrols) \ ++{ .id = snd_soc_dapm_pga, .name = wname, .reg = wreg, .shift = wshift, \ ++ .invert = winvert, .kcontrols = wcontrols, .num_kcontrols = wncontrols} ++#define SND_SOC_DAPM_MIXER(wname, wreg, wshift, winvert, \ ++ wcontrols, wncontrols)\ ++{ .id = snd_soc_dapm_mixer, .name = wname, .reg = wreg, .shift = wshift, \ ++ .invert = winvert, .kcontrols = wcontrols, .num_kcontrols = wncontrols} ++#define SND_SOC_DAPM_MICBIAS(wname, wreg, wshift, winvert) \ ++{ .id = snd_soc_dapm_micbias, .name = wname, .reg = wreg, .shift = wshift, \ ++ .invert = winvert, .kcontrols = NULL, .num_kcontrols = 0} ++#define SND_SOC_DAPM_SWITCH(wname, wreg, wshift, winvert, wcontrols) \ ++{ .id = snd_soc_dapm_switch, .name = wname, .reg = wreg, .shift = wshift, \ ++ .invert = winvert, .kcontrols = wcontrols, .num_kcontrols = 1} ++#define SND_SOC_DAPM_MUX(wname, wreg, wshift, winvert, wcontrols) \ ++{ .id = snd_soc_dapm_mux, .name = wname, .reg = wreg, .shift = wshift, \ ++ .invert = winvert, .kcontrols = wcontrols, .num_kcontrols = 1} ++ ++/* path domain with event - event handler must return 0 for success */ ++#define SND_SOC_DAPM_PGA_E(wname, wreg, wshift, winvert, wcontrols, \ ++ wncontrols, wevent, wflags) \ ++{ .id = snd_soc_dapm_pga, .name = wname, .reg = wreg, .shift = wshift, \ ++ .invert = winvert, .kcontrols = wcontrols, .num_kcontrols = wncontrols, \ ++ .event = wevent, .event_flags = wflags} ++#define SND_SOC_DAPM_MIXER_E(wname, wreg, wshift, winvert, wcontrols, \ ++ wncontrols, wevent, wflags) \ ++{ .id = snd_soc_dapm_mixer, .name = wname, .reg = wreg, .shift = wshift, \ ++ .invert = winvert, .kcontrols = wcontrols, .num_kcontrols = wncontrols, \ ++ .event = wevent, .event_flags = wflags} ++#define SND_SOC_DAPM_MICBIAS_E(wname, wreg, wshift, winvert, wevent, wflags) \ ++{ .id = snd_soc_dapm_micbias, .name = wname, .reg = wreg, .shift = wshift, \ ++ .invert = winvert, .kcontrols = NULL, .num_kcontrols = 0, \ ++ .event = wevent, .event_flags = wflags} ++#define SND_SOC_DAPM_SWITCH_E(wname, wreg, wshift, winvert, wcontrols, \ ++ wevent, wflags) \ ++{ .id = snd_soc_dapm_switch, .name = wname, .reg = wreg, .shift = wshift, \ ++ .invert = winvert, .kcontrols = wcontrols, .num_kcontrols = 1 \ ++ .event = wevent, .event_flags = wflags} ++#define SND_SOC_DAPM_MUX_E(wname, wreg, wshift, winvert, wcontrols, \ ++ wevent, wflags) \ ++{ .id = snd_soc_dapm_mux, .name = wname, .reg = wreg, .shift = wshift, \ ++ .invert = winvert, .kcontrols = wcontrols, .num_kcontrols = 1, \ ++ .event = wevent, .event_flags = wflags} ++ ++/* events that are pre and post DAPM */ ++#define SND_SOC_DAPM_PRE(wname, wevent) \ ++{ .id = snd_soc_dapm_pre, .name = wname, .kcontrols = NULL, \ ++ .num_kcontrols = 0, .event = wevent, \ ++ .event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD} ++#define SND_SOC_DAPM_POST(wname, wevent) \ ++{ .id = snd_soc_dapm_post, .name = wname, .kcontrols = NULL, \ ++ .num_kcontrols = 0, .event = wevent, \ ++ .event_flags = SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD} ++ ++/* stream domain */ ++#define SND_SOC_DAPM_DAC(wname, stname, wreg, wshift, winvert) \ ++{ .id = snd_soc_dapm_dac, .name = wname, .sname = stname, .reg = wreg, \ ++ .shift = wshift, .invert = winvert} ++#define SND_SOC_DAPM_ADC(wname, stname, wreg, wshift, winvert) \ ++{ .id = snd_soc_dapm_adc, .name = wname, .sname = stname, .reg = wreg, \ ++ .shift = wshift, .invert = winvert} ++ ++/* dapm kcontrol types */ ++#define SOC_DAPM_SINGLE(xname, reg, shift, mask, invert) \ ++{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ ++ .info = snd_soc_info_volsw, \ ++ .get = snd_soc_dapm_get_volsw, .put = snd_soc_dapm_put_volsw, \ ++ .private_value = SOC_SINGLE_VALUE(reg, shift, mask, invert) } ++#define SOC_DAPM_DOUBLE(xname, reg, shift_left, shift_right, mask, invert, \ ++ power) \ ++{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ ++ .info = snd_soc_info_volsw, \ ++ .get = snd_soc_dapm_get_volsw, .put = snd_soc_dapm_put_volsw, \ ++ .private_value = (reg) | ((shift_left) << 8) | ((shift_right) << 12) |\ ++ ((mask) << 16) | ((invert) << 24) } ++#define SOC_DAPM_ENUM(xname, xenum) \ ++{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ ++ .info = snd_soc_info_enum_double, \ ++ .get = snd_soc_dapm_get_enum_double, \ ++ .put = snd_soc_dapm_put_enum_double, \ ++ .private_value = (unsigned long)&xenum } ++ ++/* dapm stream operations */ ++#define SND_SOC_DAPM_STREAM_NOP 0x0 ++#define SND_SOC_DAPM_STREAM_START 0x1 ++#define SND_SOC_DAPM_STREAM_STOP 0x2 ++#define SND_SOC_DAPM_STREAM_SUSPEND 0x4 ++#define SND_SOC_DAPM_STREAM_RESUME 0x8 ++#define SND_SOC_DAPM_STREAM_PAUSE_PUSH 0x10 ++#define SND_SOC_DAPM_STREAM_PAUSE_RELEASE 0x20 ++ ++/* dapm event types */ ++#define SND_SOC_DAPM_PRE_PMU 0x1 /* before widget power up */ ++#define SND_SOC_DAPM_POST_PMU 0x2 /* after widget power up */ ++#define SND_SOC_DAPM_PRE_PMD 0x4 /* before widget power down */ ++#define SND_SOC_DAPM_POST_PMD 0x8 /* after widget power down */ ++#define SND_SOC_DAPM_PRE_REG 0x10 /* before audio path setup */ ++#define SND_SOC_DAPM_POST_REG 0x20 /* after audio path setup */ ++ ++/* convenience event type detection */ ++#define SND_SOC_DAPM_EVENT_ON(e) \ ++ (e & (SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU)) ++#define SND_SOC_DAPM_EVENT_OFF(e) \ ++ (e & (SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD)) ++ ++struct snd_soc_dapm_widget; ++enum snd_soc_dapm_type; ++struct snd_soc_dapm_path; ++struct snd_soc_dapm_pin; ++ ++/* dapm controls */ ++int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_value *ucontrol); ++int snd_soc_dapm_get_volsw(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_value *ucontrol); ++int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_value *ucontrol); ++int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_value *ucontrol); ++int snd_soc_dapm_new_control(struct snd_soc_codec *codec, ++ const struct snd_soc_dapm_widget *widget); ++ ++/* dapm path setup */ ++int snd_soc_dapm_connect_input(struct snd_soc_codec *codec, ++ const char *sink_name, const char *control_name, const char *src_name); ++int snd_soc_dapm_new_widgets(struct snd_soc_codec *codec); ++void snd_soc_dapm_free(struct snd_soc_device *socdev); ++ ++/* dapm events */ ++int snd_soc_dapm_stream_event(struct snd_soc_codec *codec, char *stream, ++ int event); ++ ++/* dapm sys fs - used by the core */ ++int snd_soc_dapm_sys_add(struct device *dev); ++ ++/* dapm audio endpoint control */ ++int snd_soc_dapm_set_endpoint(struct snd_soc_codec *codec, ++ char *pin, int status); ++int snd_soc_dapm_sync_endpoints(struct snd_soc_codec *codec); ++ ++/* dapm widget types */ ++enum snd_soc_dapm_type { ++ snd_soc_dapm_input = 0, /* input pin */ ++ snd_soc_dapm_output, /* output pin */ ++ snd_soc_dapm_mux, /* selects 1 analog signal from many inputs */ ++ snd_soc_dapm_mixer, /* mixes several analog signals together */ ++ snd_soc_dapm_pga, /* programmable gain/attenuation (volume) */ ++ snd_soc_dapm_adc, /* analog to digital converter */ ++ snd_soc_dapm_dac, /* digital to analog converter */ ++ snd_soc_dapm_micbias, /* microphone bias (power) */ ++ snd_soc_dapm_mic, /* microphone */ ++ snd_soc_dapm_hp, /* headphones */ ++ snd_soc_dapm_spk, /* speaker */ ++ snd_soc_dapm_line, /* line input/output */ ++ snd_soc_dapm_switch, /* analog switch */ ++ snd_soc_dapm_vmid, /* codec bias/vmid - to minimise pops */ ++ snd_soc_dapm_pre, /* machine specific pre widget - exec first */ ++ snd_soc_dapm_post, /* machine specific post widget - exec last */ ++}; ++ ++/* dapm audio path between two widgets */ ++struct snd_soc_dapm_path { ++ char *name; ++ char *long_name; ++ ++ /* source (input) and sink (output) widgets */ ++ struct snd_soc_dapm_widget *source; ++ struct snd_soc_dapm_widget *sink; ++ struct snd_kcontrol *kcontrol; ++ ++ /* status */ ++ u32 connect:1; /* source and sink widgets are connected */ ++ u32 walked:1; /* path has been walked */ ++ ++ struct list_head list_source; ++ struct list_head list_sink; ++ struct list_head list; ++}; ++ ++/* dapm widget */ ++struct snd_soc_dapm_widget { ++ enum snd_soc_dapm_type id; ++ char *name; /* widget name */ ++ char *sname; /* stream name */ ++ struct snd_soc_codec *codec; ++ struct list_head list; ++ ++ /* dapm control */ ++ short reg; /* negative reg = no direct dapm */ ++ unsigned char shift; /* bits to shift */ ++ unsigned int saved_value; /* widget saved value */ ++ unsigned int value; /* widget current value */ ++ unsigned char power:1; /* block power status */ ++ unsigned char invert:1; /* invert the power bit */ ++ unsigned char active:1; /* active stream on DAC, ADC's */ ++ unsigned char connected:1; /* connected codec pin */ ++ unsigned char new:1; /* cnew complete */ ++ unsigned char ext:1; /* has external widgets */ ++ unsigned char muted:1; /* muted for pop reduction */ ++ unsigned char suspend:1; /* was active before suspend */ ++ unsigned char pmdown:1; /* waiting for timeout */ ++ ++ /* external events */ ++ unsigned short event_flags; /* flags to specify event types */ ++ int (*event)(struct snd_soc_dapm_widget*, int); ++ ++ /* kcontrols that relate to this widget */ ++ int num_kcontrols; ++ const struct snd_kcontrol_new *kcontrols; ++ ++ /* widget input and outputs */ ++ struct list_head sources; ++ struct list_head sinks; ++}; ++ ++#endif +Index: linux-2.6-pxa-new/include/sound/soc.h +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/include/sound/soc.h +@@ -0,0 +1,487 @@ ++/* ++ * linux/sound/soc.h -- ALSA SoC Layer ++ * ++ * Author: Liam Girdwood ++ * Created: Aug 11th 2005 ++ * Copyright: Wolfson Microelectronics. PLC. ++ * ++ * This program is free software; you can redistribute it and/or modify ++ * it under the terms of the GNU General Public License version 2 as ++ * published by the Free Software Foundation. ++ */ ++ ++#ifndef __LINUX_SND_SOC_H ++#define __LINUX_SND_SOC_H ++ ++#include <linux/platform_device.h> ++#include <linux/types.h> ++#include <sound/driver.h> ++#include <sound/core.h> ++#include <sound/pcm.h> ++#include <sound/control.h> ++#include <sound/ac97_codec.h> ++ ++#define SND_SOC_VERSION "0.12.4" ++ ++/* ++ * Convenience kcontrol builders ++ */ ++#define SOC_SINGLE_VALUE(reg,shift,mask,invert) ((reg) | ((shift) << 8) |\ ++ ((shift) << 12) | ((mask) << 16) | ((invert) << 24)) ++#define SOC_SINGLE_VALUE_EXT(reg,mask,invert) ((reg) | ((mask) << 16) |\ ++ ((invert) << 31)) ++#define SOC_SINGLE(xname, reg, shift, mask, invert) \ ++{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ ++ .info = snd_soc_info_volsw, .get = snd_soc_get_volsw,\ ++ .put = snd_soc_put_volsw, \ ++ .private_value = SOC_SINGLE_VALUE(reg, shift, mask, invert) } ++#define SOC_DOUBLE(xname, reg, shift_left, shift_right, mask, invert) \ ++{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ ++ .info = snd_soc_info_volsw, .get = snd_soc_get_volsw, \ ++ .put = snd_soc_put_volsw, \ ++ .private_value = (reg) | ((shift_left) << 8) | \ ++ ((shift_right) << 12) | ((mask) << 16) | ((invert) << 24) } ++#define SOC_DOUBLE_R(xname, reg_left, reg_right, shift, mask, invert) \ ++{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ ++ .info = snd_soc_info_volsw_2r, \ ++ .get = snd_soc_get_volsw_2r, .put = snd_soc_put_volsw_2r, \ ++ .private_value = (reg_left) | ((shift) << 8) | \ ++ ((mask) << 12) | ((invert) << 20) | ((reg_right) << 24) } ++#define SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, xtexts) \ ++{ .reg = xreg, .shift_l = xshift_l, .shift_r = xshift_r, \ ++ .mask = xmask, .texts = xtexts } ++#define SOC_ENUM_SINGLE(xreg, xshift, xmask, xtexts) \ ++ SOC_ENUM_DOUBLE(xreg, xshift, xshift, xmask, xtexts) ++#define SOC_ENUM_SINGLE_EXT(xmask, xtexts) \ ++{ .mask = xmask, .texts = xtexts } ++#define SOC_ENUM(xname, xenum) \ ++{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname,\ ++ .info = snd_soc_info_enum_double, \ ++ .get = snd_soc_get_enum_double, .put = snd_soc_put_enum_double, \ ++ .private_value = (unsigned long)&xenum } ++#define SOC_SINGLE_EXT(xname, xreg, xmask, xinvert,\ ++ xhandler_get, xhandler_put) \ ++{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ ++ .info = snd_soc_info_volsw_ext, \ ++ .get = xhandler_get, .put = xhandler_put, \ ++ .private_value = SOC_SINGLE_VALUE_EXT(xreg, xmask, xinvert) } ++#define SOC_SINGLE_BOOL_EXT(xname, xdata, xhandler_get, xhandler_put) \ ++{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ ++ .info = snd_soc_info_bool_ext, \ ++ .get = xhandler_get, .put = xhandler_put, \ ++ .private_value = xdata } ++#define SOC_ENUM_EXT(xname, xenum, xhandler_get, xhandler_put) \ ++{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ ++ .info = snd_soc_info_enum_ext, \ ++ .get = xhandler_get, .put = xhandler_put, \ ++ .private_value = (unsigned long)&xenum } ++ ++/* ++ * Digital Audio Interface (DAI) types ++ */ ++#define SND_SOC_DAI_AC97 0x1 ++#define SND_SOC_DAI_I2S 0x2 ++#define SND_SOC_DAI_PCM 0x4 ++ ++/* ++ * DAI hardware audio formats ++ */ ++#define SND_SOC_DAIFMT_I2S (1 << 0) /* I2S mode */ ++#define SND_SOC_DAIFMT_RIGHT_J (1 << 1) /* Right justified mode */ ++#define SND_SOC_DAIFMT_LEFT_J (1 << 2) /* Left Justified mode */ ++#define SND_SOC_DAIFMT_DSP_A (1 << 3) /* L data msb after FRM or LRC */ ++#define SND_SOC_DAIFMT_DSP_B (1 << 4) /* L data msb during FRM or LRC */ ++#define SND_SOC_DAIFMT_AC97 (1 << 5) /* AC97 */ ++ ++/* ++ * DAI hardware signal inversions ++ */ ++#define SND_SOC_DAIFMT_NB_NF (1 << 8) /* normal bit clock + frame */ ++#define SND_SOC_DAIFMT_NB_IF (1 << 9) /* normal bclk + inv frm */ ++#define SND_SOC_DAIFMT_IB_NF (1 << 10) /* invert bclk + nor frm */ ++#define SND_SOC_DAIFMT_IB_IF (1 << 11) /* invert bclk + frm */ ++ ++/* ++ * DAI hardware clock masters ++ * This is wrt the codec, the inverse is true for the interface ++ * i.e. if the codec is clk and frm master then the interface is ++ * clk and frame slave. ++ */ ++#define SND_SOC_DAIFMT_CBM_CFM (1 << 12) /* codec clk & frm master */ ++#define SND_SOC_DAIFMT_CBS_CFM (1 << 13) /* codec clk slave & frm master */ ++#define SND_SOC_DAIFMT_CBM_CFS (1 << 14) /* codec clk master & frame slave */ ++#define SND_SOC_DAIFMT_CBS_CFS (1 << 15) /* codec clk & frm slave */ ++ ++#define SND_SOC_DAIFMT_FORMAT_MASK 0x00ff ++#define SND_SOC_DAIFMT_INV_MASK 0x0f00 ++#define SND_SOC_DAIFMT_CLOCK_MASK 0xf000 ++ ++/* ++ * DAI hardware audio direction ++ */ ++#define SND_SOC_DAIDIR_PLAYBACK 0x1 ++#define SND_SOC_DAIDIR_CAPTURE 0x2 ++ ++/* ++ * DAI hardware Time Division Multiplexing (TDM) Slots ++ * Left and Right data word positions ++ * This is measured in words (sample size) and not bits. ++ */ ++#define SND_SOC_DAITDM_LRDW(l,r) ((l << 8) | r) ++ ++/* ++ * DAI hardware clock ratios ++ * bit clock can either be a generated by dividing mclk or ++ * by multiplying sample rate, hence there are 2 definitions below ++ * depending on codec type. ++ */ ++/* ratio of sample rate to mclk/sysclk */ ++#define SND_SOC_FS_ALL 0xffff /* all mclk supported */ ++ ++/* bit clock dividers */ ++#define SND_SOC_FSBD(x) (1 << (x - 1)) /* ratio mclk:bclk */ ++#define SND_SOC_FSBD_REAL(x) (ffs(x)) ++ ++/* bit clock ratio to (sample rate * channels * word size) */ ++#define SND_SOC_FSBW(x) (1 << (x - 1)) ++#define SND_SOC_FSBW_REAL(x) (ffs(x)) ++/* all bclk ratios supported */ ++#define SND_SOC_FSB_ALL ~0ULL ++ ++/* ++ * DAI hardware flags ++ */ ++/* use bfs mclk divider mode (BCLK = MCLK / x) */ ++#define SND_SOC_DAI_BFS_DIV 0x1 ++/* use bfs rate mulitplier (BCLK = RATE * x)*/ ++#define SND_SOC_DAI_BFS_RATE 0x2 ++/* use bfs rcw multiplier (BCLK = RATE * CHN * WORD SIZE) */ ++#define SND_SOC_DAI_BFS_RCW 0x4 ++/* capture and playback can use different clocks */ ++#define SND_SOC_DAI_ASYNC 0x8 ++/* can use gated BCLK */ ++#define SND_SOC_DAI_GATED 0x10 ++ ++/* ++ * AC97 codec ID's bitmask ++ */ ++#define SND_SOC_DAI_AC97_ID0 (1 << 0) ++#define SND_SOC_DAI_AC97_ID1 (1 << 1) ++#define SND_SOC_DAI_AC97_ID2 (1 << 2) ++#define SND_SOC_DAI_AC97_ID3 (1 << 3) ++ ++struct snd_soc_device; ++struct snd_soc_pcm_stream; ++struct snd_soc_ops; ++struct snd_soc_dai_mode; ++struct snd_soc_pcm_runtime; ++struct snd_soc_codec_dai; ++struct snd_soc_cpu_dai; ++struct snd_soc_codec; ++struct snd_soc_machine_config; ++struct soc_enum; ++struct snd_soc_ac97_ops; ++struct snd_soc_clock_info; ++ ++typedef int (*hw_write_t)(void *,const char* ,int); ++typedef int (*hw_read_t)(void *,char* ,int); ++ ++extern struct snd_ac97_bus_ops soc_ac97_ops; ++ ++/* pcm <-> DAI connect */ ++void snd_soc_free_pcms(struct snd_soc_device *socdev); ++int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char *xid); ++int snd_soc_register_card(struct snd_soc_device *socdev); ++ ++/* set runtime hw params */ ++int snd_soc_set_runtime_hwparams(struct snd_pcm_substream *substream, ++ const struct snd_pcm_hardware *hw); ++int snd_soc_get_rate(int rate); ++ ++/* codec IO */ ++#define snd_soc_read(codec, reg) codec->read(codec, reg) ++#define snd_soc_write(codec, reg, value) codec->write(codec, reg, value) ++ ++/* codec register bit access */ ++int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned short reg, ++ unsigned short mask, unsigned short value); ++int snd_soc_test_bits(struct snd_soc_codec *codec, unsigned short reg, ++ unsigned short mask, unsigned short value); ++ ++int snd_soc_new_ac97_codec(struct snd_soc_codec *codec, ++ struct snd_ac97_bus_ops *ops, int num); ++void snd_soc_free_ac97_codec(struct snd_soc_codec *codec); ++ ++/* ++ *Controls ++ */ ++struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template, ++ void *data, char *long_name); ++int snd_soc_info_enum_double(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_info *uinfo); ++int snd_soc_info_enum_ext(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_info *uinfo); ++int snd_soc_get_enum_double(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_value *ucontrol); ++int snd_soc_put_enum_double(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_value *ucontrol); ++int snd_soc_info_volsw(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_info *uinfo); ++int snd_soc_info_volsw_ext(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_info *uinfo); ++int snd_soc_info_bool_ext(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_info *uinfo); ++int snd_soc_get_volsw(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_value *ucontrol); ++int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_value *ucontrol); ++int snd_soc_info_volsw_2r(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_info *uinfo); ++int snd_soc_get_volsw_2r(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_value *ucontrol); ++int snd_soc_put_volsw_2r(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_value *ucontrol); ++ ++/* SoC PCM stream information */ ++struct snd_soc_pcm_stream { ++ char *stream_name; ++ unsigned int rate_min; /* min rate */ ++ unsigned int rate_max; /* max rate */ ++ unsigned int channels_min; /* min channels */ ++ unsigned int channels_max; /* max channels */ ++ unsigned int active:1; /* stream is in use */ ++}; ++ ++/* SoC audio ops */ ++struct snd_soc_ops { ++ int (*startup)(struct snd_pcm_substream *); ++ void (*shutdown)(struct snd_pcm_substream *); ++ int (*hw_params)(struct snd_pcm_substream *, struct snd_pcm_hw_params *); ++ int (*hw_free)(struct snd_pcm_substream *); ++ int (*prepare)(struct snd_pcm_substream *); ++ int (*trigger)(struct snd_pcm_substream *, int); ++}; ++ ++/* SoC DAI hardware mode */ ++struct snd_soc_dai_mode { ++ u16 fmt; /* SND_SOC_DAIFMT_* */ ++ u16 tdm; /* SND_SOC_HWTDM_* */ ++ u64 pcmfmt; /* SNDRV_PCM_FMTBIT_* */ ++ u16 pcmrate; /* SND_SOC_HWRATE_* */ ++ u16 pcmdir:2; /* SND_SOC_HWDIR_* */ ++ u16 flags:8; /* hw flags */ ++ u16 fs; /* mclk to rate divider */ ++ u64 bfs; /* mclk to bclk dividers */ ++ unsigned long priv; /* private mode data */ ++}; ++ ++/* DAI capabilities */ ++struct snd_soc_dai_cap { ++ int num_modes; /* number of DAI modes */ ++ struct snd_soc_dai_mode *mode; /* array of supported DAI modes */ ++}; ++ ++/* SoC Codec DAI */ ++struct snd_soc_codec_dai { ++ char *name; ++ int id; ++ ++ /* DAI capabilities */ ++ struct snd_soc_pcm_stream playback; ++ struct snd_soc_pcm_stream capture; ++ struct snd_soc_dai_cap caps; ++ ++ /* DAI runtime info */ ++ struct snd_soc_dai_mode dai_runtime; ++ struct snd_soc_ops ops; ++ unsigned int (*config_sysclk)(struct snd_soc_codec_dai*, ++ struct snd_soc_clock_info *info, unsigned int clk); ++ int (*digital_mute)(struct snd_soc_codec *, ++ struct snd_soc_codec_dai*, int); ++ unsigned int mclk; /* the audio master clock */ ++ unsigned int pll_in; /* the PLL input clock */ ++ unsigned int pll_out; /* the PLL output clock */ ++ unsigned int clk_div; /* internal clock divider << 1 (for fractions) */ ++ unsigned int active; ++ unsigned char pop_wait:1; ++ ++ /* DAI private data */ ++ void *private_data; ++}; ++ ++/* SoC CPU DAI */ ++struct snd_soc_cpu_dai { ++ ++ /* DAI description */ ++ char *name; ++ unsigned int id; ++ unsigned char type; ++ ++ /* DAI callbacks */ ++ int (*probe)(struct platform_device *pdev); ++ void (*remove)(struct platform_device *pdev); ++ int (*suspend)(struct platform_device *pdev, ++ struct snd_soc_cpu_dai *cpu_dai); ++ int (*resume)(struct platform_device *pdev, ++ struct snd_soc_cpu_dai *cpu_dai); ++ unsigned int (*config_sysclk)(struct snd_soc_cpu_dai *cpu_dai, ++ struct snd_soc_clock_info *info, unsigned int clk); ++ ++ /* DAI capabilities */ ++ struct snd_soc_pcm_stream capture; ++ struct snd_soc_pcm_stream playback; ++ struct snd_soc_dai_cap caps; ++ ++ /* DAI runtime info */ ++ struct snd_soc_dai_mode dai_runtime; ++ struct snd_soc_ops ops; ++ struct snd_pcm_runtime *runtime; ++ unsigned char active:1; ++ unsigned int mclk; ++ void *dma_data; ++ ++ /* DAI private data */ ++ void *private_data; ++}; ++ ++/* SoC Audio Codec */ ++struct snd_soc_codec { ++ char *name; ++ struct module *owner; ++ struct mutex mutex; ++ ++ /* callbacks */ ++ int (*dapm_event)(struct snd_soc_codec *codec, int event); ++ ++ /* runtime */ ++ struct snd_card *card; ++ struct snd_ac97 *ac97; /* for ad-hoc ac97 devices */ ++ unsigned int active; ++ unsigned int pcm_devs; ++ void *private_data; ++ ++ /* codec IO */ ++ void *control_data; /* codec control (i2c/3wire) data */ ++ unsigned int (*read)(struct snd_soc_codec *, unsigned int); ++ int (*write)(struct snd_soc_codec *, unsigned int, unsigned int); ++ hw_write_t hw_write; ++ hw_read_t hw_read; ++ void *reg_cache; ++ short reg_cache_size; ++ short reg_cache_step; ++ ++ /* dapm */ ++ struct list_head dapm_widgets; ++ struct list_head dapm_paths; ++ unsigned int dapm_state; ++ unsigned int suspend_dapm_state; ++ ++ /* codec DAI's */ ++ struct snd_soc_codec_dai *dai; ++ unsigned int num_dai; ++}; ++ ++/* codec device */ ++struct snd_soc_codec_device { ++ int (*probe)(struct platform_device *pdev); ++ int (*remove)(struct platform_device *pdev); ++ int (*suspend)(struct platform_device *pdev, pm_message_t state); ++ int (*resume)(struct platform_device *pdev); ++}; ++ ++/* SoC platform interface */ ++struct snd_soc_platform { ++ char *name; ++ ++ int (*probe)(struct platform_device *pdev); ++ int (*remove)(struct platform_device *pdev); ++ int (*suspend)(struct platform_device *pdev, ++ struct snd_soc_cpu_dai *cpu_dai); ++ int (*resume)(struct platform_device *pdev, ++ struct snd_soc_cpu_dai *cpu_dai); ++ ++ /* pcm creation and destruction */ ++ int (*pcm_new)(struct snd_card *, struct snd_soc_codec_dai *, ++ struct snd_pcm *); ++ void (*pcm_free)(struct snd_pcm *); ++ ++ /* platform stream ops */ ++ struct snd_pcm_ops *pcm_ops; ++}; ++ ++/* SoC machine DAI configuration, glues a codec and cpu DAI together */ ++struct snd_soc_dai_link { ++ char *name; /* Codec name */ ++ char *stream_name; /* Stream name */ ++ ++ /* DAI */ ++ struct snd_soc_codec_dai *codec_dai; ++ struct snd_soc_cpu_dai *cpu_dai; ++ u32 flags; /* DAI config preference flags */ ++ ++ /* codec/machine specific init - e.g. add machine controls */ ++ int (*init)(struct snd_soc_codec *codec); ++ ++ /* audio sysclock configuration */ ++ unsigned int (*config_sysclk)(struct snd_soc_pcm_runtime *rtd, ++ struct snd_soc_clock_info *info); ++}; ++ ++/* SoC machine */ ++struct snd_soc_machine { ++ char *name; ++ ++ int (*probe)(struct platform_device *pdev); ++ int (*remove)(struct platform_device *pdev); ++ ++ /* the pre and post PM functions are used to do any PM work before and ++ * after the codec and DAI's do any PM work. */ ++ int (*suspend_pre)(struct platform_device *pdev, pm_message_t state); ++ int (*suspend_post)(struct platform_device *pdev, pm_message_t state); ++ int (*resume_pre)(struct platform_device *pdev); ++ int (*resume_post)(struct platform_device *pdev); ++ ++ /* machine stream operations */ ++ struct snd_soc_ops *ops; ++ ++ /* CPU <--> Codec DAI links */ ++ struct snd_soc_dai_link *dai_link; ++ int num_links; ++}; ++ ++/* SoC Device - the audio subsystem */ ++struct snd_soc_device { ++ struct device *dev; ++ struct snd_soc_machine *machine; ++ struct snd_soc_platform *platform; ++ struct snd_soc_codec *codec; ++ struct snd_soc_codec_device *codec_dev; ++ void *codec_data; ++}; ++ ++/* runtime channel data */ ++struct snd_soc_pcm_runtime { ++ struct snd_soc_codec_dai *codec_dai; ++ struct snd_soc_cpu_dai *cpu_dai; ++ struct snd_soc_device *socdev; ++}; ++ ++/* enumerated kcontrol */ ++struct soc_enum { ++ unsigned short reg; ++ unsigned short reg2; ++ unsigned char shift_l; ++ unsigned char shift_r; ++ unsigned int mask; ++ const char **texts; ++ void *dapm; ++}; ++ ++/* clocking configuration data */ ++struct snd_soc_clock_info { ++ unsigned int rate; ++ unsigned int fs; ++ unsigned int bclk_master; ++}; ++ ++#endif +Index: linux-2.6-pxa-new/sound/Kconfig +=================================================================== +--- linux-2.6-pxa-new.orig/sound/Kconfig ++++ linux-2.6-pxa-new/sound/Kconfig +@@ -76,6 +76,8 @@ source "sound/sparc/Kconfig" + + source "sound/parisc/Kconfig" + ++source "sound/soc/Kconfig" ++ + endmenu + + menu "Open Sound System" +Index: linux-2.6-pxa-new/sound/soc/Kconfig +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/sound/soc/Kconfig +@@ -0,0 +1,37 @@ ++# ++# SoC audio configuration ++# ++ ++menu "SoC audio support" ++ depends on SND!=n ++ ++config SND_SOC_AC97_BUS ++ bool ++ ++config SND_SOC ++ tristate "SoC audio support" ++ ---help--- ++ ++ If you want SoC support, you should say Y here and also to the ++ specific driver for your SoC below. You will also need to select the ++ specific codec(s) attached to the SoC ++ ++ This SoC audio support can also be built as a module. If so, the module ++ will be called snd-soc-core. ++ ++# All the supported Soc's ++menu "Soc Platforms" ++depends on SND_SOC ++source "sound/soc/pxa/Kconfig" ++source "sound/soc/at91/Kconfig" ++source "sound/soc/imx/Kconfig" ++source "sound/soc/s3c24xx/Kconfig" ++endmenu ++ ++# Supported codecs ++menu "Soc Codecs" ++depends on SND_SOC ++source "sound/soc/codecs/Kconfig" ++endmenu ++ ++endmenu +Index: linux-2.6-pxa-new/sound/soc/Makefile +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/sound/soc/Makefile +@@ -0,0 +1,4 @@ ++snd-soc-core-objs := soc-core.o soc-dapm.o ++ ++obj-$(CONFIG_SND_SOC) += snd-soc-core.o ++obj-$(CONFIG_SND_SOC) += pxa/ at91/ imx/ s3c24xx/ codecs/ +Index: linux-2.6-pxa-new/sound/soc/codecs/Kconfig +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/sound/soc/codecs/Kconfig +@@ -0,0 +1,90 @@ ++config SND_SOC_AC97_CODEC ++ tristate "SoC generic AC97 support" ++ depends SND_SOC ++ help ++ Say Y or M if you want generic AC97 support. This is not required ++ for the AC97 codecs listed below. ++ ++config SND_SOC_WM8711 ++ tristate "SoC driver for the WM8711 codec" ++ depends SND_SOC ++ help ++ Say Y or M if you want to support the WM8711 codec. ++ ++config SND_SOC_WM8510 ++ tristate "SoC driver for the WM8510 codec" ++ depends SND_SOC ++ help ++ Say Y or M if you want to support the WM8711 codec. ++ ++config SND_SOC_WM8731 ++ tristate "SoC driver for the WM8731 codec" ++ depends SND_SOC ++ help ++ Say Y or M if you want to support the WM8731 codec. ++ ++config SND_SOC_WM8750 ++ tristate "SoC driver for the WM8750 codec" ++ depends SND_SOC ++ help ++ Say Y or M if you want to support the WM8750 codec. ++ ++config SND_SOC_WM8753 ++ tristate "SoC driver for the WM8753 codec" ++ depends SND_SOC ++ help ++ Say Y or M if you want to support the WM8753 codec. ++ ++config SND_SOC_WM8772 ++ tristate "SoC driver for the WM8772 codec" ++ depends SND_SOC ++ help ++ Say Y or M if you want to support the WM8772 codec. ++ ++config SND_SOC_WM8971 ++ tristate "SoC driver for the WM8971 codec" ++ depends SND_SOC ++ help ++ Say Y or M if you want to support the WM8971 codec. ++ ++config SND_SOC_WM8976 ++ tristate "SoC driver for the WM8976 codec" ++ depends SND_SOC ++ help ++ Say Y or M if you want to support the WM8976 codec. ++ ++config SND_SOC_WM8974 ++ tristate "SoC driver for the WM8974 codec" ++ depends SND_SOC ++ help ++ Say Y or M if you want to support the WM8974 codec. ++ ++config SND_SOC_WM8980 ++ tristate "SoC driver for the WM8980 codec" ++ depends SND_SOC ++ help ++ Say Y or M if you want to support the WM8980 codec. ++ ++config SND_SOC_WM9713 ++ tristate "SoC driver for the WM9713 codec" ++ depends SND_SOC ++ help ++ Say Y or M if you want to support the WM9713 codec. ++ ++config SND_SOC_WM9712 ++ tristate "SoC driver for the WM9712 codec" ++ depends SND_SOC ++ help ++ Say Y or M if you want to support the WM9712 codec. ++ ++config SND_SOC_UDA1380 ++ tristate "SoC driver for the UDA1380 codec" ++ depends SND_SOC ++ help ++ Say Y or M if you want to support the UDA1380 codec. ++ ++config SND_SOC_AK4535 ++ tristate "SoC driver for the AK4535 codec" ++ depends SND_SOC ++ help ++ Say Y or M if you want to support the AK4535 codec. +Index: linux-2.6-pxa-new/sound/soc/codecs/Makefile +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/sound/soc/codecs/Makefile +@@ -0,0 +1,31 @@ ++snd-soc-ac97-objs := ac97.o ++snd-soc-wm8711-objs := wm8711.o ++snd-soc-wm8510-objs := wm8510.o ++snd-soc-wm8731-objs := wm8731.o ++snd-soc-wm8750-objs := wm8750.o ++snd-soc-wm8753-objs := wm8753.o ++snd-soc-wm8772-objs := wm8772.o ++snd-soc-wm8971-objs := wm8971.o ++snd-soc-wm8974-objs := wm8974.o ++snd-soc-wm8976-objs := wm8976.o ++snd-soc-wm8980-objs := wm8980.o ++snd-soc-uda1380-objs := uda1380.o ++snd-soc-ak4535-objs := ak4535.o ++snd-soc-wm9713-objs := wm9713.o ++snd-soc-wm9712-objs := wm9712.o ++ ++obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o ++obj-$(CONFIG_SND_SOC_WM8711) += snd-soc-wm8711.o ++obj-$(CONFIG_SND_SOC_WM8510) += snd-soc-wm8510.o ++obj-$(CONFIG_SND_SOC_WM8731) += snd-soc-wm8731.o ++obj-$(CONFIG_SND_SOC_WM8750) += snd-soc-wm8750.o ++obj-$(CONFIG_SND_SOC_WM8753) += snd-soc-wm8753.o ++obj-$(CONFIG_SND_SOC_WM8772) += snd-soc-wm8772.o ++obj-$(CONFIG_SND_SOC_WM8971) += snd-soc-wm8971.o ++obj-$(CONFIG_SND_SOC_WM8974) += snd-soc-wm8974.o ++obj-$(CONFIG_SND_SOC_WM8976) += snd-soc-wm8976.o ++obj-$(CONFIG_SND_SOC_WM8980) += snd-soc-wm8980.o ++obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o ++obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o ++obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o ++obj-$(CONFIG_SND_SOC_WM9712) += snd-soc-wm9712.o +Index: linux-2.6-pxa-new/sound/soc/codecs/ac97.c +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/sound/soc/codecs/ac97.c +@@ -0,0 +1,167 @@ ++/* ++ * ac97.c -- ALSA Soc AC97 codec support ++ * ++ * Copyright 2005 Wolfson Microelectronics PLC. ++ * Author: Liam Girdwood ++ * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com ++ * ++ * This program is free software; you can redistribute it and/or modify it ++ * under the terms of the GNU General Public License as published by the ++ * Free Software Foundation; either version 2 of the License, or (at your ++ * option) any later version. ++ * ++ * Revision history ++ * 17th Oct 2005 Initial version. ++ * ++ * Generic AC97 support. ++ */ ++ ++#include <linux/init.h> ++#include <linux/kernel.h> ++#include <linux/device.h> ++#include <sound/driver.h> ++#include <sound/core.h> ++#include <sound/pcm.h> ++#include <sound/ac97_codec.h> ++#include <sound/initval.h> ++#include <sound/soc.h> ++ ++#define AC97_VERSION "0.5" ++ ++#define AC97_DIR \ ++ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE) ++ ++#define AC97_RATES \ ++ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ ++ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ ++ SNDRV_PCM_RATE_48000) ++ ++/* may need to expand this */ ++static struct snd_soc_dai_mode soc_ac97[] = { ++ {0, 0, SNDRV_PCM_FMTBIT_S16_LE, AC97_RATES}, ++ {0, 0, SNDRV_PCM_FMTBIT_S18_3LE, AC97_RATES}, ++ {0, 0, SNDRV_PCM_FMTBIT_S20_3LE, AC97_RATES}, ++}; ++ ++static int ac97_prepare(struct snd_pcm_substream *substream) ++{ ++ struct snd_pcm_runtime *runtime = substream->runtime; ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ struct snd_soc_device *socdev = rtd->socdev; ++ struct snd_soc_codec *codec = socdev->codec; ++ ++ int reg = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? ++ AC97_PCM_FRONT_DAC_RATE : AC97_PCM_LR_ADC_RATE; ++ return snd_ac97_set_rate(codec->ac97, reg, runtime->rate); ++} ++ ++static struct snd_soc_codec_dai ac97_dai = { ++ .name = "AC97 HiFi", ++ .playback = { ++ .stream_name = "AC97 Playback", ++ .channels_min = 1, ++ .channels_max = 2,}, ++ .capture = { ++ .stream_name = "AC97 Capture", ++ .channels_min = 1, ++ .channels_max = 2,}, ++ .ops = { ++ .prepare = ac97_prepare,}, ++ .caps = { ++ .num_modes = ARRAY_SIZE(soc_ac97), ++ .mode = soc_ac97,}, ++}; ++ ++static unsigned int ac97_read(struct snd_soc_codec *codec, ++ unsigned int reg) ++{ ++ return soc_ac97_ops.read(codec->ac97, reg); ++} ++ ++static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, ++ unsigned int val) ++{ ++ soc_ac97_ops.write(codec->ac97, reg, val); ++ return 0; ++} ++ ++static int ac97_soc_probe(struct platform_device *pdev) ++{ ++ struct snd_soc_device *socdev = platform_get_drvdata(pdev); ++ struct snd_soc_codec *codec; ++ struct snd_ac97_bus *ac97_bus; ++ struct snd_ac97_template ac97_template; ++ int ret = 0; ++ ++ printk(KERN_INFO "AC97 SoC Audio Codec %s\n", AC97_VERSION); ++ ++ socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); ++ if (socdev->codec == NULL) ++ return -ENOMEM; ++ codec = socdev->codec; ++ mutex_init(&codec->mutex); ++ ++ codec->name = "AC97"; ++ codec->owner = THIS_MODULE; ++ codec->dai = &ac97_dai; ++ codec->num_dai = 1; ++ codec->write = ac97_write; ++ codec->read = ac97_read; ++ INIT_LIST_HEAD(&codec->dapm_widgets); ++ INIT_LIST_HEAD(&codec->dapm_paths); ++ ++ /* register pcms */ ++ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); ++ if(ret < 0) ++ goto err; ++ ++ /* add codec as bus device for standard ac97 */ ++ ret = snd_ac97_bus(codec->card, 0, &soc_ac97_ops, NULL, &ac97_bus); ++ if(ret < 0) ++ goto bus_err; ++ ++ memset(&ac97_template, 0, sizeof(struct snd_ac97_template)); ++ ret = snd_ac97_mixer(ac97_bus, &ac97_template, &codec->ac97); ++ if(ret < 0) ++ goto bus_err; ++ ++ ret = snd_soc_register_card(socdev); ++ if (ret < 0) ++ goto bus_err; ++ return 0; ++ ++bus_err: ++ snd_soc_free_pcms(socdev); ++ ++err: ++ kfree(socdev->codec->reg_cache); ++ kfree(socdev->codec); ++ socdev->codec = NULL; ++ return ret; ++} ++ ++static int ac97_soc_remove(struct platform_device *pdev) ++{ ++ struct snd_soc_device *socdev = platform_get_drvdata(pdev); ++ struct snd_soc_codec *codec = socdev->codec; ++ ++ if(codec == NULL) ++ return 0; ++ ++ snd_soc_free_pcms(socdev); ++ kfree(socdev->codec->reg_cache); ++ kfree(socdev->codec); ++ ++ return 0; ++} ++ ++struct snd_soc_codec_device soc_codec_dev_ac97= { ++ .probe = ac97_soc_probe, ++ .remove = ac97_soc_remove, ++}; ++ ++EXPORT_SYMBOL_GPL(soc_codec_dev_ac97); ++ ++MODULE_DESCRIPTION("Soc Generic AC97 driver"); ++MODULE_AUTHOR("Liam Girdwood"); ++MODULE_LICENSE("GPL"); +Index: linux-2.6-pxa-new/sound/soc/codecs/ac97.h +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/sound/soc/codecs/ac97.h +@@ -0,0 +1,18 @@ ++/* ++ * linux/sound/codecs/ac97.h -- ALSA SoC Layer ++ * ++ * Author: Liam Girdwood ++ * Created: Dec 1st 2005 ++ * Copyright: Wolfson Microelectronics. PLC. ++ * ++ * This program is free software; you can redistribute it and/or modify ++ * it under the terms of the GNU General Public License version 2 as ++ * published by the Free Software Foundation. ++ */ ++ ++#ifndef __LINUX_SND_SOC_AC97_H ++#define __LINUX_SND_SOC_AC97_H ++ ++extern struct snd_soc_codec_device soc_codec_dev_ac97; ++ ++#endif +Index: linux-2.6-pxa-new/sound/soc/codecs/ak4535.c +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/sound/soc/codecs/ak4535.c +@@ -0,0 +1,701 @@ ++/* ++ * ak4535.c -- AK4535 ALSA Soc Audio driver ++ * ++ * Copyright 2005 Openedhand Ltd. ++ * ++ * Author: Richard Purdie <richard@openedhand.com> ++ * ++ * Based on wm8753.c by Liam Girdwood ++ * ++ * This program is free software; you can redistribute it and/or modify ++ * it under the terms of the GNU General Public License version 2 as ++ * published by the Free Software Foundation. ++ */ ++ ++#include <linux/module.h> ++#include <linux/moduleparam.h> ++#include <linux/init.h> ++#include <linux/delay.h> ++#include <linux/pm.h> ++#include <linux/i2c.h> ++#include <linux/platform_device.h> ++#include <sound/driver.h> ++#include <sound/core.h> ++#include <sound/pcm.h> ++#include <sound/pcm_params.h> ++#include <sound/soc.h> ++#include <sound/soc-dapm.h> ++#include <sound/initval.h> ++ ++#include "ak4535.h" ++ ++#define AUDIO_NAME "ak4535" ++#define AK4535_VERSION "0.3" ++ ++struct snd_soc_codec_device soc_codec_dev_ak4535; ++ ++/* ++ * ak4535 register cache ++ */ ++static const u16 ak4535_reg[AK4535_CACHEREGNUM] = { ++ 0x0000, 0x0080, 0x0000, 0x0003, ++ 0x0002, 0x0000, 0x0011, 0x0001, ++ 0x0000, 0x0040, 0x0036, 0x0010, ++ 0x0000, 0x0000, 0x0057, 0x0000, ++}; ++ ++#define AK4535_DAIFMT \ ++ (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_CBS_CFS | \ ++ SND_SOC_DAIFMT_NB_NF) ++ ++#define AK4535_DIR \ ++ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE) ++ ++#define AK4535_RATES \ ++ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ ++ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ ++ SNDRV_PCM_RATE_48000) ++ ++static struct snd_soc_dai_mode ak4535_modes[] = { ++ /* codec frame and clock slave modes */ ++ { ++ .fmt = AK4535_DAIFMT, ++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, ++ .pcmrate = AK4535_RATES, ++ .pcmdir = AK4535_DIR, ++ .flags = SND_SOC_DAI_BFS_RATE, ++ .fs = 256, ++ .bfs = 64, ++ }, ++ { ++ .fmt = AK4535_DAIFMT, ++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, ++ .pcmrate = AK4535_RATES, ++ .pcmdir = AK4535_DIR, ++ .flags = SND_SOC_DAI_BFS_RATE, ++ .fs = 256, ++ .bfs = 32, ++ }, ++}; ++ ++/* ++ * read ak4535 register cache ++ */ ++static inline unsigned int ak4535_read_reg_cache(struct snd_soc_codec *codec, ++ unsigned int reg) ++{ ++ u16 *cache = codec->reg_cache; ++ if (reg >= AK4535_CACHEREGNUM) ++ return -1; ++ return cache[reg]; ++} ++ ++/* ++ * write ak4535 register cache ++ */ ++static inline void ak4535_write_reg_cache(struct snd_soc_codec *codec, ++ u16 reg, unsigned int value) ++{ ++ u16 *cache = codec->reg_cache; ++ if (reg >= AK4535_CACHEREGNUM) ++ return; ++ cache[reg] = value; ++} ++ ++/* ++ * write to the AK4535 register space ++ */ ++static int ak4535_write(struct snd_soc_codec *codec, unsigned int reg, ++ unsigned int value) ++{ ++ u8 data[2]; ++ ++ /* data is ++ * D15..D8 AK4535 register offset ++ * D7...D0 register data ++ */ ++ data[0] = reg & 0xff; ++ data[1] = value & 0xff; ++ ++ ak4535_write_reg_cache (codec, reg, value); ++ if (codec->hw_write(codec->control_data, data, 2) == 2) ++ return 0; ++ else ++ return -EIO; ++} ++ ++static const char *ak4535_mono_gain[] = {"+6dB", "-17dB"}; ++static const char *ak4535_mono_out[] = {"(L + R)/2", "Hi-Z"}; ++static const char *ak4535_hp_out[] = {"Stereo", "Mono"}; ++static const char *ak4535_deemp[] = {"44.1kHz", "Off", "48kHz", "32kHz"}; ++static const char *ak4535_mic_select[] = {"Internal", "External"}; ++ ++static const struct soc_enum ak4535_enum[] = { ++ SOC_ENUM_SINGLE(AK4535_SIG1, 7, 2, ak4535_mono_gain), ++ SOC_ENUM_SINGLE(AK4535_SIG1, 6, 2, ak4535_mono_out), ++ SOC_ENUM_SINGLE(AK4535_MODE2, 2, 2, ak4535_hp_out), ++ SOC_ENUM_SINGLE(AK4535_DAC, 0, 4, ak4535_deemp), ++ SOC_ENUM_SINGLE(AK4535_MIC, 1, 2, ak4535_mic_select), ++}; ++ ++static const struct snd_kcontrol_new ak4535_snd_controls[] = { ++ SOC_SINGLE("ALC2 Switch", AK4535_SIG1, 1, 1, 0), ++ SOC_ENUM("Mono 1 Output", ak4535_enum[1]), ++ SOC_ENUM("Mono 1 Gain", ak4535_enum[0]), ++ SOC_ENUM("Headphone Output", ak4535_enum[2]), ++ SOC_ENUM("Playback Deemphasis", ak4535_enum[3]), ++ SOC_SINGLE("Bass Volume", AK4535_DAC, 2, 3, 0), ++ SOC_SINGLE("Mic Boost (+20dB) Switch", AK4535_MIC, 0, 1, 0), ++ SOC_ENUM("Mic Select", ak4535_enum[4]), ++ SOC_SINGLE("ALC Operation Time", AK4535_TIMER, 0, 3, 0), ++ SOC_SINGLE("ALC Recovery Time", AK4535_TIMER, 2, 3, 0), ++ SOC_SINGLE("ALC ZC Time", AK4535_TIMER, 4, 3, 0), ++ SOC_SINGLE("ALC 1 Switch", AK4535_ALC1, 5, 1, 0), ++ SOC_SINGLE("ALC 2 Switch", AK4535_ALC1, 6, 1, 0), ++ SOC_SINGLE("ALC Volume", AK4535_ALC2, 0, 127, 0), ++ SOC_SINGLE("Capture Volume", AK4535_PGA, 0, 127, 0), ++ SOC_SINGLE("Left Playback Volume", AK4535_LATT, 0, 127, 1), ++ SOC_SINGLE("Right Playback Volume", AK4535_RATT, 0, 127, 1), ++ SOC_SINGLE("AUX Bypass Volume", AK4535_VOL, 0, 15, 0), ++ SOC_SINGLE("Mic Sidetone Volume", AK4535_VOL, 4, 7, 0), ++}; ++ ++/* add non dapm controls */ ++static int ak4535_add_controls(struct snd_soc_codec *codec) ++{ ++ int err, i; ++ ++ for (i = 0; i < ARRAY_SIZE(ak4535_snd_controls); i++) { ++ err = snd_ctl_add(codec->card, ++ snd_soc_cnew(&ak4535_snd_controls[i],codec, NULL)); ++ if (err < 0) ++ return err; ++ } ++ ++ return 0; ++} ++ ++/* Mono 1 Mixer */ ++static const struct snd_kcontrol_new ak4535_mono1_mixer_controls[] = { ++ SOC_DAPM_SINGLE("Mic Sidetone Switch", AK4535_SIG1, 4, 1, 0), ++ SOC_DAPM_SINGLE("Mono Playback Switch", AK4535_SIG1, 5, 1, 0), ++}; ++ ++/* Stereo Mixer */ ++static const struct snd_kcontrol_new ak4535_stereo_mixer_controls[] = { ++ SOC_DAPM_SINGLE("Mic Sidetone Switch", AK4535_SIG2, 4, 1, 0), ++ SOC_DAPM_SINGLE("Playback Switch", AK4535_SIG2, 7, 1, 0), ++ SOC_DAPM_SINGLE("Aux Bypass Switch", AK4535_SIG2, 5, 1, 0), ++}; ++ ++/* Input Mixer */ ++static const struct snd_kcontrol_new ak4535_input_mixer_controls[] = { ++ SOC_DAPM_SINGLE("Mic Capture Switch", AK4535_MIC, 2, 1, 0), ++ SOC_DAPM_SINGLE("Aux Capture Switch", AK4535_MIC, 5, 1, 0), ++}; ++ ++/* Input mux */ ++static const struct snd_kcontrol_new ak4535_input_mux_control = ++ SOC_DAPM_ENUM("Input Select", ak4535_enum[0]); ++ ++/* HP L switch */ ++static const struct snd_kcontrol_new ak4535_hpl_control = ++ SOC_DAPM_SINGLE("Switch", AK4535_SIG2, 1, 1, 1); ++ ++/* HP R switch */ ++static const struct snd_kcontrol_new ak4535_hpr_control = ++ SOC_DAPM_SINGLE("Switch", AK4535_SIG2, 0, 1, 1); ++ ++/* Speaker switch */ ++static const struct snd_kcontrol_new ak4535_spk_control = ++ SOC_DAPM_SINGLE("Switch", AK4535_MODE2, 0, 0, 0); ++ ++/* mono 2 switch */ ++static const struct snd_kcontrol_new ak4535_mono2_control = ++ SOC_DAPM_SINGLE("Switch", AK4535_SIG1, 0, 1, 0); ++ ++/* Line out switch */ ++static const struct snd_kcontrol_new ak4535_line_control = ++ SOC_DAPM_SINGLE("Switch", AK4535_SIG2, 6, 1, 0); ++ ++/* ak4535 dapm widgets */ ++static const struct snd_soc_dapm_widget ak4535_dapm_widgets[] = { ++ SND_SOC_DAPM_MIXER("Stereo Mixer", SND_SOC_NOPM, 0, 0, ++ &ak4535_stereo_mixer_controls[0], ++ ARRAY_SIZE(ak4535_stereo_mixer_controls)), ++ SND_SOC_DAPM_MIXER("Mono1 Mixer", SND_SOC_NOPM, 0, 0, ++ &ak4535_mono1_mixer_controls[0], ++ ARRAY_SIZE(ak4535_mono1_mixer_controls)), ++ SND_SOC_DAPM_MIXER("Input Mixer", SND_SOC_NOPM, 0, 0, ++ &ak4535_input_mixer_controls[0], ++ ARRAY_SIZE(ak4535_mono1_mixer_controls)), ++ SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0, ++ &ak4535_input_mux_control), ++ SND_SOC_DAPM_DAC("DAC", "Playback", AK4535_PM2, 0, 0), ++ SND_SOC_DAPM_SWITCH("Mono 2 Enable", SND_SOC_NOPM, 0, 0, ++ &ak4535_mono2_control), ++ SND_SOC_DAPM_SWITCH("Speaker Enable", SND_SOC_NOPM, 0, 0, ++ &ak4535_spk_control), ++ SND_SOC_DAPM_SWITCH("Line Out Enable", SND_SOC_NOPM, 0, 0, ++ &ak4535_line_control), ++ SND_SOC_DAPM_SWITCH("Left HP Enable", SND_SOC_NOPM, 0, 0, ++ &ak4535_hpl_control), ++ SND_SOC_DAPM_SWITCH("Right HP Enable", SND_SOC_NOPM, 0, 0, ++ &ak4535_hpr_control), ++ SND_SOC_DAPM_OUTPUT("LOUT"), ++ SND_SOC_DAPM_OUTPUT("HPL"), ++ SND_SOC_DAPM_OUTPUT("ROUT"), ++ SND_SOC_DAPM_OUTPUT("HPR"), ++ SND_SOC_DAPM_OUTPUT("SPP"), ++ SND_SOC_DAPM_OUTPUT("SPN"), ++ SND_SOC_DAPM_OUTPUT("MOUT1"), ++ SND_SOC_DAPM_OUTPUT("MOUT2"), ++ SND_SOC_DAPM_OUTPUT("MICOUT"), ++ SND_SOC_DAPM_ADC("ADC", "Capture", AK4535_PM1, 0, 1), ++ SND_SOC_DAPM_PGA("Spk Amp", AK4535_PM2, 3, 0, NULL, 0), ++ SND_SOC_DAPM_PGA("HP R Amp", AK4535_PM2, 1, 0, NULL, 0), ++ SND_SOC_DAPM_PGA("HP L Amp", AK4535_PM2, 2, 0, NULL, 0), ++ SND_SOC_DAPM_PGA("Mic", AK4535_PM1, 1, 0, NULL, 0), ++ SND_SOC_DAPM_PGA("Line Out", AK4535_PM1, 4, 0, NULL, 0), ++ SND_SOC_DAPM_PGA("Mono Out", AK4535_PM1, 3, 0, NULL, 0), ++ SND_SOC_DAPM_PGA("AUX In", AK4535_PM1, 2, 0, NULL, 0), ++ ++ SND_SOC_DAPM_MICBIAS("Mic Int Bias", AK4535_MIC, 3, 0), ++ SND_SOC_DAPM_MICBIAS("Mic Ext Bias", AK4535_MIC, 4, 0), ++ SND_SOC_DAPM_INPUT("MICIN"), ++ SND_SOC_DAPM_INPUT("MICEXT"), ++ SND_SOC_DAPM_INPUT("AUX"), ++ SND_SOC_DAPM_INPUT("MIN"), ++ SND_SOC_DAPM_INPUT("AIN"), ++}; ++ ++static const char *audio_map[][3] = { ++ /*stereo mixer */ ++ {"Stereo Mixer", "Playback Switch", "DAC"}, ++ {"Stereo Mixer", "Mic Sidetone Switch", "Mic"}, ++ {"Stereo Mixer", "Aux Bypass Switch", "AUX In"}, ++ ++ /* mono1 mixer */ ++ {"Mono1 Mixer", "Mic Sidetone Switch", "Mic"}, ++ {"Mono1 Mixer", "Mono Playback Switch", "DAC"}, ++ ++ /* mono2 mixer */ ++ {"Mono2 Mixer", "Mono Playback Switch", "Stereo Mixer"}, ++ ++ /* Mic */ ++ {"AIN", NULL, "Mic"}, ++ {"Input Mux", "Internal", "Mic Int Bias"}, ++ {"Input Mux", "External", "Mic Ext Bias"}, ++ {"Mic Int Bias", NULL, "MICIN"}, ++ {"Mic Ext Bias", NULL, "MICEXT"}, ++ {"MICOUT", NULL, "Input Mux"}, ++ ++ /* line out */ ++ {"LOUT", "Switch", "Line"}, ++ {"ROUT", "Switch", "Line Out Enable"}, ++ {"Line Out Enable", NULL, "Line Out"}, ++ {"Line Out", NULL, "Stereo Mixer"}, ++ ++ /* mono1 out */ ++ {"MOUT1", NULL, "Mono Out"}, ++ {"Mono Out", NULL, "Mono Mixer"}, ++ ++ /* left HP */ ++ {"HPL", "Switch", "Left HP Enable"}, ++ {"Left HP Enable", NULL, "HP L Amp"}, ++ {"HP L Amp", NULL, "Stereo Mixer"}, ++ ++ /* right HP */ ++ {"HPR", "Switch", "Right HP Enable"}, ++ {"Right HP Enable", NULL, "HP R Amp"}, ++ {"HP R Amp", NULL, "Stereo Mixer"}, ++ ++ /* speaker */ ++ {"SPP", "Switch", "Speaker Enable"}, ++ {"SPN", "Switch", "Speaker Enable"}, ++ {"Speaker Enable", NULL, "Spk Amp"}, ++ {"Spk Amp", NULL, "MIN"}, ++ ++ /* mono 2 */ ++ {"MOUT2", "Switch", "Mono 2 Enable"}, ++ {"Mono 2 Enable", NULL, "Stereo Mixer"}, ++ ++ /* Aux In */ ++ {"Aux In", NULL, "AUX"}, ++ ++ /* ADC */ ++ {"ADC", NULL, "Input Mixer"}, ++ {"Input Mixer", "Mic Capture Switch", "Mic"}, ++ {"Input Mixer", "Aux Capture Switch", "Aux In"}, ++ ++ /* terminator */ ++ {NULL, NULL, NULL}, ++}; ++ ++static int ak4535_add_widgets(struct snd_soc_codec *codec) ++{ ++ int i; ++ ++ for(i = 0; i < ARRAY_SIZE(ak4535_dapm_widgets); i++) { ++ snd_soc_dapm_new_control(codec, &ak4535_dapm_widgets[i]); ++ } ++ ++ /* set up audio path audio_mapnects */ ++ for(i = 0; audio_map[i][0] != NULL; i++) { ++ snd_soc_dapm_connect_input(codec, audio_map[i][0], ++ audio_map[i][1], audio_map[i][2]); ++ } ++ ++ snd_soc_dapm_new_widgets(codec); ++ return 0; ++} ++ ++static int ak4535_pcm_prepare(struct snd_pcm_substream *substream) ++{ ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ struct snd_soc_device *socdev = rtd->socdev; ++ struct snd_soc_codec *codec = socdev->codec; ++ u8 mode = 0, mode2; ++ int bfs; ++ ++ mode2 = ak4535_read_reg_cache(codec, AK4535_MODE2); ++ bfs = SND_SOC_FSBW_REAL(rtd->codec_dai->dai_runtime.bfs); ++ snd_assert(bfs, return -ENODEV); ++ ++ /* interface format */ ++ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_FORMAT_MASK) { ++ case SND_SOC_DAIFMT_I2S: ++ mode = 0x0002; ++ break; ++ case SND_SOC_DAIFMT_LEFT_J: ++ mode = 0x0001; ++ break; ++ } ++ ++ /* set fs */ ++ switch (rtd->codec_dai->dai_runtime.fs) { ++ case 1024: ++ mode2 |= (0x3 << 5); ++ break; ++ case 512: ++ mode2 |= (0x2 << 5); ++ break; ++ case 256: ++ mode2 |= (0x1 << 5); ++ break; ++ } ++ ++ /* bfs */ ++ if (bfs == 64) ++ mode |= 0x4; ++ ++ /* set rate */ ++ ak4535_write(codec, AK4535_MODE1, mode); ++ ak4535_write(codec, AK4535_MODE2, mode2); ++ ++ return 0; ++} ++ ++static unsigned int ak4535_config_sysclk(struct snd_soc_codec_dai *dai, ++ struct snd_soc_clock_info *info, unsigned int clk) ++{ ++ if (info->fs != 256) ++ return 0; ++ ++ /* we only support 256 FS atm */ ++ if (info->rate * info->fs == clk) { ++ dai->mclk = clk; ++ return clk; ++ } ++ ++ return 0; ++} ++ ++static int ak4535_mute(struct snd_soc_codec *codec, ++ struct snd_soc_codec_dai *dai, int mute) ++{ ++ u16 mute_reg = ak4535_read_reg_cache(codec, AK4535_DAC) & 0xffdf; ++ if (mute) ++ ak4535_write(codec, AK4535_DAC, mute_reg); ++ else ++ ak4535_write(codec, AK4535_DAC, mute_reg | 0x20); ++ return 0; ++} ++ ++static int ak4535_dapm_event(struct snd_soc_codec *codec, int event) ++{ ++ switch (event) { ++ case SNDRV_CTL_POWER_D0: /* full On */ ++ /* vref/mid, clk and osc on, dac unmute, active */ ++ case SNDRV_CTL_POWER_D1: /* partial On */ ++ case SNDRV_CTL_POWER_D2: /* partial On */ ++ break; ++ case SNDRV_CTL_POWER_D3hot: /* Off, with power */ ++ /* everything off except vref/vmid, dac mute, inactive */ ++ ak4535_write(codec, AK4535_PM1, 0x80); ++ ak4535_write(codec, AK4535_PM2, 0x0); ++ break; ++ case SNDRV_CTL_POWER_D3cold: /* Off, without power */ ++ /* everything off, inactive */ ++ ak4535_write(codec, AK4535_PM1, 0x0); ++ ak4535_write(codec, AK4535_PM2, 0x80); ++ break; ++ } ++ codec->dapm_state = event; ++ return 0; ++} ++ ++struct snd_soc_codec_dai ak4535_dai = { ++ .name = "AK4535", ++ .playback = { ++ .stream_name = "Playback", ++ .channels_min = 1, ++ .channels_max = 2, ++ }, ++ .capture = { ++ .stream_name = "Capture", ++ .channels_min = 1, ++ .channels_max = 2, ++ }, ++ .config_sysclk = ak4535_config_sysclk, ++ .digital_mute = ak4535_mute, ++ .ops = { ++ .prepare = ak4535_pcm_prepare, ++ }, ++ .caps = { ++ .num_modes = ARRAY_SIZE(ak4535_modes), ++ .mode = ak4535_modes, ++ }, ++}; ++EXPORT_SYMBOL_GPL(ak4535_dai); ++ ++static int ak4535_suspend(struct platform_device *pdev, pm_message_t state) ++{ ++ struct snd_soc_device *socdev = platform_get_drvdata(pdev); ++ struct snd_soc_codec *codec = socdev->codec; ++ ++ ak4535_dapm_event(codec, SNDRV_CTL_POWER_D3cold); ++ return 0; ++} ++ ++static int ak4535_resume(struct platform_device *pdev) ++{ ++ struct snd_soc_device *socdev = platform_get_drvdata(pdev); ++ struct snd_soc_codec *codec = socdev->codec; ++ int i; ++ u8 data[2]; ++ u16 *cache = codec->reg_cache; ++ ++ /* Sync reg_cache with the hardware */ ++ for (i = 0; i < ARRAY_SIZE(ak4535_reg); i++) { ++ data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); ++ data[1] = cache[i] & 0x00ff; ++ codec->hw_write(codec->control_data, data, 2); ++ } ++ ak4535_dapm_event(codec, SNDRV_CTL_POWER_D3hot); ++ ak4535_dapm_event(codec, codec->suspend_dapm_state); ++ return 0; ++} ++ ++/* ++ * initialise the AK4535 driver ++ * register the mixer and dsp interfaces with the kernel ++ */ ++static int ak4535_init(struct snd_soc_device *socdev) ++{ ++ struct snd_soc_codec *codec = socdev->codec; ++ int ret = 0; ++ ++ codec->name = "AK4535"; ++ codec->owner = THIS_MODULE; ++ codec->read = ak4535_read_reg_cache; ++ codec->write = ak4535_write; ++ codec->dapm_event = ak4535_dapm_event; ++ codec->dai = &ak4535_dai; ++ codec->num_dai = 1; ++ codec->reg_cache_size = ARRAY_SIZE(ak4535_reg); ++ codec->reg_cache = ++ kzalloc(sizeof(u16) * ARRAY_SIZE(ak4535_reg), GFP_KERNEL); ++ if (codec->reg_cache == NULL) ++ return -ENOMEM; ++ memcpy(codec->reg_cache, ak4535_reg, ++ sizeof(u16) * ARRAY_SIZE(ak4535_reg)); ++ codec->reg_cache_size = sizeof(u16) * ARRAY_SIZE(ak4535_reg); ++ ++ /* register pcms */ ++ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); ++ if (ret < 0) { ++ kfree(codec->reg_cache); ++ return ret; ++ } ++ ++ /* power on device */ ++ ak4535_dapm_event(codec, SNDRV_CTL_POWER_D3hot); ++ ++ ak4535_add_controls(codec); ++ ak4535_add_widgets(codec); ++ ret = snd_soc_register_card(socdev); ++ if (ret < 0) { ++ snd_soc_free_pcms(socdev); ++ snd_soc_dapm_free(socdev); ++ } ++ ++ return ret; ++} ++ ++static struct snd_soc_device *ak4535_socdev; ++ ++#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) ++ ++#define I2C_DRIVERID_AK4535 0xfefe /* liam - need a proper id */ ++ ++static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END }; ++ ++/* Magic definition of all other variables and things */ ++I2C_CLIENT_INSMOD; ++ ++static struct i2c_driver ak4535_i2c_driver; ++static struct i2c_client client_template; ++ ++/* If the i2c layer weren't so broken, we could pass this kind of data ++ around */ ++static int ak4535_codec_probe(struct i2c_adapter *adap, int addr, int kind) ++{ ++ struct snd_soc_device *socdev = ak4535_socdev; ++ struct ak4535_setup_data *setup = socdev->codec_data; ++ struct snd_soc_codec *codec = socdev->codec; ++ struct i2c_client *i2c; ++ int ret; ++ ++ if (addr != setup->i2c_address) ++ return -ENODEV; ++ ++ client_template.adapter = adap; ++ client_template.addr = addr; ++ ++ i2c = kzalloc(sizeof(struct i2c_client), GFP_KERNEL); ++ if (i2c == NULL){ ++ kfree(codec); ++ return -ENOMEM; ++ } ++ memcpy(i2c, &client_template, sizeof(struct i2c_client)); ++ i2c_set_clientdata(i2c, codec); ++ codec->control_data = i2c; ++ ++ ret = i2c_attach_client(i2c); ++ if (ret < 0) { ++ printk(KERN_ERR "failed to attach codec at addr %x\n", addr); ++ goto err; ++ } ++ ++ ret = ak4535_init(socdev); ++ if (ret < 0) { ++ printk(KERN_ERR "failed to initialise AK4535\n"); ++ goto err; ++ } ++ return ret; ++ ++err: ++ kfree(codec); ++ kfree(i2c); ++ return ret; ++} ++ ++static int ak4535_i2c_detach(struct i2c_client *client) ++{ ++ struct snd_soc_codec* codec = i2c_get_clientdata(client); ++ i2c_detach_client(client); ++ kfree(codec->reg_cache); ++ kfree(client); ++ ++ return 0; ++} ++ ++static int ak4535_i2c_attach(struct i2c_adapter *adap) ++{ ++ return i2c_probe(adap, &addr_data, ak4535_codec_probe); ++} ++ ++/* corgi i2c codec control layer */ ++static struct i2c_driver ak4535_i2c_driver = { ++ .driver = { ++ .name = "AK4535 I2C Codec", ++ .owner = THIS_MODULE, ++ }, ++ .id = I2C_DRIVERID_AK4535, ++ .attach_adapter = ak4535_i2c_attach, ++ .detach_client = ak4535_i2c_detach, ++ .command = NULL, ++}; ++ ++static struct i2c_client client_template = { ++ .name = "AK4535", ++ .driver = &ak4535_i2c_driver, ++}; ++#endif ++ ++static int ak4535_probe(struct platform_device *pdev) ++{ ++ struct snd_soc_device *socdev = platform_get_drvdata(pdev); ++ struct ak4535_setup_data *setup; ++ struct snd_soc_codec* codec; ++ int ret = 0; ++ ++ printk(KERN_INFO "AK4535 Audio Codec %s", AK4535_VERSION); ++ ++ setup = socdev->codec_data; ++ codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); ++ if (codec == NULL) ++ return -ENOMEM; ++ ++ socdev->codec = codec; ++ mutex_init(&codec->mutex); ++ INIT_LIST_HEAD(&codec->dapm_widgets); ++ INIT_LIST_HEAD(&codec->dapm_paths); ++ ++ ak4535_socdev = socdev; ++#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) ++ if (setup->i2c_address) { ++ normal_i2c[0] = setup->i2c_address; ++ codec->hw_write = (hw_write_t)i2c_master_send; ++ ret = i2c_add_driver(&ak4535_i2c_driver); ++ if (ret != 0) ++ printk(KERN_ERR "can't add i2c driver"); ++ } ++#else ++ /* Add other interfaces here */ ++#endif ++ return ret; ++} ++ ++/* power down chip */ ++static int ak4535_remove(struct platform_device *pdev) ++{ ++ struct snd_soc_device *socdev = platform_get_drvdata(pdev); ++ struct snd_soc_codec* codec = socdev->codec; ++ ++ if (codec->control_data) ++ ak4535_dapm_event(codec, SNDRV_CTL_POWER_D3cold); ++ ++ snd_soc_free_pcms(socdev); ++ snd_soc_dapm_free(socdev); ++#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) ++ i2c_del_driver(&ak4535_i2c_driver); ++#endif ++ kfree(codec); ++ ++ return 0; ++} ++ ++struct snd_soc_codec_device soc_codec_dev_ak4535 = { ++ .probe = ak4535_probe, ++ .remove = ak4535_remove, ++ .suspend = ak4535_suspend, ++ .resume = ak4535_resume, ++}; ++ ++EXPORT_SYMBOL_GPL(soc_codec_dev_ak4535); ++ ++MODULE_DESCRIPTION("Soc AK4535 driver"); ++MODULE_AUTHOR("Richard Purdie"); ++MODULE_LICENSE("GPL"); +Index: linux-2.6-pxa-new/sound/soc/codecs/ak4535.h +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/sound/soc/codecs/ak4535.h +@@ -0,0 +1,46 @@ ++/* ++ * ak4535.h -- AK4535 Soc Audio driver ++ * ++ * Copyright 2005 Openedhand Ltd. ++ * ++ * Author: Richard Purdie <richard@openedhand.com> ++ * ++ * Based on wm8753.h ++ * ++ * This program is free software; you can redistribute it and/or modify ++ * it under the terms of the GNU General Public License version 2 as ++ * published by the Free Software Foundation. ++ */ ++ ++#ifndef _AK4535_H ++#define _AK4535_H ++ ++/* AK4535 register space */ ++ ++#define AK4535_PM1 0x0 ++#define AK4535_PM2 0x1 ++#define AK4535_SIG1 0x2 ++#define AK4535_SIG2 0x3 ++#define AK4535_MODE1 0x4 ++#define AK4535_MODE2 0x5 ++#define AK4535_DAC 0x6 ++#define AK4535_MIC 0x7 ++#define AK4535_TIMER 0x8 ++#define AK4535_ALC1 0x9 ++#define AK4535_ALC2 0xa ++#define AK4535_PGA 0xb ++#define AK4535_LATT 0xc ++#define AK4535_RATT 0xd ++#define AK4535_VOL 0xe ++#define AK4535_STATUS 0xf ++ ++#define AK4535_CACHEREGNUM 0x10 ++ ++struct ak4535_setup_data { ++ unsigned short i2c_address; ++}; ++ ++extern struct snd_soc_codec_dai ak4535_dai; ++extern struct snd_soc_codec_device soc_codec_dev_ak4535; ++ ++#endif +Index: linux-2.6-pxa-new/sound/soc/codecs/uda1380.c +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/sound/soc/codecs/uda1380.c +@@ -0,0 +1,582 @@ ++/* ++ * uda1380.c - Philips UDA1380 ALSA SoC audio driver ++ * ++ * This program is free software; you can redistribute it and/or modify ++ * it under the terms of the GNU General Public License version 2 as ++ * published by the Free Software Foundation. ++ * ++ * Modified by Richard Purdie <richard@openedhand.com> to fit into SoC ++ * codec model. ++ * ++ * Copyright (c) 2005 Giorgio Padrin <giorgio@mandarinlogiq.org> ++ * Copyright 2005 Openedhand Ltd. ++ */ ++ ++#include <linux/module.h> ++#include <linux/init.h> ++#include <linux/types.h> ++#include <linux/string.h> ++#include <linux/slab.h> ++#include <linux/errno.h> ++#include <linux/ioctl.h> ++#include <linux/delay.h> ++#include <linux/i2c.h> ++#include <sound/driver.h> ++#include <sound/core.h> ++#include <sound/control.h> ++#include <sound/initval.h> ++#include <sound/info.h> ++#include <sound/soc.h> ++#include <sound/soc-dapm.h> ++ ++#include "uda1380.h" ++ ++#define UDA1380_VERSION "0.4" ++ ++/* ++ * uda1380 register cache ++ */ ++static const u16 uda1380_reg[UDA1380_CACHEREGNUM] = { ++ 0x0502, 0x0000, 0x0000, 0x3f3f, ++ 0x0202, 0x0000, 0x0000, 0x0000, ++ 0x0000, 0x0000, 0x0000, 0x0000, ++ 0x0000, 0x0000, 0x0000, 0x0000, ++ 0x0000, 0xff00, 0x0000, 0x4800, ++ 0x0000, 0x0000, 0x0000, 0x0000, ++ 0x0000, 0x0000, 0x0000, 0x0000, ++ 0x0000, 0x0000, 0x0000, 0x0000, ++ 0x0000, 0x8000, 0x0002, 0x0000, ++}; ++ ++#define UDA1380_DAIFMT \ ++ (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_CBS_CFS | \ ++ SND_SOC_DAIFMT_NB_NF) ++ ++#define UDA1380_DIR \ ++ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE) ++ ++#define UDA1380_RATES \ ++ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ ++ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ ++ SNDRV_PCM_RATE_48000) ++ ++static struct snd_soc_dai_mode uda1380_modes[] = { ++ /* slave rates capture & playback */ ++ { ++ .fmt = UDA1380_DAIFMT, ++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, ++ .pcmrate = UDA1380_RATES, ++ .pcmdir = UDA1380_DIR, ++ .flags = SND_SOC_DAI_BFS_RATE, ++ .fs = 256, ++ .bfs = 64, ++ }, ++ ++ /* slave rates playback */ ++ { ++ .fmt = UDA1380_DAIFMT, ++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, ++ .pcmrate = SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000, ++ .pcmdir = SND_SOC_DAIDIR_PLAYBACK, ++ .flags = SND_SOC_DAI_BFS_RATE, ++ .fs = 256, ++ .bfs = 64, ++ }, ++}; ++ ++/* ++ * read uda1380 register cache ++ */ ++static inline unsigned int uda1380_read_reg_cache(struct snd_soc_codec *codec, ++ unsigned int reg) ++{ ++ u16 *cache = codec->reg_cache; ++ if (reg == UDA1380_RESET) ++ return 0; ++ if (reg >= UDA1380_CACHEREGNUM) ++ return -1; ++ return cache[reg]; ++} ++ ++/* ++ * write uda1380 register cache ++ */ ++static inline void uda1380_write_reg_cache(struct snd_soc_codec *codec, ++ u16 reg, unsigned int value) ++{ ++ u16 *cache = codec->reg_cache; ++ if (reg >= UDA1380_CACHEREGNUM) ++ return; ++ cache[reg] = value; ++} ++ ++/* ++ * write to the UDA1380 register space ++ */ ++static int uda1380_write(struct snd_soc_codec *codec, unsigned int reg, ++ unsigned int value) ++{ ++ u8 data[3]; ++ ++ /* data is ++ * data[0] is register offset ++ * data[1] is MS byte ++ * data[2] is LS byte ++ */ ++ data[0] = reg; ++ data[1] = (value & 0xff00) >> 8; ++ data[2] = value & 0x00ff; ++ ++ uda1380_write_reg_cache (codec, reg, value); ++ if (codec->hw_write(codec->control_data, data, 3) == 3) ++ return 0; ++ else ++ return -EIO; ++} ++ ++#define uda1380_reset(c) uda1380_write(c, UDA1380_RESET, 0) ++ ++/* declarations of ALSA reg_elem_REAL controls */ ++static const char *uda1380_deemp[] = {"None", "32kHz", "44.1kHz", "48kHz", ++ "96kHz"}; ++static const char *uda1380_input_sel[] = {"Line", "Mic"}; ++ ++static const struct soc_enum uda1380_enum[] = { ++ SOC_ENUM_DOUBLE(UDA1380_DEEMP, 0, 8, 5, uda1380_deemp), ++ SOC_ENUM_SINGLE(UDA1380_ADC, 3, 2, uda1380_input_sel), ++}; ++ ++static const struct snd_kcontrol_new uda1380_snd_controls[] = { ++ SOC_DOUBLE("Playback Volume", UDA1380_MVOL, 0, 8, 127, 0), ++ SOC_DOUBLE("Treble Volume", UDA1380_MODE, 4, 12, 3, 0), ++ SOC_DOUBLE("Bass Volume", UDA1380_MODE, 0, 8, 15, 0), ++ SOC_ENUM("Playback De-emphasis", uda1380_enum[0]), ++ SOC_DOUBLE("Capture Volume", UDA1380_DEC, 0, 8, 127, 0), ++ SOC_DOUBLE("Line Capture Volume", UDA1380_PGA, 0, 8, 15, 0), ++ SOC_SINGLE("Mic Capture Volume", UDA1380_PGA, 8, 11, 0), ++ SOC_DOUBLE("Playback Switch", UDA1380_DEEMP, 3, 11, 1, 0), ++ SOC_SINGLE("Capture Switch", UDA1380_PGA, 15, 1, 0), ++ SOC_SINGLE("AGC Timing", UDA1380_AGC, 8, 7, 0), ++ SOC_SINGLE("AGC Target level", UDA1380_AGC, 2, 3, 1), ++ SOC_SINGLE("AGC Switch", UDA1380_AGC, 0, 1, 0), ++}; ++ ++/* add non dapm controls */ ++static int uda1380_add_controls(struct snd_soc_codec *codec) ++{ ++ int err, i; ++ ++ for (i = 0; i < ARRAY_SIZE(uda1380_snd_controls); i++) { ++ err = snd_ctl_add(codec->card, ++ snd_soc_cnew(&uda1380_snd_controls[i],codec, NULL)); ++ if (err < 0) ++ return err; ++ } ++ ++ return 0; ++} ++ ++/* Input mux */ ++static const struct snd_kcontrol_new uda1380_input_mux_control = ++ SOC_DAPM_ENUM("Input Select", uda1380_enum[1]); ++ ++static const struct snd_soc_dapm_widget uda1380_dapm_widgets[] = { ++ SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0, ++ &uda1380_input_mux_control), ++ SND_SOC_DAPM_PGA("Left PGA", UDA1380_PM, 3, 0, NULL, 0), ++ SND_SOC_DAPM_PGA("Right PGA", UDA1380_PM, 1, 0, NULL, 0), ++ SND_SOC_DAPM_PGA("Mic LNA", UDA1380_PM, 4, 0, NULL, 0), ++ SND_SOC_DAPM_ADC("Left ADC", "Left Capture", UDA1380_PM, 2, 0), ++ SND_SOC_DAPM_ADC("Right ADC", "Right Capture", UDA1380_PM, 0, 0), ++ SND_SOC_DAPM_INPUT("VINM"), ++ SND_SOC_DAPM_INPUT("VINL"), ++ SND_SOC_DAPM_INPUT("VINR"), ++ SND_SOC_DAPM_MIXER("Analog Mixer", UDA1380_PM, 6, 0, NULL, 0), ++ SND_SOC_DAPM_OUTPUT("VOUTLHP"), ++ SND_SOC_DAPM_OUTPUT("VOUTRHP"), ++ SND_SOC_DAPM_OUTPUT("VOUTL"), ++ SND_SOC_DAPM_OUTPUT("VOUTR"), ++ SND_SOC_DAPM_DAC("DAC", "Playback", UDA1380_PM, 10, 0), ++ SND_SOC_DAPM_PGA("HeadPhone Driver", UDA1380_PM, 13, 0, NULL, 0), ++}; ++ ++static const char *audio_map[][3] = { ++ ++ /* analog mixer setup is different from diagram for dapm */ ++ {"HeadPhone Driver", NULL, "Analog Mixer"}, ++ {"VOUTR", NULL, "Analog Mixer"}, ++ {"VOUTL", NULL, "Analog Mixer"}, ++ {"Analog Mixer", NULL, "VINR"}, ++ {"Analog Mixer", NULL, "VINL"}, ++ {"Analog Mixer", NULL, "DAC"}, ++ ++ /* headphone driver */ ++ {"VOUTLHP", NULL, "HeadPhone Driver"}, ++ {"VOUTRHP", NULL, "HeadPhone Driver"}, ++ ++ /* input mux */ ++ {"Left ADC", NULL, "Input Mux"}, ++ {"Input Mux", "Mic", "Mic LNA"}, ++ {"Input Mux", "Line", "Left PGA"}, ++ ++ /* right input */ ++ {"Right ADC", NULL, "Right PGA"}, ++ ++ /* inputs */ ++ {"Mic LNA", NULL, "VINM"}, ++ {"Left PGA", NULL, "VINL"}, ++ {"Right PGA", NULL, "VINR"}, ++ ++ /* terminator */ ++ {NULL, NULL, NULL}, ++}; ++ ++static int uda1380_add_widgets(struct snd_soc_codec *codec) ++{ ++ int i; ++ ++ for(i = 0; i < ARRAY_SIZE(uda1380_dapm_widgets); i++) { ++ snd_soc_dapm_new_control(codec, &uda1380_dapm_widgets[i]); ++ } ++ ++ /* set up audio path audio_mapnects */ ++ for(i = 0; audio_map[i][0] != NULL; i++) { ++ snd_soc_dapm_connect_input(codec, audio_map[i][0], ++ audio_map[i][1], audio_map[i][2]); ++ } ++ ++ snd_soc_dapm_new_widgets(codec); ++ return 0; ++} ++ ++static int uda1380_pcm_prepare(struct snd_pcm_substream *substream) ++{ ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ struct snd_soc_device *socdev = rtd->socdev; ++ struct snd_soc_codec *codec = socdev->codec; ++ u16 clk = uda1380_read_reg_cache(codec, UDA1380_CLK); ++ ++ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ++ uda1380_write(codec, UDA1380_CLK, R00_EN_DAC | R00_EN_INT | clk); ++ else ++ uda1380_write(codec, UDA1380_CLK, R00_EN_ADC | R00_EN_DEC | clk); ++ ++ return 0; ++} ++ ++static void uda1380_pcm_shutdown(struct snd_pcm_substream *substream) ++{ ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ struct snd_soc_device *socdev = rtd->socdev; ++ struct snd_soc_codec *codec = socdev->codec; ++ u16 clk = uda1380_read_reg_cache(codec, UDA1380_CLK); ++ ++ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ++ uda1380_write(codec, UDA1380_CLK, ~(R00_EN_DAC | R00_EN_INT) & clk); ++ else ++ uda1380_write(codec, UDA1380_CLK, ~(R00_EN_ADC | R00_EN_DEC) & clk); ++} ++ ++static unsigned int uda1380_config_sysclk(struct snd_soc_codec_dai *dai, ++ struct snd_soc_clock_info *info, unsigned int clk) ++{ ++ if(info->fs != 256) ++ return 0; ++ ++ /* we only support 256 FS atm */ ++ if(info->rate * info->fs == clk) { ++ dai->mclk = clk; ++ return clk; ++ } ++ ++ return 0; ++} ++ ++static int uda1380_mute(struct snd_soc_codec *codec, ++ struct snd_soc_codec_dai *dai, int mute) ++{ ++ u16 mute_reg = uda1380_read_reg_cache(codec, UDA1380_DEEMP) & 0xbfff; ++ if(mute) ++ uda1380_write(codec, UDA1380_DEEMP, mute_reg | 0x4000); ++ else ++ uda1380_write(codec, UDA1380_DEEMP, mute_reg); ++ return 0; ++} ++ ++static int uda1380_dapm_event(struct snd_soc_codec *codec, int event) ++{ ++ switch (event) { ++ case SNDRV_CTL_POWER_D0: /* full On */ ++ case SNDRV_CTL_POWER_D1: /* partial On */ ++ case SNDRV_CTL_POWER_D2: /* partial On */ ++ case SNDRV_CTL_POWER_D3hot: /* Off, with power */ ++ /* everything off except internal bias */ ++ uda1380_write(codec, UDA1380_PM, R02_PON_BIAS); ++ break; ++ case SNDRV_CTL_POWER_D3cold: /* Off, without power */ ++ /* everything off, inactive */ ++ uda1380_write(codec, UDA1380_PM, 0x0); ++ break; ++ } ++ codec->dapm_state = event; ++ return 0; ++} ++ ++struct snd_soc_codec_dai uda1380_dai = { ++ .name = "UDA1380", ++ .playback = { ++ .stream_name = "Playback", ++ .channels_min = 1, ++ .channels_max = 2, ++ }, ++ .capture = { ++ .stream_name = "Capture", ++ .channels_min = 1, ++ .channels_max = 2, ++ }, ++ .config_sysclk = uda1380_config_sysclk, ++ .digital_mute = uda1380_mute, ++ .ops = { ++ .prepare = uda1380_pcm_prepare, ++ .shutdown = uda1380_pcm_shutdown, ++ }, ++ .caps = { ++ .num_modes = ARRAY_SIZE(uda1380_modes), ++ .mode = uda1380_modes, ++ }, ++}; ++EXPORT_SYMBOL_GPL(uda1380_dai); ++ ++static int uda1380_suspend(struct platform_device *pdev, pm_message_t state) ++{ ++ struct snd_soc_device *socdev = platform_get_drvdata(pdev); ++ struct snd_soc_codec *codec = socdev->codec; ++ ++ uda1380_dapm_event(codec, SNDRV_CTL_POWER_D3cold); ++ return 0; ++} ++ ++static int uda1380_resume(struct platform_device *pdev) ++{ ++ struct snd_soc_device *socdev = platform_get_drvdata(pdev); ++ struct snd_soc_codec *codec = socdev->codec; ++ int i; ++ u8 data[2]; ++ u16 *cache = codec->reg_cache; ++ ++ /* Sync reg_cache with the hardware */ ++ for (i = 0; i < ARRAY_SIZE(uda1380_reg); i++) { ++ data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); ++ data[1] = cache[i] & 0x00ff; ++ codec->hw_write(codec->control_data, data, 2); ++ } ++ uda1380_dapm_event(codec, SNDRV_CTL_POWER_D3hot); ++ uda1380_dapm_event(codec, codec->suspend_dapm_state); ++ return 0; ++} ++ ++/* ++ * initialise the UDA1380 driver ++ * register the mixer and dsp interfaces with the kernel ++ */ ++static int uda1380_init(struct snd_soc_device *socdev) ++{ ++ struct snd_soc_codec *codec = socdev->codec; ++ int ret = 0; ++ ++ codec->name = "UDA1380"; ++ codec->owner = THIS_MODULE; ++ codec->read = uda1380_read_reg_cache; ++ codec->write = uda1380_write; ++ codec->dapm_event = uda1380_dapm_event; ++ codec->dai = &uda1380_dai; ++ codec->num_dai = 1; ++ codec->reg_cache_size = ARRAY_SIZE(uda1380_reg); ++ codec->reg_cache = ++ kzalloc(sizeof(u16) * ARRAY_SIZE(uda1380_reg), GFP_KERNEL); ++ if (codec->reg_cache == NULL) ++ return -ENOMEM; ++ memcpy(codec->reg_cache, uda1380_reg, ++ sizeof(u16) * ARRAY_SIZE(uda1380_reg)); ++ codec->reg_cache_size = sizeof(u16) * ARRAY_SIZE(uda1380_reg); ++ uda1380_reset(codec); ++ ++ /* register pcms */ ++ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); ++ if(ret < 0) { ++ kfree(codec->reg_cache); ++ return ret; ++ } ++ ++ /* power on device */ ++ uda1380_dapm_event(codec, SNDRV_CTL_POWER_D3hot); ++ uda1380_write(codec, UDA1380_CLK, 0); ++ ++ /* uda1380 init */ ++ uda1380_add_controls(codec); ++ uda1380_add_widgets(codec); ++ ret = snd_soc_register_card(socdev); ++ if(ret < 0) { ++ snd_soc_free_pcms(socdev); ++ snd_soc_dapm_free(socdev); ++ } ++ ++ return ret; ++} ++ ++static struct snd_soc_device *uda1380_socdev; ++ ++#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) ++ ++#define I2C_DRIVERID_UDA1380 0xfefe /* liam - need a proper id */ ++ ++static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END }; ++ ++/* Magic definition of all other variables and things */ ++I2C_CLIENT_INSMOD; ++ ++static struct i2c_driver uda1380_i2c_driver; ++static struct i2c_client client_template; ++ ++/* If the i2c layer weren't so broken, we could pass this kind of data ++ around */ ++ ++static int uda1380_codec_probe(struct i2c_adapter *adap, int addr, int kind) ++{ ++ struct snd_soc_device *socdev = uda1380_socdev; ++ struct uda1380_setup_data *setup = socdev->codec_data; ++ struct snd_soc_codec *codec = socdev->codec; ++ struct i2c_client *i2c; ++ int ret; ++ ++ if (addr != setup->i2c_address) ++ return -ENODEV; ++ ++ client_template.adapter = adap; ++ client_template.addr = addr; ++ ++ i2c = kzalloc(sizeof(struct i2c_client), GFP_KERNEL); ++ if (i2c == NULL){ ++ kfree(codec); ++ return -ENOMEM; ++ } ++ memcpy(i2c, &client_template, sizeof(struct i2c_client)); ++ i2c_set_clientdata(i2c, codec); ++ codec->control_data = i2c; ++ ++ ret = i2c_attach_client(i2c); ++ if(ret < 0) { ++ printk(KERN_ERR "failed to attach codec at addr %x\n", addr); ++ goto err; ++ } ++ ++ ret = uda1380_init(socdev); ++ if(ret < 0) { ++ printk(KERN_ERR "failed to initialise UDA1380\n"); ++ goto err; ++ } ++ return ret; ++ ++err: ++ kfree(codec); ++ kfree(i2c); ++ return ret; ++} ++ ++static int uda1380_i2c_detach(struct i2c_client *client) ++{ ++ struct snd_soc_codec* codec = i2c_get_clientdata(client); ++ i2c_detach_client(client); ++ kfree(codec->reg_cache); ++ kfree(client); ++ return 0; ++} ++ ++static int uda1380_i2c_attach(struct i2c_adapter *adap) ++{ ++ return i2c_probe(adap, &addr_data, uda1380_codec_probe); ++} ++ ++/* corgi i2c codec control layer */ ++static struct i2c_driver uda1380_i2c_driver = { ++ .driver = { ++ .name = "UDA1380 I2C Codec", ++ .owner = THIS_MODULE, ++ }, ++ .id = I2C_DRIVERID_UDA1380, ++ .attach_adapter = uda1380_i2c_attach, ++ .detach_client = uda1380_i2c_detach, ++ .command = NULL, ++}; ++ ++static struct i2c_client client_template = { ++ .name = "UDA1380", ++ .driver = &uda1380_i2c_driver, ++}; ++#endif ++ ++static int uda1380_probe(struct platform_device *pdev) ++{ ++ struct snd_soc_device *socdev = platform_get_drvdata(pdev); ++ struct uda1380_setup_data *setup; ++ struct snd_soc_codec* codec; ++ int ret = 0; ++ ++ printk(KERN_INFO "UDA1380 Audio Codec %s", UDA1380_VERSION); ++ ++ setup = socdev->codec_data; ++ codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); ++ if (codec == NULL) ++ return -ENOMEM; ++ ++ socdev->codec = codec; ++ mutex_init(&codec->mutex); ++ INIT_LIST_HEAD(&codec->dapm_widgets); ++ INIT_LIST_HEAD(&codec->dapm_paths); ++ ++ uda1380_socdev = socdev; ++#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) ++ if (setup->i2c_address) { ++ normal_i2c[0] = setup->i2c_address; ++ codec->hw_write = (hw_write_t)i2c_master_send; ++ ret = i2c_add_driver(&uda1380_i2c_driver); ++ if (ret != 0) ++ printk(KERN_ERR "can't add i2c driver"); ++ } ++#else ++ /* Add other interfaces here */ ++#endif ++ return ret; ++} ++ ++/* power down chip */ ++static int uda1380_remove(struct platform_device *pdev) ++{ ++ struct snd_soc_device *socdev = platform_get_drvdata(pdev); ++ struct snd_soc_codec* codec = socdev->codec; ++ ++ if (codec->control_data) ++ uda1380_dapm_event(codec, SNDRV_CTL_POWER_D3cold); ++ ++ snd_soc_free_pcms(socdev); ++ snd_soc_dapm_free(socdev); ++#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) ++ i2c_del_driver(&uda1380_i2c_driver); ++#endif ++ kfree(codec); ++ ++ return 0; ++} ++ ++struct snd_soc_codec_device soc_codec_dev_uda1380 = { ++ .probe = uda1380_probe, ++ .remove = uda1380_remove, ++ .suspend = uda1380_suspend, ++ .resume = uda1380_resume, ++}; ++ ++EXPORT_SYMBOL_GPL(soc_codec_dev_uda1380); ++ ++MODULE_AUTHOR("Giorgio Padrin"); ++MODULE_DESCRIPTION("Audio support for codec Philips UDA1380"); ++MODULE_LICENSE("GPL"); +Index: linux-2.6-pxa-new/sound/soc/codecs/uda1380.h +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/sound/soc/codecs/uda1380.h +@@ -0,0 +1,56 @@ ++/* ++ * Audio support for Philips UDA1380 ++ * ++ * This program is free software; you can redistribute it and/or modify ++ * it under the terms of the GNU General Public License version 2 as ++ * published by the Free Software Foundation. ++ * ++ * Copyright (c) 2005 Giorgio Padrin <giorgio@mandarinlogiq.org> ++ */ ++ ++#define UDA1380_CLK 0x00 ++#define UDA1380_IFACE 0x01 ++#define UDA1380_PM 0x02 ++#define UDA1380_AMIX 0x03 ++#define UDA1380_HP 0x04 ++#define UDA1380_MVOL 0x10 ++#define UDA1380_MIXVOL 0x11 ++#define UDA1380_MODE 0x12 ++#define UDA1380_DEEMP 0x13 ++#define UDA1380_MIXER 0x14 ++#define UDA1380_INTSTAT 0x18 ++#define UDA1380_DEC 0x20 ++#define UDA1380_PGA 0x21 ++#define UDA1380_ADC 0x22 ++#define UDA1380_AGC 0x23 ++#define UDA1380_DECSTAT 0x28 ++#define UDA1380_RESET 0x7f ++ ++#define UDA1380_CACHEREGNUM 0x24 ++ ++/* Register flags */ ++#define R00_EN_ADC 0x0800 ++#define R00_EN_DEC 0x0400 ++#define R00_EN_DAC 0x0200 ++#define R00_EN_INT 0x0100 ++#define R02_PON_HP 0x2000 ++#define R02_PON_DAC 0x0400 ++#define R02_PON_BIAS 0x0100 ++#define R02_PON_LNA 0x0010 ++#define R02_PON_PGAL 0x0008 ++#define R02_PON_ADCL 0x0004 ++#define R02_PON_PGAR 0x0002 ++#define R02_PON_ADCR 0x0001 ++#define R13_MTM 0x4000 ++#define R21_MT_ADC 0x8000 ++#define R22_SEL_LNA 0x0008 ++#define R22_SEL_MIC 0x0004 ++#define R22_SKIP_DCFIL 0x0002 ++#define R23_AGC_EN 0x0001 ++ ++struct uda1380_setup_data { ++ unsigned short i2c_address; ++}; ++ ++extern struct snd_soc_codec_dai uda1380_dai; ++extern struct snd_soc_codec_device soc_codec_dev_uda1380; +Index: linux-2.6-pxa-new/sound/soc/codecs/wm8731.c +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/sound/soc/codecs/wm8731.c +@@ -0,0 +1,886 @@ ++/* ++ * wm8731.c -- WM8731 ALSA SoC Audio driver ++ * ++ * Copyright 2005 Openedhand Ltd. ++ * ++ * Author: Richard Purdie <richard@openedhand.com> ++ * ++ * Based on wm8753.c by Liam Girdwood ++ * ++ * This program is free software; you can redistribute it and/or modify ++ * it under the terms of the GNU General Public License version 2 as ++ * published by the Free Software Foundation. ++ */ ++ ++#include <linux/module.h> ++#include <linux/moduleparam.h> ++#include <linux/init.h> ++#include <linux/delay.h> ++#include <linux/pm.h> ++#include <linux/i2c.h> ++#include <linux/platform_device.h> ++#include <sound/driver.h> ++#include <sound/core.h> ++#include <sound/pcm.h> ++#include <sound/pcm_params.h> ++#include <sound/soc.h> ++#include <sound/soc-dapm.h> ++#include <sound/initval.h> ++ ++#include "wm8731.h" ++ ++#define AUDIO_NAME "wm8731" ++#define WM8731_VERSION "0.12" ++ ++/* ++ * Debug ++ */ ++ ++#define WM8731_DEBUG 0 ++ ++#ifdef WM8731_DEBUG ++#define dbg(format, arg...) \ ++ printk(KERN_DEBUG AUDIO_NAME ": " format "\n" , ## arg) ++#else ++#define dbg(format, arg...) do {} while (0) ++#endif ++#define err(format, arg...) \ ++ printk(KERN_ERR AUDIO_NAME ": " format "\n" , ## arg) ++#define info(format, arg...) \ ++ printk(KERN_INFO AUDIO_NAME ": " format "\n" , ## arg) ++#define warn(format, arg...) \ ++ printk(KERN_WARNING AUDIO_NAME ": " format "\n" , ## arg) ++ ++struct snd_soc_codec_device soc_codec_dev_wm8731; ++ ++/* ++ * wm8731 register cache ++ * We can't read the WM8731 register space when we are ++ * using 2 wire for device control, so we cache them instead. ++ * There is no point in caching the reset register ++ */ ++static const u16 wm8731_reg[WM8731_CACHEREGNUM] = { ++ 0x0097, 0x0097, 0x0079, 0x0079, ++ 0x000a, 0x0008, 0x009f, 0x000a, ++ 0x0000, 0x0000 ++}; ++ ++#define WM8731_DAIFMT \ ++ (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_RIGHT_J | \ ++ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_NB_IF | SND_SOC_DAIFMT_IB_NF | \ ++ SND_SOC_DAIFMT_IB_IF) ++ ++#define WM8731_DIR \ ++ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE) ++ ++#define WM8731_RATES \ ++ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ ++ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ ++ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) ++ ++#define WM8731_HIFI_BITS \ ++ (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ ++ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) ++ ++static struct snd_soc_dai_mode wm8731_modes[] = { ++ /* codec frame and clock master modes */ ++ /* 8k */ ++ { ++ .fmt = WM8731_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8731_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_8000, ++ .pcmdir = WM8731_DIR, ++ .flags = SND_SOC_DAI_BFS_RATE, ++ .fs = 1536, ++ .bfs = 64, ++ }, ++ { ++ .fmt = WM8731_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8731_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_8000, ++ .pcmdir = WM8731_DIR, ++ .flags = SND_SOC_DAI_BFS_RATE, ++ .fs = 2304, ++ .bfs = 64, ++ }, ++ { ++ .fmt = WM8731_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8731_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_8000, ++ .pcmdir = WM8731_DIR, ++ .flags = SND_SOC_DAI_BFS_RATE, ++ .fs = 1408, ++ .bfs = 64, ++ }, ++ { ++ .fmt = WM8731_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8731_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_8000, ++ .pcmdir = WM8731_DIR, ++ .flags = SND_SOC_DAI_BFS_RATE, ++ .fs = 2112, ++ .bfs = 64, ++ }, ++ ++ /* 32k */ ++ { ++ .fmt = WM8731_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8731_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_32000, ++ .pcmdir = WM8731_DIR, ++ .flags = SND_SOC_DAI_BFS_RATE, ++ .fs = 384, ++ .bfs = 64, ++ }, ++ { ++ .fmt = WM8731_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8731_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_32000, ++ .pcmdir = WM8731_DIR, ++ .flags = SND_SOC_DAI_BFS_RATE, ++ .fs = 576, ++ .bfs = 64, ++ }, ++ ++ /* 44.1k & 48k */ ++ { ++ .fmt = WM8731_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8731_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000, ++ .pcmdir = WM8731_DIR, ++ .flags = SND_SOC_DAI_BFS_RATE, ++ .fs = 256, ++ .bfs = 64, ++ }, ++ { ++ .fmt = WM8731_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8731_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000, ++ .pcmdir = WM8731_DIR, ++ .flags = SND_SOC_DAI_BFS_RATE, ++ .fs = 384, ++ .bfs = 64, ++ }, ++ ++ /* 88.2 & 96k */ ++ { ++ .fmt = WM8731_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8731_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000, ++ .pcmdir = WM8731_DIR, ++ .flags = SND_SOC_DAI_BFS_RATE, ++ .fs = 128, ++ .bfs = 64, ++ }, ++ { ++ .fmt = WM8731_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8731_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000, ++ .pcmdir = WM8731_DIR, ++ .flags = SND_SOC_DAI_BFS_RATE, ++ .fs = 192, ++ .bfs = 64, ++ }, ++ ++ /* USB codec frame and clock master modes */ ++ /* 8k */ ++ { ++ .fmt = WM8731_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8731_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_8000, ++ .pcmdir = WM8731_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 1500, ++ .bfs = SND_SOC_FSBD(1), ++ }, ++ ++ /* 44.1k */ ++ { ++ .fmt = WM8731_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8731_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_44100, ++ .pcmdir = WM8731_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 272, ++ .bfs = SND_SOC_FSBD(1), ++ }, ++ ++ /* 48k */ ++ { ++ .fmt = WM8731_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8731_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_48000, ++ .pcmdir = WM8731_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 250, ++ .bfs = SND_SOC_FSBD(1), ++ }, ++ ++ /* 88.2k */ ++ { ++ .fmt = WM8731_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8731_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_88200, ++ .pcmdir = WM8731_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 136, ++ .bfs = SND_SOC_FSBD(1), ++ }, ++ ++ /* 96k */ ++ { ++ .fmt = WM8731_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8731_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_96000, ++ .pcmdir = WM8731_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 125, ++ .bfs = SND_SOC_FSBD(1), ++ }, ++ ++ /* codec frame and clock slave modes */ ++ { ++ .fmt = WM8731_DAIFMT | SND_SOC_DAIFMT_CBS_CFS, ++ .pcmfmt = WM8731_HIFI_BITS, ++ .pcmrate = WM8731_RATES, ++ .pcmdir = WM8731_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = SND_SOC_FS_ALL, ++ .bfs = SND_SOC_FSB_ALL, ++ }, ++}; ++ ++/* ++ * read wm8731 register cache ++ */ ++static inline unsigned int wm8731_read_reg_cache(struct snd_soc_codec *codec, ++ unsigned int reg) ++{ ++ u16 *cache = codec->reg_cache; ++ if (reg == WM8731_RESET) ++ return 0; ++ if (reg >= WM8731_CACHEREGNUM) ++ return -1; ++ return cache[reg]; ++} ++ ++/* ++ * write wm8731 register cache ++ */ ++static inline void wm8731_write_reg_cache(struct snd_soc_codec *codec, ++ u16 reg, unsigned int value) ++{ ++ u16 *cache = codec->reg_cache; ++ if (reg >= WM8731_CACHEREGNUM) ++ return; ++ cache[reg] = value; ++} ++ ++/* ++ * write to the WM8731 register space ++ */ ++static int wm8731_write(struct snd_soc_codec *codec, unsigned int reg, ++ unsigned int value) ++{ ++ u8 data[2]; ++ ++ /* data is ++ * D15..D9 WM8731 register offset ++ * D8...D0 register data ++ */ ++ data[0] = (reg << 1) | ((value >> 8) & 0x0001); ++ data[1] = value & 0x00ff; ++ ++ wm8731_write_reg_cache (codec, reg, value); ++ if (codec->hw_write(codec->control_data, data, 2) == 2) ++ return 0; ++ else ++ return -EIO; ++} ++ ++#define wm8731_reset(c) wm8731_write(c, WM8731_RESET, 0) ++ ++static const char *wm8731_input_select[] = {"Line In", "Mic"}; ++static const char *wm8731_deemph[] = {"None", "32Khz", "44.1Khz", "48Khz"}; ++ ++static const struct soc_enum wm8731_enum[] = { ++ SOC_ENUM_SINGLE(WM8731_APANA, 2, 2, wm8731_input_select), ++ SOC_ENUM_SINGLE(WM8731_APDIGI, 1, 4, wm8731_deemph), ++}; ++ ++static const struct snd_kcontrol_new wm8731_snd_controls[] = { ++ ++SOC_DOUBLE_R("Master Playback Volume", WM8731_LOUT1V, WM8731_ROUT1V, ++ 0, 127, 0), ++SOC_DOUBLE_R("Master Playback ZC Switch", WM8731_LOUT1V, WM8731_ROUT1V, ++ 7, 1, 0), ++ ++SOC_DOUBLE_R("Capture Volume", WM8731_LINVOL, WM8731_RINVOL, 0, 31, 0), ++SOC_DOUBLE_R("Line Capture Switch", WM8731_LINVOL, WM8731_RINVOL, 7, 1, 1), ++ ++SOC_SINGLE("Mic Boost (+20dB)", WM8731_APANA, 0, 1, 0), ++SOC_SINGLE("Capture Mic Switch", WM8731_APANA, 1, 1, 1), ++ ++SOC_SINGLE("Sidetone Playback Volume", WM8731_APANA, 6, 3, 1), ++ ++SOC_SINGLE("ADC High Pass Filter Switch", WM8731_APDIGI, 0, 1, 1), ++SOC_SINGLE("Store DC Offset Switch", WM8731_APDIGI, 4, 1, 0), ++ ++SOC_ENUM("Playback De-emphasis", wm8731_enum[1]), ++}; ++ ++/* add non dapm controls */ ++static int wm8731_add_controls(struct snd_soc_codec *codec) ++{ ++ int err, i; ++ ++ for (i = 0; i < ARRAY_SIZE(wm8731_snd_controls); i++) { ++ if ((err = snd_ctl_add(codec->card, ++ snd_soc_cnew(&wm8731_snd_controls[i],codec, NULL))) < 0) ++ return err; ++ } ++ ++ return 0; ++} ++ ++/* Output Mixer */ ++static const struct snd_kcontrol_new wm8731_output_mixer_controls[] = { ++SOC_DAPM_SINGLE("Line Bypass Switch", WM8731_APANA, 3, 1, 0), ++SOC_DAPM_SINGLE("Mic Sidetone Switch", WM8731_APANA, 5, 1, 0), ++SOC_DAPM_SINGLE("HiFi Playback Switch", WM8731_APANA, 4, 1, 0), ++}; ++ ++/* Input mux */ ++static const struct snd_kcontrol_new wm8731_input_mux_controls = ++SOC_DAPM_ENUM("Input Select", wm8731_enum[0]); ++ ++static const struct snd_soc_dapm_widget wm8731_dapm_widgets[] = { ++SND_SOC_DAPM_MIXER("Output Mixer", WM8731_PWR, 4, 1, ++ &wm8731_output_mixer_controls[0], ++ ARRAY_SIZE(wm8731_output_mixer_controls)), ++SND_SOC_DAPM_DAC("DAC", "HiFi Playback", WM8731_PWR, 3, 1), ++SND_SOC_DAPM_OUTPUT("LOUT"), ++SND_SOC_DAPM_OUTPUT("LHPOUT"), ++SND_SOC_DAPM_OUTPUT("ROUT"), ++SND_SOC_DAPM_OUTPUT("RHPOUT"), ++SND_SOC_DAPM_ADC("ADC", "HiFi Capture", WM8731_PWR, 2, 1), ++SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0, &wm8731_input_mux_controls), ++SND_SOC_DAPM_PGA("Line Input", WM8731_PWR, 0, 1, NULL, 0), ++SND_SOC_DAPM_MICBIAS("Mic Bias", WM8731_PWR, 1, 1), ++SND_SOC_DAPM_INPUT("MICIN"), ++SND_SOC_DAPM_INPUT("RLINEIN"), ++SND_SOC_DAPM_INPUT("LLINEIN"), ++}; ++ ++static const char *intercon[][3] = { ++ /* output mixer */ ++ {"Output Mixer", "Line Bypass Switch", "Line Input"}, ++ {"Output Mixer", "HiFi Playback Switch", "DAC"}, ++ {"Output Mixer", "Mic Sidetone Switch", "Mic Bias"}, ++ ++ /* outputs */ ++ {"RHPOUT", NULL, "Output Mixer"}, ++ {"ROUT", NULL, "Output Mixer"}, ++ {"LHPOUT", NULL, "Output Mixer"}, ++ {"LOUT", NULL, "Output Mixer"}, ++ ++ /* input mux */ ++ {"Input Mux", "Line In", "Line Input"}, ++ {"Input Mux", "Mic", "Mic Bias"}, ++ {"ADC", NULL, "Input Mux"}, ++ ++ /* inputs */ ++ {"Line Input", NULL, "LLINEIN"}, ++ {"Line Input", NULL, "RLINEIN"}, ++ {"Mic Bias", NULL, "MICIN"}, ++ ++ /* terminator */ ++ {NULL, NULL, NULL}, ++}; ++ ++static int wm8731_add_widgets(struct snd_soc_codec *codec) ++{ ++ int i; ++ ++ for(i = 0; i < ARRAY_SIZE(wm8731_dapm_widgets); i++) { ++ snd_soc_dapm_new_control(codec, &wm8731_dapm_widgets[i]); ++ } ++ ++ /* set up audio path interconnects */ ++ for(i = 0; intercon[i][0] != NULL; i++) { ++ snd_soc_dapm_connect_input(codec, intercon[i][0], ++ intercon[i][1], intercon[i][2]); ++ } ++ ++ snd_soc_dapm_new_widgets(codec); ++ return 0; ++} ++ ++struct _coeff_div { ++ u32 mclk; ++ u32 rate; ++ u16 fs; ++ u8 sr:4; ++ u8 bosr:1; ++ u8 usb:1; ++}; ++ ++/* codec mclk clock divider coefficients */ ++static const struct _coeff_div coeff_div[] = { ++ /* 48k */ ++ {12288000, 48000, 256, 0x0, 0x0, 0x0}, ++ {18432000, 48000, 384, 0x0, 0x1, 0x0}, ++ {12000000, 48000, 250, 0x0, 0x0, 0x1}, ++ ++ /* 32k */ ++ {12288000, 32000, 384, 0x6, 0x0, 0x0}, ++ {18432000, 32000, 576, 0x6, 0x1, 0x0}, ++ ++ /* 8k */ ++ {12288000, 8000, 1536, 0x3, 0x0, 0x0}, ++ {18432000, 8000, 2304, 0x3, 0x1, 0x0}, ++ {11289600, 8000, 1408, 0xb, 0x0, 0x0}, ++ {16934400, 8000, 2112, 0xb, 0x1, 0x0}, ++ {12000000, 8000, 1500, 0x3, 0x0, 0x1}, ++ ++ /* 96k */ ++ {12288000, 96000, 128, 0x7, 0x0, 0x0}, ++ {18432000, 96000, 192, 0x7, 0x1, 0x0}, ++ {12000000, 96000, 125, 0x7, 0x0, 0x1}, ++ ++ /* 44.1k */ ++ {11289600, 44100, 256, 0x8, 0x0, 0x0}, ++ {16934400, 44100, 384, 0x8, 0x1, 0x0}, ++ {12000000, 44100, 272, 0x8, 0x1, 0x1}, ++ ++ /* 88.2k */ ++ {11289600, 88200, 128, 0xf, 0x0, 0x0}, ++ {16934400, 88200, 192, 0xf, 0x1, 0x0}, ++ {12000000, 88200, 136, 0xf, 0x1, 0x1}, ++}; ++ ++static inline int get_coeff(int mclk, int rate) ++{ ++ int i; ++ ++ for (i = 0; i < ARRAY_SIZE(coeff_div); i++) { ++ if (coeff_div[i].rate == rate && coeff_div[i].mclk == mclk) ++ return i; ++ } ++ return 0; ++} ++ ++/* WM8731 supports numerous clocks per sample rate */ ++static unsigned int wm8731_config_sysclk(struct snd_soc_codec_dai *dai, ++ struct snd_soc_clock_info *info, unsigned int clk) ++{ ++ dai->mclk = 0; ++ ++ /* check that the calculated FS and rate actually match a clock from ++ * the machine driver */ ++ if (info->fs * info->rate == clk) ++ dai->mclk = clk; ++ ++ return dai->mclk; ++} ++ ++static int wm8731_pcm_prepare(struct snd_pcm_substream *substream) ++{ ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ struct snd_soc_device *socdev = rtd->socdev; ++ struct snd_soc_codec *codec = socdev->codec; ++ u16 iface = 0, srate; ++ int i = get_coeff(rtd->codec_dai->mclk, ++ snd_soc_get_rate(rtd->codec_dai->dai_runtime.pcmrate)); ++ ++ /* set master/slave audio interface */ ++ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_CLOCK_MASK) { ++ case SND_SOC_DAIFMT_CBM_CFM: ++ iface |= 0x0040; ++ break; ++ case SND_SOC_DAIFMT_CBS_CFS: ++ break; ++ } ++ srate = (coeff_div[i].sr << 2) | ++ (coeff_div[i].bosr << 1) | coeff_div[i].usb; ++ wm8731_write(codec, WM8731_SRATE, srate); ++ ++ /* interface format */ ++ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_FORMAT_MASK) { ++ case SND_SOC_DAIFMT_I2S: ++ iface |= 0x0002; ++ break; ++ case SND_SOC_DAIFMT_RIGHT_J: ++ break; ++ case SND_SOC_DAIFMT_LEFT_J: ++ iface |= 0x0001; ++ break; ++ case SND_SOC_DAIFMT_DSP_A: ++ iface |= 0x0003; ++ break; ++ case SND_SOC_DAIFMT_DSP_B: ++ iface |= 0x0013; ++ break; ++ } ++ ++ /* bit size */ ++ switch (rtd->codec_dai->dai_runtime.pcmfmt) { ++ case SNDRV_PCM_FMTBIT_S16_LE: ++ break; ++ case SNDRV_PCM_FMTBIT_S20_3LE: ++ iface |= 0x0004; ++ break; ++ case SNDRV_PCM_FMTBIT_S24_LE: ++ iface |= 0x0008; ++ break; ++ case SNDRV_PCM_FMTBIT_S32_LE: ++ iface |= 0x000c; ++ break; ++ } ++ ++ /* clock inversion */ ++ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_INV_MASK) { ++ case SND_SOC_DAIFMT_NB_NF: ++ break; ++ case SND_SOC_DAIFMT_IB_IF: ++ iface |= 0x0090; ++ break; ++ case SND_SOC_DAIFMT_IB_NF: ++ iface |= 0x0080; ++ break; ++ case SND_SOC_DAIFMT_NB_IF: ++ iface |= 0x0010; ++ break; ++ } ++ ++ /* set iface */ ++ wm8731_write(codec, WM8731_IFACE, iface); ++ ++ /* set active */ ++ wm8731_write(codec, WM8731_ACTIVE, 0x0001); ++ return 0; ++} ++ ++static void wm8731_shutdown(struct snd_pcm_substream *substream) ++{ ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ struct snd_soc_device *socdev = rtd->socdev; ++ struct snd_soc_codec *codec = socdev->codec; ++ ++ /* deactivate */ ++ if (!codec->active) { ++ udelay(50); ++ wm8731_write(codec, WM8731_ACTIVE, 0x0); ++ } ++} ++ ++static int wm8731_mute(struct snd_soc_codec *codec, ++ struct snd_soc_codec_dai *dai, int mute) ++{ ++ u16 mute_reg = wm8731_read_reg_cache(codec, WM8731_APDIGI) & 0xfff7; ++ if (mute) ++ wm8731_write(codec, WM8731_APDIGI, mute_reg | 0x8); ++ else ++ wm8731_write(codec, WM8731_APDIGI, mute_reg); ++ return 0; ++} ++ ++static int wm8731_dapm_event(struct snd_soc_codec *codec, int event) ++{ ++ u16 reg = wm8731_read_reg_cache(codec, WM8731_PWR) & 0xff7f; ++ ++ switch (event) { ++ case SNDRV_CTL_POWER_D0: /* full On */ ++ /* vref/mid, osc on, dac unmute */ ++ wm8731_write(codec, WM8731_PWR, reg); ++ break; ++ case SNDRV_CTL_POWER_D1: /* partial On */ ++ case SNDRV_CTL_POWER_D2: /* partial On */ ++ break; ++ case SNDRV_CTL_POWER_D3hot: /* Off, with power */ ++ /* everything off except vref/vmid, */ ++ wm8731_write(codec, WM8731_PWR, reg | 0x0040); ++ break; ++ case SNDRV_CTL_POWER_D3cold: /* Off, without power */ ++ /* everything off, dac mute, inactive */ ++ wm8731_write(codec, WM8731_ACTIVE, 0x0); ++ wm8731_write(codec, WM8731_PWR, 0xffff); ++ break; ++ } ++ codec->dapm_state = event; ++ return 0; ++} ++ ++struct snd_soc_codec_dai wm8731_dai = { ++ .name = "WM8731", ++ .playback = { ++ .stream_name = "Playback", ++ .channels_min = 1, ++ .channels_max = 2, ++ }, ++ .capture = { ++ .stream_name = "Capture", ++ .channels_min = 1, ++ .channels_max = 2, ++ }, ++ .config_sysclk = wm8731_config_sysclk, ++ .digital_mute = wm8731_mute, ++ .ops = { ++ .prepare = wm8731_pcm_prepare, ++ .shutdown = wm8731_shutdown, ++ }, ++ .caps = { ++ .num_modes = ARRAY_SIZE(wm8731_modes), ++ .mode = wm8731_modes, ++ }, ++}; ++EXPORT_SYMBOL_GPL(wm8731_dai); ++ ++static int wm8731_suspend(struct platform_device *pdev, pm_message_t state) ++{ ++ struct snd_soc_device *socdev = platform_get_drvdata(pdev); ++ struct snd_soc_codec *codec = socdev->codec; ++ ++ wm8731_write(codec, WM8731_ACTIVE, 0x0); ++ wm8731_dapm_event(codec, SNDRV_CTL_POWER_D3cold); ++ return 0; ++} ++ ++static int wm8731_resume(struct platform_device *pdev) ++{ ++ struct snd_soc_device *socdev = platform_get_drvdata(pdev); ++ struct snd_soc_codec *codec = socdev->codec; ++ int i; ++ u8 data[2]; ++ u16 *cache = codec->reg_cache; ++ ++ /* Sync reg_cache with the hardware */ ++ for (i = 0; i < ARRAY_SIZE(wm8731_reg); i++) { ++ data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); ++ data[1] = cache[i] & 0x00ff; ++ codec->hw_write(codec->control_data, data, 2); ++ } ++ wm8731_dapm_event(codec, SNDRV_CTL_POWER_D3hot); ++ wm8731_dapm_event(codec, codec->suspend_dapm_state); ++ return 0; ++} ++ ++/* ++ * initialise the WM8731 driver ++ * register the mixer and dsp interfaces with the kernel ++ */ ++static int wm8731_init(struct snd_soc_device *socdev) ++{ ++ struct snd_soc_codec *codec = socdev->codec; ++ int reg, ret = 0; ++ ++ codec->name = "WM8731"; ++ codec->owner = THIS_MODULE; ++ codec->read = wm8731_read_reg_cache; ++ codec->write = wm8731_write; ++ codec->dapm_event = wm8731_dapm_event; ++ codec->dai = &wm8731_dai; ++ codec->num_dai = 1; ++ codec->reg_cache_size = ARRAY_SIZE(wm8731_reg); ++ ++ codec->reg_cache = ++ kzalloc(sizeof(u16) * ARRAY_SIZE(wm8731_reg), GFP_KERNEL); ++ if (codec->reg_cache == NULL) ++ return -ENOMEM; ++ memcpy(codec->reg_cache, ++ wm8731_reg, sizeof(u16) * ARRAY_SIZE(wm8731_reg)); ++ codec->reg_cache_size = sizeof(u16) * ARRAY_SIZE(wm8731_reg); ++ ++ wm8731_reset(codec); ++ ++ /* register pcms */ ++ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); ++ if (ret < 0) { ++ kfree(codec->reg_cache); ++ return ret; ++ } ++ ++ /* power on device */ ++ wm8731_dapm_event(codec, SNDRV_CTL_POWER_D3hot); ++ ++ /* set the update bits */ ++ reg = wm8731_read_reg_cache(codec, WM8731_LOUT1V); ++ wm8731_write(codec, WM8731_LOUT1V, reg | 0x0100); ++ reg = wm8731_read_reg_cache(codec, WM8731_ROUT1V); ++ wm8731_write(codec, WM8731_ROUT1V, reg | 0x0100); ++ reg = wm8731_read_reg_cache(codec, WM8731_LINVOL); ++ wm8731_write(codec, WM8731_LINVOL, reg | 0x0100); ++ reg = wm8731_read_reg_cache(codec, WM8731_RINVOL); ++ wm8731_write(codec, WM8731_RINVOL, reg | 0x0100); ++ ++ wm8731_add_controls(codec); ++ wm8731_add_widgets(codec); ++ ret = snd_soc_register_card(socdev); ++ if (ret < 0) { ++ snd_soc_free_pcms(socdev); ++ snd_soc_dapm_free(socdev); ++ } ++ ++ return ret; ++} ++ ++static struct snd_soc_device *wm8731_socdev; ++ ++#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) ++ ++/* ++ * WM8731 2 wire address is determined by GPIO5 ++ * state during powerup. ++ * low = 0x1a ++ * high = 0x1b ++ */ ++static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END }; ++ ++/* Magic definition of all other variables and things */ ++I2C_CLIENT_INSMOD; ++ ++static struct i2c_driver wm8731_i2c_driver; ++static struct i2c_client client_template; ++ ++/* If the i2c layer weren't so broken, we could pass this kind of data ++ around */ ++ ++static int wm8731_codec_probe(struct i2c_adapter *adap, int addr, int kind) ++{ ++ struct snd_soc_device *socdev = wm8731_socdev; ++ struct wm8731_setup_data *setup = socdev->codec_data; ++ struct snd_soc_codec *codec = socdev->codec; ++ struct i2c_client *i2c; ++ int ret; ++ ++ if (addr != setup->i2c_address) ++ return -ENODEV; ++ ++ client_template.adapter = adap; ++ client_template.addr = addr; ++ ++ i2c = kzalloc(sizeof(struct i2c_client), GFP_KERNEL); ++ if (i2c == NULL) { ++ kfree(codec); ++ return -ENOMEM; ++ } ++ memcpy(i2c, &client_template, sizeof(struct i2c_client)); ++ i2c_set_clientdata(i2c, codec); ++ codec->control_data = i2c; ++ ++ ret = i2c_attach_client(i2c); ++ if (ret < 0) { ++ err("failed to attach codec at addr %x\n", addr); ++ goto err; ++ } ++ ++ ret = wm8731_init(socdev); ++ if (ret < 0) { ++ err("failed to initialise WM8731\n"); ++ goto err; ++ } ++ return ret; ++ ++err: ++ kfree(codec); ++ kfree(i2c); ++ return ret; ++} ++ ++static int wm8731_i2c_detach(struct i2c_client *client) ++{ ++ struct snd_soc_codec* codec = i2c_get_clientdata(client); ++ i2c_detach_client(client); ++ kfree(codec->reg_cache); ++ kfree(client); ++ return 0; ++} ++ ++static int wm8731_i2c_attach(struct i2c_adapter *adap) ++{ ++ return i2c_probe(adap, &addr_data, wm8731_codec_probe); ++} ++ ++/* corgi i2c codec control layer */ ++static struct i2c_driver wm8731_i2c_driver = { ++ .driver = { ++ .name = "WM8731 I2C Codec", ++ .owner = THIS_MODULE, ++ }, ++ .id = I2C_DRIVERID_WM8731, ++ .attach_adapter = wm8731_i2c_attach, ++ .detach_client = wm8731_i2c_detach, ++ .command = NULL, ++}; ++ ++static struct i2c_client client_template = { ++ .name = "WM8731", ++ .driver = &wm8731_i2c_driver, ++}; ++#endif ++ ++static int wm8731_probe(struct platform_device *pdev) ++{ ++ struct snd_soc_device *socdev = platform_get_drvdata(pdev); ++ struct wm8731_setup_data *setup; ++ struct snd_soc_codec *codec; ++ int ret = 0; ++ ++ info("WM8731 Audio Codec %s", WM8731_VERSION); ++ ++ setup = socdev->codec_data; ++ codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); ++ if (codec == NULL) ++ return -ENOMEM; ++ ++ socdev->codec = codec; ++ mutex_init(&codec->mutex); ++ INIT_LIST_HEAD(&codec->dapm_widgets); ++ INIT_LIST_HEAD(&codec->dapm_paths); ++ ++ wm8731_socdev = socdev; ++#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) ++ if (setup->i2c_address) { ++ normal_i2c[0] = setup->i2c_address; ++ codec->hw_write = (hw_write_t)i2c_master_send; ++ ret = i2c_add_driver(&wm8731_i2c_driver); ++ if (ret != 0) ++ printk(KERN_ERR "can't add i2c driver"); ++ } ++#else ++ /* Add other interfaces here */ ++#endif ++ return ret; ++} ++ ++/* power down chip */ ++static int wm8731_remove(struct platform_device *pdev) ++{ ++ struct snd_soc_device *socdev = platform_get_drvdata(pdev); ++ struct snd_soc_codec *codec = socdev->codec; ++ ++ if (codec->control_data) ++ wm8731_dapm_event(codec, SNDRV_CTL_POWER_D3cold); ++ ++ snd_soc_free_pcms(socdev); ++ snd_soc_dapm_free(socdev); ++#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) ++ i2c_del_driver(&wm8731_i2c_driver); ++#endif ++ kfree(codec); ++ ++ return 0; ++} ++ ++struct snd_soc_codec_device soc_codec_dev_wm8731 = { ++ .probe = wm8731_probe, ++ .remove = wm8731_remove, ++ .suspend = wm8731_suspend, ++ .resume = wm8731_resume, ++}; ++ ++EXPORT_SYMBOL_GPL(soc_codec_dev_wm8731); ++ ++MODULE_DESCRIPTION("ASoC WM8731 driver"); ++MODULE_AUTHOR("Richard Purdie"); ++MODULE_LICENSE("GPL"); +Index: linux-2.6-pxa-new/sound/soc/codecs/wm8731.h +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/sound/soc/codecs/wm8731.h +@@ -0,0 +1,41 @@ ++/* ++ * wm8731.h -- WM8731 Soc Audio driver ++ * ++ * Copyright 2005 Openedhand Ltd. ++ * ++ * Author: Richard Purdie <richard@openedhand.com> ++ * ++ * Based on wm8753.h ++ * ++ * This program is free software; you can redistribute it and/or modify ++ * it under the terms of the GNU General Public License version 2 as ++ * published by the Free Software Foundation. ++ */ ++ ++#ifndef _WM8731_H ++#define _WM8731_H ++ ++/* WM8731 register space */ ++ ++#define WM8731_LINVOL 0x00 ++#define WM8731_RINVOL 0x01 ++#define WM8731_LOUT1V 0x02 ++#define WM8731_ROUT1V 0x03 ++#define WM8731_APANA 0x04 ++#define WM8731_APDIGI 0x05 ++#define WM8731_PWR 0x06 ++#define WM8731_IFACE 0x07 ++#define WM8731_SRATE 0x08 ++#define WM8731_ACTIVE 0x09 ++#define WM8731_RESET 0x0f ++ ++#define WM8731_CACHEREGNUM 10 ++ ++struct wm8731_setup_data { ++ unsigned short i2c_address; ++}; ++ ++extern struct snd_soc_codec_dai wm8731_dai; ++extern struct snd_soc_codec_device soc_codec_dev_wm8731; ++ ++#endif +Index: linux-2.6-pxa-new/sound/soc/codecs/wm8750.c +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/sound/soc/codecs/wm8750.c +@@ -0,0 +1,1282 @@ ++/* ++ * wm8750.c -- WM8750 ALSA SoC audio driver ++ * ++ * Copyright 2005 Openedhand Ltd. ++ * ++ * Author: Richard Purdie <richard@openedhand.com> ++ * ++ * Based on WM8753.c ++ * ++ * This program is free software; you can redistribute it and/or modify ++ * it under the terms of the GNU General Public License version 2 as ++ * published by the Free Software Foundation. ++ */ ++ ++#include <linux/module.h> ++#include <linux/moduleparam.h> ++#include <linux/init.h> ++#include <linux/delay.h> ++#include <linux/pm.h> ++#include <linux/i2c.h> ++#include <linux/platform_device.h> ++#include <sound/driver.h> ++#include <sound/core.h> ++#include <sound/pcm.h> ++#include <sound/pcm_params.h> ++#include <sound/soc.h> ++#include <sound/soc-dapm.h> ++#include <sound/initval.h> ++ ++#include "wm8750.h" ++ ++#define AUDIO_NAME "WM8750" ++#define WM8750_VERSION "0.11" ++ ++/* ++ * Debug ++ */ ++ ++#define WM8750_DEBUG 0 ++ ++#ifdef WM8750_DEBUG ++#define dbg(format, arg...) \ ++ printk(KERN_DEBUG AUDIO_NAME ": " format "\n" , ## arg) ++#else ++#define dbg(format, arg...) do {} while (0) ++#endif ++#define err(format, arg...) \ ++ printk(KERN_ERR AUDIO_NAME ": " format "\n" , ## arg) ++#define info(format, arg...) \ ++ printk(KERN_INFO AUDIO_NAME ": " format "\n" , ## arg) ++#define warn(format, arg...) \ ++ printk(KERN_WARNING AUDIO_NAME ": " format "\n" , ## arg) ++ ++static struct workqueue_struct *wm8750_workq = NULL; ++static struct work_struct wm8750_dapm_work; ++ ++/* ++ * wm8750 register cache ++ * We can't read the WM8750 register space when we ++ * are using 2 wire for device control, so we cache them instead. ++ */ ++static const u16 wm8750_reg[] = { ++ 0x0097, 0x0097, 0x0079, 0x0079, /* 0 */ ++ 0x0000, 0x0008, 0x0000, 0x000a, /* 4 */ ++ 0x0000, 0x0000, 0x00ff, 0x00ff, /* 8 */ ++ 0x000f, 0x000f, 0x0000, 0x0000, /* 12 */ ++ 0x0000, 0x007b, 0x0000, 0x0032, /* 16 */ ++ 0x0000, 0x00c3, 0x00c3, 0x00c0, /* 20 */ ++ 0x0000, 0x0000, 0x0000, 0x0000, /* 24 */ ++ 0x0000, 0x0000, 0x0000, 0x0000, /* 28 */ ++ 0x0000, 0x0000, 0x0050, 0x0050, /* 32 */ ++ 0x0050, 0x0050, 0x0050, 0x0050, /* 36 */ ++ 0x0079, 0x0079, 0x0079, /* 40 */ ++}; ++ ++#define WM8750_HIFI_DAIFMT \ ++ (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_RIGHT_J | \ ++ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_NB_IF | SND_SOC_DAIFMT_IB_NF | \ ++ SND_SOC_DAIFMT_IB_IF) ++ ++#define WM8750_DIR \ ++ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE) ++ ++#define WM8750_HIFI_FSB \ ++ (SND_SOC_FSBD(1) | SND_SOC_FSBD(2) | SND_SOC_FSBD(4) | \ ++ SND_SOC_FSBD(8) | SND_SOC_FSBD(16)) ++ ++#define WM8750_HIFI_RATES \ ++ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ ++ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ ++ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) ++ ++#define WM8750_HIFI_BITS \ ++ (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ ++ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) ++ ++static struct snd_soc_dai_mode wm8750_modes[] = { ++ /* common codec frame and clock master modes */ ++ /* 8k */ ++ { ++ .fmt = WM8750_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8750_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_8000, ++ .pcmdir = WM8750_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 1536, ++ .bfs = WM8750_HIFI_FSB, ++ }, ++ { ++ .fmt = WM8750_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8750_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_8000, ++ .pcmdir = WM8750_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 1408, ++ .bfs = WM8750_HIFI_FSB, ++ }, ++ { ++ .fmt = WM8750_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8750_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_8000, ++ .pcmdir = WM8750_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 2304, ++ .bfs = WM8750_HIFI_FSB, ++ }, ++ { ++ .fmt = WM8750_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8750_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_8000, ++ .pcmdir = WM8750_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 2112, ++ .bfs = WM8750_HIFI_FSB, ++ }, ++ { ++ .fmt = WM8750_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8750_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_8000, ++ .pcmdir = WM8750_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 1500, ++ .bfs = WM8750_HIFI_FSB, ++ }, ++ ++ /* 11.025k */ ++ { ++ .fmt = WM8750_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8750_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_11025, ++ .pcmdir = WM8750_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 1024, ++ .bfs = WM8750_HIFI_FSB, ++ }, ++ { ++ .fmt = WM8750_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8750_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_11025, ++ .pcmdir = WM8750_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 1536, ++ .bfs = WM8750_HIFI_FSB, ++ }, ++ { ++ .fmt = WM8750_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8750_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_11025, ++ .pcmdir = WM8750_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 1088, ++ .bfs = WM8750_HIFI_FSB, ++ }, ++ ++ /* 16k */ ++ { ++ .fmt = WM8750_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8750_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_16000, ++ .pcmdir = WM8750_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 768, ++ .bfs = WM8750_HIFI_FSB, ++ }, ++ { ++ .fmt = WM8750_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8750_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_16000, ++ .pcmdir = WM8750_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 1152, ++ .bfs = WM8750_HIFI_FSB ++ }, ++ { ++ .fmt = WM8750_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8750_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_16000, ++ .pcmdir = WM8750_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 750, ++ .bfs = WM8750_HIFI_FSB, ++ }, ++ ++ /* 22.05k */ ++ { ++ .fmt = WM8750_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8750_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_22050, ++ .pcmdir = WM8750_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 512, ++ .bfs = WM8750_HIFI_FSB, ++ }, ++ { ++ .fmt = WM8750_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8750_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_22050, ++ .pcmdir = WM8750_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 768, ++ .bfs = WM8750_HIFI_FSB, ++ }, ++ { ++ .fmt = WM8750_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8750_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_22050, ++ .pcmdir = WM8750_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 544, ++ .bfs = WM8750_HIFI_FSB, ++ }, ++ ++ /* 32k */ ++ { ++ .fmt = WM8750_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8750_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_32000, ++ .pcmdir = WM8750_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 384, ++ .bfs = WM8750_HIFI_FSB, ++ }, ++ { ++ .fmt = WM8750_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8750_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_32000, ++ .pcmdir = WM8750_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 576, ++ .bfs = WM8750_HIFI_FSB, ++ }, ++ { ++ .fmt = WM8750_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8750_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_32000, ++ .pcmdir = WM8750_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 375, ++ .bfs = WM8750_HIFI_FSB, ++ }, ++ ++ /* 44.1k & 48k */ ++ { ++ .fmt = WM8750_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8750_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000, ++ .pcmdir = WM8750_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 256, ++ .bfs = WM8750_HIFI_FSB, ++ }, ++ { ++ .fmt = WM8750_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8750_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000, ++ .pcmdir = WM8750_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 384, ++ .bfs = WM8750_HIFI_FSB, ++ }, ++ { ++ .fmt = WM8750_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8750_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_44100, ++ .pcmdir = WM8750_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 272, ++ .bfs = WM8750_HIFI_FSB, ++ }, ++ { ++ .fmt = WM8750_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8750_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_48000, ++ .pcmdir = WM8750_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 250, ++ .bfs = WM8750_HIFI_FSB, ++ }, ++ ++ /* 88.2k & 96k */ ++ { ++ .fmt = WM8750_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8750_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000, ++ .pcmdir = WM8750_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 128, ++ .bfs = WM8750_HIFI_FSB, ++ }, ++ { ++ .fmt = WM8750_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8750_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000, ++ .pcmdir = WM8750_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 192, ++ .bfs = WM8750_HIFI_FSB, ++ }, ++ { ++ .fmt = WM8750_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8750_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_88200, ++ .pcmdir = WM8750_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 136, ++ .bfs = WM8750_HIFI_FSB, ++ }, ++ { ++ .fmt = WM8750_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8750_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_96000, ++ .pcmdir = WM8750_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 125, ++ .bfs = WM8750_HIFI_FSB, ++ }, ++ ++ /* codec frame and clock slave modes */ ++ { ++ .fmt = WM8750_HIFI_DAIFMT | SND_SOC_DAIFMT_CBS_CFS, ++ .pcmfmt = WM8750_HIFI_BITS, ++ .pcmrate = WM8750_HIFI_RATES, ++ .pcmdir = WM8750_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = SND_SOC_FS_ALL, ++ .bfs = SND_SOC_FSB_ALL, ++ }, ++}; ++ ++/* ++ * read wm8750 register cache ++ */ ++static inline unsigned int wm8750_read_reg_cache(struct snd_soc_codec *codec, ++ unsigned int reg) ++{ ++ u16 *cache = codec->reg_cache; ++ if (reg > WM8750_CACHE_REGNUM) ++ return -1; ++ return cache[reg]; ++} ++ ++/* ++ * write wm8750 register cache ++ */ ++static inline void wm8750_write_reg_cache(struct snd_soc_codec *codec, ++ unsigned int reg, unsigned int value) ++{ ++ u16 *cache = codec->reg_cache; ++ if (reg > WM8750_CACHE_REGNUM) ++ return; ++ cache[reg] = value; ++} ++ ++static int wm8750_write(struct snd_soc_codec *codec, unsigned int reg, ++ unsigned int value) ++{ ++ u8 data[2]; ++ ++ /* data is ++ * D15..D9 WM8753 register offset ++ * D8...D0 register data ++ */ ++ data[0] = (reg << 1) | ((value >> 8) & 0x0001); ++ data[1] = value & 0x00ff; ++ ++ wm8750_write_reg_cache (codec, reg, value); ++ if (codec->hw_write(codec->control_data, data, 2) == 2) ++ return 0; ++ else ++ return -EIO; ++} ++ ++#define wm8750_reset(c) wm8750_write(c, WM8750_RESET, 0) ++ ++/* ++ * WM8750 Controls ++ */ ++static const char *wm8750_bass[] = {"Linear Control", "Adaptive Boost"}; ++static const char *wm8750_bass_filter[] = { "130Hz @ 48kHz", "200Hz @ 48kHz" }; ++static const char *wm8750_treble[] = {"8kHz", "4kHz"}; ++static const char *wm8750_3d_lc[] = {"200Hz", "500Hz"}; ++static const char *wm8750_3d_uc[] = {"2.2kHz", "1.5kHz"}; ++static const char *wm8750_3d_func[] = {"Capture", "Playback"}; ++static const char *wm8750_alc_func[] = {"Off", "Right", "Left", "Stereo"}; ++static const char *wm8750_ng_type[] = {"Constant PGA Gain", ++ "Mute ADC Output"}; ++static const char *wm8750_line_mux[] = {"Line 1", "Line 2", "Line 3", "PGA", ++ "Differential"}; ++static const char *wm8750_pga_sel[] = {"Line 1", "Line 2", "Line 3", ++ "Differential"}; ++static const char *wm8750_out3[] = {"VREF", "ROUT1 + Vol", "MonoOut", ++ "ROUT1"}; ++static const char *wm8750_diff_sel[] = {"Line 1", "Line 2"}; ++static const char *wm8750_adcpol[] = {"Normal", "L Invert", "R Invert", ++ "L + R Invert"}; ++static const char *wm8750_deemph[] = {"None", "32Khz", "44.1Khz", "48Khz"}; ++static const char *wm8750_mono_mux[] = {"Stereo", "Mono (Left)", ++ "Mono (Right)", "Digital Mono"}; ++ ++static const struct soc_enum wm8750_enum[] = { ++SOC_ENUM_SINGLE(WM8750_BASS, 7, 2, wm8750_bass), ++SOC_ENUM_SINGLE(WM8750_BASS, 6, 2, wm8750_bass_filter), ++SOC_ENUM_SINGLE(WM8750_TREBLE, 6, 2, wm8750_treble), ++SOC_ENUM_SINGLE(WM8750_3D, 5, 2, wm8750_3d_lc), ++SOC_ENUM_SINGLE(WM8750_3D, 6, 2, wm8750_3d_uc), ++SOC_ENUM_SINGLE(WM8750_3D, 7, 2, wm8750_3d_func), ++SOC_ENUM_SINGLE(WM8750_ALC1, 7, 4, wm8750_alc_func), ++SOC_ENUM_SINGLE(WM8750_NGATE, 1, 2, wm8750_ng_type), ++SOC_ENUM_SINGLE(WM8750_LOUTM1, 0, 5, wm8750_line_mux), ++SOC_ENUM_SINGLE(WM8750_ROUTM1, 0, 5, wm8750_line_mux), ++SOC_ENUM_SINGLE(WM8750_LADCIN, 6, 4, wm8750_pga_sel), /* 10 */ ++SOC_ENUM_SINGLE(WM8750_RADCIN, 6, 4, wm8750_pga_sel), ++SOC_ENUM_SINGLE(WM8750_ADCTL2, 7, 4, wm8750_out3), ++SOC_ENUM_SINGLE(WM8750_ADCIN, 8, 2, wm8750_diff_sel), ++SOC_ENUM_SINGLE(WM8750_ADCDAC, 5, 4, wm8750_adcpol), ++SOC_ENUM_SINGLE(WM8750_ADCDAC, 1, 4, wm8750_deemph), ++SOC_ENUM_SINGLE(WM8750_ADCIN, 6, 4, wm8750_mono_mux), /* 16 */ ++ ++}; ++ ++static const struct snd_kcontrol_new wm8750_snd_controls[] = { ++ ++SOC_DOUBLE_R("Capture Volume", WM8750_LINVOL, WM8750_RINVOL, 0, 63, 0), ++SOC_DOUBLE_R("Capture ZC Switch", WM8750_LINVOL, WM8750_RINVOL, 6, 1, 0), ++SOC_DOUBLE_R("Capture Switch", WM8750_LINVOL, WM8750_RINVOL, 7, 1, 1), ++ ++SOC_DOUBLE_R("Headphone Playback ZC Switch", WM8750_LOUT1V, ++ WM8750_ROUT1V, 7, 1, 0), ++SOC_DOUBLE_R("Speaker Playback ZC Switch", WM8750_LOUT2V, ++ WM8750_ROUT2V, 7, 1, 0), ++ ++SOC_ENUM("Playback De-emphasis", wm8750_enum[15]), ++ ++SOC_ENUM("Capture Polarity", wm8750_enum[14]), ++SOC_SINGLE("Playback 6dB Attenuate", WM8750_ADCDAC, 7, 1, 0), ++SOC_SINGLE("Capture 6dB Attenuate", WM8750_ADCDAC, 8, 1, 0), ++ ++SOC_DOUBLE_R("PCM Volume", WM8750_LDAC, WM8750_RDAC, 0, 255, 0), ++ ++SOC_ENUM("Bass Boost", wm8750_enum[0]), ++SOC_ENUM("Bass Filter", wm8750_enum[1]), ++SOC_SINGLE("Bass Volume", WM8750_BASS, 0, 15, 1), ++ ++SOC_SINGLE("Treble Volume", WM8750_TREBLE, 0, 15, 0), ++SOC_ENUM("Treble Cut-off", wm8750_enum[2]), ++ ++SOC_SINGLE("3D Switch", WM8750_3D, 0, 1, 0), ++SOC_SINGLE("3D Volume", WM8750_3D, 1, 15, 0), ++SOC_ENUM("3D Lower Cut-off", wm8750_enum[3]), ++SOC_ENUM("3D Upper Cut-off", wm8750_enum[4]), ++SOC_ENUM("3D Mode", wm8750_enum[5]), ++ ++SOC_SINGLE("ALC Capture Target Volume", WM8750_ALC1, 0, 7, 0), ++SOC_SINGLE("ALC Capture Max Volume", WM8750_ALC1, 4, 7, 0), ++SOC_ENUM("ALC Capture Function", wm8750_enum[6]), ++SOC_SINGLE("ALC Capture ZC Switch", WM8750_ALC2, 7, 1, 0), ++SOC_SINGLE("ALC Capture Hold Time", WM8750_ALC2, 0, 15, 0), ++SOC_SINGLE("ALC Capture Decay Time", WM8750_ALC3, 4, 15, 0), ++SOC_SINGLE("ALC Capture Attack Time", WM8750_ALC3, 0, 15, 0), ++SOC_SINGLE("ALC Capture NG Threshold", WM8750_NGATE, 3, 31, 0), ++SOC_ENUM("ALC Capture NG Type", wm8750_enum[4]), ++SOC_SINGLE("ALC Capture NG Switch", WM8750_NGATE, 0, 1, 0), ++ ++SOC_SINGLE("Left ADC Capture Volume", WM8750_LADC, 0, 255, 0), ++SOC_SINGLE("Right ADC Capture Volume", WM8750_RADC, 0, 255, 0), ++ ++SOC_SINGLE("ZC Timeout Switch", WM8750_ADCTL1, 0, 1, 0), ++SOC_SINGLE("Playback Invert Switch", WM8750_ADCTL1, 1, 1, 0), ++ ++SOC_SINGLE("Right Speaker Playback Invert Switch", WM8750_ADCTL2, 4, 1, 0), ++ ++/* Unimplemented */ ++/* ADCDAC Bit 0 - ADCHPD */ ++/* ADCDAC Bit 4 - HPOR */ ++/* ADCTL1 Bit 2,3 - DATSEL */ ++/* ADCTL1 Bit 4,5 - DMONOMIX */ ++/* ADCTL1 Bit 6,7 - VSEL */ ++/* ADCTL2 Bit 2 - LRCM */ ++/* ADCTL2 Bit 3 - TRI */ ++/* ADCTL3 Bit 5 - HPFLREN */ ++/* ADCTL3 Bit 6 - VROI */ ++/* ADCTL3 Bit 7,8 - ADCLRM */ ++/* ADCIN Bit 4 - LDCM */ ++/* ADCIN Bit 5 - RDCM */ ++ ++SOC_DOUBLE_R("Mic Boost", WM8750_LADCIN, WM8750_RADCIN, 4, 3, 0), ++ ++SOC_DOUBLE_R("Bypass Left Playback Volume", WM8750_LOUTM1, ++ WM8750_LOUTM2, 4, 7, 1), ++SOC_DOUBLE_R("Bypass Right Playback Volume", WM8750_ROUTM1, ++ WM8750_ROUTM2, 4, 7, 1), ++SOC_DOUBLE_R("Bypass Mono Playback Volume", WM8750_MOUTM1, ++ WM8750_MOUTM2, 4, 7, 1), ++ ++SOC_SINGLE("Mono Playback ZC Switch", WM8750_MOUTV, 7, 1, 0), ++ ++SOC_DOUBLE_R("Headphone Playback Volume", WM8750_LOUT1V, WM8750_ROUT1V, ++ 0, 127, 0), ++SOC_DOUBLE_R("Speaker Playback Volume", WM8750_LOUT2V, WM8750_ROUT2V, ++ 0, 127, 0), ++ ++SOC_SINGLE("Mono Playback Volume", WM8750_MOUTV, 0, 127, 0), ++ ++}; ++ ++/* add non dapm controls */ ++static int wm8750_add_controls(struct snd_soc_codec *codec) ++{ ++ int err, i; ++ ++ for (i = 0; i < ARRAY_SIZE(wm8750_snd_controls); i++) { ++ err = snd_ctl_add(codec->card, ++ snd_soc_cnew(&wm8750_snd_controls[i],codec, NULL)); ++ if (err < 0) ++ return err; ++ } ++ return 0; ++} ++ ++/* ++ * DAPM Controls ++ */ ++ ++/* Left Mixer */ ++static const struct snd_kcontrol_new wm8750_left_mixer_controls[] = { ++SOC_DAPM_SINGLE("Playback Switch", WM8750_LOUTM1, 8, 1, 0), ++SOC_DAPM_SINGLE("Left Bypass Switch", WM8750_LOUTM1, 7, 1, 0), ++SOC_DAPM_SINGLE("Right Playback Switch", WM8750_LOUTM2, 8, 1, 0), ++SOC_DAPM_SINGLE("Right Bypass Switch", WM8750_LOUTM2, 7, 1, 0), ++}; ++ ++/* Right Mixer */ ++static const struct snd_kcontrol_new wm8750_right_mixer_controls[] = { ++SOC_DAPM_SINGLE("Left Playback Switch", WM8750_ROUTM1, 8, 1, 0), ++SOC_DAPM_SINGLE("Left Bypass Switch", WM8750_ROUTM1, 7, 1, 0), ++SOC_DAPM_SINGLE("Playback Switch", WM8750_ROUTM2, 8, 1, 0), ++SOC_DAPM_SINGLE("Right Bypass Switch", WM8750_ROUTM2, 7, 1, 0), ++}; ++ ++/* Mono Mixer */ ++static const struct snd_kcontrol_new wm8750_mono_mixer_controls[] = { ++SOC_DAPM_SINGLE("Left Playback Switch", WM8750_MOUTM1, 8, 1, 0), ++SOC_DAPM_SINGLE("Left Bypass Switch", WM8750_MOUTM1, 7, 1, 0), ++SOC_DAPM_SINGLE("Right Playback Switch", WM8750_MOUTM2, 8, 1, 0), ++SOC_DAPM_SINGLE("Right Bypass Switch", WM8750_MOUTM2, 7, 1, 0), ++}; ++ ++/* Left Line Mux */ ++static const struct snd_kcontrol_new wm8750_left_line_controls = ++SOC_DAPM_ENUM("Route", wm8750_enum[8]); ++ ++/* Right Line Mux */ ++static const struct snd_kcontrol_new wm8750_right_line_controls = ++SOC_DAPM_ENUM("Route", wm8750_enum[9]); ++ ++/* Left PGA Mux */ ++static const struct snd_kcontrol_new wm8750_left_pga_controls = ++SOC_DAPM_ENUM("Route", wm8750_enum[10]); ++ ++/* Right PGA Mux */ ++static const struct snd_kcontrol_new wm8750_right_pga_controls = ++SOC_DAPM_ENUM("Route", wm8750_enum[11]); ++ ++/* Out 3 Mux */ ++static const struct snd_kcontrol_new wm8750_out3_controls = ++SOC_DAPM_ENUM("Route", wm8750_enum[12]); ++ ++/* Differential Mux */ ++static const struct snd_kcontrol_new wm8750_diffmux_controls = ++SOC_DAPM_ENUM("Route", wm8750_enum[13]); ++ ++/* Mono ADC Mux */ ++static const struct snd_kcontrol_new wm8750_monomux_controls = ++SOC_DAPM_ENUM("Route", wm8750_enum[16]); ++ ++static const struct snd_soc_dapm_widget wm8750_dapm_widgets[] = { ++ SND_SOC_DAPM_MIXER("Left Mixer", SND_SOC_NOPM, 0, 0, ++ &wm8750_left_mixer_controls[0], ++ ARRAY_SIZE(wm8750_left_mixer_controls)), ++ SND_SOC_DAPM_MIXER("Right Mixer", SND_SOC_NOPM, 0, 0, ++ &wm8750_right_mixer_controls[0], ++ ARRAY_SIZE(wm8750_right_mixer_controls)), ++ SND_SOC_DAPM_MIXER("Mono Mixer", WM8750_PWR2, 2, 0, ++ &wm8750_mono_mixer_controls[0], ++ ARRAY_SIZE(wm8750_mono_mixer_controls)), ++ ++ SND_SOC_DAPM_PGA("Right Out 2", WM8750_PWR2, 3, 0, NULL, 0), ++ SND_SOC_DAPM_PGA("Left Out 2", WM8750_PWR2, 4, 0, NULL, 0), ++ SND_SOC_DAPM_PGA("Right Out 1", WM8750_PWR2, 5, 0, NULL, 0), ++ SND_SOC_DAPM_PGA("Left Out 1", WM8750_PWR2, 6, 0, NULL, 0), ++ SND_SOC_DAPM_DAC("Right DAC", "Right Playback", WM8750_PWR2, 7, 0), ++ SND_SOC_DAPM_DAC("Left DAC", "Left Playback", WM8750_PWR2, 8, 0), ++ ++ SND_SOC_DAPM_MICBIAS("Mic Bias", WM8750_PWR1, 1, 0), ++ SND_SOC_DAPM_ADC("Right ADC", "Right Capture", WM8750_PWR1, 2, 0), ++ SND_SOC_DAPM_ADC("Left ADC", "Left Capture", WM8750_PWR1, 3, 0), ++ ++ SND_SOC_DAPM_MUX("Left PGA Mux", WM8750_PWR1, 5, 0, ++ &wm8750_left_pga_controls), ++ SND_SOC_DAPM_MUX("Right PGA Mux", WM8750_PWR1, 4, 0, ++ &wm8750_right_pga_controls), ++ SND_SOC_DAPM_MUX("Left Line Mux", SND_SOC_NOPM, 0, 0, ++ &wm8750_left_line_controls), ++ SND_SOC_DAPM_MUX("Right Line Mux", SND_SOC_NOPM, 0, 0, ++ &wm8750_right_line_controls), ++ ++ SND_SOC_DAPM_MUX("Out3 Mux", SND_SOC_NOPM, 0, 0, &wm8750_out3_controls), ++ SND_SOC_DAPM_PGA("Out 3", WM8750_PWR2, 1, 0, NULL, 0), ++ SND_SOC_DAPM_PGA("Mono Out 1", WM8750_PWR2, 2, 0, NULL, 0), ++ ++ SND_SOC_DAPM_MUX("Differential Mux", SND_SOC_NOPM, 0, 0, ++ &wm8750_diffmux_controls), ++ SND_SOC_DAPM_MUX("Left ADC Mux", SND_SOC_NOPM, 0, 0, ++ &wm8750_monomux_controls), ++ SND_SOC_DAPM_MUX("Right ADC Mux", SND_SOC_NOPM, 0, 0, ++ &wm8750_monomux_controls), ++ ++ SND_SOC_DAPM_OUTPUT("LOUT1"), ++ SND_SOC_DAPM_OUTPUT("ROUT1"), ++ SND_SOC_DAPM_OUTPUT("LOUT2"), ++ SND_SOC_DAPM_OUTPUT("ROUT2"), ++ SND_SOC_DAPM_OUTPUT("MONO"), ++ SND_SOC_DAPM_OUTPUT("OUT3"), ++ ++ SND_SOC_DAPM_INPUT("LINPUT1"), ++ SND_SOC_DAPM_INPUT("LINPUT2"), ++ SND_SOC_DAPM_INPUT("LINPUT3"), ++ SND_SOC_DAPM_INPUT("RINPUT1"), ++ SND_SOC_DAPM_INPUT("RINPUT2"), ++ SND_SOC_DAPM_INPUT("RINPUT3"), ++}; ++ ++static const char *audio_map[][3] = { ++ /* left mixer */ ++ {"Left Mixer", "Playback Switch", "Left DAC"}, ++ {"Left Mixer", "Left Bypass Switch", "Left Line Mux"}, ++ {"Left Mixer", "Right Playback Switch", "Right DAC"}, ++ {"Left Mixer", "Right Bypass Switch", "Right Line Mux"}, ++ ++ /* right mixer */ ++ {"Right Mixer", "Left Playback Switch", "Left DAC"}, ++ {"Right Mixer", "Left Bypass Switch", "Left Line Mux"}, ++ {"Right Mixer", "Playback Switch", "Right DAC"}, ++ {"Right Mixer", "Right Bypass Switch", "Right Line Mux"}, ++ ++ /* left out 1 */ ++ {"Left Out 1", NULL, "Left Mixer"}, ++ {"LOUT1", NULL, "Left Out 1"}, ++ ++ /* left out 2 */ ++ {"Left Out 2", NULL, "Left Mixer"}, ++ {"LOUT2", NULL, "Left Out 2"}, ++ ++ /* right out 1 */ ++ {"Right Out 1", NULL, "Right Mixer"}, ++ {"ROUT1", NULL, "Right Out 1"}, ++ ++ /* right out 2 */ ++ {"Right Out 2", NULL, "Right Mixer"}, ++ {"ROUT2", NULL, "Right Out 2"}, ++ ++ /* mono mixer */ ++ {"Mono Mixer", "Left Playback Switch", "Left DAC"}, ++ {"Mono Mixer", "Left Bypass Switch", "Left Line Mux"}, ++ {"Mono Mixer", "Right Playback Switch", "Right DAC"}, ++ {"Mono Mixer", "Right Bypass Switch", "Right Line Mux"}, ++ ++ /* mono out */ ++ {"Mono Out 1", NULL, "Mono Mixer"}, ++ {"MONO1", NULL, "Mono Out 1"}, ++ ++ /* out 3 */ ++ {"Out3 Mux", "VREF", "VREF"}, ++ {"Out3 Mux", "ROUT1 + Vol", "ROUT1"}, ++ {"Out3 Mux", "ROUT1", "Right Mixer"}, ++ {"Out3 Mux", "MonoOut", "MONO1"}, ++ {"Out 3", NULL, "Out3 Mux"}, ++ {"OUT3", NULL, "Out 3"}, ++ ++ /* Left Line Mux */ ++ {"Left Line Mux", "Line 1", "LINPUT1"}, ++ {"Left Line Mux", "Line 2", "LINPUT2"}, ++ {"Left Line Mux", "Line 3", "LINPUT3"}, ++ {"Left Line Mux", "PGA", "Left PGA Mux"}, ++ {"Left Line Mux", "Differential", "Differential Mux"}, ++ ++ /* Right Line Mux */ ++ {"Right Line Mux", "Line 1", "RINPUT1"}, ++ {"Right Line Mux", "Line 2", "RINPUT2"}, ++ {"Right Line Mux", "Line 3", "RINPUT3"}, ++ {"Right Line Mux", "PGA", "Right PGA Mux"}, ++ {"Right Line Mux", "Differential", "Differential Mux"}, ++ ++ /* Left PGA Mux */ ++ {"Left PGA Mux", "Line 1", "LINPUT1"}, ++ {"Left PGA Mux", "Line 2", "LINPUT2"}, ++ {"Left PGA Mux", "Line 3", "LINPUT3"}, ++ {"Left PGA Mux", "Differential", "Differential Mux"}, ++ ++ /* Right PGA Mux */ ++ {"Right PGA Mux", "Line 1", "RINPUT1"}, ++ {"Right PGA Mux", "Line 2", "RINPUT2"}, ++ {"Right PGA Mux", "Line 3", "RINPUT3"}, ++ {"Right PGA Mux", "Differential", "Differential Mux"}, ++ ++ /* Differential Mux */ ++ {"Differential Mux", "Line 1", "LINPUT1"}, ++ {"Differential Mux", "Line 1", "RINPUT1"}, ++ {"Differential Mux", "Line 2", "LINPUT2"}, ++ {"Differential Mux", "Line 2", "RINPUT2"}, ++ ++ /* Left ADC Mux */ ++ {"Left ADC Mux", "Stereo", "Left PGA Mux"}, ++ {"Left ADC Mux", "Mono (Left)", "Left PGA Mux"}, ++ {"Left ADC Mux", "Digital Mono", "Left PGA Mux"}, ++ ++ /* Right ADC Mux */ ++ {"Right ADC Mux", "Stereo", "Right PGA Mux"}, ++ {"Right ADC Mux", "Mono (Right)", "Right PGA Mux"}, ++ {"Right ADC Mux", "Digital Mono", "Right PGA Mux"}, ++ ++ /* ADC */ ++ {"Left ADC", NULL, "Left ADC Mux"}, ++ {"Right ADC", NULL, "Right ADC Mux"}, ++ ++ /* terminator */ ++ {NULL, NULL, NULL}, ++}; ++ ++static int wm8750_add_widgets(struct snd_soc_codec *codec) ++{ ++ int i; ++ ++ for(i = 0; i < ARRAY_SIZE(wm8750_dapm_widgets); i++) { ++ snd_soc_dapm_new_control(codec, &wm8750_dapm_widgets[i]); ++ } ++ ++ /* set up audio path audio_mapnects */ ++ for(i = 0; audio_map[i][0] != NULL; i++) { ++ snd_soc_dapm_connect_input(codec, audio_map[i][0], ++ audio_map[i][1], audio_map[i][2]); ++ } ++ ++ snd_soc_dapm_new_widgets(codec); ++ return 0; ++} ++ ++struct _coeff_div { ++ u32 mclk; ++ u32 rate; ++ u16 fs; ++ u8 sr:5; ++ u8 usb:1; ++}; ++ ++/* codec hifi mclk clock divider coefficients */ ++static const struct _coeff_div coeff_div[] = { ++ /* 8k */ ++ {12288000, 8000, 1536, 0x6, 0x0}, ++ {11289600, 8000, 1408, 0x16, 0x0}, ++ {18432000, 8000, 2304, 0x7, 0x0}, ++ {16934400, 8000, 2112, 0x17, 0x0}, ++ {12000000, 8000, 1500, 0x6, 0x1}, ++ ++ /* 11.025k */ ++ {11289600, 11025, 1024, 0x18, 0x0}, ++ {16934400, 11025, 1536, 0x19, 0x0}, ++ {12000000, 11025, 1088, 0x19, 0x1}, ++ ++ /* 16k */ ++ {12288000, 16000, 768, 0xa, 0x0}, ++ {18432000, 16000, 1152, 0xb, 0x0}, ++ {12000000, 16000, 750, 0xa, 0x1}, ++ ++ /* 22.05k */ ++ {11289600, 22050, 512, 0x1a, 0x0}, ++ {16934400, 22050, 768, 0x1b, 0x0}, ++ {12000000, 22050, 544, 0x1b, 0x1}, ++ ++ /* 32k */ ++ {12288000, 32000, 384, 0xc, 0x0}, ++ {18432000, 32000, 576, 0xd, 0x0}, ++ {12000000, 32000, 375, 0xa, 0x1}, ++ ++ /* 44.1k */ ++ {11289600, 44100, 256, 0x10, 0x0}, ++ {16934400, 44100, 384, 0x11, 0x0}, ++ {12000000, 44100, 272, 0x11, 0x1}, ++ ++ /* 48k */ ++ {12288000, 48000, 256, 0x0, 0x0}, ++ {18432000, 48000, 384, 0x1, 0x0}, ++ {12000000, 48000, 250, 0x0, 0x1}, ++ ++ /* 88.2k */ ++ {11289600, 88200, 128, 0x1e, 0x0}, ++ {16934400, 88200, 192, 0x1f, 0x0}, ++ {12000000, 88200, 136, 0x1f, 0x1}, ++ ++ /* 96k */ ++ {12288000, 96000, 128, 0xe, 0x0}, ++ {18432000, 96000, 192, 0xf, 0x0}, ++ {12000000, 96000, 125, 0xe, 0x1}, ++}; ++ ++static inline int get_coeff(int mclk, int rate) ++{ ++ int i; ++ ++ for (i = 0; i < ARRAY_SIZE(coeff_div); i++) { ++ if (coeff_div[i].rate == rate && coeff_div[i].mclk == mclk) ++ return i; ++ } ++ ++ printk(KERN_ERR "wm8750: could not get coeff for mclk %d @ rate %d\n", ++ mclk, rate); ++ return -EINVAL; ++} ++ ++/* WM8750 supports numerous input clocks per sample rate */ ++static unsigned int wm8750_config_sysclk(struct snd_soc_codec_dai *dai, ++ struct snd_soc_clock_info *info, unsigned int clk) ++{ ++ dai->mclk = clk; ++ return dai->mclk; ++} ++ ++static int wm8750_pcm_prepare(struct snd_pcm_substream *substream) ++{ ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ struct snd_soc_device *socdev = rtd->socdev; ++ struct snd_soc_codec *codec = socdev->codec; ++ u16 iface = 0, bfs, srate = 0; ++ int i = get_coeff(rtd->codec_dai->mclk, ++ snd_soc_get_rate(rtd->codec_dai->dai_runtime.pcmrate)); ++ ++ /* is coefficient valid ? */ ++ if (i < 0) ++ return i; ++ ++ bfs = SND_SOC_FSBD_REAL(rtd->codec_dai->dai_runtime.bfs); ++ ++ /* set master/slave audio interface */ ++ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_CLOCK_MASK) { ++ case SND_SOC_DAIFMT_CBM_CFM: ++ iface = 0x0040; ++ break; ++ case SND_SOC_DAIFMT_CBS_CFS: ++ break; ++ } ++ ++ /* interface format */ ++ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_FORMAT_MASK) { ++ case SND_SOC_DAIFMT_I2S: ++ iface |= 0x0002; ++ break; ++ case SND_SOC_DAIFMT_RIGHT_J: ++ break; ++ case SND_SOC_DAIFMT_LEFT_J: ++ iface |= 0x0001; ++ break; ++ case SND_SOC_DAIFMT_DSP_A: ++ iface |= 0x0003; ++ break; ++ case SND_SOC_DAIFMT_DSP_B: ++ iface |= 0x0013; ++ break; ++ } ++ ++ /* bit size */ ++ switch (rtd->codec_dai->dai_runtime.pcmfmt) { ++ case SNDRV_PCM_FMTBIT_S16_LE: ++ break; ++ case SNDRV_PCM_FMTBIT_S20_3LE: ++ iface |= 0x0004; ++ break; ++ case SNDRV_PCM_FMTBIT_S24_LE: ++ iface |= 0x0008; ++ break; ++ case SNDRV_PCM_FMTBIT_S32_LE: ++ iface |= 0x000c; ++ break; ++ } ++ ++ /* clock inversion */ ++ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_INV_MASK) { ++ case SND_SOC_DAIFMT_NB_NF: ++ break; ++ case SND_SOC_DAIFMT_IB_IF: ++ iface |= 0x0090; ++ break; ++ case SND_SOC_DAIFMT_IB_NF: ++ iface |= 0x0080; ++ break; ++ case SND_SOC_DAIFMT_NB_IF: ++ iface |= 0x0010; ++ break; ++ } ++ ++ /* set bclk divisor rate */ ++ switch (bfs) { ++ case 1: ++ break; ++ case 4: ++ srate |= (0x1 << 7); ++ break; ++ case 8: ++ srate |= (0x2 << 7); ++ break; ++ case 16: ++ srate |= (0x3 << 7); ++ break; ++ } ++ ++ /* set iface & srate */ ++ wm8750_write(codec, WM8750_IFACE, iface); ++ wm8750_write(codec, WM8750_SRATE, srate | ++ (coeff_div[i].sr << 1) | coeff_div[i].usb); ++ ++ return 0; ++} ++ ++static int wm8750_mute(struct snd_soc_codec *codec, ++ struct snd_soc_codec_dai *dai, int mute) ++{ ++ u16 mute_reg = wm8750_read_reg_cache(codec, WM8750_ADCDAC) & 0xfff7; ++ if (mute) ++ wm8750_write(codec, WM8750_ADCDAC, mute_reg | 0x8); ++ else ++ wm8750_write(codec, WM8750_ADCDAC, mute_reg); ++ return 0; ++} ++ ++static int wm8750_dapm_event(struct snd_soc_codec *codec, int event) ++{ ++ u16 pwr_reg = wm8750_read_reg_cache(codec, WM8750_PWR1) & 0xfe3e; ++ ++ switch (event) { ++ case SNDRV_CTL_POWER_D0: /* full On */ ++ /* set vmid to 50k and unmute dac */ ++ wm8750_write(codec, WM8750_PWR1, pwr_reg | 0x00c0); ++ break; ++ case SNDRV_CTL_POWER_D1: /* partial On */ ++ case SNDRV_CTL_POWER_D2: /* partial On */ ++ /* set vmid to 5k for quick power up */ ++ wm8750_write(codec, WM8750_PWR1, pwr_reg | 0x01c1); ++ break; ++ case SNDRV_CTL_POWER_D3hot: /* Off, with power */ ++ /* mute dac and set vmid to 500k, enable VREF */ ++ wm8750_write(codec, WM8750_PWR1, pwr_reg | 0x0141); ++ break; ++ case SNDRV_CTL_POWER_D3cold: /* Off, without power */ ++ wm8750_write(codec, WM8750_PWR1, 0x0001); ++ break; ++ } ++ codec->dapm_state = event; ++ return 0; ++} ++ ++struct snd_soc_codec_dai wm8750_dai = { ++ .name = "WM8750", ++ .playback = { ++ .stream_name = "Playback", ++ .channels_min = 1, ++ .channels_max = 2, ++ }, ++ .capture = { ++ .stream_name = "Capture", ++ .channels_min = 1, ++ .channels_max = 2, ++ }, ++ .config_sysclk = wm8750_config_sysclk, ++ .digital_mute = wm8750_mute, ++ .ops = { ++ .prepare = wm8750_pcm_prepare, ++ }, ++ .caps = { ++ .num_modes = ARRAY_SIZE(wm8750_modes), ++ .mode = wm8750_modes, ++ }, ++}; ++EXPORT_SYMBOL_GPL(wm8750_dai); ++ ++static void wm8750_work(void *data) ++{ ++ struct snd_soc_codec *codec = (struct snd_soc_codec *)data; ++ wm8750_dapm_event(codec, codec->dapm_state); ++} ++ ++static int wm8750_suspend(struct platform_device *pdev, pm_message_t state) ++{ ++ struct snd_soc_device *socdev = platform_get_drvdata(pdev); ++ struct snd_soc_codec *codec = socdev->codec; ++ ++ wm8750_dapm_event(codec, SNDRV_CTL_POWER_D3cold); ++ return 0; ++} ++ ++static int wm8750_resume(struct platform_device *pdev) ++{ ++ struct snd_soc_device *socdev = platform_get_drvdata(pdev); ++ struct snd_soc_codec *codec = socdev->codec; ++ int i; ++ u8 data[2]; ++ u16 *cache = codec->reg_cache; ++ ++ /* Sync reg_cache with the hardware */ ++ for (i = 0; i < ARRAY_SIZE(wm8750_reg); i++) { ++ if (i == WM8750_RESET) ++ continue; ++ data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); ++ data[1] = cache[i] & 0x00ff; ++ codec->hw_write(codec->control_data, data, 2); ++ } ++ ++ wm8750_dapm_event(codec, SNDRV_CTL_POWER_D3hot); ++ ++ /* charge wm8750 caps */ ++ if (codec->suspend_dapm_state == SNDRV_CTL_POWER_D0) { ++ wm8750_dapm_event(codec, SNDRV_CTL_POWER_D2); ++ codec->dapm_state = SNDRV_CTL_POWER_D0; ++ queue_delayed_work(wm8750_workq, &wm8750_dapm_work, ++ msecs_to_jiffies(1000)); ++ } ++ ++ return 0; ++} ++ ++/* ++ * initialise the WM8750 driver ++ * register the mixer and dsp interfaces with the kernel ++ */ ++static int wm8750_init(struct snd_soc_device *socdev) ++{ ++ struct snd_soc_codec *codec = socdev->codec; ++ int reg, ret = 0; ++ ++ codec->name = "WM8750"; ++ codec->owner = THIS_MODULE; ++ codec->read = wm8750_read_reg_cache; ++ codec->write = wm8750_write; ++ codec->dapm_event = wm8750_dapm_event; ++ codec->dai = &wm8750_dai; ++ codec->num_dai = 1; ++ codec->reg_cache_size = ARRAY_SIZE(wm8750_reg); ++ ++ codec->reg_cache = ++ kzalloc(sizeof(u16) * ARRAY_SIZE(wm8750_reg), GFP_KERNEL); ++ if (codec->reg_cache == NULL) ++ return -ENOMEM; ++ memcpy(codec->reg_cache, wm8750_reg, ++ sizeof(u16) * ARRAY_SIZE(wm8750_reg)); ++ codec->reg_cache_size = sizeof(u16) * ARRAY_SIZE(wm8750_reg); ++ ++ wm8750_reset(codec); ++ ++ /* register pcms */ ++ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); ++ if (ret < 0) { ++ kfree(codec->reg_cache); ++ return ret; ++ } ++ ++ /* charge output caps */ ++ wm8750_dapm_event(codec, SNDRV_CTL_POWER_D2); ++ codec->dapm_state = SNDRV_CTL_POWER_D3hot; ++ queue_delayed_work(wm8750_workq, &wm8750_dapm_work, ++ msecs_to_jiffies(1000)); ++ ++ /* set the update bits */ ++ reg = wm8750_read_reg_cache(codec, WM8750_LDAC); ++ wm8750_write(codec, WM8750_LDAC, reg | 0x0100); ++ reg = wm8750_read_reg_cache(codec, WM8750_RDAC); ++ wm8750_write(codec, WM8750_RDAC, reg | 0x0100); ++ reg = wm8750_read_reg_cache(codec, WM8750_LOUT1V); ++ wm8750_write(codec, WM8750_LOUT1V, reg | 0x0100); ++ reg = wm8750_read_reg_cache(codec, WM8750_ROUT1V); ++ wm8750_write(codec, WM8750_ROUT1V, reg | 0x0100); ++ reg = wm8750_read_reg_cache(codec, WM8750_LOUT2V); ++ wm8750_write(codec, WM8750_LOUT2V, reg | 0x0100); ++ reg = wm8750_read_reg_cache(codec, WM8750_ROUT2V); ++ wm8750_write(codec, WM8750_ROUT2V, reg | 0x0100); ++ reg = wm8750_read_reg_cache(codec, WM8750_LINVOL); ++ wm8750_write(codec, WM8750_LINVOL, reg | 0x0100); ++ reg = wm8750_read_reg_cache(codec, WM8750_RINVOL); ++ wm8750_write(codec, WM8750_RINVOL, reg | 0x0100); ++ ++ wm8750_add_controls(codec); ++ wm8750_add_widgets(codec); ++ ret = snd_soc_register_card(socdev); ++ if (ret < 0) { ++ snd_soc_free_pcms(socdev); ++ snd_soc_dapm_free(socdev); ++ } ++ ++ return ret; ++} ++ ++/* If the i2c layer weren't so broken, we could pass this kind of data ++ around */ ++static struct snd_soc_device *wm8750_socdev; ++ ++#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) ++ ++/* ++ * WM8731 2 wire address is determined by GPIO5 ++ * state during powerup. ++ * low = 0x1a ++ * high = 0x1b ++ */ ++static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END }; ++ ++/* Magic definition of all other variables and things */ ++I2C_CLIENT_INSMOD; ++ ++static struct i2c_driver wm8750_i2c_driver; ++static struct i2c_client client_template; ++ ++static int wm8750_codec_probe(struct i2c_adapter *adap, int addr, int kind) ++{ ++ struct snd_soc_device *socdev = wm8750_socdev; ++ struct wm8750_setup_data *setup = socdev->codec_data; ++ struct snd_soc_codec *codec = socdev->codec; ++ struct i2c_client *i2c; ++ int ret; ++ ++ if (addr != setup->i2c_address) ++ return -ENODEV; ++ ++ client_template.adapter = adap; ++ client_template.addr = addr; ++ ++ i2c = kzalloc(sizeof(struct i2c_client), GFP_KERNEL); ++ if (i2c == NULL) { ++ kfree(codec); ++ return -ENOMEM; ++ } ++ memcpy(i2c, &client_template, sizeof(struct i2c_client)); ++ i2c_set_clientdata(i2c, codec); ++ codec->control_data = i2c; ++ ++ ret = i2c_attach_client(i2c); ++ if (ret < 0) { ++ err("failed to attach codec at addr %x\n", addr); ++ goto err; ++ } ++ ++ ret = wm8750_init(socdev); ++ if (ret < 0) { ++ err("failed to initialise WM8750\n"); ++ goto err; ++ } ++ return ret; ++ ++err: ++ kfree(codec); ++ kfree(i2c); ++ return ret; ++} ++ ++static int wm8750_i2c_detach(struct i2c_client *client) ++{ ++ struct snd_soc_codec *codec = i2c_get_clientdata(client); ++ i2c_detach_client(client); ++ kfree(codec->reg_cache); ++ kfree(client); ++ return 0; ++} ++ ++static int wm8750_i2c_attach(struct i2c_adapter *adap) ++{ ++ return i2c_probe(adap, &addr_data, wm8750_codec_probe); ++} ++ ++/* corgi i2c codec control layer */ ++static struct i2c_driver wm8750_i2c_driver = { ++ .driver = { ++ .name = "WM8750 I2C Codec", ++ .owner = THIS_MODULE, ++ }, ++ .id = I2C_DRIVERID_WM8750, ++ .attach_adapter = wm8750_i2c_attach, ++ .detach_client = wm8750_i2c_detach, ++ .command = NULL, ++}; ++ ++static struct i2c_client client_template = { ++ .name = "WM8750", ++ .driver = &wm8750_i2c_driver, ++}; ++#endif ++ ++static int wm8750_probe(struct platform_device *pdev) ++{ ++ struct snd_soc_device *socdev = platform_get_drvdata(pdev); ++ struct wm8750_setup_data *setup = socdev->codec_data; ++ struct snd_soc_codec *codec; ++ int ret = 0; ++ ++ info("WM8750 Audio Codec %s", WM8750_VERSION); ++ codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); ++ if (codec == NULL) ++ return -ENOMEM; ++ ++ socdev->codec = codec; ++ mutex_init(&codec->mutex); ++ INIT_LIST_HEAD(&codec->dapm_widgets); ++ INIT_LIST_HEAD(&codec->dapm_paths); ++ wm8750_socdev = socdev; ++ INIT_WORK(&wm8750_dapm_work, wm8750_work, codec); ++ wm8750_workq = create_workqueue("wm8750"); ++ if (wm8750_workq == NULL) { ++ kfree(codec); ++ return -ENOMEM; ++ } ++#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) ++ if (setup->i2c_address) { ++ normal_i2c[0] = setup->i2c_address; ++ codec->hw_write = (hw_write_t)i2c_master_send; ++ ret = i2c_add_driver(&wm8750_i2c_driver); ++ if (ret != 0) ++ printk(KERN_ERR "can't add i2c driver"); ++ } ++#else ++ /* Add other interfaces here */ ++#endif ++ ++ return ret; ++} ++ ++/* power down chip */ ++static int wm8750_remove(struct platform_device *pdev) ++{ ++ struct snd_soc_device *socdev = platform_get_drvdata(pdev); ++ struct snd_soc_codec *codec = socdev->codec; ++ ++ if (codec->control_data) ++ wm8750_dapm_event(codec, SNDRV_CTL_POWER_D3cold); ++ if (wm8750_workq) ++ destroy_workqueue(wm8750_workq); ++ snd_soc_free_pcms(socdev); ++ snd_soc_dapm_free(socdev); ++#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) ++ i2c_del_driver(&wm8750_i2c_driver); ++#endif ++ kfree(codec); ++ ++ return 0; ++} ++ ++struct snd_soc_codec_device soc_codec_dev_wm8750 = { ++ .probe = wm8750_probe, ++ .remove = wm8750_remove, ++ .suspend = wm8750_suspend, ++ .resume = wm8750_resume, ++}; ++ ++EXPORT_SYMBOL_GPL(soc_codec_dev_wm8750); ++ ++MODULE_DESCRIPTION("ASoC WM8750 driver"); ++MODULE_AUTHOR("Liam Girdwood"); ++MODULE_LICENSE("GPL"); +Index: linux-2.6-pxa-new/sound/soc/codecs/wm8750.h +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/sound/soc/codecs/wm8750.h +@@ -0,0 +1,66 @@ ++/* ++ * Copyright 2005 Openedhand Ltd. ++ * ++ * Author: Richard Purdie <richard@openedhand.com> ++ * ++ * Based on WM8753.h ++ * ++ * This program is free software; you can redistribute it and/or modify ++ * it under the terms of the GNU General Public License version 2 as ++ * published by the Free Software Foundation. ++ * ++ */ ++ ++#ifndef _WM8750_H ++#define _WM8750_H ++ ++/* WM8750 register space */ ++ ++#define WM8750_LINVOL 0x00 ++#define WM8750_RINVOL 0x01 ++#define WM8750_LOUT1V 0x02 ++#define WM8750_ROUT1V 0x03 ++#define WM8750_ADCDAC 0x05 ++#define WM8750_IFACE 0x07 ++#define WM8750_SRATE 0x08 ++#define WM8750_LDAC 0x0a ++#define WM8750_RDAC 0x0b ++#define WM8750_BASS 0x0c ++#define WM8750_TREBLE 0x0d ++#define WM8750_RESET 0x0f ++#define WM8750_3D 0x10 ++#define WM8750_ALC1 0x11 ++#define WM8750_ALC2 0x12 ++#define WM8750_ALC3 0x13 ++#define WM8750_NGATE 0x14 ++#define WM8750_LADC 0x15 ++#define WM8750_RADC 0x16 ++#define WM8750_ADCTL1 0x17 ++#define WM8750_ADCTL2 0x18 ++#define WM8750_PWR1 0x19 ++#define WM8750_PWR2 0x1a ++#define WM8750_ADCTL3 0x1b ++#define WM8750_ADCIN 0x1f ++#define WM8750_LADCIN 0x20 ++#define WM8750_RADCIN 0x21 ++#define WM8750_LOUTM1 0x22 ++#define WM8750_LOUTM2 0x23 ++#define WM8750_ROUTM1 0x24 ++#define WM8750_ROUTM2 0x25 ++#define WM8750_MOUTM1 0x26 ++#define WM8750_MOUTM2 0x27 ++#define WM8750_LOUT2V 0x28 ++#define WM8750_ROUT2V 0x29 ++#define WM8750_MOUTV 0x2a ++ ++#define WM8750_CACHE_REGNUM 0x2a ++ ++struct wm8750_setup_data { ++ unsigned short i2c_address; ++ unsigned int mclk; ++}; ++ ++extern struct snd_soc_codec_dai wm8750_dai; ++extern struct snd_soc_codec_device soc_codec_dev_wm8750; ++ ++#endif +Index: linux-2.6-pxa-new/sound/soc/codecs/wm8753.c +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/sound/soc/codecs/wm8753.c +@@ -0,0 +1,2128 @@ ++/* ++ * wm8753.c -- WM8753 ALSA Soc Audio driver ++ * ++ * Copyright 2003 Wolfson Microelectronics PLC. ++ * Author: Liam Girdwood ++ * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com ++ * ++ * This program is free software; you can redistribute it and/or modify it ++ * under the terms of the GNU General Public License as published by the ++ * Free Software Foundation; either version 2 of the License, or (at your ++ * option) any later version. ++ * ++ * Notes: ++ * The WM8753 is a low power, high quality stereo codec with integrated PCM ++ * codec designed for portable digital telephony applications. ++ * ++ * Dual DAI:- ++ * ++ * This driver support 2 DAI PCM's. This makes the default PCM available for ++ * HiFi audio (e.g. MP3, ogg) playback/capture and the other PCM available for ++ * voice. ++ * ++ * Please note that the voice PCM can be connected directly to a Bluetooth ++ * codec or GSM modem and thus cannot be read or written to, although it is ++ * available to be configured with snd_hw_params(), etc and kcontrols in the ++ * normal alsa manner. ++ * ++ * Fast DAI switching:- ++ * ++ * The driver can now fast switch between the DAI configurations via a ++ * an alsa kcontrol. This allows the PCM to remain open. ++ * ++ */ ++ ++#include <linux/module.h> ++#include <linux/moduleparam.h> ++#include <linux/version.h> ++#include <linux/kernel.h> ++#include <linux/init.h> ++#include <linux/delay.h> ++#include <linux/pm.h> ++#include <linux/i2c.h> ++#include <linux/platform_device.h> ++#include <sound/driver.h> ++#include <sound/core.h> ++#include <sound/pcm.h> ++#include <sound/pcm_params.h> ++#include <sound/soc.h> ++#include <sound/soc-dapm.h> ++#include <sound/initval.h> ++ ++#include "wm8753.h" ++ ++#define AUDIO_NAME "wm8753" ++#define WM8753_VERSION "0.16" ++ ++/* ++ * Debug ++ */ ++ ++#define WM8753_DEBUG 0 ++ ++#ifdef WM8753_DEBUG ++#define dbg(format, arg...) \ ++ printk(KERN_DEBUG AUDIO_NAME ": " format "\n" , ## arg) ++#else ++#define dbg(format, arg...) do {} while (0) ++#endif ++#define err(format, arg...) \ ++ printk(KERN_ERR AUDIO_NAME ": " format "\n" , ## arg) ++#define info(format, arg...) \ ++ printk(KERN_INFO AUDIO_NAME ": " format "\n" , ## arg) ++#define warn(format, arg...) \ ++ printk(KERN_WARNING AUDIO_NAME ": " format "\n" , ## arg) ++ ++static int caps_charge = 2000; ++module_param(caps_charge, int, 0); ++MODULE_PARM_DESC(caps_charge, "WM8753 cap charge time (msecs)"); ++ ++static struct workqueue_struct *wm8753_workq = NULL; ++static struct work_struct wm8753_dapm_work; ++static void wm8753_set_dai_mode(struct snd_soc_codec *codec, ++ unsigned int mode); ++ ++/* ++ * wm8753 register cache ++ * We can't read the WM8753 register space when we ++ * are using 2 wire for device control, so we cache them instead. ++ */ ++static const u16 wm8753_reg[] = { ++ 0x0008, 0x0000, 0x000a, 0x000a, ++ 0x0033, 0x0000, 0x0007, 0x00ff, ++ 0x00ff, 0x000f, 0x000f, 0x007b, ++ 0x0000, 0x0032, 0x0000, 0x00c3, ++ 0x00c3, 0x00c0, 0x0000, 0x0000, ++ 0x0000, 0x0000, 0x0000, 0x0000, ++ 0x0000, 0x0000, 0x0000, 0x0000, ++ 0x0000, 0x0000, 0x0000, 0x0055, ++ 0x0005, 0x0050, 0x0055, 0x0050, ++ 0x0055, 0x0050, 0x0055, 0x0079, ++ 0x0079, 0x0079, 0x0079, 0x0079, ++ 0x0000, 0x0000, 0x0000, 0x0000, ++ 0x0097, 0x0097, 0x0000, 0x0004, ++ 0x0000, 0x0083, 0x0024, 0x01ba, ++ 0x0000, 0x0083, 0x0024, 0x01ba, ++ 0x0000, 0x0000 ++}; ++ ++#define WM8753_DAIFMT \ ++ (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_RIGHT_J | \ ++ SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF | \ ++ SND_SOC_DAIFMT_NB_IF | SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_IB_IF) ++ ++#define WM8753_DIR \ ++ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE) ++ ++#define WM8753_HIFI_FSB \ ++ (SND_SOC_FSBD(1) | SND_SOC_FSBD(2) | SND_SOC_FSBD(4) | \ ++ SND_SOC_FSBD(8) | SND_SOC_FSBD(16)) ++ ++#define WM8753_HIFI_RATES \ ++ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ ++ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ ++ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) ++ ++#define WM8753_HIFI_BITS \ ++ (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ ++ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) ++ ++/* ++ * HiFi modes ++ */ ++static struct snd_soc_dai_mode wm8753_hifi_modes[] = { ++ /* codec frame and clock master modes */ ++ /* 8k */ ++ { ++ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS, ++ .pcmfmt = WM8753_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_8000, ++ .pcmdir = WM8753_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 1536, ++ .bfs = WM8753_HIFI_FSB, ++ }, ++ { ++ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS, ++ .pcmfmt = WM8753_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_8000, ++ .pcmdir = WM8753_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 1408, ++ .bfs = WM8753_HIFI_FSB, ++ }, ++ { ++ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS, ++ .pcmfmt = WM8753_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_8000, ++ .pcmdir = WM8753_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 2304, ++ .bfs = WM8753_HIFI_FSB, ++ }, ++ { ++ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS, ++ .pcmfmt = WM8753_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_8000, ++ .pcmdir = WM8753_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 2112, ++ .bfs = WM8753_HIFI_FSB, ++ }, ++ { ++ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS, ++ .pcmfmt = WM8753_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_8000, ++ .pcmdir = WM8753_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 1500, ++ .bfs = WM8753_HIFI_FSB, ++ }, ++ ++ /* 11.025k */ ++ { ++ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS, ++ .pcmfmt = WM8753_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_11025, ++ .pcmdir = WM8753_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 1024, ++ .bfs = WM8753_HIFI_FSB, ++ }, ++ { ++ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS, ++ .pcmfmt = WM8753_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_11025, ++ .pcmdir = WM8753_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 1536, ++ .bfs = WM8753_HIFI_FSB, ++ }, ++ { ++ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS, ++ .pcmfmt = WM8753_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_11025, ++ .pcmdir = WM8753_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 1088, ++ .bfs = WM8753_HIFI_FSB, ++ }, ++ ++ /* 16k */ ++ { ++ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS, ++ .pcmfmt = WM8753_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_16000, ++ .pcmdir = WM8753_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 768, ++ .bfs = WM8753_HIFI_FSB, ++ }, ++ { ++ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS, ++ .pcmfmt= WM8753_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_16000, ++ .pcmdir = WM8753_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 1152, ++ .bfs = WM8753_HIFI_FSB, ++ }, ++ { ++ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS, ++ .pcmfmt = WM8753_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_16000, ++ .pcmdir = WM8753_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 750, ++ .bfs = WM8753_HIFI_FSB, ++ }, ++ ++ /* 22.05k */ ++ { ++ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS, ++ .pcmfmt = WM8753_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_22050, ++ .pcmdir = WM8753_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 512, ++ .bfs = WM8753_HIFI_FSB, ++ }, ++ { ++ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS, ++ .pcmfmt = WM8753_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_22050, ++ .pcmdir = WM8753_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 768, ++ .bfs = WM8753_HIFI_FSB, ++ }, ++ { ++ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS, ++ .pcmfmt = WM8753_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_22050, ++ .pcmdir = WM8753_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 544, ++ .bfs = WM8753_HIFI_FSB, ++ }, ++ ++ /* 32k */ ++ { ++ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS, ++ .pcmfmt = WM8753_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_32000, ++ .pcmdir = WM8753_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 384, ++ .bfs = WM8753_HIFI_FSB, ++ }, ++ { ++ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS, ++ .pcmfmt = WM8753_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_32000, ++ .pcmdir = WM8753_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 576, ++ .bfs = WM8753_HIFI_FSB, ++ }, ++ { ++ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS, ++ .pcmfmt = WM8753_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_32000, ++ .pcmdir = WM8753_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 375, ++ .bfs = WM8753_HIFI_FSB, ++ }, ++ ++ /* 44.1k & 48k */ ++ { ++ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS, ++ .pcmfmt = WM8753_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000, ++ .pcmdir = WM8753_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 256, ++ .bfs = WM8753_HIFI_FSB, ++ }, ++ { ++ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS, ++ .pcmfmt = WM8753_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000, ++ .pcmdir = WM8753_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 384, ++ .bfs = WM8753_HIFI_FSB, ++ }, ++ { ++ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS, ++ .pcmfmt = WM8753_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_48000, ++ .pcmdir = WM8753_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 250, ++ .bfs = WM8753_HIFI_FSB, ++ }, ++ { ++ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS, ++ .pcmfmt = WM8753_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_44100, ++ .pcmdir = WM8753_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 272, ++ .bfs = WM8753_HIFI_FSB, ++ }, ++ ++ /* 88.2k & 96k */ ++ { ++ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS, ++ .pcmfmt = WM8753_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000, ++ .pcmdir = WM8753_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 128, ++ .bfs = WM8753_HIFI_FSB, ++ }, ++ { ++ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS, ++ .pcmfmt = WM8753_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000, ++ .pcmdir = WM8753_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 192, ++ .bfs = WM8753_HIFI_FSB, ++ }, ++ { ++ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS, ++ .pcmfmt = WM8753_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_88200, ++ .pcmdir = WM8753_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 136, ++ .bfs = WM8753_HIFI_FSB, ++ }, ++ { ++ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS, ++ .pcmfmt = WM8753_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_96000, ++ .pcmdir = WM8753_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 125, ++ .bfs = WM8753_HIFI_FSB, ++ }, ++ ++ /* codec frame and clock slave modes */ ++ { ++ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBS_CFS, ++ .pcmfmt = WM8753_HIFI_BITS, ++ .pcmrate = WM8753_HIFI_RATES, ++ .pcmdir = WM8753_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = SND_SOC_FS_ALL, ++ .bfs = SND_SOC_FSB_ALL, ++ }, ++}; ++ ++#define WM8753_VOICE_FSB \ ++ (SND_SOC_FSBD(1) | SND_SOC_FSBD(2) | SND_SOC_FSBD(4) | \ ++ SND_SOC_FSBD(8) | SND_SOC_FSBD(16)) ++ ++#define WM8753_VOICE_RATES \ ++ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ ++ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ ++ SNDRV_PCM_RATE_48000) ++ ++#define WM8753_VOICE_BITS \ ++ (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ ++ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) ++ ++/* ++ * Voice modes ++ */ ++static struct snd_soc_dai_mode wm8753_voice_modes[] = { ++ ++ /* master modes */ ++ { ++ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS, ++ .pcmfmt = WM8753_VOICE_BITS, ++ .pcmrate = WM8753_VOICE_RATES, ++ .pcmdir = WM8753_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 256, ++ .bfs = WM8753_VOICE_FSB, ++ }, ++ { ++ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS, ++ .pcmfmt = WM8753_VOICE_BITS, ++ .pcmrate = WM8753_VOICE_RATES, ++ .pcmdir = WM8753_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 384, ++ .bfs = WM8753_VOICE_FSB, ++ }, ++ ++ /* slave modes */ ++ { ++ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBS_CFS, ++ .pcmfmt = WM8753_VOICE_BITS, ++ .pcmrate = WM8753_VOICE_RATES, ++ .pcmdir = WM8753_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = SND_SOC_FS_ALL, ++ .bfs = SND_SOC_FSB_ALL, ++ }, ++}; ++ ++ ++/* ++ * Mode 4 ++ */ ++static struct snd_soc_dai_mode wm8753_mixed_modes[] = { ++ /* slave modes */ ++ { ++ .fmt = WM8753_DAIFMT | SND_SOC_DAIFMT_CBS_CFS, ++ .pcmfmt = WM8753_HIFI_BITS, ++ .pcmrate = WM8753_HIFI_RATES, ++ .pcmdir = WM8753_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = SND_SOC_FS_ALL, ++ .bfs = SND_SOC_FSB_ALL, ++ }, ++}; ++ ++/* ++ * read wm8753 register cache ++ */ ++static inline unsigned int wm8753_read_reg_cache(struct snd_soc_codec *codec, ++ unsigned int reg) ++{ ++ u16 *cache = codec->reg_cache; ++ if (reg < 1 || reg > (ARRAY_SIZE(wm8753_reg) + 1)) ++ return -1; ++ return cache[reg - 1]; ++} ++ ++/* ++ * write wm8753 register cache ++ */ ++static inline void wm8753_write_reg_cache(struct snd_soc_codec *codec, ++ unsigned int reg, unsigned int value) ++{ ++ u16 *cache = codec->reg_cache; ++ if (reg < 1 || reg > 0x3f) ++ return; ++ cache[reg - 1] = value; ++} ++ ++/* ++ * write to the WM8753 register space ++ */ ++static int wm8753_write(struct snd_soc_codec *codec, unsigned int reg, ++ unsigned int value) ++{ ++ u8 data[2]; ++ ++ /* data is ++ * D15..D9 WM8753 register offset ++ * D8...D0 register data ++ */ ++ data[0] = (reg << 1) | ((value >> 8) & 0x0001); ++ data[1] = value & 0x00ff; ++ ++ wm8753_write_reg_cache (codec, reg, value); ++ if (codec->hw_write(codec->control_data, data, 2) == 2) ++ return 0; ++ else ++ return -EIO; ++} ++ ++#define wm8753_reset(c) wm8753_write(c, WM8753_RESET, 0) ++ ++/* ++ * WM8753 Controls ++ */ ++static const char *wm8753_base[] = {"Linear Control", "Adaptive Boost"}; ++static const char *wm8753_base_filter[] = ++ {"130Hz @ 48kHz", "200Hz @ 48kHz", "100Hz @ 16kHz", "400Hz @ 48kHz", ++ "100Hz @ 8kHz", "200Hz @ 8kHz"}; ++static const char *wm8753_treble[] = {"8kHz", "4kHz"}; ++static const char *wm8753_alc_func[] = {"Off", "Right", "Left", "Stereo"}; ++static const char *wm8753_ng_type[] = {"Constant PGA Gain", "Mute ADC Output"}; ++static const char *wm8753_3d_func[] = {"Capture", "Playback"}; ++static const char *wm8753_3d_uc[] = {"2.2kHz", "1.5kHz"}; ++static const char *wm8753_3d_lc[] = {"200Hz", "500Hz"}; ++static const char *wm8753_deemp[] = {"None", "32kHz", "44.1kHz", "48kHz"}; ++static const char *wm8753_mono_mix[] = {"Stereo", "Left", "Right", "Mono"}; ++static const char *wm8753_dac_phase[] = {"Non Inverted", "Inverted"}; ++static const char *wm8753_line_mix[] = {"Line 1 + 2", "Line 1 - 2", ++ "Line 1", "Line 2"}; ++static const char *wm8753_mono_mux[] = {"Line Mix", "Rx Mix"}; ++static const char *wm8753_right_mux[] = {"Line 2", "Rx Mix"}; ++static const char *wm8753_left_mux[] = {"Line 1", "Rx Mix"}; ++static const char *wm8753_rxmsel[] = {"RXP - RXN", "RXP + RXN", "RXP", "RXN"}; ++static const char *wm8753_sidetone_mux[] = {"Left PGA", "Mic 1", "Mic 2", ++ "Right PGA"}; ++static const char *wm8753_mono2_src[] = {"Inverted Mono 1", "Left", "Right", ++ "Left + Right"}; ++static const char *wm8753_out3[] = {"VREF", "ROUT2", "Left + Right"}; ++static const char *wm8753_out4[] = {"VREF", "Capture ST", "LOUT2"}; ++static const char *wm8753_radcsel[] = {"PGA", "Line or RXP-RXN", "Sidetone"}; ++static const char *wm8753_ladcsel[] = {"PGA", "Line or RXP-RXN", "Line"}; ++static const char *wm8753_mono_adc[] = {"Stereo", "Analogue Mix Left", ++ "Analogue Mix Right", "Digital Mono Mix"}; ++static const char *wm8753_adc_hp[] = {"3.4Hz @ 48kHz", "82Hz @ 16k", ++ "82Hz @ 8kHz", "170Hz @ 8kHz"}; ++static const char *wm8753_adc_filter[] = {"HiFi", "Voice"}; ++static const char *wm8753_mic_sel[] = {"Mic 1", "Mic 2", "Mic 3"}; ++static const char *wm8753_dai_mode[] = {"DAI 0", "DAI 1", "DAI 2", "DAI 3"}; ++ ++static const struct soc_enum wm8753_enum[] = { ++SOC_ENUM_SINGLE(WM8753_BASS, 7, 2, wm8753_base), // 0 ++SOC_ENUM_SINGLE(WM8753_BASS, 4, 6, wm8753_base_filter), // 1 ++SOC_ENUM_SINGLE(WM8753_TREBLE, 6, 2, wm8753_treble), // 2 ++SOC_ENUM_SINGLE(WM8753_ALC1, 7, 4, wm8753_alc_func), // 3 ++SOC_ENUM_SINGLE(WM8753_NGATE, 1, 2, wm8753_ng_type), // 4 ++SOC_ENUM_SINGLE(WM8753_3D, 7, 2, wm8753_3d_func), // 5 ++SOC_ENUM_SINGLE(WM8753_3D, 6, 2, wm8753_3d_uc), // 6 ++SOC_ENUM_SINGLE(WM8753_3D, 5, 2, wm8753_3d_lc), // 7 ++SOC_ENUM_SINGLE(WM8753_DAC, 1, 4, wm8753_deemp), // 8 ++SOC_ENUM_SINGLE(WM8753_DAC, 4, 4, wm8753_mono_mix), // 9 ++SOC_ENUM_SINGLE(WM8753_DAC, 6, 2, wm8753_dac_phase), // 10 ++SOC_ENUM_SINGLE(WM8753_INCTL1, 3, 4, wm8753_line_mix), // 11 ++SOC_ENUM_SINGLE(WM8753_INCTL1, 2, 2, wm8753_mono_mux), // 12 ++SOC_ENUM_SINGLE(WM8753_INCTL1, 1, 2, wm8753_right_mux), // 13 ++SOC_ENUM_SINGLE(WM8753_INCTL1, 0, 2, wm8753_left_mux), // 14 ++SOC_ENUM_SINGLE(WM8753_INCTL2, 6, 4, wm8753_rxmsel), // 15 ++SOC_ENUM_SINGLE(WM8753_INCTL2, 4, 4, wm8753_sidetone_mux),// 16 ++SOC_ENUM_SINGLE(WM8753_OUTCTL, 7, 4, wm8753_mono2_src), // 17 ++SOC_ENUM_SINGLE(WM8753_OUTCTL, 0, 3, wm8753_out3), // 18 ++SOC_ENUM_SINGLE(WM8753_ADCTL2, 7, 3, wm8753_out4), // 19 ++SOC_ENUM_SINGLE(WM8753_ADCIN, 2, 3, wm8753_radcsel), // 20 ++SOC_ENUM_SINGLE(WM8753_ADCIN, 0, 3, wm8753_ladcsel), // 21 ++SOC_ENUM_SINGLE(WM8753_ADCIN, 4, 4, wm8753_mono_adc), // 22 ++SOC_ENUM_SINGLE(WM8753_ADC, 2, 4, wm8753_adc_hp), // 23 ++SOC_ENUM_SINGLE(WM8753_ADC, 4, 2, wm8753_adc_filter), // 24 ++SOC_ENUM_SINGLE(WM8753_MICBIAS, 6, 3, wm8753_mic_sel), // 25 ++SOC_ENUM_SINGLE(WM8753_IOCTL, 2, 4, wm8753_dai_mode), // 26 ++}; ++ ++ ++static int wm8753_get_dai(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_value *ucontrol) ++{ ++ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); ++ int mode = wm8753_read_reg_cache(codec, WM8753_IOCTL); ++ ++ ucontrol->value.integer.value[0] = (mode & 0xc) >> 2; ++ return 0; ++} ++ ++static int wm8753_set_dai(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_value *ucontrol) ++{ ++ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); ++ int mode = wm8753_read_reg_cache(codec, WM8753_IOCTL); ++ ++ if (((mode &0xc) >> 2) == ucontrol->value.integer.value[0]) ++ return 0; ++ ++ mode &= 0xfff3; ++ mode |= (ucontrol->value.integer.value[0] << 2); ++ ++ wm8753_write(codec, WM8753_IOCTL, mode); ++ wm8753_set_dai_mode(codec, ucontrol->value.integer.value[0]); ++ return 1; ++} ++ ++static const struct snd_kcontrol_new wm8753_snd_controls[] = { ++SOC_DOUBLE_R("PCM Volume", WM8753_LDAC, WM8753_RDAC, 0, 255, 0), ++ ++SOC_DOUBLE_R("ADC Capture Volume", WM8753_LADC, WM8753_RADC, 0, 63, 0), ++SOC_DOUBLE_R("ADC Capture Switch", WM8753_LINVOL, WM8753_RINVOL, 7, 1, 0), ++SOC_DOUBLE_R("ADC Capture ZC Switch", WM8753_LINVOL, WM8753_RINVOL, 6, 1, 0), ++ ++SOC_DOUBLE_R("Headphone Playback Volume", WM8753_LOUT1V, WM8753_ROUT1V, 0, 127, 0), ++SOC_DOUBLE_R("Speaker Playback Volume", WM8753_LOUT2V, WM8753_ROUT2V, 0, 127, 0), ++ ++SOC_SINGLE("Mono Playback Volume", WM8753_MOUTV, 0, 127, 0), ++ ++SOC_DOUBLE_R("Bypass Playback Volume", WM8753_LOUTM1, WM8753_ROUTM1, 4, 7, 1), ++SOC_DOUBLE_R("Sidetone Playback Volume", WM8753_LOUTM2, WM8753_ROUTM2, 4, 7, 1), ++SOC_DOUBLE_R("Voice Playback Volume", WM8753_LOUTM2, WM8753_ROUTM2, 0, 7, 1), ++ ++SOC_DOUBLE_R("Headphone Playback ZC Switch", WM8753_LOUT1V, WM8753_ROUT1V, 7, 1, 0), ++SOC_DOUBLE_R("Speaker Playback ZC Switch", WM8753_LOUT2V, WM8753_ROUT2V, 7, 1, 0), ++ ++SOC_SINGLE("Mono Bypass Playback Volume", WM8753_MOUTM1, 4, 7, 1), ++SOC_SINGLE("Mono Sidetone Playback Volume", WM8753_MOUTM2, 4, 7, 1), ++SOC_SINGLE("Mono Voice Playback Volume", WM8753_MOUTM2, 4, 7, 1), ++SOC_SINGLE("Mono Playback ZC Switch", WM8753_MOUTV, 7, 1, 0), ++ ++SOC_ENUM("Bass Boost", wm8753_enum[0]), ++SOC_ENUM("Bass Filter", wm8753_enum[1]), ++SOC_SINGLE("Bass Volume", WM8753_BASS, 0, 7, 1), ++ ++SOC_SINGLE("Treble Volume", WM8753_TREBLE, 0, 7, 0), ++SOC_ENUM("Treble Cut-off", wm8753_enum[2]), ++ ++SOC_DOUBLE("Sidetone Capture Volume", WM8753_RECMIX1, 0, 4, 7, 1), ++SOC_SINGLE("Voice Sidetone Capture Volume", WM8753_RECMIX2, 0, 7, 1), ++ ++SOC_DOUBLE_R("Capture Volume", WM8753_LINVOL, WM8753_RINVOL, 0, 63, 0), ++SOC_DOUBLE_R("Capture ZC Switch", WM8753_LINVOL, WM8753_RINVOL, 6, 1, 0), ++SOC_DOUBLE_R("Capture Switch", WM8753_LINVOL, WM8753_RINVOL, 7, 1, 0), ++ ++SOC_ENUM("Capture Filter Select", wm8753_enum[23]), ++SOC_ENUM("Capture Filter Cut-off", wm8753_enum[24]), ++SOC_SINGLE("Capture Filter Switch", WM8753_ADC, 0, 1, 1), ++ ++SOC_SINGLE("ALC Capture Target Volume", WM8753_ALC1, 0, 7, 0), ++SOC_SINGLE("ALC Capture Max Volume", WM8753_ALC1, 4, 7, 0), ++SOC_ENUM("ALC Capture Function", wm8753_enum[3]), ++SOC_SINGLE("ALC Capture ZC Switch", WM8753_ALC2, 8, 1, 0), ++SOC_SINGLE("ALC Capture Hold Time", WM8753_ALC2, 0, 15, 1), ++SOC_SINGLE("ALC Capture Decay Time", WM8753_ALC3, 4, 15, 1), ++SOC_SINGLE("ALC Capture Attack Time", WM8753_ALC3, 0, 15, 0), ++SOC_SINGLE("ALC Capture NG Threshold", WM8753_NGATE, 3, 31, 0), ++SOC_ENUM("ALC Capture NG Type", wm8753_enum[4]), ++SOC_SINGLE("ALC Capture NG Switch", WM8753_NGATE, 0, 1, 0), ++ ++SOC_ENUM("3D Function", wm8753_enum[5]), ++SOC_ENUM("3D Upper Cut-off", wm8753_enum[6]), ++SOC_ENUM("3D Lower Cut-off", wm8753_enum[7]), ++SOC_SINGLE("3D Volume", WM8753_3D, 1, 15, 0), ++SOC_SINGLE("3D Switch", WM8753_3D, 0, 1, 0), ++ ++SOC_SINGLE("Capture 6dB Attenuate", WM8753_ADCTL1, 2, 1, 0), ++SOC_SINGLE("Playback 6dB Attenuate", WM8753_ADCTL1, 1, 1, 0), ++ ++SOC_ENUM("De-emphasis", wm8753_enum[8]), ++SOC_ENUM("Playback Mono Mix", wm8753_enum[9]), ++SOC_ENUM("Playback Phase", wm8753_enum[10]), ++ ++SOC_SINGLE("Mic2 Capture Volume", WM8753_INCTL1, 7, 3, 0), ++SOC_SINGLE("Mic1 Capture Volume", WM8753_INCTL1, 5, 3, 0), ++ ++SOC_ENUM_EXT("DAI Mode", wm8753_enum[26], wm8753_get_dai, wm8753_set_dai), ++}; ++ ++/* add non dapm controls */ ++static int wm8753_add_controls(struct snd_soc_codec *codec) ++{ ++ int err, i; ++ ++ for (i = 0; i < ARRAY_SIZE(wm8753_snd_controls); i++) { ++ err = snd_ctl_add(codec->card, ++ snd_soc_cnew(&wm8753_snd_controls[i],codec, NULL)); ++ if (err < 0) ++ return err; ++ } ++ return 0; ++} ++ ++/* ++ * _DAPM_ Controls ++ */ ++ ++/* Left Mixer */ ++static const struct snd_kcontrol_new wm8753_left_mixer_controls[] = { ++SOC_DAPM_SINGLE("Voice Playback Switch", WM8753_LOUTM2, 8, 1, 0), ++SOC_DAPM_SINGLE("Sidetone Playback Switch", WM8753_LOUTM2, 7, 1, 0), ++SOC_DAPM_SINGLE("Left Playback Switch", WM8753_LOUTM1, 8, 1, 0), ++SOC_DAPM_SINGLE("Bypass Playback Switch", WM8753_LOUTM1, 7, 1, 0), ++}; ++ ++/* Right mixer */ ++static const struct snd_kcontrol_new wm8753_right_mixer_controls[] = { ++SOC_DAPM_SINGLE("Voice Playback Switch", WM8753_ROUTM2, 8, 1, 0), ++SOC_DAPM_SINGLE("Sidetone Playback Switch", WM8753_ROUTM2, 7, 1, 0), ++SOC_DAPM_SINGLE("Right Playback Switch", WM8753_ROUTM1, 8, 1, 0), ++SOC_DAPM_SINGLE("Bypass Playback Switch", WM8753_ROUTM1, 7, 1, 0), ++}; ++ ++/* Mono mixer */ ++static const struct snd_kcontrol_new wm8753_mono_mixer_controls[] = { ++SOC_DAPM_SINGLE("Left Playback Switch", WM8753_MOUTM1, 8, 1, 0), ++SOC_DAPM_SINGLE("Right Playback Switch", WM8753_MOUTM2, 8, 1, 0), ++SOC_DAPM_SINGLE("Voice Playback Switch", WM8753_MOUTM2, 3, 1, 0), ++SOC_DAPM_SINGLE("Sidetone Playback Switch", WM8753_MOUTM2, 7, 1, 0), ++SOC_DAPM_SINGLE("Bypass Playback Switch", WM8753_MOUTM1, 7, 1, 0), ++}; ++ ++/* Mono 2 Mux */ ++static const struct snd_kcontrol_new wm8753_mono2_controls = ++SOC_DAPM_ENUM("Route", wm8753_enum[17]); ++ ++/* Out 3 Mux */ ++static const struct snd_kcontrol_new wm8753_out3_controls = ++SOC_DAPM_ENUM("Route", wm8753_enum[18]); ++ ++/* Out 4 Mux */ ++static const struct snd_kcontrol_new wm8753_out4_controls = ++SOC_DAPM_ENUM("Route", wm8753_enum[19]); ++ ++/* ADC Mono Mix */ ++static const struct snd_kcontrol_new wm8753_adc_mono_controls = ++SOC_DAPM_ENUM("Route", wm8753_enum[22]); ++ ++/* Record mixer */ ++static const struct snd_kcontrol_new wm8753_record_mixer_controls[] = { ++SOC_DAPM_SINGLE("Voice Capture Switch", WM8753_RECMIX2, 3, 1, 0), ++SOC_DAPM_SINGLE("Left Capture Switch", WM8753_RECMIX1, 3, 1, 0), ++SOC_DAPM_SINGLE("Right Capture Switch", WM8753_RECMIX1, 7, 1, 0), ++}; ++ ++/* Left ADC mux */ ++static const struct snd_kcontrol_new wm8753_adc_left_controls = ++SOC_DAPM_ENUM("Route", wm8753_enum[21]); ++ ++/* Right ADC mux */ ++static const struct snd_kcontrol_new wm8753_adc_right_controls = ++SOC_DAPM_ENUM("Route", wm8753_enum[20]); ++ ++/* MIC mux */ ++static const struct snd_kcontrol_new wm8753_mic_mux_controls = ++SOC_DAPM_ENUM("Route", wm8753_enum[16]); ++ ++/* ALC mixer */ ++static const struct snd_kcontrol_new wm8753_alc_mixer_controls[] = { ++SOC_DAPM_SINGLE("Line Capture Switch", WM8753_INCTL2, 3, 1, 0), ++SOC_DAPM_SINGLE("Mic2 Capture Switch", WM8753_INCTL2, 2, 1, 0), ++SOC_DAPM_SINGLE("Mic1 Capture Switch", WM8753_INCTL2, 1, 1, 0), ++SOC_DAPM_SINGLE("Rx Capture Switch", WM8753_INCTL2, 0, 1, 0), ++}; ++ ++/* Left Line mux */ ++static const struct snd_kcontrol_new wm8753_line_left_controls = ++SOC_DAPM_ENUM("Route", wm8753_enum[14]); ++ ++/* Right Line mux */ ++static const struct snd_kcontrol_new wm8753_line_right_controls = ++SOC_DAPM_ENUM("Route", wm8753_enum[13]); ++ ++/* Mono Line mux */ ++static const struct snd_kcontrol_new wm8753_line_mono_controls = ++SOC_DAPM_ENUM("Route", wm8753_enum[12]); ++ ++/* Line mux and mixer */ ++static const struct snd_kcontrol_new wm8753_line_mux_mix_controls = ++SOC_DAPM_ENUM("Route", wm8753_enum[11]); ++ ++/* Rx mux and mixer */ ++static const struct snd_kcontrol_new wm8753_rx_mux_mix_controls = ++SOC_DAPM_ENUM("Route", wm8753_enum[15]); ++ ++/* Mic Selector Mux */ ++static const struct snd_kcontrol_new wm8753_mic_sel_mux_controls = ++SOC_DAPM_ENUM("Route", wm8753_enum[25]); ++ ++static const struct snd_soc_dapm_widget wm8753_dapm_widgets[] = { ++SND_SOC_DAPM_MICBIAS("Mic Bias", WM8753_PWR1, 5, 0), ++SND_SOC_DAPM_MIXER("Left Mixer", WM8753_PWR4, 0, 0, ++ &wm8753_left_mixer_controls[0], ARRAY_SIZE(wm8753_left_mixer_controls)), ++SND_SOC_DAPM_PGA("Left Out 1", WM8753_PWR3, 8, 0, NULL, 0), ++SND_SOC_DAPM_PGA("Left Out 2", WM8753_PWR3, 6, 0, NULL, 0), ++SND_SOC_DAPM_DAC("Left DAC", "Left HiFi Playback", WM8753_PWR1, 3, 0), ++SND_SOC_DAPM_OUTPUT("LOUT1"), ++SND_SOC_DAPM_OUTPUT("LOUT2"), ++SND_SOC_DAPM_MIXER("Right Mixer", WM8753_PWR4, 1, 0, ++ &wm8753_right_mixer_controls[0], ARRAY_SIZE(wm8753_right_mixer_controls)), ++SND_SOC_DAPM_PGA("Right Out 1", WM8753_PWR3, 7, 0, NULL, 0), ++SND_SOC_DAPM_PGA("Right Out 2", WM8753_PWR3, 5, 0, NULL, 0), ++SND_SOC_DAPM_DAC("Right DAC", "Right HiFi Playback", WM8753_PWR1, 2, 0), ++SND_SOC_DAPM_OUTPUT("ROUT1"), ++SND_SOC_DAPM_OUTPUT("ROUT2"), ++SND_SOC_DAPM_MIXER("Mono Mixer", WM8753_PWR4, 2, 0, ++ &wm8753_mono_mixer_controls[0], ARRAY_SIZE(wm8753_mono_mixer_controls)), ++SND_SOC_DAPM_PGA("Mono Out 1", WM8753_PWR3, 2, 0, NULL, 0), ++SND_SOC_DAPM_PGA("Mono Out 2", WM8753_PWR3, 1, 0, NULL, 0), ++SND_SOC_DAPM_DAC("Voice DAC", "Voice Playback", WM8753_PWR1, 4, 0), ++SND_SOC_DAPM_OUTPUT("MONO1"), ++SND_SOC_DAPM_MUX("Mono 2 Mux", SND_SOC_NOPM, 0, 0, &wm8753_mono2_controls), ++SND_SOC_DAPM_OUTPUT("MONO2"), ++SND_SOC_DAPM_MIXER("Out3 Left + Right", -1, 0, 0, NULL, 0), ++SND_SOC_DAPM_MUX("Out3 Mux", SND_SOC_NOPM, 0, 0, &wm8753_out3_controls), ++SND_SOC_DAPM_PGA("Out 3", WM8753_PWR3, 4, 0, NULL, 0), ++SND_SOC_DAPM_OUTPUT("OUT3"), ++SND_SOC_DAPM_MUX("Out4 Mux", SND_SOC_NOPM, 0, 0, &wm8753_out4_controls), ++SND_SOC_DAPM_PGA("Out 4", WM8753_PWR3, 3, 0, NULL, 0), ++SND_SOC_DAPM_OUTPUT("OUT4"), ++SND_SOC_DAPM_MIXER("Playback Mixer", WM8753_PWR4, 3, 0, ++ &wm8753_record_mixer_controls[0], ++ ARRAY_SIZE(wm8753_record_mixer_controls)), ++SND_SOC_DAPM_ADC("Left ADC", "Left Voice Capture", WM8753_PWR2, 3, 0), ++SND_SOC_DAPM_ADC("Right ADC", "Right Voice Capture", WM8753_PWR2, 2, 0), ++SND_SOC_DAPM_MUX("Capture Left Mixer", SND_SOC_NOPM, 0, 0, ++ &wm8753_adc_mono_controls), ++SND_SOC_DAPM_MUX("Capture Right Mixer", SND_SOC_NOPM, 0, 0, ++ &wm8753_adc_mono_controls), ++SND_SOC_DAPM_MUX("Capture Left Mux", SND_SOC_NOPM, 0, 0, ++ &wm8753_adc_left_controls), ++SND_SOC_DAPM_MUX("Capture Right Mux", SND_SOC_NOPM, 0, 0, ++ &wm8753_adc_right_controls), ++SND_SOC_DAPM_MUX("Mic Sidetone Mux", SND_SOC_NOPM, 0, 0, ++ &wm8753_mic_mux_controls), ++SND_SOC_DAPM_PGA("Left Capture Volume", WM8753_PWR2, 5, 0, NULL, 0), ++SND_SOC_DAPM_PGA("Right Capture Volume", WM8753_PWR2, 4, 0, NULL, 0), ++SND_SOC_DAPM_MIXER("ALC Mixer", WM8753_PWR2, 6, 0, ++ &wm8753_alc_mixer_controls[0], ARRAY_SIZE(wm8753_alc_mixer_controls)), ++SND_SOC_DAPM_MUX("Line Left Mux", SND_SOC_NOPM, 0, 0, ++ &wm8753_line_left_controls), ++SND_SOC_DAPM_MUX("Line Right Mux", SND_SOC_NOPM, 0, 0, ++ &wm8753_line_right_controls), ++SND_SOC_DAPM_MUX("Line Mono Mux", SND_SOC_NOPM, 0, 0, ++ &wm8753_line_mono_controls), ++SND_SOC_DAPM_MUX("Line Mixer", SND_SOC_NOPM, 0, 0, ++ &wm8753_line_mux_mix_controls), ++SND_SOC_DAPM_MUX("Rx Mixer", SND_SOC_NOPM, 0, 0, ++ &wm8753_rx_mux_mix_controls), ++SND_SOC_DAPM_PGA("Mic 1 Volume", WM8753_PWR2, 8, 0, NULL, 0), ++SND_SOC_DAPM_PGA("Mic 2 Volume", WM8753_PWR2, 7, 0, NULL, 0), ++SND_SOC_DAPM_MUX("Mic Selection Mux", SND_SOC_NOPM, 0, 0, ++ &wm8753_mic_sel_mux_controls), ++SND_SOC_DAPM_INPUT("LINE1"), ++SND_SOC_DAPM_INPUT("LINE2"), ++SND_SOC_DAPM_INPUT("RXP"), ++SND_SOC_DAPM_INPUT("RXN"), ++SND_SOC_DAPM_INPUT("ACIN"), ++SND_SOC_DAPM_INPUT("ACOP"), ++SND_SOC_DAPM_INPUT("MIC1N"), ++SND_SOC_DAPM_INPUT("MIC1"), ++SND_SOC_DAPM_INPUT("MIC2N"), ++SND_SOC_DAPM_INPUT("MIC2"), ++SND_SOC_DAPM_VMID("VREF"), ++}; ++ ++static const char *audio_map[][3] = { ++ /* left mixer */ ++ {"Left Mixer", "Left Playback Switch", "Left DAC"}, ++ {"Left Mixer", "Voice Playback Switch", "Voice DAC"}, ++ {"Left Mixer", "Sidetone Playback Switch", "Mic Sidetone Mux"}, ++ {"Left Mixer", "Bypass Playback Switch", "Line Left Mux"}, ++ ++ /* right mixer */ ++ {"Right Mixer", "Right Playback Switch", "Right DAC"}, ++ {"Right Mixer", "Voice Playback Switch", "Voice DAC"}, ++ {"Right Mixer", "Sidetone Playback Switch", "Mic Sidetone Mux"}, ++ {"Right Mixer", "Bypass Playback Switch", "Line Right Mux"}, ++ ++ /* mono mixer */ ++ {"Mono Mixer", "Voice Playback Switch", "Voice DAC"}, ++ {"Mono Mixer", "Left Playback Switch", "Left DAC"}, ++ {"Mono Mixer", "Right Playback Switch", "Right DAC"}, ++ {"Mono Mixer", "Sidetone Playback Switch", "Mic Sidetone Mux"}, ++ {"Mono Mixer", "Bypass Playback Switch", "Line Mono Mux"}, ++ ++ /* left out */ ++ {"Left Out 1", NULL, "Left Mixer"}, ++ {"Left Out 2", NULL, "Left Mixer"}, ++ {"LOUT1", NULL, "Left Out 1"}, ++ {"LOUT2", NULL, "Left Out 2"}, ++ ++ /* right out */ ++ {"Right Out 1", NULL, "Right Mixer"}, ++ {"Right Out 2", NULL, "Right Mixer"}, ++ {"ROUT1", NULL, "Right Out 1"}, ++ {"ROUT2", NULL, "Right Out 2"}, ++ ++ /* mono 1 out */ ++ {"Mono Out 1", NULL, "Mono Mixer"}, ++ {"MONO1", NULL, "Mono Out 1"}, ++ ++ /* mono 2 out */ ++ {"Mono 2 Mux", "Left + Right", "Out3 Left + Right"}, ++ {"Mono 2 Mux", "Inverted Mono 1", "MONO1"}, ++ {"Mono 2 Mux", "Left", "Left Mixer"}, ++ {"Mono 2 Mux", "Right", "Right Mixer"}, ++ {"Mono Out 2", NULL, "Mono 2 Mux"}, ++ {"MONO2", NULL, "Mono Out 2"}, ++ ++ /* out 3 */ ++ {"Out3 Left + Right", NULL, "Left Mixer"}, ++ {"Out3 Left + Right", NULL, "Right Mixer"}, ++ {"Out3 Mux", "VREF", "VREF"}, ++ {"Out3 Mux", "Left + Right", "Out3 Left + Right"}, ++ {"Out3 Mux", "ROUT2", "ROUT2"}, ++ {"Out 3", NULL, "Out3 Mux"}, ++ {"OUT3", NULL, "Out 3"}, ++ ++ /* out 4 */ ++ {"Out4 Mux", "VREF", "VREF"}, ++ {"Out4 Mux", "Capture ST", "Capture ST Mixer"}, ++ {"Out4 Mux", "LOUT2", "LOUT2"}, ++ {"Out 4", NULL, "Out4 Mux"}, ++ {"OUT4", NULL, "Out 4"}, ++ ++ /* record mixer */ ++ {"Playback Mixer", "Left Capture Switch", "Left Mixer"}, ++ {"Playback Mixer", "Voice Capture Switch", "Mono Mixer"}, ++ {"Playback Mixer", "Right Capture Switch", "Right Mixer"}, ++ ++ /* Mic/SideTone Mux */ ++ {"Mic Sidetone Mux", "Left PGA", "Left Capture Volume"}, ++ {"Mic Sidetone Mux", "Right PGA", "Right Capture Volume"}, ++ {"Mic Sidetone Mux", "Mic 1", "Mic 1 Volume"}, ++ {"Mic Sidetone Mux", "Mic 2", "Mic 2 Volume"}, ++ ++ /* Capture Left Mux */ ++ {"Capture Left Mux", "PGA", "Left Capture Volume"}, ++ {"Capture Left Mux", "Line or RXP-RXN", "Line Left Mux"}, ++ {"Capture Left Mux", "Line", "LINE1"}, ++ ++ /* Capture Right Mux */ ++ {"Capture Right Mux", "PGA", "Right Capture Volume"}, ++ {"Capture Right Mux", "Line or RXP-RXN", "Line Right Mux"}, ++ {"Capture Right Mux", "Sidetone", "Capture ST Mixer"}, ++ ++ /* Mono Capture mixer-mux */ ++ {"Capture Right Mixer", "Stereo", "Capture Right Mux"}, ++ {"Capture Left Mixer", "Analogue Mix Left", "Capture Left Mux"}, ++ {"Capture Left Mixer", "Analogue Mix Left", "Capture Right Mux"}, ++ {"Capture Right Mixer", "Analogue Mix Right", "Capture Left Mux"}, ++ {"Capture Right Mixer", "Analogue Mix Right", "Capture Right Mux"}, ++ {"Capture Left Mixer", "Digital Mono Mix", "Capture Left Mux"}, ++ {"Capture Left Mixer", "Digital Mono Mix", "Capture Right Mux"}, ++ {"Capture Right Mixer", "Digital Mono Mix", "Capture Left Mux"}, ++ {"Capture Right Mixer", "Digital Mono Mix", "Capture Right Mux"}, ++ ++ /* ADC */ ++ {"Left ADC", NULL, "Capture Left Mixer"}, ++ {"Right ADC", NULL, "Capture Right Mixer"}, ++ ++ /* Left Capture Volume */ ++ {"Left Capture Volume", NULL, "ACIN"}, ++ ++ /* Right Capture Volume */ ++ {"Right Capture Volume", NULL, "Mic 2 Volume"}, ++ ++ /* ALC Mixer */ ++ {"ALC Mixer", "Line Capture Switch", "Line Mixer"}, ++ {"ALC Mixer", "Mic2 Capture Switch", "Mic 2 Volume"}, ++ {"ALC Mixer", "Mic1 Capture Switch", "Mic 1 Volume"}, ++ {"ALC Mixer", "Rx Capture Switch", "Rx Mixer"}, ++ ++ /* Line Left Mux */ ++ {"Line Left Mux", "Line 1", "LINE1"}, ++ {"Line Left Mux", "Rx Mix", "Rx Mixer"}, ++ ++ /* Line Right Mux */ ++ {"Line Right Mux", "Line 2", "LINE2"}, ++ {"Line Right Mux", "Rx Mix", "Rx Mixer"}, ++ ++ /* Line Mono Mux */ ++ {"Line Mono Mux", "Line Mix", "Line Mixer"}, ++ {"Line Mono Mux", "Rx Mix", "Rx Mixer"}, ++ ++ /* Line Mixer/Mux */ ++ {"Line Mixer", "Line 1 + 2", "LINE1"}, ++ {"Line Mixer", "Line 1 - 2", "LINE1"}, ++ {"Line Mixer", "Line 1 + 2", "LINE2"}, ++ {"Line Mixer", "Line 1 - 2", "LINE2"}, ++ {"Line Mixer", "Line 1", "LINE1"}, ++ {"Line Mixer", "Line 2", "LINE2"}, ++ ++ /* Rx Mixer/Mux */ ++ {"Rx Mixer", "RXP - RXN", "RXP"}, ++ {"Rx Mixer", "RXP + RXN", "RXP"}, ++ {"Rx Mixer", "RXP - RXN", "RXN"}, ++ {"Rx Mixer", "RXP + RXN", "RXN"}, ++ {"Rx Mixer", "RXP", "RXP"}, ++ {"Rx Mixer", "RXN", "RXN"}, ++ ++ /* Mic 1 Volume */ ++ {"Mic 1 Volume", NULL, "MIC1N"}, ++ {"Mic 1 Volume", NULL, "Mic Selection Mux"}, ++ ++ /* Mic 2 Volume */ ++ {"Mic 2 Volume", NULL, "MIC2N"}, ++ {"Mic 2 Volume", NULL, "MIC2"}, ++ ++ /* Mic Selector Mux */ ++ {"Mic Selection Mux", "Mic 1", "MIC1"}, ++ {"Mic Selection Mux", "Mic 2", "MIC2N"}, ++ {"Mic Selection Mux", "Mic 3", "MIC2"}, ++ ++ /* ACOP */ ++ {"ACOP", NULL, "ALC Mixer"}, ++ ++ /* terminator */ ++ {NULL, NULL, NULL}, ++}; ++ ++static int wm8753_add_widgets(struct snd_soc_codec *codec) ++{ ++ int i; ++ ++ for(i = 0; i < ARRAY_SIZE(wm8753_dapm_widgets); i++) { ++ snd_soc_dapm_new_control(codec, &wm8753_dapm_widgets[i]); ++ } ++ ++ /* set up the WM8753 audio map */ ++ for(i = 0; audio_map[i][0] != NULL; i++) { ++ snd_soc_dapm_connect_input(codec, audio_map[i][0], ++ audio_map[i][1], audio_map[i][2]); ++ } ++ ++ snd_soc_dapm_new_widgets(codec); ++ return 0; ++} ++ ++/* PLL divisors */ ++struct _pll_div { ++ u32 pll_in; /* ext clock input */ ++ u32 pll_out; /* pll out freq */ ++ u32 div2:1; ++ u32 n:4; ++ u32 k:24; ++}; ++ ++/* ++ * PLL divisors - ++ */ ++static const struct _pll_div pll_div[] = { ++ {13000000, 12288000, 0, 0x7, 0x23F54A}, ++ {13000000, 11289600, 0, 0x6, 0x3CA2F5}, ++ {12000000, 12288000, 0, 0x8, 0x0C49BA}, ++ {12000000, 11289600, 0, 0x7, 0x21B08A}, ++ {24000000, 12288000, 1, 0x8, 0x0C49BA}, ++ {24000000, 11289600, 1, 0x7, 0x21B08A}, ++ {12288000, 11289600, 0, 0x7, 0x166667}, ++ {26000000, 11289600, 1, 0x6, 0x3CA2F5}, ++ {26000000, 12288000, 1, 0x7, 0x23F54A}, ++}; ++ ++static u32 wm8753_config_pll(struct snd_soc_codec *codec, ++ struct snd_soc_codec_dai *dai, int pll) ++{ ++ u16 reg; ++ int found = 0; ++ ++ if (pll == 1) { ++ reg = wm8753_read_reg_cache(codec, WM8753_CLOCK) & 0xffef; ++ if (!dai->pll_in || !dai->mclk) { ++ /* disable PLL1 */ ++ wm8753_write(codec, WM8753_PLL1CTL1, 0x0026); ++ wm8753_write(codec, WM8753_CLOCK, reg); ++ return 0; ++ } else { ++ u16 value = 0; ++ int i = 0; ++ ++ /* if we cant match, then use good values for N and K */ ++ for (;i < ARRAY_SIZE(pll_div); i++) { ++ if (pll_div[i].pll_out == dai->pll_out && ++ pll_div[i].pll_in == dai->pll_in) { ++ found = 1; ++ break; ++ } ++ } ++ ++ if (!found) ++ goto err; ++ ++ /* set up N and K PLL divisor ratios */ ++ /* bits 8:5 = PLL_N, bits 3:0 = PLL_K[21:18] */ ++ value = (pll_div[i].n << 5) + ((pll_div[i].k & 0x3c0000) >> 18); ++ wm8753_write(codec, WM8753_PLL1CTL2, value); ++ ++ /* bits 8:0 = PLL_K[17:9] */ ++ value = (pll_div[i].k & 0x03fe00) >> 9; ++ wm8753_write(codec, WM8753_PLL1CTL3, value); ++ ++ /* bits 8:0 = PLL_K[8:0] */ ++ value = pll_div[i].k & 0x0001ff; ++ wm8753_write(codec, WM8753_PLL1CTL4, value); ++ ++ /* set PLL1 as input and enable */ ++ wm8753_write(codec, WM8753_PLL1CTL1, 0x0027 | ++ (pll_div[i].div2 << 3)); ++ wm8753_write(codec, WM8753_CLOCK, reg | 0x0010); ++ } ++ } else { ++ reg = wm8753_read_reg_cache(codec, WM8753_CLOCK) & 0xfff7; ++ if (!dai->pll_in || !dai->mclk) { ++ /* disable PLL2 */ ++ wm8753_write(codec, WM8753_PLL2CTL1, 0x0026); ++ wm8753_write(codec, WM8753_CLOCK, reg); ++ return 0; ++ } else { ++ u16 value = 0; ++ int i = 0; ++ ++ /* if we cant match, then use good values for N and K */ ++ for (;i < ARRAY_SIZE(pll_div); i++) { ++ if (pll_div[i].pll_out == dai->pll_out && ++ pll_div[i].pll_in == dai->pll_in) { ++ found = 1; ++ break; ++ } ++ } ++ ++ if (!found) ++ goto err; ++ ++ /* set up N and K PLL divisor ratios */ ++ /* bits 8:5 = PLL_N, bits 3:0 = PLL_K[21:18] */ ++ value = (pll_div[i].n << 5) + ((pll_div[i].k & 0x3c0000) >> 18); ++ wm8753_write(codec, WM8753_PLL2CTL2, value); ++ ++ /* bits 8:0 = PLL_K[17:9] */ ++ value = (pll_div[i].k & 0x03fe00) >> 9; ++ wm8753_write(codec, WM8753_PLL2CTL3, value); ++ ++ /* bits 8:0 = PLL_K[8:0] */ ++ value = pll_div[i].k & 0x0001ff; ++ wm8753_write(codec, WM8753_PLL2CTL4, value); ++ ++ /* set PLL1 as input and enable */ ++ wm8753_write(codec, WM8753_PLL2CTL1, 0x0027 | ++ (pll_div[i].div2 << 3)); ++ wm8753_write(codec, WM8753_CLOCK, reg | 0x0008); ++ } ++ } ++ ++ return dai->pll_in; ++err: ++ return 0; ++} ++ ++struct _coeff_div { ++ u32 mclk; ++ u32 rate; ++ u16 fs; ++ u8 sr:5; ++ u8 usb:1; ++}; ++ ++/* codec hifi mclk (after PLL) clock divider coefficients */ ++static const struct _coeff_div coeff_div[] = { ++ /* 8k */ ++ {12288000, 8000, 1536, 0x6, 0x0}, ++ {11289600, 8000, 1408, 0x16, 0x0}, ++ {18432000, 8000, 2304, 0x7, 0x0}, ++ {16934400, 8000, 2112, 0x17, 0x0}, ++ {12000000, 8000, 1500, 0x6, 0x1}, ++ ++ /* 11.025k */ ++ {11289600, 11025, 1024, 0x18, 0x0}, ++ {16934400, 11025, 1536, 0x19, 0x0}, ++ {12000000, 11025, 1088, 0x19, 0x1}, ++ ++ /* 16k */ ++ {12288000, 16000, 768, 0xa, 0x0}, ++ {18432000, 16000, 1152, 0xb, 0x0}, ++ {12000000, 16000, 750, 0xa, 0x1}, ++ ++ /* 22.05k */ ++ {11289600, 22050, 512, 0x1a, 0x0}, ++ {16934400, 22050, 768, 0x1b, 0x0}, ++ {12000000, 22050, 544, 0x1b, 0x1}, ++ ++ /* 32k */ ++ {12288000, 32000, 384, 0xc, 0x0}, ++ {18432000, 32000, 576, 0xd, 0x0}, ++ {12000000, 32000, 375, 0xa, 0x1}, ++ ++ /* 44.1k */ ++ {11289600, 44100, 256, 0x10, 0x0}, ++ {16934400, 44100, 384, 0x11, 0x0}, ++ {12000000, 44100, 272, 0x11, 0x1}, ++ ++ /* 48k */ ++ {12288000, 48000, 256, 0x0, 0x0}, ++ {18432000, 48000, 384, 0x1, 0x0}, ++ {12000000, 48000, 250, 0x0, 0x1}, ++ ++ /* 88.2k */ ++ {11289600, 88200, 128, 0x1e, 0x0}, ++ {16934400, 88200, 192, 0x1f, 0x0}, ++ {12000000, 88200, 136, 0x1f, 0x1}, ++ ++ /* 96k */ ++ {12288000, 96000, 128, 0xe, 0x0}, ++ {18432000, 96000, 192, 0xf, 0x0}, ++ {12000000, 96000, 125, 0xe, 0x1}, ++}; ++ ++static int get_coeff(int mclk, int rate) ++{ ++ int i; ++ ++ for (i = 0; i < ARRAY_SIZE(coeff_div); i++) { ++ if (coeff_div[i].rate == rate && coeff_div[i].mclk == mclk) ++ return i; ++ } ++ return -EINVAL; ++} ++ ++/* supported HiFi input clocks (that don't use PLL) */ ++const static int hifi_clks[] = {11289600, 12000000, 12288000, ++ 16934400, 18432000}; ++ ++/* The HiFi interface can be clocked in one of two ways:- ++ * o No PLL - MCLK is used directly. ++ * o PLL - PLL is used to generate audio MCLK from input clock. ++ * ++ * We use the direct method if we can as it saves power. ++ */ ++static unsigned int wm8753_config_i2s_sysclk(struct snd_soc_codec_dai *dai, ++ struct snd_soc_clock_info *info, unsigned int clk) ++{ ++ int i, pll_out; ++ ++ /* is clk supported without the PLL */ ++ for(i = 0; i < ARRAY_SIZE(hifi_clks); i++) { ++ if (clk == hifi_clks[i]) { ++ dai->mclk = clk; ++ dai->pll_in = dai->pll_out = 0; ++ dai->clk_div = 1; ++ return clk; ++ } ++ } ++ ++ /* determine best PLL output speed */ ++ if (info->bclk_master & ++ (SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS)) { ++ pll_out = info->fs * info->rate; ++ } else { ++ /* calc slave clock */ ++ switch (info->rate){ ++ case 11025: ++ case 22050: ++ case 44100: ++ case 88200: ++ pll_out = 11289600; ++ break; ++ default: ++ pll_out = 12288000; ++ break; ++ } ++ } ++ ++ /* are input & output clocks supported by PLL */ ++ for (i = 0;i < ARRAY_SIZE(pll_div); i++) { ++ if (pll_div[i].pll_in == clk && pll_div[i].pll_out == pll_out) { ++ dai->pll_in = clk; ++ dai->pll_out = dai->mclk = pll_out; ++ return pll_out; ++ } ++ } ++ ++ /* this clk is not supported */ ++ return 0; ++} ++ ++/* valid PCM clock dividers * 2 */ ++static int pcm_divs[] = {2, 6, 11, 4, 8, 12, 16}; ++ ++/* The Voice interface can be clocked in one of four ways:- ++ * o No PLL - MCLK is used directly. ++ * o Div - MCLK is directly divided. ++ * o PLL - PLL is used to generate audio MCLK from input clock. ++ * o PLL & Div - PLL and post divider are used. ++ * ++ * We use the non PLL methods if we can, as it saves power. ++ */ ++ ++static unsigned int wm8753_config_pcm_sysclk(struct snd_soc_codec_dai *dai, ++ struct snd_soc_clock_info *info, unsigned int clk) ++{ ++ int i, j, best_clk = info->fs * info->rate; ++ ++ /* can we run at this clk without the PLL ? */ ++ for (i = 0; i < ARRAY_SIZE(pcm_divs); i++) { ++ if ((best_clk >> 1) * pcm_divs[i] == clk) { ++ dai->pll_in = 0; ++ dai->clk_div = pcm_divs[i]; ++ dai->mclk = best_clk; ++ return dai->mclk; ++ } ++ } ++ ++ /* now check for PLL support */ ++ for (i = 0; i < ARRAY_SIZE(pll_div); i++) { ++ if (pll_div[i].pll_in == clk) { ++ for (j = 0; j < ARRAY_SIZE(pcm_divs); j++) { ++ if (pll_div[i].pll_out == pcm_divs[j] * (best_clk >> 1)) { ++ dai->pll_in = clk; ++ dai->pll_out = pll_div[i].pll_out; ++ dai->clk_div = pcm_divs[j]; ++ dai->mclk = best_clk; ++ return dai->mclk; ++ } ++ } ++ } ++ } ++ ++ /* this clk is not supported */ ++ return 0; ++} ++ ++/* set the format and bit size for ADC and Voice DAC */ ++static void wm8753_adc_vdac_prepare(struct snd_pcm_substream *substream) ++{ ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ struct snd_soc_device *socdev = rtd->socdev; ++ struct snd_soc_codec *codec = socdev->codec; ++ u16 voice = wm8753_read_reg_cache(codec, WM8753_PCM) & 0x01e0; ++ ++ /* interface format */ ++ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_FORMAT_MASK) { ++ case SND_SOC_DAIFMT_I2S: ++ voice |= 0x0002; ++ break; ++ case SND_SOC_DAIFMT_RIGHT_J: ++ break; ++ case SND_SOC_DAIFMT_LEFT_J: ++ voice |= 0x0001; ++ break; ++ case SND_SOC_DAIFMT_DSP_A: ++ voice |= 0x0003; ++ break; ++ case SND_SOC_DAIFMT_DSP_B: ++ voice |= 0x0013; ++ break; ++ } ++ ++ /* bit size */ ++ switch (rtd->codec_dai->dai_runtime.pcmfmt) { ++ case SNDRV_PCM_FMTBIT_S16_LE: ++ break; ++ case SNDRV_PCM_FMTBIT_S20_3LE: ++ voice |= 0x0004; ++ break; ++ case SNDRV_PCM_FMTBIT_S24_LE: ++ voice |= 0x0008; ++ break; ++ case SNDRV_PCM_FMTBIT_S32_LE: ++ voice |= 0x000c; ++ break; ++ } ++ ++ wm8753_write(codec, WM8753_PCM, voice); ++} ++ ++/* configure PCM DAI */ ++static int wm8753_pcm_dai_prepare(struct snd_pcm_substream *substream) ++{ ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ struct snd_soc_device *socdev = rtd->socdev; ++ struct snd_soc_codec *codec = socdev->codec; ++ u16 voice, ioctl, srate, srate2, fs, bfs, clock; ++ unsigned int rate; ++ ++ bfs = SND_SOC_FSBD_REAL(rtd->codec_dai->dai_runtime.bfs); ++ fs = rtd->codec_dai->dai_runtime.fs; ++ rate = snd_soc_get_rate(rtd->codec_dai->dai_runtime.pcmrate); ++ voice = wm8753_read_reg_cache(codec, WM8753_PCM) & 0x001f; ++ ++ /* set master/slave audio interface */ ++ ioctl = wm8753_read_reg_cache(codec, WM8753_IOCTL) & 0x01fd; ++ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_CLOCK_MASK) { ++ case SND_SOC_DAIFMT_CBM_CFM: ++ ioctl |= 0x0002; ++ case SND_SOC_DAIFMT_CBM_CFS: ++ voice |= 0x0040; ++ break; ++ } ++ ++ /* do we need to enable the PLL */ ++ if (rtd->codec_dai->pll_in) { ++ if (wm8753_config_pll(codec, rtd->codec_dai, 2) != ++ rtd->codec_dai->pll_in) { ++ err("could not set pll to %d --> %d", ++ rtd->codec_dai->pll_in, rtd->codec_dai->pll_out); ++ return -ENODEV; ++ } ++ } ++ ++ /* set up PCM divider */ ++ clock = wm8753_read_reg_cache(codec, WM8753_CLOCK) & 0x003f; ++ switch (rtd->codec_dai->clk_div) { ++ case 2: /* 1 */ ++ break; ++ case 6: /* 3 */ ++ clock |= (0x2 << 6); ++ break; ++ case 11: /* 5.5 */ ++ clock |= (0x3 << 6); ++ break; ++ case 4: /* 2 */ ++ clock |= (0x4 << 6); ++ break; ++ case 8: /* 4 */ ++ clock |= (0x5 << 6); ++ break; ++ case 12: /* 6 */ ++ clock |= (0x6 << 6); ++ break; ++ case 16: /* 8 */ ++ clock |= (0x7 << 6); ++ break; ++ default: ++ printk(KERN_ERR "wm8753: invalid PCM clk divider %d\n", ++ rtd->codec_dai->clk_div); ++ break; ++ } ++ wm8753_write(codec, WM8753_CLOCK, clock); ++ ++ /* set bclk divisor rate */ ++ srate2 = wm8753_read_reg_cache(codec, WM8753_SRATE2) & 0x003f; ++ switch (bfs) { ++ case 1: ++ break; ++ case 2: ++ srate2 |= (0x1 << 6); ++ break; ++ case 4: ++ srate2 |= (0x2 << 6); ++ break; ++ case 8: ++ srate2 |= (0x3 << 6); ++ break; ++ case 16: ++ srate2 |= (0x4 << 6); ++ break; ++ } ++ wm8753_write(codec, WM8753_SRATE2, srate2); ++ ++ srate = wm8753_read_reg_cache(codec, WM8753_SRATE1) & 0x017f; ++ if (rtd->codec_dai->dai_runtime.fs == 384) ++ srate |= 0x80; ++ wm8753_write(codec, WM8753_SRATE1, srate); ++ ++ /* clock inversion */ ++ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_INV_MASK) { ++ case SND_SOC_DAIFMT_IB_IF: ++ voice |= 0x0090; ++ break; ++ case SND_SOC_DAIFMT_IB_NF: ++ voice |= 0x0080; ++ break; ++ case SND_SOC_DAIFMT_NB_IF: ++ voice |= 0x0010; ++ break; ++ } ++ //printk("voice %x %x ioctl %x %x srate2 %x %x srate1 %x %x\n", ++ //WM8753_PCM, voice, WM8753_IOCTL, ioctl, WM8753_SRATE2, ++ //srate2, WM8753_SRATE1, srate); ++ ++ wm8753_write(codec, WM8753_IOCTL, ioctl); ++ wm8753_write(codec, WM8753_PCM, voice); ++ return 0; ++} ++ ++/* configure hifi DAC wordlength and format */ ++static void wm8753_hdac_prepare(struct snd_pcm_substream *substream) ++{ ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ struct snd_soc_device *socdev = rtd->socdev; ++ struct snd_soc_codec *codec = socdev->codec; ++ u16 hifi = wm8753_read_reg_cache(codec, WM8753_HIFI) & 0x01e0; ++ ++ /* interface format */ ++ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_FORMAT_MASK) { ++ case SND_SOC_DAIFMT_I2S: ++ hifi |= 0x0002; ++ break; ++ case SND_SOC_DAIFMT_RIGHT_J: ++ break; ++ case SND_SOC_DAIFMT_LEFT_J: ++ hifi |= 0x0001; ++ break; ++ case SND_SOC_DAIFMT_DSP_A: ++ hifi |= 0x0003; ++ break; ++ case SND_SOC_DAIFMT_DSP_B: ++ hifi |= 0x0013; ++ break; ++ } ++ ++ /* bit size */ ++ switch (rtd->codec_dai->dai_runtime.pcmfmt) { ++ case SNDRV_PCM_FMTBIT_S16_LE: ++ break; ++ case SNDRV_PCM_FMTBIT_S20_3LE: ++ hifi |= 0x0004; ++ break; ++ case SNDRV_PCM_FMTBIT_S24_LE: ++ hifi |= 0x0008; ++ break; ++ case SNDRV_PCM_FMTBIT_S32_LE: ++ hifi |= 0x000c; ++ break; ++ } ++ ++ wm8753_write(codec, WM8753_HIFI, hifi); ++} ++ ++/* configure i2s (hifi) DAI clocking */ ++static int wm8753_i2s_dai_prepare(struct snd_pcm_substream *substream) ++{ ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ struct snd_soc_device *socdev = rtd->socdev; ++ struct snd_soc_codec *codec = socdev->codec; ++ u16 srate, bfs, hifi, ioctl; ++ unsigned int rate; ++ int i = 0; ++ ++ bfs = SND_SOC_FSBD_REAL(rtd->codec_dai->dai_runtime.bfs); ++ rate = snd_soc_get_rate(rtd->codec_dai->dai_runtime.pcmrate); ++ hifi = wm8753_read_reg_cache(codec, WM8753_HIFI) & 0x001f; ++ ++ /* is coefficient valid ? */ ++ if ((i = get_coeff(rtd->codec_dai->mclk, rate)) < 0) ++ return i; ++ ++ srate = wm8753_read_reg_cache(codec, WM8753_SRATE1) & 0x01c0; ++ wm8753_write(codec, WM8753_SRATE1, srate | (coeff_div[i].sr << 1) | ++ coeff_div[i].usb); ++ ++ /* do we need to enable the PLL */ ++ if (rtd->codec_dai->pll_in) { ++ if (wm8753_config_pll(codec, rtd->codec_dai, 1) != ++ rtd->codec_dai->pll_in) { ++ err("could not set pll to %d --> %d", ++ rtd->codec_dai->pll_in, rtd->codec_dai->pll_out); ++ return -ENODEV; ++ } ++ } ++ ++ /* set bclk divisor rate */ ++ srate = wm8753_read_reg_cache(codec, WM8753_SRATE2) & 0x01c7; ++ switch (bfs) { ++ case 1: ++ break; ++ case 2: ++ srate |= (0x1 << 3); ++ break; ++ case 4: ++ srate |= (0x2 << 3); ++ break; ++ case 8: ++ srate |= (0x3 << 3); ++ break; ++ case 16: ++ srate |= (0x4 << 3); ++ break; ++ } ++ wm8753_write(codec, WM8753_SRATE2, srate); ++ ++ /* set master/slave audio interface */ ++ ioctl = wm8753_read_reg_cache(codec, WM8753_IOCTL) & 0x00fe; ++ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_CLOCK_MASK) { ++ case SND_SOC_DAIFMT_CBM_CFM: ++ ioctl |= 0x0001; ++ case SND_SOC_DAIFMT_CBM_CFS: ++ hifi |= 0x0040; ++ break; ++ } ++ ++ /* clock inversion */ ++ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_INV_MASK) { ++ case SND_SOC_DAIFMT_IB_IF: ++ hifi |= 0x0090; ++ break; ++ case SND_SOC_DAIFMT_IB_NF: ++ hifi |= 0x0080; ++ break; ++ case SND_SOC_DAIFMT_NB_IF: ++ hifi |= 0x0010; ++ break; ++ } ++ wm8753_write(codec, WM8753_IOCTL, ioctl); ++ wm8753_write(codec, WM8753_HIFI, hifi); ++ return 0; ++} ++ ++static int wm8753_mode1v_prepare (struct snd_pcm_substream *substream) ++{ ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ struct snd_soc_device *socdev = rtd->socdev; ++ struct snd_soc_codec *codec = socdev->codec; ++ u16 clock; ++ ++ /* set clk source as pcmclk */ ++ clock = wm8753_read_reg_cache(codec, WM8753_CLOCK) & 0xfffb; ++ wm8753_write(codec, WM8753_CLOCK, clock); ++ ++ wm8753_adc_vdac_prepare(substream); ++ return wm8753_pcm_dai_prepare(substream); ++} ++ ++static int wm8753_mode1h_prepare (struct snd_pcm_substream *substream) ++{ ++ wm8753_hdac_prepare(substream); ++ return wm8753_i2s_dai_prepare(substream); ++} ++ ++static int wm8753_mode2_prepare (struct snd_pcm_substream *substream) ++{ ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ struct snd_soc_device *socdev = rtd->socdev; ++ struct snd_soc_codec *codec = socdev->codec; ++ u16 clock; ++ ++ /* set clk source as pcmclk */ ++ clock = wm8753_read_reg_cache(codec, WM8753_CLOCK) & 0xfffb; ++ wm8753_write(codec, WM8753_CLOCK, clock); ++ ++ wm8753_adc_vdac_prepare(substream); ++ return wm8753_i2s_dai_prepare(substream); ++} ++ ++static int wm8753_mode3_prepare (struct snd_pcm_substream *substream) ++{ ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ struct snd_soc_device *socdev = rtd->socdev; ++ struct snd_soc_codec *codec = socdev->codec; ++ u16 clock; ++ ++ /* set clk source as mclk */ ++ clock = wm8753_read_reg_cache(codec, WM8753_CLOCK) & 0xfffb; ++ wm8753_write(codec, WM8753_CLOCK, clock | 0x4); ++ ++ wm8753_hdac_prepare(substream); ++ wm8753_adc_vdac_prepare(substream); ++ return wm8753_i2s_dai_prepare(substream); ++} ++ ++static int wm8753_mode4_prepare (struct snd_pcm_substream *substream) ++{ ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ struct snd_soc_device *socdev = rtd->socdev; ++ struct snd_soc_codec *codec = socdev->codec; ++ u16 clock; ++ ++ /* set clk source as mclk */ ++ clock = wm8753_read_reg_cache(codec, WM8753_CLOCK) & 0xfffb; ++ wm8753_write(codec, WM8753_CLOCK, clock | 0x4); ++ ++ wm8753_hdac_prepare(substream); ++ wm8753_adc_vdac_prepare(substream); ++ return wm8753_i2s_dai_prepare(substream); ++} ++ ++static int wm8753_mute(struct snd_soc_codec *codec, ++ struct snd_soc_codec_dai *dai, int mute) ++{ ++ u16 mute_reg = wm8753_read_reg_cache(codec, WM8753_DAC) & 0xfff7; ++ ++ /* the digital mute covers the HiFi and Voice DAC's on the WM8753. ++ * make sure we check if they are not both active when we mute */ ++ if (mute && dai->id == 1) { ++ if (!wm8753_dai[WM8753_DAI_VOICE].playback.active || ++ !wm8753_dai[WM8753_DAI_HIFI].playback.active) ++ wm8753_write(codec, WM8753_DAC, mute_reg | 0x8); ++ } else { ++ if (mute) ++ wm8753_write(codec, WM8753_DAC, mute_reg | 0x8); ++ else ++ wm8753_write(codec, WM8753_DAC, mute_reg); ++ } ++ ++ return 0; ++} ++ ++static int wm8753_dapm_event(struct snd_soc_codec *codec, int event) ++{ ++ u16 pwr_reg = wm8753_read_reg_cache(codec, WM8753_PWR1) & 0xfe3e; ++ ++ switch (event) { ++ case SNDRV_CTL_POWER_D0: /* full On */ ++ /* set vmid to 50k and unmute dac */ ++ wm8753_write(codec, WM8753_PWR1, pwr_reg | 0x00c0); ++ break; ++ case SNDRV_CTL_POWER_D1: /* partial On */ ++ case SNDRV_CTL_POWER_D2: /* partial On */ ++ /* set vmid to 5k for quick power up */ ++ wm8753_write(codec, WM8753_PWR1, pwr_reg | 0x01c1); ++ break; ++ case SNDRV_CTL_POWER_D3hot: /* Off, with power */ ++ /* mute dac and set vmid to 500k, enable VREF */ ++ wm8753_write(codec, WM8753_PWR1, pwr_reg | 0x0141); ++ break; ++ case SNDRV_CTL_POWER_D3cold: /* Off, without power */ ++ wm8753_write(codec, WM8753_PWR1, 0x0001); ++ break; ++ } ++ codec->dapm_state = event; ++ return 0; ++} ++ ++/* ++ * The WM8753 supports upto 4 different and mutually exclusive DAI ++ * configurations. This gives 2 PCM's available for use, hifi and voice. ++ * NOTE: The Voice PCM cannot play or caputure audio to the CPU as it's DAI ++ * is connected between the wm8753 and a BT codec or GSM modem. ++ * ++ * 1. Voice over PCM DAI - HIFI DAC over HIFI DAI ++ * 2. Voice over HIFI DAI - HIFI disabled ++ * 3. Voice disabled - HIFI over HIFI ++ * 4. Voice disabled - HIFI over HIFI, uses voice DAI LRC for capture ++ */ ++static const struct snd_soc_codec_dai wm8753_all_dai[] = { ++/* DAI HiFi mode 1 */ ++{ .name = "WM8753 HiFi", ++ .id = 1, ++ .playback = { ++ .stream_name = "HiFi Playback", ++ .channels_min = 1, ++ .channels_max = 2,}, ++ .capture = { /* dummy for fast DAI switching */ ++ .stream_name = "HiFi Capture", ++ .channels_min = 1, ++ .channels_max = 2,}, ++ .config_sysclk = wm8753_config_i2s_sysclk, ++ .digital_mute = wm8753_mute, ++ .ops = { ++ .prepare = wm8753_mode1h_prepare,}, ++ .caps = { ++ .num_modes = ARRAY_SIZE(wm8753_hifi_modes), ++ .mode = wm8753_hifi_modes,}, ++}, ++/* DAI Voice mode 1 */ ++{ .name = "WM8753 Voice", ++ .id = 1, ++ .playback = { ++ .stream_name = "Voice Playback", ++ .channels_min = 1, ++ .channels_max = 1,}, ++ .capture = { ++ .stream_name = "Voice Capture", ++ .channels_min = 1, ++ .channels_max = 2,}, ++ .config_sysclk = wm8753_config_pcm_sysclk, ++ .digital_mute = wm8753_mute, ++ .ops = { ++ .prepare = wm8753_mode1v_prepare,}, ++ .caps = { ++ .num_modes = ARRAY_SIZE(wm8753_voice_modes), ++ .mode = wm8753_voice_modes,}, ++}, ++/* DAI HiFi mode 2 - dummy */ ++{ .name = "WM8753 HiFi", ++ .id = 2, ++}, ++/* DAI Voice mode 2 */ ++{ .name = "WM8753 Voice", ++ .id = 2, ++ .playback = { ++ .stream_name = "Voice Playback", ++ .channels_min = 1, ++ .channels_max = 1,}, ++ .capture = { ++ .stream_name = "Voice Capture", ++ .channels_min = 1, ++ .channels_max = 2,}, ++ .config_sysclk = wm8753_config_i2s_sysclk, ++ .digital_mute = wm8753_mute, ++ .ops = { ++ .prepare = wm8753_mode2_prepare,}, ++ .caps = { ++ .num_modes = ARRAY_SIZE(wm8753_voice_modes), ++ .mode = wm8753_voice_modes,}, ++}, ++/* DAI HiFi mode 3 */ ++{ .name = "WM8753 HiFi", ++ .id = 3, ++ .playback = { ++ .stream_name = "HiFi Playback", ++ .channels_min = 1, ++ .channels_max = 2,}, ++ .capture = { ++ .stream_name = "HiFi Capture", ++ .channels_min = 1, ++ .channels_max = 2,}, ++ .config_sysclk = wm8753_config_i2s_sysclk, ++ .digital_mute = wm8753_mute, ++ .ops = { ++ .prepare = wm8753_mode3_prepare,}, ++ .caps = { ++ .num_modes = ARRAY_SIZE(wm8753_hifi_modes), ++ .mode = wm8753_hifi_modes,}, ++}, ++/* DAI Voice mode 3 - dummy */ ++{ .name = "WM8753 Voice", ++ .id = 3, ++}, ++/* DAI HiFi mode 4 */ ++{ .name = "WM8753 HiFi", ++ .id = 4, ++ .playback = { ++ .stream_name = "HiFi Playback", ++ .channels_min = 1, ++ .channels_max = 2,}, ++ .capture = { ++ .stream_name = "HiFi Capture", ++ .channels_min = 1, ++ .channels_max = 2,}, ++ .config_sysclk = wm8753_config_i2s_sysclk, ++ .digital_mute = wm8753_mute, ++ .ops = { ++ .prepare = wm8753_mode4_prepare,}, ++ .caps = { ++ .num_modes = ARRAY_SIZE(wm8753_mixed_modes), ++ .mode = wm8753_mixed_modes,}, ++}, ++/* DAI Voice mode 4 - dummy */ ++{ .name = "WM8753 Voice", ++ .id = 4, ++}, ++}; ++ ++struct snd_soc_codec_dai wm8753_dai[2]; ++EXPORT_SYMBOL_GPL(wm8753_dai); ++ ++static void wm8753_set_dai_mode(struct snd_soc_codec *codec, unsigned int mode) ++{ ++ if (mode < 4) { ++ wm8753_dai[0] = wm8753_all_dai[mode << 1]; ++ wm8753_dai[1] = wm8753_all_dai[(mode << 1) + 1]; ++ } ++} ++ ++static void wm8753_work(void *data) ++{ ++ struct snd_soc_codec *codec = (struct snd_soc_codec *)data; ++ wm8753_dapm_event(codec, codec->dapm_state); ++} ++ ++static int wm8753_suspend(struct platform_device *pdev, pm_message_t state) ++{ ++ struct snd_soc_device *socdev = platform_get_drvdata(pdev); ++ struct snd_soc_codec *codec = socdev->codec; ++ ++ wm8753_dapm_event(codec, SNDRV_CTL_POWER_D3cold); ++ return 0; ++} ++ ++static int wm8753_resume(struct platform_device *pdev) ++{ ++ struct snd_soc_device *socdev = platform_get_drvdata(pdev); ++ struct snd_soc_codec *codec = socdev->codec; ++ int i; ++ u8 data[2]; ++ u16 *cache = codec->reg_cache; ++ ++ /* Sync reg_cache with the hardware */ ++ for (i = 0; i < ARRAY_SIZE(wm8753_reg); i++) { ++ if (i + 1 == WM8753_RESET) ++ continue; ++ data[0] = ((i + 1) << 1) | ((cache[i] >> 8) & 0x0001); ++ data[1] = cache[i] & 0x00ff; ++ codec->hw_write(codec->control_data, data, 2); ++ } ++ ++ wm8753_dapm_event(codec, SNDRV_CTL_POWER_D3hot); ++ ++ /* charge wm8753 caps */ ++ if (codec->suspend_dapm_state == SNDRV_CTL_POWER_D0) { ++ wm8753_dapm_event(codec, SNDRV_CTL_POWER_D2); ++ codec->dapm_state = SNDRV_CTL_POWER_D0; ++ queue_delayed_work(wm8753_workq, &wm8753_dapm_work, ++ msecs_to_jiffies(caps_charge)); ++ } ++ ++ return 0; ++} ++ ++/* ++ * initialise the WM8753 driver ++ * register the mixer and dsp interfaces with the kernel ++ */ ++static int wm8753_init(struct snd_soc_device *socdev) ++{ ++ struct snd_soc_codec *codec = socdev->codec; ++ int reg, ret = 0; ++ ++ codec->name = "WM8753"; ++ codec->owner = THIS_MODULE; ++ codec->read = wm8753_read_reg_cache; ++ codec->write = wm8753_write; ++ codec->dapm_event = wm8753_dapm_event; ++ codec->dai = wm8753_dai; ++ codec->num_dai = 2; ++ codec->reg_cache_size = ARRAY_SIZE(wm8753_reg); ++ ++ codec->reg_cache = ++ kzalloc(sizeof(u16) * ARRAY_SIZE(wm8753_reg), GFP_KERNEL); ++ if (codec->reg_cache == NULL) ++ return -ENOMEM; ++ memcpy(codec->reg_cache, wm8753_reg, ++ sizeof(u16) * ARRAY_SIZE(wm8753_reg)); ++ codec->reg_cache_size = sizeof(u16) * ARRAY_SIZE(wm8753_reg); ++ wm8753_set_dai_mode(codec, 0); ++ ++ wm8753_reset(codec); ++ ++ /* register pcms */ ++ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); ++ if (ret < 0) { ++ kfree(codec->reg_cache); ++ return ret; ++ } ++ ++ /* charge output caps */ ++ wm8753_dapm_event(codec, SNDRV_CTL_POWER_D2); ++ codec->dapm_state = SNDRV_CTL_POWER_D3hot; ++ queue_delayed_work(wm8753_workq, ++ &wm8753_dapm_work, msecs_to_jiffies(caps_charge)); ++ ++ /* set the update bits */ ++ reg = wm8753_read_reg_cache(codec, WM8753_LDAC); ++ wm8753_write(codec, WM8753_LDAC, reg | 0x0100); ++ reg = wm8753_read_reg_cache(codec, WM8753_RDAC); ++ wm8753_write(codec, WM8753_RDAC, reg | 0x0100); ++ reg = wm8753_read_reg_cache(codec, WM8753_LOUT1V); ++ wm8753_write(codec, WM8753_LOUT1V, reg | 0x0100); ++ reg = wm8753_read_reg_cache(codec, WM8753_ROUT1V); ++ wm8753_write(codec, WM8753_ROUT1V, reg | 0x0100); ++ reg = wm8753_read_reg_cache(codec, WM8753_LOUT2V); ++ wm8753_write(codec, WM8753_LOUT2V, reg | 0x0100); ++ reg = wm8753_read_reg_cache(codec, WM8753_ROUT2V); ++ wm8753_write(codec, WM8753_ROUT2V, reg | 0x0100); ++ reg = wm8753_read_reg_cache(codec, WM8753_LINVOL); ++ wm8753_write(codec, WM8753_LINVOL, reg | 0x0100); ++ reg = wm8753_read_reg_cache(codec, WM8753_RINVOL); ++ wm8753_write(codec, WM8753_RINVOL, reg | 0x0100); ++ ++ wm8753_add_controls(codec); ++ wm8753_add_widgets(codec); ++ ret = snd_soc_register_card(socdev); ++ if (ret < 0) { ++ snd_soc_free_pcms(socdev); ++ snd_soc_dapm_free(socdev); ++ } ++ ++ return ret; ++} ++ ++/* If the i2c layer weren't so broken, we could pass this kind of data ++ around */ ++static struct snd_soc_device *wm8753_socdev; ++ ++#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) ++ ++/* ++ * WM8753 2 wire address is determined by GPIO5 ++ * state during powerup. ++ * low = 0x1a ++ * high = 0x1b ++ */ ++#define I2C_DRIVERID_WM8753 0xfefe /* liam - need a proper id */ ++ ++static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END }; ++ ++/* Magic definition of all other variables and things */ ++I2C_CLIENT_INSMOD; ++ ++static struct i2c_driver wm8753_i2c_driver; ++static struct i2c_client client_template; ++ ++static int wm8753_codec_probe(struct i2c_adapter *adap, int addr, int kind) ++{ ++ struct snd_soc_device *socdev = wm8753_socdev; ++ struct wm8753_setup_data *setup = socdev->codec_data; ++ struct snd_soc_codec *codec = socdev->codec; ++ struct i2c_client *i2c; ++ int ret; ++ ++ if (addr != setup->i2c_address) ++ return -ENODEV; ++ ++ client_template.adapter = adap; ++ client_template.addr = addr; ++ ++ i2c = kzalloc(sizeof(struct i2c_client), GFP_KERNEL); ++ if (i2c == NULL){ ++ kfree(codec); ++ return -ENOMEM; ++ } ++ memcpy(i2c, &client_template, sizeof(struct i2c_client)); ++ i2c_set_clientdata(i2c, codec); ++ codec->control_data = i2c; ++ ++ ret = i2c_attach_client(i2c); ++ if (ret < 0) { ++ err("failed to attach codec at addr %x\n", addr); ++ goto err; ++ } ++ ++ ret = wm8753_init(socdev); ++ if (ret < 0) { ++ err("failed to initialise WM8753\n"); ++ goto err; ++ } ++ ++ return ret; ++ ++err: ++ kfree(codec); ++ kfree(i2c); ++ return ret; ++} ++ ++static int wm8753_i2c_detach(struct i2c_client *client) ++{ ++ struct snd_soc_codec *codec = i2c_get_clientdata(client); ++ i2c_detach_client(client); ++ kfree(codec->reg_cache); ++ kfree(client); ++ return 0; ++} ++ ++static int wm8753_i2c_attach(struct i2c_adapter *adap) ++{ ++ return i2c_probe(adap, &addr_data, wm8753_codec_probe); ++} ++ ++/* corgi i2c codec control layer */ ++static struct i2c_driver wm8753_i2c_driver = { ++ .driver = { ++ .name = "WM8753 I2C Codec", ++ .owner = THIS_MODULE, ++ }, ++ .id = I2C_DRIVERID_WM8753, ++ .attach_adapter = wm8753_i2c_attach, ++ .detach_client = wm8753_i2c_detach, ++ .command = NULL, ++}; ++ ++static struct i2c_client client_template = { ++ .name = "WM8753", ++ .driver = &wm8753_i2c_driver, ++}; ++#endif ++ ++static int wm8753_probe(struct platform_device *pdev) ++{ ++ struct snd_soc_device *socdev = platform_get_drvdata(pdev); ++ struct wm8753_setup_data *setup; ++ struct snd_soc_codec *codec; ++ int ret = 0; ++ ++ info("WM8753 Audio Codec %s", WM8753_VERSION); ++ ++ setup = socdev->codec_data; ++ codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); ++ if (codec == NULL) ++ return -ENOMEM; ++ ++ socdev->codec = codec; ++ mutex_init(&codec->mutex); ++ INIT_LIST_HEAD(&codec->dapm_widgets); ++ INIT_LIST_HEAD(&codec->dapm_paths); ++ wm8753_socdev = socdev; ++ INIT_WORK(&wm8753_dapm_work, wm8753_work, codec); ++ wm8753_workq = create_workqueue("wm8753"); ++ if (wm8753_workq == NULL) { ++ kfree(codec); ++ return -ENOMEM; ++ } ++#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) ++ if (setup->i2c_address) { ++ normal_i2c[0] = setup->i2c_address; ++ codec->hw_write = (hw_write_t)i2c_master_send; ++ ret = i2c_add_driver(&wm8753_i2c_driver); ++ if (ret != 0) ++ printk(KERN_ERR "can't add i2c driver"); ++ } ++#else ++ /* Add other interfaces here */ ++#endif ++ return ret; ++} ++ ++/* power down chip */ ++static int wm8753_remove(struct platform_device *pdev) ++{ ++ struct snd_soc_device *socdev = platform_get_drvdata(pdev); ++ struct snd_soc_codec *codec = socdev->codec; ++ ++ if (codec->control_data) ++ wm8753_dapm_event(codec, SNDRV_CTL_POWER_D3cold); ++ if (wm8753_workq) ++ destroy_workqueue(wm8753_workq); ++ snd_soc_free_pcms(socdev); ++ snd_soc_dapm_free(socdev); ++#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) ++ i2c_del_driver(&wm8753_i2c_driver); ++#endif ++ kfree(codec); ++ ++ return 0; ++} ++ ++struct snd_soc_codec_device soc_codec_dev_wm8753 = { ++ .probe = wm8753_probe, ++ .remove = wm8753_remove, ++ .suspend = wm8753_suspend, ++ .resume = wm8753_resume, ++}; ++ ++EXPORT_SYMBOL_GPL(soc_codec_dev_wm8753); ++ ++MODULE_DESCRIPTION("ASoC WM8753 driver"); ++MODULE_AUTHOR("Liam Girdwood"); ++MODULE_LICENSE("GPL"); +Index: linux-2.6-pxa-new/sound/soc/codecs/wm8753.h +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/sound/soc/codecs/wm8753.h +@@ -0,0 +1,91 @@ ++/* ++ * wm8753.h -- audio driver for WM8753 ++ * ++ * Copyright 2003 Wolfson Microelectronics PLC. ++ * Author: Liam Girdwood ++ * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com ++ * ++ * This program is free software; you can redistribute it and/or modify it ++ * under the terms of the GNU General Public License as published by the ++ * Free Software Foundation; either version 2 of the License, or (at your ++ * option) any later version. ++ * ++ */ ++ ++#ifndef _WM8753_H ++#define _WM8753_H ++ ++/* WM8753 register space */ ++ ++#define WM8753_DAC 0x01 ++#define WM8753_ADC 0x02 ++#define WM8753_PCM 0x03 ++#define WM8753_HIFI 0x04 ++#define WM8753_IOCTL 0x05 ++#define WM8753_SRATE1 0x06 ++#define WM8753_SRATE2 0x07 ++#define WM8753_LDAC 0x08 ++#define WM8753_RDAC 0x09 ++#define WM8753_BASS 0x0a ++#define WM8753_TREBLE 0x0b ++#define WM8753_ALC1 0x0c ++#define WM8753_ALC2 0x0d ++#define WM8753_ALC3 0x0e ++#define WM8753_NGATE 0x0f ++#define WM8753_LADC 0x10 ++#define WM8753_RADC 0x11 ++#define WM8753_ADCTL1 0x12 ++#define WM8753_3D 0x13 ++#define WM8753_PWR1 0x14 ++#define WM8753_PWR2 0x15 ++#define WM8753_PWR3 0x16 ++#define WM8753_PWR4 0x17 ++#define WM8753_ID 0x18 ++#define WM8753_INTPOL 0x19 ++#define WM8753_INTEN 0x1a ++#define WM8753_GPIO1 0x1b ++#define WM8753_GPIO2 0x1c ++#define WM8753_RESET 0x1f ++#define WM8753_RECMIX1 0x20 ++#define WM8753_RECMIX2 0x21 ++#define WM8753_LOUTM1 0x22 ++#define WM8753_LOUTM2 0x23 ++#define WM8753_ROUTM1 0x24 ++#define WM8753_ROUTM2 0x25 ++#define WM8753_MOUTM1 0x26 ++#define WM8753_MOUTM2 0x27 ++#define WM8753_LOUT1V 0x28 ++#define WM8753_ROUT1V 0x29 ++#define WM8753_LOUT2V 0x2a ++#define WM8753_ROUT2V 0x2b ++#define WM8753_MOUTV 0x2c ++#define WM8753_OUTCTL 0x2d ++#define WM8753_ADCIN 0x2e ++#define WM8753_INCTL1 0x2f ++#define WM8753_INCTL2 0x30 ++#define WM8753_LINVOL 0x31 ++#define WM8753_RINVOL 0x32 ++#define WM8753_MICBIAS 0x33 ++#define WM8753_CLOCK 0x34 ++#define WM8753_PLL1CTL1 0x35 ++#define WM8753_PLL1CTL2 0x36 ++#define WM8753_PLL1CTL3 0x37 ++#define WM8753_PLL1CTL4 0x38 ++#define WM8753_PLL2CTL1 0x39 ++#define WM8753_PLL2CTL2 0x3a ++#define WM8753_PLL2CTL3 0x3b ++#define WM8753_PLL2CTL4 0x3c ++#define WM8753_BIASCTL 0x3d ++#define WM8753_ADCTL2 0x3f ++ ++struct wm8753_setup_data { ++ unsigned short i2c_address; ++}; ++ ++#define WM8753_DAI_HIFI 0 ++#define WM8753_DAI_VOICE 1 ++ ++extern struct snd_soc_codec_dai wm8753_dai[2]; ++extern struct snd_soc_codec_device soc_codec_dev_wm8753; ++ ++#endif +Index: linux-2.6-pxa-new/sound/soc/codecs/wm8772.c +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/sound/soc/codecs/wm8772.c +@@ -0,0 +1,806 @@ ++/* ++ * wm8772.c -- WM8772 ALSA Soc Audio driver ++ * ++ * Copyright 2005 Wolfson Microelectronics PLC. ++ * Author: Liam Girdwood ++ * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com ++ * ++ * This program is free software; you can redistribute it and/or modify it ++ * under the terms of the GNU General Public License as published by the ++ * Free Software Foundation; either version 2 of the License, or (at your ++ * option) any later version. ++ * ++ */ ++ ++#include <linux/module.h> ++#include <linux/moduleparam.h> ++#include <linux/version.h> ++#include <linux/kernel.h> ++#include <linux/init.h> ++#include <linux/delay.h> ++#include <linux/pm.h> ++#include <linux/i2c.h> ++ ++#include <sound/driver.h> ++#include <sound/core.h> ++#include <sound/pcm.h> ++#include <sound/pcm_params.h> ++#include <sound/soc.h> ++#include <sound/soc-dapm.h> ++#include <sound/initval.h> ++ ++#include "wm8772.h" ++ ++#define AUDIO_NAME "WM8772" ++#define WM8772_VERSION "0.3" ++ ++/* ++ * wm8772 register cache ++ * We can't read the WM8772 register space when we ++ * are using 2 wire for device control, so we cache them instead. ++ */ ++static const u16 wm8772_reg[] = { ++ 0x00ff, 0x00ff, 0x0120, 0x0000, /* 0 */ ++ 0x00ff, 0x00ff, 0x00ff, 0x00ff, /* 4 */ ++ 0x00ff, 0x0000, 0x0080, 0x0040, /* 8 */ ++ 0x0000 ++}; ++ ++#define WM8772_DAIFMT \ ++ (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_RIGHT_J | \ ++ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_IB_NF) ++ ++#define WM8772_DIR \ ++ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE) ++ ++#define WM8772_PRATES \ ++ (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\ ++ SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000) ++ ++#define WM8772_CRATES \ ++ (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\ ++ SNDRV_PCM_RATE_96000) ++ ++static struct snd_soc_dai_mode wm8772_modes[] = { ++ /* common codec frame and clock master modes */ ++ /* 32k */ ++ { ++ .fmt = WM8772_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, ++ .pcmrate = SNDRV_PCM_RATE_32000, ++ .pcmdir = WM8772_DIR, ++ .flags = SND_SOC_DAI_BFS_RATE, ++ .fs = 768, ++ .bfs = 64, ++ }, ++ { ++ .fmt = WM8772_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, ++ .pcmrate = SNDRV_PCM_RATE_32000, ++ .pcmdir = WM8772_DIR, ++ .flags = SND_SOC_DAI_BFS_RATE, ++ .fs = 512, ++ .bfs = 64, ++ }, ++ { ++ .fmt = WM8772_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, ++ .pcmrate = SNDRV_PCM_RATE_32000, ++ .pcmdir = WM8772_DIR, ++ .flags = SND_SOC_DAI_BFS_RATE, ++ .fs = 384, ++ .bfs = 64, ++ }, ++ { ++ .fmt = WM8772_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, ++ .pcmrate = SNDRV_PCM_RATE_32000, ++ .pcmdir = WM8772_DIR, ++ .flags = SND_SOC_DAI_BFS_RATE, ++ .fs = 256, ++ .bfs = 64, ++ }, ++ { ++ .fmt = WM8772_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, ++ .pcmrate = SNDRV_PCM_RATE_32000, ++ .pcmdir = WM8772_DIR, ++ .flags = SND_SOC_DAI_BFS_RATE, ++ .fs = 192, ++ .bfs = 64, ++ }, ++ { ++ .fmt = WM8772_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, ++ .pcmrate = SNDRV_PCM_RATE_32000, ++ .pcmdir = WM8772_DIR, ++ .flags = SND_SOC_DAI_BFS_RATE, ++ .fs = 128, ++ .bfs = 64, ++ }, ++ ++ /* 44.1k */ ++ { ++ .fmt = WM8772_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, ++ .pcmrate = SNDRV_PCM_RATE_44100, ++ .pcmdir = WM8772_DIR, ++ .flags = SND_SOC_DAI_BFS_RATE, ++ .fs = 768, ++ .bfs = 64, ++ }, ++ { ++ .fmt = WM8772_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, ++ .pcmrate = SNDRV_PCM_RATE_44100, ++ .pcmdir = WM8772_DIR, ++ .flags = SND_SOC_DAI_BFS_RATE, ++ .fs = 512, ++ .bfs = 64, ++ }, ++ { ++ .fmt = WM8772_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, ++ .pcmrate = SNDRV_PCM_RATE_44100, ++ .pcmdir = WM8772_DIR, ++ .flags = SND_SOC_DAI_BFS_RATE, ++ .fs = 384, ++ .bfs = 64, ++ }, ++ { ++ .fmt = WM8772_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, ++ .pcmrate = SNDRV_PCM_RATE_44100, ++ .pcmdir = WM8772_DIR, ++ .flags = SND_SOC_DAI_BFS_RATE, ++ .fs = 256, ++ .bfs = 64, ++ }, ++ { ++ .fmt = WM8772_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, ++ .pcmrate = SNDRV_PCM_RATE_44100, ++ .pcmdir = WM8772_DIR, ++ .flags = SND_SOC_DAI_BFS_RATE, ++ .fs = 192, ++ .bfs = 64, ++ }, ++ { ++ .fmt = WM8772_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, ++ .pcmrate = SNDRV_PCM_RATE_44100, ++ .pcmdir = WM8772_DIR, ++ .flags = SND_SOC_DAI_BFS_RATE, ++ .fs = 128, ++ .bfs = 64, ++ }, ++ ++ /* 48k */ ++ { ++ .fmt = WM8772_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, ++ .pcmrate = SNDRV_PCM_RATE_48000, ++ .pcmdir = WM8772_DIR, ++ .flags = SND_SOC_DAI_BFS_RATE, ++ .fs = 768, ++ .bfs = 64, ++ }, ++ { ++ .fmt = WM8772_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, ++ .pcmrate = SNDRV_PCM_RATE_48000, ++ .pcmdir = WM8772_DIR, ++ .flags = SND_SOC_DAI_BFS_RATE, ++ .fs = 512, ++ .bfs = 64, ++ }, ++ { ++ .fmt = WM8772_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, ++ .pcmrate = SNDRV_PCM_RATE_48000, ++ .pcmdir = WM8772_DIR, ++ .flags = SND_SOC_DAI_BFS_RATE, ++ .fs = 384, ++ .bfs = 64, ++ }, ++ { ++ .fmt = WM8772_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, ++ .pcmrate = SNDRV_PCM_RATE_48000, ++ .pcmdir = WM8772_DIR, ++ .flags = SND_SOC_DAI_BFS_RATE, ++ .fs = 256, ++ .bfs = 64, ++ }, ++ { ++ .fmt = WM8772_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, ++ .pcmrate = SNDRV_PCM_RATE_48000, ++ .pcmdir = WM8772_DIR, ++ .flags = SND_SOC_DAI_BFS_RATE, ++ .fs = 192, ++ .bfs = 64, ++ }, ++ { ++ .fmt = WM8772_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, ++ .pcmrate = SNDRV_PCM_RATE_48000, ++ .pcmdir = WM8772_DIR, ++ .flags = SND_SOC_DAI_BFS_RATE, ++ .fs = 128, ++ .bfs = 64, ++ }, ++ ++ /* 96k */ ++ { ++ .fmt = WM8772_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, ++ .pcmrate = SNDRV_PCM_RATE_96000, ++ .pcmdir = WM8772_DIR, ++ .flags = SND_SOC_DAI_BFS_RATE, ++ .fs = 384, ++ .bfs = 64, ++ }, ++ { ++ .fmt = WM8772_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, ++ .pcmrate = SNDRV_PCM_RATE_96000, ++ .pcmdir = WM8772_DIR, ++ .flags = SND_SOC_DAI_BFS_RATE, ++ .fs = 256, ++ .bfs = 64, ++ }, ++ { ++ .fmt = WM8772_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, ++ .pcmrate = SNDRV_PCM_RATE_96000, ++ .pcmdir = WM8772_DIR, ++ .flags = SND_SOC_DAI_BFS_RATE, ++ .fs = 192, ++ .bfs = 64, ++ }, ++ { ++ .fmt = WM8772_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, ++ .pcmrate = SNDRV_PCM_RATE_96000, ++ .pcmdir = WM8772_DIR, ++ .pcmrate = SND_SOC_DAI_BFS_RATE, ++ .fs = 128, ++ .bfs = 64, ++ }, ++ ++ /* 192k */ ++ { ++ .fmt = WM8772_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, ++ .pcmrate = SNDRV_PCM_RATE_192000, ++ .pcmdir = SND_SOC_DAIDIR_PLAYBACK, ++ .flags = SND_SOC_DAI_BFS_RATE, ++ .fs = 192, ++ .bfs = 64, ++ }, ++ { ++ .fmt = WM8772_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, ++ .pcmrate = SNDRV_PCM_RATE_192000, ++ .pcmdir = SND_SOC_DAIDIR_PLAYBACK, ++ .flags = SND_SOC_DAI_BFS_RATE, ++ .fs = 128, ++ .bfs = 64, ++ }, ++ ++ /* slave mode */ ++ { ++ .fmt = WM8772_DAIFMT, ++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, ++ .pcmrate = WM8772_PRATES, ++ .pcmdir = SND_SOC_DAIDIR_PLAYBACK, ++ .fs = SND_SOC_FS_ALL, ++ .bfs = SND_SOC_FSB_ALL, ++ }, ++ { ++ .fmt = WM8772_DAIFMT, ++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, ++ .pcmrate = WM8772_CRATES, ++ .pcmdir = SND_SOC_DAIDIR_CAPTURE, ++ .fs = SND_SOC_FS_ALL, ++ .bfs = SND_SOC_FSB_ALL, ++ }, ++}; ++ ++/* ++ * read wm8772 register cache ++ */ ++static inline unsigned int wm8772_read_reg_cache(struct snd_soc_codec * codec, ++ unsigned int reg) ++{ ++ u16 *cache = codec->reg_cache; ++ if (reg > WM8772_CACHE_REGNUM) ++ return -1; ++ return cache[reg]; ++} ++ ++/* ++ * write wm8772 register cache ++ */ ++static inline void wm8772_write_reg_cache(struct snd_soc_codec * codec, ++ unsigned int reg, unsigned int value) ++{ ++ u16 *cache = codec->reg_cache; ++ if (reg > WM8772_CACHE_REGNUM) ++ return; ++ cache[reg] = value; ++} ++ ++static int wm8772_write(struct snd_soc_codec * codec, unsigned int reg, ++ unsigned int value) ++{ ++ u8 data[2]; ++ ++ /* data is ++ * D15..D9 WM8772 register offset ++ * D8...D0 register data ++ */ ++ data[0] = (reg << 1) | ((value >> 8) & 0x0001); ++ data[1] = value & 0x00ff; ++ ++ wm8772_write_reg_cache (codec, reg, value); ++ if (codec->hw_write(codec->control_data, data, 2) == 2) ++ return 0; ++ else ++ return -1; ++} ++ ++#define wm8772_reset(c) wm8772_write(c, WM8772_RESET, 0) ++ ++/* ++ * WM8772 Controls ++ */ ++static const char *wm8772_zero_flag[] = {"All Ch", "Ch 1", "Ch 2", "Ch3"}; ++ ++static const struct soc_enum wm8772_enum[] = { ++SOC_ENUM_SINGLE(WM8772_DACCTRL, 0, 4, wm8772_zero_flag), ++}; ++ ++static const struct snd_kcontrol_new wm8772_snd_controls[] = { ++ ++SOC_SINGLE("Left1 Playback Volume", WM8772_LDAC1VOL, 0, 255, 0), ++SOC_SINGLE("Left2 Playback Volume", WM8772_LDAC2VOL, 0, 255, 0), ++SOC_SINGLE("Left3 Playback Volume", WM8772_LDAC3VOL, 0, 255, 0), ++SOC_SINGLE("Right1 Playback Volume", WM8772_RDAC1VOL, 0, 255, 0), ++SOC_SINGLE("Right1 Playback Volume", WM8772_RDAC2VOL, 0, 255, 0), ++SOC_SINGLE("Right1 Playback Volume", WM8772_RDAC3VOL, 0, 255, 0), ++SOC_SINGLE("Master Playback Volume", WM8772_MDACVOL, 0, 255, 0), ++ ++SOC_SINGLE("Playback Switch", WM8772_DACCH, 0, 1, 0), ++SOC_SINGLE("Capture Switch", WM8772_ADCCTRL, 2, 1, 0), ++ ++SOC_SINGLE("Demp1 Playback Switch", WM8772_DACCTRL, 6, 1, 0), ++SOC_SINGLE("Demp2 Playback Switch", WM8772_DACCTRL, 7, 1, 0), ++SOC_SINGLE("Demp3 Playback Switch", WM8772_DACCTRL, 8, 1, 0), ++ ++SOC_SINGLE("Phase Invert 1 Switch", WM8772_IFACE, 6, 1, 0), ++SOC_SINGLE("Phase Invert 2 Switch", WM8772_IFACE, 7, 1, 0), ++SOC_SINGLE("Phase Invert 3 Switch", WM8772_IFACE, 8, 1, 0), ++ ++SOC_SINGLE("Playback ZC Switch", WM8772_DACCTRL, 0, 1, 0), ++ ++SOC_SINGLE("Capture High Pass Switch", WM8772_ADCCTRL, 3, 1, 0), ++}; ++ ++/* add non dapm controls */ ++static int wm8772_add_controls(struct snd_soc_codec *codec) ++{ ++ int err, i; ++ ++ for (i = 0; i < ARRAY_SIZE(wm8772_snd_controls); i++) { ++ err = snd_ctl_add(codec->card, ++ snd_soc_cnew(&wm8772_snd_controls[i],codec, NULL)); ++ if (err < 0) ++ return err; ++ } ++ return 0; ++} ++ ++/* valid wm8772 mclk frequencies */ ++static const int freq_table[5][6] = { ++ {4096000, 6144000, 8192000, 12288000, 16384000, 24576000}, ++ {5644800, 8467000, 11289600, 16934000, 22579200, 33868800}, ++ {6144000, 9216000, 12288000, 18432000, 24576000, 36864000}, ++ {12288000, 18432000, 24576000, 36864000, 0, 0}, ++ {24576000, 36864000, 0, 0, 0}, ++}; ++ ++static unsigned int check_freq(int rate, unsigned int freq) ++{ ++ int i; ++ ++ for(i = 0; i < 6; i++) { ++ if(freq == freq_table[i][rate]) ++ return freq; ++ } ++ return 0; ++} ++ ++static unsigned int wm8772_config_sysclk(struct snd_soc_codec_dai *dai, ++ struct snd_soc_clock_info *info, unsigned int clk) ++{ ++ switch (info->rate){ ++ case 32000: ++ dai->mclk = check_freq(0, clk); ++ break; ++ case 44100: ++ dai->mclk = check_freq(1, clk); ++ break; ++ case 48000: ++ dai->mclk = check_freq(2, clk); ++ break; ++ case 96000: ++ dai->mclk = check_freq(3, clk); ++ break; ++ case 192000: ++ dai->mclk = check_freq(4, clk); ++ break; ++ default: ++ dai->mclk = 0; ++ } ++ return dai->mclk; ++} ++ ++static int wm8772_pcm_prepare(struct snd_pcm_substream *substream) ++{ ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ struct snd_soc_device *socdev = rtd->socdev; ++ struct snd_soc_codec *codec = socdev->codec; ++ u16 diface = wm8772_read_reg_cache(codec, WM8772_IFACE) & 0xffc0; ++ u16 diface_ctrl = wm8772_read_reg_cache(codec, WM8772_DACRATE) & 0xfe1f; ++ u16 aiface = 0; ++ u16 aiface_ctrl = wm8772_read_reg_cache(codec, WM8772_ADCCTRL) & 0xfcff; ++ ++ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { ++ ++ /* set master/slave audio interface */ ++ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_CLOCK_MASK) { ++ case SND_SOC_DAIFMT_CBM_CFM: ++ diface_ctrl |= 0x0010; ++ break; ++ case SND_SOC_DAIFMT_CBS_CFS: ++ break; ++ } ++ ++ /* interface format */ ++ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_FORMAT_MASK) { ++ case SND_SOC_DAIFMT_I2S: ++ diface |= 0x0002; ++ break; ++ case SND_SOC_DAIFMT_RIGHT_J: ++ break; ++ case SND_SOC_DAIFMT_LEFT_J: ++ diface |= 0x0001; ++ break; ++ case SND_SOC_DAIFMT_DSP_A: ++ diface |= 0x0003; ++ break; ++ case SND_SOC_DAIFMT_DSP_B: ++ diface |= 0x0007; ++ break; ++ } ++ ++ /* bit size */ ++ switch (rtd->codec_dai->dai_runtime.pcmfmt) { ++ case SNDRV_PCM_FMTBIT_S16_LE: ++ break; ++ case SNDRV_PCM_FORMAT_S20_3LE: ++ diface |= 0x0010; ++ break; ++ case SNDRV_PCM_FORMAT_S24_3LE: ++ diface |= 0x0020; ++ break; ++ case SNDRV_PCM_FORMAT_S32_LE: ++ diface |= 0x0030; ++ break; ++ } ++ ++ /* clock inversion */ ++ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_INV_MASK) { ++ case SND_SOC_DAIFMT_NB_NF: ++ break; ++ case SND_SOC_DAIFMT_IB_NF: ++ diface |= 0x0008; ++ break; ++ } ++ ++ /* set rate */ ++ switch (rtd->codec_dai->dai_runtime.fs) { ++ case 768: ++ diface_ctrl |= (0x5 << 6); ++ break; ++ case 512: ++ diface_ctrl |= (0x4 << 6); ++ break; ++ case 384: ++ diface_ctrl |= (0x3 << 6); ++ break; ++ case 256: ++ diface_ctrl |= (0x2 << 6); ++ break; ++ case 192: ++ diface_ctrl |= (0x1 << 6); ++ break; ++ } ++ ++ wm8772_write(codec, WM8772_DACRATE, diface_ctrl); ++ wm8772_write(codec, WM8772_IFACE, diface); ++ ++ } else { ++ ++ /* set master/slave audio interface */ ++ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_CLOCK_MASK) { ++ case SND_SOC_DAIFMT_CBM_CFM: ++ aiface |= 0x0010; ++ break; ++ case SND_SOC_DAIFMT_CBS_CFS: ++ break; ++ } ++ ++ /* interface format */ ++ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_FORMAT_MASK) { ++ case SND_SOC_DAIFMT_I2S: ++ aiface |= 0x0002; ++ break; ++ case SND_SOC_DAIFMT_RIGHT_J: ++ break; ++ case SND_SOC_DAIFMT_LEFT_J: ++ aiface |= 0x0001; ++ break; ++ case SND_SOC_DAIFMT_DSP_A: ++ aiface |= 0x0003; ++ break; ++ case SND_SOC_DAIFMT_DSP_B: ++ aiface |= 0x0003; ++ aiface_ctrl |= 0x0010; ++ break; ++ } ++ ++ /* bit size */ ++ switch (rtd->codec_dai->dai_runtime.pcmfmt) { ++ case SNDRV_PCM_FMTBIT_S16_LE: ++ break; ++ case SNDRV_PCM_FMTBIT_S20_3LE: ++ aiface |= 0x0004; ++ break; ++ case SNDRV_PCM_FMTBIT_S24_LE: ++ aiface |= 0x0008; ++ break; ++ case SNDRV_PCM_FMTBIT_S32_LE: ++ aiface |= 0x000c; ++ break; ++ } ++ ++ /* clock inversion */ ++ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_INV_MASK) { ++ case SND_SOC_DAIFMT_NB_NF: ++ break; ++ case SND_SOC_DAIFMT_IB_NF: ++ aiface_ctrl |= 0x0020; ++ break; ++ } ++ ++ /* set rate */ ++ switch (rtd->codec_dai->dai_runtime.fs) { ++ case 768: ++ aiface |= (0x5 << 5); ++ break; ++ case 512: ++ aiface |= (0x4 << 5); ++ break; ++ case 384: ++ aiface |= (0x3 << 5); ++ break; ++ case 256: ++ aiface |= (0x2 << 5); ++ break; ++ } ++ ++ wm8772_write(codec, WM8772_ADCCTRL, aiface_ctrl); ++ wm8772_write(codec, WM8772_ADCRATE, aiface); ++ } ++ ++ return 0; ++} ++ ++static int wm8772_dapm_event(struct snd_soc_codec *codec, int event) ++{ ++ u16 master = wm8772_read_reg_cache(codec, WM8772_DACRATE) & 0xffe0; ++ ++ switch (event) { ++ case SNDRV_CTL_POWER_D0: /* full On */ ++ /* vref/mid, clk and osc on, dac unmute, active */ ++ wm8772_write(codec, WM8772_DACRATE, master); ++ break; ++ case SNDRV_CTL_POWER_D1: /* partial On */ ++ case SNDRV_CTL_POWER_D2: /* partial On */ ++ break; ++ case SNDRV_CTL_POWER_D3hot: /* Off, with power */ ++ /* everything off except vref/vmid, dac mute, inactive */ ++ wm8772_write(codec, WM8772_DACRATE, master | 0x0f); ++ break; ++ case SNDRV_CTL_POWER_D3cold: /* Off, without power */ ++ /* everything off, dac mute, inactive */ ++ wm8772_write(codec, WM8772_DACRATE, master | 0x1f); ++ break; ++ } ++ codec->dapm_state = event; ++ return 0; ++} ++ ++struct snd_soc_codec_dai wm8772_dai = { ++ .name = "WM8772", ++ .playback = { ++ .stream_name = "Playback", ++ .channels_min = 1, ++ .channels_max = 6, ++ }, ++ .capture = { ++ .stream_name = "Capture", ++ .channels_min = 1, ++ .channels_max = 2, ++ }, ++ .config_sysclk = wm8772_config_sysclk, ++ .ops = { ++ .prepare = wm8772_pcm_prepare, ++ }, ++ .caps = { ++ .num_modes = ARRAY_SIZE(wm8772_modes), ++ .mode = wm8772_modes, ++ }, ++}; ++EXPORT_SYMBOL_GPL(wm8772_dai); ++ ++static int wm8772_suspend(struct platform_device *pdev, pm_message_t state) ++{ ++ struct snd_soc_device *socdev = platform_get_drvdata(pdev); ++ struct snd_soc_codec *codec = socdev->codec; ++ ++ wm8772_dapm_event(codec, SNDRV_CTL_POWER_D3cold); ++ return 0; ++} ++ ++static int wm8772_resume(struct platform_device *pdev) ++{ ++ struct snd_soc_device *socdev = platform_get_drvdata(pdev); ++ struct snd_soc_codec *codec = socdev->codec; ++ int i; ++ u8 data[2]; ++ u16 *cache = codec->reg_cache; ++ ++ /* Sync reg_cache with the hardware */ ++ for (i = 0; i < ARRAY_SIZE(wm8772_reg); i++) { ++ data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); ++ data[1] = cache[i] & 0x00ff; ++ codec->hw_write(codec->control_data, data, 2); ++ } ++ wm8772_dapm_event(codec, SNDRV_CTL_POWER_D3hot); ++ wm8772_dapm_event(codec, codec->suspend_dapm_state); ++ return 0; ++} ++ ++/* ++ * initialise the WM8772 driver ++ * register the mixer and dsp interfaces with the kernel ++ */ ++static int wm8772_init(struct snd_soc_device *socdev) ++{ ++ struct snd_soc_codec *codec = socdev->codec; ++ int reg, ret = 0; ++ ++ codec->name = "WM8772"; ++ codec->owner = THIS_MODULE; ++ codec->read = wm8772_read_reg_cache; ++ codec->write = wm8772_write; ++ codec->dapm_event = wm8772_dapm_event; ++ codec->dai = &wm8772_dai; ++ codec->num_dai = 1; ++ codec->reg_cache_size = ARRAY_SIZE(wm8772_reg); ++ codec->reg_cache = ++ kzalloc(sizeof(u16) * ARRAY_SIZE(wm8772_reg), GFP_KERNEL); ++ if (codec->reg_cache == NULL) ++ return -ENOMEM; ++ memcpy(codec->reg_cache, wm8772_reg, ++ sizeof(u16) * ARRAY_SIZE(wm8772_reg)); ++ codec->reg_cache_size = sizeof(u16) * ARRAY_SIZE(wm8772_reg); ++ ++ wm8772_reset(codec); ++ ++ /* register pcms */ ++ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); ++ if(ret < 0) { ++ kfree(codec->reg_cache); ++ return ret; ++ } ++ ++ /* power on device */ ++ wm8772_dapm_event(codec, SNDRV_CTL_POWER_D3hot); ++ ++ /* set the update bits */ ++ reg = wm8772_read_reg_cache(codec, WM8772_MDACVOL); ++ wm8772_write(codec, WM8772_MDACVOL, reg | 0x0100); ++ reg = wm8772_read_reg_cache(codec, WM8772_LDAC1VOL); ++ wm8772_write(codec, WM8772_LDAC1VOL, reg | 0x0100); ++ reg = wm8772_read_reg_cache(codec, WM8772_LDAC2VOL); ++ wm8772_write(codec, WM8772_LDAC2VOL, reg | 0x0100); ++ reg = wm8772_read_reg_cache(codec, WM8772_LDAC3VOL); ++ wm8772_write(codec, WM8772_LDAC3VOL, reg | 0x0100); ++ reg = wm8772_read_reg_cache(codec, WM8772_RDAC1VOL); ++ wm8772_write(codec, WM8772_RDAC1VOL, reg | 0x0100); ++ reg = wm8772_read_reg_cache(codec, WM8772_RDAC2VOL); ++ wm8772_write(codec, WM8772_RDAC2VOL, reg | 0x0100); ++ reg = wm8772_read_reg_cache(codec, WM8772_RDAC3VOL); ++ wm8772_write(codec, WM8772_RDAC3VOL, reg | 0x0100); ++ ++ wm8772_add_controls(codec); ++ ret = snd_soc_register_card(socdev); ++ if (ret < 0) { ++ snd_soc_free_pcms(socdev); ++ snd_soc_dapm_free(socdev); ++ } ++ ++ return ret; ++} ++ ++static struct snd_soc_device *wm8772_socdev; ++ ++static int wm8772_probe(struct platform_device *pdev) ++{ ++ struct snd_soc_device *socdev = platform_get_drvdata(pdev); ++ struct wm8772_setup_data *setup; ++ struct snd_soc_codec *codec; ++ int ret = 0; ++ ++ printk(KERN_INFO "WM8772 Audio Codec %s", WM8772_VERSION); ++ ++ setup = socdev->codec_data; ++ codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); ++ if (codec == NULL) ++ return -ENOMEM; ++ ++ socdev->codec = codec; ++ mutex_init(&codec->mutex); ++ INIT_LIST_HEAD(&codec->dapm_widgets); ++ INIT_LIST_HEAD(&codec->dapm_paths); ++ ++ wm8772_socdev = socdev; ++ ++ /* Add other interfaces here */ ++#warning do SPI device probe here and then call wm8772_init() ++ ++ return ret; ++} ++ ++/* power down chip */ ++static int wm8772_remove(struct platform_device *pdev) ++{ ++ struct snd_soc_device *socdev = platform_get_drvdata(pdev); ++ struct snd_soc_codec *codec = socdev->codec; ++ ++ if (codec->control_data) ++ wm8772_dapm_event(codec, SNDRV_CTL_POWER_D3cold); ++ ++ snd_soc_free_pcms(socdev); ++ kfree(codec); ++ ++ return 0; ++} ++ ++struct snd_soc_codec_device soc_codec_dev_wm8772 = { ++ .probe = wm8772_probe, ++ .remove = wm8772_remove, ++ .suspend = wm8772_suspend, ++ .resume = wm8772_resume, ++}; ++ ++EXPORT_SYMBOL_GPL(soc_codec_dev_wm8772); ++ ++MODULE_DESCRIPTION("ASoC WM8772 driver"); ++MODULE_AUTHOR("Liam Girdwood"); ++MODULE_LICENSE("GPL"); +Index: linux-2.6-pxa-new/sound/soc/codecs/wm8772.h +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/sound/soc/codecs/wm8772.h +@@ -0,0 +1,40 @@ ++/* ++ * wm8772.h -- audio driver for WM8772 ++ * ++ * Copyright 2005 Wolfson Microelectronics PLC. ++ * Author: Liam Girdwood ++ * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com ++ * ++ * This program is free software; you can redistribute it and/or modify it ++ * under the terms of the GNU General Public License as published by the ++ * Free Software Foundation; either version 2 of the License, or (at your ++ * option) any later version. ++ * ++ */ ++ ++#ifndef _WM8772_H ++#define _WM8772_H ++ ++/* WM8772 register space */ ++ ++#define WM8772_LDAC1VOL 0x00 ++#define WM8772_RDAC1VOL 0x01 ++#define WM8772_DACCH 0x02 ++#define WM8772_IFACE 0x03 ++#define WM8772_LDAC2VOL 0x04 ++#define WM8772_RDAC2VOL 0x05 ++#define WM8772_LDAC3VOL 0x06 ++#define WM8772_RDAC3VOL 0x07 ++#define WM8772_MDACVOL 0x08 ++#define WM8772_DACCTRL 0x09 ++#define WM8772_DACRATE 0x0a ++#define WM8772_ADCRATE 0x0b ++#define WM8772_ADCCTRL 0x0c ++#define WM8772_RESET 0x1f ++ ++#define WM8772_CACHE_REGNUM 10 ++ ++extern struct snd_soc_codec_dai wm8772_dai; ++extern struct snd_soc_codec_device soc_codec_dev_wm8772; ++ ++#endif +Index: linux-2.6-pxa-new/sound/soc/codecs/wm8971.c +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/sound/soc/codecs/wm8971.c +@@ -0,0 +1,1214 @@ ++/* ++ * wm8971.c -- WM8971 ALSA SoC Audio driver ++ * ++ * Copyright 2005 Lab126, Inc. ++ * ++ * Author: Kenneth Kiraly <kiraly@lab126.com> ++ * ++ * Based on wm8753.c by Liam Girdwood ++ * ++ * This program is free software; you can redistribute it and/or modify it ++ * under the terms of the GNU General Public License as published by the ++ * Free Software Foundation; either version 2 of the License, or (at your ++ * option) any later version. ++ */ ++ ++#include <linux/module.h> ++#include <linux/moduleparam.h> ++#include <linux/init.h> ++#include <linux/delay.h> ++#include <linux/pm.h> ++#include <linux/i2c.h> ++#include <linux/platform_device.h> ++#include <sound/driver.h> ++#include <sound/core.h> ++#include <sound/pcm.h> ++#include <sound/pcm_params.h> ++#include <sound/soc.h> ++#include <sound/soc-dapm.h> ++#include <sound/initval.h> ++ ++#include "wm8971.h" ++ ++#define AUDIO_NAME "wm8971" ++#define WM8971_VERSION "0.8" ++ ++#undef WM8971_DEBUG ++ ++#ifdef WM8971_DEBUG ++#define dbg(format, arg...) \ ++ printk(KERN_DEBUG AUDIO_NAME ": " format "\n" , ## arg) ++#else ++#define dbg(format, arg...) do {} while (0) ++#endif ++#define err(format, arg...) \ ++ printk(KERN_ERR AUDIO_NAME ": " format "\n" , ## arg) ++#define info(format, arg...) \ ++ printk(KERN_INFO AUDIO_NAME ": " format "\n" , ## arg) ++#define warn(format, arg...) \ ++ printk(KERN_WARNING AUDIO_NAME ": " format "\n" , ## arg) ++ ++#define WM8971_REG_COUNT 43 ++ ++static struct workqueue_struct *wm8971_workq = NULL; ++static struct work_struct wm8971_dapm_work; ++ ++/* ++ * wm8971 register cache ++ * We can't read the WM8971 register space when we ++ * are using 2 wire for device control, so we cache them instead. ++ */ ++static const u16 wm8971_reg[] = { ++ 0x0097, 0x0097, 0x0079, 0x0079, /* 0 */ ++ 0x0000, 0x0008, 0x0000, 0x000a, /* 4 */ ++ 0x0000, 0x0000, 0x00ff, 0x00ff, /* 8 */ ++ 0x000f, 0x000f, 0x0000, 0x0000, /* 12 */ ++ 0x0000, 0x007b, 0x0000, 0x0032, /* 16 */ ++ 0x0000, 0x00c3, 0x00c3, 0x00c0, /* 20 */ ++ 0x0000, 0x0000, 0x0000, 0x0000, /* 24 */ ++ 0x0000, 0x0000, 0x0000, 0x0000, /* 28 */ ++ 0x0000, 0x0000, 0x0050, 0x0050, /* 32 */ ++ 0x0050, 0x0050, 0x0050, 0x0050, /* 36 */ ++ 0x0079, 0x0079, 0x0079, /* 40 */ ++}; ++ ++#define WM8971_HIFI_DAIFMT \ ++ (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_RIGHT_J | \ ++ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_NB_IF | SND_SOC_DAIFMT_IB_NF | \ ++ SND_SOC_DAIFMT_IB_IF) ++ ++#define WM8971_DIR \ ++ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE) ++ ++#define WM8971_HIFI_FSB \ ++ (SND_SOC_FSBD(1) | SND_SOC_FSBD(2) | SND_SOC_FSBD(4) | \ ++ SND_SOC_FSBD(8) | SND_SOC_FSBD(16)) ++ ++#define WM8971_HIFI_RATES \ ++ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ ++ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ ++ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) ++ ++#define WM8971_HIFI_BITS \ ++ (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ ++ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) ++ ++static struct snd_soc_dai_mode wm8971_modes[] = { ++ /* common codec frame and clock master modes */ ++ /* 8k */ ++ { ++ .fmt = WM8971_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8971_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_8000, ++ .pcmdir = WM8971_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 1536, ++ .bfs = WM8971_HIFI_FSB, ++ }, ++ { ++ .fmt = WM8971_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8971_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_8000, ++ .pcmdir = WM8971_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 1408, ++ .bfs = WM8971_HIFI_FSB, ++ }, ++ { ++ .fmt = WM8971_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8971_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_8000, ++ .pcmdir = WM8971_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 2304, ++ .bfs = WM8971_HIFI_FSB, ++ }, ++ { ++ .fmt = WM8971_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8971_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_8000, ++ .pcmdir = WM8971_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 2112, ++ .bfs = WM8971_HIFI_FSB, ++ }, ++ { ++ .fmt = WM8971_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8971_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_8000, ++ .pcmdir = WM8971_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 1500, ++ .bfs = WM8971_HIFI_FSB, ++ }, ++ ++ /* 11.025k */ ++ { ++ .fmt = WM8971_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8971_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_11025, ++ .pcmdir = WM8971_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 1024, ++ .bfs = WM8971_HIFI_FSB, ++ }, ++ { ++ .fmt = WM8971_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8971_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_11025, ++ .pcmdir = WM8971_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 1536, ++ .bfs = WM8971_HIFI_FSB, ++ }, ++ { ++ .fmt = WM8971_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8971_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_11025, ++ .pcmdir = WM8971_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 1088, ++ .bfs = WM8971_HIFI_FSB, ++ }, ++ ++ /* 16k */ ++ { ++ .fmt = WM8971_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8971_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_16000, ++ .pcmdir = WM8971_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 768, ++ .bfs = WM8971_HIFI_FSB, ++ }, ++ { ++ .fmt = WM8971_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8971_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_16000, ++ .pcmdir = WM8971_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 1152, ++ .bfs = WM8971_HIFI_FSB ++ }, ++ { ++ .fmt = WM8971_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8971_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_16000, ++ .pcmdir = WM8971_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 750, ++ .bfs = WM8971_HIFI_FSB, ++ }, ++ ++ /* 22.05k */ ++ { ++ .fmt = WM8971_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8971_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_22050, ++ .pcmdir = WM8971_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 512, ++ .bfs = WM8971_HIFI_FSB, ++ }, ++ { ++ .fmt = WM8971_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8971_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_22050, ++ .pcmdir = WM8971_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 768, ++ .bfs = WM8971_HIFI_FSB, ++ }, ++ { ++ .fmt = WM8971_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8971_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_22050, ++ .pcmdir = WM8971_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 544, ++ .bfs = WM8971_HIFI_FSB, ++ }, ++ ++ /* 32k */ ++ { ++ .fmt = WM8971_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8971_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_32000, ++ .pcmdir = WM8971_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 384, ++ .bfs = WM8971_HIFI_FSB, ++ }, ++ { ++ .fmt = WM8971_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8971_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_32000, ++ .pcmdir = WM8971_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 576, ++ .bfs = WM8971_HIFI_FSB, ++ }, ++ { ++ .fmt = WM8971_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8971_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_32000, ++ .pcmdir = WM8971_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 375, ++ .bfs = WM8971_HIFI_FSB, ++ }, ++ ++ /* 44.1k & 48k */ ++ { ++ .fmt = WM8971_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8971_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000, ++ .pcmdir = WM8971_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 256, ++ .bfs = WM8971_HIFI_FSB, ++ }, ++ { ++ .fmt = WM8971_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8971_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000, ++ .pcmdir = WM8971_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 384, ++ .bfs = WM8971_HIFI_FSB, ++ }, ++ { ++ .fmt = WM8971_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8971_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_44100, ++ .pcmdir = WM8971_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 272, ++ .bfs = WM8971_HIFI_FSB, ++ }, ++ { ++ .fmt = WM8971_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8971_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_48000, ++ .pcmdir = WM8971_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 250, ++ .bfs = WM8971_HIFI_FSB, ++ }, ++ ++ /* 88.2k & 96k */ ++ { ++ .fmt = WM8971_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8971_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000, ++ .pcmdir = WM8971_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 128, ++ .bfs = WM8971_HIFI_FSB, ++ }, ++ { ++ .fmt = WM8971_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8971_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000, ++ .pcmdir = WM8971_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 192, ++ .bfs = WM8971_HIFI_FSB, ++ }, ++ { ++ .fmt = WM8971_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8971_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_88200, ++ .pcmdir = WM8971_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 136, ++ .bfs = WM8971_HIFI_FSB, ++ }, ++ { ++ .fmt = WM8971_HIFI_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8971_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_96000, ++ .pcmdir = WM8971_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 125, ++ .bfs = WM8971_HIFI_FSB, ++ }, ++ ++ /* codec frame and clock slave modes */ ++ { ++ .fmt = WM8971_HIFI_DAIFMT | SND_SOC_DAIFMT_CBS_CFS, ++ .pcmfmt = WM8971_HIFI_BITS, ++ .pcmrate = WM8971_HIFI_RATES, ++ .pcmdir = WM8971_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = SND_SOC_FS_ALL, ++ .bfs = SND_SOC_FSB_ALL, ++ }, ++}; ++ ++static inline unsigned int wm8971_read_reg_cache(struct snd_soc_codec *codec, ++ unsigned int reg) ++{ ++ u16 *cache = codec->reg_cache; ++ if (reg < WM8971_REG_COUNT) ++ return cache[reg]; ++ ++ return -1; ++} ++ ++static inline void wm8971_write_reg_cache(struct snd_soc_codec *codec, ++ unsigned int reg, unsigned int value) ++{ ++ u16 *cache = codec->reg_cache; ++ if (reg < WM8971_REG_COUNT) ++ cache[reg] = value; ++} ++ ++static int wm8971_write(struct snd_soc_codec *codec, unsigned int reg, ++ unsigned int value) ++{ ++ u8 data[2]; ++ ++ /* data is ++ * D15..D9 WM8753 register offset ++ * D8...D0 register data ++ */ ++ data[0] = (reg << 1) | ((value >> 8) & 0x0001); ++ data[1] = value & 0x00ff; ++ ++ wm8971_write_reg_cache (codec, reg, value); ++ if (codec->hw_write(codec->control_data, data, 2) == 2) ++ return 0; ++ else ++ return -EIO; ++} ++ ++#define wm8971_reset(c) wm8971_write(c, WM8971_RESET, 0) ++ ++/* WM8971 Controls */ ++static const char *wm8971_bass[] = { "Linear Control", "Adaptive Boost" }; ++static const char *wm8971_bass_filter[] = { "130Hz @ 48kHz", ++ "200Hz @ 48kHz" }; ++static const char *wm8971_treble[] = { "8kHz", "4kHz" }; ++static const char *wm8971_alc_func[] = { "Off", "Right", "Left", "Stereo" }; ++static const char *wm8971_ng_type[] = { "Constant PGA Gain", ++ "Mute ADC Output" }; ++static const char *wm8971_deemp[] = { "None", "32kHz", "44.1kHz", "48kHz" }; ++static const char *wm8971_mono_mux[] = {"Stereo", "Mono (Left)", ++ "Mono (Right)", "Digital Mono"}; ++static const char *wm8971_dac_phase[] = { "Non Inverted", "Inverted" }; ++static const char *wm8971_lline_mux[] = {"Line", "NC", "NC", "PGA", ++ "Differential"}; ++static const char *wm8971_rline_mux[] = {"Line", "Mic", "NC", "PGA", ++ "Differential"}; ++static const char *wm8971_lpga_sel[] = {"Line", "NC", "NC", "Differential"}; ++static const char *wm8971_rpga_sel[] = {"Line", "Mic", "NC", "Differential"}; ++static const char *wm8971_adcpol[] = {"Normal", "L Invert", "R Invert", ++ "L + R Invert"}; ++ ++static const struct soc_enum wm8971_enum[] = { ++ SOC_ENUM_SINGLE(WM8971_BASS, 7, 2, wm8971_bass), /* 0 */ ++ SOC_ENUM_SINGLE(WM8971_BASS, 6, 2, wm8971_bass_filter), ++ SOC_ENUM_SINGLE(WM8971_TREBLE, 6, 2, wm8971_treble), ++ SOC_ENUM_SINGLE(WM8971_ALC1, 7, 4, wm8971_alc_func), ++ SOC_ENUM_SINGLE(WM8971_NGATE, 1, 2, wm8971_ng_type), /* 4 */ ++ SOC_ENUM_SINGLE(WM8971_ADCDAC, 1, 4, wm8971_deemp), ++ SOC_ENUM_SINGLE(WM8971_ADCTL1, 4, 4, wm8971_mono_mux), ++ SOC_ENUM_SINGLE(WM8971_ADCTL1, 1, 2, wm8971_dac_phase), ++ SOC_ENUM_SINGLE(WM8971_LOUTM1, 0, 5, wm8971_lline_mux), /* 8 */ ++ SOC_ENUM_SINGLE(WM8971_ROUTM1, 0, 5, wm8971_rline_mux), ++ SOC_ENUM_SINGLE(WM8971_LADCIN, 6, 4, wm8971_lpga_sel), ++ SOC_ENUM_SINGLE(WM8971_RADCIN, 6, 4, wm8971_rpga_sel), ++ SOC_ENUM_SINGLE(WM8971_ADCDAC, 5, 4, wm8971_adcpol), /* 12 */ ++ SOC_ENUM_SINGLE(WM8971_ADCIN, 6, 4, wm8971_mono_mux), ++}; ++ ++static const struct snd_kcontrol_new wm8971_snd_controls[] = { ++ SOC_DOUBLE_R("Capture Volume", WM8971_LINVOL, WM8971_RINVOL, 0, 63, 0), ++ SOC_DOUBLE_R("Capture ZC Switch", WM8971_LINVOL, WM8971_RINVOL, 6, 1, 0), ++ SOC_DOUBLE_R("Capture Switch", WM8971_LINVOL, WM8971_RINVOL, 7, 1, 1), ++ ++ SOC_DOUBLE_R("Headphone Playback ZC Switch", WM8971_LOUT1V, ++ WM8971_ROUT1V, 7, 1, 0), ++ SOC_DOUBLE_R("Speaker Playback ZC Switch", WM8971_LOUT2V, ++ WM8971_ROUT2V, 7, 1, 0), ++ SOC_SINGLE("Mono Playback ZC Switch", WM8971_MOUTV, 7, 1, 0), ++ ++ SOC_DOUBLE_R("PCM Volume", WM8971_LDAC, WM8971_RDAC, 0, 255, 0), ++ ++ SOC_DOUBLE_R("Bypass Left Playback Volume", WM8971_LOUTM1, ++ WM8971_LOUTM2, 4, 7, 1), ++ SOC_DOUBLE_R("Bypass Right Playback Volume", WM8971_ROUTM1, ++ WM8971_ROUTM2, 4, 7, 1), ++ SOC_DOUBLE_R("Bypass Mono Playback Volume", WM8971_MOUTM1, ++ WM8971_MOUTM2, 4, 7, 1), ++ ++ SOC_DOUBLE_R("Headphone Playback Volume", WM8971_LOUT1V, ++ WM8971_ROUT1V, 0, 127, 0), ++ SOC_DOUBLE_R("Speaker Playback Volume", WM8971_LOUT2V, ++ WM8971_ROUT2V, 0, 127, 0), ++ ++ SOC_ENUM("Bass Boost", wm8971_enum[0]), ++ SOC_ENUM("Bass Filter", wm8971_enum[1]), ++ SOC_SINGLE("Bass Volume", WM8971_BASS, 0, 7, 1), ++ ++ SOC_SINGLE("Treble Volume", WM8971_TREBLE, 0, 7, 0), ++ SOC_ENUM("Treble Cut-off", wm8971_enum[2]), ++ ++ SOC_SINGLE("Capture Filter Switch", WM8971_ADCDAC, 0, 1, 1), ++ ++ SOC_SINGLE("ALC Target Volume", WM8971_ALC1, 0, 7, 0), ++ SOC_SINGLE("ALC Max Volume", WM8971_ALC1, 4, 7, 0), ++ ++ SOC_SINGLE("ALC Capture Target Volume", WM8971_ALC1, 0, 7, 0), ++ SOC_SINGLE("ALC Capture Max Volume", WM8971_ALC1, 4, 7, 0), ++ SOC_ENUM("ALC Capture Function", wm8971_enum[3]), ++ SOC_SINGLE("ALC Capture ZC Switch", WM8971_ALC2, 7, 1, 0), ++ SOC_SINGLE("ALC Capture Hold Time", WM8971_ALC2, 0, 15, 0), ++ SOC_SINGLE("ALC Capture Decay Time", WM8971_ALC3, 4, 15, 0), ++ SOC_SINGLE("ALC Capture Attack Time", WM8971_ALC3, 0, 15, 0), ++ SOC_SINGLE("ALC Capture NG Threshold", WM8971_NGATE, 3, 31, 0), ++ SOC_ENUM("ALC Capture NG Type", wm8971_enum[4]), ++ SOC_SINGLE("ALC Capture NG Switch", WM8971_NGATE, 0, 1, 0), ++ ++ SOC_SINGLE("Capture 6dB Attenuate", WM8971_ADCDAC, 8, 1, 0), ++ SOC_SINGLE("Playback 6dB Attenuate", WM8971_ADCDAC, 7, 1, 0), ++ ++ SOC_ENUM("Playback De-emphasis", wm8971_enum[5]), ++ SOC_ENUM("Playback Function", wm8971_enum[6]), ++ SOC_ENUM("Playback Phase", wm8971_enum[7]), ++ ++ SOC_DOUBLE_R("Mic Boost", WM8971_LADCIN, WM8971_RADCIN, 4, 3, 0), ++}; ++ ++/* add non-DAPM controls */ ++static int wm8971_add_controls(struct snd_soc_codec *codec) ++{ ++ int err, i; ++ ++ for (i = 0; i < ARRAY_SIZE(wm8971_snd_controls); i++) { ++ err = snd_ctl_add(codec->card, ++ snd_soc_cnew(&wm8971_snd_controls[i],codec, NULL)); ++ if (err < 0) ++ return err; ++ } ++ ++ return 0; ++} ++ ++/* ++ * DAPM Controls ++ */ ++ ++/* Left Mixer */ ++static const struct snd_kcontrol_new wm8971_left_mixer_controls[] = { ++SOC_DAPM_SINGLE("Playback Switch", WM8971_LOUTM1, 8, 1, 0), ++SOC_DAPM_SINGLE("Left Bypass Switch", WM8971_LOUTM1, 7, 1, 0), ++SOC_DAPM_SINGLE("Right Playback Switch", WM8971_LOUTM2, 8, 1, 0), ++SOC_DAPM_SINGLE("Right Bypass Switch", WM8971_LOUTM2, 7, 1, 0), ++}; ++ ++/* Right Mixer */ ++static const struct snd_kcontrol_new wm8971_right_mixer_controls[] = { ++SOC_DAPM_SINGLE("Left Playback Switch", WM8971_ROUTM1, 8, 1, 0), ++SOC_DAPM_SINGLE("Left Bypass Switch", WM8971_ROUTM1, 7, 1, 0), ++SOC_DAPM_SINGLE("Playback Switch", WM8971_ROUTM2, 8, 1, 0), ++SOC_DAPM_SINGLE("Right Bypass Switch", WM8971_ROUTM2, 7, 1, 0), ++}; ++ ++/* Mono Mixer */ ++static const struct snd_kcontrol_new wm8971_mono_mixer_controls[] = { ++SOC_DAPM_SINGLE("Left Playback Switch", WM8971_MOUTM1, 8, 1, 0), ++SOC_DAPM_SINGLE("Left Bypass Switch", WM8971_MOUTM1, 7, 1, 0), ++SOC_DAPM_SINGLE("Right Playback Switch", WM8971_MOUTM2, 8, 1, 0), ++SOC_DAPM_SINGLE("Right Bypass Switch", WM8971_MOUTM2, 7, 1, 0), ++}; ++ ++/* Left Line Mux */ ++static const struct snd_kcontrol_new wm8971_left_line_controls = ++SOC_DAPM_ENUM("Route", wm8971_enum[8]); ++ ++/* Right Line Mux */ ++static const struct snd_kcontrol_new wm8971_right_line_controls = ++SOC_DAPM_ENUM("Route", wm8971_enum[9]); ++ ++/* Left PGA Mux */ ++static const struct snd_kcontrol_new wm8971_left_pga_controls = ++SOC_DAPM_ENUM("Route", wm8971_enum[10]); ++ ++/* Right PGA Mux */ ++static const struct snd_kcontrol_new wm8971_right_pga_controls = ++SOC_DAPM_ENUM("Route", wm8971_enum[11]); ++ ++/* Mono ADC Mux */ ++static const struct snd_kcontrol_new wm8971_monomux_controls = ++SOC_DAPM_ENUM("Route", wm8971_enum[13]); ++ ++static const struct snd_soc_dapm_widget wm8971_dapm_widgets[] = { ++ SND_SOC_DAPM_MIXER("Left Mixer", SND_SOC_NOPM, 0, 0, ++ &wm8971_left_mixer_controls[0], ++ ARRAY_SIZE(wm8971_left_mixer_controls)), ++ SND_SOC_DAPM_MIXER("Right Mixer", SND_SOC_NOPM, 0, 0, ++ &wm8971_right_mixer_controls[0], ++ ARRAY_SIZE(wm8971_right_mixer_controls)), ++ SND_SOC_DAPM_MIXER("Mono Mixer", WM8971_PWR2, 2, 0, ++ &wm8971_mono_mixer_controls[0], ++ ARRAY_SIZE(wm8971_mono_mixer_controls)), ++ ++ SND_SOC_DAPM_PGA("Right Out 2", WM8971_PWR2, 3, 0, NULL, 0), ++ SND_SOC_DAPM_PGA("Left Out 2", WM8971_PWR2, 4, 0, NULL, 0), ++ SND_SOC_DAPM_PGA("Right Out 1", WM8971_PWR2, 5, 0, NULL, 0), ++ SND_SOC_DAPM_PGA("Left Out 1", WM8971_PWR2, 6, 0, NULL, 0), ++ SND_SOC_DAPM_DAC("Right DAC", "Right Playback", WM8971_PWR2, 7, 0), ++ SND_SOC_DAPM_DAC("Left DAC", "Left Playback", WM8971_PWR2, 8, 0), ++ SND_SOC_DAPM_PGA("Mono Out 1", WM8971_PWR2, 2, 0, NULL, 0), ++ ++ SND_SOC_DAPM_MICBIAS("Mic Bias", WM8971_PWR1, 1, 0), ++ SND_SOC_DAPM_ADC("Right ADC", "Right Capture", WM8971_PWR1, 2, 0), ++ SND_SOC_DAPM_ADC("Left ADC", "Left Capture", WM8971_PWR1, 3, 0), ++ ++ SND_SOC_DAPM_MUX("Left PGA Mux", WM8971_PWR1, 5, 0, ++ &wm8971_left_pga_controls), ++ SND_SOC_DAPM_MUX("Right PGA Mux", WM8971_PWR1, 4, 0, ++ &wm8971_right_pga_controls), ++ SND_SOC_DAPM_MUX("Left Line Mux", SND_SOC_NOPM, 0, 0, ++ &wm8971_left_line_controls), ++ SND_SOC_DAPM_MUX("Right Line Mux", SND_SOC_NOPM, 0, 0, ++ &wm8971_right_line_controls), ++ ++ SND_SOC_DAPM_MUX("Left ADC Mux", SND_SOC_NOPM, 0, 0, ++ &wm8971_monomux_controls), ++ SND_SOC_DAPM_MUX("Right ADC Mux", SND_SOC_NOPM, 0, 0, ++ &wm8971_monomux_controls), ++ ++ SND_SOC_DAPM_OUTPUT("LOUT1"), ++ SND_SOC_DAPM_OUTPUT("ROUT1"), ++ SND_SOC_DAPM_OUTPUT("LOUT2"), ++ SND_SOC_DAPM_OUTPUT("ROUT2"), ++ SND_SOC_DAPM_OUTPUT("MONO"), ++ ++ SND_SOC_DAPM_INPUT("LINPUT1"), ++ SND_SOC_DAPM_INPUT("RINPUT1"), ++ SND_SOC_DAPM_INPUT("MIC"), ++}; ++ ++static const char *audio_map[][3] = { ++ /* left mixer */ ++ {"Left Mixer", "Playback Switch", "Left DAC"}, ++ {"Left Mixer", "Left Bypass Switch", "Left Line Mux"}, ++ {"Left Mixer", "Right Playback Switch", "Right DAC"}, ++ {"Left Mixer", "Right Bypass Switch", "Right Line Mux"}, ++ ++ /* right mixer */ ++ {"Right Mixer", "Left Playback Switch", "Left DAC"}, ++ {"Right Mixer", "Left Bypass Switch", "Left Line Mux"}, ++ {"Right Mixer", "Playback Switch", "Right DAC"}, ++ {"Right Mixer", "Right Bypass Switch", "Right Line Mux"}, ++ ++ /* left out 1 */ ++ {"Left Out 1", NULL, "Left Mixer"}, ++ {"LOUT1", NULL, "Left Out 1"}, ++ ++ /* left out 2 */ ++ {"Left Out 2", NULL, "Left Mixer"}, ++ {"LOUT2", NULL, "Left Out 2"}, ++ ++ /* right out 1 */ ++ {"Right Out 1", NULL, "Right Mixer"}, ++ {"ROUT1", NULL, "Right Out 1"}, ++ ++ /* right out 2 */ ++ {"Right Out 2", NULL, "Right Mixer"}, ++ {"ROUT2", NULL, "Right Out 2"}, ++ ++ /* mono mixer */ ++ {"Mono Mixer", "Left Playback Switch", "Left DAC"}, ++ {"Mono Mixer", "Left Bypass Switch", "Left Line Mux"}, ++ {"Mono Mixer", "Right Playback Switch", "Right DAC"}, ++ {"Mono Mixer", "Right Bypass Switch", "Right Line Mux"}, ++ ++ /* mono out */ ++ {"Mono Out", NULL, "Mono Mixer"}, ++ {"MONO1", NULL, "Mono Out"}, ++ ++ /* Left Line Mux */ ++ {"Left Line Mux", "Line", "LINPUT1"}, ++ {"Left Line Mux", "PGA", "Left PGA Mux"}, ++ {"Left Line Mux", "Differential", "Differential Mux"}, ++ ++ /* Right Line Mux */ ++ {"Right Line Mux", "Line", "RINPUT1"}, ++ {"Right Line Mux", "Mic", "MIC"}, ++ {"Right Line Mux", "PGA", "Right PGA Mux"}, ++ {"Right Line Mux", "Differential", "Differential Mux"}, ++ ++ /* Left PGA Mux */ ++ {"Left PGA Mux", "Line", "LINPUT1"}, ++ {"Left PGA Mux", "Differential", "Differential Mux"}, ++ ++ /* Right PGA Mux */ ++ {"Right PGA Mux", "Line", "RINPUT1"}, ++ {"Right PGA Mux", "Differential", "Differential Mux"}, ++ ++ /* Differential Mux */ ++ {"Differential Mux", "Line", "LINPUT1"}, ++ {"Differential Mux", "Line", "RINPUT1"}, ++ ++ /* Left ADC Mux */ ++ {"Left ADC Mux", "Stereo", "Left PGA Mux"}, ++ {"Left ADC Mux", "Mono (Left)", "Left PGA Mux"}, ++ {"Left ADC Mux", "Digital Mono", "Left PGA Mux"}, ++ ++ /* Right ADC Mux */ ++ {"Right ADC Mux", "Stereo", "Right PGA Mux"}, ++ {"Right ADC Mux", "Mono (Right)", "Right PGA Mux"}, ++ {"Right ADC Mux", "Digital Mono", "Right PGA Mux"}, ++ ++ /* ADC */ ++ {"Left ADC", NULL, "Left ADC Mux"}, ++ {"Right ADC", NULL, "Right ADC Mux"}, ++ ++ /* terminator */ ++ {NULL, NULL, NULL}, ++}; ++ ++static int wm8971_add_widgets(struct snd_soc_codec *codec) ++{ ++ int i; ++ ++ for(i = 0; i < ARRAY_SIZE(wm8971_dapm_widgets); i++) { ++ snd_soc_dapm_new_control(codec, &wm8971_dapm_widgets[i]); ++ } ++ ++ /* set up audio path audio_mapnects */ ++ for(i = 0; audio_map[i][0] != NULL; i++) { ++ snd_soc_dapm_connect_input(codec, audio_map[i][0], ++ audio_map[i][1], audio_map[i][2]); ++ } ++ ++ snd_soc_dapm_new_widgets(codec); ++ return 0; ++} ++ ++struct _coeff_div { ++ u32 mclk; ++ u32 rate; ++ u16 fs; ++ u8 sr:5; ++ u8 usb:1; ++}; ++ ++/* codec hifi mclk clock divider coefficients */ ++static const struct _coeff_div coeff_div[] = { ++ /* 8k */ ++ {12288000, 8000, 1536, 0x6, 0x0}, ++ {11289600, 8000, 1408, 0x16, 0x0}, ++ {18432000, 8000, 2304, 0x7, 0x0}, ++ {16934400, 8000, 2112, 0x17, 0x0}, ++ {12000000, 8000, 1500, 0x6, 0x1}, ++ ++ /* 11.025k */ ++ {11289600, 11025, 1024, 0x18, 0x0}, ++ {16934400, 11025, 1536, 0x19, 0x0}, ++ {12000000, 11025, 1088, 0x19, 0x1}, ++ ++ /* 16k */ ++ {12288000, 16000, 768, 0xa, 0x0}, ++ {18432000, 16000, 1152, 0xb, 0x0}, ++ {12000000, 16000, 750, 0xa, 0x1}, ++ ++ /* 22.05k */ ++ {11289600, 22050, 512, 0x1a, 0x0}, ++ {16934400, 22050, 768, 0x1b, 0x0}, ++ {12000000, 22050, 544, 0x1b, 0x1}, ++ ++ /* 32k */ ++ {12288000, 32000, 384, 0xc, 0x0}, ++ {18432000, 32000, 576, 0xd, 0x0}, ++ {12000000, 32000, 375, 0xa, 0x1}, ++ ++ /* 44.1k */ ++ {11289600, 44100, 256, 0x10, 0x0}, ++ {16934400, 44100, 384, 0x11, 0x0}, ++ {12000000, 44100, 272, 0x11, 0x1}, ++ ++ /* 48k */ ++ {12288000, 48000, 256, 0x0, 0x0}, ++ {18432000, 48000, 384, 0x1, 0x0}, ++ {12000000, 48000, 250, 0x0, 0x1}, ++ ++ /* 88.2k */ ++ {11289600, 88200, 128, 0x1e, 0x0}, ++ {16934400, 88200, 192, 0x1f, 0x0}, ++ {12000000, 88200, 136, 0x1f, 0x1}, ++ ++ /* 96k */ ++ {12288000, 96000, 128, 0xe, 0x0}, ++ {18432000, 96000, 192, 0xf, 0x0}, ++ {12000000, 96000, 125, 0xe, 0x1}, ++}; ++ ++static int get_coeff(int mclk, int rate) ++{ ++ int i; ++ ++ for (i = 0; i < ARRAY_SIZE(coeff_div); i++) { ++ if (coeff_div[i].rate == rate && coeff_div[i].mclk == mclk) ++ return i; ++ } ++ return -EINVAL; ++} ++ ++/* WM8971 supports numerous input clocks per sample rate */ ++static unsigned int wm8971_config_sysclk(struct snd_soc_codec_dai *dai, ++ struct snd_soc_clock_info *info, unsigned int clk) ++{ ++ dai->mclk = 0; ++ ++ /* check that the calculated FS and rate actually match a clock from ++ * the machine driver */ ++ if (info->fs * info->rate == clk) ++ dai->mclk = clk; ++ ++ return dai->mclk; ++} ++ ++static int wm8971_pcm_prepare(struct snd_pcm_substream *substream) ++{ ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ struct snd_soc_device *socdev = rtd->socdev; ++ struct snd_soc_codec *codec = socdev->codec; ++ u16 iface = 0, bfs, srate = 0; ++ int i = get_coeff(rtd->codec_dai->mclk, ++ snd_soc_get_rate(rtd->codec_dai->dai_runtime.pcmrate)); ++ ++ /* is coefficient valid ? */ ++ if (i < 0) ++ return i; ++ ++ bfs = SND_SOC_FSBD_REAL(rtd->codec_dai->dai_runtime.bfs); ++ ++ /* set master/slave audio interface */ ++ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_CLOCK_MASK) { ++ case SND_SOC_DAIFMT_CBM_CFM: ++ iface |= 0x0040; ++ break; ++ case SND_SOC_DAIFMT_CBS_CFS: ++ break; ++ } ++ ++ /* interface format */ ++ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_FORMAT_MASK) { ++ case SND_SOC_DAIFMT_I2S: ++ iface |= 0x0002; ++ break; ++ case SND_SOC_DAIFMT_RIGHT_J: ++ break; ++ case SND_SOC_DAIFMT_LEFT_J: ++ iface |= 0x0001; ++ break; ++ case SND_SOC_DAIFMT_DSP_A: ++ iface |= 0x0003; ++ break; ++ case SND_SOC_DAIFMT_DSP_B: ++ iface |= 0x0013; ++ break; ++ } ++ ++ /* bit size */ ++ switch (rtd->codec_dai->dai_runtime.pcmfmt) { ++ case SNDRV_PCM_FMTBIT_S16_LE: ++ break; ++ case SNDRV_PCM_FMTBIT_S20_3LE: ++ iface |= 0x0004; ++ break; ++ case SNDRV_PCM_FMTBIT_S24_LE: ++ iface |= 0x0008; ++ break; ++ case SNDRV_PCM_FMTBIT_S32_LE: ++ iface |= 0x000c; ++ break; ++ } ++ ++ /* clock inversion */ ++ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_INV_MASK) { ++ case SND_SOC_DAIFMT_NB_NF: ++ break; ++ case SND_SOC_DAIFMT_IB_IF: ++ iface |= 0x0090; ++ break; ++ case SND_SOC_DAIFMT_IB_NF: ++ iface |= 0x0080; ++ break; ++ case SND_SOC_DAIFMT_NB_IF: ++ iface |= 0x0010; ++ break; ++ } ++ ++ /* set bclk divisor rate */ ++ switch (bfs) { ++ case 1: ++ break; ++ case 4: ++ srate |= (0x1 << 7); ++ break; ++ case 8: ++ srate |= (0x2 << 7); ++ break; ++ case 16: ++ srate |= (0x3 << 7); ++ break; ++ } ++ ++ /* set iface & srate */ ++ wm8971_write(codec, WM8971_AUDIO, iface); ++ wm8971_write(codec, WM8971_SRATE, srate | ++ (coeff_div[i].sr << 1) | coeff_div[i].usb); ++ return 0; ++} ++ ++static int wm8971_mute(struct snd_soc_codec *codec, ++ struct snd_soc_codec_dai *dai, int mute) ++{ ++ u16 mute_reg = wm8971_read_reg_cache(codec, WM8971_ADCDAC) & 0xfff7; ++ if (mute) ++ wm8971_write(codec, WM8971_ADCDAC, mute_reg | 0x8); ++ else ++ wm8971_write(codec, WM8971_ADCDAC, mute_reg); ++ return 0; ++} ++ ++static int wm8971_dapm_event(struct snd_soc_codec *codec, int event) ++{ ++ u16 pwr_reg = wm8971_read_reg_cache(codec, WM8971_PWR1) & 0xfe3e; ++ ++ switch (event) { ++ case SNDRV_CTL_POWER_D0: /* full On */ ++ /* set vmid to 50k and unmute dac */ ++ wm8971_write(codec, WM8971_PWR1, pwr_reg | 0x00c1); ++ break; ++ case SNDRV_CTL_POWER_D1: /* partial On */ ++ case SNDRV_CTL_POWER_D2: /* partial On */ ++ /* set vmid to 5k for quick power up */ ++ wm8971_write(codec, WM8971_PWR1, pwr_reg | 0x01c0); ++ break; ++ case SNDRV_CTL_POWER_D3hot: /* Off, with power */ ++ /* mute dac and set vmid to 500k, enable VREF */ ++ wm8971_write(codec, WM8971_PWR1, pwr_reg | 0x0140); ++ break; ++ case SNDRV_CTL_POWER_D3cold: /* Off, without power */ ++ wm8971_write(codec, WM8971_PWR1, 0x0001); ++ break; ++ } ++ codec->dapm_state = event; ++ return 0; ++} ++ ++struct snd_soc_codec_dai wm8971_dai = { ++ .name = "WM8971", ++ .playback = { ++ .stream_name = "Playback", ++ .channels_min = 1, ++ .channels_max = 2, ++ }, ++ .capture = { ++ .stream_name = "Capture", ++ .channels_min = 1, ++ .channels_max = 2, ++ }, ++ .config_sysclk = wm8971_config_sysclk, ++ .digital_mute = wm8971_mute, ++ .ops = { ++ .prepare = wm8971_pcm_prepare, ++ }, ++ .caps = { ++ .num_modes = ARRAY_SIZE(wm8971_modes), ++ .mode = wm8971_modes, ++ }, ++}; ++EXPORT_SYMBOL_GPL(wm8971_dai); ++ ++static void wm8971_work(void *data) ++{ ++ struct snd_soc_codec *codec = (struct snd_soc_codec *)data; ++ wm8971_dapm_event(codec, codec->dapm_state); ++} ++ ++static int wm8971_suspend(struct platform_device *pdev, pm_message_t state) ++{ ++ struct snd_soc_device *socdev = platform_get_drvdata(pdev); ++ struct snd_soc_codec *codec = socdev->codec; ++ ++ wm8971_dapm_event(codec, SNDRV_CTL_POWER_D3cold); ++ return 0; ++} ++ ++static int wm8971_resume(struct platform_device *pdev) ++{ ++ struct snd_soc_device *socdev = platform_get_drvdata(pdev); ++ struct snd_soc_codec *codec = socdev->codec; ++ int i; ++ u8 data[2]; ++ u16 *cache = codec->reg_cache; ++ ++ /* Sync reg_cache with the hardware */ ++ for (i = 0; i < ARRAY_SIZE(wm8971_reg); i++) { ++ if (i + 1 == WM8971_RESET) ++ continue; ++ data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); ++ data[1] = cache[i] & 0x00ff; ++ codec->hw_write(codec->control_data, data, 2); ++ } ++ ++ wm8971_dapm_event(codec, SNDRV_CTL_POWER_D3hot); ++ ++ /* charge wm8971 caps */ ++ if (codec->suspend_dapm_state == SNDRV_CTL_POWER_D0) { ++ wm8971_dapm_event(codec, SNDRV_CTL_POWER_D2); ++ codec->dapm_state = SNDRV_CTL_POWER_D0; ++ queue_delayed_work(wm8971_workq, &wm8971_dapm_work, ++ msecs_to_jiffies(1000)); ++ } ++ ++ return 0; ++} ++ ++static int wm8971_init(struct snd_soc_device *socdev) ++{ ++ struct snd_soc_codec *codec = socdev->codec; ++ int reg, ret = 0; ++ ++ codec->name = "WM8971"; ++ codec->owner = THIS_MODULE; ++ codec->read = wm8971_read_reg_cache; ++ codec->write = wm8971_write; ++ codec->dapm_event = wm8971_dapm_event; ++ codec->dai = &wm8971_dai; ++ codec->reg_cache_size = ARRAY_SIZE(wm8971_reg); ++ codec->num_dai = 1; ++ codec->reg_cache = ++ kzalloc(sizeof(u16) * ARRAY_SIZE(wm8971_reg), GFP_KERNEL); ++ if (codec->reg_cache == NULL) ++ return -ENOMEM; ++ memcpy(codec->reg_cache, wm8971_reg, ++ sizeof(u16) * ARRAY_SIZE(wm8971_reg)); ++ codec->reg_cache_size = sizeof(u16) * ARRAY_SIZE(wm8971_reg); ++ ++ wm8971_reset(codec); ++ ++ /* register pcms */ ++ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); ++ if (ret < 0) { ++ kfree(codec->reg_cache); ++ return ret; ++ } ++ ++ /* charge output caps */ ++ wm8971_dapm_event(codec, SNDRV_CTL_POWER_D2); ++ codec->dapm_state = SNDRV_CTL_POWER_D3hot; ++ queue_delayed_work(wm8971_workq, &wm8971_dapm_work, ++ msecs_to_jiffies(1000)); ++ ++ /* set the update bits */ ++ reg = wm8971_read_reg_cache(codec, WM8971_LDAC); ++ wm8971_write(codec, WM8971_LDAC, reg | 0x0100); ++ reg = wm8971_read_reg_cache(codec, WM8971_RDAC); ++ wm8971_write(codec, WM8971_RDAC, reg | 0x0100); ++ ++ reg = wm8971_read_reg_cache(codec, WM8971_LOUT1V); ++ wm8971_write(codec, WM8971_LOUT1V, reg | 0x0100); ++ reg = wm8971_read_reg_cache(codec, WM8971_ROUT1V); ++ wm8971_write(codec, WM8971_ROUT1V, reg | 0x0100); ++ ++ reg = wm8971_read_reg_cache(codec, WM8971_LOUT2V); ++ wm8971_write(codec, WM8971_LOUT2V, reg | 0x0100); ++ reg = wm8971_read_reg_cache(codec, WM8971_ROUT2V); ++ wm8971_write(codec, WM8971_ROUT2V, reg | 0x0100); ++ ++ reg = wm8971_read_reg_cache(codec, WM8971_LINVOL); ++ wm8971_write(codec, WM8971_LINVOL, reg | 0x0100); ++ reg = wm8971_read_reg_cache(codec, WM8971_RINVOL); ++ wm8971_write(codec, WM8971_RINVOL, reg | 0x0100); ++ ++ wm8971_add_controls(codec); ++ wm8971_add_widgets(codec); ++ ret = snd_soc_register_card(socdev); ++ if (ret < 0) { ++ snd_soc_free_pcms(socdev); ++ snd_soc_dapm_free(socdev); ++ } ++ ++ return ret; ++} ++ ++/* If the i2c layer weren't so broken, we could pass this kind of data ++ around */ ++static struct snd_soc_device *wm8971_socdev; ++ ++#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) ++ ++/* ++ * WM8731 2 wire address is determined by GPIO5 ++ * state during powerup. ++ * low = 0x1a ++ * high = 0x1b ++ */ ++#define I2C_DRIVERID_WM8971 0xfefe /* liam - need a proper id */ ++ ++static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END }; ++ ++/* Magic definition of all other variables and things */ ++I2C_CLIENT_INSMOD; ++ ++static struct i2c_driver wm8971_i2c_driver; ++static struct i2c_client client_template; ++ ++static int wm8971_codec_probe(struct i2c_adapter *adap, int addr, int kind) ++{ ++ struct snd_soc_device *socdev = wm8971_socdev; ++ struct wm8971_setup_data *setup = socdev->codec_data; ++ struct snd_soc_codec *codec = socdev->codec; ++ struct i2c_client *i2c; ++ int ret; ++ ++ if (addr != setup->i2c_address) ++ return -ENODEV; ++ ++ client_template.adapter = adap; ++ client_template.addr = addr; ++ ++ i2c = kzalloc(sizeof(struct i2c_client), GFP_KERNEL); ++ if (i2c == NULL) { ++ kfree(codec); ++ return -ENOMEM; ++ } ++ memcpy(i2c, &client_template, sizeof(struct i2c_client)); ++ ++ i2c_set_clientdata(i2c, codec); ++ ++ codec->control_data = i2c; ++ ++ ret = i2c_attach_client(i2c); ++ if (ret < 0) { ++ err("failed to attach codec at addr %x\n", addr); ++ goto err; ++ } ++ ++ ret = wm8971_init(socdev); ++ if (ret < 0) { ++ err("failed to initialise WM8971\n"); ++ goto err; ++ } ++ return ret; ++ ++err: ++ kfree(codec); ++ kfree(i2c); ++ return ret; ++} ++ ++static int wm8971_i2c_detach(struct i2c_client *client) ++{ ++ struct snd_soc_codec* codec = i2c_get_clientdata(client); ++ i2c_detach_client(client); ++ kfree(codec->reg_cache); ++ kfree(client); ++ return 0; ++} ++ ++static int wm8971_i2c_attach(struct i2c_adapter *adap) ++{ ++ return i2c_probe(adap, &addr_data, wm8971_codec_probe); ++} ++ ++/* corgi i2c codec control layer */ ++static struct i2c_driver wm8971_i2c_driver = { ++ .driver = { ++ .name = "WM8971 I2C Codec", ++ .owner = THIS_MODULE, ++ }, ++ .id = I2C_DRIVERID_WM8971, ++ .attach_adapter = wm8971_i2c_attach, ++ .detach_client = wm8971_i2c_detach, ++ .command = NULL, ++}; ++ ++static struct i2c_client client_template = { ++ .name = "WM8971", ++ .driver = &wm8971_i2c_driver, ++}; ++#endif ++ ++static int wm8971_probe(struct platform_device *pdev) ++{ ++ struct snd_soc_device *socdev = platform_get_drvdata(pdev); ++ struct wm8971_setup_data *setup; ++ struct snd_soc_codec *codec; ++ int ret = 0; ++ ++ info("WM8971 Audio Codec %s", WM8971_VERSION); ++ ++ setup = socdev->codec_data; ++ codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); ++ if (codec == NULL) ++ return -ENOMEM; ++ ++ socdev->codec = codec; ++ mutex_init(&codec->mutex); ++ INIT_LIST_HEAD(&codec->dapm_widgets); ++ INIT_LIST_HEAD(&codec->dapm_paths); ++ wm8971_socdev = socdev; ++ ++ INIT_WORK(&wm8971_dapm_work, wm8971_work, codec); ++ wm8971_workq = create_workqueue("wm8971"); ++ if (wm8971_workq == NULL) { ++ kfree(codec); ++ return -ENOMEM; ++ } ++#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) ++ if (setup->i2c_address) { ++ normal_i2c[0] = setup->i2c_address; ++ codec->hw_write = (hw_write_t)i2c_master_send; ++ ret = i2c_add_driver(&wm8971_i2c_driver); ++ if (ret != 0) ++ printk(KERN_ERR "can't add i2c driver"); ++ } ++#else ++ /* Add other interfaces here */ ++#endif ++ ++ return ret; ++} ++ ++/* power down chip */ ++static int wm8971_remove(struct platform_device *pdev) ++{ ++ struct snd_soc_device *socdev = platform_get_drvdata(pdev); ++ struct snd_soc_codec *codec = socdev->codec; ++ ++ if (codec->control_data) ++ wm8971_dapm_event(codec, SNDRV_CTL_POWER_D3cold); ++ if (wm8971_workq) ++ destroy_workqueue(wm8971_workq); ++ snd_soc_free_pcms(socdev); ++ snd_soc_dapm_free(socdev); ++#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) ++ i2c_del_driver(&wm8971_i2c_driver); ++#endif ++ kfree(codec); ++ ++ return 0; ++} ++ ++struct snd_soc_codec_device soc_codec_dev_wm8971 = { ++ .probe = wm8971_probe, ++ .remove = wm8971_remove, ++ .suspend = wm8971_suspend, ++ .resume = wm8971_resume, ++}; ++ ++EXPORT_SYMBOL_GPL(soc_codec_dev_wm8971); ++ ++MODULE_DESCRIPTION("ASoC WM8971 driver"); ++MODULE_AUTHOR("Lab126"); ++MODULE_LICENSE("GPL"); +Index: linux-2.6-pxa-new/sound/soc/codecs/wm8971.h +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/sound/soc/codecs/wm8971.h +@@ -0,0 +1,61 @@ ++/* ++ * wm8971.h -- audio driver for WM8971 ++ * ++ * Copyright 2005 Lab126, Inc. ++ * ++ * Author: Kenneth Kiraly <kiraly@lab126.com> ++ * ++ * This program is free software; you can redistribute it and/or modify it ++ * under the terms of the GNU General Public License as published by the ++ * Free Software Foundation; either version 2 of the License, or (at your ++ * option) any later version. ++ * ++ */ ++ ++#ifndef _WM8971_H ++#define _WM8971_H ++ ++#define WM8971_LINVOL 0x00 ++#define WM8971_RINVOL 0x01 ++#define WM8971_LOUT1V 0x02 ++#define WM8971_ROUT1V 0x03 ++#define WM8971_ADCDAC 0x05 ++#define WM8971_AUDIO 0x07 ++#define WM8971_SRATE 0x08 ++#define WM8971_LDAC 0x0a ++#define WM8971_RDAC 0x0b ++#define WM8971_BASS 0x0c ++#define WM8971_TREBLE 0x0d ++#define WM8971_RESET 0x0f ++#define WM8971_ALC1 0x11 ++#define WM8971_ALC2 0x12 ++#define WM8971_ALC3 0x13 ++#define WM8971_NGATE 0x14 ++#define WM8971_LADC 0x15 ++#define WM8971_RADC 0x16 ++#define WM8971_ADCTL1 0x17 ++#define WM8971_ADCTL2 0x18 ++#define WM8971_PWR1 0x19 ++#define WM8971_PWR2 0x1a ++#define WM8971_ADCTL3 0x1b ++#define WM8971_ADCIN 0x1f ++#define WM8971_LADCIN 0x20 ++#define WM8971_RADCIN 0x21 ++#define WM8971_LOUTM1 0x22 ++#define WM8971_LOUTM2 0x23 ++#define WM8971_ROUTM1 0x24 ++#define WM8971_ROUTM2 0x25 ++#define WM8971_MOUTM1 0x26 ++#define WM8971_MOUTM2 0x27 ++#define WM8971_LOUT2V 0x28 ++#define WM8971_ROUT2V 0x29 ++#define WM8971_MOUTV 0x2A ++ ++struct wm8971_setup_data { ++ unsigned short i2c_address; ++}; ++ ++extern struct snd_soc_codec_dai wm8971_dai; ++extern struct snd_soc_codec_device soc_codec_dev_wm8971; ++ ++#endif +Index: linux-2.6-pxa-new/sound/soc/codecs/wm8974.c +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/sound/soc/codecs/wm8974.c +@@ -0,0 +1,935 @@ ++/* ++ * wm8974.c -- WM8974 ALSA Soc Audio driver ++ * ++ * Copyright 2006 Wolfson Microelectronics PLC. ++ * ++ * Author: Liam Girdwood <liam.girdwood@wolfsonmicro.com> ++ * ++ * This program is free software; you can redistribute it and/or modify ++ * it under the terms of the GNU General Public License version 2 as ++ * published by the Free Software Foundation. ++ */ ++ ++#include <linux/module.h> ++#include <linux/moduleparam.h> ++#include <linux/version.h> ++#include <linux/kernel.h> ++#include <linux/init.h> ++#include <linux/delay.h> ++#include <linux/pm.h> ++#include <linux/i2c.h> ++#include <linux/platform_device.h> ++#include <sound/driver.h> ++#include <sound/core.h> ++#include <sound/pcm.h> ++#include <sound/pcm_params.h> ++#include <sound/soc.h> ++#include <sound/soc-dapm.h> ++#include <sound/initval.h> ++ ++#include "wm8974.h" ++ ++#define AUDIO_NAME "wm8974" ++#define WM8974_VERSION "0.5" ++ ++/* ++ * Debug ++ */ ++ ++#define WM8974_DEBUG 0 ++ ++#ifdef WM8974_DEBUG ++#define dbg(format, arg...) \ ++ printk(KERN_DEBUG AUDIO_NAME ": " format "\n" , ## arg) ++#else ++#define dbg(format, arg...) do {} while (0) ++#endif ++#define err(format, arg...) \ ++ printk(KERN_ERR AUDIO_NAME ": " format "\n" , ## arg) ++#define info(format, arg...) \ ++ printk(KERN_INFO AUDIO_NAME ": " format "\n" , ## arg) ++#define warn(format, arg...) \ ++ printk(KERN_WARNING AUDIO_NAME ": " format "\n" , ## arg) ++ ++struct snd_soc_codec_device soc_codec_dev_wm8974; ++ ++/* ++ * wm8974 register cache ++ * We can't read the WM8974 register space when we are ++ * using 2 wire for device control, so we cache them instead. ++ */ ++static const u16 wm8974_reg[WM8974_CACHEREGNUM] = { ++ 0x0000, 0x0000, 0x0000, 0x0000, ++ 0x0050, 0x0000, 0x0140, 0x0000, ++ 0x0000, 0x0000, 0x0000, 0x00ff, ++ 0x0000, 0x0000, 0x0100, 0x00ff, ++ 0x0000, 0x0000, 0x012c, 0x002c, ++ 0x002c, 0x002c, 0x002c, 0x0000, ++ 0x0032, 0x0000, 0x0000, 0x0000, ++ 0x0000, 0x0000, 0x0000, 0x0000, ++ 0x0038, 0x000b, 0x0032, 0x0000, ++ 0x0008, 0x000c, 0x0093, 0x00e9, ++ 0x0000, 0x0000, 0x0000, 0x0000, ++ 0x0003, 0x0010, 0x0000, 0x0000, ++ 0x0000, 0x0002, 0x0000, 0x0000, ++ 0x0000, 0x0000, 0x0039, 0x0000, ++ 0x0000, ++}; ++ ++#define WM8974_DAIFMT \ ++ (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_RIGHT_J | \ ++ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_NB_IF | SND_SOC_DAIFMT_IB_NF | \ ++ SND_SOC_DAIFMT_IB_IF) ++ ++#define WM8974_DIR \ ++ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE) ++ ++#define WM8974_RATES \ ++ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ ++ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ ++ SNDRV_PCM_RATE_48000) ++ ++#define WM8794_BCLK \ ++ (SND_SOC_FSBD(1) | SND_SOC_FSBD(2) | SND_SOC_FSBD(4) | SND_SOC_FSBD(8) |\ ++ SND_SOC_FSBD(16) | SND_SOC_FSBD(32)) ++ ++#define WM8794_HIFI_BITS \ ++ (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ ++ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) ++ ++static struct snd_soc_dai_mode wm8974_modes[] = { ++ /* codec frame and clock master modes */ ++ { ++ .fmt = WM8974_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8794_HIFI_BITS, ++ .pcmrate = WM8974_RATES, ++ .pcmdir = WM8974_DIR, ++ .fs = 256, ++ .bfs = WM8794_BCLK, ++ }, ++ ++ /* codec frame and clock slave modes */ ++ { ++ .fmt = WM8974_DAIFMT | SND_SOC_DAIFMT_CBS_CFS, ++ .pcmfmt = WM8794_HIFI_BITS, ++ .pcmrate = WM8974_RATES, ++ .pcmdir = WM8974_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = SND_SOC_FS_ALL, ++ .bfs = SND_SOC_FSB_ALL, ++ }, ++}; ++ ++/* ++ * read wm8974 register cache ++ */ ++static inline unsigned int wm8974_read_reg_cache(struct snd_soc_codec * codec, ++ unsigned int reg) ++{ ++ u16 *cache = codec->reg_cache; ++ if (reg == WM8974_RESET) ++ return 0; ++ if (reg >= WM8974_CACHEREGNUM) ++ return -1; ++ return cache[reg]; ++} ++ ++/* ++ * write wm8974 register cache ++ */ ++static inline void wm8974_write_reg_cache(struct snd_soc_codec *codec, ++ u16 reg, unsigned int value) ++{ ++ u16 *cache = codec->reg_cache; ++ if (reg >= WM8974_CACHEREGNUM) ++ return; ++ cache[reg] = value; ++} ++ ++/* ++ * write to the WM8974 register space ++ */ ++static int wm8974_write(struct snd_soc_codec *codec, unsigned int reg, ++ unsigned int value) ++{ ++ u8 data[2]; ++ ++ /* data is ++ * D15..D9 WM8974 register offset ++ * D8...D0 register data ++ */ ++ data[0] = (reg << 1) | ((value >> 8) & 0x0001); ++ data[1] = value & 0x00ff; ++ ++ wm8974_write_reg_cache (codec, reg, value); ++ if (codec->hw_write(codec->control_data, data, 2) == 2) ++ return 0; ++ else ++ return -EIO; ++} ++ ++#define wm8974_reset(c) wm8974_write(c, WM8974_RESET, 0) ++ ++static const char *wm8974_companding[] = {"Off", "NC", "u-law", "A-law" }; ++static const char *wm8974_deemp[] = {"None", "32kHz", "44.1kHz", "48kHz" }; ++static const char *wm8974_eqmode[] = {"Capture", "Playback" }; ++static const char *wm8974_bw[] = {"Narrow", "Wide" }; ++static const char *wm8974_eq1[] = {"80Hz", "105Hz", "135Hz", "175Hz" }; ++static const char *wm8974_eq2[] = {"230Hz", "300Hz", "385Hz", "500Hz" }; ++static const char *wm8974_eq3[] = {"650Hz", "850Hz", "1.1kHz", "1.4kHz" }; ++static const char *wm8974_eq4[] = {"1.8kHz", "2.4kHz", "3.2kHz", "4.1kHz" }; ++static const char *wm8974_eq5[] = {"5.3kHz", "6.9kHz", "9kHz", "11.7kHz" }; ++static const char *wm8974_alc[] = {"ALC", "Limiter" }; ++ ++static const struct soc_enum wm8974_enum[] = { ++ SOC_ENUM_SINGLE(WM8974_COMP, 1, 4, wm8974_companding), /* adc */ ++ SOC_ENUM_SINGLE(WM8974_COMP, 3, 4, wm8974_companding), /* dac */ ++ SOC_ENUM_SINGLE(WM8974_DAC, 4, 4, wm8974_deemp), ++ SOC_ENUM_SINGLE(WM8974_EQ1, 8, 2, wm8974_eqmode), ++ ++ SOC_ENUM_SINGLE(WM8974_EQ1, 5, 4, wm8974_eq1), ++ SOC_ENUM_SINGLE(WM8974_EQ2, 8, 2, wm8974_bw), ++ SOC_ENUM_SINGLE(WM8974_EQ2, 5, 4, wm8974_eq2), ++ SOC_ENUM_SINGLE(WM8974_EQ3, 8, 2, wm8974_bw), ++ ++ SOC_ENUM_SINGLE(WM8974_EQ3, 5, 4, wm8974_eq3), ++ SOC_ENUM_SINGLE(WM8974_EQ4, 8, 2, wm8974_bw), ++ SOC_ENUM_SINGLE(WM8974_EQ4, 5, 4, wm8974_eq4), ++ SOC_ENUM_SINGLE(WM8974_EQ5, 8, 2, wm8974_bw), ++ ++ SOC_ENUM_SINGLE(WM8974_EQ5, 5, 4, wm8974_eq5), ++ SOC_ENUM_SINGLE(WM8974_ALC3, 8, 2, wm8974_alc), ++}; ++ ++static const struct snd_kcontrol_new wm8974_snd_controls[] = { ++ ++SOC_SINGLE("Digital Loopback Switch", WM8974_COMP, 0, 1, 0), ++ ++SOC_ENUM("DAC Companding", wm8974_enum[1]), ++SOC_ENUM("ADC Companding", wm8974_enum[0]), ++ ++SOC_ENUM("Playback De-emphasis", wm8974_enum[2]), ++SOC_SINGLE("DAC Inversion Switch", WM8974_DAC, 0, 1, 0), ++ ++SOC_SINGLE("PCM Volume", WM8974_DACVOL, 0, 127, 0), ++ ++SOC_SINGLE("High Pass Filter Switch", WM8974_ADC, 8, 1, 0), ++SOC_SINGLE("High Pass Cut Off", WM8974_ADC, 4, 7, 0), ++SOC_SINGLE("ADC Inversion Switch", WM8974_COMP, 0, 1, 0), ++ ++SOC_SINGLE("Capture Volume", WM8974_ADCVOL, 0, 127, 0), ++ ++SOC_ENUM("Equaliser Function", wm8974_enum[3]), ++SOC_ENUM("EQ1 Cut Off", wm8974_enum[4]), ++SOC_SINGLE("EQ1 Volume", WM8974_EQ1, 0, 31, 1), ++ ++SOC_ENUM("Equaliser EQ2 Bandwith", wm8974_enum[5]), ++SOC_ENUM("EQ2 Cut Off", wm8974_enum[6]), ++SOC_SINGLE("EQ2 Volume", WM8974_EQ2, 0, 31, 1), ++ ++SOC_ENUM("Equaliser EQ3 Bandwith", wm8974_enum[7]), ++SOC_ENUM("EQ3 Cut Off", wm8974_enum[8]), ++SOC_SINGLE("EQ3 Volume", WM8974_EQ3, 0, 31, 1), ++ ++SOC_ENUM("Equaliser EQ4 Bandwith", wm8974_enum[9]), ++SOC_ENUM("EQ4 Cut Off", wm8974_enum[10]), ++SOC_SINGLE("EQ4 Volume", WM8974_EQ4, 0, 31, 1), ++ ++SOC_ENUM("Equaliser EQ5 Bandwith", wm8974_enum[11]), ++SOC_ENUM("EQ5 Cut Off", wm8974_enum[12]), ++SOC_SINGLE("EQ5 Volume", WM8974_EQ5, 0, 31, 1), ++ ++SOC_SINGLE("DAC Playback Limiter Switch", WM8974_DACLIM1, 8, 1, 0), ++SOC_SINGLE("DAC Playback Limiter Decay", WM8974_DACLIM1, 4, 15, 0), ++SOC_SINGLE("DAC Playback Limiter Attack", WM8974_DACLIM1, 0, 15, 0), ++ ++SOC_SINGLE("DAC Playback Limiter Threshold", WM8974_DACLIM2, 4, 7, 0), ++SOC_SINGLE("DAC Playback Limiter Boost", WM8974_DACLIM2, 0, 15, 0), ++ ++SOC_SINGLE("ALC Enable Switch", WM8974_ALC1, 8, 1, 0), ++SOC_SINGLE("ALC Capture Max Gain", WM8974_ALC1, 3, 7, 0), ++SOC_SINGLE("ALC Capture Min Gain", WM8974_ALC1, 0, 7, 0), ++ ++SOC_SINGLE("ALC Capture ZC Switch", WM8974_ALC2, 8, 1, 0), ++SOC_SINGLE("ALC Capture Hold", WM8974_ALC2, 4, 7, 0), ++SOC_SINGLE("ALC Capture Target", WM8974_ALC2, 0, 15, 0), ++ ++SOC_ENUM("ALC Capture Mode", wm8974_enum[13]), ++SOC_SINGLE("ALC Capture Decay", WM8974_ALC3, 4, 15, 0), ++SOC_SINGLE("ALC Capture Attack", WM8974_ALC3, 0, 15, 0), ++ ++SOC_SINGLE("ALC Capture Noise Gate Switch", WM8974_NGATE, 3, 1, 0), ++SOC_SINGLE("ALC Capture Noise Gate Threshold", WM8974_NGATE, 0, 7, 0), ++ ++SOC_SINGLE("Capture PGA ZC Switch", WM8974_INPPGA, 7, 1, 0), ++SOC_SINGLE("Capture PGA Volume", WM8974_INPPGA, 0, 63, 0), ++ ++SOC_SINGLE("Speaker Playback ZC Switch", WM8974_SPKVOL, 7, 1, 0), ++SOC_SINGLE("Speaker Playback Switch", WM8974_SPKVOL, 6, 1, 1), ++SOC_SINGLE("Speaker Playback Volume", WM8974_SPKVOL, 0, 63, 0), ++ ++SOC_SINGLE("Capture Boost(+20dB)", WM8974_ADCBOOST, 8, 1, 0), ++SOC_SINGLE("Mono Playback Switch", WM8974_MONOMIX, 6, 1, 0), ++}; ++ ++/* add non dapm controls */ ++static int wm8974_add_controls(struct snd_soc_codec *codec) ++{ ++ int err, i; ++ ++ for (i = 0; i < ARRAY_SIZE(wm8974_snd_controls); i++) { ++ err = snd_ctl_add(codec->card, ++ snd_soc_cnew(&wm8974_snd_controls[i],codec, NULL)); ++ if (err < 0) ++ return err; ++ } ++ ++ return 0; ++} ++ ++/* Speaker Output Mixer */ ++static const struct snd_kcontrol_new wm8974_speaker_mixer_controls[] = { ++SOC_DAPM_SINGLE("Line Bypass Switch", WM8974_SPKMIX, 1, 1, 0), ++SOC_DAPM_SINGLE("Aux Playback Switch", WM8974_SPKMIX, 5, 1, 0), ++SOC_DAPM_SINGLE("PCM Playback Switch", WM8974_SPKMIX, 0, 1, 1), ++}; ++ ++/* Mono Output Mixer */ ++static const struct snd_kcontrol_new wm8974_mono_mixer_controls[] = { ++SOC_DAPM_SINGLE("Line Bypass Switch", WM8974_MONOMIX, 1, 1, 0), ++SOC_DAPM_SINGLE("Aux Playback Switch", WM8974_MONOMIX, 2, 1, 0), ++SOC_DAPM_SINGLE("PCM Playback Switch", WM8974_MONOMIX, 0, 1, 1), ++}; ++ ++/* AUX Input boost vol */ ++static const struct snd_kcontrol_new wm8974_aux_boost_controls = ++SOC_DAPM_SINGLE("Aux Volume", WM8974_ADCBOOST, 0, 7, 0); ++ ++/* Mic Input boost vol */ ++static const struct snd_kcontrol_new wm8974_mic_boost_controls = ++SOC_DAPM_SINGLE("Mic Volume", WM8974_ADCBOOST, 4, 7, 0); ++ ++/* Capture boost switch */ ++static const struct snd_kcontrol_new wm8974_capture_boost_controls = ++SOC_DAPM_SINGLE("Capture Boost Switch", WM8974_INPPGA, 6, 1, 0); ++ ++/* Aux In to PGA */ ++static const struct snd_kcontrol_new wm8974_aux_capture_boost_controls = ++SOC_DAPM_SINGLE("Aux Capture Boost Switch", WM8974_INPPGA, 2, 1, 0); ++ ++/* Mic P In to PGA */ ++static const struct snd_kcontrol_new wm8974_micp_capture_boost_controls = ++SOC_DAPM_SINGLE("Mic P Capture Boost Switch", WM8974_INPPGA, 0, 1, 0); ++ ++/* Mic N In to PGA */ ++static const struct snd_kcontrol_new wm8974_micn_capture_boost_controls = ++SOC_DAPM_SINGLE("Mic N Capture Boost Switch", WM8974_INPPGA, 1, 1, 0); ++ ++static const struct snd_soc_dapm_widget wm8974_dapm_widgets[] = { ++SND_SOC_DAPM_MIXER("Speaker Mixer", WM8974_POWER3, 2, 0, ++ &wm8974_speaker_mixer_controls[0], ++ ARRAY_SIZE(wm8974_speaker_mixer_controls)), ++SND_SOC_DAPM_MIXER("Mono Mixer", WM8974_POWER3, 3, 0, ++ &wm8974_mono_mixer_controls[0], ++ ARRAY_SIZE(wm8974_mono_mixer_controls)), ++SND_SOC_DAPM_DAC("DAC", "HiFi Playback", WM8974_POWER3, 0, 0), ++SND_SOC_DAPM_ADC("ADC", "HiFi Capture", WM8974_POWER3, 0, 0), ++SND_SOC_DAPM_PGA("Aux Input", WM8974_POWER1, 6, 0, NULL, 0), ++SND_SOC_DAPM_PGA("SpkN Out", WM8974_POWER3, 5, 0, NULL, 0), ++SND_SOC_DAPM_PGA("SpkP Out", WM8974_POWER3, 6, 0, NULL, 0), ++SND_SOC_DAPM_PGA("Mono Out", WM8974_POWER3, 7, 0, NULL, 0), ++SND_SOC_DAPM_PGA("Mic PGA", WM8974_POWER2, 2, 0, NULL, 0), ++ ++SND_SOC_DAPM_PGA("Aux Boost", SND_SOC_NOPM, 0, 0, ++ &wm8974_aux_boost_controls, 1), ++SND_SOC_DAPM_PGA("Mic Boost", SND_SOC_NOPM, 0, 0, ++ &wm8974_mic_boost_controls, 1), ++SND_SOC_DAPM_SWITCH("Capture Boost", SND_SOC_NOPM, 0, 0, ++ &wm8974_capture_boost_controls), ++ ++SND_SOC_DAPM_MIXER("Boost Mixer", WM8974_POWER2, 4, 0, NULL, 0), ++ ++SND_SOC_DAPM_MICBIAS("Mic Bias", WM8974_POWER1, 4, 0), ++ ++SND_SOC_DAPM_INPUT("MICN"), ++SND_SOC_DAPM_INPUT("MICP"), ++SND_SOC_DAPM_INPUT("AUX"), ++SND_SOC_DAPM_OUTPUT("MONOOUT"), ++SND_SOC_DAPM_OUTPUT("SPKOUTP"), ++SND_SOC_DAPM_OUTPUT("SPKOUTN"), ++}; ++ ++static const char *audio_map[][3] = { ++ /* Mono output mixer */ ++ {"Mono Mixer", "PCM Playback Switch", "DAC"}, ++ {"Mono Mixer", "Aux Playback Switch", "Aux Input"}, ++ {"Mono Mixer", "Line Bypass Switch", "Boost Mixer"}, ++ ++ /* Speaker output mixer */ ++ {"Speaker Mixer", "PCM Playback Switch", "DAC"}, ++ {"Speaker Mixer", "Aux Playback Switch", "Aux Input"}, ++ {"Speaker Mixer", "Line Bypass Switch", "Boost Mixer"}, ++ ++ /* Outputs */ ++ {"Mono Out", NULL, "Mono Mixer"}, ++ {"MONOOUT", NULL, "Mono Out"}, ++ {"SpkN Out", NULL, "Speaker Mixer"}, ++ {"SpkP Out", NULL, "Speaker Mixer"}, ++ {"SPKOUTN", NULL, "SpkN Out"}, ++ {"SPKOUTP", NULL, "SpkP Out"}, ++ ++ /* Boost Mixer */ ++ {"Boost Mixer", NULL, "ADC"}, ++ {"Capture Boost Switch", "Aux Capture Boost Switch", "AUX"}, ++ {"Aux Boost", "Aux Volume", "Boost Mixer"}, ++ {"Capture Boost", "Capture Switch", "Boost Mixer"}, ++ {"Mic Boost", "Mic Volume", "Boost Mixer"}, ++ ++ /* Inputs */ ++ {"MICP", NULL, "Mic Boost"}, ++ {"MICN", NULL, "Mic PGA"}, ++ {"Mic PGA", NULL, "Capture Boost"}, ++ {"AUX", NULL, "Aux Input"}, ++ ++ /* terminator */ ++ {NULL, NULL, NULL}, ++}; ++ ++static int wm8974_add_widgets(struct snd_soc_codec *codec) ++{ ++ int i; ++ ++ for(i = 0; i < ARRAY_SIZE(wm8974_dapm_widgets); i++) { ++ snd_soc_dapm_new_control(codec, &wm8974_dapm_widgets[i]); ++ } ++ ++ /* set up audio path audio_mapnects */ ++ for(i = 0; audio_map[i][0] != NULL; i++) { ++ snd_soc_dapm_connect_input(codec, audio_map[i][0], ++ audio_map[i][1], audio_map[i][2]); ++ } ++ ++ snd_soc_dapm_new_widgets(codec); ++ return 0; ++} ++ ++struct pll_ { ++ unsigned int in_hz, out_hz; ++ unsigned int pre:4; /* prescale - 1 */ ++ unsigned int n:4; ++ unsigned int k; ++}; ++ ++struct pll_ pll[] = { ++ {12000000, 11289600, 0, 7, 0x86c220}, ++ {12000000, 12288000, 0, 8, 0x3126e8}, ++ {13000000, 11289600, 0, 6, 0xf28bd4}, ++ {13000000, 12288000, 0, 7, 0x8fd525}, ++ {12288000, 11289600, 0, 7, 0x59999a}, ++ {11289600, 12288000, 0, 8, 0x80dee9}, ++ /* liam - add more entries */ ++}; ++ ++static int set_pll(struct snd_soc_codec *codec, unsigned int in, ++ unsigned int out) ++{ ++ int i; ++ u16 reg; ++ ++ if(out == 0) { ++ reg = wm8974_read_reg_cache(codec, WM8974_POWER1); ++ wm8974_write(codec, WM8974_POWER1, reg & 0x1df); ++ return 0; ++ } ++ ++ for(i = 0; i < ARRAY_SIZE(pll); i++) { ++ if (in == pll[i].in_hz && out == pll[i].out_hz) { ++ wm8974_write(codec, WM8974_PLLN, (pll[i].pre << 4) | pll[i].n); ++ wm8974_write(codec, WM8974_PLLK1, pll[i].k >> 18); ++ wm8974_write(codec, WM8974_PLLK1, (pll[i].k >> 9) && 0x1ff); ++ wm8974_write(codec, WM8974_PLLK1, pll[i].k && 0x1ff); ++ reg = wm8974_read_reg_cache(codec, WM8974_POWER1); ++ wm8974_write(codec, WM8974_POWER1, reg | 0x020); ++ return 0; ++ } ++ } ++ return -EINVAL; ++} ++ ++/* mclk dividers * 2 */ ++static unsigned char mclk_div[] = {2, 3, 4, 6, 8, 12, 16, 24}; ++ ++/* we need 256FS to drive the DAC's and ADC's */ ++static unsigned int wm8974_config_sysclk(struct snd_soc_codec_dai *dai, ++ struct snd_soc_clock_info *info, unsigned int clk) ++{ ++ int i, j, best_clk = info->fs * info->rate; ++ ++ /* can we run at this clk without the PLL ? */ ++ for (i = 0; i < ARRAY_SIZE(mclk_div); i++) { ++ if ((best_clk >> 1) * mclk_div[i] == clk) { ++ dai->pll_in = 0; ++ dai->clk_div = mclk_div[i]; ++ dai->mclk = best_clk; ++ return dai->mclk; ++ } ++ } ++ ++ /* now check for PLL support */ ++ for (i = 0; i < ARRAY_SIZE(pll); i++) { ++ if (pll[i].in_hz == clk) { ++ for (j = 0; j < ARRAY_SIZE(mclk_div); j++) { ++ if (pll[i].out_hz == mclk_div[j] * (best_clk >> 1)) { ++ dai->pll_in = clk; ++ dai->pll_out = pll[i].out_hz; ++ dai->clk_div = mclk_div[j]; ++ dai->mclk = best_clk; ++ return dai->mclk; ++ } ++ } ++ } ++ } ++ ++ /* this clk is not supported */ ++ return 0; ++} ++ ++static int wm8974_pcm_prepare(struct snd_pcm_substream *substream) ++{ ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ struct snd_soc_device *socdev = rtd->socdev; ++ struct snd_soc_codec *codec = socdev->codec; ++ struct snd_soc_codec_dai *dai = rtd->codec_dai; ++ u16 iface = 0, bfs, clk = 0, adn; ++ int fs = 48000 << 7, i; ++ ++ bfs = SND_SOC_FSBD_REAL(rtd->codec_dai->dai_runtime.bfs); ++ switch (bfs) { ++ case 2: ++ clk |= 0x1 << 2; ++ break; ++ case 4: ++ clk |= 0x2 << 2; ++ break; ++ case 8: ++ clk |= 0x3 << 2; ++ break; ++ case 16: ++ clk |= 0x4 << 2; ++ break; ++ case 32: ++ clk |= 0x5 << 2; ++ break; ++ } ++ ++ /* set master/slave audio interface */ ++ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_CLOCK_MASK) { ++ case SND_SOC_DAIFMT_CBM_CFM: ++ clk |= 0x0001; ++ break; ++ case SND_SOC_DAIFMT_CBS_CFS: ++ break; ++ } ++ ++ /* interface format */ ++ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_FORMAT_MASK) { ++ case SND_SOC_DAIFMT_I2S: ++ iface |= 0x0010; ++ break; ++ case SND_SOC_DAIFMT_RIGHT_J: ++ break; ++ case SND_SOC_DAIFMT_LEFT_J: ++ iface |= 0x0008; ++ break; ++ case SND_SOC_DAIFMT_DSP_A: ++ iface |= 0x00018; ++ break; ++ } ++ ++ /* bit size */ ++ switch (rtd->codec_dai->dai_runtime.pcmfmt) { ++ case SNDRV_PCM_FMTBIT_S16_LE: ++ break; ++ case SNDRV_PCM_FMTBIT_S20_3LE: ++ iface |= 0x0020; ++ break; ++ case SNDRV_PCM_FMTBIT_S24_LE: ++ iface |= 0x0040; ++ break; ++ case SNDRV_PCM_FMTBIT_S32_LE: ++ iface |= 0x0060; ++ break; ++ } ++ ++ /* clock inversion */ ++ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_INV_MASK) { ++ case SND_SOC_DAIFMT_NB_NF: ++ break; ++ case SND_SOC_DAIFMT_IB_IF: ++ iface |= 0x0180; ++ break; ++ case SND_SOC_DAIFMT_IB_NF: ++ iface |= 0x0100; ++ break; ++ case SND_SOC_DAIFMT_NB_IF: ++ iface |= 0x0080; ++ break; ++ } ++ ++ /* filter coefficient */ ++ adn = wm8974_read_reg_cache(codec, WM8974_ADD) & 0x1f1; ++ switch (rtd->codec_dai->dai_runtime.pcmrate) { ++ case SNDRV_PCM_RATE_8000: ++ adn |= 0x5 << 1; ++ fs = 8000 << 7; ++ break; ++ case SNDRV_PCM_RATE_11025: ++ adn |= 0x4 << 1; ++ fs = 11025 << 7; ++ break; ++ case SNDRV_PCM_RATE_16000: ++ adn |= 0x3 << 1; ++ fs = 16000 << 7; ++ break; ++ case SNDRV_PCM_RATE_22050: ++ adn |= 0x2 << 1; ++ fs = 22050 << 7; ++ break; ++ case SNDRV_PCM_RATE_32000: ++ adn |= 0x1 << 1; ++ fs = 32000 << 7; ++ break; ++ case SNDRV_PCM_RATE_44100: ++ fs = 44100 << 7; ++ break; ++ } ++ ++ /* do we need to enable the PLL */ ++ if(dai->pll_in) ++ set_pll(codec, dai->pll_in, dai->pll_out); ++ ++ /* divide the clock to 256 fs */ ++ for(i = 0; i < ARRAY_SIZE(mclk_div); i++) { ++ if (dai->clk_div == mclk_div[i]) { ++ clk |= i << 5; ++ clk &= 0xff; ++ goto set; ++ } ++ } ++ ++set: ++ /* set iface */ ++ wm8974_write(codec, WM8974_IFACE, iface); ++ wm8974_write(codec, WM8974_CLOCK, clk); ++ ++ return 0; ++} ++ ++static int wm8974_hw_free(struct snd_pcm_substream *substream) ++{ ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ struct snd_soc_device *socdev = rtd->socdev; ++ struct snd_soc_codec *codec = socdev->codec; ++ set_pll(codec, 0, 0); ++ return 0; ++} ++ ++static int wm8974_mute(struct snd_soc_codec *codec, ++ struct snd_soc_codec_dai *dai, int mute) ++{ ++ u16 mute_reg = wm8974_read_reg_cache(codec, WM8974_DAC) & 0xffbf; ++ if(mute) ++ wm8974_write(codec, WM8974_DAC, mute_reg | 0x40); ++ else ++ wm8974_write(codec, WM8974_DAC, mute_reg); ++ return 0; ++} ++ ++/* liam need to make this lower power with dapm */ ++static int wm8974_dapm_event(struct snd_soc_codec *codec, int event) ++{ ++ ++ switch (event) { ++ case SNDRV_CTL_POWER_D0: /* full On */ ++ /* vref/mid, clk and osc on, dac unmute, active */ ++ wm8974_write(codec, WM8974_POWER1, 0x1ff); ++ wm8974_write(codec, WM8974_POWER2, 0x1ff); ++ wm8974_write(codec, WM8974_POWER3, 0x1ff); ++ break; ++ case SNDRV_CTL_POWER_D1: /* partial On */ ++ case SNDRV_CTL_POWER_D2: /* partial On */ ++ break; ++ case SNDRV_CTL_POWER_D3hot: /* Off, with power */ ++ /* everything off except vref/vmid, dac mute, inactive */ ++ ++ break; ++ case SNDRV_CTL_POWER_D3cold: /* Off, without power */ ++ /* everything off, dac mute, inactive */ ++ wm8974_write(codec, WM8974_POWER1, 0x0); ++ wm8974_write(codec, WM8974_POWER2, 0x0); ++ wm8974_write(codec, WM8974_POWER3, 0x0); ++ break; ++ } ++ codec->dapm_state = event; ++ return 0; ++} ++ ++struct snd_soc_codec_dai wm8974_dai = { ++ .name = "WM8974 HiFi", ++ .playback = { ++ .stream_name = "Playback", ++ .channels_min = 1, ++ .channels_max = 1, ++ }, ++ .capture = { ++ .stream_name = "Capture", ++ .channels_min = 1, ++ .channels_max = 1, ++ }, ++ .config_sysclk = wm8974_config_sysclk, ++ .digital_mute = wm8974_mute, ++ .ops = { ++ .prepare = wm8974_pcm_prepare, ++ .hw_free = wm8974_hw_free, ++ }, ++ .caps = { ++ .num_modes = ARRAY_SIZE(wm8974_modes), ++ .mode = wm8974_modes, ++ }, ++}; ++EXPORT_SYMBOL_GPL(wm8974_dai); ++ ++static int wm8974_suspend(struct platform_device *pdev, pm_message_t state) ++{ ++ struct snd_soc_device *socdev = platform_get_drvdata(pdev); ++ struct snd_soc_codec *codec = socdev->codec; ++ ++ wm8974_dapm_event(codec, SNDRV_CTL_POWER_D3cold); ++ return 0; ++} ++ ++static int wm8974_resume(struct platform_device *pdev) ++{ ++ struct snd_soc_device *socdev = platform_get_drvdata(pdev); ++ struct snd_soc_codec *codec = socdev->codec; ++ int i; ++ u8 data[2]; ++ u16 *cache = codec->reg_cache; ++ ++ /* Sync reg_cache with the hardware */ ++ for (i = 0; i < ARRAY_SIZE(wm8974_reg); i++) { ++ data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); ++ data[1] = cache[i] & 0x00ff; ++ codec->hw_write(codec->control_data, data, 2); ++ } ++ wm8974_dapm_event(codec, SNDRV_CTL_POWER_D3hot); ++ wm8974_dapm_event(codec, codec->suspend_dapm_state); ++ return 0; ++} ++ ++/* ++ * initialise the WM8974 driver ++ * register the mixer and dsp interfaces with the kernel ++ */ ++static int wm8974_init(struct snd_soc_device *socdev) ++{ ++ struct snd_soc_codec *codec = socdev->codec; ++ int ret = 0; ++ ++ codec->name = "WM8974"; ++ codec->owner = THIS_MODULE; ++ codec->read = wm8974_read_reg_cache; ++ codec->write = wm8974_write; ++ codec->dapm_event = wm8974_dapm_event; ++ codec->dai = &wm8974_dai; ++ codec->num_dai = 1; ++ codec->reg_cache_size = ARRAY_SIZE(wm8974_reg); ++ codec->reg_cache = ++ kzalloc(sizeof(u16) * ARRAY_SIZE(wm8974_reg), GFP_KERNEL); ++ if (codec->reg_cache == NULL) ++ return -ENOMEM; ++ memcpy(codec->reg_cache, wm8974_reg, ++ sizeof(u16) * ARRAY_SIZE(wm8974_reg)); ++ codec->reg_cache_size = sizeof(u16) * ARRAY_SIZE(wm8974_reg); ++ ++ wm8974_reset(codec); ++ ++ /* register pcms */ ++ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); ++ if(ret < 0) { ++ kfree(codec->reg_cache); ++ return ret; ++ } ++ ++ /* power on device */ ++ wm8974_dapm_event(codec, SNDRV_CTL_POWER_D3hot); ++ wm8974_add_controls(codec); ++ wm8974_add_widgets(codec); ++ ret = snd_soc_register_card(socdev); ++ if(ret < 0) { ++ snd_soc_free_pcms(socdev); ++ snd_soc_dapm_free(socdev); ++ } ++ ++ return ret; ++} ++ ++static struct snd_soc_device *wm8974_socdev; ++ ++#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) ++ ++/* ++ * WM8974 2 wire address is 0x1a ++ */ ++#define I2C_DRIVERID_WM8974 0xfefe /* liam - need a proper id */ ++ ++static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END }; ++ ++/* Magic definition of all other variables and things */ ++I2C_CLIENT_INSMOD; ++ ++static struct i2c_driver wm8974_i2c_driver; ++static struct i2c_client client_template; ++ ++/* If the i2c layer weren't so broken, we could pass this kind of data ++ around */ ++ ++static int wm8974_codec_probe(struct i2c_adapter *adap, int addr, int kind) ++{ ++ struct snd_soc_device *socdev = wm8974_socdev; ++ struct wm8974_setup_data *setup = socdev->codec_data; ++ struct snd_soc_codec *codec = socdev->codec; ++ struct i2c_client *i2c; ++ int ret; ++ ++ if (addr != setup->i2c_address) ++ return -ENODEV; ++ ++ client_template.adapter = adap; ++ client_template.addr = addr; ++ ++ i2c = kzalloc(sizeof(struct i2c_client), GFP_KERNEL); ++ if (i2c == NULL) { ++ kfree(codec); ++ return -ENOMEM; ++ } ++ memcpy(i2c, &client_template, sizeof(struct i2c_client)); ++ i2c_set_clientdata(i2c, codec); ++ codec->control_data = i2c; ++ ++ ret = i2c_attach_client(i2c); ++ if(ret < 0) { ++ err("failed to attach codec at addr %x\n", addr); ++ goto err; ++ } ++ ++ ret = wm8974_init(socdev); ++ if(ret < 0) { ++ err("failed to initialise WM8974\n"); ++ goto err; ++ } ++ return ret; ++ ++err: ++ kfree(codec); ++ kfree(i2c); ++ return ret; ++} ++ ++static int wm8974_i2c_detach(struct i2c_client *client) ++{ ++ struct snd_soc_codec *codec = i2c_get_clientdata(client); ++ i2c_detach_client(client); ++ kfree(codec->reg_cache); ++ kfree(client); ++ return 0; ++} ++ ++static int wm8974_i2c_attach(struct i2c_adapter *adap) ++{ ++ return i2c_probe(adap, &addr_data, wm8974_codec_probe); ++} ++ ++/* corgi i2c codec control layer */ ++static struct i2c_driver wm8974_i2c_driver = { ++ .driver = { ++ .name = "WM8974 I2C Codec", ++ .owner = THIS_MODULE, ++ }, ++ .id = I2C_DRIVERID_WM8974, ++ .attach_adapter = wm8974_i2c_attach, ++ .detach_client = wm8974_i2c_detach, ++ .command = NULL, ++}; ++ ++static struct i2c_client client_template = { ++ .name = "WM8974", ++ .driver = &wm8974_i2c_driver, ++}; ++#endif ++ ++static int wm8974_probe(struct platform_device *pdev) ++{ ++ struct snd_soc_device *socdev = platform_get_drvdata(pdev); ++ struct wm8974_setup_data *setup; ++ struct snd_soc_codec *codec; ++ int ret = 0; ++ ++ info("WM8974 Audio Codec %s", WM8974_VERSION); ++ ++ setup = socdev->codec_data; ++ codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); ++ if (codec == NULL) ++ return -ENOMEM; ++ ++ socdev->codec = codec; ++ mutex_init(&codec->mutex); ++ INIT_LIST_HEAD(&codec->dapm_widgets); ++ INIT_LIST_HEAD(&codec->dapm_paths); ++ ++ wm8974_socdev = socdev; ++#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) ++ if (setup->i2c_address) { ++ normal_i2c[0] = setup->i2c_address; ++ codec->hw_write = (hw_write_t)i2c_master_send; ++ ret = i2c_add_driver(&wm8974_i2c_driver); ++ if (ret != 0) ++ printk(KERN_ERR "can't add i2c driver"); ++ } ++#else ++ /* Add other interfaces here */ ++#endif ++ return ret; ++} ++ ++/* power down chip */ ++static int wm8974_remove(struct platform_device *pdev) ++{ ++ struct snd_soc_device *socdev = platform_get_drvdata(pdev); ++ struct snd_soc_codec *codec = socdev->codec; ++ ++ if (codec->control_data) ++ wm8974_dapm_event(codec, SNDRV_CTL_POWER_D3cold); ++ ++ snd_soc_free_pcms(socdev); ++ snd_soc_dapm_free(socdev); ++#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) ++ i2c_del_driver(&wm8974_i2c_driver); ++#endif ++ kfree(codec); ++ ++ return 0; ++} ++ ++struct snd_soc_codec_device soc_codec_dev_wm8974 = { ++ .probe = wm8974_probe, ++ .remove = wm8974_remove, ++ .suspend = wm8974_suspend, ++ .resume = wm8974_resume, ++}; ++ ++EXPORT_SYMBOL_GPL(soc_codec_dev_wm8974); ++ ++MODULE_DESCRIPTION("ASoC WM8974 driver"); ++MODULE_AUTHOR("Liam Girdwood"); ++MODULE_LICENSE("GPL"); +Index: linux-2.6-pxa-new/sound/soc/codecs/wm8974.h +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/sound/soc/codecs/wm8974.h +@@ -0,0 +1,64 @@ ++/* ++ * wm8974.h -- WM8974 Soc Audio driver ++ * ++ * This program is free software; you can redistribute it and/or modify ++ * it under the terms of the GNU General Public License version 2 as ++ * published by the Free Software Foundation. ++ */ ++ ++#ifndef _WM8974_H ++#define _WM8974_H ++ ++/* WM8974 register space */ ++ ++#define WM8974_RESET 0x0 ++#define WM8974_POWER1 0x1 ++#define WM8974_POWER2 0x2 ++#define WM8974_POWER3 0x3 ++#define WM8974_IFACE 0x4 ++#define WM8974_COMP 0x5 ++#define WM8974_CLOCK 0x6 ++#define WM8974_ADD 0x7 ++#define WM8974_GPIO 0x8 ++#define WM8974_DAC 0xa ++#define WM8974_DACVOL 0xb ++#define WM8974_ADC 0xe ++#define WM8974_ADCVOL 0xf ++#define WM8974_EQ1 0x12 ++#define WM8974_EQ2 0x13 ++#define WM8974_EQ3 0x14 ++#define WM8974_EQ4 0x15 ++#define WM8974_EQ5 0x16 ++#define WM8974_DACLIM1 0x18 ++#define WM8974_DACLIM2 0x19 ++#define WM8974_NOTCH1 0x1b ++#define WM8974_NOTCH2 0x1c ++#define WM8974_NOTCH3 0x1d ++#define WM8974_NOTCH4 0x1e ++#define WM8974_ALC1 0x20 ++#define WM8974_ALC2 0x21 ++#define WM8974_ALC3 0x22 ++#define WM8974_NGATE 0x23 ++#define WM8974_PLLN 0x24 ++#define WM8974_PLLK1 0x25 ++#define WM8974_PLLK2 0x26 ++#define WM8974_PLLK3 0x27 ++#define WM8974_ATTEN 0x28 ++#define WM8974_INPUT 0x2c ++#define WM8974_INPPGA 0x2d ++#define WM8974_ADCBOOST 0x2f ++#define WM8974_OUTPUT 0x31 ++#define WM8974_SPKMIX 0x32 ++#define WM8974_SPKVOL 0x36 ++#define WM8974_MONOMIX 0x38 ++ ++#define WM8974_CACHEREGNUM 57 ++ ++struct wm8974_setup_data { ++ unsigned short i2c_address; ++}; ++ ++extern struct snd_soc_codec_dai wm8974_dai; ++extern struct snd_soc_codec_device soc_codec_dev_wm8974; ++ ++#endif +Index: linux-2.6-pxa-new/sound/soc/codecs/wm9712.c +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/sound/soc/codecs/wm9712.c +@@ -0,0 +1,781 @@ ++/* ++ * wm9712.c -- ALSA Soc WM9712 codec support ++ * ++ * Copyright 2006 Wolfson Microelectronics PLC. ++ * Author: Liam Girdwood ++ * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com ++ * ++ * This program is free software; you can redistribute it and/or modify it ++ * under the terms of the GNU General Public License as published by the ++ * Free Software Foundation; either version 2 of the License, or (at your ++ * option) any later version. ++ * ++ * Revision history ++ * 4th Feb 2006 Initial version. ++ */ ++ ++#include <linux/init.h> ++#include <linux/module.h> ++#include <linux/version.h> ++#include <linux/kernel.h> ++#include <linux/device.h> ++#include <sound/driver.h> ++#include <sound/core.h> ++#include <sound/pcm.h> ++#include <sound/ac97_codec.h> ++#include <sound/initval.h> ++#include <sound/soc.h> ++#include <sound/soc-dapm.h> ++ ++#define WM9712_VERSION "0.4" ++ ++static unsigned int ac97_read(struct snd_soc_codec *codec, ++ unsigned int reg); ++static int ac97_write(struct snd_soc_codec *codec, ++ unsigned int reg, unsigned int val); ++ ++#define AC97_DIR \ ++ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE) ++ ++#define AC97_RATES \ ++ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ ++ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ ++ SNDRV_PCM_RATE_48000) ++ ++/* may need to expand this */ ++static struct snd_soc_dai_mode ac97_modes[] = { ++ { ++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S18_3LE, ++ .pcmrate = AC97_RATES, ++ .pcmdir = AC97_DIR, ++ }, ++}; ++ ++/* ++ * WM9712 register cache ++ */ ++static const u16 wm9712_reg[] = { ++ 0x6174, 0x8000, 0x8000, 0x8000, // 6 ++ 0xf0f0, 0xaaa0, 0xc008, 0x6808, // e ++ 0xe808, 0xaaa0, 0xad00, 0x8000, // 16 ++ 0xe808, 0x3000, 0x8000, 0x0000, // 1e ++ 0x0000, 0x0000, 0x0000, 0x000f, // 26 ++ 0x0405, 0x0410, 0xbb80, 0xbb80, // 2e ++ 0x0000, 0xbb80, 0x0000, 0x0000, // 36 ++ 0x0000, 0x2000, 0x0000, 0x0000, // 3e ++ 0x0000, 0x0000, 0x0000, 0x0000, // 46 ++ 0x0000, 0x0000, 0xf83e, 0xffff, // 4e ++ 0x0000, 0x0000, 0x0000, 0xf83e, // 56 ++ 0x0008, 0x0000, 0x0000, 0x0000, // 5e ++ 0xb032, 0x3e00, 0x0000, 0x0000, // 66 ++ 0x0000, 0x0000, 0x0000, 0x0000, // 6e ++ 0x0000, 0x0000, 0x0000, 0x0006, // 76 ++ 0x0001, 0x0000, 0x574d, 0x4c12, // 7e ++ 0x0000, 0x0000 // virtual hp mixers ++}; ++ ++/* virtual HP mixers regs */ ++#define HPL_MIXER 0x80 ++#define HPR_MIXER 0x82 ++ ++static const char *wm9712_alc_select[] = {"None", "Left", "Right", "Stereo"}; ++static const char *wm9712_alc_mux[] = {"Stereo", "Left", "Right", "None"}; ++static const char *wm9712_out3_src[] = {"Left", "VREF", "Left + Right", ++ "Mono"}; ++static const char *wm9712_spk_src[] = {"Speaker Mix", "Headphone Mix"}; ++static const char *wm9712_rec_adc[] = {"Stereo", "Left", "Right", "Mute"}; ++static const char *wm9712_base[] = {"Linear Control", "Adaptive Boost"}; ++static const char *wm9712_rec_gain[] = {"+1.5dB Steps", "+0.75dB Steps"}; ++static const char *wm9712_mic[] = {"Mic 1", "Differential", "Mic 2", ++ "Stereo"}; ++static const char *wm9712_rec_sel[] = {"Mic", "NC", "NC", "Speaker Mixer", ++ "Line", "Headphone Mixer", "Phone Mixer", "Phone"}; ++static const char *wm9712_ng_type[] = {"Constant Gain", "Mute"}; ++static const char *wm9712_diff_sel[] = {"Mic", "Line"}; ++ ++static const struct soc_enum wm9712_enum[] = { ++SOC_ENUM_SINGLE(AC97_PCI_SVID, 14, 4, wm9712_alc_select), ++SOC_ENUM_SINGLE(AC97_VIDEO, 12, 4, wm9712_alc_mux), ++SOC_ENUM_SINGLE(AC97_AUX, 9, 4, wm9712_out3_src), ++SOC_ENUM_SINGLE(AC97_AUX, 8, 2, wm9712_spk_src), ++SOC_ENUM_SINGLE(AC97_REC_SEL, 12, 4, wm9712_rec_adc), ++SOC_ENUM_SINGLE(AC97_MASTER_TONE, 15, 2, wm9712_base), ++SOC_ENUM_DOUBLE(AC97_REC_GAIN, 14, 6, 2, wm9712_rec_gain), ++SOC_ENUM_SINGLE(AC97_MIC, 5, 4, wm9712_mic), ++SOC_ENUM_SINGLE(AC97_REC_SEL, 8, 8, wm9712_rec_sel), ++SOC_ENUM_SINGLE(AC97_REC_SEL, 0, 8, wm9712_rec_sel), ++SOC_ENUM_SINGLE(AC97_PCI_SVID, 5, 2, wm9712_ng_type), ++SOC_ENUM_SINGLE(0x5c, 8, 2, wm9712_diff_sel), ++}; ++ ++static const struct snd_kcontrol_new wm9712_snd_ac97_controls[] = { ++SOC_DOUBLE("Speaker Playback Volume", AC97_MASTER, 8, 0, 31, 1), ++SOC_SINGLE("Speaker Playback Switch", AC97_MASTER, 15, 1, 1), ++SOC_DOUBLE("Headphone Playback Volume", AC97_HEADPHONE, 8, 0, 31, 1), ++SOC_SINGLE("Headphone Playback Switch", AC97_HEADPHONE,15, 1, 1), ++ ++SOC_SINGLE("Speaker Playback ZC Switch", AC97_MASTER, 7, 1, 0), ++SOC_SINGLE("Speaker Playback Invert Switch", AC97_MASTER, 6, 1, 0), ++SOC_SINGLE("Headphone Playback ZC Switch", AC97_HEADPHONE, 7, 1, 0), ++SOC_SINGLE("Mono Playback ZC Switch", AC97_MASTER_MONO, 7, 1, 0), ++SOC_SINGLE("Mono Playback Volume", AC97_MASTER_MONO, 0, 31, 0), ++ ++SOC_SINGLE("ALC Target Volume", AC97_CODEC_CLASS_REV, 12, 15, 0), ++SOC_SINGLE("ALC Hold Time", AC97_CODEC_CLASS_REV, 8, 15, 0), ++SOC_SINGLE("ALC Decay Time", AC97_CODEC_CLASS_REV, 4, 15, 0), ++SOC_SINGLE("ALC Attack Time", AC97_CODEC_CLASS_REV, 0, 15, 0), ++SOC_ENUM("ALC Function", wm9712_enum[0]), ++SOC_SINGLE("ALC Max Volume", AC97_PCI_SVID, 11, 7, 0), ++SOC_SINGLE("ALC ZC Timeout", AC97_PCI_SVID, 9, 3, 1), ++SOC_SINGLE("ALC ZC Switch", AC97_PCI_SVID, 8, 1, 0), ++SOC_SINGLE("ALC NG Switch", AC97_PCI_SVID, 7, 1, 0), ++SOC_ENUM("ALC NG Type", wm9712_enum[10]), ++SOC_SINGLE("ALC NG Threshold", AC97_PCI_SVID, 0, 31, 1), ++ ++SOC_SINGLE("Mic Headphone Volume", AC97_VIDEO, 12, 7, 1), ++SOC_SINGLE("ALC Headphone Volume", AC97_VIDEO, 7, 7, 1), ++ ++SOC_SINGLE("Out3 Switch", AC97_AUX, 15, 1, 1), ++SOC_SINGLE("Out3 ZC Switch", AC97_AUX, 7, 1, 1), ++SOC_SINGLE("Out3 Volume", AC97_AUX, 0, 31, 1), ++ ++SOC_SINGLE("PCBeep Bypass Headphone Volume", AC97_PC_BEEP, 12, 7, 1), ++SOC_SINGLE("PCBeep Bypass Speaker Volume", AC97_PC_BEEP, 8, 7, 1), ++SOC_SINGLE("PCBeep Bypass Phone Volume", AC97_PC_BEEP, 4, 7, 1), ++ ++SOC_SINGLE("Aux Playback Headphone Volume", AC97_CD, 12, 7, 1), ++SOC_SINGLE("Aux Playback Speaker Volume", AC97_CD, 8, 7, 1), ++SOC_SINGLE("Aux Playback Phone Volume", AC97_CD, 4, 7, 1), ++ ++SOC_SINGLE("Phone Volume", AC97_PHONE, 0, 15, 0), ++SOC_DOUBLE("Line Capture Volume", AC97_LINE, 8, 0, 31, 1), ++ ++SOC_SINGLE("Capture 20dB Boost Switch", AC97_REC_SEL, 14, 1, 0), ++SOC_SINGLE("Capture to Phone 20dB Boost Switch", AC97_REC_SEL, 11, 1, 1), ++ ++SOC_SINGLE("3D Upper Cut-off Switch", AC97_3D_CONTROL, 5, 1, 1), ++SOC_SINGLE("3D Lower Cut-off Switch", AC97_3D_CONTROL, 4, 1, 1), ++SOC_SINGLE("3D Playback Volume", AC97_3D_CONTROL, 0, 15, 0), ++ ++SOC_ENUM("Bass Control", wm9712_enum[5]), ++SOC_SINGLE("Bass Cut-off Switch", AC97_MASTER_TONE, 12, 1, 1), ++SOC_SINGLE("Tone Cut-off Switch", AC97_MASTER_TONE, 4, 1, 1), ++SOC_SINGLE("Playback Attenuate (-6dB) Switch", AC97_MASTER_TONE, 6, 1, 0), ++SOC_SINGLE("Bass Volume", AC97_MASTER_TONE, 8, 15, 0), ++SOC_SINGLE("Treble Volume", AC97_MASTER_TONE, 0, 15, 0), ++ ++SOC_SINGLE("Capture ADC Switch", AC97_REC_GAIN, 15, 1, 1), ++SOC_ENUM("Capture Volume Steps", wm9712_enum[6]), ++SOC_DOUBLE("Capture Volume", AC97_REC_GAIN, 8, 0, 63, 1), ++SOC_SINGLE("Capture ZC Switch", AC97_REC_GAIN, 7, 1, 0), ++ ++SOC_SINGLE("Mic 1 Volume", AC97_MIC, 8, 31, 1), ++SOC_SINGLE("Mic 2 Volume", AC97_MIC, 0, 31, 1), ++SOC_SINGLE("Mic 20dB Boost Switch", AC97_MIC, 7, 1, 0), ++}; ++ ++/* add non dapm controls */ ++static int wm9712_add_controls(struct snd_soc_codec *codec) ++{ ++ int err, i; ++ ++ for (i = 0; i < ARRAY_SIZE(wm9712_snd_ac97_controls); i++) { ++ err = snd_ctl_add(codec->card, ++ snd_soc_cnew(&wm9712_snd_ac97_controls[i],codec, NULL)); ++ if (err < 0) ++ return err; ++ } ++ return 0; ++} ++ ++/* We have to create a fake left and right HP mixers because ++ * the codec only has a single control that is shared by both channels. ++ * This makes it impossible to determine the audio path. ++ */ ++static int mixer_event (struct snd_soc_dapm_widget *w, int event) ++{ ++ u16 l, r, beep, line, phone, mic, pcm, aux; ++ ++ l = ac97_read(w->codec, HPL_MIXER); ++ r = ac97_read(w->codec, HPR_MIXER); ++ beep = ac97_read(w->codec, AC97_PC_BEEP); ++ mic = ac97_read(w->codec, AC97_VIDEO); ++ phone = ac97_read(w->codec, AC97_PHONE); ++ line = ac97_read(w->codec, AC97_LINE); ++ pcm = ac97_read(w->codec, AC97_PCM); ++ aux = ac97_read(w->codec, AC97_CD); ++ ++ if (l & 0x1 || r & 0x1) ++ ac97_write(w->codec, AC97_VIDEO, mic & 0x7fff); ++ else ++ ac97_write(w->codec, AC97_VIDEO, mic | 0x8000); ++ ++ if (l & 0x2 || r & 0x2) ++ ac97_write(w->codec, AC97_PCM, pcm & 0x7fff); ++ else ++ ac97_write(w->codec, AC97_PCM, pcm | 0x8000); ++ ++ if (l & 0x4 || r & 0x4) ++ ac97_write(w->codec, AC97_LINE, line & 0x7fff); ++ else ++ ac97_write(w->codec, AC97_LINE, line | 0x8000); ++ ++ if (l & 0x8 || r & 0x8) ++ ac97_write(w->codec, AC97_PHONE, phone & 0x7fff); ++ else ++ ac97_write(w->codec, AC97_PHONE, phone | 0x8000); ++ ++ if (l & 0x10 || r & 0x10) ++ ac97_write(w->codec, AC97_CD, aux & 0x7fff); ++ else ++ ac97_write(w->codec, AC97_CD, aux | 0x8000); ++ ++ if (l & 0x20 || r & 0x20) ++ ac97_write(w->codec, AC97_PC_BEEP, beep & 0x7fff); ++ else ++ ac97_write(w->codec, AC97_PC_BEEP, beep | 0x8000); ++ ++ return 0; ++} ++ ++/* Left Headphone Mixers */ ++static const struct snd_kcontrol_new wm9712_hpl_mixer_controls[] = { ++ SOC_DAPM_SINGLE("PCBeep Bypass Switch", HPL_MIXER, 5, 1, 0), ++ SOC_DAPM_SINGLE("Aux Playback Switch", HPL_MIXER, 4, 1, 0), ++ SOC_DAPM_SINGLE("Phone Bypass Switch", HPL_MIXER, 3, 1, 0), ++ SOC_DAPM_SINGLE("Line Bypass Switch", HPL_MIXER, 2, 1, 0), ++ SOC_DAPM_SINGLE("PCM Playback Switch", HPL_MIXER, 1, 1, 0), ++ SOC_DAPM_SINGLE("Mic Sidetone Switch", HPL_MIXER, 0, 1, 0), ++}; ++ ++/* Right Headphone Mixers */ ++static const struct snd_kcontrol_new wm9712_hpr_mixer_controls[] = { ++ SOC_DAPM_SINGLE("PCBeep Bypass Switch", HPR_MIXER, 5, 1, 0), ++ SOC_DAPM_SINGLE("Aux Playback Switch", HPR_MIXER, 4, 1, 0), ++ SOC_DAPM_SINGLE("Phone Bypass Switch", HPR_MIXER, 3, 1, 0), ++ SOC_DAPM_SINGLE("Line Bypass Switch", HPR_MIXER, 2, 1, 0), ++ SOC_DAPM_SINGLE("PCM Playback Switch", HPR_MIXER, 1, 1, 0), ++ SOC_DAPM_SINGLE("Mic Sidetone Switch", HPR_MIXER, 0, 1, 0), ++}; ++ ++/* Speaker Mixer */ ++static const struct snd_kcontrol_new wm9712_speaker_mixer_controls[] = { ++ SOC_DAPM_SINGLE("PCBeep Bypass Switch", AC97_PC_BEEP, 11, 1, 1), ++ SOC_DAPM_SINGLE("Aux Playback Switch", AC97_CD, 11, 1, 1), ++ SOC_DAPM_SINGLE("Phone Bypass Switch", AC97_PHONE, 14, 1, 1), ++ SOC_DAPM_SINGLE("Line Bypass Switch", AC97_LINE, 14, 1, 1), ++ SOC_DAPM_SINGLE("PCM Playback Switch", AC97_PCM, 14, 1, 1), ++}; ++ ++/* Phone Mixer */ ++static const struct snd_kcontrol_new wm9712_phone_mixer_controls[] = { ++ SOC_DAPM_SINGLE("PCBeep Bypass Switch", AC97_PC_BEEP, 7, 1, 1), ++ SOC_DAPM_SINGLE("Aux Playback Switch", AC97_CD, 7, 1, 1), ++ SOC_DAPM_SINGLE("Line Bypass Switch", AC97_LINE, 13, 1, 1), ++ SOC_DAPM_SINGLE("PCM Playback Switch", AC97_PCM, 13, 1, 1), ++ SOC_DAPM_SINGLE("Mic 1 Sidetone Switch", AC97_MIC, 14, 1, 1), ++ SOC_DAPM_SINGLE("Mic 2 Sidetone Switch", AC97_MIC, 13, 1, 1), ++}; ++ ++/* ALC headphone mux */ ++static const struct snd_kcontrol_new wm9712_alc_mux_controls = ++SOC_DAPM_ENUM("Route", wm9712_enum[1]); ++ ++/* out 3 mux */ ++static const struct snd_kcontrol_new wm9712_out3_mux_controls = ++SOC_DAPM_ENUM("Route", wm9712_enum[2]); ++ ++/* spk mux */ ++static const struct snd_kcontrol_new wm9712_spk_mux_controls = ++SOC_DAPM_ENUM("Route", wm9712_enum[3]); ++ ++/* Capture to Phone mux */ ++static const struct snd_kcontrol_new wm9712_capture_phone_mux_controls = ++SOC_DAPM_ENUM("Route", wm9712_enum[4]); ++ ++/* Capture left select */ ++static const struct snd_kcontrol_new wm9712_capture_selectl_controls = ++SOC_DAPM_ENUM("Route", wm9712_enum[8]); ++ ++/* Capture right select */ ++static const struct snd_kcontrol_new wm9712_capture_selectr_controls = ++SOC_DAPM_ENUM("Route", wm9712_enum[9]); ++ ++/* Mic select */ ++static const struct snd_kcontrol_new wm9712_mic_src_controls = ++SOC_DAPM_ENUM("Route", wm9712_enum[7]); ++ ++/* diff select */ ++static const struct snd_kcontrol_new wm9712_diff_sel_controls = ++SOC_DAPM_ENUM("Route", wm9712_enum[11]); ++ ++static const struct snd_soc_dapm_widget wm9712_dapm_widgets[] = { ++SND_SOC_DAPM_MUX("ALC Sidetone Mux", SND_SOC_NOPM, 0, 0, ++ &wm9712_alc_mux_controls), ++SND_SOC_DAPM_MUX("Out3 Mux", SND_SOC_NOPM, 0, 0, ++ &wm9712_out3_mux_controls), ++SND_SOC_DAPM_MUX("Speaker Mux", SND_SOC_NOPM, 0, 0, ++ &wm9712_spk_mux_controls), ++SND_SOC_DAPM_MUX("Capture Phone Mux", SND_SOC_NOPM, 0, 0, ++ &wm9712_capture_phone_mux_controls), ++SND_SOC_DAPM_MUX("Left Capture Select", SND_SOC_NOPM, 0, 0, ++ &wm9712_capture_selectl_controls), ++SND_SOC_DAPM_MUX("Right Capture Select", SND_SOC_NOPM, 0, 0, ++ &wm9712_capture_selectr_controls), ++SND_SOC_DAPM_MUX("Mic Select Source", SND_SOC_NOPM, 0, 0, ++ &wm9712_mic_src_controls), ++SND_SOC_DAPM_MUX("Differential Source", SND_SOC_NOPM, 0, 0, ++ &wm9712_diff_sel_controls), ++SND_SOC_DAPM_MIXER("AC97 Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), ++SND_SOC_DAPM_MIXER_E("Left HP Mixer", AC97_INT_PAGING, 9, 1, ++ &wm9712_hpl_mixer_controls[0], ARRAY_SIZE(wm9712_hpl_mixer_controls), ++ mixer_event, SND_SOC_DAPM_POST_REG), ++SND_SOC_DAPM_MIXER_E("Right HP Mixer", AC97_INT_PAGING, 8, 1, ++ &wm9712_hpr_mixer_controls[0], ARRAY_SIZE(wm9712_hpr_mixer_controls), ++ mixer_event, SND_SOC_DAPM_POST_REG), ++SND_SOC_DAPM_MIXER("Phone Mixer", AC97_INT_PAGING, 6, 1, ++ &wm9712_phone_mixer_controls[0], ARRAY_SIZE(wm9712_phone_mixer_controls)), ++SND_SOC_DAPM_MIXER("Speaker Mixer", AC97_INT_PAGING, 7, 1, ++ &wm9712_speaker_mixer_controls[0], ++ ARRAY_SIZE(wm9712_speaker_mixer_controls)), ++SND_SOC_DAPM_MIXER("Mono Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), ++SND_SOC_DAPM_DAC("Left DAC", "Left HiFi Playback", AC97_INT_PAGING, 14, 1), ++SND_SOC_DAPM_DAC("Right DAC", "Right HiFi Playback", AC97_INT_PAGING, 13, 1), ++SND_SOC_DAPM_DAC("Aux DAC", "Aux Playback", SND_SOC_NOPM, 0, 0), ++SND_SOC_DAPM_ADC("Left ADC", "Left HiFi Capture", AC97_INT_PAGING, 12, 1), ++SND_SOC_DAPM_ADC("Right ADC", "Right HiFi Capture", AC97_INT_PAGING, 11, 1), ++SND_SOC_DAPM_PGA("Headphone PGA", AC97_INT_PAGING, 4, 1, NULL, 0), ++SND_SOC_DAPM_PGA("Speaker PGA", AC97_INT_PAGING, 3, 1, NULL, 0), ++SND_SOC_DAPM_PGA("Out 3 PGA", AC97_INT_PAGING, 5, 1, NULL, 0), ++SND_SOC_DAPM_PGA("Line PGA", AC97_INT_PAGING, 2, 1, NULL, 0), ++SND_SOC_DAPM_PGA("Phone PGA", AC97_INT_PAGING, 1, 1, NULL, 0), ++SND_SOC_DAPM_PGA("Mic PGA", AC97_INT_PAGING, 0, 1, NULL, 0), ++SND_SOC_DAPM_MICBIAS("Mic Bias", AC97_INT_PAGING, 10, 1), ++SND_SOC_DAPM_OUTPUT("MONOOUT"), ++SND_SOC_DAPM_OUTPUT("HPOUTL"), ++SND_SOC_DAPM_OUTPUT("HPOUTR"), ++SND_SOC_DAPM_OUTPUT("LOUT2"), ++SND_SOC_DAPM_OUTPUT("ROUT2"), ++SND_SOC_DAPM_OUTPUT("OUT3"), ++SND_SOC_DAPM_INPUT("LINEINL"), ++SND_SOC_DAPM_INPUT("LINEINR"), ++SND_SOC_DAPM_INPUT("PHONE"), ++SND_SOC_DAPM_INPUT("PCBEEP"), ++SND_SOC_DAPM_INPUT("MIC1"), ++SND_SOC_DAPM_INPUT("MIC2"), ++}; ++ ++static const char *audio_map[][3] = { ++ /* virtual mixer - mixes left & right channels for spk and mono */ ++ {"AC97 Mixer", NULL, "Left DAC"}, ++ {"AC97 Mixer", NULL, "Right DAC"}, ++ ++ /* Left HP mixer */ ++ {"Left HP Mixer", "PCBeep Bypass Switch", "PCBEEP"}, ++ {"Left HP Mixer", "Aux Playback Switch", "Aux DAC"}, ++ {"Left HP Mixer", "Phone Bypass Switch", "Phone PGA"}, ++ {"Left HP Mixer", "Line Bypass Switch", "Line PGA"}, ++ {"Left HP Mixer", "PCM Playback Switch", "Left DAC"}, ++ {"Left HP Mixer", "Mic Sidetone Switch", "Mic PGA"}, ++ {"Left HP Mixer", NULL, "ALC Sidetone Mux"}, ++ //{"Right HP Mixer", NULL, "HP Mixer"}, ++ ++ /* Right HP mixer */ ++ {"Right HP Mixer", "PCBeep Bypass Switch", "PCBEEP"}, ++ {"Right HP Mixer", "Aux Playback Switch", "Aux DAC"}, ++ {"Right HP Mixer", "Phone Bypass Switch", "Phone PGA"}, ++ {"Right HP Mixer", "Line Bypass Switch", "Line PGA"}, ++ {"Right HP Mixer", "PCM Playback Switch", "Right DAC"}, ++ {"Right HP Mixer", "Mic Sidetone Switch", "Mic PGA"}, ++ {"Right HP Mixer", NULL, "ALC Sidetone Mux"}, ++ ++ /* speaker mixer */ ++ {"Speaker Mixer", "PCBeep Bypass Switch", "PCBEEP"}, ++ {"Speaker Mixer", "Line Bypass Switch", "Line PGA"}, ++ {"Speaker Mixer", "PCM Playback Switch", "AC97 Mixer"}, ++ {"Speaker Mixer", "Phone Bypass Switch", "Phone PGA"}, ++ {"Speaker Mixer", "Aux Playback Switch", "Aux DAC"}, ++ ++ /* Phone mixer */ ++ {"Phone Mixer", "PCBeep Bypass Switch", "PCBEEP"}, ++ {"Phone Mixer", "Line Bypass Switch", "Line PGA"}, ++ {"Phone Mixer", "Aux Playback Switch", "Aux DAC"}, ++ {"Phone Mixer", "PCM Playback Switch", "AC97 Mixer"}, ++ {"Phone Mixer", "Mic 1 Sidetone Switch", "Mic PGA"}, ++ {"Phone Mixer", "Mic 2 Sidetone Switch", "Mic PGA"}, ++ ++ /* inputs */ ++ {"Line PGA", NULL, "LINEINL"}, ++ {"Line PGA", NULL, "LINEINR"}, ++ {"Phone PGA", NULL, "PHONE"}, ++ {"Mic PGA", NULL, "MIC1"}, ++ {"Mic PGA", NULL, "MIC2"}, ++ ++ /* left capture selector */ ++ {"Left Capture Select", "Mic", "MIC1"}, ++ {"Left Capture Select", "Speaker Mixer", "Speaker Mixer"}, ++ {"Left Capture Select", "Line", "LINEINL"}, ++ {"Left Capture Select", "Headphone Mixer", "Left HP Mixer"}, ++ {"Left Capture Select", "Phone Mixer", "Phone Mixer"}, ++ {"Left Capture Select", "Phone", "PHONE"}, ++ ++ /* right capture selector */ ++ {"Right Capture Select", "Mic", "MIC2"}, ++ {"Right Capture Select", "Speaker Mixer", "Speaker Mixer"}, ++ {"Right Capture Select", "Line", "LINEINR"}, ++ {"Right Capture Select", "Headphone Mixer", "Right HP Mixer"}, ++ {"Right Capture Select", "Phone Mixer", "Phone Mixer"}, ++ {"Right Capture Select", "Phone", "PHONE"}, ++ ++ /* ALC Sidetone */ ++ {"ALC Sidetone Mux", "Stereo", "Left Capture Select"}, ++ {"ALC Sidetone Mux", "Stereo", "Right Capture Select"}, ++ {"ALC Sidetone Mux", "Left", "Left Capture Select"}, ++ {"ALC Sidetone Mux", "Right", "Right Capture Select"}, ++ ++ /* ADC's */ ++ {"Left ADC", NULL, "Left Capture Select"}, ++ {"Right ADC", NULL, "Right Capture Select"}, ++ ++ /* outputs */ ++ {"MONOOUT", NULL, "Phone Mixer"}, ++ {"HPOUTL", NULL, "Headphone PGA"}, ++ {"Headphone PGA", NULL, "Left HP Mixer"}, ++ {"HPOUTR", NULL, "Headphone PGA"}, ++ {"Headphone PGA", NULL, "Right HP Mixer"}, ++ ++ /* mono hp mixer */ ++ {"Mono HP Mixer", NULL, "Left HP Mixer"}, ++ {"Mono HP Mixer", NULL, "Right HP Mixer"}, ++ ++ /* Out3 Mux */ ++ {"Out3 Mux", "Left", "Left HP Mixer"}, ++ {"Out3 Mux", "Mono", "Phone Mixer"}, ++ {"Out3 Mux", "Left + Right", "Mono HP Mixer"}, ++ {"Out 3 PGA", NULL, "Out3 Mux"}, ++ {"OUT3", NULL, "Out 3 PGA"}, ++ ++ /* speaker Mux */ ++ {"Speaker Mux", "Speaker Mix", "Speaker Mixer"}, ++ {"Speaker Mux", "Headphone Mix", "Mono HP Mixer"}, ++ {"Speaker PGA", NULL, "Speaker Mux"}, ++ {"LOUT2", NULL, "Speaker PGA"}, ++ {"ROUT2", NULL, "Speaker PGA"}, ++ ++ {NULL, NULL, NULL}, ++}; ++ ++static int wm9712_add_widgets(struct snd_soc_codec *codec) ++{ ++ int i; ++ ++ for(i = 0; i < ARRAY_SIZE(wm9712_dapm_widgets); i++) { ++ snd_soc_dapm_new_control(codec, &wm9712_dapm_widgets[i]); ++ } ++ ++ /* set up audio path audio_mapnects */ ++ for(i = 0; audio_map[i][0] != NULL; i++) { ++ snd_soc_dapm_connect_input(codec, audio_map[i][0], ++ audio_map[i][1], audio_map[i][2]); ++ } ++ ++ snd_soc_dapm_new_widgets(codec); ++ return 0; ++} ++ ++static unsigned int ac97_read(struct snd_soc_codec *codec, ++ unsigned int reg) ++{ ++ u16 *cache = codec->reg_cache; ++ ++ if (reg == AC97_RESET || reg == AC97_GPIO_STATUS || ++ reg == AC97_VENDOR_ID1 || reg == AC97_VENDOR_ID2 || ++ reg == AC97_REC_GAIN) ++ return soc_ac97_ops.read(codec->ac97, reg); ++ else { ++ reg = reg >> 1; ++ ++ if (reg > (ARRAY_SIZE(wm9712_reg))) ++ return -EIO; ++ ++ return cache[reg]; ++ } ++} ++ ++static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, ++ unsigned int val) ++{ ++ u16 *cache = codec->reg_cache; ++ ++ soc_ac97_ops.write(codec->ac97, reg, val); ++ reg = reg >> 1; ++ if (reg <= (ARRAY_SIZE(wm9712_reg))) ++ cache[reg] = val; ++ ++ return 0; ++} ++ ++static int ac97_prepare(struct snd_pcm_substream *substream) ++{ ++ struct snd_pcm_runtime *runtime = substream->runtime; ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ struct snd_soc_device *socdev = rtd->socdev; ++ struct snd_soc_codec *codec = socdev->codec; ++ int reg; ++ u16 vra; ++ ++ vra = ac97_read(codec, AC97_EXTENDED_STATUS); ++ ac97_write(codec, AC97_EXTENDED_STATUS, vra | 0x1); ++ ++ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ++ reg = AC97_PCM_FRONT_DAC_RATE; ++ else ++ reg = AC97_PCM_LR_ADC_RATE; ++ ++ return ac97_write(codec, reg, runtime->rate); ++} ++ ++static int ac97_aux_prepare(struct snd_pcm_substream *substream) ++{ ++ struct snd_pcm_runtime *runtime = substream->runtime; ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ struct snd_soc_device *socdev = rtd->socdev; ++ struct snd_soc_codec *codec = socdev->codec; ++ u16 vra, xsle; ++ ++ vra = ac97_read(codec, AC97_EXTENDED_STATUS); ++ ac97_write(codec, AC97_EXTENDED_STATUS, vra | 0x1); ++ xsle = ac97_read(codec, AC97_PCI_SID); ++ ac97_write(codec, AC97_PCI_SID, xsle | 0x8000); ++ ++ if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) ++ return -ENODEV; ++ ++ return ac97_write(codec, AC97_PCM_SURR_DAC_RATE, runtime->rate); ++} ++ ++struct snd_soc_codec_dai wm9712_dai[] = { ++{ ++ .name = "AC97 HiFi", ++ .playback = { ++ .stream_name = "HiFi Playback", ++ .channels_min = 1, ++ .channels_max = 2,}, ++ .capture = { ++ .stream_name = "HiFi Capture", ++ .channels_min = 1, ++ .channels_max = 2,}, ++ .ops = { ++ .prepare = ac97_prepare,}, ++ .caps = { ++ .num_modes = ARRAY_SIZE(ac97_modes), ++ .mode = ac97_modes,}, ++ }, ++ { ++ .name = "AC97 Aux", ++ .playback = { ++ .stream_name = "Aux Playback", ++ .channels_min = 1, ++ .channels_max = 1,}, ++ .ops = { ++ .prepare = ac97_aux_prepare,}, ++ .caps = { ++ .num_modes = ARRAY_SIZE(ac97_modes), ++ .mode = ac97_modes,}, ++ }, ++}; ++EXPORT_SYMBOL_GPL(wm9712_dai); ++ ++static int wm9712_dapm_event(struct snd_soc_codec *codec, int event) ++{ ++ u16 reg; ++ ++ switch (event) { ++ case SNDRV_CTL_POWER_D0: /* full On */ ++ /* liam - maybe enable thermal shutdown */ ++ reg = ac97_read(codec, AC97_EXTENDED_MID) & 0xdfff; ++ ac97_write(codec, AC97_EXTENDED_MID, reg); ++ break; ++ case SNDRV_CTL_POWER_D1: /* partial On */ ++ case SNDRV_CTL_POWER_D2: /* partial On */ ++ break; ++ case SNDRV_CTL_POWER_D3hot: /* Off, with power */ ++ /* enable master bias and vmid */ ++ reg = ac97_read(codec, AC97_EXTENDED_MID) & 0xbbff; ++ ac97_write(codec, AC97_EXTENDED_MID, reg); ++ ac97_write(codec, AC97_POWERDOWN, 0x0000); ++ break; ++ case SNDRV_CTL_POWER_D3cold: /* Off, without power */ ++ /* disable everything including AC link */ ++ ac97_write(codec, AC97_EXTENDED_MID, 0xffff); ++ ac97_write(codec, AC97_EXTENDED_MSTATUS, 0xffff); ++ ac97_write(codec, AC97_POWERDOWN, 0xffff); ++ break; ++ } ++ codec->dapm_state = event; ++ return 0; ++} ++ ++static int wm9712_reset(struct snd_soc_codec *codec, int try_warm) ++{ ++ if (try_warm && soc_ac97_ops.warm_reset) { ++ soc_ac97_ops.warm_reset(codec->ac97); ++ if (!(ac97_read(codec, 0) & 0x8000)) ++ return 1; ++ } ++ ++ soc_ac97_ops.reset(codec->ac97); ++ if (ac97_read(codec, 0) & 0x8000) ++ goto err; ++ return 0; ++ ++err: ++ printk(KERN_ERR "WM9712 AC97 reset failed\n"); ++ return -EIO; ++} ++ ++static int wm9712_soc_suspend(struct platform_device *pdev, ++ pm_message_t state) ++{ ++ struct snd_soc_device *socdev = platform_get_drvdata(pdev); ++ struct snd_soc_codec *codec = socdev->codec; ++ ++ wm9712_dapm_event(codec, SNDRV_CTL_POWER_D3cold); ++ return 0; ++} ++ ++static int wm9712_soc_resume(struct platform_device *pdev) ++{ ++ struct snd_soc_device *socdev = platform_get_drvdata(pdev); ++ struct snd_soc_codec *codec = socdev->codec; ++ int i, ret; ++ u16 *cache = codec->reg_cache; ++ ++ ret = wm9712_reset(codec, 1); ++ if (ret < 0){ ++ printk(KERN_ERR "could not reset AC97 codec\n"); ++ return ret; ++ } ++ ++ wm9712_dapm_event(codec, SNDRV_CTL_POWER_D3hot); ++ ++ if (ret == 0) { ++ /* Sync reg_cache with the hardware after cold reset */ ++ for (i = 2; i < ARRAY_SIZE(wm9712_reg) << 1; i+=2) { ++ if (i == AC97_INT_PAGING || i == AC97_POWERDOWN || ++ (i > 0x58 && i != 0x5c)) ++ continue; ++ soc_ac97_ops.write(codec->ac97, i, cache[i>>1]); ++ } ++ } ++ ++ if (codec->suspend_dapm_state == SNDRV_CTL_POWER_D0) ++ wm9712_dapm_event(codec, SNDRV_CTL_POWER_D0); ++ ++ return ret; ++} ++ ++static int wm9712_soc_probe(struct platform_device *pdev) ++{ ++ struct snd_soc_device *socdev = platform_get_drvdata(pdev); ++ struct snd_soc_codec *codec; ++ int ret = 0; ++ ++ printk(KERN_INFO "WM9711/WM9712 SoC Audio Codec %s\n", WM9712_VERSION); ++ ++ socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); ++ if (socdev->codec == NULL) ++ return -ENOMEM; ++ codec = socdev->codec; ++ mutex_init(&codec->mutex); ++ ++ codec->reg_cache = ++ kzalloc(sizeof(u16) * ARRAY_SIZE(wm9712_reg), GFP_KERNEL); ++ if (codec->reg_cache == NULL) { ++ kfree(codec->ac97); ++ kfree(socdev->codec); ++ socdev->codec = NULL; ++ return -ENOMEM; ++ } ++ memcpy(codec->reg_cache, wm9712_reg, sizeof(u16) * ARRAY_SIZE(wm9712_reg)); ++ codec->reg_cache_size = sizeof(u16) * ARRAY_SIZE(wm9712_reg); ++ codec->reg_cache_step = 2; ++ ++ codec->name = "WM9712"; ++ codec->owner = THIS_MODULE; ++ codec->dai = wm9712_dai; ++ codec->num_dai = ARRAY_SIZE(wm9712_dai); ++ codec->write = ac97_write; ++ codec->read = ac97_read; ++ codec->dapm_event = wm9712_dapm_event; ++ INIT_LIST_HEAD(&codec->dapm_widgets); ++ INIT_LIST_HEAD(&codec->dapm_paths); ++ ++ ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); ++ if (ret < 0) ++ goto err; ++ ++ /* register pcms */ ++ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); ++ if (ret < 0) ++ goto pcm_err; ++ ++ ret = wm9712_reset(codec, 0); ++ if (ret < 0) { ++ printk(KERN_ERR "AC97 link error\n"); ++ goto reset_err; ++ } ++ ++ /* set alc mux to none */ ++ ac97_write(codec, AC97_VIDEO, ac97_read(codec, AC97_VIDEO) | 0x3000); ++ ++ wm9712_dapm_event(codec, SNDRV_CTL_POWER_D3hot); ++ wm9712_add_controls(codec); ++ wm9712_add_widgets(codec); ++ ret = snd_soc_register_card(socdev); ++ if (ret < 0) ++ goto reset_err; ++ ++ return 0; ++ ++reset_err: ++ snd_soc_free_pcms(socdev); ++ ++pcm_err: ++ snd_soc_free_ac97_codec(codec); ++ ++err: ++ kfree(socdev->codec->reg_cache); ++ kfree(socdev->codec); ++ socdev->codec = NULL; ++ return ret; ++} ++ ++static int wm9712_soc_remove(struct platform_device *pdev) ++{ ++ struct snd_soc_device *socdev = platform_get_drvdata(pdev); ++ struct snd_soc_codec *codec = socdev->codec; ++ ++ if (codec == NULL) ++ return 0; ++ ++ snd_soc_dapm_free(socdev); ++ snd_soc_free_pcms(socdev); ++ snd_soc_free_ac97_codec(codec); ++ kfree(codec->reg_cache); ++ kfree(codec); ++ return 0; ++} ++ ++struct snd_soc_codec_device soc_codec_dev_wm9712 = { ++ .probe = wm9712_soc_probe, ++ .remove = wm9712_soc_remove, ++ .suspend = wm9712_soc_suspend, ++ .resume = wm9712_soc_resume, ++}; ++ ++EXPORT_SYMBOL_GPL(soc_codec_dev_wm9712); ++ ++MODULE_DESCRIPTION("ASoC WM9711/WM9712 driver"); ++MODULE_AUTHOR("Liam Girdwood"); ++MODULE_LICENSE("GPL"); +Index: linux-2.6-pxa-new/sound/soc/codecs/wm9712.h +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/sound/soc/codecs/wm9712.h +@@ -0,0 +1,14 @@ ++/* ++ * wm9712.h -- WM9712 Soc Audio driver ++ */ ++ ++#ifndef _WM9712_H ++#define _WM9712_H ++ ++#define WM9712_DAI_AC97_HIFI 0 ++#define WM9712_DAI_AC97_AUX 1 ++ ++extern struct snd_soc_codec_dai wm9712_dai[2]; ++extern struct snd_soc_codec_device soc_codec_dev_wm9712; ++ ++#endif +Index: linux-2.6-pxa-new/sound/soc/codecs/wm9713.c +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/sound/soc/codecs/wm9713.c +@@ -0,0 +1,1313 @@ ++/* ++ * wm9713.c -- ALSA Soc WM9713 codec support ++ * ++ * Copyright 2006 Wolfson Microelectronics PLC. ++ * Author: Liam Girdwood ++ * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com ++ * ++ * This program is free software; you can redistribute it and/or modify it ++ * under the terms of the GNU General Public License as published by the ++ * Free Software Foundation; either version 2 of the License, or (at your ++ * option) any later version. ++ * ++ * Revision history ++ * 4th Feb 2006 Initial version. ++ * ++ * Features:- ++ * ++ * o Support for AC97 Codec, Voice DAC and Aux DAC ++ * o Support for DAPM ++ */ ++ ++#include <linux/init.h> ++#include <linux/module.h> ++#include <linux/device.h> ++#include <sound/driver.h> ++#include <sound/core.h> ++#include <sound/pcm.h> ++#include <sound/ac97_codec.h> ++#include <sound/initval.h> ++#include <sound/soc.h> ++#include <sound/soc-dapm.h> ++ ++#define WM9713_VERSION "0.12" ++ ++struct wm9713 { ++ u32 pll; /* current PLL frequency */ ++ u32 pll_resume; /* PLL resume frequency */ ++}; ++ ++static unsigned int ac97_read(struct snd_soc_codec *codec, ++ unsigned int reg); ++static int ac97_write(struct snd_soc_codec *codec, ++ unsigned int reg, unsigned int val); ++ ++#define AC97_DIR \ ++ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE) ++ ++#define AC97_RATES \ ++ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | \ ++ SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | \ ++ SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ ++ SNDRV_PCM_RATE_48000) ++ ++/* may need to expand this */ ++static struct snd_soc_dai_mode ac97_modes[] = { ++ { ++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S18_3LE, ++ .pcmrate = AC97_RATES, ++ }, ++}; ++ ++#define WM9713_VOICE_DAIFMT \ ++ (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J | \ ++ SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_DSP_A | \ ++ SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF | \ ++ SND_SOC_DAIFMT_NB_IF | SND_SOC_DAIFMT_IB_NF | \ ++ SND_SOC_DAIFMT_IB_IF) ++ ++#define WM9713_DIR \ ++ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE) ++ ++#define WM9713_VOICE_FSB \ ++ (SND_SOC_FSBD(1) | SND_SOC_FSBD(2) | SND_SOC_FSBD(4) | \ ++ SND_SOC_FSBD(8) | SND_SOC_FSBD(16)) ++ ++#define WM9713_VOICE_RATES \ ++ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 | \ ++ SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_48000 | \ ++ SNDRV_PCM_RATE_96000) ++ ++#define WM9713_HIFI_BITS \ ++ (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ ++ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) ++ ++/* ++ * Voice modes ++ */ ++static struct snd_soc_dai_mode wm9713_voice_modes[] = { ++ /* master modes */ ++ { ++ .fmt = WM9713_VOICE_DAIFMT | SND_SOC_DAIFMT_CBM_CFM | \ ++ SND_SOC_DAIFMT_CBM_CFS, ++ .pcmfmt = WM9713_HIFI_BITS, ++ .pcmrate = WM9713_VOICE_RATES, ++ .pcmdir = WM9713_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 256, ++ .bfs = WM9713_VOICE_FSB, ++ }, ++ ++ /* slave modes */ ++ { ++ .fmt = WM9713_VOICE_DAIFMT | SND_SOC_DAIFMT_CBS_CFS, ++ .pcmfmt = WM9713_HIFI_BITS, ++ .pcmrate = WM9713_VOICE_RATES, ++ .pcmdir = WM9713_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = SND_SOC_FS_ALL, ++ .bfs = SND_SOC_FSB_ALL, ++ }, ++}; ++ ++/* ++ * WM9713 register cache ++ * Reg 0x3c bit 15 is used by touch driver. ++ */ ++static const u16 wm9713_reg[] = { ++ 0x6174, 0x8080, 0x8080, 0x8080, // 6 ++ 0xc880, 0xe808, 0xe808, 0x0808, // e ++ 0x00da, 0x8000, 0xd600, 0xaaa0, // 16 ++ 0xaaa0, 0xaaa0, 0x0000, 0x0000, // 1e ++ 0x0f0f, 0x0040, 0x0000, 0x7f00, // 26 ++ 0x0405, 0x0410, 0xbb80, 0xbb80, // 2e ++ 0x0000, 0xbb80, 0x0000, 0x4523, // 36 ++ 0x0000, 0x2000, 0x7eff, 0xffff, // 3e ++ 0x0000, 0x0000, 0x0080, 0x0000, // 46 ++ 0x0000, 0x0000, 0xfffe, 0xffff, // 4e ++ 0x0000, 0x0000, 0x0000, 0xfffe, // 56 ++ 0x4000, 0x0000, 0x0000, 0x0000, // 5e ++ 0xb032, 0x3e00, 0x0000, 0x0000, // 66 ++ 0x0000, 0x0000, 0x0000, 0x0000, // 6e ++ 0x0000, 0x0000, 0x0000, 0x0006, // 76 ++ 0x0001, 0x0000, 0x574d, 0x4c13, // 7e ++ 0x0000, 0x0000, 0x0000 // virtual hp & mic mixers ++}; ++ ++/* virtual HP mixers regs */ ++#define HPL_MIXER 0x80 ++#define HPR_MIXER 0x82 ++#define MICB_MUX 0x82 ++ ++static const char *wm9713_mic_mixer[] = {"Stereo", "Mic 1", "Mic 2", "Mute"}; ++static const char *wm9713_rec_mux[] = {"Stereo", "Left", "Right", "Mute"}; ++static const char *wm9713_rec_src[] = ++ {"Mic 1", "Mic 2", "Line", "Mono In", "Headphone", "Speaker", ++ "Mono Out", "Zh"}; ++static const char *wm9713_rec_gain[] = {"+1.5dB Steps", "+0.75dB Steps"}; ++static const char *wm9713_alc_select[] = {"None", "Left", "Right", "Stereo"}; ++static const char *wm9713_mono_pga[] = {"Vmid", "Zh", "Mono", "Inv", ++ "Mono Vmid", "Inv Vmid"}; ++static const char *wm9713_spk_pga[] = ++ {"Vmid", "Zh", "Headphone", "Speaker", "Inv", "Headphone Vmid", ++ "Speaker Vmid", "Inv Vmid"}; ++static const char *wm9713_hp_pga[] = {"Vmid", "Zh", "Headphone", ++ "Headphone Vmid"}; ++static const char *wm9713_out3_pga[] = {"Vmid", "Zh", "Inv 1", "Inv 1 Vmid"}; ++static const char *wm9713_out4_pga[] = {"Vmid", "Zh", "Inv 2", "Inv 2 Vmid"}; ++static const char *wm9713_dac_inv[] = ++ {"Off", "Mono", "Speaker", "Left Headphone", "Right Headphone", ++ "Headphone Mono", "NC", "Vmid"}; ++static const char *wm9713_bass[] = {"Linear Control", "Adaptive Boost"}; ++static const char *wm9713_ng_type[] = {"Constant Gain", "Mute"}; ++static const char *wm9713_mic_select[] = {"Mic 1", "Mic 2 A", "Mic 2 B"}; ++static const char *wm9713_micb_select[] = {"MPB", "MPA"}; ++ ++static const struct soc_enum wm9713_enum[] = { ++SOC_ENUM_SINGLE(AC97_LINE, 3, 4, wm9713_mic_mixer), /* record mic mixer 0 */ ++SOC_ENUM_SINGLE(AC97_VIDEO, 14, 4, wm9713_rec_mux), /* record mux hp 1 */ ++SOC_ENUM_SINGLE(AC97_VIDEO, 9, 4, wm9713_rec_mux), /* record mux mono 2 */ ++SOC_ENUM_SINGLE(AC97_VIDEO, 3, 8, wm9713_rec_src), /* record mux left 3 */ ++SOC_ENUM_SINGLE(AC97_VIDEO, 0, 8, wm9713_rec_src), /* record mux right 4*/ ++SOC_ENUM_DOUBLE(AC97_CD, 14, 6, 2, wm9713_rec_gain), /* record step size 5 */ ++SOC_ENUM_SINGLE(AC97_PCI_SVID, 14, 4, wm9713_alc_select), /* alc source select 6*/ ++SOC_ENUM_SINGLE(AC97_REC_GAIN, 14, 4, wm9713_mono_pga), /* mono input select 7 */ ++SOC_ENUM_SINGLE(AC97_REC_GAIN, 11, 8, wm9713_spk_pga), /* speaker left input select 8 */ ++SOC_ENUM_SINGLE(AC97_REC_GAIN, 8, 8, wm9713_spk_pga), /* speaker right input select 9 */ ++SOC_ENUM_SINGLE(AC97_REC_GAIN, 6, 3, wm9713_hp_pga), /* headphone left input 10 */ ++SOC_ENUM_SINGLE(AC97_REC_GAIN, 4, 3, wm9713_hp_pga), /* headphone right input 11 */ ++SOC_ENUM_SINGLE(AC97_REC_GAIN, 2, 4, wm9713_out3_pga), /* out 3 source 12 */ ++SOC_ENUM_SINGLE(AC97_REC_GAIN, 0, 4, wm9713_out4_pga), /* out 4 source 13 */ ++SOC_ENUM_SINGLE(AC97_REC_GAIN_MIC, 13, 8, wm9713_dac_inv), /* dac invert 1 14 */ ++SOC_ENUM_SINGLE(AC97_REC_GAIN_MIC, 10, 8, wm9713_dac_inv), /* dac invert 2 15 */ ++SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 15, 2, wm9713_bass), /* bass control 16 */ ++SOC_ENUM_SINGLE(AC97_PCI_SVID, 5, 2, wm9713_ng_type), /* noise gate type 17 */ ++SOC_ENUM_SINGLE(AC97_3D_CONTROL, 12, 3, wm9713_mic_select), /* mic selection 18 */ ++SOC_ENUM_SINGLE(MICB_MUX, 0, 2, wm9713_micb_select), /* mic selection 19 */ ++}; ++ ++static const struct snd_kcontrol_new wm9713_snd_ac97_controls[] = { ++SOC_DOUBLE("Speaker Playback Volume", AC97_MASTER, 8, 0, 31, 1), ++SOC_DOUBLE("Speaker Playback Switch", AC97_MASTER, 15, 7, 1, 1), ++SOC_DOUBLE("Headphone Playback Volume", AC97_HEADPHONE, 8, 0, 31, 1), ++SOC_DOUBLE("Headphone Playback Switch", AC97_HEADPHONE,15, 7, 1, 1), ++SOC_DOUBLE("Line In Volume", AC97_PC_BEEP, 8, 0, 31, 1), ++SOC_DOUBLE("PCM Playback Volume", AC97_PHONE, 8, 0, 31, 1), ++SOC_SINGLE("Mic 1 Volume", AC97_MIC, 8, 31, 1), ++SOC_SINGLE("Mic 2 Volume", AC97_MIC, 0, 31, 1), ++ ++SOC_SINGLE("Mic Boost (+20dB) Switch", AC97_LINE, 5, 1, 0), ++SOC_SINGLE("Mic Headphone Mixer Volume", AC97_LINE, 0, 7, 1), ++ ++SOC_SINGLE("Capture Switch", AC97_CD, 15, 1, 1), ++SOC_ENUM("Capture Volume Steps", wm9713_enum[5]), ++SOC_DOUBLE("Capture Volume", AC97_CD, 8, 0, 63, 0), ++SOC_SINGLE("Capture ZC Switch", AC97_CD, 7, 1, 0), ++ ++SOC_SINGLE("Capture to Headphone Volume", AC97_VIDEO, 11, 7, 1), ++SOC_SINGLE("Capture to Mono Boost (+20dB) Switch", AC97_VIDEO, 8, 1, 0), ++SOC_SINGLE("Capture ADC Boost (+20dB) Switch", AC97_VIDEO, 6, 1, 0), ++ ++SOC_SINGLE("ALC Target Volume", AC97_CODEC_CLASS_REV, 12, 15, 0), ++SOC_SINGLE("ALC Hold Time", AC97_CODEC_CLASS_REV, 8, 15, 0), ++SOC_SINGLE("ALC Decay Time ", AC97_CODEC_CLASS_REV, 4, 15, 0), ++SOC_SINGLE("ALC Attack Time", AC97_CODEC_CLASS_REV, 0, 15, 0), ++SOC_ENUM("ALC Function", wm9713_enum[6]), ++SOC_SINGLE("ALC Max Volume", AC97_PCI_SVID, 11, 7, 0), ++SOC_SINGLE("ALC ZC Timeout", AC97_PCI_SVID, 9, 3, 0), ++SOC_SINGLE("ALC ZC Switch", AC97_PCI_SVID, 8, 1, 0), ++SOC_SINGLE("ALC NG Switch", AC97_PCI_SVID, 7, 1, 0), ++SOC_ENUM("ALC NG Type", wm9713_enum[17]), ++SOC_SINGLE("ALC NG Threshold", AC97_PCI_SVID, 0, 31, 0), ++ ++SOC_DOUBLE("Speaker Playback ZC Switch", AC97_MASTER, 14, 6, 1, 0), ++SOC_DOUBLE("Headphone Playback ZC Switch", AC97_HEADPHONE, 14, 6, 1, 0), ++ ++SOC_SINGLE("Out4 Playback Switch", AC97_MASTER_MONO, 15, 1, 1), ++SOC_SINGLE("Out4 Playback ZC Switch", AC97_MASTER_MONO, 14, 1, 0), ++SOC_SINGLE("Out4 Playback Volume", AC97_MASTER_MONO, 8, 63, 1), ++ ++SOC_SINGLE("Out3 Playback Switch", AC97_MASTER_MONO, 7, 1, 1), ++SOC_SINGLE("Out3 Playback ZC Switch", AC97_MASTER_MONO, 6, 1, 0), ++SOC_SINGLE("Out3 Playback Volume", AC97_MASTER_MONO, 0, 63, 1), ++ ++SOC_SINGLE("Mono Capture Volume", AC97_MASTER_TONE, 8, 31, 1), ++SOC_SINGLE("Mono Playback Switch", AC97_MASTER_TONE, 7, 1, 1), ++SOC_SINGLE("Mono Playback ZC Switch", AC97_MASTER_TONE, 6, 1, 0), ++SOC_SINGLE("Mono Playback Volume", AC97_MASTER_TONE, 0, 31, 1), ++ ++SOC_SINGLE("PC Beep Playback Headphone Volume", AC97_AUX, 12, 7, 1), ++SOC_SINGLE("PC Beep Playback Speaker Volume", AC97_AUX, 8, 7, 1), ++SOC_SINGLE("PC Beep Playback Mono Volume", AC97_AUX, 4, 7, 1), ++ ++SOC_SINGLE("Voice Playback Headphone Volume", AC97_PCM, 12, 7, 1), ++SOC_SINGLE("Voice Playback Master Volume", AC97_PCM, 8, 7, 1), ++SOC_SINGLE("Voice Playback Mono Volume", AC97_PCM, 4, 7, 1), ++ ++SOC_SINGLE("Aux Playback Headphone Volume", AC97_REC_SEL, 12, 7, 1), ++SOC_SINGLE("Aux Playback Master Volume", AC97_REC_SEL, 8, 7, 1), ++SOC_SINGLE("Aux Playback Mono Volume", AC97_REC_SEL, 4, 7, 1), ++ ++SOC_ENUM("Bass Control", wm9713_enum[16]), ++SOC_SINGLE("Bass Cut-off Switch", AC97_GENERAL_PURPOSE, 12, 1, 1), ++SOC_SINGLE("Tone Cut-off Switch", AC97_GENERAL_PURPOSE, 4, 1, 1), ++SOC_SINGLE("Playback Attenuate (-6dB) Switch", AC97_GENERAL_PURPOSE, 6, 1, 0), ++SOC_SINGLE("Bass Volume", AC97_GENERAL_PURPOSE, 8, 15, 1), ++SOC_SINGLE("Tone Volume", AC97_GENERAL_PURPOSE, 0, 15, 1), ++ ++SOC_SINGLE("3D Upper Cut-off Switch", AC97_REC_GAIN_MIC, 5, 1, 0), ++SOC_SINGLE("3D Lower Cut-off Switch", AC97_REC_GAIN_MIC, 4, 1, 0), ++SOC_SINGLE("3D Depth", AC97_REC_GAIN_MIC, 0, 15, 1), ++}; ++ ++/* add non dapm controls */ ++static int wm9713_add_controls(struct snd_soc_codec *codec) ++{ ++ int err, i; ++ ++ for (i = 0; i < ARRAY_SIZE(wm9713_snd_ac97_controls); i++) { ++ err = snd_ctl_add(codec->card, ++ snd_soc_cnew(&wm9713_snd_ac97_controls[i],codec, NULL)); ++ if (err < 0) ++ return err; ++ } ++ return 0; ++} ++ ++/* We have to create a fake left and right HP mixers because ++ * the codec only has a single control that is shared by both channels. ++ * This makes it impossible to determine the audio path using the current ++ * register map, thus we add a new (virtual) register to help determine the ++ * audio route within the device. ++ */ ++static int mixer_event (struct snd_soc_dapm_widget *w, int event) ++{ ++ u16 l, r, beep, tone, phone, rec, pcm, aux; ++ ++ l = ac97_read(w->codec, HPL_MIXER); ++ r = ac97_read(w->codec, HPR_MIXER); ++ beep = ac97_read(w->codec, AC97_PC_BEEP); ++ tone = ac97_read(w->codec, AC97_MASTER_TONE); ++ phone = ac97_read(w->codec, AC97_PHONE); ++ rec = ac97_read(w->codec, AC97_REC_SEL); ++ pcm = ac97_read(w->codec, AC97_PCM); ++ aux = ac97_read(w->codec, AC97_AUX); ++ ++ if (event & SND_SOC_DAPM_PRE_REG) ++ return 0; ++ if (l & 0x1 || r & 0x1) ++ ac97_write(w->codec, AC97_PC_BEEP, beep & 0x7fff); ++ else ++ ac97_write(w->codec, AC97_PC_BEEP, beep | 0x8000); ++ ++ if (l & 0x2 || r & 0x2) ++ ac97_write(w->codec, AC97_MASTER_TONE, tone & 0x7fff); ++ else ++ ac97_write(w->codec, AC97_MASTER_TONE, tone | 0x8000); ++ ++ if (l & 0x4 || r & 0x4) ++ ac97_write(w->codec, AC97_PHONE, phone & 0x7fff); ++ else ++ ac97_write(w->codec, AC97_PHONE, phone | 0x8000); ++ ++ if (l & 0x8 || r & 0x8) ++ ac97_write(w->codec, AC97_REC_SEL, rec & 0x7fff); ++ else ++ ac97_write(w->codec, AC97_REC_SEL, rec | 0x8000); ++ ++ if (l & 0x10 || r & 0x10) ++ ac97_write(w->codec, AC97_PCM, pcm & 0x7fff); ++ else ++ ac97_write(w->codec, AC97_PCM, pcm | 0x8000); ++ ++ if (l & 0x20 || r & 0x20) ++ ac97_write(w->codec, AC97_AUX, aux & 0x7fff); ++ else ++ ac97_write(w->codec, AC97_AUX, aux | 0x8000); ++ ++ return 0; ++} ++ ++/* Left Headphone Mixers */ ++static const struct snd_kcontrol_new wm9713_hpl_mixer_controls[] = { ++SOC_DAPM_SINGLE("PC Beep Playback Switch", HPL_MIXER, 5, 1, 0), ++SOC_DAPM_SINGLE("Voice Playback Switch", HPL_MIXER, 4, 1, 0), ++SOC_DAPM_SINGLE("Aux Playback Switch", HPL_MIXER, 3, 1, 0), ++SOC_DAPM_SINGLE("PCM Playback Switch", HPL_MIXER, 2, 1, 0), ++SOC_DAPM_SINGLE("MonoIn Playback Switch", HPL_MIXER, 1, 1, 0), ++SOC_DAPM_SINGLE("Bypass Playback Switch", HPL_MIXER, 0, 1, 0), ++}; ++ ++/* Right Headphone Mixers */ ++static const struct snd_kcontrol_new wm9713_hpr_mixer_controls[] = { ++SOC_DAPM_SINGLE("PC Beep Playback Switch", HPR_MIXER, 5, 1, 0), ++SOC_DAPM_SINGLE("Voice Playback Switch", HPR_MIXER, 4, 1, 0), ++SOC_DAPM_SINGLE("Aux Playback Switch", HPR_MIXER, 3, 1, 0), ++SOC_DAPM_SINGLE("PCM Playback Switch", HPR_MIXER, 2, 1, 0), ++SOC_DAPM_SINGLE("MonoIn Playback Switch", HPR_MIXER, 1, 1, 0), ++SOC_DAPM_SINGLE("Bypass Playback Switch", HPR_MIXER, 0, 1, 0), ++}; ++ ++/* headphone capture mux */ ++static const struct snd_kcontrol_new wm9713_hp_rec_mux_controls = ++SOC_DAPM_ENUM("Route", wm9713_enum[1]); ++ ++/* headphone mic mux */ ++static const struct snd_kcontrol_new wm9713_hp_mic_mux_controls = ++SOC_DAPM_ENUM("Route", wm9713_enum[0]); ++ ++/* Speaker Mixer */ ++static const struct snd_kcontrol_new wm9713_speaker_mixer_controls[] = { ++SOC_DAPM_SINGLE("PC Beep Playback Switch", AC97_AUX, 11, 1, 1), ++SOC_DAPM_SINGLE("Voice Playback Switch", AC97_PCM, 11, 1, 1), ++SOC_DAPM_SINGLE("Aux Playback Switch", AC97_REC_SEL, 11, 1, 1), ++SOC_DAPM_SINGLE("PCM Playback Switch", AC97_PHONE, 14, 1, 1), ++SOC_DAPM_SINGLE("MonoIn Playback Switch", AC97_MASTER_TONE, 14, 1, 1), ++SOC_DAPM_SINGLE("Bypass Playback Switch", AC97_PC_BEEP, 14, 1, 1), ++}; ++ ++/* Mono Mixer */ ++static const struct snd_kcontrol_new wm9713_mono_mixer_controls[] = { ++SOC_DAPM_SINGLE("PC Beep Playback Switch", AC97_AUX, 7, 1, 1), ++SOC_DAPM_SINGLE("Voice Playback Switch", AC97_PCM, 7, 1, 1), ++SOC_DAPM_SINGLE("Aux Playback Switch", AC97_REC_SEL, 7, 1, 1), ++SOC_DAPM_SINGLE("PCM Playback Switch", AC97_PHONE, 13, 1, 1), ++SOC_DAPM_SINGLE("MonoIn Playback Switch", AC97_MASTER_TONE, 13, 1, 1), ++SOC_DAPM_SINGLE("Bypass Playback Switch", AC97_PC_BEEP, 13, 1, 1), ++SOC_DAPM_SINGLE("Mic 1 Sidetone Switch", AC97_LINE, 7, 1, 1), ++SOC_DAPM_SINGLE("Mic 2 Sidetone Switch", AC97_LINE, 6, 1, 1), ++}; ++ ++/* mono mic mux */ ++static const struct snd_kcontrol_new wm9713_mono_mic_mux_controls = ++SOC_DAPM_ENUM("Route", wm9713_enum[2]); ++ ++/* mono output mux */ ++static const struct snd_kcontrol_new wm9713_mono_mux_controls = ++SOC_DAPM_ENUM("Route", wm9713_enum[7]); ++ ++/* speaker left output mux */ ++static const struct snd_kcontrol_new wm9713_hp_spkl_mux_controls = ++SOC_DAPM_ENUM("Route", wm9713_enum[8]); ++ ++/* speaker right output mux */ ++static const struct snd_kcontrol_new wm9713_hp_spkr_mux_controls = ++SOC_DAPM_ENUM("Route", wm9713_enum[9]); ++ ++/* headphone left output mux */ ++static const struct snd_kcontrol_new wm9713_hpl_out_mux_controls = ++SOC_DAPM_ENUM("Route", wm9713_enum[10]); ++ ++/* headphone right output mux */ ++static const struct snd_kcontrol_new wm9713_hpr_out_mux_controls = ++SOC_DAPM_ENUM("Route", wm9713_enum[11]); ++ ++/* Out3 mux */ ++static const struct snd_kcontrol_new wm9713_out3_mux_controls = ++SOC_DAPM_ENUM("Route", wm9713_enum[12]); ++ ++/* Out4 mux */ ++static const struct snd_kcontrol_new wm9713_out4_mux_controls = ++SOC_DAPM_ENUM("Route", wm9713_enum[13]); ++ ++/* DAC inv mux 1 */ ++static const struct snd_kcontrol_new wm9713_dac_inv1_mux_controls = ++SOC_DAPM_ENUM("Route", wm9713_enum[14]); ++ ++/* DAC inv mux 2 */ ++static const struct snd_kcontrol_new wm9713_dac_inv2_mux_controls = ++SOC_DAPM_ENUM("Route", wm9713_enum[15]); ++ ++/* Capture source left */ ++static const struct snd_kcontrol_new wm9713_rec_srcl_mux_controls = ++SOC_DAPM_ENUM("Route", wm9713_enum[3]); ++ ++/* Capture source right */ ++static const struct snd_kcontrol_new wm9713_rec_srcr_mux_controls = ++SOC_DAPM_ENUM("Route", wm9713_enum[4]); ++ ++/* mic source */ ++static const struct snd_kcontrol_new wm9713_mic_sel_mux_controls = ++SOC_DAPM_ENUM("Route", wm9713_enum[18]); ++ ++/* mic source B virtual control */ ++static const struct snd_kcontrol_new wm9713_micb_sel_mux_controls = ++SOC_DAPM_ENUM("Route", wm9713_enum[19]); ++ ++static const struct snd_soc_dapm_widget wm9713_dapm_widgets[] = { ++SND_SOC_DAPM_MUX("Capture Headphone Mux", SND_SOC_NOPM, 0, 0, ++ &wm9713_hp_rec_mux_controls), ++SND_SOC_DAPM_MUX("Sidetone Mux", SND_SOC_NOPM, 0, 0, ++ &wm9713_hp_mic_mux_controls), ++SND_SOC_DAPM_MUX("Capture Mono Mux", SND_SOC_NOPM, 0, 0, ++ &wm9713_mono_mic_mux_controls), ++SND_SOC_DAPM_MUX("Mono Out Mux", SND_SOC_NOPM, 0, 0, ++ &wm9713_mono_mux_controls), ++SND_SOC_DAPM_MUX("Left Speaker Out Mux", SND_SOC_NOPM, 0, 0, ++ &wm9713_hp_spkl_mux_controls), ++SND_SOC_DAPM_MUX("Right Speaker Out Mux", SND_SOC_NOPM, 0, 0, ++ &wm9713_hp_spkr_mux_controls), ++SND_SOC_DAPM_MUX("Left Headphone Out Mux", SND_SOC_NOPM, 0, 0, ++ &wm9713_hpl_out_mux_controls), ++SND_SOC_DAPM_MUX("Right Headphone Out Mux", SND_SOC_NOPM, 0, 0, ++ &wm9713_hpr_out_mux_controls), ++SND_SOC_DAPM_MUX("Out 3 Mux", SND_SOC_NOPM, 0, 0, ++ &wm9713_out3_mux_controls), ++SND_SOC_DAPM_MUX("Out 4 Mux", SND_SOC_NOPM, 0, 0, ++ &wm9713_out4_mux_controls), ++SND_SOC_DAPM_MUX("DAC Inv Mux 1", SND_SOC_NOPM, 0, 0, ++ &wm9713_dac_inv1_mux_controls), ++SND_SOC_DAPM_MUX("DAC Inv Mux 2", SND_SOC_NOPM, 0, 0, ++ &wm9713_dac_inv2_mux_controls), ++SND_SOC_DAPM_MUX("Left Capture Source", SND_SOC_NOPM, 0, 0, ++ &wm9713_rec_srcl_mux_controls), ++SND_SOC_DAPM_MUX("Right Capture Source", SND_SOC_NOPM, 0, 0, ++ &wm9713_rec_srcr_mux_controls), ++SND_SOC_DAPM_MUX("Mic A Source", SND_SOC_NOPM, 0, 0, ++ &wm9713_mic_sel_mux_controls ), ++SND_SOC_DAPM_MUX("Mic B Source", SND_SOC_NOPM, 0, 0, ++ &wm9713_micb_sel_mux_controls ), ++SND_SOC_DAPM_MIXER_E("Left HP Mixer", AC97_EXTENDED_MID, 3, 1, ++ &wm9713_hpl_mixer_controls[0], ARRAY_SIZE(wm9713_hpl_mixer_controls), ++ mixer_event, SND_SOC_DAPM_POST_REG), ++SND_SOC_DAPM_MIXER_E("Right HP Mixer", AC97_EXTENDED_MID, 2, 1, ++ &wm9713_hpr_mixer_controls[0], ARRAY_SIZE(wm9713_hpr_mixer_controls), ++ mixer_event, SND_SOC_DAPM_POST_REG), ++SND_SOC_DAPM_MIXER("Mono Mixer", AC97_EXTENDED_MID, 0, 1, ++ &wm9713_mono_mixer_controls[0], ARRAY_SIZE(wm9713_mono_mixer_controls)), ++SND_SOC_DAPM_MIXER("Speaker Mixer", AC97_EXTENDED_MID, 1, 1, ++ &wm9713_speaker_mixer_controls[0], ++ ARRAY_SIZE(wm9713_speaker_mixer_controls)), ++SND_SOC_DAPM_DAC("Left DAC", "Left HiFi Playback", AC97_EXTENDED_MID, 7, 1), ++SND_SOC_DAPM_DAC("Right DAC", "Right HiFi Playback", AC97_EXTENDED_MID, 6, 1), ++SND_SOC_DAPM_MIXER("AC97 Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), ++SND_SOC_DAPM_MIXER("HP Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), ++SND_SOC_DAPM_MIXER("Capture Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), ++SND_SOC_DAPM_DAC("Voice DAC", "Voice Playback", AC97_EXTENDED_MID, 12, 1), ++SND_SOC_DAPM_DAC("Aux DAC", "Aux Playback", AC97_EXTENDED_MID, 11, 1), ++SND_SOC_DAPM_ADC("Left ADC", "Left HiFi Capture", AC97_EXTENDED_MID, 5, 1), ++SND_SOC_DAPM_ADC("Right ADC", "Right HiFi Capture", AC97_EXTENDED_MID, 4, 1), ++SND_SOC_DAPM_PGA("Left Headphone", AC97_EXTENDED_MSTATUS, 10, 1, NULL, 0), ++SND_SOC_DAPM_PGA("Right Headphone", AC97_EXTENDED_MSTATUS, 9, 1, NULL, 0), ++SND_SOC_DAPM_PGA("Left Speaker", AC97_EXTENDED_MSTATUS, 8, 1, NULL, 0), ++SND_SOC_DAPM_PGA("Right Speaker", AC97_EXTENDED_MSTATUS, 7, 1, NULL, 0), ++SND_SOC_DAPM_PGA("Out 3", AC97_EXTENDED_MSTATUS, 11, 1, NULL, 0), ++SND_SOC_DAPM_PGA("Out 4", AC97_EXTENDED_MSTATUS, 12, 1, NULL, 0), ++SND_SOC_DAPM_PGA("Mono Out", AC97_EXTENDED_MSTATUS, 13, 1, NULL, 0), ++SND_SOC_DAPM_PGA("Left Line In", AC97_EXTENDED_MSTATUS, 6, 1, NULL, 0), ++SND_SOC_DAPM_PGA("Right Line In", AC97_EXTENDED_MSTATUS, 5, 1, NULL, 0), ++SND_SOC_DAPM_PGA("Mono In", AC97_EXTENDED_MSTATUS, 4, 1, NULL, 0), ++SND_SOC_DAPM_PGA("Mic A PGA", AC97_EXTENDED_MSTATUS, 3, 1, NULL, 0), ++SND_SOC_DAPM_PGA("Mic B PGA", AC97_EXTENDED_MSTATUS, 2, 1, NULL, 0), ++SND_SOC_DAPM_PGA("Mic A Pre Amp", AC97_EXTENDED_MSTATUS, 1, 1, NULL, 0), ++SND_SOC_DAPM_PGA("Mic B Pre Amp", AC97_EXTENDED_MSTATUS, 0, 1, NULL, 0), ++SND_SOC_DAPM_MICBIAS("Mic Bias", AC97_EXTENDED_MSTATUS, 14, 1), ++SND_SOC_DAPM_OUTPUT("MONO"), ++SND_SOC_DAPM_OUTPUT("HPL"), ++SND_SOC_DAPM_OUTPUT("HPR"), ++SND_SOC_DAPM_OUTPUT("SPKL"), ++SND_SOC_DAPM_OUTPUT("SPKR"), ++SND_SOC_DAPM_OUTPUT("OUT3"), ++SND_SOC_DAPM_OUTPUT("OUT4"), ++SND_SOC_DAPM_INPUT("LINEL"), ++SND_SOC_DAPM_INPUT("LINER"), ++SND_SOC_DAPM_INPUT("MONOIN"), ++SND_SOC_DAPM_INPUT("PCBEEP"), ++SND_SOC_DAPM_INPUT("MIC1"), ++SND_SOC_DAPM_INPUT("MIC2A"), ++SND_SOC_DAPM_INPUT("MIC2B"), ++SND_SOC_DAPM_VMID("VMID"), ++}; ++ ++static const char *audio_map[][3] = { ++ /* left HP mixer */ ++ {"Left HP Mixer", "PC Beep Playback Switch", "PCBEEP"}, ++ {"Left HP Mixer", "Voice Playback Switch", "Voice DAC"}, ++ {"Left HP Mixer", "Aux Playback Switch", "Aux DAC"}, ++ {"Left HP Mixer", "Bypass Playback Switch", "Left Line In"}, ++ {"Left HP Mixer", "PCM Playback Switch", "Left DAC"}, ++ {"Left HP Mixer", "MonoIn Playback Switch", "Mono In"}, ++ {"Left HP Mixer", NULL, "Capture Headphone Mux"}, ++ ++ /* right HP mixer */ ++ {"Right HP Mixer", "PC Beep Playback Switch", "PCBEEP"}, ++ {"Right HP Mixer", "Voice Playback Switch", "Voice DAC"}, ++ {"Right HP Mixer", "Aux Playback Switch", "Aux DAC"}, ++ {"Right HP Mixer", "Bypass Playback Switch", "Right Line In"}, ++ {"Right HP Mixer", "PCM Playback Switch", "Right DAC"}, ++ {"Right HP Mixer", "MonoIn Playback Switch", "Mono In"}, ++ {"Right HP Mixer", NULL, "Capture Headphone Mux"}, ++ ++ /* virtual mixer - mixes left & right channels for spk and mono */ ++ {"AC97 Mixer", NULL, "Left DAC"}, ++ {"AC97 Mixer", NULL, "Right DAC"}, ++ {"Line Mixer", NULL, "Right Line In"}, ++ {"Line Mixer", NULL, "Left Line In"}, ++ {"HP Mixer", NULL, "Left HP Mixer"}, ++ {"HP Mixer", NULL, "Right HP Mixer"}, ++ {"Capture Mixer", NULL, "Left Capture Source"}, ++ {"Capture Mixer", NULL, "Right Capture Source"}, ++ ++ /* speaker mixer */ ++ {"Speaker Mixer", "PC Beep Playback Switch", "PCBEEP"}, ++ {"Speaker Mixer", "Voice Playback Switch", "Voice DAC"}, ++ {"Speaker Mixer", "Aux Playback Switch", "Aux DAC"}, ++ {"Speaker Mixer", "Bypass Playback Switch", "Line Mixer"}, ++ {"Speaker Mixer", "PCM Playback Switch", "AC97 Mixer"}, ++ {"Speaker Mixer", "MonoIn Playback Switch", "Mono In"}, ++ ++ /* mono mixer */ ++ {"Mono Mixer", "PC Beep Playback Switch", "PCBEEP"}, ++ {"Mono Mixer", "Voice Playback Switch", "Voice DAC"}, ++ {"Mono Mixer", "Aux Playback Switch", "Aux DAC"}, ++ {"Mono Mixer", "Bypass Playback Switch", "Line Mixer"}, ++ {"Mono Mixer", "PCM Playback Switch", "AC97 Mixer"}, ++ {"Mono Mixer", NULL, "Capture Mono Mux"}, ++ ++ /* DAC inv mux 1 */ ++ {"DAC Inv Mux 1", "Mono", "Mono Mixer"}, ++ {"DAC Inv Mux 1", "Speaker", "Speaker Mixer"}, ++ {"DAC Inv Mux 1", "Left Headphone", "Left HP Mixer"}, ++ {"DAC Inv Mux 1", "Right Headphone", "Right HP Mixer"}, ++ {"DAC Inv Mux 1", "Headphone Mono", "HP Mixer"}, ++ ++ /* DAC inv mux 2 */ ++ {"DAC Inv Mux 2", "Mono", "Mono Mixer"}, ++ {"DAC Inv Mux 2", "Speaker", "Speaker Mixer"}, ++ {"DAC Inv Mux 2", "Left Headphone", "Left HP Mixer"}, ++ {"DAC Inv Mux 2", "Right Headphone", "Right HP Mixer"}, ++ {"DAC Inv Mux 2", "Headphone Mono", "HP Mixer"}, ++ ++ /* headphone left mux */ ++ {"Left Headphone Out Mux", "Headphone", "Left HP Mixer"}, ++ ++ /* headphone right mux */ ++ {"Right Headphone Out Mux", "Headphone", "Right HP Mixer"}, ++ ++ /* speaker left mux */ ++ {"Left Speaker Out Mux", "Headphone", "Left HP Mixer"}, ++ {"Left Speaker Out Mux", "Speaker", "Speaker Mixer"}, ++ {"Left Speaker Out Mux", "Inv", "DAC Inv Mux 1"}, ++ ++ /* speaker right mux */ ++ {"Right Speaker Out Mux", "Headphone", "Right HP Mixer"}, ++ {"Right Speaker Out Mux", "Speaker", "Speaker Mixer"}, ++ {"Right Speaker Out Mux", "Inv", "DAC Inv Mux 2"}, ++ ++ /* mono mux */ ++ {"Mono Out Mux", "Mono", "Mono Mixer"}, ++ {"Mono Out Mux", "Inv", "DAC Inv Mux 1"}, ++ ++ /* out 3 mux */ ++ {"Out 3 Mux", "Inv 1", "DAC Inv Mux 1"}, ++ ++ /* out 4 mux */ ++ {"Out 4 Mux", "Inv 2", "DAC Inv Mux 2"}, ++ ++ /* output pga */ ++ {"HPL", NULL, "Left Headphone"}, ++ {"Left Headphone", NULL, "Left Headphone Out Mux"}, ++ {"HPR", NULL, "Right Headphone"}, ++ {"Right Headphone", NULL, "Right Headphone Out Mux"}, ++ {"OUT3", NULL, "Out 3"}, ++ {"Out 3", NULL, "Out 3 Mux"}, ++ {"OUT4", NULL, "Out 4"}, ++ {"Out 4", NULL, "Out 4 Mux"}, ++ {"SPKL", NULL, "Left Speaker"}, ++ {"Left Speaker", NULL, "Left Speaker Out Mux"}, ++ {"SPKR", NULL, "Right Speaker"}, ++ {"Right Speaker", NULL, "Right Speaker Out Mux"}, ++ {"MONO", NULL, "Mono Out"}, ++ {"Mono Out", NULL, "Mono Out Mux"}, ++ ++ /* input pga */ ++ {"Left Line In", NULL, "LINEL"}, ++ {"Right Line In", NULL, "LINER"}, ++ {"Mono In", NULL, "MONOIN"}, ++ {"Mic A PGA", NULL, "Mic A Pre Amp"}, ++ {"Mic B PGA", NULL, "Mic B Pre Amp"}, ++ ++ /* left capture select */ ++ {"Left Capture Source", "Mic 1", "Mic A Pre Amp"}, ++ {"Left Capture Source", "Mic 2", "Mic B Pre Amp"}, ++ {"Left Capture Source", "Line", "LINEL"}, ++ {"Left Capture Source", "Mono In", "MONOIN"}, ++ {"Left Capture Source", "Headphone", "Left HP Mixer"}, ++ {"Left Capture Source", "Speaker", "Speaker Mixer"}, ++ {"Left Capture Source", "Mono Out", "Mono Mixer"}, ++ ++ /* right capture select */ ++ {"Right Capture Source", "Mic 1", "Mic A Pre Amp"}, ++ {"Right Capture Source", "Mic 2", "Mic B Pre Amp"}, ++ {"Right Capture Source", "Line", "LINER"}, ++ {"Right Capture Source", "Mono In", "MONOIN"}, ++ {"Right Capture Source", "Headphone", "Right HP Mixer"}, ++ {"Right Capture Source", "Speaker", "Speaker Mixer"}, ++ {"Right Capture Source", "Mono Out", "Mono Mixer"}, ++ ++ /* left ADC */ ++ {"Left ADC", NULL, "Left Capture Source"}, ++ ++ /* right ADC */ ++ {"Right ADC", NULL, "Right Capture Source"}, ++ ++ /* mic */ ++ {"Mic A Pre Amp", NULL, "Mic A Source"}, ++ {"Mic A Source", "Mic 1", "MIC1"}, ++ {"Mic A Source", "Mic 2 A", "MIC2A"}, ++ {"Mic A Source", "Mic 2 B", "Mic B Source"}, ++ {"Mic B Pre Amp", "MPB", "Mic B Source"}, ++ {"Mic B Source", NULL, "MIC2B"}, ++ ++ /* headphone capture */ ++ {"Capture Headphone Mux", "Stereo", "Capture Mixer"}, ++ {"Capture Headphone Mux", "Left", "Left Capture Source"}, ++ {"Capture Headphone Mux", "Right", "Right Capture Source"}, ++ ++ /* mono capture */ ++ {"Capture Mono Mux", "Stereo", "Capture Mixer"}, ++ {"Capture Mono Mux", "Left", "Left Capture Source"}, ++ {"Capture Mono Mux", "Right", "Right Capture Source"}, ++ ++ {NULL, NULL, NULL}, ++}; ++ ++static int wm9713_add_widgets(struct snd_soc_codec *codec) ++{ ++ int i; ++ ++ for(i = 0; i < ARRAY_SIZE(wm9713_dapm_widgets); i++) { ++ snd_soc_dapm_new_control(codec, &wm9713_dapm_widgets[i]); ++ } ++ ++ /* set up audio path audio_mapnects */ ++ for(i = 0; audio_map[i][0] != NULL; i++) { ++ snd_soc_dapm_connect_input(codec, audio_map[i][0], ++ audio_map[i][1], audio_map[i][2]); ++ } ++ ++ snd_soc_dapm_new_widgets(codec); ++ return 0; ++} ++ ++static unsigned int ac97_read(struct snd_soc_codec *codec, ++ unsigned int reg) ++{ ++ u16 *cache = codec->reg_cache; ++ ++ if (reg == AC97_RESET || reg == AC97_GPIO_STATUS || ++ reg == AC97_VENDOR_ID1 || reg == AC97_VENDOR_ID2 || ++ reg == AC97_CD) ++ return soc_ac97_ops.read(codec->ac97, reg); ++ else { ++ reg = reg >> 1; ++ ++ if (reg > (ARRAY_SIZE(wm9713_reg))) ++ return -EIO; ++ ++ return cache[reg]; ++ } ++} ++ ++static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, ++ unsigned int val) ++{ ++ u16 *cache = codec->reg_cache; ++ if (reg < 0x7c) ++ soc_ac97_ops.write(codec->ac97, reg, val); ++ reg = reg >> 1; ++ if (reg <= (ARRAY_SIZE(wm9713_reg))) ++ cache[reg] = val; ++ ++ return 0; ++} ++ ++struct pll_ { ++ unsigned int in_hz; ++ unsigned int lf:1; /* allows low frequency use */ ++ unsigned int sdm:1; /* allows fraction n div */ ++ unsigned int divsel:1; /* enables input clock div */ ++ unsigned int divctl:1; /* input clock divider */ ++ unsigned int n:4; ++ unsigned int k; ++}; ++ ++struct pll_ pll[] = { ++ {13000000, 0, 1, 0, 0, 7, 0x23f488}, ++ {2048000, 1, 0, 0, 0, 12, 0x0}, ++ {4096000, 1, 0, 0, 0, 6, 0x0}, ++ {12288000, 0, 0, 0, 0, 8, 0x0}, ++ /* liam - add more entries */ ++}; ++ ++/* we must have either 24.576MHz or a PLL freq */ ++static unsigned int wm9713_config_ac97sysclk(struct snd_soc_codec_dai *dai, ++ struct snd_soc_clock_info *info, unsigned int clk) ++{ ++ int i; ++ dai->mclk = 0; ++ ++ /* first check if we can get away witout burning any PLL power */ ++ if (24576000 == clk) { ++ /* standard AC97 clock */ ++ dai->mclk = clk; ++ goto out; ++ } ++ ++ /* ok no standard clock, so we must now try the PLL */ ++ for(i = 0; i < ARRAY_SIZE(pll); i++) { ++ if (clk == pll[i].in_hz) { ++ dai->mclk = clk; /* clock out */ ++ goto out; ++ } ++ } ++ ++out: ++ return dai->mclk; ++} ++ ++/* The WM9713 voice DAC can only run at 256FS. This interface and DAC are ++ * clocked by the main AC97 clock divided down to 256 FS. ++ */ ++static unsigned int wm9713_config_vsysclk(struct snd_soc_codec_dai *dai, ++ struct snd_soc_clock_info *info, unsigned int clk) ++{ ++ ++ int i, j, best_clk = info->fs * info->rate; ++ ++ /* can we run at this clk without the PLL ? */ ++ for (i = 1; i <= 16; i++) { ++ if (best_clk * i == clk) { ++ dai->pll_in = 0; ++ dai->clk_div = i << 1; ++ dai->mclk = best_clk; ++ return dai->mclk; ++ } ++ } ++ ++ /* now check for PLL support */ ++ for (i = 0; i < ARRAY_SIZE(pll); i++) { ++ if (pll[i].in_hz == clk) { ++ for (j = 1; j <= 16; j++) { ++ if (24576000 == j * best_clk) { ++ dai->pll_in = clk; ++ dai->pll_out = 24576000; ++ dai->clk_div = j << 1; ++ dai->mclk = best_clk; ++ return dai->mclk; ++ } ++ } ++ } ++ } ++ ++ /* this clk is not supported */ ++ return 0; ++} ++ ++u32 wm9713_set_pll(struct snd_soc_codec *codec, u32 in) ++{ ++ struct wm9713 *wm = (struct wm9713*)codec->private_data; ++ int i; ++ u16 reg, reg2; ++ ++ /* turn PLL off ? */ ++ if (in == 0) { ++ /* disable PLL power and select ext source */ ++ reg = ac97_read(codec, AC97_HANDSET_RATE); ++ ac97_write(codec, AC97_HANDSET_RATE, reg | 0x0080); ++ reg = ac97_read(codec, AC97_EXTENDED_MID); ++ ac97_write(codec, AC97_EXTENDED_MID, reg | 0x0200); ++ wm->pll = 0; ++ return 0; ++ } ++ ++ for (i = 0; i < ARRAY_SIZE(pll); i++) { ++ if (pll[i].in_hz == in) ++ goto found; ++ } ++ return -EINVAL; ++ ++found: ++ if (pll[i].sdm == 0) { ++ reg = (pll[i].n << 12) | (pll[i].lf << 11) | ++ (pll[i].divsel << 9) | (pll[i].divctl << 8); ++ ac97_write(codec, AC97_LINE1_LEVEL, reg); ++ } else { ++ /* write the fractional k to the reg 0x46 pages */ ++ reg2 = (pll[i].n << 12) | (pll[i].lf << 11) | (pll[i].sdm << 10) | ++ (pll[i].divsel << 9) | (pll[i].divctl << 8); ++ ++ reg = reg2 | (0x5 << 4) | (pll[i].k >> 20); /* K [21:20] */ ++ ac97_write(codec, AC97_LINE1_LEVEL, reg); ++ ++ reg = reg2 | (0x4 << 4) | ((pll[i].k >> 16) & 0xf); /* K [19:16] */ ++ ac97_write(codec, AC97_LINE1_LEVEL, reg); ++ ++ reg = reg2 | (0x3 << 4) | ((pll[i].k >> 12) & 0xf); /* K [15:12] */ ++ ac97_write(codec, AC97_LINE1_LEVEL, reg); ++ ++ reg = reg2 | (0x2 << 4) | ((pll[i].k >> 8) & 0xf); /* K [11:8] */ ++ ac97_write(codec, AC97_LINE1_LEVEL, reg); ++ ++ reg = reg2 | (0x1 << 4) | ((pll[i].k >> 4) & 0xf); /* K [7:4] */ ++ ac97_write(codec, AC97_LINE1_LEVEL, reg); ++ ++ reg = reg2 | (0x0 << 4) | (pll[i].k & 0xf); /* K [3:0] */ ++ ac97_write(codec, AC97_LINE1_LEVEL, reg); ++ } ++ ++ /* turn PLL on and select as source */ ++ reg = ac97_read(codec, AC97_EXTENDED_MID); ++ ac97_write(codec, AC97_EXTENDED_MID, reg & 0xfdff); ++ reg = ac97_read(codec, AC97_HANDSET_RATE); ++ ac97_write(codec, AC97_HANDSET_RATE, reg & 0xff7f); ++ /* wait 10ms AC97 link frames for the link to stabilise */ ++ schedule_timeout_interruptible(msecs_to_jiffies(10)); ++ wm->pll = in; ++ return 0; ++} ++EXPORT_SYMBOL_GPL(wm9713_set_pll); ++ ++static int wm9713_voice_prepare(struct snd_pcm_substream *substream) ++{ ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ struct snd_soc_device *socdev = rtd->socdev; ++ struct snd_soc_codec *codec = socdev->codec; ++ u16 reg = 0x8000, bfs, div, gpio; ++ ++ bfs = SND_SOC_FSBD_REAL(rtd->codec_dai->dai_runtime.bfs); ++ gpio = ac97_read(codec, AC97_GPIO_CFG) & 0xffe2; ++ ++ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_CLOCK_MASK){ ++ case SND_SOC_DAIFMT_CBM_CFM: ++ reg |= 0x4000; ++ gpio |= 0x0008; ++ break; ++ case SND_SOC_DAIFMT_CBM_CFS: ++ reg |= 0x6000; ++ gpio |= 0x000c; ++ break; ++ case SND_SOC_DAIFMT_CBS_CFS: ++ reg |= 0x0200; ++ gpio |= 0x000d; ++ break; ++ case SND_SOC_DAIFMT_CBS_CFM: ++ gpio |= 0x0009; ++ break; ++ } ++ ac97_write(codec, AC97_GPIO_CFG, gpio); ++ ++ /* enable PLL if needed */ ++ if (rtd->codec_dai->pll_in) ++ wm9713_set_pll(codec, rtd->codec_dai->pll_in); ++ ++ /* set the PCM divider */ ++ div = ac97_read(codec, AC97_HANDSET_RATE) & 0xf0ff; ++ ac97_write(codec, AC97_HANDSET_RATE, div | ++ ((rtd->codec_dai->clk_div >> 1) -1) << 8); ++ ++ /* clock inversion */ ++ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_INV_MASK) { ++ case SND_SOC_DAIFMT_IB_IF: ++ reg |= 0x00c0; ++ break; ++ case SND_SOC_DAIFMT_IB_NF: ++ reg |= 0x0080; ++ break; ++ case SND_SOC_DAIFMT_NB_IF: ++ reg |= 0x0040; ++ break; ++ } ++ ++ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_FORMAT_MASK) { ++ case SND_SOC_DAIFMT_I2S: ++ reg |= 0x0002; ++ break; ++ case SND_SOC_DAIFMT_RIGHT_J: ++ break; ++ case SND_SOC_DAIFMT_LEFT_J: ++ reg |= 0x0001; ++ break; ++ case SND_SOC_DAIFMT_DSP_A: ++ reg |= 0x0003; ++ break; ++ case SND_SOC_DAIFMT_DSP_B: ++ reg |= 0x0043; ++ break; ++ } ++ ++ switch (rtd->codec_dai->dai_runtime.pcmfmt) { ++ case SNDRV_PCM_FMTBIT_S16_LE: ++ break; ++ case SNDRV_PCM_FMTBIT_S20_3LE: ++ reg |= 0x0004; ++ break; ++ case SNDRV_PCM_FMTBIT_S24_LE: ++ reg |= 0x0008; ++ break; ++ case SNDRV_PCM_FMTBIT_S32_LE: ++ reg |= 0x000c; ++ break; ++ } ++ ++ switch (bfs) { ++ case 2: ++ reg |= (0x1 << 9); ++ break; ++ case 4: ++ reg |= (0x2 << 9); ++ break; ++ case 8: ++ reg |= (0x3 << 9); ++ break; ++ case 16: ++ reg |= (0x4 << 9); ++ break; ++ } ++ ++ /* enable PCM interface in master mode */ ++ ac97_write(codec, AC97_CENTER_LFE_MASTER, reg); ++ return 0; ++} ++ ++static void wm9713_shutdown(struct snd_pcm_substream *substream) ++{ ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ struct snd_soc_device *socdev = rtd->socdev; ++ struct snd_soc_codec *codec = socdev->codec; ++ ++ if (!codec->active) ++ wm9713_set_pll(codec, 0); ++} ++ ++static void wm9713_voiceshutdown(snd_pcm_substream_t *substream) ++{ ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ struct snd_soc_device *socdev = rtd->socdev; ++ struct snd_soc_codec *codec = socdev->codec; ++ u16 status; ++ ++ wm9713_shutdown(substream); ++ ++ /* Gracefully shut down the voice interface. */ ++ status = ac97_read(codec, AC97_EXTENDED_STATUS) | 0x1000; ++ ac97_write(codec,AC97_HANDSET_RATE,0x0280); ++ schedule_timeout_interruptible(msecs_to_jiffies(1)); ++ ac97_write(codec,AC97_HANDSET_RATE,0x0F80); ++ ac97_write(codec,AC97_EXTENDED_MID,status); ++} ++ ++static int ac97_hifi_prepare(struct snd_pcm_substream *substream) ++{ ++ struct snd_pcm_runtime *runtime = substream->runtime; ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ struct snd_soc_device *socdev = rtd->socdev; ++ struct snd_soc_codec *codec = socdev->codec; ++ int reg; ++ u16 vra; ++ ++ /* we need a 24576000Hz clock to run at the correct speed */ ++ if (rtd->codec_dai->mclk != 24576000) ++ wm9713_set_pll(codec, rtd->codec_dai->mclk); ++ ++ vra = ac97_read(codec, AC97_EXTENDED_STATUS); ++ ac97_write(codec, AC97_EXTENDED_STATUS, vra | 0x1); ++ ++ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ++ reg = AC97_PCM_FRONT_DAC_RATE; ++ else ++ reg = AC97_PCM_LR_ADC_RATE; ++ ++ return ac97_write(codec, reg, runtime->rate); ++} ++ ++static int ac97_aux_prepare(struct snd_pcm_substream *substream) ++{ ++ struct snd_pcm_runtime *runtime = substream->runtime; ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ struct snd_soc_device *socdev = rtd->socdev; ++ struct snd_soc_codec *codec = socdev->codec; ++ u16 vra, xsle; ++ ++ /* we need a 24576000Hz clock to run at the correct speed */ ++ if (rtd->codec_dai->mclk != 24576000) ++ wm9713_set_pll(codec, rtd->codec_dai->mclk); ++ ++ vra = ac97_read(codec, AC97_EXTENDED_STATUS); ++ ac97_write(codec, AC97_EXTENDED_STATUS, vra | 0x1); ++ xsle = ac97_read(codec, AC97_PCI_SID); ++ ac97_write(codec, AC97_PCI_SID, xsle | 0x8000); ++ ++ if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) ++ return -ENODEV; ++ ++ return ac97_write(codec, AC97_PCM_SURR_DAC_RATE, runtime->rate); ++} ++ ++struct snd_soc_codec_dai wm9713_dai[] = { ++{ ++ .name = "AC97 HiFi", ++ .playback = { ++ .stream_name = "HiFi Playback", ++ .channels_min = 1, ++ .channels_max = 2,}, ++ .capture = { ++ .stream_name = "HiFi Capture", ++ .channels_min = 1, ++ .channels_max = 2,}, ++ .config_sysclk = wm9713_config_ac97sysclk, ++ .ops = { ++ .shutdown = wm9713_shutdown, ++ .prepare = ac97_hifi_prepare,}, ++ .caps = { ++ .num_modes = ARRAY_SIZE(ac97_modes), ++ .mode = ac97_modes,},}, ++ { ++ .name = "AC97 Aux", ++ .playback = { ++ .stream_name = "Aux Playback", ++ .channels_min = 1, ++ .channels_max = 1,}, ++ .config_sysclk = wm9713_config_ac97sysclk, ++ .ops = { ++ .shutdown = wm9713_shutdown, ++ .prepare = ac97_aux_prepare,}, ++ .caps = { ++ .num_modes = ARRAY_SIZE(ac97_modes), ++ .mode = ac97_modes,} ++ }, ++ { ++ .name = "WM9713 Voice", ++ .playback = { ++ .stream_name = "Voice Playback", ++ .channels_min = 1, ++ .channels_max = 1,}, ++ .capture = { ++ .stream_name = "Voice Capture", ++ .channels_min = 1, ++ .channels_max = 2,}, ++ .config_sysclk = wm9713_config_vsysclk, ++ .ops = { ++ .prepare = wm9713_voice_prepare, ++ .shutdown = wm9713_voiceshutdown,}, ++ .caps = { ++ .num_modes = ARRAY_SIZE(wm9713_voice_modes), ++ .mode = wm9713_voice_modes,}, ++ }, ++}; ++EXPORT_SYMBOL_GPL(wm9713_dai); ++ ++int wm9713_reset(struct snd_soc_codec *codec, int try_warm) ++{ ++ if (try_warm && soc_ac97_ops.warm_reset) { ++ soc_ac97_ops.warm_reset(codec->ac97); ++ if (!(ac97_read(codec, 0) & 0x8000)) ++ return 1; ++ } ++ ++ soc_ac97_ops.reset(codec->ac97); ++ if (ac97_read(codec, 0) & 0x8000) ++ return -EIO; ++ return 0; ++} ++EXPORT_SYMBOL_GPL(wm9713_reset); ++ ++static int wm9713_dapm_event(struct snd_soc_codec *codec, int event) ++{ ++ u16 reg; ++ ++ switch (event) { ++ case SNDRV_CTL_POWER_D0: /* full On */ ++ /* enable thermal shutdown */ ++ reg = ac97_read(codec, AC97_EXTENDED_MID) & 0x1bff; ++ ac97_write(codec, AC97_EXTENDED_MID, reg); ++ break; ++ case SNDRV_CTL_POWER_D1: /* partial On */ ++ case SNDRV_CTL_POWER_D2: /* partial On */ ++ break; ++ case SNDRV_CTL_POWER_D3hot: /* Off, with power */ ++ /* enable master bias and vmid */ ++ reg = ac97_read(codec, AC97_EXTENDED_MID) & 0x3bff; ++ ac97_write(codec, AC97_EXTENDED_MID, reg); ++ ac97_write(codec, AC97_POWERDOWN, 0x0000); ++ break; ++ case SNDRV_CTL_POWER_D3cold: /* Off, without power */ ++ /* disable everything including AC link */ ++ ac97_write(codec, AC97_EXTENDED_MID, 0xffff); ++ ac97_write(codec, AC97_EXTENDED_MSTATUS, 0xffff); ++ ac97_write(codec, AC97_POWERDOWN, 0xffff); ++ break; ++ } ++ codec->dapm_state = event; ++ return 0; ++} ++ ++static int wm9713_soc_suspend(struct platform_device *pdev, ++ pm_message_t state) ++{ ++ struct snd_soc_device *socdev = platform_get_drvdata(pdev); ++ struct snd_soc_codec *codec = socdev->codec; ++ struct wm9713 *wm = (struct wm9713*)codec->private_data; ++ ++ if (wm->pll) { ++ wm->pll_resume = wm->pll; ++ wm9713_set_pll(codec, 0); ++ } ++ wm9713_dapm_event(codec, SNDRV_CTL_POWER_D3cold); ++ return 0; ++} ++ ++static int wm9713_soc_resume(struct platform_device *pdev) ++{ ++ struct snd_soc_device *socdev = platform_get_drvdata(pdev); ++ struct snd_soc_codec *codec = socdev->codec; ++ struct wm9713 *wm = (struct wm9713*)codec->private_data; ++ int i, ret; ++ u16 *cache = codec->reg_cache; ++ ++ if ((ret = wm9713_reset(codec, 1)) < 0){ ++ printk(KERN_ERR "could not reset AC97 codec\n"); ++ return ret; ++ } ++ ++ wm9713_dapm_event(codec, SNDRV_CTL_POWER_D3hot); ++ ++ /* only synchronise the codec if warm reset failed */ ++ if (ret == 0) { ++ for (i = 2; i < ARRAY_SIZE(wm9713_reg) << 1; i+=2) { ++ if (i == AC97_POWERDOWN || i == AC97_EXTENDED_MID || ++ i == AC97_EXTENDED_MSTATUS || i > 0x66) ++ continue; ++ soc_ac97_ops.write(codec->ac97, i, cache[i>>1]); ++ } ++ } ++ ++ if (wm->pll_resume) { ++ wm9713_set_pll(codec, wm->pll_resume); ++ wm->pll_resume = 0; ++ } ++ ++ if (codec->suspend_dapm_state == SNDRV_CTL_POWER_D0) ++ wm9713_dapm_event(codec, SNDRV_CTL_POWER_D0); ++ ++ return ret; ++} ++ ++static int wm9713_soc_probe(struct platform_device *pdev) ++{ ++ struct snd_soc_device *socdev = platform_get_drvdata(pdev); ++ struct snd_soc_codec *codec; ++ int ret = 0, reg; ++ ++ printk(KERN_INFO "WM9713/WM9714 SoC Audio Codec %s\n", WM9713_VERSION); ++ ++ socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); ++ if (socdev->codec == NULL) ++ return -ENOMEM; ++ codec = socdev->codec; ++ mutex_init(&codec->mutex); ++ ++ codec->reg_cache = ++ kzalloc(sizeof(u16) * ARRAY_SIZE(wm9713_reg), GFP_KERNEL); ++ if (codec->reg_cache == NULL){ ++ kfree(socdev->codec); ++ socdev->codec = NULL; ++ return -ENOMEM; ++ } ++ memcpy(codec->reg_cache, wm9713_reg, ++ sizeof(u16) * ARRAY_SIZE(wm9713_reg)); ++ codec->reg_cache_size = sizeof(u16) * ARRAY_SIZE(wm9713_reg); ++ codec->reg_cache_step = 2; ++ ++ codec->private_data = kzalloc(sizeof(struct wm9713), GFP_KERNEL); ++ if (codec->private_data == NULL) { ++ kfree(codec->reg_cache); ++ kfree(socdev->codec); ++ socdev->codec = NULL; ++ return -ENOMEM; ++ } ++ ++ codec->name = "WM9713"; ++ codec->owner = THIS_MODULE; ++ codec->dai = wm9713_dai; ++ codec->num_dai = ARRAY_SIZE(wm9713_dai); ++ codec->write = ac97_write; ++ codec->read = ac97_read; ++ codec->dapm_event = wm9713_dapm_event; ++ INIT_LIST_HEAD(&codec->dapm_widgets); ++ INIT_LIST_HEAD(&codec->dapm_paths); ++ ++ ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); ++ if (ret < 0) ++ goto err; ++ ++ /* register pcms */ ++ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); ++ if (ret < 0) ++ goto pcm_err; ++ ++ /* do a cold reset for the controller and then try ++ * a warm reset followed by an optional cold reset for codec */ ++ wm9713_reset(codec, 0); ++ ret = wm9713_reset(codec, 1); ++ if (ret < 0) { ++ printk(KERN_ERR "AC97 link error\n"); ++ goto reset_err; ++ } ++ ++ wm9713_dapm_event(codec, SNDRV_CTL_POWER_D3hot); ++ ++ /* unmute the adc - move to kcontrol */ ++ reg = ac97_read(codec, AC97_CD) & 0x7fff; ++ ac97_write(codec, AC97_CD, reg); ++ ++ wm9713_add_controls(codec); ++ wm9713_add_widgets(codec); ++ ret = snd_soc_register_card(socdev); ++ if (ret < 0) ++ goto reset_err; ++ return 0; ++ ++reset_err: ++ snd_soc_free_pcms(socdev); ++ ++pcm_err: ++ snd_soc_free_ac97_codec(codec); ++ ++err: ++ kfree(socdev->codec->private_data); ++ kfree(socdev->codec->reg_cache); ++ kfree(socdev->codec); ++ socdev->codec = NULL; ++ return ret; ++} ++ ++static int wm9713_soc_remove(struct platform_device *pdev) ++{ ++ struct snd_soc_device *socdev = platform_get_drvdata(pdev); ++ struct snd_soc_codec *codec = socdev->codec; ++ ++ if (codec == NULL) ++ return 0; ++ ++ snd_soc_dapm_free(socdev); ++ snd_soc_free_pcms(socdev); ++ snd_soc_free_ac97_codec(codec); ++ kfree(codec->private_data); ++ kfree(codec->reg_cache); ++ kfree(codec); ++ return 0; ++} ++ ++struct snd_soc_codec_device soc_codec_dev_wm9713= { ++ .probe = wm9713_soc_probe, ++ .remove = wm9713_soc_remove, ++ .suspend = wm9713_soc_suspend, ++ .resume = wm9713_soc_resume, ++}; ++ ++EXPORT_SYMBOL_GPL(soc_codec_dev_wm9713); ++ ++MODULE_DESCRIPTION("ASoC WM9713/WM9714 driver"); ++MODULE_AUTHOR("Liam Girdwood"); ++MODULE_LICENSE("GPL"); +Index: linux-2.6-pxa-new/sound/soc/codecs/wm9713.h +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/sound/soc/codecs/wm9713.h +@@ -0,0 +1,18 @@ ++/* ++ * wm9713.h -- WM9713 Soc Audio driver ++ */ ++ ++#ifndef _WM9713_H ++#define _WM9713_H ++ ++#define WM9713_DAI_AC97_HIFI 0 ++#define WM9713_DAI_AC97_AUX 1 ++#define WM9713_DAI_PCM_VOICE 2 ++ ++extern struct snd_soc_codec_device soc_codec_dev_wm9713; ++extern struct snd_soc_codec_dai wm9713_dai[3]; ++ ++u32 wm9713_set_pll(struct snd_soc_codec *codec, u32 in); ++int wm9713_reset(struct snd_soc_codec *codec, int try_warm); ++ ++#endif +Index: linux-2.6-pxa-new/sound/soc/pxa/Kconfig +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/sound/soc/pxa/Kconfig +@@ -0,0 +1,125 @@ ++menu "SoC Audio for the Intel PXA2xx" ++ ++config SND_PXA2xx_SOC ++ tristate "SoC Audio for the Intel PXA2xx chip" ++ depends on ARCH_PXA && SND ++ select SND_PCM ++ help ++ Say Y or M if you want to add support for codecs attached to ++ the PXA2xx AC97, I2S or SSP interface. You will also need ++ to select the audio interfaces to support below. ++ ++config SND_PXA2xx_AC97 ++ tristate ++ select SND_AC97_CODEC ++ ++config SND_PXA2xx_SOC_AC97 ++ tristate ++ select SND_AC97_BUS ++ select SND_SOC_AC97_BUS ++ ++config SND_PXA2xx_SOC_I2S ++ tristate ++ ++config SND_PXA2xx_SOC_SSP ++ tristate ++ select PXA_SSP ++ ++config SND_PXA2xx_SOC_MAINSTONE ++ tristate "SoC AC97 Audio support for Intel Mainstone" ++ depends on SND_PXA2xx_SOC && MACH_MAINSTONE ++ select SND_PXA2xx_AC97 ++ help ++ Say Y if you want to add support for generic AC97 SoC audio on Mainstone. ++ ++config SND_PXA2xx_SOC_MAINSTONE_WM8731 ++ tristate "SoC I2S Audio support for Intel Mainstone - WM8731" ++ depends on SND_PXA2xx_SOC && MACH_MAINSTONE ++ select SND_PXA2xx_SOC_I2S ++ help ++ Say Y if you want to add support for SoC audio on Mainstone ++ with the WM8731. ++ ++config SND_PXA2xx_SOC_MAINSTONE_WM8753 ++ tristate "SoC I2S/SSP Audio support for Intel Mainstone - WM8753" ++ depends on SND_PXA2xx_SOC && MACH_MAINSTONE ++ select SND_PXA2xx_SOC_I2S ++ select SND_PXA2xx_SOC_SSP ++ help ++ Say Y if you want to add support for SoC audio on Mainstone ++ with the WM8753. ++ ++config SND_PXA2xx_SOC_MAINSTONE_WM8974 ++ tristate "SoC I2S Audio support for Intel Mainstone - WM8974" ++ depends on SND_PXA2xx_SOC && MACH_MAINSTONE ++ select SND_PXA2xx_SOC_I2S ++ help ++ Say Y if you want to add support for SoC audio on Mainstone ++ with the WM8974. ++ ++config SND_PXA2xx_SOC_MAINSTONE_WM9713 ++ tristate "SoC I2S/SSP Audio support for Intel Mainstone - WM9713" ++ depends on SND_PXA2xx_SOC && MACH_MAINSTONE ++ select SND_PXA2xx_SOC_AC97 ++ select SND_PXA2xx_SOC_SSP ++ help ++ Say Y if you want to add support for SoC voice audio on Mainstone ++ with the WM9713. ++ ++config SND_MAINSTONE_BASEBAND ++ tristate "Example SoC Baseband Audio support for Intel Mainstone" ++ depends on SND_PXA2xx_SOC && MACH_MAINSTONE ++ select SND_PXA2xx_SOC_AC97 ++ help ++ Say Y if you want to add support for SoC baseband on Mainstone ++ with the WM9713 and example Baseband modem. ++ ++config SND_MAINSTONE_BLUETOOTH ++ tristate "Example SoC Bluetooth Audio support for Intel Mainstone" ++ depends on SND_PXA2xx_SOC && MACH_MAINSTONE ++ select SND_PXA2xx_SOC_I2S ++ help ++ Say Y if you want to add support for SoC bluetooth on Mainstone ++ with the WM8753 and example Bluetooth codec. ++ ++config SND_PXA2xx_SOC_MAINSTONE_WM9712 ++ tristate "SoC I2S/SSP Audio support for Intel Mainstone - WM9712" ++ depends on SND_PXA2xx_SOC && MACH_MAINSTONE ++ select SND_PXA2xx_SOC_AC97 ++ help ++ Say Y if you want to add support for SoC voice audio on Mainstone ++ with the WM9712. ++ ++config SND_PXA2xx_SOC_CORGI ++ tristate "SoC Audio support for Sharp Zaurus SL-C7x0" ++ depends on SND_PXA2xx_SOC && PXA_SHARP_C7xx ++ select SND_PXA2xx_SOC_I2S ++ help ++ Say Y if you want to add support for SoC audio on Sharp ++ Zaurus SL-C7x0 models (Corgi, Shepherd, Husky). ++ ++config SND_PXA2xx_SOC_SPITZ ++ tristate "SoC Audio support for Sharp Zaurus SL-Cxx00" ++ depends on SND_PXA2xx_SOC && PXA_SHARP_Cxx00 ++ select SND_PXA2xx_SOC_I2S ++ help ++ Say Y if you want to add support for SoC audio on Sharp ++ Zaurus SL-Cxx00 models (Spitz, Borzoi and Akita). ++ ++config SND_PXA2xx_SOC_POODLE ++ tristate "SoC Audio support for Poodle" ++ depends on SND_PXA2xx_SOC && MACH_POODLE ++ select SND_PXA2xx_SOC_I2S ++ help ++ Say Y if you want to add support for SoC audio on Sharp ++ Zaurus SL-5600 model (Poodle). ++ ++config SND_PXA2xx_SOC_TOSA ++ tristate "SoC AC97 Audio support for Tosa" ++ depends on SND_PXA2xx_SOC && MACH_TOSA ++ select SND_PXA2xx_SOC_AC97 ++ help ++ Say Y if you want to add support for SoC audio on Sharp ++ Zaurus SL-C6000x models (Tosa). ++ ++endmenu +Index: linux-2.6-pxa-new/sound/soc/pxa/Makefile +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/sound/soc/pxa/Makefile +@@ -0,0 +1,36 @@ ++# PXA Platform Support ++snd-soc-pxa2xx-objs := pxa2xx-pcm.o ++snd-soc-pxa2xx-ac97-objs := pxa2xx-ac97.o ++snd-soc-pxa2xx-i2s-objs := pxa2xx-i2s.o ++snd-soc-pxa2xx-ssp-objs := pxa2xx-ssp.o ++ ++obj-$(CONFIG_SND_PXA2xx_SOC) += snd-soc-pxa2xx.o ++obj-$(CONFIG_SND_PXA2xx_SOC_AC97) += snd-soc-pxa2xx-ac97.o ++obj-$(CONFIG_SND_PXA2xx_SOC_I2S) += snd-soc-pxa2xx-i2s.o ++obj-$(CONFIG_SND_PXA2xx_SOC_SSP) += snd-soc-pxa2xx-ssp.o ++ ++# PXA Machine Support ++snd-soc-corgi-objs := corgi.o ++snd-soc-mainstone-wm8731-objs := mainstone_wm8731.o ++snd-soc-mainstone-wm8753-objs := mainstone_wm8753.o ++snd-soc-mainstone-wm8974-objs := mainstone_wm8974.o ++snd-soc-mainstone-wm9713-objs := mainstone_wm9713.o ++snd-soc-mainstone-wm9712-objs := mainstone_wm9712.o ++snd-soc-mainstone-baseband-objs := mainstone_baseband.o ++snd-soc-mainstone-bluetooth-objs := mainstone_bluetooth.o ++snd-soc-poodle-objs := poodle.o ++snd-soc-tosa-objs := tosa.o ++snd-soc-spitz-objs := spitz.o ++ ++obj-$(CONFIG_SND_PXA2xx_SOC_CORGI) += snd-soc-corgi.o ++obj-$(CONFIG_SND_PXA2xx_SOC_MAINSTONE_WM8731) += snd-soc-mainstone-wm8731.o ++obj-$(CONFIG_SND_PXA2xx_SOC_MAINSTONE_WM8753) += snd-soc-mainstone-wm8753.o ++obj-$(CONFIG_SND_PXA2xx_SOC_MAINSTONE_WM8974) += snd-soc-mainstone-wm8974.o ++obj-$(CONFIG_SND_PXA2xx_SOC_MAINSTONE_WM9713) += snd-soc-mainstone-wm9713.o ++obj-$(CONFIG_SND_PXA2xx_SOC_MAINSTONE_WM9712) += snd-soc-mainstone-wm9712.o ++obj-$(CONFIG_SND_MAINSTONE_BASEBAND) += snd-soc-mainstone-baseband.o ++obj-$(CONFIG_SND_MAINSTONE_BLUETOOTH) += snd-soc-mainstone-bluetooth.o ++obj-$(CONFIG_SND_PXA2xx_SOC_POODLE) += snd-soc-poodle.o ++obj-$(CONFIG_SND_PXA2xx_SOC_TOSA) += snd-soc-tosa.o ++obj-$(CONFIG_SND_PXA2xx_SOC_SPITZ) += snd-soc-spitz.o ++ +Index: linux-2.6-pxa-new/sound/soc/pxa/corgi.c +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/sound/soc/pxa/corgi.c +@@ -0,0 +1,361 @@ ++/* ++ * corgi.c -- SoC audio for Corgi ++ * ++ * Copyright 2005 Wolfson Microelectronics PLC. ++ * Copyright 2005 Openedhand Ltd. ++ * ++ * Authors: Liam Girdwood <liam.girdwood@wolfsonmicro.com> ++ * Richard Purdie <richard@openedhand.com> ++ * ++ * This program is free software; you can redistribute it and/or modify it ++ * under the terms of the GNU General Public License as published by the ++ * Free Software Foundation; either version 2 of the License, or (at your ++ * option) any later version. ++ * ++ * Revision history ++ * 30th Nov 2005 Initial version. ++ * ++ */ ++ ++#include <linux/module.h> ++#include <linux/moduleparam.h> ++#include <linux/timer.h> ++#include <linux/interrupt.h> ++#include <linux/platform_device.h> ++#include <sound/driver.h> ++#include <sound/core.h> ++#include <sound/pcm.h> ++#include <sound/soc.h> ++#include <sound/soc-dapm.h> ++ ++#include <asm/mach-types.h> ++#include <asm/hardware/scoop.h> ++#include <asm/arch/pxa-regs.h> ++#include <asm/arch/hardware.h> ++#include <asm/arch/corgi.h> ++#include <asm/arch/audio.h> ++ ++#include "../codecs/wm8731.h" ++#include "pxa2xx-pcm.h" ++ ++#define CORGI_HP 0 ++#define CORGI_MIC 1 ++#define CORGI_LINE 2 ++#define CORGI_HEADSET 3 ++#define CORGI_HP_OFF 4 ++#define CORGI_SPK_ON 0 ++#define CORGI_SPK_OFF 1 ++ ++ /* audio clock in Hz - rounded from 12.235MHz */ ++#define CORGI_AUDIO_CLOCK 12288000 ++ ++static int corgi_jack_func; ++static int corgi_spk_func; ++ ++static void corgi_ext_control(struct snd_soc_codec *codec) ++{ ++ int spk = 0, mic = 0, line = 0, hp = 0, hs = 0; ++ ++ /* set up jack connection */ ++ switch (corgi_jack_func) { ++ case CORGI_HP: ++ hp = 1; ++ /* set = unmute headphone */ ++ set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L); ++ set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R); ++ break; ++ case CORGI_MIC: ++ mic = 1; ++ /* reset = mute headphone */ ++ reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L); ++ reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R); ++ break; ++ case CORGI_LINE: ++ line = 1; ++ reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L); ++ reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R); ++ break; ++ case CORGI_HEADSET: ++ hs = 1; ++ mic = 1; ++ reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L); ++ set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R); ++ break; ++ } ++ ++ if (corgi_spk_func == CORGI_SPK_ON) ++ spk = 1; ++ ++ /* set the enpoints to their new connetion states */ ++ snd_soc_dapm_set_endpoint(codec, "Ext Spk", spk); ++ snd_soc_dapm_set_endpoint(codec, "Mic Jack", mic); ++ snd_soc_dapm_set_endpoint(codec, "Line Jack", line); ++ snd_soc_dapm_set_endpoint(codec, "Headphone Jack", hp); ++ snd_soc_dapm_set_endpoint(codec, "Headset Jack", hs); ++ ++ /* signal a DAPM event */ ++ snd_soc_dapm_sync_endpoints(codec); ++} ++ ++static int corgi_startup(struct snd_pcm_substream *substream) ++{ ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ struct snd_soc_codec *codec = rtd->socdev->codec; ++ ++ /* check the jack status at stream startup */ ++ corgi_ext_control(codec); ++ return 0; ++} ++ ++/* we need to unmute the HP at shutdown as the mute burns power on corgi */ ++static int corgi_shutdown(struct snd_pcm_substream *substream) ++{ ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ struct snd_soc_codec *codec = rtd->socdev->codec; ++ ++ /* set = unmute headphone */ ++ set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_L); ++ set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MUTE_R); ++ return 0; ++} ++ ++static struct snd_soc_ops corgi_ops = { ++ .startup = corgi_startup, ++ .shutdown = corgi_shutdown, ++}; ++ ++static int corgi_get_jack(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_value *ucontrol) ++{ ++ ucontrol->value.integer.value[0] = corgi_jack_func; ++ return 0; ++} ++ ++static int corgi_set_jack(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_value *ucontrol) ++{ ++ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); ++ ++ if (corgi_jack_func == ucontrol->value.integer.value[0]) ++ return 0; ++ ++ corgi_jack_func = ucontrol->value.integer.value[0]; ++ corgi_ext_control(codec); ++ return 1; ++} ++ ++static int corgi_get_spk(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_value *ucontrol) ++{ ++ ucontrol->value.integer.value[0] = corgi_spk_func; ++ return 0; ++} ++ ++static int corgi_set_spk(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_value *ucontrol) ++{ ++ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); ++ ++ if (corgi_spk_func == ucontrol->value.integer.value[0]) ++ return 0; ++ ++ corgi_spk_func = ucontrol->value.integer.value[0]; ++ corgi_ext_control(codec); ++ return 1; ++} ++ ++static int corgi_amp_event(struct snd_soc_dapm_widget *w, int event) ++{ ++ if (SND_SOC_DAPM_EVENT_ON(event)) ++ set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_APM_ON); ++ else ++ reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_APM_ON); ++ ++ return 0; ++} ++ ++static int corgi_mic_event(struct snd_soc_dapm_widget *w, int event) ++{ ++ if (SND_SOC_DAPM_EVENT_ON(event)) ++ set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MIC_BIAS); ++ else ++ reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MIC_BIAS); ++ ++ return 0; ++} ++ ++/* corgi machine dapm widgets */ ++static const struct snd_soc_dapm_widget wm8731_dapm_widgets[] = { ++SND_SOC_DAPM_HP("Headphone Jack", NULL), ++SND_SOC_DAPM_MIC("Mic Jack", corgi_mic_event), ++SND_SOC_DAPM_SPK("Ext Spk", corgi_amp_event), ++SND_SOC_DAPM_LINE("Line Jack", NULL), ++SND_SOC_DAPM_HP("Headset Jack", NULL), ++}; ++ ++/* Corgi machine audio map (connections to the codec pins) */ ++static const char *audio_map[][3] = { ++ ++ /* headset Jack - in = micin, out = LHPOUT*/ ++ {"Headset Jack", NULL, "LHPOUT"}, ++ ++ /* headphone connected to LHPOUT1, RHPOUT1 */ ++ {"Headphone Jack", NULL, "LHPOUT"}, ++ {"Headphone Jack", NULL, "RHPOUT"}, ++ ++ /* speaker connected to LOUT, ROUT */ ++ {"Ext Spk", NULL, "ROUT"}, ++ {"Ext Spk", NULL, "LOUT"}, ++ ++ /* mic is connected to MICIN (via right channel of headphone jack) */ ++ {"MICIN", NULL, "Mic Jack"}, ++ ++ /* Same as the above but no mic bias for line signals */ ++ {"MICIN", NULL, "Line Jack"}, ++ ++ {NULL, NULL, NULL}, ++}; ++ ++static const char *jack_function[] = {"Headphone", "Mic", "Line", "Headset", ++ "Off"}; ++static const char *spk_function[] = {"On", "Off"}; ++static const struct soc_enum corgi_enum[] = { ++ SOC_ENUM_SINGLE_EXT(5, jack_function), ++ SOC_ENUM_SINGLE_EXT(2, spk_function), ++}; ++ ++static const struct snd_kcontrol_new wm8731_corgi_controls[] = { ++ SOC_ENUM_EXT("Jack Function", corgi_enum[0], corgi_get_jack, ++ corgi_set_jack), ++ SOC_ENUM_EXT("Speaker Function", corgi_enum[1], corgi_get_spk, ++ corgi_set_spk), ++}; ++ ++/* ++ * Logic for a wm8731 as connected on a Sharp SL-C7x0 Device ++ */ ++static int corgi_wm8731_init(struct snd_soc_codec *codec) ++{ ++ int i, err; ++ ++ snd_soc_dapm_set_endpoint(codec, "LLINEIN", 0); ++ snd_soc_dapm_set_endpoint(codec, "RLINEIN", 0); ++ ++ /* Add corgi specific controls */ ++ for (i = 0; i < ARRAY_SIZE(wm8731_corgi_controls); i++) { ++ err = snd_ctl_add(codec->card, ++ snd_soc_cnew(&wm8731_corgi_controls[i],codec, NULL)); ++ if (err < 0) ++ return err; ++ } ++ ++ /* Add corgi specific widgets */ ++ for(i = 0; i < ARRAY_SIZE(wm8731_dapm_widgets); i++) { ++ snd_soc_dapm_new_control(codec, &wm8731_dapm_widgets[i]); ++ } ++ ++ /* Set up corgi specific audio path audio_map */ ++ for(i = 0; audio_map[i][0] != NULL; i++) { ++ snd_soc_dapm_connect_input(codec, audio_map[i][0], ++ audio_map[i][1], audio_map[i][2]); ++ } ++ ++ snd_soc_dapm_sync_endpoints(codec); ++ return 0; ++} ++ ++static unsigned int corgi_config_sysclk(struct snd_soc_pcm_runtime *rtd, ++ struct snd_soc_clock_info *info) ++{ ++ if (info->bclk_master & SND_SOC_DAIFMT_CBS_CFS) { ++ /* pxa2xx is i2s master */ ++ switch (info->rate) { ++ case 44100: ++ case 88200: ++ /* configure codec digital filters for 44.1, 88.2 */ ++ rtd->codec_dai->config_sysclk(rtd->codec_dai, info, ++ 11289600); ++ break; ++ default: ++ /* configure codec digital filters for all other rates */ ++ rtd->codec_dai->config_sysclk(rtd->codec_dai, info, ++ CORGI_AUDIO_CLOCK); ++ break; ++ } ++ /* config pxa i2s as master */ ++ return rtd->cpu_dai->config_sysclk(rtd->cpu_dai, info, ++ CORGI_AUDIO_CLOCK); ++ } else { ++ /* codec is i2s master - ++ * only configure codec DAI clock and filters */ ++ return rtd->codec_dai->config_sysclk(rtd->codec_dai, info, ++ CORGI_AUDIO_CLOCK); ++ } ++} ++ ++/* corgi digital audio interface glue - connects codec <--> CPU */ ++static struct snd_soc_dai_link corgi_dai = { ++ .name = "WM8731", ++ .stream_name = "WM8731", ++ .cpu_dai = &pxa_i2s_dai, ++ .codec_dai = &wm8731_dai, ++ .init = corgi_wm8731_init, ++ .config_sysclk = corgi_config_sysclk, ++}; ++ ++/* corgi audio machine driver */ ++static struct snd_soc_machine snd_soc_machine_corgi = { ++ .name = "Corgi", ++ .dai_link = &corgi_dai, ++ .num_links = 1, ++ .ops = &corgi_ops, ++}; ++ ++/* corgi audio private data */ ++static struct wm8731_setup_data corgi_wm8731_setup = { ++ .i2c_address = 0x1b, ++}; ++ ++/* corgi audio subsystem */ ++static struct snd_soc_device corgi_snd_devdata = { ++ .machine = &snd_soc_machine_corgi, ++ .platform = &pxa2xx_soc_platform, ++ .codec_dev = &soc_codec_dev_wm8731, ++ .codec_data = &corgi_wm8731_setup, ++}; ++ ++static struct platform_device *corgi_snd_device; ++ ++static int __init corgi_init(void) ++{ ++ int ret; ++ ++ if (!(machine_is_corgi() || machine_is_shepherd() || machine_is_husky())) ++ return -ENODEV; ++ ++ corgi_snd_device = platform_device_alloc("soc-audio", -1); ++ if (!corgi_snd_device) ++ return -ENOMEM; ++ ++ platform_set_drvdata(corgi_snd_device, &corgi_snd_devdata); ++ corgi_snd_devdata.dev = &corgi_snd_device->dev; ++ ret = platform_device_add(corgi_snd_device); ++ ++ if (ret) ++ platform_device_put(corgi_snd_device); ++ ++ return ret; ++} ++ ++static void __exit corgi_exit(void) ++{ ++ platform_device_unregister(corgi_snd_device); ++} ++ ++module_init(corgi_init); ++module_exit(corgi_exit); ++ ++/* Module information */ ++MODULE_AUTHOR("Richard Purdie"); ++MODULE_DESCRIPTION("ALSA SoC Corgi"); ++MODULE_LICENSE("GPL"); +Index: linux-2.6-pxa-new/sound/soc/pxa/mainstone.c +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/sound/soc/pxa/mainstone.c +@@ -0,0 +1,126 @@ ++/* ++ * mainstone.c -- SoC audio for Mainstone ++ * ++ * Copyright 2005 Wolfson Microelectronics PLC. ++ * Author: Liam Girdwood ++ * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com ++ * ++ * Mainstone audio amplifier code taken from arch/arm/mach-pxa/mainstone.c ++ * Copyright: MontaVista Software Inc. ++ * ++ * This program is free software; you can redistribute it and/or modify it ++ * under the terms of the GNU General Public License as published by the ++ * Free Software Foundation; either version 2 of the License, or (at your ++ * option) any later version. ++ * ++ * Revision history ++ * 30th Oct 2005 Initial version. ++ * ++ */ ++ ++#include <linux/module.h> ++#include <linux/moduleparam.h> ++#include <linux/device.h> ++#include <linux/i2c.h> ++#include <sound/driver.h> ++#include <sound/core.h> ++#include <sound/pcm.h> ++#include <sound/soc.h> ++#include <sound/soc-dapm.h> ++ ++#include <asm/arch/pxa-regs.h> ++#include <asm/arch/mainstone.h> ++#include <asm/arch/audio.h> ++ ++#include "../codecs/ac97.h" ++#include "pxa2xx-pcm.h" ++ ++static struct snd_soc_machine mainstone; ++static long mst_audio_suspend_mask; ++ ++static int mainstone_suspend(struct platform_device *pdev, pm_message_t state) ++{ ++ mst_audio_suspend_mask = MST_MSCWR2; ++ MST_MSCWR2 |= MST_MSCWR2_AC97_SPKROFF; ++ return 0; ++} ++ ++static int mainstone_resume(struct platform_device *pdev) ++{ ++ MST_MSCWR2 &= mst_audio_suspend_mask | ~MST_MSCWR2_AC97_SPKROFF; ++ return 0; ++} ++ ++static int mainstone_probe(struct platform_device *pdev) ++{ ++ MST_MSCWR2 &= ~MST_MSCWR2_AC97_SPKROFF; ++ return 0; ++} ++ ++static int mainstone_remove(struct platform_device *pdev) ++{ ++ MST_MSCWR2 |= MST_MSCWR2_AC97_SPKROFF; ++ return 0; ++} ++ ++static struct snd_soc_machine_config codecs[] = { ++{ ++ .name = "AC97", ++ .sname = "AC97 HiFi", ++ .iface = &pxa_ac97_interface[0], ++}, ++{ ++ .name = "AC97 Aux", ++ .sname = "AC97 Aux", ++ .iface = &pxa_ac97_interface[1], ++}, ++}; ++ ++static struct snd_soc_machine mainstone = { ++ .name = "Mainstone", ++ .probe = mainstone_probe, ++ .remove = mainstone_remove, ++ .suspend_pre = mainstone_suspend, ++ .resume_post = mainstone_resume, ++ .config = codecs, ++ .nconfigs = ARRAY_SIZE(codecs), ++}; ++ ++static struct snd_soc_device mainstone_snd_devdata = { ++ .machine = &mainstone, ++ .platform = &pxa2xx_soc_platform, ++ .codec_dev = &soc_codec_dev_ac97, ++}; ++ ++static struct platform_device *mainstone_snd_device; ++ ++static int __init mainstone_init(void) ++{ ++ int ret; ++ ++ mainstone_snd_device = platform_device_alloc("soc-audio", -1); ++ if (!mainstone_snd_device) ++ return -ENOMEM; ++ ++ platform_set_drvdata(mainstone_snd_device, &mainstone_snd_devdata); ++ mainstone_snd_devdata.dev = &mainstone_snd_device->dev; ++ ret = platform_device_add(mainstone_snd_device); ++ ++ if (ret) ++ platform_device_put(mainstone_snd_device); ++ ++ return ret; ++} ++ ++static void __exit mainstone_exit(void) ++{ ++ platform_device_unregister(mainstone_snd_device); ++} ++ ++module_init(mainstone_init); ++module_exit(mainstone_exit); ++ ++/* Module information */ ++MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com"); ++MODULE_DESCRIPTION("ALSA SoC Mainstone"); ++MODULE_LICENSE("GPL"); +Index: linux-2.6-pxa-new/sound/soc/pxa/mainstone_baseband.c +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/sound/soc/pxa/mainstone_baseband.c +@@ -0,0 +1,249 @@ ++/* ++ * mainstone_baseband.c ++ * Mainstone Example Baseband modem -- ALSA Soc Audio Layer ++ * ++ * Copyright 2006 Wolfson Microelectronics PLC. ++ * Author: Liam Girdwood ++ * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com ++ * ++ * This program is free software; you can redistribute it and/or modify it ++ * under the terms of the GNU General Public License as published by the ++ * Free Software Foundation; either version 2 of the License, or (at your ++ * option) any later version. ++ * ++ * Revision history ++ * 15th Apr 2006 Initial version. ++ * ++ * This is example code to demonstrate connecting a baseband modem to the PCM ++ * DAI on the WM9713 codec on the Intel Mainstone platform. It is by no means ++ * complete as it requires code to control the modem. ++ * ++ * The architecture consists of the WM9713 AC97 DAI connected to the PXA27x ++ * AC97 controller and the WM9713 PCM DAI connected to the basebands DAI. The ++ * baseband is controlled via a serial port. Audio is routed between the PXA27x ++ * and the baseband via internal WM9713 analog paths. ++ * ++ * This driver is not the baseband modem driver. This driver only calls ++ * functions from the Baseband driver to set up it's PCM DAI. ++ * ++ * It's intended to use this driver as follows:- ++ * ++ * 1. open() WM9713 PCM audio device. ++ * 2. open() serial device (for AT commands). ++ * 3. configure PCM audio device (rate etc) - sets up WM9713 PCM DAI, ++ * this will also set up the baseband PCM DAI (via calling baseband driver). ++ * 4. send any further AT commands to set up baseband. ++ * 5. configure codec audio mixer paths. ++ * 6. open(), configure and read/write AC97 audio device - to Tx/Rx voice ++ * ++ * The PCM audio device is opened but IO is never performed on it as the IO is ++ * directly between the codec and the baseband (and not the CPU). ++ * ++ * TODO: ++ * o Implement callbacks ++ */ ++ ++#include <linux/init.h> ++#include <linux/module.h> ++#include <linux/platform_device.h> ++ ++#include <sound/driver.h> ++#include <sound/core.h> ++#include <sound/pcm.h> ++#include <sound/soc.h> ++#include <sound/soc-dapm.h> ++ ++#include <asm/hardware.h> ++#include <asm/arch/pxa-regs.h> ++#include <asm/arch/audio.h> ++#include <asm/arch/ssp.h> ++ ++#include "../codecs/wm9713.h" ++#include "pxa2xx-pcm.h" ++ ++static struct snd_soc_machine mainstone; ++ ++#define BASEBAND_XXX_DAIFMT \ ++ (SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBS_CFS |\ ++ SND_SOC_DAIFMT_NB_NF) ++ ++#define BASEBAND_XXX_DIR \ ++ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE) ++ ++/* ++ * PCM modes - 8k 16bit mono baseband modem is master ++ */ ++static struct snd_soc_dai_mode mainstone_example_modes[] = { ++ /* port master clk & frame modes */ ++ {BASEBAND_XXX_DAIFMT, SND_SOC_DAITDM_LRDW(0,0), SNDRV_PCM_FORMAT_S16_LE, ++ SNDRV_PCM_RATE_8000, BASEBAND_XXX_DIR, SND_SOC_DAI_BFS_RATE, 256, 64}, ++}; ++ ++/* Do specific baseband PCM voice startup here */ ++static int mainstone_baseband_startup(struct snd_pcm_substream *substream) ++{ ++ return 0; ++} ++ ++/* Do specific baseband PCM voice shutdown here */ ++static void mainstone_baseband_shutdown (struct snd_pcm_substream *substream) ++{ ++} ++ ++/* Do specific baseband modem PCM voice hw params init here */ ++static int mainstone_baseband_hw_params(struct snd_pcm_substream *substream, ++ struct snd_pcm_hw_params *params) ++{ ++ return 0; ++} ++ ++/* Do specific baseband modem PCM voice hw params free here */ ++static int mainstone_baseband_hw_free(struct snd_pcm_substream *substream) ++{ ++ return 0; ++} ++ ++static struct snd_soc_cpu_dai mainstone_example_dai[] = { ++ { .name = "Baseband", ++ .id = 0, ++ .type = SND_SOC_DAI_PCM, ++ .playback = { ++ .channels_min = 1, ++ .channels_max = 1,}, ++ .capture = { ++ .channels_min = 1, ++ .channels_max = 1,}, ++ .ops = { ++ .startup = mainstone_baseband_startup, ++ .shutdown = mainstone_baseband_shutdown, ++ .hw_params = mainstone_baseband_hw_params, ++ .hw_free = mainstone_baseband_hw_free, ++ }, ++ .caps = { ++ .mode = mainstone_example_modes, ++ .num_modes = ARRAY_SIZE(mainstone_example_modes),}, ++ }, ++}; ++ ++/* do we need to do any thing on the mainstone when the stream is ++ * started and stopped ++ */ ++static int mainstone_startup(struct snd_pcm_substream *substream) ++{ ++ return 0; ++} ++ ++static void mainstone_shutdown(struct snd_pcm_substream *substream) ++{ ++} ++ ++static struct snd_soc_ops mainstone_ops = { ++ .startup = mainstone_startup, ++ .shutdown = mainstone_shutdown, ++}; ++ ++/* PM */ ++static int mainstone_suspend(struct platform_device *pdev, pm_message_t state) ++{ ++ return 0; ++} ++ ++static int mainstone_resume(struct platform_device *pdev) ++{ ++ return 0; ++} ++ ++static int mainstone_probe(struct platform_device *pdev) ++{ ++ return 0; ++} ++ ++static int mainstone_remove(struct platform_device *pdev) ++{ ++ return 0; ++} ++ ++static int mainstone_wm9713_init(struct snd_soc_codec *codec) ++{ ++ return 0; ++} ++ ++unsigned int mainstone_config_sysclk(struct snd_soc_pcm_runtime *rtd, ++ struct snd_soc_clock_info *info) ++{ ++ /* wm8753 has pll that generates mclk from 13MHz xtal */ ++ return rtd->codec_dai->config_sysclk(rtd->codec_dai, info, 13000000); ++} ++ ++/* the physical audio connections between the WM9713, Baseband and pxa2xx */ ++static struct snd_soc_dai_link mainstone_dai[] = { ++{ ++ .name = "AC97", ++ .stream_name = "AC97 HiFi", ++ .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI], ++ .codec_dai = &wm9713_dai[WM9713_DAI_AC97_HIFI], ++ .init = mainstone_wm9713_init, ++}, ++{ ++ .name = "AC97 Aux", ++ .stream_name = "AC97 Aux", ++ .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX], ++ .codec_dai = &wm9713_dai[WM9713_DAI_AC97_AUX], ++}, ++{ ++ .name = "Baseband", ++ .stream_name = "Voice", ++ .cpu_dai = mainstone_example_dai, ++ .codec_dai = &wm9713_dai[WM9713_DAI_PCM_VOICE], ++ .config_sysclk = mainstone_config_sysclk, ++}, ++}; ++ ++static struct snd_soc_machine mainstone = { ++ .name = "Mainstone", ++ .probe = mainstone_probe, ++ .remove = mainstone_remove, ++ .suspend_pre = mainstone_suspend, ++ .resume_post = mainstone_resume, ++ .ops = &mainstone_ops, ++ .dai_link = mainstone_dai, ++ .num_links = ARRAY_SIZE(mainstone_dai), ++}; ++ ++static struct snd_soc_device mainstone_snd_ac97_devdata = { ++ .machine = &mainstone, ++ .platform = &pxa2xx_soc_platform, ++ .codec_dev = &soc_codec_dev_wm9713, ++}; ++ ++static struct platform_device *mainstone_snd_ac97_device; ++ ++static int __init mainstone_init(void) ++{ ++ int ret; ++ ++ mainstone_snd_ac97_device = platform_device_alloc("soc-audio", -1); ++ if (!mainstone_snd_ac97_device) ++ return -ENOMEM; ++ ++ platform_set_drvdata(mainstone_snd_ac97_device, &mainstone_snd_ac97_devdata); ++ mainstone_snd_ac97_devdata.dev = &mainstone_snd_ac97_device->dev; ++ ++ if((ret = platform_device_add(mainstone_snd_ac97_device)) != 0) ++ platform_device_put(mainstone_snd_ac97_device); ++ ++ return ret; ++} ++ ++static void __exit mainstone_exit(void) ++{ ++ platform_device_unregister(mainstone_snd_ac97_device); ++} ++ ++module_init(mainstone_init); ++module_exit(mainstone_exit); ++ ++/* Module information */ ++MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com"); ++MODULE_DESCRIPTION("Mainstone Example Baseband PCM Interface"); ++MODULE_LICENSE("GPL"); +Index: linux-2.6-pxa-new/sound/soc/pxa/mainstone_bluetooth.c +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/sound/soc/pxa/mainstone_bluetooth.c +@@ -0,0 +1,399 @@ ++/* ++ * mainstone_bluetooth.c ++ * Mainstone Example Bluetooth -- ALSA Soc Audio Layer ++ * ++ * Copyright 2006 Wolfson Microelectronics PLC. ++ * Author: Liam Girdwood ++ * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com ++ * ++ * This program is free software; you can redistribute it and/or modify it ++ * under the terms of the GNU General Public License as published by the ++ * Free Software Foundation; either version 2 of the License, or (at your ++ * option) any later version. ++ * ++ * Revision history ++ * 15th May 2006 Initial version. ++ * ++ * This is example code to demonstrate connecting a bluetooth codec to the PCM ++ * DAI on the WM8753 codec on the Intel Mainstone platform. It is by no means ++ * complete as it requires code to control the BT codec. ++ * ++ * The architecture consists of the WM8753 HIFI DAI connected to the PXA27x ++ * I2S controller and the WM8753 PCM DAI connected to the bluetooth DAI. The ++ * bluetooth codec and wm8753 are controlled via I2C. Audio is routed between ++ * the PXA27x and the bluetooth via internal WM8753 analog paths. ++ * ++ * This example supports the following audio input/outputs. ++ * ++ * o Board mounted Mic and Speaker (spk has amplifier) ++ * o Headphones via jack socket ++ * o BT source and sink ++ * ++ * This driver is not the bluetooth codec driver. This driver only calls ++ * functions from the Bluetooth driver to set up it's PCM DAI. ++ * ++ * It's intended to use the driver as follows:- ++ * ++ * 1. open() WM8753 PCM audio device. ++ * 2. configure PCM audio device (rate etc) - sets up WM8753 PCM DAI, ++ * this should also set up the BT codec DAI (via calling bt driver). ++ * 3. configure codec audio mixer paths. ++ * 4. open(), configure and read/write HIFI audio device - to Tx/Rx voice ++ * ++ * The PCM audio device is opened but IO is never performed on it as the IO is ++ * directly between the codec and the BT codec (and not the CPU). ++ * ++ * TODO: ++ * o Implement callbacks ++ */ ++ ++#include <linux/init.h> ++#include <linux/module.h> ++#include <linux/platform_device.h> ++ ++#include <sound/driver.h> ++#include <sound/core.h> ++#include <sound/pcm.h> ++#include <sound/soc.h> ++#include <sound/soc-dapm.h> ++ ++#include <asm/hardware.h> ++#include <asm/arch/pxa-regs.h> ++#include <asm/arch/audio.h> ++#include <asm/arch/ssp.h> ++ ++#include "../codecs/wm8753.h" ++#include "pxa2xx-pcm.h" ++ ++static struct snd_soc_machine mainstone; ++ ++#define BLUETOOTH_DAIFMT \ ++ (SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBS_CFS |\ ++ SND_SOC_DAIFMT_NB_NF) ++ ++#define BLUETOOTH_DIR \ ++ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE) ++ ++/* ++ * PCM modes - 8k 16bit mono BT codec is master ++ */ ++static struct snd_soc_dai_mode mainstone_bt_modes[] = { ++ /* port master clk & frame modes */ ++ {BLUETOOTH_DAIFMT, SND_SOC_DAITDM_LRDW(0,0), SNDRV_PCM_FORMAT_S16_LE, ++ SNDRV_PCM_RATE_8000, BLUETOOTH_DIR, SND_SOC_DAI_BFS_RATE, 256, 64}, ++}; ++ ++/* Do specific bluetooth PCM startup here */ ++static int mainstone_bt_startup(struct snd_pcm_substream *substream) ++{ ++ return 0; ++} ++ ++/* Do specific bluetooth PCM shutdown here */ ++static void mainstone_bt_shutdown (struct snd_pcm_substream *substream) ++{ ++} ++ ++/* Do pecific bluetooth PCM hw params init here */ ++static int mainstone_bt_hw_params(struct snd_pcm_substream *substream, ++ struct snd_pcm_hw_params *params) ++{ ++ return 0; ++} ++ ++/* Do specific bluetooth PCM hw params free here */ ++static int mainstone_bt_hw_free(struct snd_pcm_substream *substream) ++{ ++ return 0; ++} ++ ++static struct snd_soc_cpu_dai mainstone_bt_dai[] = { ++ { .name = "Bluetooth", ++ .id = 0, ++ .type = SND_SOC_DAI_PCM, ++ .playback = { ++ .channels_min = 1, ++ .channels_max = 1,}, ++ .capture = { ++ .channels_min = 1, ++ .channels_max = 1,}, ++ .ops = { ++ .startup = mainstone_bt_startup, ++ .shutdown = mainstone_bt_shutdown, ++ .hw_params = mainstone_bt_hw_params, ++ .hw_free = mainstone_bt_hw_free, ++ }, ++ .caps = { ++ .mode = mainstone_bt_modes, ++ .num_modes = ARRAY_SIZE(mainstone_bt_modes),}, ++ }, ++}; ++ ++/* do we need to do any thing on the mainstone when the stream is ++ * started and stopped ++ */ ++static int mainstone_startup(struct snd_pcm_substream *substream) ++{ ++ return 0; ++} ++ ++static void mainstone_shutdown(struct snd_pcm_substream *substream) ++{ ++} ++ ++static struct snd_soc_ops mainstone_ops = { ++ .startup = mainstone_startup, ++ .shutdown = mainstone_shutdown, ++}; ++ ++/* PM */ ++static int mainstone_suspend(struct platform_device *pdev, pm_message_t state) ++{ ++ return 0; ++} ++ ++static int mainstone_resume(struct platform_device *pdev) ++{ ++ return 0; ++} ++ ++static int mainstone_probe(struct platform_device *pdev) ++{ ++ return 0; ++} ++ ++static int mainstone_remove(struct platform_device *pdev) ++{ ++ return 0; ++} ++ ++/* ++ * Machine audio functions. ++ * ++ * The machine now has 3 extra audio controls. ++ * ++ * Jack function: Sets function (device plugged into Jack) to nothing (Off) ++ * or Headphones. ++ * ++ * Mic function: Set the on board Mic to On or Off ++ * Spk function: Set the on board Spk to On or Off ++ * ++ * example: BT playback (of far end) and capture (of near end) ++ * Set Mic and Speaker to On, open BT alsa interface as above and set up ++ * internal audio paths. ++ */ ++ ++static int machine_jack_func = 0; ++static int machine_spk_func = 0; ++static int machine_mic_func = 0; ++ ++static int machine_get_jack(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_value *ucontrol) ++{ ++ ucontrol->value.integer.value[0] = machine_jack_func; ++ return 0; ++} ++ ++static int machine_set_jack(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_value *ucontrol) ++{ ++ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); ++ machine_jack_func = ucontrol->value.integer.value[0]; ++ snd_soc_dapm_set_endpoint(codec, "Headphone Jack", machine_jack_func); ++ return 0; ++} ++ ++static int machine_get_spk(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_value *ucontrol) ++{ ++ ucontrol->value.integer.value[0] = machine_spk_func; ++ return 0; ++} ++ ++static int machine_set_spk(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_value *ucontrol) ++{ ++ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); ++ machine_spk_func = ucontrol->value.integer.value[0]; ++ snd_soc_dapm_set_endpoint(codec, "Spk", machine_spk_func); ++ return 0; ++} ++ ++static int machine_get_mic(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_value *ucontrol) ++{ ++ ucontrol->value.integer.value[0] = machine_spk_func; ++ return 0; ++} ++ ++static int machine_set_mic(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_value *ucontrol) ++{ ++ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); ++ machine_spk_func = ucontrol->value.integer.value[0]; ++ snd_soc_dapm_set_endpoint(codec, "Mic", machine_mic_func); ++ return 0; ++} ++ ++/* turns on board speaker amp on/off */ ++static int machine_amp_event(struct snd_soc_dapm_widget *w, int event) ++{ ++#if 0 ++ if (SND_SOC_DAPM_EVENT_ON(event)) ++ /* on */ ++ else ++ /* off */ ++#endif ++ return 0; ++} ++ ++/* machine dapm widgets */ ++static const struct snd_soc_dapm_widget machine_dapm_widgets[] = { ++SND_SOC_DAPM_HP("Headphone Jack", NULL), ++SND_SOC_DAPM_SPK("Spk", machine_amp_event), ++SND_SOC_DAPM_MIC("Mic", NULL), ++}; ++ ++/* machine connections to the codec pins */ ++static const char* audio_map[][3] = { ++ ++ /* headphone connected to LOUT1, ROUT1 */ ++ {"Headphone Jack", NULL, "LOUT"}, ++ {"Headphone Jack", NULL, "ROUT"}, ++ ++ /* speaker connected to LOUT2, ROUT2 */ ++ {"Spk", NULL, "ROUT2"}, ++ {"Spk", NULL, "LOUT2"}, ++ ++ /* mic is connected to MIC1 (via Mic Bias) */ ++ {"MIC1", NULL, "Mic Bias"}, ++ {"Mic Bias", NULL, "Mic"}, ++ ++ {NULL, NULL, NULL}, ++}; ++ ++static const char* jack_function[] = {"Off", "Headphone"}; ++static const char* spk_function[] = {"Off", "On"}; ++static const char* mic_function[] = {"Off", "On"}; ++static const struct soc_enum machine_ctl_enum[] = { ++ SOC_ENUM_SINGLE_EXT(2, jack_function), ++ SOC_ENUM_SINGLE_EXT(2, spk_function), ++ SOC_ENUM_SINGLE_EXT(2, mic_function), ++}; ++ ++static const struct snd_kcontrol_new wm8753_machine_controls[] = { ++ SOC_ENUM_EXT("Jack Function", machine_ctl_enum[0], machine_get_jack, machine_set_jack), ++ SOC_ENUM_EXT("Speaker Function", machine_ctl_enum[1], machine_get_spk, machine_set_spk), ++ SOC_ENUM_EXT("Mic Function", machine_ctl_enum[2], machine_get_mic, machine_set_mic), ++}; ++ ++static int mainstone_wm8753_init(struct snd_soc_codec *codec) ++{ ++ int i, err; ++ ++ /* not used on this machine - e.g. will never be powered up */ ++ snd_soc_dapm_set_endpoint(codec, "OUT3", 0); ++ snd_soc_dapm_set_endpoint(codec, "OUT4", 0); ++ snd_soc_dapm_set_endpoint(codec, "MONO2", 0); ++ snd_soc_dapm_set_endpoint(codec, "MONO1", 0); ++ snd_soc_dapm_set_endpoint(codec, "LINE1", 0); ++ snd_soc_dapm_set_endpoint(codec, "LINE2", 0); ++ snd_soc_dapm_set_endpoint(codec, "RXP", 0); ++ snd_soc_dapm_set_endpoint(codec, "RXN", 0); ++ snd_soc_dapm_set_endpoint(codec, "MIC2", 0); ++ ++ /* Add machine specific controls */ ++ for (i = 0; i < ARRAY_SIZE(wm8753_machine_controls); i++) { ++ if ((err = snd_ctl_add(codec->card, ++ snd_soc_cnew(&wm8753_machine_controls[i],codec, NULL))) < 0) ++ return err; ++ } ++ ++ /* Add machine specific widgets */ ++ for(i = 0; i < ARRAY_SIZE(machine_dapm_widgets); i++) { ++ snd_soc_dapm_new_control(codec, &machine_dapm_widgets[i]); ++ } ++ ++ /* Set up machine specific audio path audio_mapnects */ ++ for(i = 0; audio_map[i][0] != NULL; i++) { ++ snd_soc_dapm_connect_input(codec, audio_map[i][0], audio_map[i][1], audio_map[i][2]); ++ } ++ ++ snd_soc_dapm_sync_endpoints(codec); ++ return 0; ++} ++ ++/* this configures the clocking between the WM8753 and the BT codec */ ++unsigned int mainstone_config_sysclk(struct snd_soc_pcm_runtime *rtd, ++ struct snd_soc_clock_info *info) ++{ ++ /* wm8753 has pll that generates mclk from 13MHz xtal */ ++ return rtd->codec_dai->config_sysclk(rtd->codec_dai, info, 13000000); ++} ++ ++static struct snd_soc_dai_link mainstone_dai[] = { ++{ /* Hifi Playback - for similatious use with voice below */ ++ .name = "WM8753", ++ .stream_name = "WM8753 HiFi", ++ .cpu_dai = &pxa_i2s_dai, ++ .codec_dai = &wm8753_dai[WM8753_DAI_HIFI], ++ .init = mainstone_wm8753_init, ++ .config_sysclk = mainstone_config_sysclk, ++}, ++{ /* Voice via BT */ ++ .name = "Bluetooth", ++ .stream_name = "Voice", ++ .cpu_dai = mainstone_bt_dai, ++ .codec_dai = &wm8753_dai[WM8753_DAI_VOICE], ++ .config_sysclk = mainstone_config_sysclk, ++}, ++}; ++ ++static struct snd_soc_machine mainstone = { ++ .name = "Mainstone", ++ .probe = mainstone_probe, ++ .remove = mainstone_remove, ++ .suspend_pre = mainstone_suspend, ++ .resume_post = mainstone_resume, ++ .ops = &mainstone_ops, ++ .dai_link = mainstone_dai, ++ .num_links = ARRAY_SIZE(mainstone_dai), ++}; ++ ++static struct snd_soc_device mainstone_snd_wm8753_devdata = { ++ .machine = &mainstone, ++ .platform = &pxa2xx_soc_platform, ++ .codec_dev = &soc_codec_dev_wm8753, ++}; ++ ++static struct platform_device *mainstone_snd_wm8753_device; ++ ++static int __init mainstone_init(void) ++{ ++ int ret; ++ ++ mainstone_snd_wm8753_device = platform_device_alloc("soc-audio", -1); ++ if (!mainstone_snd_wm8753_device) ++ return -ENOMEM; ++ ++ platform_set_drvdata(mainstone_snd_wm8753_device, &mainstone_snd_wm8753_devdata); ++ mainstone_snd_wm8753_devdata.dev = &mainstone_snd_wm8753_device->dev; ++ ++ if((ret = platform_device_add(mainstone_snd_wm8753_device)) != 0) ++ platform_device_put(mainstone_snd_wm8753_device); ++ ++ return ret; ++} ++ ++static void __exit mainstone_exit(void) ++{ ++ platform_device_unregister(mainstone_snd_wm8753_device); ++} ++ ++module_init(mainstone_init); ++module_exit(mainstone_exit); ++ ++/* Module information */ ++MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com"); ++MODULE_DESCRIPTION("Mainstone Example Bluetooth PCM Interface"); ++MODULE_LICENSE("GPL"); +Index: linux-2.6-pxa-new/sound/soc/pxa/mainstone_wm8731.c +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/sound/soc/pxa/mainstone_wm8731.c +@@ -0,0 +1,156 @@ ++/* ++ * mainstone.c -- SoC audio for Mainstone ++ * ++ * Copyright 2005 Wolfson Microelectronics PLC. ++ * Author: Liam Girdwood ++ * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com ++ * ++ * Mainstone audio amplifier code taken from arch/arm/mach-pxa/mainstone.c ++ * Copyright: MontaVista Software Inc. ++ * ++ * This program is free software; you can redistribute it and/or modify it ++ * under the terms of the GNU General Public License as published by the ++ * Free Software Foundation; either version 2 of the License, or (at your ++ * option) any later version. ++ * ++ * Revision history ++ * 5th June 2006 Initial version. ++ * ++ */ ++ ++#include <linux/module.h> ++#include <linux/moduleparam.h> ++#include <linux/device.h> ++#include <linux/i2c.h> ++#include <sound/driver.h> ++#include <sound/core.h> ++#include <sound/pcm.h> ++#include <sound/soc.h> ++#include <sound/soc-dapm.h> ++ ++#include <asm/arch/pxa-regs.h> ++#include <asm/arch/mainstone.h> ++#include <asm/arch/audio.h> ++ ++#include "../codecs/wm8731.h" ++#include "pxa2xx-pcm.h" ++ ++static struct snd_soc_machine mainstone; ++ ++ ++static const struct snd_soc_dapm_widget dapm_widgets[] = { ++ SND_SOC_DAPM_MIC("Int Mic", NULL), ++ SND_SOC_DAPM_SPK("Ext Spk", NULL), ++}; ++ ++static const char* intercon[][3] = { ++ ++ /* speaker connected to LHPOUT */ ++ {"Ext Spk", NULL, "LHPOUT"}, ++ ++ /* mic is connected to Mic Jack, with WM8731 Mic Bias */ ++ {"MICIN", NULL, "Mic Bias"}, ++ {"Mic Bias", NULL, "Int Mic"}, ++ ++ /* terminator */ ++ {NULL, NULL, NULL}, ++}; ++ ++/* ++ * Logic for a wm8731 as connected on a Endrelia ETI-B1 board. ++ */ ++static int mainstone_wm8731_init(struct snd_soc_codec *codec) ++{ ++ int i; ++ ++ ++ /* Add specific widgets */ ++ for(i = 0; i < ARRAY_SIZE(dapm_widgets); i++) { ++ snd_soc_dapm_new_control(codec, &dapm_widgets[i]); ++ } ++ ++ /* Set up specific audio path interconnects */ ++ for(i = 0; intercon[i][0] != NULL; i++) { ++ snd_soc_dapm_connect_input(codec, intercon[i][0], intercon[i][1], intercon[i][2]); ++ } ++ ++ /* not connected */ ++ snd_soc_dapm_set_endpoint(codec, "RLINEIN", 0); ++ snd_soc_dapm_set_endpoint(codec, "LLINEIN", 0); ++ ++ /* always connected */ ++ snd_soc_dapm_set_endpoint(codec, "Int Mic", 1); ++ snd_soc_dapm_set_endpoint(codec, "Ext Spk", 1); ++ ++ snd_soc_dapm_sync_endpoints(codec); ++ ++ return 0; ++} ++ ++unsigned int mainstone_config_sysclk(struct snd_soc_pcm_runtime *rtd, ++ struct snd_soc_clock_info *info) ++{ ++ /* we have a 12.288MHz crystal */ ++ return rtd->codec_dai->config_sysclk(rtd->codec_dai, info, 12288000); ++} ++ ++static struct snd_soc_dai_link mainstone_dai[] = { ++{ ++ .name = "WM8731", ++ .stream_name = "WM8731 HiFi", ++ .cpu_dai = &pxa_i2s_dai, ++ .codec_dai = &wm8731_dai, ++ .init = mainstone_wm8731_init, ++ .config_sysclk = mainstone_config_sysclk, ++}, ++}; ++ ++static struct snd_soc_machine mainstone = { ++ .name = "Mainstone", ++ .dai_link = mainstone_dai, ++ .num_links = ARRAY_SIZE(mainstone_dai), ++}; ++ ++static struct wm8731_setup_data corgi_wm8731_setup = { ++ .i2c_address = 0x1b, ++}; ++ ++static struct snd_soc_device mainstone_snd_devdata = { ++ .machine = &mainstone, ++ .platform = &pxa2xx_soc_platform, ++ .codec_dev = &soc_codec_dev_wm8731, ++ .codec_data = &corgi_wm8731_setup, ++}; ++ ++static struct platform_device *mainstone_snd_device; ++ ++static int __init mainstone_init(void) ++{ ++ int ret; ++ ++ mainstone_snd_device = platform_device_alloc("soc-audio", -1); ++ if (!mainstone_snd_device) ++ return -ENOMEM; ++ ++ platform_set_drvdata(mainstone_snd_device, &mainstone_snd_devdata); ++ mainstone_snd_devdata.dev = &mainstone_snd_device->dev; ++ ret = platform_device_add(mainstone_snd_device); ++ ++ if (ret) ++ platform_device_put(mainstone_snd_device); ++ ++ return ret; ++} ++ ++static void __exit mainstone_exit(void) ++{ ++ platform_device_unregister(mainstone_snd_device); ++} ++ ++module_init(mainstone_init); ++module_exit(mainstone_exit); ++ ++/* Module information */ ++MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com"); ++MODULE_DESCRIPTION("ALSA SoC WM8731 Mainstone"); ++MODULE_LICENSE("GPL"); +Index: linux-2.6-pxa-new/sound/soc/pxa/mainstone_wm8753.c +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/sound/soc/pxa/mainstone_wm8753.c +@@ -0,0 +1,226 @@ ++/* ++ * mainstone.c -- SoC audio for Mainstone ++ * ++ * Copyright 2005 Wolfson Microelectronics PLC. ++ * Author: Liam Girdwood ++ * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com ++ * ++ * Mainstone audio amplifier code taken from arch/arm/mach-pxa/mainstone.c ++ * Copyright: MontaVista Software Inc. ++ * ++ * This program is free software; you can redistribute it and/or modify it ++ * under the terms of the GNU General Public License as published by the ++ * Free Software Foundation; either version 2 of the License, or (at your ++ * option) any later version. ++ * ++ * Revision history ++ * 30th Oct 2005 Initial version. ++ * ++ */ ++ ++#include <linux/module.h> ++#include <linux/moduleparam.h> ++#include <linux/device.h> ++#include <linux/i2c.h> ++#include <sound/driver.h> ++#include <sound/core.h> ++#include <sound/pcm.h> ++#include <sound/soc.h> ++#include <sound/soc-dapm.h> ++ ++#include <asm/arch/pxa-regs.h> ++#include <asm/arch/mainstone.h> ++#include <asm/arch/audio.h> ++ ++#include "../codecs/wm8753.h" ++#include "pxa2xx-pcm.h" ++ ++static struct snd_soc_machine mainstone; ++ ++static int mainstone_startup(struct snd_pcm_substream *substream) ++{ ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ ++ if(rtd->cpu_dai->type == SND_SOC_DAI_PCM && rtd->cpu_dai->id == 1) { ++ /* enable USB on the go MUX so we can use SSPFRM2 */ ++ MST_MSCWR2 |= MST_MSCWR2_USB_OTG_SEL; ++ MST_MSCWR2 &= ~MST_MSCWR2_USB_OTG_RST; ++ } ++ return 0; ++} ++ ++static void mainstone_shutdown(struct snd_pcm_substream *substream) ++{ ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ ++ if(rtd->cpu_dai->type == SND_SOC_DAI_PCM && rtd->cpu_dai->id == 1) { ++ /* disable USB on the go MUX so we can use ttyS0 */ ++ MST_MSCWR2 &= ~MST_MSCWR2_USB_OTG_SEL; ++ MST_MSCWR2 |= MST_MSCWR2_USB_OTG_RST; ++ } ++} ++ ++static struct snd_soc_ops mainstone_ops = { ++ .startup = mainstone_startup, ++ .shutdown = mainstone_shutdown, ++}; ++ ++static long mst_audio_suspend_mask; ++ ++static int mainstone_suspend(struct platform_device *pdev, pm_message_t state) ++{ ++ mst_audio_suspend_mask = MST_MSCWR2; ++ MST_MSCWR2 |= MST_MSCWR2_AC97_SPKROFF; ++ return 0; ++} ++ ++static int mainstone_resume(struct platform_device *pdev) ++{ ++ MST_MSCWR2 &= mst_audio_suspend_mask | ~MST_MSCWR2_AC97_SPKROFF; ++ return 0; ++} ++ ++static int mainstone_probe(struct platform_device *pdev) ++{ ++ MST_MSCWR2 &= ~MST_MSCWR2_AC97_SPKROFF; ++ return 0; ++} ++ ++static int mainstone_remove(struct platform_device *pdev) ++{ ++ MST_MSCWR2 |= MST_MSCWR2_AC97_SPKROFF; ++ return 0; ++} ++ ++/* example machine audio_mapnections */ ++static const char* audio_map[][3] = { ++ ++ /* mic is connected to mic1 - with bias */ ++ {"MIC1", NULL, "Mic Bias"}, ++ {"MIC1N", NULL, "Mic Bias"}, ++ {"Mic Bias", NULL, "Mic1 Jack"}, ++ {"Mic Bias", NULL, "Mic1 Jack"}, ++ ++ {"ACIN", NULL, "ACOP"}, ++ {NULL, NULL, NULL}, ++}; ++ ++/* headphone detect support on my board */ ++static const char * hp_pol[] = {"Headphone", "Speaker"}; ++static const struct soc_enum wm8753_enum = ++ SOC_ENUM_SINGLE(WM8753_OUTCTL, 1, 2, hp_pol); ++ ++static const struct snd_kcontrol_new wm8753_mainstone_controls[] = { ++ SOC_SINGLE("Headphone Detect Switch", WM8753_OUTCTL, 6, 1, 0), ++ SOC_ENUM("Headphone Detect Polarity", wm8753_enum), ++}; ++ ++/* ++ * This is an example machine initialisation for a wm8753 connected to a ++ * Mainstone II. It is missing logic to detect hp/mic insertions and logic ++ * to re-route the audio in such an event. ++ */ ++static int mainstone_wm8753_init(struct snd_soc_codec *codec) ++{ ++ int i, err; ++ ++ /* set up mainstone codec pins */ ++ snd_soc_dapm_set_endpoint(codec, "RXP", 0); ++ snd_soc_dapm_set_endpoint(codec, "RXN", 0); ++ snd_soc_dapm_set_endpoint(codec, "MIC2", 0); ++ ++ /* add mainstone specific controls */ ++ for (i = 0; i < ARRAY_SIZE(wm8753_mainstone_controls); i++) { ++ if ((err = snd_ctl_add(codec->card, ++ snd_soc_cnew(&wm8753_mainstone_controls[i],codec, NULL))) < 0) ++ return err; ++ } ++ ++ /* set up mainstone specific audio path audio_mapnects */ ++ for(i = 0; audio_map[i][0] != NULL; i++) { ++ snd_soc_dapm_connect_input(codec, audio_map[i][0], audio_map[i][1], audio_map[i][2]); ++ } ++ ++ snd_soc_dapm_sync_endpoints(codec); ++ return 0; ++} ++ ++unsigned int mainstone_config_sysclk(struct snd_soc_pcm_runtime *rtd, ++ struct snd_soc_clock_info *info) ++{ ++ /* wm8753 has pll that generates mclk from 13MHz xtal */ ++ return rtd->codec_dai->config_sysclk(rtd->codec_dai, info, 13000000); ++} ++ ++static struct snd_soc_dai_link mainstone_dai[] = { ++{ /* Hifi Playback - for similatious use with voice below */ ++ .name = "WM8753", ++ .stream_name = "WM8753 HiFi", ++ .cpu_dai = &pxa_i2s_dai, ++ .codec_dai = &wm8753_dai[WM8753_DAI_HIFI], ++ .init = mainstone_wm8753_init, ++ .config_sysclk = mainstone_config_sysclk, ++}, ++{ /* Voice via BT */ ++ .name = "Bluetooth", ++ .stream_name = "Voice", ++ .cpu_dai = &pxa_ssp_dai[1], ++ .codec_dai = &wm8753_dai[WM8753_DAI_VOICE], ++ .config_sysclk = mainstone_config_sysclk, ++}, ++}; ++ ++static struct snd_soc_machine mainstone = { ++ .name = "Mainstone", ++ .probe = mainstone_probe, ++ .remove = mainstone_remove, ++ .suspend_pre = mainstone_suspend, ++ .resume_post = mainstone_resume, ++ .ops = &mainstone_ops, ++ .dai_link = mainstone_dai, ++ .num_links = ARRAY_SIZE(mainstone_dai), ++}; ++ ++static struct wm8753_setup_data mainstone_wm8753_setup = { ++ .i2c_address = 0x1a, ++}; ++ ++static struct snd_soc_device mainstone_snd_devdata = { ++ .machine = &mainstone, ++ .platform = &pxa2xx_soc_platform, ++ .codec_dev = &soc_codec_dev_wm8753, ++ .codec_data = &mainstone_wm8753_setup, ++}; ++ ++static struct platform_device *mainstone_snd_device; ++ ++static int __init mainstone_init(void) ++{ ++ int ret; ++ ++ mainstone_snd_device = platform_device_alloc("soc-audio", -1); ++ if (!mainstone_snd_device) ++ return -ENOMEM; ++ ++ platform_set_drvdata(mainstone_snd_device, &mainstone_snd_devdata); ++ mainstone_snd_devdata.dev = &mainstone_snd_device->dev; ++ ret = platform_device_add(mainstone_snd_device); ++ ++ if (ret) ++ platform_device_put(mainstone_snd_device); ++ ++ return ret; ++} ++ ++static void __exit mainstone_exit(void) ++{ ++ platform_device_unregister(mainstone_snd_device); ++} ++ ++module_init(mainstone_init); ++module_exit(mainstone_exit); ++ ++/* Module information */ ++MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com"); ++MODULE_DESCRIPTION("ALSA SoC WM8753 Mainstone"); ++MODULE_LICENSE("GPL"); +Index: linux-2.6-pxa-new/sound/soc/pxa/mainstone_wm8974.c +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/sound/soc/pxa/mainstone_wm8974.c +@@ -0,0 +1,112 @@ ++/* ++ * mainstone.c -- SoC audio for Mainstone ++ * ++ * Copyright 2005 Wolfson Microelectronics PLC. ++ * Author: Liam Girdwood ++ * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com ++ * ++ * Mainstone audio amplifier code taken from arch/arm/mach-pxa/mainstone.c ++ * Copyright: MontaVista Software Inc. ++ * ++ * This program is free software; you can redistribute it and/or modify it ++ * under the terms of the GNU General Public License as published by the ++ * Free Software Foundation; either version 2 of the License, or (at your ++ * option) any later version. ++ * ++ * Revision history ++ * 30th Oct 2005 Initial version. ++ * ++ */ ++ ++#include <linux/module.h> ++#include <linux/moduleparam.h> ++#include <linux/device.h> ++#include <linux/i2c.h> ++#include <sound/driver.h> ++#include <sound/core.h> ++#include <sound/pcm.h> ++#include <sound/soc.h> ++#include <sound/soc-dapm.h> ++ ++#include <asm/arch/pxa-regs.h> ++#include <asm/arch/mainstone.h> ++#include <asm/arch/audio.h> ++ ++#include "../codecs/wm8974.h" ++#include "pxa2xx-pcm.h" ++ ++static struct snd_soc_machine mainstone; ++ ++static int mainstone_wm8974_init(struct snd_soc_codec *codec) ++{ ++ return 0; ++} ++ ++unsigned int mainstone_config_sysclk(struct snd_soc_pcm_runtime *rtd, ++ struct snd_soc_clock_info *info) ++{ ++ /* we have a PLL */ ++ return rtd->codec_dai->config_sysclk(rtd->codec_dai, info, 12288000); ++ ++} ++ ++static struct snd_soc_dai_link mainstone_dai[] = { ++{ ++ .name = "WM8974", ++ .stream_name = "WM8974 HiFi", ++ .cpu_dai = &pxa_i2s_dai, ++ .codec_dai = &wm8974_dai, ++ .init = mainstone_wm8974_init, ++ .config_sysclk = mainstone_config_sysclk, ++}, ++}; ++ ++static struct snd_soc_machine mainstone = { ++ .name = "Mainstone", ++ .dai_link = mainstone_dai, ++ .num_links = ARRAY_SIZE(mainstone_dai), ++}; ++ ++static struct wm8974_setup_data mainstone_wm8974_setup = { ++ .i2c_address = 0x1a, ++}; ++ ++static struct snd_soc_device mainstone_snd_devdata = { ++ .machine = &mainstone, ++ .platform = &pxa2xx_soc_platform, ++ .codec_dev = &soc_codec_dev_wm8974, ++ .codec_data = &mainstone_wm8974_setup, ++}; ++ ++static struct platform_device *mainstone_snd_device; ++ ++static int __init mainstone_init(void) ++{ ++ int ret; ++ ++ mainstone_snd_device = platform_device_alloc("soc-audio", -1); ++ if (!mainstone_snd_device) ++ return -ENOMEM; ++ ++ platform_set_drvdata(mainstone_snd_device, &mainstone_snd_devdata); ++ mainstone_snd_devdata.dev = &mainstone_snd_device->dev; ++ ret = platform_device_add(mainstone_snd_device); ++ ++ if (ret) ++ platform_device_put(mainstone_snd_device); ++ ++ return ret; ++} ++ ++static void __exit mainstone_exit(void) ++{ ++ platform_device_unregister(mainstone_snd_device); ++} ++ ++module_init(mainstone_init); ++module_exit(mainstone_exit); ++ ++/* Module information */ ++MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com"); ++MODULE_DESCRIPTION("ALSA SoC Mainstone"); ++MODULE_LICENSE("GPL"); +Index: linux-2.6-pxa-new/sound/soc/pxa/mainstone_wm9712.c +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/sound/soc/pxa/mainstone_wm9712.c +@@ -0,0 +1,171 @@ ++/* ++ * mainstone.c -- SoC audio for Mainstone ++ * ++ * Copyright 2006 Wolfson Microelectronics PLC. ++ * Author: Liam Girdwood ++ * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com ++ * ++ * Mainstone audio amplifier code taken from arch/arm/mach-pxa/mainstone.c ++ * Copyright: MontaVista Software Inc. ++ * ++ * This program is free software; you can redistribute it and/or modify it ++ * under the terms of the GNU General Public License as published by the ++ * Free Software Foundation; either version 2 of the License, or (at your ++ * option) any later version. ++ * ++ * Revision history ++ * 29th Jan 2006 Initial version. ++ * ++ */ ++ ++#include <linux/module.h> ++#include <linux/moduleparam.h> ++#include <linux/device.h> ++#include <linux/i2c.h> ++#include <sound/driver.h> ++#include <sound/core.h> ++#include <sound/pcm.h> ++#include <sound/soc.h> ++#include <sound/soc-dapm.h> ++ ++#include <asm/arch/pxa-regs.h> ++#include <asm/arch/mainstone.h> ++#include <asm/arch/audio.h> ++ ++#include "../codecs/wm9712.h" ++#include "pxa2xx-pcm.h" ++ ++static struct snd_soc_machine mainstone; ++static long mst_audio_suspend_mask; ++ ++static int mainstone_suspend(struct platform_device *pdev, pm_message_t state) ++{ ++ mst_audio_suspend_mask = MST_MSCWR2; ++ MST_MSCWR2 |= MST_MSCWR2_AC97_SPKROFF; ++ return 0; ++} ++ ++static int mainstone_resume(struct platform_device *pdev) ++{ ++ MST_MSCWR2 &= mst_audio_suspend_mask | ~MST_MSCWR2_AC97_SPKROFF; ++ return 0; ++} ++ ++static int mainstone_probe(struct platform_device *pdev) ++{ ++ MST_MSCWR2 &= ~MST_MSCWR2_AC97_SPKROFF; ++ return 0; ++} ++ ++static int mainstone_remove(struct platform_device *pdev) ++{ ++ MST_MSCWR2 |= MST_MSCWR2_AC97_SPKROFF; ++ return 0; ++} ++ ++/* mainstone machine dapm widgets */ ++static const struct snd_soc_dapm_widget mainstone_dapm_widgets[] = { ++ SND_SOC_DAPM_MIC("Mic (Internal)", NULL), ++}; ++ ++/* example machine interconnections */ ++static const char* intercon[][3] = { ++ ++ /* mic is connected to mic1 - with bias */ ++ {"MIC1", NULL, "Mic Bias"}, ++ {"Mic Bias", NULL, "Mic (Internal)"}, ++ ++ {NULL, NULL, NULL}, ++}; ++ ++/* ++ * This is an example machine initialisation for a wm8753 connected to a ++ * Mainstone II. It is missing logic to detect hp/mic insertions and logic ++ * to re-route the audio in such an event. ++ */ ++static int mainstone_wm9712_init(struct snd_soc_codec *codec) ++{ ++ int i; ++ ++ /* set up mainstone codec pins */ ++ snd_soc_dapm_set_endpoint(codec, "RXP", 0); ++ snd_soc_dapm_set_endpoint(codec, "RXN", 0); ++ //snd_soc_dapm_set_endpoint(codec, "MIC2", 0); ++ ++ /* Add mainstone specific widgets */ ++ for(i = 0; i < ARRAY_SIZE(mainstone_dapm_widgets); i++) { ++ snd_soc_dapm_new_control(codec, &mainstone_dapm_widgets[i]); ++ } ++ ++ /* set up mainstone specific audio path interconnects */ ++ for(i = 0; intercon[i][0] != NULL; i++) { ++ snd_soc_dapm_connect_input(codec, intercon[i][0], intercon[i][1], intercon[i][2]); ++ } ++ ++ snd_soc_dapm_sync_endpoints(codec); ++ return 0; ++} ++ ++static struct snd_soc_dai_link mainstone_dai[] = { ++{ ++ .name = "AC97", ++ .stream_name = "AC97 HiFi", ++ .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI], ++ .codec_dai = &wm9712_dai[WM9712_DAI_AC97_HIFI], ++ .init = mainstone_wm9712_init, ++}, ++{ ++ .name = "AC97 Aux", ++ .stream_name = "AC97 Aux", ++ .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX], ++ .codec_dai = &wm9712_dai[WM9712_DAI_AC97_AUX], ++}, ++}; ++ ++static struct snd_soc_machine mainstone = { ++ .name = "Mainstone", ++ .probe = mainstone_probe, ++ .remove = mainstone_remove, ++ .suspend_pre = mainstone_suspend, ++ .resume_post = mainstone_resume, ++ .dai_link = mainstone_dai, ++ .num_links = ARRAY_SIZE(mainstone_dai), ++}; ++ ++static struct snd_soc_device mainstone_snd_ac97_devdata = { ++ .machine = &mainstone, ++ .platform = &pxa2xx_soc_platform, ++ .codec_dev = &soc_codec_dev_wm9712, ++}; ++ ++static struct platform_device *mainstone_snd_ac97_device; ++ ++static int __init mainstone_init(void) ++{ ++ int ret; ++ ++ mainstone_snd_ac97_device = platform_device_alloc("soc-audio", -1); ++ if (!mainstone_snd_ac97_device) ++ return -ENOMEM; ++ ++ platform_set_drvdata(mainstone_snd_ac97_device, &mainstone_snd_ac97_devdata); ++ mainstone_snd_ac97_devdata.dev = &mainstone_snd_ac97_device->dev; ++ ++ if((ret = platform_device_add(mainstone_snd_ac97_device)) != 0) ++ platform_device_put(mainstone_snd_ac97_device); ++ ++ return ret; ++} ++ ++static void __exit mainstone_exit(void) ++{ ++ platform_device_unregister(mainstone_snd_ac97_device); ++} ++ ++module_init(mainstone_init); ++module_exit(mainstone_exit); ++ ++/* Module information */ ++MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com"); ++MODULE_DESCRIPTION("ALSA SoC WM9712 Mainstone"); ++MODULE_LICENSE("GPL"); +Index: linux-2.6-pxa-new/sound/soc/pxa/mainstone_wm9713.c +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/sound/soc/pxa/mainstone_wm9713.c +@@ -0,0 +1,263 @@ ++/* ++ * mainstone.c -- SoC audio for Mainstone ++ * ++ * Copyright 2006 Wolfson Microelectronics PLC. ++ * Author: Liam Girdwood ++ * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com ++ * ++ * Mainstone audio amplifier code taken from arch/arm/mach-pxa/mainstone.c ++ * Copyright: MontaVista Software Inc. ++ * ++ * This program is free software; you can redistribute it and/or modify it ++ * under the terms of the GNU General Public License as published by the ++ * Free Software Foundation; either version 2 of the License, or (at your ++ * option) any later version. ++ * ++ * Revision history ++ * 29th Jan 2006 Initial version. ++ * ++ */ ++ ++#include <linux/module.h> ++#include <linux/moduleparam.h> ++#include <linux/device.h> ++#include <linux/i2c.h> ++#include <sound/driver.h> ++#include <sound/core.h> ++#include <sound/pcm.h> ++#include <sound/soc.h> ++#include <sound/soc-dapm.h> ++ ++#include <asm/arch/pxa-regs.h> ++#include <asm/arch/mainstone.h> ++#include <asm/arch/audio.h> ++ ++#include "../codecs/wm9713.h" ++#include "pxa2xx-pcm.h" ++ ++static struct snd_soc_machine mainstone; ++ ++static int mainstone_startup(struct snd_pcm_substream *substream) ++{ ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ ++ if(rtd->cpu_dai->type == SND_SOC_DAI_PCM && rtd->cpu_dai->id == 1) { ++ /* enable USB on the go MUX so we can use SSPFRM2 */ ++ MST_MSCWR2 |= MST_MSCWR2_USB_OTG_SEL; ++ MST_MSCWR2 &= ~MST_MSCWR2_USB_OTG_RST; ++ } ++ return 0; ++} ++ ++static void mainstone_shutdown(struct snd_pcm_substream *substream) ++{ ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ ++ if(rtd->cpu_dai->type == SND_SOC_DAI_PCM && rtd->cpu_dai->id == 1) { ++ /* disable USB on the go MUX so we can use ttyS0 */ ++ MST_MSCWR2 &= ~MST_MSCWR2_USB_OTG_SEL; ++ MST_MSCWR2 |= MST_MSCWR2_USB_OTG_RST; ++ } ++} ++ ++static struct snd_soc_ops mainstone_ops = { ++ .startup = mainstone_startup, ++ .shutdown = mainstone_shutdown, ++}; ++ ++static int test = 0; ++static int get_test(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_value *ucontrol) ++{ ++ ucontrol->value.integer.value[0] = test; ++ return 0; ++} ++ ++static int set_test(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_value *ucontrol) ++{ ++ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); ++ ++ test = ucontrol->value.integer.value[0]; ++ if(test) { ++ ++ } else { ++ ++ } ++ return 0; ++} ++ ++static long mst_audio_suspend_mask; ++ ++static int mainstone_suspend(struct platform_device *pdev, pm_message_t state) ++{ ++ mst_audio_suspend_mask = MST_MSCWR2; ++ MST_MSCWR2 |= MST_MSCWR2_AC97_SPKROFF; ++ return 0; ++} ++ ++static int mainstone_resume(struct platform_device *pdev) ++{ ++ MST_MSCWR2 &= mst_audio_suspend_mask | ~MST_MSCWR2_AC97_SPKROFF; ++ return 0; ++} ++ ++static int mainstone_probe(struct platform_device *pdev) ++{ ++ MST_MSCWR2 &= ~MST_MSCWR2_AC97_SPKROFF; ++ return 0; ++} ++ ++static int mainstone_remove(struct platform_device *pdev) ++{ ++ MST_MSCWR2 |= MST_MSCWR2_AC97_SPKROFF; ++ return 0; ++} ++ ++static const char* test_function[] = {"Off", "On"}; ++static const struct soc_enum mainstone_enum[] = { ++ SOC_ENUM_SINGLE_EXT(2, test_function), ++}; ++ ++static const struct snd_kcontrol_new mainstone_controls[] = { ++ SOC_ENUM_EXT("ATest Function", mainstone_enum[0], get_test, set_test), ++}; ++ ++/* mainstone machine dapm widgets */ ++static const struct snd_soc_dapm_widget mainstone_dapm_widgets[] = { ++ SND_SOC_DAPM_MIC("Mic 1", NULL), ++ SND_SOC_DAPM_MIC("Mic 2", NULL), ++ SND_SOC_DAPM_MIC("Mic 3", NULL), ++}; ++ ++/* example machine audio_mapnections */ ++static const char* audio_map[][3] = { ++ ++ /* mic is connected to mic1 - with bias */ ++ {"MIC1", NULL, "Mic Bias"}, ++ {"Mic Bias", NULL, "Mic 1"}, ++ /* mic is connected to mic2A - with bias */ ++ {"MIC2A", NULL, "Mic Bias"}, ++ {"Mic Bias", NULL, "Mic 2"}, ++ /* mic is connected to mic2B - with bias */ ++ {"MIC2B", NULL, "Mic Bias"}, ++ {"Mic Bias", NULL, "Mic 3"}, ++ ++ {NULL, NULL, NULL}, ++}; ++ ++/* ++ * This is an example machine initialisation for a wm9713 connected to a ++ * Mainstone II. It is missing logic to detect hp/mic insertions and logic ++ * to re-route the audio in such an event. ++ */ ++static int mainstone_wm9713_init(struct snd_soc_codec *codec) ++{ ++ int i, err; ++ ++ /* set up mainstone codec pins */ ++ snd_soc_dapm_set_endpoint(codec, "RXP", 0); ++ snd_soc_dapm_set_endpoint(codec, "RXN", 0); ++ //snd_soc_dapm_set_endpoint(codec, "MIC2", 0); ++ ++ /* Add test specific controls */ ++ for (i = 0; i < ARRAY_SIZE(mainstone_controls); i++) { ++ if ((err = snd_ctl_add(codec->card, ++ snd_soc_cnew(&mainstone_controls[i],codec, NULL))) < 0) ++ return err; ++ } ++ ++ /* Add mainstone specific widgets */ ++ for(i = 0; i < ARRAY_SIZE(mainstone_dapm_widgets); i++) { ++ snd_soc_dapm_new_control(codec, &mainstone_dapm_widgets[i]); ++ } ++ ++ /* set up mainstone specific audio path audio_mapnects */ ++ for(i = 0; audio_map[i][0] != NULL; i++) { ++ snd_soc_dapm_connect_input(codec, audio_map[i][0], audio_map[i][1], audio_map[i][2]); ++ } ++ ++ snd_soc_dapm_sync_endpoints(codec); ++ return 0; ++} ++ ++/* configure the system audio clock */ ++unsigned int mainstone_config_sysclk(struct snd_soc_pcm_runtime *rtd, ++ struct snd_soc_clock_info *info) ++{ ++ return rtd->codec_dai->config_sysclk(rtd->codec_dai, info, 24576000); ++} ++ ++static struct snd_soc_dai_link mainstone_dai[] = { ++{ ++ .name = "AC97", ++ .stream_name = "AC97 HiFi", ++ .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI], ++ .codec_dai = &wm9713_dai[WM9713_DAI_AC97_HIFI], ++ .init = mainstone_wm9713_init, ++ .config_sysclk = mainstone_config_sysclk, ++}, ++{ ++ .name = "AC97 Aux", ++ .stream_name = "AC97 Aux", ++ .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX], ++ .codec_dai = &wm9713_dai[WM9713_DAI_AC97_AUX], ++ .config_sysclk = mainstone_config_sysclk, ++}, ++{ ++ .name = "WM9713", ++ .stream_name = "WM9713 Voice", ++ .cpu_dai = &pxa_ssp_dai[PXA2XX_DAI_SSP2], ++ .codec_dai = &wm9713_dai[WM9713_DAI_PCM_VOICE], ++ .config_sysclk = mainstone_config_sysclk, ++}, ++}; ++ ++static struct snd_soc_machine mainstone = { ++ .name = "Mainstone", ++ .probe = mainstone_probe, ++ .remove = mainstone_remove, ++ .suspend_pre = mainstone_suspend, ++ .resume_post = mainstone_resume, ++ .ops = &mainstone_ops, ++ .dai_link = mainstone_dai, ++ .num_links = ARRAY_SIZE(mainstone_dai), ++}; ++ ++static struct snd_soc_device mainstone_snd_ac97_devdata = { ++ .machine = &mainstone, ++ .platform = &pxa2xx_soc_platform, ++ .codec_dev = &soc_codec_dev_wm9713, ++}; ++ ++static struct platform_device *mainstone_snd_ac97_device; ++ ++static int __init mainstone_init(void) ++{ ++ int ret; ++ ++ mainstone_snd_ac97_device = platform_device_alloc("soc-audio", -1); ++ if (!mainstone_snd_ac97_device) ++ return -ENOMEM; ++ ++ platform_set_drvdata(mainstone_snd_ac97_device, &mainstone_snd_ac97_devdata); ++ mainstone_snd_ac97_devdata.dev = &mainstone_snd_ac97_device->dev; ++ ++ if((ret = platform_device_add(mainstone_snd_ac97_device)) != 0) ++ platform_device_put(mainstone_snd_ac97_device); ++ ++ return ret; ++} ++ ++static void __exit mainstone_exit(void) ++{ ++ platform_device_unregister(mainstone_snd_ac97_device); ++} ++ ++module_init(mainstone_init); ++module_exit(mainstone_exit); ++ ++/* Module information */ ++MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com"); ++MODULE_DESCRIPTION("ALSA SoC WM9713 Mainstone"); ++MODULE_LICENSE("GPL"); +Index: linux-2.6-pxa-new/sound/soc/pxa/poodle.c +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/sound/soc/pxa/poodle.c +@@ -0,0 +1,329 @@ ++/* ++ * poodle.c -- SoC audio for Poodle ++ * ++ * Copyright 2005 Wolfson Microelectronics PLC. ++ * Copyright 2005 Openedhand Ltd. ++ * ++ * Authors: Liam Girdwood <liam.girdwood@wolfsonmicro.com> ++ * Richard Purdie <richard@openedhand.com> ++ * ++ * This program is free software; you can redistribute it and/or modify it ++ * under the terms of the GNU General Public License as published by the ++ * Free Software Foundation; either version 2 of the License, or (at your ++ * option) any later version. ++ * ++ */ ++ ++#include <linux/module.h> ++#include <linux/moduleparam.h> ++#include <linux/timer.h> ++#include <linux/interrupt.h> ++#include <linux/platform_device.h> ++#include <sound/driver.h> ++#include <sound/core.h> ++#include <sound/pcm.h> ++#include <sound/soc.h> ++#include <sound/soc-dapm.h> ++ ++#include <asm/mach-types.h> ++#include <asm/hardware/locomo.h> ++#include <asm/arch/pxa-regs.h> ++#include <asm/arch/hardware.h> ++#include <asm/arch/poodle.h> ++#include <asm/arch/audio.h> ++ ++#include "../codecs/wm8731.h" ++#include "pxa2xx-pcm.h" ++ ++#define POODLE_HP 1 ++#define POODLE_HP_OFF 0 ++#define POODLE_SPK_ON 1 ++#define POODLE_SPK_OFF 0 ++ ++ /* audio clock in Hz - rounded from 12.235MHz */ ++#define POODLE_AUDIO_CLOCK 12288000 ++ ++static int poodle_jack_func; ++static int poodle_spk_func; ++ ++static void poodle_ext_control(struct snd_soc_codec *codec) ++{ ++ int spk = 0; ++ ++ /* set up jack connection */ ++ if (poodle_jack_func == POODLE_HP) { ++ /* set = unmute headphone */ ++ locomo_gpio_write(&poodle_locomo_device.dev, ++ POODLE_LOCOMO_GPIO_MUTE_L, 1); ++ locomo_gpio_write(&poodle_locomo_device.dev, ++ POODLE_LOCOMO_GPIO_MUTE_R, 1); ++ snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 1); ++ } else { ++ locomo_gpio_write(&poodle_locomo_device.dev, ++ POODLE_LOCOMO_GPIO_MUTE_L, 0); ++ locomo_gpio_write(&poodle_locomo_device.dev, ++ POODLE_LOCOMO_GPIO_MUTE_R, 0); ++ snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 0); ++ } ++ ++ if (poodle_spk_func == POODLE_SPK_ON) ++ spk = 1; ++ ++ /* set the enpoints to their new connetion states */ ++ snd_soc_dapm_set_endpoint(codec, "Ext Spk", spk); ++ ++ /* signal a DAPM event */ ++ snd_soc_dapm_sync_endpoints(codec); ++} ++ ++static int poodle_startup(struct snd_pcm_substream *substream) ++{ ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ struct snd_soc_codec *codec = rtd->socdev->codec; ++ ++ /* check the jack status at stream startup */ ++ poodle_ext_control(codec); ++ return 0; ++} ++ ++/* we need to unmute the HP at shutdown as the mute burns power on poodle */ ++static int poodle_shutdown(struct snd_pcm_substream *substream) ++{ ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ struct snd_soc_codec *codec = rtd->socdev->codec; ++ ++ /* set = unmute headphone */ ++ locomo_gpio_write(&poodle_locomo_device.dev, ++ POODLE_LOCOMO_GPIO_MUTE_L, 1); ++ locomo_gpio_write(&poodle_locomo_device.dev, ++ POODLE_LOCOMO_GPIO_MUTE_R, 1); ++ return 0; ++} ++ ++static struct snd_soc_ops poodle_ops = { ++ .startup = poodle_startup, ++ .shutdown = poodle_shutdown, ++}; ++ ++static int poodle_get_jack(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_value *ucontrol) ++{ ++ ucontrol->value.integer.value[0] = poodle_jack_func; ++ return 0; ++} ++ ++static int poodle_set_jack(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_value *ucontrol) ++{ ++ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); ++ ++ if (poodle_jack_func == ucontrol->value.integer.value[0]) ++ return 0; ++ ++ poodle_jack_func = ucontrol->value.integer.value[0]; ++ poodle_ext_control(codec); ++ return 1; ++} ++ ++static int poodle_get_spk(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_value *ucontrol) ++{ ++ ucontrol->value.integer.value[0] = poodle_spk_func; ++ return 0; ++} ++ ++static int poodle_set_spk(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_value *ucontrol) ++{ ++ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); ++ ++ if (poodle_spk_func == ucontrol->value.integer.value[0]) ++ return 0; ++ ++ poodle_spk_func = ucontrol->value.integer.value[0]; ++ poodle_ext_control(codec); ++ return 1; ++} ++ ++static int poodle_amp_event(struct snd_soc_dapm_widget *w, int event) ++{ ++ if (SND_SOC_DAPM_EVENT_ON(event)) ++ locomo_gpio_write(&poodle_locomo_device.dev, ++ POODLE_LOCOMO_GPIO_AMP_ON, 0); ++ else ++ locomo_gpio_write(&poodle_locomo_device.dev, ++ POODLE_LOCOMO_GPIO_AMP_ON, 1); ++ ++ return 0; ++} ++ ++/* poodle machine dapm widgets */ ++static const struct snd_soc_dapm_widget wm8731_dapm_widgets[] = { ++SND_SOC_DAPM_HP("Headphone Jack", NULL), ++SND_SOC_DAPM_SPK("Ext Spk", poodle_amp_event), ++}; ++ ++/* Corgi machine audio_mapnections to the codec pins */ ++static const char *audio_map[][3] = { ++ ++ /* headphone connected to LHPOUT1, RHPOUT1 */ ++ {"Headphone Jack", NULL, "LHPOUT"}, ++ {"Headphone Jack", NULL, "RHPOUT"}, ++ ++ /* speaker connected to LOUT, ROUT */ ++ {"Ext Spk", NULL, "ROUT"}, ++ {"Ext Spk", NULL, "LOUT"}, ++ ++ {NULL, NULL, NULL}, ++}; ++ ++static const char *jack_function[] = {"Off", "Headphone"}; ++static const char *spk_function[] = {"Off", "On"}; ++static const struct soc_enum poodle_enum[] = { ++ SOC_ENUM_SINGLE_EXT(2, jack_function), ++ SOC_ENUM_SINGLE_EXT(2, spk_function), ++}; ++ ++static const snd_kcontrol_new_t wm8731_poodle_controls[] = { ++ SOC_ENUM_EXT("Jack Function", poodle_enum[0], poodle_get_jack, ++ poodle_set_jack), ++ SOC_ENUM_EXT("Speaker Function", poodle_enum[1], poodle_get_spk, ++ poodle_set_spk), ++}; ++ ++/* ++ * Logic for a wm8731 as connected on a Sharp SL-C7x0 Device ++ */ ++static int poodle_wm8731_init(struct snd_soc_codec *codec) ++{ ++ int i, err; ++ ++ snd_soc_dapm_set_endpoint(codec, "LLINEIN", 0); ++ snd_soc_dapm_set_endpoint(codec, "RLINEIN", 0); ++ snd_soc_dapm_set_endpoint(codec, "MICIN", 1); ++ ++ /* Add poodle specific controls */ ++ for (i = 0; i < ARRAY_SIZE(wm8731_poodle_controls); i++) { ++ err = snd_ctl_add(codec->card, ++ snd_soc_cnew(&wm8731_poodle_controls[i],codec, NULL)); ++ if (err < 0) ++ return err; ++ } ++ ++ /* Add poodle specific widgets */ ++ for (i = 0; i < ARRAY_SIZE(wm8731_dapm_widgets); i++) { ++ snd_soc_dapm_new_control(codec, &wm8731_dapm_widgets[i]); ++ } ++ ++ /* Set up poodle specific audio path audio_map */ ++ for (i = 0; audio_map[i][0] != NULL; i++) { ++ snd_soc_dapm_connect_input(codec, audio_map[i][0], ++ audio_map[i][1], audio_map[i][2]); ++ } ++ ++ snd_soc_dapm_sync_endpoints(codec); ++ return 0; ++} ++ ++static unsigned int poodle_config_sysclk(struct snd_soc_pcm_runtime *rtd, ++ struct snd_soc_clock_info *info) ++{ ++ if (info->bclk_master & SND_SOC_DAIFMT_CBS_CFS) { ++ /* pxa2xx is i2s master */ ++ switch (info->rate) { ++ case 44100: ++ case 88200: ++ /* configure codec digital filters for 44.1, 88.2 */ ++ rtd->codec_dai->config_sysclk(rtd->codec_dai, info, ++ 11289600); ++ break; ++ default: ++ /* configure codec digital filters for all other rates */ ++ rtd->codec_dai->config_sysclk(rtd->codec_dai, info, ++ POODLE_AUDIO_CLOCK); ++ break; ++ } ++ return rtd->cpu_dai->config_sysclk(rtd->cpu_dai, info, ++ POODLE_AUDIO_CLOCK); ++ } else { ++ /* codec is i2s master - ++ * only configure codec DAI clock and filters */ ++ return rtd->codec_dai->config_sysclk(rtd->codec_dai, info, ++ POODLE_AUDIO_CLOCK); ++ } ++} ++ ++/* poodle digital audio interface glue - connects codec <--> CPU */ ++static struct snd_soc_dai_link poodle_dai = { ++ .name = "WM8731", ++ .stream_name = "WM8731", ++ .cpu_dai = &pxa_i2s_dai, ++ .codec_dai = &wm8731_dai, ++ .init = poodle_wm8731_init, ++ .config_sysclk = poodle_config_sysclk, ++}; ++ ++/* poodle audio machine driver */ ++static struct snd_soc_machine snd_soc_machine_poodle = { ++ .name = "Poodle", ++ .dai_link = &poodle_dai, ++ .num_links = 1, ++ .ops = &poodle_ops, ++}; ++ ++/* poodle audio private data */ ++static struct wm8731_setup_data poodle_wm8731_setup = { ++ .i2c_address = 0x1b, ++}; ++ ++/* poodle audio subsystem */ ++static struct snd_soc_device poodle_snd_devdata = { ++ .machine = &snd_soc_machine_poodle, ++ .platform = &pxa2xx_soc_platform, ++ .codec_dev = &soc_codec_dev_wm8731, ++ .codec_data = &poodle_wm8731_setup, ++}; ++ ++static struct platform_device *poodle_snd_device; ++ ++static int __init poodle_init(void) ++{ ++ int ret; ++ ++ if (!machine_is_poodle()) ++ return -ENODEV; ++ ++ locomo_gpio_set_dir(&poodle_locomo_device.dev, ++ POODLE_LOCOMO_GPIO_AMP_ON, 0); ++ /* should we mute HP at startup - burning power ?*/ ++ locomo_gpio_set_dir(&poodle_locomo_device.dev, ++ POODLE_LOCOMO_GPIO_MUTE_L, 0); ++ locomo_gpio_set_dir(&poodle_locomo_device.dev, ++ POODLE_LOCOMO_GPIO_MUTE_R, 0); ++ ++ poodle_snd_device = platform_device_alloc("soc-audio", -1); ++ if (!poodle_snd_device) ++ return -ENOMEM; ++ ++ platform_set_drvdata(poodle_snd_device, &poodle_snd_devdata); ++ poodle_snd_devdata.dev = &poodle_snd_device->dev; ++ ret = platform_device_add(poodle_snd_device); ++ ++ if (ret) ++ platform_device_put(poodle_snd_device); ++ ++ return ret; ++} ++ ++static void __exit poodle_exit(void) ++{ ++ platform_device_unregister(poodle_snd_device); ++} ++ ++module_init(poodle_init); ++module_exit(poodle_exit); ++ ++/* Module information */ ++MODULE_AUTHOR("Richard Purdie"); ++MODULE_DESCRIPTION("ALSA SoC Poodle"); ++MODULE_LICENSE("GPL"); +Index: linux-2.6-pxa-new/sound/soc/pxa/pxa2xx-ac97.c +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/sound/soc/pxa/pxa2xx-ac97.c +@@ -0,0 +1,437 @@ ++/* ++ * linux/sound/pxa2xx-ac97.c -- AC97 support for the Intel PXA2xx chip. ++ * ++ * Author: Nicolas Pitre ++ * Created: Dec 02, 2004 ++ * Copyright: MontaVista Software Inc. ++ * ++ * This program is free software; you can redistribute it and/or modify ++ * it under the terms of the GNU General Public License version 2 as ++ * published by the Free Software Foundation. ++ */ ++ ++#include <linux/init.h> ++#include <linux/module.h> ++#include <linux/platform_device.h> ++#include <linux/interrupt.h> ++#include <linux/wait.h> ++#include <linux/delay.h> ++ ++#include <sound/driver.h> ++#include <sound/core.h> ++#include <sound/pcm.h> ++#include <sound/ac97_codec.h> ++#include <sound/initval.h> ++#include <sound/soc.h> ++ ++#include <asm/irq.h> ++#include <linux/mutex.h> ++#include <asm/hardware.h> ++#include <asm/arch/pxa-regs.h> ++#include <asm/arch/audio.h> ++ ++#include "pxa2xx-pcm.h" ++ ++static DEFINE_MUTEX(car_mutex); ++static DECLARE_WAIT_QUEUE_HEAD(gsr_wq); ++static volatile long gsr_bits; ++ ++#define AC97_DIR \ ++ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE) ++ ++#define AC97_RATES \ ++ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ ++ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) ++ ++/* may need to expand this */ ++static struct snd_soc_dai_mode pxa2xx_ac97_modes[] = { ++ { ++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, ++ .pcmrate = AC97_RATES, ++ .pcmdir = AC97_DIR, ++ }, ++}; ++ ++/* ++ * Beware PXA27x bugs: ++ * ++ * o Slot 12 read from modem space will hang controller. ++ * o CDONE, SDONE interrupt fails after any slot 12 IO. ++ * ++ * We therefore have an hybrid approach for waiting on SDONE (interrupt or ++ * 1 jiffy timeout if interrupt never comes). ++ */ ++ ++static unsigned short pxa2xx_ac97_read(struct snd_ac97 *ac97, ++ unsigned short reg) ++{ ++ unsigned short val = -1; ++ volatile u32 *reg_addr; ++ ++ mutex_lock(&car_mutex); ++ ++ /* set up primary or secondary codec/modem space */ ++#ifdef CONFIG_PXA27x ++ reg_addr = ac97->num ? &SAC_REG_BASE : &PAC_REG_BASE; ++#else ++ if (reg == AC97_GPIO_STATUS) ++ reg_addr = ac97->num ? &SMC_REG_BASE : &PMC_REG_BASE; ++ else ++ reg_addr = ac97->num ? &SAC_REG_BASE : &PAC_REG_BASE; ++#endif ++ reg_addr += (reg >> 1); ++ ++#ifndef CONFIG_PXA27x ++ if (reg == AC97_GPIO_STATUS) { ++ /* read from controller cache */ ++ val = *reg_addr; ++ goto out; ++ } ++#endif ++ ++ /* start read access across the ac97 link */ ++ GSR = GSR_CDONE | GSR_SDONE; ++ gsr_bits = 0; ++ val = *reg_addr; ++ ++ wait_event_timeout(gsr_wq, (GSR | gsr_bits) & GSR_SDONE, 1); ++ if (!((GSR | gsr_bits) & GSR_SDONE)) { ++ printk(KERN_ERR "%s: read error (ac97_reg=%x GSR=%#lx)\n", ++ __FUNCTION__, reg, GSR | gsr_bits); ++ val = -1; ++ goto out; ++ } ++ ++ /* valid data now */ ++ GSR = GSR_CDONE | GSR_SDONE; ++ gsr_bits = 0; ++ val = *reg_addr; ++ /* but we've just started another cycle... */ ++ wait_event_timeout(gsr_wq, (GSR | gsr_bits) & GSR_SDONE, 1); ++ ++out: mutex_unlock(&car_mutex); ++ return val; ++} ++ ++static void pxa2xx_ac97_write(struct snd_ac97 *ac97, unsigned short reg, ++ unsigned short val) ++{ ++ volatile u32 *reg_addr; ++ ++ mutex_lock(&car_mutex); ++ ++ /* set up primary or secondary codec/modem space */ ++#ifdef CONFIG_PXA27x ++ reg_addr = ac97->num ? &SAC_REG_BASE : &PAC_REG_BASE; ++#else ++ if (reg == AC97_GPIO_STATUS) ++ reg_addr = ac97->num ? &SMC_REG_BASE : &PMC_REG_BASE; ++ else ++ reg_addr = ac97->num ? &SAC_REG_BASE : &PAC_REG_BASE; ++#endif ++ reg_addr += (reg >> 1); ++ ++ GSR = GSR_CDONE | GSR_SDONE; ++ gsr_bits = 0; ++ *reg_addr = val; ++ wait_event_timeout(gsr_wq, (GSR | gsr_bits) & GSR_CDONE, 1); ++ if (!((GSR | gsr_bits) & GSR_CDONE)) ++ printk(KERN_ERR "%s: write error (ac97_reg=%x GSR=%#lx)\n", ++ __FUNCTION__, reg, GSR | gsr_bits); ++ ++ mutex_unlock(&car_mutex); ++} ++ ++static void pxa2xx_ac97_warm_reset(struct snd_ac97 *ac97) ++{ ++ gsr_bits = 0; ++ ++#ifdef CONFIG_PXA27x ++ /* warm reset broken on Bulverde, ++ so manually keep AC97 reset high */ ++ pxa_gpio_mode(113 | GPIO_OUT | GPIO_DFLT_HIGH); ++ udelay(10); ++ GCR |= GCR_WARM_RST; ++ pxa_gpio_mode(113 | GPIO_ALT_FN_2_OUT); ++ udelay(500); ++#else ++ GCR |= GCR_WARM_RST | GCR_PRIRDY_IEN | GCR_SECRDY_IEN; ++ wait_event_timeout(gsr_wq, gsr_bits & (GSR_PCR | GSR_SCR), 1); ++#endif ++ ++ if (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR))) ++ printk(KERN_INFO "%s: warm reset timeout (GSR=%#lx)\n", ++ __FUNCTION__, gsr_bits); ++ ++ GCR &= ~(GCR_PRIRDY_IEN|GCR_SECRDY_IEN); ++ GCR |= GCR_SDONE_IE|GCR_CDONE_IE; ++} ++ ++static void pxa2xx_ac97_cold_reset(struct snd_ac97 *ac97) ++{ ++ GCR &= GCR_COLD_RST; /* clear everything but nCRST */ ++ GCR &= ~GCR_COLD_RST; /* then assert nCRST */ ++ ++ gsr_bits = 0; ++#ifdef CONFIG_PXA27x ++ /* PXA27x Developers Manual section 13.5.2.2.1 */ ++ pxa_set_cken(1 << 31, 1); ++ udelay(5); ++ pxa_set_cken(1 << 31, 0); ++ GCR = GCR_COLD_RST; ++ udelay(50); ++#else ++ GCR = GCR_COLD_RST; ++ GCR |= GCR_CDONE_IE|GCR_SDONE_IE; ++ wait_event_timeout(gsr_wq, gsr_bits & (GSR_PCR | GSR_SCR), 1); ++#endif ++ ++ if (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR))) ++ printk(KERN_INFO "%s: cold reset timeout (GSR=%#lx)\n", ++ __FUNCTION__, gsr_bits); ++ ++ GCR &= ~(GCR_PRIRDY_IEN|GCR_SECRDY_IEN); ++ GCR |= GCR_SDONE_IE|GCR_CDONE_IE; ++} ++ ++static irqreturn_t pxa2xx_ac97_irq(int irq, void *dev_id) ++{ ++ long status; ++ ++ status = GSR; ++ if (status) { ++ GSR = status; ++ gsr_bits |= status; ++ wake_up(&gsr_wq); ++ ++#ifdef CONFIG_PXA27x ++ /* Although we don't use those we still need to clear them ++ since they tend to spuriously trigger when MMC is used ++ (hardware bug? go figure)... */ ++ MISR = MISR_EOC; ++ PISR = PISR_EOC; ++ MCSR = MCSR_EOC; ++#endif ++ ++ return IRQ_HANDLED; ++ } ++ ++ return IRQ_NONE; ++} ++ ++struct snd_ac97_bus_ops soc_ac97_ops = { ++ .read = pxa2xx_ac97_read, ++ .write = pxa2xx_ac97_write, ++ .warm_reset = pxa2xx_ac97_warm_reset, ++ .reset = pxa2xx_ac97_cold_reset, ++}; ++ ++static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_stereo_out = { ++ .name = "AC97 PCM Stereo out", ++ .dev_addr = __PREG(PCDR), ++ .drcmr = &DRCMRTXPCDR, ++ .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | ++ DCMD_BURST32 | DCMD_WIDTH4, ++}; ++ ++static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_stereo_in = { ++ .name = "AC97 PCM Stereo in", ++ .dev_addr = __PREG(PCDR), ++ .drcmr = &DRCMRRXPCDR, ++ .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | ++ DCMD_BURST32 | DCMD_WIDTH4, ++}; ++ ++static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_aux_mono_out = { ++ .name = "AC97 Aux PCM (Slot 5) Mono out", ++ .dev_addr = __PREG(MODR), ++ .drcmr = &DRCMRTXMODR, ++ .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | ++ DCMD_BURST16 | DCMD_WIDTH2, ++}; ++ ++static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_aux_mono_in = { ++ .name = "AC97 Aux PCM (Slot 5) Mono in", ++ .dev_addr = __PREG(MODR), ++ .drcmr = &DRCMRRXMODR, ++ .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | ++ DCMD_BURST16 | DCMD_WIDTH2, ++}; ++ ++static struct pxa2xx_pcm_dma_params pxa2xx_ac97_pcm_mic_mono_in = { ++ .name = "AC97 Mic PCM (Slot 6) Mono in", ++ .dev_addr = __PREG(MCDR), ++ .drcmr = &DRCMRRXMCDR, ++ .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | ++ DCMD_BURST16 | DCMD_WIDTH2, ++}; ++ ++#ifdef CONFIG_PM ++static int pxa2xx_ac97_suspend(struct platform_device *pdev, ++ struct snd_soc_cpu_dai *dai) ++{ ++ GCR |= GCR_ACLINK_OFF; ++ pxa_set_cken(CKEN2_AC97, 0); ++ return 0; ++} ++ ++static int pxa2xx_ac97_resume(struct platform_device *pdev, ++ struct snd_soc_cpu_dai *dai) ++{ ++ pxa_gpio_mode(GPIO31_SYNC_AC97_MD); ++ pxa_gpio_mode(GPIO30_SDATA_OUT_AC97_MD); ++ pxa_gpio_mode(GPIO28_BITCLK_AC97_MD); ++ pxa_gpio_mode(GPIO29_SDATA_IN_AC97_MD); ++#ifdef CONFIG_PXA27x ++ /* Use GPIO 113 as AC97 Reset on Bulverde */ ++ pxa_gpio_mode(113 | GPIO_ALT_FN_2_OUT); ++#endif ++ pxa_set_cken(CKEN2_AC97, 1); ++ return 0; ++} ++ ++#else ++#define pxa2xx_ac97_suspend NULL ++#define pxa2xx_ac97_resume NULL ++#endif ++ ++static int pxa2xx_ac97_probe(struct platform_device *pdev) ++{ ++ int ret; ++ ++ ret = request_irq(IRQ_AC97, pxa2xx_ac97_irq, IRQF_DISABLED, "AC97", NULL); ++ if (ret < 0) ++ goto err; ++ ++ pxa_gpio_mode(GPIO31_SYNC_AC97_MD); ++ pxa_gpio_mode(GPIO30_SDATA_OUT_AC97_MD); ++ pxa_gpio_mode(GPIO28_BITCLK_AC97_MD); ++ pxa_gpio_mode(GPIO29_SDATA_IN_AC97_MD); ++#ifdef CONFIG_PXA27x ++ /* Use GPIO 113 as AC97 Reset on Bulverde */ ++ pxa_gpio_mode(113 | GPIO_ALT_FN_2_OUT); ++#endif ++ pxa_set_cken(CKEN2_AC97, 1); ++ return 0; ++ ++ err: ++ if (CKEN & CKEN2_AC97) { ++ GCR |= GCR_ACLINK_OFF; ++ free_irq(IRQ_AC97, NULL); ++ pxa_set_cken(CKEN2_AC97, 0); ++ } ++ return ret; ++} ++ ++static void pxa2xx_ac97_remove(struct platform_device *pdev) ++{ ++ GCR |= GCR_ACLINK_OFF; ++ free_irq(IRQ_AC97, NULL); ++ pxa_set_cken(CKEN2_AC97, 0); ++} ++ ++static int pxa2xx_ac97_hw_params(struct snd_pcm_substream *substream, ++ struct snd_pcm_hw_params *params) ++{ ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ ++ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ++ rtd->cpu_dai->dma_data = &pxa2xx_ac97_pcm_stereo_out; ++ else ++ rtd->cpu_dai->dma_data = &pxa2xx_ac97_pcm_stereo_in; ++ ++ return 0; ++} ++ ++static int pxa2xx_ac97_hw_aux_params(struct snd_pcm_substream *substream, ++ struct snd_pcm_hw_params *params) ++{ ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ ++ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ++ rtd->cpu_dai->dma_data = &pxa2xx_ac97_pcm_aux_mono_out; ++ else ++ rtd->cpu_dai->dma_data = &pxa2xx_ac97_pcm_aux_mono_in; ++ ++ return 0; ++} ++ ++static int pxa2xx_ac97_hw_mic_params(struct snd_pcm_substream *substream, ++ struct snd_pcm_hw_params *params) ++{ ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ ++ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ++ return -ENODEV; ++ else ++ rtd->cpu_dai->dma_data = &pxa2xx_ac97_pcm_mic_mono_in; ++ ++ return 0; ++} ++ ++/* ++ * There is only 1 physical AC97 interface for pxa2xx, but it ++ * has extra fifo's that can be used for aux DACs and ADCs. ++ */ ++struct snd_soc_cpu_dai pxa_ac97_dai[] = { ++{ ++ .name = "pxa2xx-ac97", ++ .id = 0, ++ .type = SND_SOC_DAI_AC97, ++ .probe = pxa2xx_ac97_probe, ++ .remove = pxa2xx_ac97_remove, ++ .suspend = pxa2xx_ac97_suspend, ++ .resume = pxa2xx_ac97_resume, ++ .playback = { ++ .stream_name = "AC97 Playback", ++ .channels_min = 2, ++ .channels_max = 2,}, ++ .capture = { ++ .stream_name = "AC97 Capture", ++ .channels_min = 2, ++ .channels_max = 2,}, ++ .ops = { ++ .hw_params = pxa2xx_ac97_hw_params,}, ++ .caps = { ++ .num_modes = ARRAY_SIZE(pxa2xx_ac97_modes), ++ .mode = pxa2xx_ac97_modes,}, ++}, ++{ ++ .name = "pxa2xx-ac97-aux", ++ .id = 1, ++ .type = SND_SOC_DAI_AC97, ++ .playback = { ++ .stream_name = "AC97 Aux Playback", ++ .channels_min = 1, ++ .channels_max = 1,}, ++ .capture = { ++ .stream_name = "AC97 Aux Capture", ++ .channels_min = 1, ++ .channels_max = 1,}, ++ .ops = { ++ .hw_params = pxa2xx_ac97_hw_aux_params,}, ++ .caps = { ++ .num_modes = ARRAY_SIZE(pxa2xx_ac97_modes), ++ .mode = pxa2xx_ac97_modes,}, ++}, ++{ ++ .name = "pxa2xx-ac97-mic", ++ .id = 2, ++ .type = SND_SOC_DAI_AC97, ++ .capture = { ++ .stream_name = "AC97 Mic Capture", ++ .channels_min = 1, ++ .channels_max = 1,}, ++ .ops = { ++ .hw_params = pxa2xx_ac97_hw_mic_params,}, ++ .caps = { ++ .num_modes = ARRAY_SIZE(pxa2xx_ac97_modes), ++ .mode = pxa2xx_ac97_modes,},}, ++}; ++ ++EXPORT_SYMBOL_GPL(pxa_ac97_dai); ++EXPORT_SYMBOL_GPL(soc_ac97_ops); ++ ++MODULE_AUTHOR("Nicolas Pitre"); ++MODULE_DESCRIPTION("AC97 driver for the Intel PXA2xx chip"); ++MODULE_LICENSE("GPL"); +Index: linux-2.6-pxa-new/sound/soc/pxa/pxa2xx-i2s.c +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/sound/soc/pxa/pxa2xx-i2s.c +@@ -0,0 +1,354 @@ ++/* ++ * pxa2xx-i2s.c -- ALSA Soc Audio Layer ++ * ++ * Copyright 2005 Wolfson Microelectronics PLC. ++ * Author: Liam Girdwood ++ * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com ++ * ++ * This program is free software; you can redistribute it and/or modify it ++ * under the terms of the GNU General Public License as published by the ++ * Free Software Foundation; either version 2 of the License, or (at your ++ * option) any later version. ++ * ++ * Revision history ++ * 12th Aug 2005 Initial version. ++ */ ++ ++#include <linux/init.h> ++#include <linux/module.h> ++#include <linux/device.h> ++#include <linux/delay.h> ++#include <sound/driver.h> ++#include <sound/core.h> ++#include <sound/pcm.h> ++#include <sound/initval.h> ++#include <sound/soc.h> ++ ++#include <asm/hardware.h> ++#include <asm/arch/pxa-regs.h> ++#include <asm/arch/audio.h> ++ ++#include "pxa2xx-pcm.h" ++ ++/* used to disable sysclk if external crystal is used */ ++static int extclk; ++module_param(extclk, int, 0); ++MODULE_PARM_DESC(extclk, "set to 1 to disable pxa2xx i2s sysclk"); ++ ++struct pxa_i2s_port { ++ u32 sadiv; ++ u32 sacr0; ++ u32 sacr1; ++ u32 saimr; ++ int master; ++}; ++static struct pxa_i2s_port pxa_i2s; ++ ++#define PXA_I2S_DAIFMT \ ++ (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF) ++ ++#define PXA_I2S_DIR \ ++ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE) ++ ++#define PXA_I2S_RATES \ ++ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ ++ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ ++ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) ++ ++/* priv is divider */ ++static struct snd_soc_dai_mode pxa2xx_i2s_modes[] = { ++ /* pxa2xx I2S frame and clock master modes */ ++ { ++ .fmt = PXA_I2S_DAIFMT | SND_SOC_DAIFMT_CBS_CFS, ++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, ++ .pcmrate = SNDRV_PCM_RATE_8000, ++ .pcmdir = PXA_I2S_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 256, ++ .bfs = SND_SOC_FSBD(4), ++ .priv = 0x48, ++ }, ++ { ++ .fmt = PXA_I2S_DAIFMT | SND_SOC_DAIFMT_CBS_CFS, ++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, ++ .pcmrate = SNDRV_PCM_RATE_11025, ++ .pcmdir = PXA_I2S_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 256, ++ .bfs = SND_SOC_FSBD(4), ++ .priv = 0x34, ++ }, ++ { ++ .fmt = PXA_I2S_DAIFMT | SND_SOC_DAIFMT_CBS_CFS, ++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, ++ .pcmrate = SNDRV_PCM_RATE_16000, ++ .pcmdir = PXA_I2S_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 256, ++ .bfs = SND_SOC_FSBD(4), ++ .priv = 0x24, ++ }, ++ { ++ .fmt = PXA_I2S_DAIFMT | SND_SOC_DAIFMT_CBS_CFS, ++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, ++ .pcmrate = SNDRV_PCM_RATE_22050, ++ .pcmdir = PXA_I2S_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 256, ++ .bfs = SND_SOC_FSBD(4), ++ .priv = 0x1a, ++ }, ++ { ++ .fmt = PXA_I2S_DAIFMT | SND_SOC_DAIFMT_CBS_CFS, ++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, ++ .pcmrate = SNDRV_PCM_RATE_44100, ++ .pcmdir = PXA_I2S_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 256, ++ .bfs = SND_SOC_FSBD(4), ++ .priv = 0xd, ++ }, ++ { ++ .fmt = PXA_I2S_DAIFMT | SND_SOC_DAIFMT_CBS_CFS, ++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, ++ .pcmrate = SNDRV_PCM_RATE_48000, ++ .pcmdir = PXA_I2S_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 256, ++ .bfs = SND_SOC_FSBD(4), ++ .priv = 0xc, ++ }, ++ ++ /* pxa2xx I2S frame master and clock slave mode */ ++ { ++ .fmt = PXA_I2S_DAIFMT | SND_SOC_DAIFMT_CBM_CFS, ++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, ++ .pcmrate = PXA_I2S_RATES, ++ .pcmdir = PXA_I2S_DIR, ++ .fs = SND_SOC_FS_ALL, ++ .flags = SND_SOC_DAI_BFS_RATE, ++ .bfs = 64, ++ .priv = 0x48, ++ }, ++}; ++ ++static struct pxa2xx_pcm_dma_params pxa2xx_i2s_pcm_stereo_out = { ++ .name = "I2S PCM Stereo out", ++ .dev_addr = __PREG(SADR), ++ .drcmr = &DRCMRTXSADR, ++ .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | ++ DCMD_BURST32 | DCMD_WIDTH4, ++}; ++ ++static struct pxa2xx_pcm_dma_params pxa2xx_i2s_pcm_stereo_in = { ++ .name = "I2S PCM Stereo in", ++ .dev_addr = __PREG(SADR), ++ .drcmr = &DRCMRRXSADR, ++ .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | ++ DCMD_BURST32 | DCMD_WIDTH4, ++}; ++ ++static struct pxa2xx_gpio gpio_bus[] = { ++ { /* I2S SoC Slave */ ++ .rx = GPIO29_SDATA_IN_I2S_MD, ++ .tx = GPIO30_SDATA_OUT_I2S_MD, ++ .clk = GPIO28_BITCLK_IN_I2S_MD, ++ .frm = GPIO31_SYNC_I2S_MD, ++ }, ++ { /* I2S SoC Master */ ++#ifdef CONFIG_PXA27x ++ .sys = GPIO113_I2S_SYSCLK_MD, ++#else ++ .sys = GPIO32_SYSCLK_I2S_MD, ++#endif ++ .rx = GPIO29_SDATA_IN_I2S_MD, ++ .tx = GPIO30_SDATA_OUT_I2S_MD, ++ .clk = GPIO28_BITCLK_OUT_I2S_MD, ++ .frm = GPIO31_SYNC_I2S_MD, ++ }, ++}; ++ ++static int pxa2xx_i2s_startup(struct snd_pcm_substream *substream) ++{ ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ ++ if (!rtd->cpu_dai->active) { ++ SACR0 |= SACR0_RST; ++ SACR0 = 0; ++ } ++ ++ return 0; ++} ++ ++/* wait for I2S controller to be ready */ ++static int pxa_i2s_wait(void) ++{ ++ int i; ++ ++ /* flush the Rx FIFO */ ++ for(i = 0; i < 16; i++) ++ SADR; ++ return 0; ++} ++ ++static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream, ++ struct snd_pcm_hw_params *params) ++{ ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ ++ pxa_i2s.master = 0; ++ if (rtd->cpu_dai->dai_runtime.fmt & SND_SOC_DAIFMT_CBS_CFS) ++ pxa_i2s.master = 1; ++ ++ if (pxa_i2s.master && !extclk) ++ pxa_gpio_mode(gpio_bus[pxa_i2s.master].sys); ++ ++ pxa_gpio_mode(gpio_bus[pxa_i2s.master].rx); ++ pxa_gpio_mode(gpio_bus[pxa_i2s.master].tx); ++ pxa_gpio_mode(gpio_bus[pxa_i2s.master].frm); ++ pxa_gpio_mode(gpio_bus[pxa_i2s.master].clk); ++ pxa_set_cken(CKEN8_I2S, 1); ++ pxa_i2s_wait(); ++ ++ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ++ rtd->cpu_dai->dma_data = &pxa2xx_i2s_pcm_stereo_out; ++ else ++ rtd->cpu_dai->dma_data = &pxa2xx_i2s_pcm_stereo_in; ++ ++ /* is port used by another stream */ ++ if (!(SACR0 & SACR0_ENB)) { ++ ++ SACR0 = 0; ++ SACR1 = 0; ++ if (pxa_i2s.master) ++ SACR0 |= SACR0_BCKD; ++ ++ SACR0 |= SACR0_RFTH(14) | SACR0_TFTH(1); ++ ++ if (rtd->cpu_dai->dai_runtime.fmt & SND_SOC_DAIFMT_LEFT_J) ++ SACR1 |= SACR1_AMSL; ++ } ++ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ++ SAIMR |= SAIMR_TFS; ++ else ++ SAIMR |= SAIMR_RFS; ++ ++ SADIV = rtd->cpu_dai->dai_runtime.priv; ++ return 0; ++} ++ ++static int pxa2xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd) ++{ ++ int ret = 0; ++ ++ switch (cmd) { ++ case SNDRV_PCM_TRIGGER_START: ++ SACR0 |= SACR0_ENB; ++ break; ++ case SNDRV_PCM_TRIGGER_RESUME: ++ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: ++ case SNDRV_PCM_TRIGGER_STOP: ++ case SNDRV_PCM_TRIGGER_SUSPEND: ++ case SNDRV_PCM_TRIGGER_PAUSE_PUSH: ++ break; ++ default: ++ ret = -EINVAL; ++ } ++ ++ return ret; ++} ++ ++static void pxa2xx_i2s_shutdown(struct snd_pcm_substream *substream) ++{ ++ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { ++ SACR1 |= SACR1_DRPL; ++ SAIMR &= ~SAIMR_TFS; ++ } else { ++ SACR1 |= SACR1_DREC; ++ SAIMR &= ~SAIMR_RFS; ++ } ++ ++ if (SACR1 & (SACR1_DREC | SACR1_DRPL)) { ++ SACR0 &= ~SACR0_ENB; ++ pxa_i2s_wait(); ++ pxa_set_cken(CKEN8_I2S, 0); ++ } ++} ++ ++#ifdef CONFIG_PM ++static int pxa2xx_i2s_suspend(struct platform_device *dev, ++ struct snd_soc_cpu_dai *dai) ++{ ++ if (!dai->active) ++ return 0; ++ ++ /* store registers */ ++ pxa_i2s.sacr0 = SACR0; ++ pxa_i2s.sacr1 = SACR1; ++ pxa_i2s.saimr = SAIMR; ++ pxa_i2s.sadiv = SADIV; ++ ++ /* deactivate link */ ++ SACR0 &= ~SACR0_ENB; ++ pxa_i2s_wait(); ++ return 0; ++} ++ ++static int pxa2xx_i2s_resume(struct platform_device *pdev, ++ struct snd_soc_cpu_dai *dai) ++{ ++ if (!dai->active) ++ return 0; ++ ++ pxa_i2s_wait(); ++ ++ SACR0 = pxa_i2s.sacr0 &= ~SACR0_ENB; ++ SACR1 = pxa_i2s.sacr1; ++ SAIMR = pxa_i2s.saimr; ++ SADIV = pxa_i2s.sadiv; ++ SACR0 |= SACR0_ENB; ++ ++ return 0; ++} ++ ++#else ++#define pxa2xx_i2s_suspend NULL ++#define pxa2xx_i2s_resume NULL ++#endif ++ ++/* pxa2xx I2S sysclock is always 256 FS */ ++static unsigned int pxa_i2s_config_sysclk(struct snd_soc_cpu_dai *iface, ++ struct snd_soc_clock_info *info, unsigned int clk) ++{ ++ return info->rate << 8; ++} ++ ++struct snd_soc_cpu_dai pxa_i2s_dai = { ++ .name = "pxa2xx-i2s", ++ .id = 0, ++ .type = SND_SOC_DAI_I2S, ++ .suspend = pxa2xx_i2s_suspend, ++ .resume = pxa2xx_i2s_resume, ++ .config_sysclk = pxa_i2s_config_sysclk, ++ .playback = { ++ .channels_min = 2, ++ .channels_max = 2,}, ++ .capture = { ++ .channels_min = 2, ++ .channels_max = 2,}, ++ .ops = { ++ .startup = pxa2xx_i2s_startup, ++ .shutdown = pxa2xx_i2s_shutdown, ++ .trigger = pxa2xx_i2s_trigger, ++ .hw_params = pxa2xx_i2s_hw_params,}, ++ .caps = { ++ .num_modes = ARRAY_SIZE(pxa2xx_i2s_modes), ++ .mode = pxa2xx_i2s_modes,}, ++}; ++ ++EXPORT_SYMBOL_GPL(pxa_i2s_dai); ++ ++/* Module information */ ++MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com"); ++MODULE_DESCRIPTION("pxa2xx I2S SoC Interface"); ++MODULE_LICENSE("GPL"); +Index: linux-2.6-pxa-new/sound/soc/pxa/pxa2xx-pcm.c +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/sound/soc/pxa/pxa2xx-pcm.c +@@ -0,0 +1,363 @@ ++/* ++ * linux/sound/arm/pxa2xx-pcm.c -- ALSA PCM interface for the Intel PXA2xx chip ++ * ++ * Author: Nicolas Pitre ++ * Created: Nov 30, 2004 ++ * Copyright: (C) 2004 MontaVista Software, Inc. ++ * ++ * This program is free software; you can redistribute it and/or modify ++ * it under the terms of the GNU General Public License version 2 as ++ * published by the Free Software Foundation. ++ */ ++ ++#include <linux/module.h> ++#include <linux/init.h> ++#include <linux/platform_device.h> ++#include <linux/slab.h> ++#include <linux/dma-mapping.h> ++ ++#include <sound/driver.h> ++#include <sound/core.h> ++#include <sound/pcm.h> ++#include <sound/pcm_params.h> ++#include <sound/soc.h> ++ ++#include <asm/dma.h> ++#include <asm/hardware.h> ++#include <asm/arch/pxa-regs.h> ++#include <asm/arch/audio.h> ++ ++#include "pxa2xx-pcm.h" ++ ++static const struct snd_pcm_hardware pxa2xx_pcm_hardware = { ++ .info = SNDRV_PCM_INFO_MMAP | ++ SNDRV_PCM_INFO_MMAP_VALID | ++ SNDRV_PCM_INFO_INTERLEAVED | ++ SNDRV_PCM_INFO_PAUSE | ++ SNDRV_PCM_INFO_RESUME, ++ .formats = SNDRV_PCM_FMTBIT_S16_LE | ++ SNDRV_PCM_FMTBIT_S24_LE | ++ SNDRV_PCM_FMTBIT_S32_LE, ++ .period_bytes_min = 32, ++ .period_bytes_max = 8192 - 32, ++ .periods_min = 1, ++ .periods_max = PAGE_SIZE/sizeof(pxa_dma_desc), ++ .buffer_bytes_max = 128 * 1024, ++ .fifo_size = 32, ++}; ++ ++struct pxa2xx_runtime_data { ++ int dma_ch; ++ struct pxa2xx_pcm_dma_params *params; ++ pxa_dma_desc *dma_desc_array; ++ dma_addr_t dma_desc_array_phys; ++}; ++ ++static void pxa2xx_pcm_dma_irq(int dma_ch, void *dev_id) ++{ ++ struct snd_pcm_substream *substream = dev_id; ++ struct pxa2xx_runtime_data *prtd = substream->runtime->private_data; ++ int dcsr; ++ ++ dcsr = DCSR(dma_ch); ++ DCSR(dma_ch) = dcsr & ~DCSR_STOPIRQEN; ++ ++ if (dcsr & DCSR_ENDINTR) { ++ snd_pcm_period_elapsed(substream); ++ } else { ++ printk( KERN_ERR "%s: DMA error on channel %d (DCSR=%#x)\n", ++ prtd->params->name, dma_ch, dcsr ); ++ } ++} ++ ++static int pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream, ++ struct snd_pcm_hw_params *params) ++{ ++ struct snd_pcm_runtime *runtime = substream->runtime; ++ struct pxa2xx_runtime_data *prtd = runtime->private_data; ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ struct pxa2xx_pcm_dma_params *dma = rtd->cpu_dai->dma_data; ++ size_t totsize = params_buffer_bytes(params); ++ size_t period = params_period_bytes(params); ++ pxa_dma_desc *dma_desc; ++ dma_addr_t dma_buff_phys, next_desc_phys; ++ int ret; ++ ++ /* this may get called several times by oss emulation ++ * with different params */ ++ if (prtd->params == NULL) { ++ prtd->params = dma; ++ ret = pxa_request_dma(prtd->params->name, DMA_PRIO_LOW, ++ pxa2xx_pcm_dma_irq, substream); ++ if (ret < 0) ++ return ret; ++ prtd->dma_ch = ret; ++ } else if (prtd->params != dma) { ++ pxa_free_dma(prtd->dma_ch); ++ prtd->params = dma; ++ ret = pxa_request_dma(prtd->params->name, DMA_PRIO_LOW, ++ pxa2xx_pcm_dma_irq, substream); ++ if (ret < 0) ++ return ret; ++ prtd->dma_ch = ret; ++ } ++ ++ snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); ++ runtime->dma_bytes = totsize; ++ ++ dma_desc = prtd->dma_desc_array; ++ next_desc_phys = prtd->dma_desc_array_phys; ++ dma_buff_phys = runtime->dma_addr; ++ do { ++ next_desc_phys += sizeof(pxa_dma_desc); ++ dma_desc->ddadr = next_desc_phys; ++ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { ++ dma_desc->dsadr = dma_buff_phys; ++ dma_desc->dtadr = prtd->params->dev_addr; ++ } else { ++ dma_desc->dsadr = prtd->params->dev_addr; ++ dma_desc->dtadr = dma_buff_phys; ++ } ++ if (period > totsize) ++ period = totsize; ++ dma_desc->dcmd = prtd->params->dcmd | period | DCMD_ENDIRQEN; ++ dma_desc++; ++ dma_buff_phys += period; ++ } while (totsize -= period); ++ dma_desc[-1].ddadr = prtd->dma_desc_array_phys; ++ ++ return 0; ++} ++ ++static int pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream) ++{ ++ struct pxa2xx_runtime_data *prtd = substream->runtime->private_data; ++ ++ if (prtd && prtd->params) ++ *prtd->params->drcmr = 0; ++ ++ if (prtd->dma_ch) { ++ snd_pcm_set_runtime_buffer(substream, NULL); ++ pxa_free_dma(prtd->dma_ch); ++ prtd->dma_ch = 0; ++ } ++ ++ return 0; ++} ++ ++static int pxa2xx_pcm_prepare(struct snd_pcm_substream *substream) ++{ ++ struct pxa2xx_runtime_data *prtd = substream->runtime->private_data; ++ ++ DCSR(prtd->dma_ch) &= ~DCSR_RUN; ++ DCSR(prtd->dma_ch) = 0; ++ DCMD(prtd->dma_ch) = 0; ++ *prtd->params->drcmr = prtd->dma_ch | DRCMR_MAPVLD; ++ ++ return 0; ++} ++ ++static int pxa2xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) ++{ ++ struct pxa2xx_runtime_data *prtd = substream->runtime->private_data; ++ int ret = 0; ++ ++ switch (cmd) { ++ case SNDRV_PCM_TRIGGER_START: ++ DDADR(prtd->dma_ch) = prtd->dma_desc_array_phys; ++ DCSR(prtd->dma_ch) = DCSR_RUN; ++ break; ++ ++ case SNDRV_PCM_TRIGGER_STOP: ++ case SNDRV_PCM_TRIGGER_SUSPEND: ++ case SNDRV_PCM_TRIGGER_PAUSE_PUSH: ++ DCSR(prtd->dma_ch) &= ~DCSR_RUN; ++ break; ++ ++ case SNDRV_PCM_TRIGGER_RESUME: ++ DCSR(prtd->dma_ch) |= DCSR_RUN; ++ break; ++ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: ++ DDADR(prtd->dma_ch) = prtd->dma_desc_array_phys; ++ DCSR(prtd->dma_ch) |= DCSR_RUN; ++ break; ++ ++ default: ++ ret = -EINVAL; ++ } ++ ++ return ret; ++} ++ ++static snd_pcm_uframes_t ++pxa2xx_pcm_pointer(struct snd_pcm_substream *substream) ++{ ++ struct snd_pcm_runtime *runtime = substream->runtime; ++ struct pxa2xx_runtime_data *prtd = runtime->private_data; ++ ++ dma_addr_t ptr = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? ++ DSADR(prtd->dma_ch) : DTADR(prtd->dma_ch); ++ snd_pcm_uframes_t x = bytes_to_frames(runtime, ptr - runtime->dma_addr); ++ ++ if (x == runtime->buffer_size) ++ x = 0; ++ return x; ++} ++ ++static int pxa2xx_pcm_open(struct snd_pcm_substream *substream) ++{ ++ struct snd_pcm_runtime *runtime = substream->runtime; ++ struct pxa2xx_runtime_data *prtd; ++ int ret; ++ ++ snd_soc_set_runtime_hwparams(substream, &pxa2xx_pcm_hardware); ++ ++ /* ++ * For mysterious reasons (and despite what the manual says) ++ * playback samples are lost if the DMA count is not a multiple ++ * of the DMA burst size. Let's add a rule to enforce that. ++ */ ++ ret = snd_pcm_hw_constraint_step(runtime, 0, ++ SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 32); ++ if (ret) ++ goto out; ++ ++ ret = snd_pcm_hw_constraint_step(runtime, 0, ++ SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 32); ++ if (ret) ++ goto out; ++ ++ prtd = kzalloc(sizeof(struct pxa2xx_runtime_data), GFP_KERNEL); ++ if (prtd == NULL) { ++ ret = -ENOMEM; ++ goto out; ++ } ++ ++ prtd->dma_desc_array = ++ dma_alloc_writecombine(substream->pcm->card->dev, PAGE_SIZE, ++ &prtd->dma_desc_array_phys, GFP_KERNEL); ++ if (!prtd->dma_desc_array) { ++ ret = -ENOMEM; ++ goto err1; ++ } ++ ++ runtime->private_data = prtd; ++ return 0; ++ ++ err1: ++ kfree(prtd); ++ out: ++ return ret; ++} ++ ++static int pxa2xx_pcm_close(struct snd_pcm_substream *substream) ++{ ++ struct snd_pcm_runtime *runtime = substream->runtime; ++ struct pxa2xx_runtime_data *prtd = runtime->private_data; ++ ++ dma_free_writecombine(substream->pcm->card->dev, PAGE_SIZE, ++ prtd->dma_desc_array, prtd->dma_desc_array_phys); ++ kfree(prtd); ++ return 0; ++} ++ ++static int pxa2xx_pcm_mmap(struct snd_pcm_substream *substream, ++ struct vm_area_struct *vma) ++{ ++ struct snd_pcm_runtime *runtime = substream->runtime; ++ return dma_mmap_writecombine(substream->pcm->card->dev, vma, ++ runtime->dma_area, ++ runtime->dma_addr, ++ runtime->dma_bytes); ++} ++ ++struct snd_pcm_ops pxa2xx_pcm_ops = { ++ .open = pxa2xx_pcm_open, ++ .close = pxa2xx_pcm_close, ++ .ioctl = snd_pcm_lib_ioctl, ++ .hw_params = pxa2xx_pcm_hw_params, ++ .hw_free = pxa2xx_pcm_hw_free, ++ .prepare = pxa2xx_pcm_prepare, ++ .trigger = pxa2xx_pcm_trigger, ++ .pointer = pxa2xx_pcm_pointer, ++ .mmap = pxa2xx_pcm_mmap, ++}; ++ ++static int pxa2xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) ++{ ++ struct snd_pcm_substream *substream = pcm->streams[stream].substream; ++ struct snd_dma_buffer *buf = &substream->dma_buffer; ++ size_t size = pxa2xx_pcm_hardware.buffer_bytes_max; ++ buf->dev.type = SNDRV_DMA_TYPE_DEV; ++ buf->dev.dev = pcm->card->dev; ++ buf->private_data = NULL; ++ buf->area = dma_alloc_writecombine(pcm->card->dev, size, ++ &buf->addr, GFP_KERNEL); ++ if (!buf->area) ++ return -ENOMEM; ++ buf->bytes = size; ++ return 0; ++} ++ ++static void pxa2xx_pcm_free_dma_buffers(struct snd_pcm *pcm) ++{ ++ struct snd_pcm_substream *substream; ++ struct snd_dma_buffer *buf; ++ int stream; ++ ++ for (stream = 0; stream < 2; stream++) { ++ substream = pcm->streams[stream].substream; ++ if (!substream) ++ continue; ++ ++ buf = &substream->dma_buffer; ++ if (!buf->area) ++ continue; ++ ++ dma_free_writecombine(pcm->card->dev, buf->bytes, ++ buf->area, buf->addr); ++ buf->area = NULL; ++ } ++} ++ ++static u64 pxa2xx_pcm_dmamask = DMA_32BIT_MASK; ++ ++int pxa2xx_pcm_new(struct snd_card *card, struct snd_soc_codec_dai *dai, ++ struct snd_pcm *pcm) ++{ ++ int ret = 0; ++ ++ if (!card->dev->dma_mask) ++ card->dev->dma_mask = &pxa2xx_pcm_dmamask; ++ if (!card->dev->coherent_dma_mask) ++ card->dev->coherent_dma_mask = DMA_32BIT_MASK; ++ ++ if (dai->playback.channels_min) { ++ ret = pxa2xx_pcm_preallocate_dma_buffer(pcm, ++ SNDRV_PCM_STREAM_PLAYBACK); ++ if (ret) ++ goto out; ++ } ++ ++ if (dai->capture.channels_min) { ++ ret = pxa2xx_pcm_preallocate_dma_buffer(pcm, ++ SNDRV_PCM_STREAM_CAPTURE); ++ if (ret) ++ goto out; ++ } ++ out: ++ return ret; ++} ++ ++struct snd_soc_platform pxa2xx_soc_platform = { ++ .name = "pxa2xx-audio", ++ .pcm_ops = &pxa2xx_pcm_ops, ++ .pcm_new = pxa2xx_pcm_new, ++ .pcm_free = pxa2xx_pcm_free_dma_buffers, ++}; ++ ++EXPORT_SYMBOL_GPL(pxa2xx_soc_platform); ++ ++MODULE_AUTHOR("Nicolas Pitre"); ++MODULE_DESCRIPTION("Intel PXA2xx PCM DMA module"); ++MODULE_LICENSE("GPL"); +Index: linux-2.6-pxa-new/sound/soc/pxa/pxa2xx-pcm.h +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/sound/soc/pxa/pxa2xx-pcm.h +@@ -0,0 +1,48 @@ ++/* ++ * linux/sound/arm/pxa2xx-pcm.h -- ALSA PCM interface for the Intel PXA2xx chip ++ * ++ * Author: Nicolas Pitre ++ * Created: Nov 30, 2004 ++ * Copyright: MontaVista Software, Inc. ++ * ++ * This program is free software; you can redistribute it and/or modify ++ * it under the terms of the GNU General Public License version 2 as ++ * published by the Free Software Foundation. ++ */ ++ ++#ifndef _PXA2XX_PCM_H ++#define _PXA2XX_PCM_H ++ ++struct pxa2xx_pcm_dma_params { ++ char *name; /* stream identifier */ ++ u32 dcmd; /* DMA descriptor dcmd field */ ++ volatile u32 *drcmr; /* the DMA request channel to use */ ++ u32 dev_addr; /* device physical address for DMA */ ++}; ++ ++struct pxa2xx_gpio { ++ u32 sys; ++ u32 rx; ++ u32 tx; ++ u32 clk; ++ u32 frm; ++}; ++ ++/* pxa2xx DAI ID's */ ++#define PXA2XX_DAI_AC97_HIFI 0 ++#define PXA2XX_DAI_AC97_AUX 1 ++#define PXA2XX_DAI_AC97_MIC 2 ++#define PXA2XX_DAI_I2S 0 ++#define PXA2XX_DAI_SSP1 0 ++#define PXA2XX_DAI_SSP2 1 ++#define PXA2XX_DAI_SSP3 2 ++ ++extern struct snd_soc_cpu_dai pxa_ac97_dai[3]; ++extern struct snd_soc_cpu_dai pxa_i2s_dai; ++extern struct snd_soc_cpu_dai pxa_ssp_dai[3]; ++ ++/* platform data */ ++extern struct snd_soc_platform pxa2xx_soc_platform; ++extern struct snd_ac97_bus_ops pxa2xx_ac97_ops; ++ ++#endif +Index: linux-2.6-pxa-new/sound/soc/pxa/pxa2xx-ssp.c +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/sound/soc/pxa/pxa2xx-ssp.c +@@ -0,0 +1,767 @@ ++/* ++ * pxa2xx-ssp.c -- ALSA Soc Audio Layer ++ * ++ * Copyright 2005 Wolfson Microelectronics PLC. ++ * Author: Liam Girdwood ++ * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com ++ * ++ * This program is free software; you can redistribute it and/or modify it ++ * under the terms of the GNU General Public License as published by the ++ * Free Software Foundation; either version 2 of the License, or (at your ++ * option) any later version. ++ * ++ * Revision history ++ * 12th Aug 2005 Initial version. ++ * ++ * TODO: ++ * o Fix master mode (bug) ++ * o Fix resume (bug) ++ * o Add support for other clocks ++ * o Test network mode for > 16bit sample size ++ */ ++ ++#include <linux/init.h> ++#include <linux/module.h> ++#include <linux/platform_device.h> ++ ++#include <sound/driver.h> ++#include <sound/core.h> ++#include <sound/pcm.h> ++#include <sound/initval.h> ++#include <sound/soc.h> ++ ++#include <asm/hardware.h> ++#include <asm/arch/pxa-regs.h> ++#include <asm/arch/audio.h> ++#include <asm/arch/ssp.h> ++ ++#include "pxa2xx-pcm.h" ++ ++/* ++ * SSP sysclock frequency in Hz ++ * Neither default pxa2xx PLL clocks are good for audio, hence pxa27x ++ * has audio clock. I would recommend using the pxa27x audio clock or an ++ * external clock or making the codec master to gurantee better sample rates. ++ */ ++#ifdef CONFIG_PXA27x ++static int sysclk[3] = {13000000, 13000000, 13000000}; ++#else ++static int sysclk[3] = {1843200, 1843200, 1843200}; ++#endif ++module_param_array(sysclk, int, NULL, 0); ++MODULE_PARM_DESC(sysclk, "sysclk frequency in Hz"); ++ ++/* ++ * SSP sysclock source. ++ * sysclk is ignored if audio clock is used ++ */ ++#ifdef CONFIG_PXA27x ++static int clksrc[3] = {0, 0, 0}; ++#else ++static int clksrc[3] = {0, 0, 0}; ++#endif ++module_param_array(clksrc, int, NULL, 0); ++MODULE_PARM_DESC(clksrc, ++ "sysclk source, 0 = internal PLL, 1 = ext, 2 = network, 3 = audio clock"); ++ ++/* ++ * SSP GPIO's ++ */ ++#define GPIO26_SSP1RX_MD (26 | GPIO_ALT_FN_1_IN) ++#define GPIO25_SSP1TX_MD (25 | GPIO_ALT_FN_2_OUT) ++#define GPIO23_SSP1CLKS_MD (23 | GPIO_ALT_FN_2_IN) ++#define GPIO24_SSP1FRMS_MD (24 | GPIO_ALT_FN_2_IN) ++#define GPIO23_SSP1CLKM_MD (23 | GPIO_ALT_FN_2_OUT) ++#define GPIO24_SSP1FRMM_MD (24 | GPIO_ALT_FN_2_OUT) ++ ++#define GPIO11_SSP2RX_MD (11 | GPIO_ALT_FN_2_IN) ++#define GPIO13_SSP2TX_MD (13 | GPIO_ALT_FN_1_OUT) ++#define GPIO22_SSP2CLKS_MD (22 | GPIO_ALT_FN_3_IN) ++#define GPIO88_SSP2FRMS_MD (88 | GPIO_ALT_FN_3_IN) ++#define GPIO22_SSP2CLKM_MD (22 | GPIO_ALT_FN_3_OUT) ++#define GPIO88_SSP2FRMM_MD (88 | GPIO_ALT_FN_3_OUT) ++ ++#define GPIO82_SSP3RX_MD (82 | GPIO_ALT_FN_1_IN) ++#define GPIO81_SSP3TX_MD (81 | GPIO_ALT_FN_1_OUT) ++#define GPIO84_SSP3CLKS_MD (84 | GPIO_ALT_FN_1_IN) ++#define GPIO83_SSP3FRMS_MD (83 | GPIO_ALT_FN_1_IN) ++#define GPIO84_SSP3CLKM_MD (84 | GPIO_ALT_FN_1_OUT) ++#define GPIO83_SSP3FRMM_MD (83 | GPIO_ALT_FN_1_OUT) ++ ++#define PXA_SSP_MDAIFMT \ ++ (SND_SOC_DAIFMT_DSP_B |SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBS_CFS | SND_SOC_DAIFMT_CBM_CFS | \ ++ SND_SOC_DAIFMT_CBS_CFM | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_NB_IF) ++ ++#define PXA_SSP_SDAIFMT \ ++ (SND_SOC_DAIFMT_DSP_B |SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_CBM_CFS | \ ++ SND_SOC_DAIFMT_CBS_CFM | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_NB_IF) ++ ++#define PXA_SSP_DIR \ ++ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE) ++ ++#define PXA_SSP_RATES \ ++ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ ++ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ ++ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | \ ++ SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000) ++ ++#define PXA_SSP_BITS \ ++ (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ ++ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) ++ ++/* ++ * SSP modes ++ */ ++static struct snd_soc_dai_mode pxa2xx_ssp_modes[] = { ++ /* port slave clk & frame modes */ ++ { ++ .fmt = PXA_SSP_SDAIFMT, ++ .pcmfmt = PXA_SSP_BITS, ++ .pcmrate = PXA_SSP_RATES, ++ .pcmdir = PXA_SSP_DIR, ++ .fs = SND_SOC_FS_ALL, ++ .bfs = SND_SOC_FSB_ALL, ++ }, ++ ++ /* port master clk & frame modes */ ++#ifdef CONFIG_PXA27x ++ { ++ .fmt = PXA_SSP_MDAIFMT, ++ .pcmfmt = PXA_SSP_BITS, ++ .pcmrate = SNDRV_PCM_RATE_8000, ++ .pcmdir = PXA_SSP_DIR, ++ .flags = SND_SOC_DAI_BFS_RCW, ++ .fs = 256, ++ .bfs = SND_SOC_FSBW(1), ++ }, ++ { ++ .fmt = PXA_SSP_MDAIFMT, ++ .pcmfmt = PXA_SSP_BITS, ++ .pcmrate = SNDRV_PCM_RATE_11025, ++ .pcmdir = PXA_SSP_DIR, ++ .flags = SND_SOC_DAI_BFS_RCW, ++ .fs = 256, ++ .bfs = SND_SOC_FSBW(1), ++ }, ++ { ++ .fmt = PXA_SSP_MDAIFMT, ++ .pcmfmt = PXA_SSP_BITS, ++ .pcmrate = SNDRV_PCM_RATE_16000, ++ .pcmdir = PXA_SSP_DIR, ++ .flags = SND_SOC_DAI_BFS_RCW, ++ .fs = 256, ++ .bfs = SND_SOC_FSBW(1), ++ }, ++ { ++ .fmt = PXA_SSP_MDAIFMT, ++ .pcmfmt = PXA_SSP_BITS, ++ .pcmrate = SNDRV_PCM_RATE_22050, ++ .pcmdir = PXA_SSP_DIR, ++ .flags = SND_SOC_DAI_BFS_RCW, ++ .fs = 256, ++ .bfs = SND_SOC_FSBW(1), ++ }, ++ { ++ .fmt = PXA_SSP_MDAIFMT, ++ .pcmfmt = PXA_SSP_BITS, ++ .pcmrate = SNDRV_PCM_RATE_32000, ++ .pcmdir = PXA_SSP_DIR, ++ .flags = SND_SOC_DAI_BFS_RCW, ++ .fs = 256, ++ .bfs = SND_SOC_FSBW(1), ++ }, ++ { ++ .fmt = PXA_SSP_MDAIFMT, ++ .pcmfmt = PXA_SSP_BITS, ++ .pcmrate = SNDRV_PCM_RATE_44100, ++ .pcmdir = PXA_SSP_DIR, ++ .flags = SND_SOC_DAI_BFS_RCW, ++ .fs = 256, ++ .bfs = SND_SOC_FSBW(1), ++ }, ++ { ++ .fmt = PXA_SSP_MDAIFMT, ++ .pcmfmt = PXA_SSP_BITS, ++ .pcmrate = SNDRV_PCM_RATE_48000, ++ .pcmdir = PXA_SSP_DIR, ++ .flags = SND_SOC_DAI_BFS_RCW, ++ .fs = 256, ++ .bfs = SND_SOC_FSBW(1), ++ }, ++ { ++ .fmt = PXA_SSP_MDAIFMT, ++ .pcmfmt = PXA_SSP_BITS, ++ .pcmrate = SNDRV_PCM_RATE_88200, ++ .pcmdir = PXA_SSP_DIR, ++ .flags = SND_SOC_DAI_BFS_RCW, ++ .fs = 128, ++ .bfs = SND_SOC_FSBW(1), ++ }, ++ { ++ .fmt = PXA_SSP_MDAIFMT, ++ .pcmfmt = PXA_SSP_BITS, ++ .pcmrate = SNDRV_PCM_RATE_96000, ++ .pcmdir = PXA_SSP_DIR, ++ .flags = SND_SOC_DAI_BFS_RCW, ++ .fs = 128, ++ .bfs = SND_SOC_FSBW(1), ++ }, ++#endif ++}; ++ ++static struct ssp_dev ssp[3]; ++#ifdef CONFIG_PM ++static struct ssp_state ssp_state[3]; ++#endif ++ ++static struct pxa2xx_pcm_dma_params pxa2xx_ssp1_pcm_mono_out = { ++ .name = "SSP1 PCM Mono out", ++ .dev_addr = __PREG(SSDR_P1), ++ .drcmr = &DRCMRTXSSDR, ++ .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | ++ DCMD_BURST16 | DCMD_WIDTH2, ++}; ++ ++static struct pxa2xx_pcm_dma_params pxa2xx_ssp1_pcm_mono_in = { ++ .name = "SSP1 PCM Mono in", ++ .dev_addr = __PREG(SSDR_P1), ++ .drcmr = &DRCMRRXSSDR, ++ .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | ++ DCMD_BURST16 | DCMD_WIDTH2, ++}; ++ ++static struct pxa2xx_pcm_dma_params pxa2xx_ssp1_pcm_stereo_out = { ++ .name = "SSP1 PCM Stereo out", ++ .dev_addr = __PREG(SSDR_P1), ++ .drcmr = &DRCMRTXSSDR, ++ .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | ++ DCMD_BURST16 | DCMD_WIDTH4, ++}; ++ ++static struct pxa2xx_pcm_dma_params pxa2xx_ssp1_pcm_stereo_in = { ++ .name = "SSP1 PCM Stereo in", ++ .dev_addr = __PREG(SSDR_P1), ++ .drcmr = &DRCMRRXSSDR, ++ .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | ++ DCMD_BURST16 | DCMD_WIDTH4, ++}; ++ ++static struct pxa2xx_pcm_dma_params pxa2xx_ssp2_pcm_mono_out = { ++ .name = "SSP2 PCM Mono out", ++ .dev_addr = __PREG(SSDR_P2), ++ .drcmr = &DRCMRTXSS2DR, ++ .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | ++ DCMD_BURST16 | DCMD_WIDTH2, ++}; ++ ++static struct pxa2xx_pcm_dma_params pxa2xx_ssp2_pcm_mono_in = { ++ .name = "SSP2 PCM Mono in", ++ .dev_addr = __PREG(SSDR_P2), ++ .drcmr = &DRCMRRXSS2DR, ++ .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | ++ DCMD_BURST16 | DCMD_WIDTH2, ++}; ++ ++static struct pxa2xx_pcm_dma_params pxa2xx_ssp2_pcm_stereo_out = { ++ .name = "SSP2 PCM Stereo out", ++ .dev_addr = __PREG(SSDR_P2), ++ .drcmr = &DRCMRTXSS2DR, ++ .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | ++ DCMD_BURST16 | DCMD_WIDTH4, ++}; ++ ++static struct pxa2xx_pcm_dma_params pxa2xx_ssp2_pcm_stereo_in = { ++ .name = "SSP2 PCM Stereo in", ++ .dev_addr = __PREG(SSDR_P2), ++ .drcmr = &DRCMRRXSS2DR, ++ .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | ++ DCMD_BURST16 | DCMD_WIDTH4, ++}; ++ ++static struct pxa2xx_pcm_dma_params pxa2xx_ssp3_pcm_mono_out = { ++ .name = "SSP3 PCM Mono out", ++ .dev_addr = __PREG(SSDR_P3), ++ .drcmr = &DRCMRTXSS3DR, ++ .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | ++ DCMD_BURST16 | DCMD_WIDTH2, ++}; ++ ++static struct pxa2xx_pcm_dma_params pxa2xx_ssp3_pcm_mono_in = { ++ .name = "SSP3 PCM Mono in", ++ .dev_addr = __PREG(SSDR_P3), ++ .drcmr = &DRCMRRXSS3DR, ++ .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | ++ DCMD_BURST16 | DCMD_WIDTH2, ++}; ++ ++static struct pxa2xx_pcm_dma_params pxa2xx_ssp3_pcm_stereo_out = { ++ .name = "SSP3 PCM Stereo out", ++ .dev_addr = __PREG(SSDR_P3), ++ .drcmr = &DRCMRTXSS3DR, ++ .dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG | ++ DCMD_BURST16 | DCMD_WIDTH4, ++}; ++ ++static struct pxa2xx_pcm_dma_params pxa2xx_ssp3_pcm_stereo_in = { ++ .name = "SSP3 PCM Stereo in", ++ .dev_addr = __PREG(SSDR_P3), ++ .drcmr = &DRCMRRXSS3DR, ++ .dcmd = DCMD_INCTRGADDR | DCMD_FLOWSRC | ++ DCMD_BURST16 | DCMD_WIDTH4, ++}; ++ ++static struct pxa2xx_pcm_dma_params *ssp_dma_params[3][4] = { ++ {&pxa2xx_ssp1_pcm_mono_out, &pxa2xx_ssp1_pcm_mono_in, ++ &pxa2xx_ssp1_pcm_stereo_out,&pxa2xx_ssp1_pcm_stereo_in,}, ++ {&pxa2xx_ssp2_pcm_mono_out, &pxa2xx_ssp2_pcm_mono_in, ++ &pxa2xx_ssp2_pcm_stereo_out, &pxa2xx_ssp2_pcm_stereo_in,}, ++ {&pxa2xx_ssp3_pcm_mono_out, &pxa2xx_ssp3_pcm_mono_in, ++ &pxa2xx_ssp3_pcm_stereo_out,&pxa2xx_ssp3_pcm_stereo_in,}, ++}; ++ ++static struct pxa2xx_gpio ssp_gpios[3][4] = { ++ {{ /* SSP1 SND_SOC_DAIFMT_CBM_CFM */ ++ .rx = GPIO26_SSP1RX_MD, ++ .tx = GPIO25_SSP1TX_MD, ++ .clk = (23 | GPIO_ALT_FN_2_IN), ++ .frm = (24 | GPIO_ALT_FN_2_IN), ++ }, ++ { /* SSP1 SND_SOC_DAIFMT_CBS_CFS */ ++ .rx = GPIO26_SSP1RX_MD, ++ .tx = GPIO25_SSP1TX_MD, ++ .clk = (23 | GPIO_ALT_FN_2_OUT), ++ .frm = (24 | GPIO_ALT_FN_2_OUT), ++ }, ++ { /* SSP1 SND_SOC_DAIFMT_CBS_CFM */ ++ .rx = GPIO26_SSP1RX_MD, ++ .tx = GPIO25_SSP1TX_MD, ++ .clk = (23 | GPIO_ALT_FN_2_OUT), ++ .frm = (24 | GPIO_ALT_FN_2_IN), ++ }, ++ { /* SSP1 SND_SOC_DAIFMT_CBM_CFS */ ++ .rx = GPIO26_SSP1RX_MD, ++ .tx = GPIO25_SSP1TX_MD, ++ .clk = (23 | GPIO_ALT_FN_2_IN), ++ .frm = (24 | GPIO_ALT_FN_2_OUT), ++ }}, ++ {{ /* SSP2 SND_SOC_DAIFMT_CBM_CFM */ ++ .rx = GPIO11_SSP2RX_MD, ++ .tx = GPIO13_SSP2TX_MD, ++ .clk = (22 | GPIO_ALT_FN_3_IN), ++ .frm = (88 | GPIO_ALT_FN_3_IN), ++ }, ++ { /* SSP2 SND_SOC_DAIFMT_CBS_CFS */ ++ .rx = GPIO11_SSP2RX_MD, ++ .tx = GPIO13_SSP2TX_MD, ++ .clk = (22 | GPIO_ALT_FN_3_OUT), ++ .frm = (88 | GPIO_ALT_FN_3_OUT), ++ }, ++ { /* SSP2 SND_SOC_DAIFMT_CBS_CFM */ ++ .rx = GPIO11_SSP2RX_MD, ++ .tx = GPIO13_SSP2TX_MD, ++ .clk = (22 | GPIO_ALT_FN_3_OUT), ++ .frm = (88 | GPIO_ALT_FN_3_IN), ++ }, ++ { /* SSP2 SND_SOC_DAIFMT_CBM_CFS */ ++ .rx = GPIO11_SSP2RX_MD, ++ .tx = GPIO13_SSP2TX_MD, ++ .clk = (22 | GPIO_ALT_FN_3_IN), ++ .frm = (88 | GPIO_ALT_FN_3_OUT), ++ }}, ++ {{ /* SSP3 SND_SOC_DAIFMT_CBM_CFM */ ++ .rx = GPIO82_SSP3RX_MD, ++ .tx = GPIO81_SSP3TX_MD, ++ .clk = (84 | GPIO_ALT_FN_3_IN), ++ .frm = (83 | GPIO_ALT_FN_3_IN), ++ }, ++ { /* SSP3 SND_SOC_DAIFMT_CBS_CFS */ ++ .rx = GPIO82_SSP3RX_MD, ++ .tx = GPIO81_SSP3TX_MD, ++ .clk = (84 | GPIO_ALT_FN_3_OUT), ++ .frm = (83 | GPIO_ALT_FN_3_OUT), ++ }, ++ { /* SSP3 SND_SOC_DAIFMT_CBS_CFM */ ++ .rx = GPIO82_SSP3RX_MD, ++ .tx = GPIO81_SSP3TX_MD, ++ .clk = (84 | GPIO_ALT_FN_3_OUT), ++ .frm = (83 | GPIO_ALT_FN_3_IN), ++ }, ++ { /* SSP3 SND_SOC_DAIFMT_CBM_CFS */ ++ .rx = GPIO82_SSP3RX_MD, ++ .tx = GPIO81_SSP3TX_MD, ++ .clk = (84 | GPIO_ALT_FN_3_IN), ++ .frm = (83 | GPIO_ALT_FN_3_OUT), ++ }}, ++}; ++ ++static int pxa2xx_ssp_startup(struct snd_pcm_substream *substream) ++{ ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ int ret = 0; ++ ++ if (!rtd->cpu_dai->active) { ++ ret = ssp_init (&ssp[rtd->cpu_dai->id], rtd->cpu_dai->id + 1, ++ SSP_NO_IRQ); ++ if (ret < 0) ++ return ret; ++ ssp_disable(&ssp[rtd->cpu_dai->id]); ++ } ++ return ret; ++} ++ ++static void pxa2xx_ssp_shutdown(struct snd_pcm_substream *substream) ++{ ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ ++ if (!rtd->cpu_dai->active) { ++ ssp_disable(&ssp[rtd->cpu_dai->id]); ++ ssp_exit(&ssp[rtd->cpu_dai->id]); ++ } ++} ++ ++#ifdef CONFIG_PM ++ ++#if defined (CONFIG_PXA27x) ++static int cken[3] = {CKEN23_SSP1, CKEN3_SSP2, CKEN4_SSP3}; ++#else ++static int cken[3] = {CKEN3_SSP, CKEN9_NSSP, CKEN10_ASSP}; ++#endif ++ ++static int pxa2xx_ssp_suspend(struct platform_device *pdev, ++ struct snd_soc_cpu_dai *dai) ++{ ++ if (!dai->active) ++ return 0; ++ ++ ssp_save_state(&ssp[dai->id], &ssp_state[dai->id]); ++ pxa_set_cken(cken[dai->id], 0); ++ return 0; ++} ++ ++static int pxa2xx_ssp_resume(struct platform_device *pdev, ++ struct snd_soc_cpu_dai *dai) ++{ ++ if (!dai->active) ++ return 0; ++ ++ pxa_set_cken(cken[dai->id], 1); ++ ssp_restore_state(&ssp[dai->id], &ssp_state[dai->id]); ++ ssp_enable(&ssp[dai->id]); ++ ++ return 0; ++} ++ ++#else ++#define pxa2xx_ssp_suspend NULL ++#define pxa2xx_ssp_resume NULL ++#endif ++ ++/* todo - check clk source and PLL before returning clock rate */ ++static unsigned int pxa_ssp_config_sysclk(struct snd_soc_cpu_dai *dai, ++ struct snd_soc_clock_info *info, unsigned int clk) ++{ ++ /* audio clock ? (divide by 1) */ ++ if (clksrc[dai->id] == 3) { ++ switch(info->rate){ ++ case 8000: ++ case 16000: ++ case 32000: ++ case 48000: ++ case 96000: ++ return 12288000; ++ break; ++ case 11025: ++ case 22050: ++ case 44100: ++ case 88200: ++ return 11289600; ++ break; ++ } ++ } ++ ++ /* pll */ ++ return sysclk[dai->id]; ++} ++ ++#ifdef CONFIG_PXA27x ++static u32 pxa27x_set_audio_clk(unsigned int rate, unsigned int fs) ++{ ++ u32 aclk = 0, div = 0; ++ ++ if (rate == 0 || fs == 0) ++ return 0; ++ ++ switch(rate){ ++ case 8000: ++ case 16000: ++ case 32000: ++ case 48000: ++ case 96000: ++ aclk = 0x2 << 4; ++ div = 12288000 / (rate * fs); ++ break; ++ case 11025: ++ case 22050: ++ case 44100: ++ case 88200: ++ aclk = 0x1 << 4; ++ div = 11289600 / (rate * fs); ++ break; ++ } ++ ++ aclk |= ffs(div) - 1; ++ return aclk; ++} ++#endif ++ ++static inline int get_scr(int srate, int id) ++{ ++ if (srate == 0) ++ return 0; ++ return (sysclk[id] / srate) - 1; ++} ++ ++static int pxa2xx_ssp_hw_params(struct snd_pcm_substream *substream, ++ struct snd_pcm_hw_params *params) ++{ ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ int fmt = 0, dma = 0, fs, chn = params_channels(params); ++ u32 ssp_mode = 0, ssp_setup = 0, psp_mode = 0, rate = 0; ++ ++ fs = rtd->cpu_dai->dai_runtime.fs; ++ ++ /* select correct DMA params */ ++ if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) ++ dma = 1; ++ if (chn == 2 || rtd->cpu_dai->dai_runtime.pcmfmt != PXA_SSP_BITS) ++ dma += 2; ++ rtd->cpu_dai->dma_data = ssp_dma_params[rtd->cpu_dai->id][dma]; ++ ++ /* is port used by another stream */ ++ if (SSCR0 & SSCR0_SSE) ++ return 0; ++ ++ /* bit size */ ++ switch(rtd->cpu_dai->dai_runtime.pcmfmt) { ++ case SNDRV_PCM_FMTBIT_S16_LE: ++ ssp_mode |=SSCR0_DataSize(16); ++ break; ++ case SNDRV_PCM_FMTBIT_S24_LE: ++ ssp_mode |=(SSCR0_EDSS | SSCR0_DataSize(8)); ++ /* use network mode for stereo samples > 16 bits */ ++ if (chn == 2) { ++ ssp_mode |= (SSCR0_MOD | SSCR0_SlotsPerFrm(2) << 24); ++ /* active slots 0,1 */ ++ SSTSA_P(rtd->cpu_dai->id +1) = 0x3; ++ SSRSA_P(rtd->cpu_dai->id +1) = 0x3; ++ } ++ break; ++ case SNDRV_PCM_FMTBIT_S32_LE: ++ ssp_mode |= (SSCR0_EDSS | SSCR0_DataSize(16)); ++ /* use network mode for stereo samples > 16 bits */ ++ if (chn == 2) { ++ ssp_mode |= (SSCR0_MOD | SSCR0_SlotsPerFrm(2) << 24); ++ /* active slots 0,1 */ ++ SSTSA_P(rtd->cpu_dai->id +1) = 0x3; ++ SSRSA_P(rtd->cpu_dai->id +1) = 0x3; ++ } ++ break; ++ } ++ ++ ssp_mode |= SSCR0_PSP; ++ ssp_setup = SSCR1_RxTresh(14) | SSCR1_TxTresh(1) | ++ SSCR1_TRAIL | SSCR1_RWOT; ++ ++ switch(rtd->cpu_dai->dai_runtime.fmt & SND_SOC_DAIFMT_CLOCK_MASK) { ++ case SND_SOC_DAIFMT_CBM_CFM: ++ ssp_setup |= (SSCR1_SCLKDIR | SSCR1_SFRMDIR); ++ break; ++ case SND_SOC_DAIFMT_CBM_CFS: ++ ssp_setup |= SSCR1_SCLKDIR; ++ break; ++ case SND_SOC_DAIFMT_CBS_CFM: ++ ssp_setup |= SSCR1_SFRMDIR; ++ break; ++ } ++ ++ switch(rtd->cpu_dai->dai_runtime.fmt) { ++ case SND_SOC_DAIFMT_CBS_CFS: ++ fmt = 1; ++ break; ++ case SND_SOC_DAIFMT_CBS_CFM: ++ fmt = 2; ++ break; ++ case SND_SOC_DAIFMT_CBM_CFS: ++ fmt = 3; ++ break; ++ } ++ ++ pxa_gpio_mode(ssp_gpios[rtd->cpu_dai->id][fmt].rx); ++ pxa_gpio_mode(ssp_gpios[rtd->cpu_dai->id][fmt].tx); ++ pxa_gpio_mode(ssp_gpios[rtd->cpu_dai->id][fmt].frm); ++ pxa_gpio_mode(ssp_gpios[rtd->cpu_dai->id][fmt].clk); ++ ++ switch (rtd->cpu_dai->dai_runtime.fmt & SND_SOC_DAIFMT_INV_MASK) { ++ case SND_SOC_DAIFMT_NB_NF: ++ psp_mode |= SSPSP_SFRMP | SSPSP_FSRT; ++ break; ++ } ++ ++ if (rtd->cpu_dai->dai_runtime.fmt & SND_SOC_DAIFMT_DSP_A) ++ psp_mode |= SSPSP_SCMODE(2); ++ if (rtd->cpu_dai->dai_runtime.fmt & SND_SOC_DAIFMT_DSP_B) ++ psp_mode |= SSPSP_SCMODE(3); ++ ++ switch(clksrc[rtd->cpu_dai->id]) { ++ case 2: /* network clock */ ++ ssp_mode |= SSCR0_NCS | SSCR0_MOD; ++ case 1: /* external clock */ ++ ssp_mode |= SSCR0_ECS; ++ case 0: /* internal clock */ ++ rate = get_scr(snd_soc_get_rate(rtd->cpu_dai->dai_runtime.pcmrate), ++ rtd->cpu_dai->id); ++ break; ++#ifdef CONFIG_PXA27x ++ case 3: /* audio clock */ ++ ssp_mode |= (1 << 30); ++ SSACD_P(rtd->cpu_dai->id) = (0x1 << 3) | ++ pxa27x_set_audio_clk( ++ snd_soc_get_rate(rtd->cpu_dai->dai_runtime.pcmrate), fs); ++ break; ++#endif ++ } ++ ++ ssp_config(&ssp[rtd->cpu_dai->id], ssp_mode, ssp_setup, psp_mode, ++ SSCR0_SerClkDiv(rate)); ++#if 0 ++ printk("SSCR0 %x SSCR1 %x SSTO %x SSPSP %x SSSR %x\n", ++ SSCR0_P(rtd->cpu_dai->id+1), SSCR1_P(rtd->cpu_dai->id+1), ++ SSTO_P(rtd->cpu_dai->id+1), SSPSP_P(rtd->cpu_dai->id+1), ++ SSSR_P(rtd->cpu_dai->id+1)); ++#endif ++ return 0; ++} ++ ++static int pxa2xx_ssp_trigger(struct snd_pcm_substream *substream, int cmd) ++{ ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ int ret = 0; ++ ++ switch (cmd) { ++ case SNDRV_PCM_TRIGGER_RESUME: ++ ssp_enable(&ssp[rtd->cpu_dai->id]); ++ break; ++ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: ++ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ++ SSCR1_P(rtd->cpu_dai->id+1) |= SSCR1_TSRE; ++ else ++ SSCR1_P(rtd->cpu_dai->id+1) |= SSCR1_RSRE; ++ SSSR_P(rtd->cpu_dai->id+1) |= SSSR_P(rtd->cpu_dai->id+1); ++ break; ++ case SNDRV_PCM_TRIGGER_START: ++ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ++ SSCR1_P(rtd->cpu_dai->id+1) |= SSCR1_TSRE; ++ else ++ SSCR1_P(rtd->cpu_dai->id+1) |= SSCR1_RSRE; ++ ssp_enable(&ssp[rtd->cpu_dai->id]); ++ break; ++ case SNDRV_PCM_TRIGGER_STOP: ++ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ++ SSCR1_P(rtd->cpu_dai->id+1) &= ~SSCR1_TSRE; ++ else ++ SSCR1_P(rtd->cpu_dai->id+1) &= ~SSCR1_RSRE; ++ break; ++ case SNDRV_PCM_TRIGGER_SUSPEND: ++ ssp_disable(&ssp[rtd->cpu_dai->id]); ++ break; ++ case SNDRV_PCM_TRIGGER_PAUSE_PUSH: ++ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ++ SSCR1_P(rtd->cpu_dai->id+1) &= ~SSCR1_TSRE; ++ else ++ SSCR1_P(rtd->cpu_dai->id+1) &= ~SSCR1_RSRE; ++ break; ++ ++ default: ++ ret = -EINVAL; ++ } ++#if 0 ++ printk("SSCR0 %x SSCR1 %x SSTO %x SSPSP %x SSSR %x\n", ++ SSCR0_P(rtd->cpu_dai->id+1), SSCR1_P(rtd->cpu_dai->id+1), ++ SSTO_P(rtd->cpu_dai->id+1), SSPSP_P(rtd->cpu_dai->id+1), ++ SSSR_P(rtd->cpu_dai->id+1)); ++#endif ++ return ret; ++} ++ ++struct snd_soc_cpu_dai pxa_ssp_dai[] = { ++ { .name = "pxa2xx-ssp1", ++ .id = 0, ++ .type = SND_SOC_DAI_PCM, ++ .suspend = pxa2xx_ssp_suspend, ++ .resume = pxa2xx_ssp_resume, ++ .config_sysclk = pxa_ssp_config_sysclk, ++ .playback = { ++ .channels_min = 1, ++ .channels_max = 2,}, ++ .capture = { ++ .channels_min = 1, ++ .channels_max = 2,}, ++ .ops = { ++ .startup = pxa2xx_ssp_startup, ++ .shutdown = pxa2xx_ssp_shutdown, ++ .trigger = pxa2xx_ssp_trigger, ++ .hw_params = pxa2xx_ssp_hw_params,}, ++ .caps = { ++ .mode = pxa2xx_ssp_modes, ++ .num_modes = ARRAY_SIZE(pxa2xx_ssp_modes),}, ++ }, ++ { .name = "pxa2xx-ssp2", ++ .id = 1, ++ .type = SND_SOC_DAI_PCM, ++ .suspend = pxa2xx_ssp_suspend, ++ .resume = pxa2xx_ssp_resume, ++ .config_sysclk = pxa_ssp_config_sysclk, ++ .playback = { ++ .channels_min = 1, ++ .channels_max = 2,}, ++ .capture = { ++ .channels_min = 1, ++ .channels_max = 2,}, ++ .ops = { ++ .startup = pxa2xx_ssp_startup, ++ .shutdown = pxa2xx_ssp_shutdown, ++ .trigger = pxa2xx_ssp_trigger, ++ .hw_params = pxa2xx_ssp_hw_params,}, ++ .caps = { ++ .mode = pxa2xx_ssp_modes, ++ .num_modes = ARRAY_SIZE(pxa2xx_ssp_modes),}, ++ }, ++ { .name = "pxa2xx-ssp3", ++ .id = 2, ++ .type = SND_SOC_DAI_PCM, ++ .suspend = pxa2xx_ssp_suspend, ++ .resume = pxa2xx_ssp_resume, ++ .config_sysclk = pxa_ssp_config_sysclk, ++ .playback = { ++ .channels_min = 1, ++ .channels_max = 2,}, ++ .capture = { ++ .channels_min = 1, ++ .channels_max = 2,}, ++ .ops = { ++ .startup = pxa2xx_ssp_startup, ++ .shutdown = pxa2xx_ssp_shutdown, ++ .trigger = pxa2xx_ssp_trigger, ++ .hw_params = pxa2xx_ssp_hw_params,}, ++ .caps = { ++ .mode = pxa2xx_ssp_modes, ++ .num_modes = ARRAY_SIZE(pxa2xx_ssp_modes),}, ++ }, ++}; ++ ++EXPORT_SYMBOL_GPL(pxa_ssp_dai); ++ ++/* Module information */ ++MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com"); ++MODULE_DESCRIPTION("pxa2xx SSP/PCM SoC Interface"); ++MODULE_LICENSE("GPL"); +Index: linux-2.6-pxa-new/sound/soc/pxa/spitz.c +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/sound/soc/pxa/spitz.c +@@ -0,0 +1,374 @@ ++/* ++ * spitz.c -- SoC audio for Sharp SL-Cxx00 models Spitz, Borzoi and Akita ++ * ++ * Copyright 2005 Wolfson Microelectronics PLC. ++ * Copyright 2005 Openedhand Ltd. ++ * ++ * Authors: Liam Girdwood <liam.girdwood@wolfsonmicro.com> ++ * Richard Purdie <richard@openedhand.com> ++ * ++ * This program is free software; you can redistribute it and/or modify it ++ * under the terms of the GNU General Public License as published by the ++ * Free Software Foundation; either version 2 of the License, or (at your ++ * option) any later version. ++ * ++ * Revision history ++ * 30th Nov 2005 Initial version. ++ * ++ */ ++ ++#include <linux/module.h> ++#include <linux/moduleparam.h> ++#include <linux/timer.h> ++#include <linux/interrupt.h> ++#include <linux/platform_device.h> ++#include <sound/driver.h> ++#include <sound/core.h> ++#include <sound/pcm.h> ++#include <sound/soc.h> ++#include <sound/soc-dapm.h> ++ ++#include <asm/mach-types.h> ++#include <asm/hardware/scoop.h> ++#include <asm/arch/pxa-regs.h> ++#include <asm/arch/hardware.h> ++#include <asm/arch/akita.h> ++#include <asm/arch/spitz.h> ++#include <asm/mach-types.h> ++#include "../codecs/wm8750.h" ++#include "pxa2xx-pcm.h" ++ ++#define SPITZ_HP 0 ++#define SPITZ_MIC 1 ++#define SPITZ_LINE 2 ++#define SPITZ_HEADSET 3 ++#define SPITZ_HP_OFF 4 ++#define SPITZ_SPK_ON 0 ++#define SPITZ_SPK_OFF 1 ++ ++ /* audio clock in Hz - rounded from 12.235MHz */ ++#define SPITZ_AUDIO_CLOCK 12288000 ++ ++static int spitz_jack_func; ++static int spitz_spk_func; ++ ++static void spitz_ext_control(struct snd_soc_codec *codec) ++{ ++ if (spitz_spk_func == SPITZ_SPK_ON) ++ snd_soc_dapm_set_endpoint(codec, "Ext Spk", 1); ++ else ++ snd_soc_dapm_set_endpoint(codec, "Ext Spk", 0); ++ ++ /* set up jack connection */ ++ switch (spitz_jack_func) { ++ case SPITZ_HP: ++ /* enable and unmute hp jack, disable mic bias */ ++ snd_soc_dapm_set_endpoint(codec, "Headset Jack", 0); ++ snd_soc_dapm_set_endpoint(codec, "Mic Jack", 0); ++ snd_soc_dapm_set_endpoint(codec, "Line Jack", 0); ++ snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 1); ++ set_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L); ++ set_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R); ++ break; ++ case SPITZ_MIC: ++ /* enable mic jack and bias, mute hp */ ++ snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 0); ++ snd_soc_dapm_set_endpoint(codec, "Headset Jack", 0); ++ snd_soc_dapm_set_endpoint(codec, "Line Jack", 0); ++ snd_soc_dapm_set_endpoint(codec, "Mic Jack", 1); ++ reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L); ++ reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R); ++ break; ++ case SPITZ_LINE: ++ /* enable line jack, disable mic bias and mute hp */ ++ snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 0); ++ snd_soc_dapm_set_endpoint(codec, "Headset Jack", 0); ++ snd_soc_dapm_set_endpoint(codec, "Mic Jack", 0); ++ snd_soc_dapm_set_endpoint(codec, "Line Jack", 1); ++ reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L); ++ reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R); ++ break; ++ case SPITZ_HEADSET: ++ /* enable and unmute headset jack enable mic bias, mute L hp */ ++ snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 0); ++ snd_soc_dapm_set_endpoint(codec, "Mic Jack", 1); ++ snd_soc_dapm_set_endpoint(codec, "Line Jack", 0); ++ snd_soc_dapm_set_endpoint(codec, "Headset Jack", 1); ++ reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L); ++ set_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R); ++ break; ++ case SPITZ_HP_OFF: ++ ++ /* jack removed, everything off */ ++ snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 0); ++ snd_soc_dapm_set_endpoint(codec, "Headset Jack", 0); ++ snd_soc_dapm_set_endpoint(codec, "Mic Jack", 0); ++ snd_soc_dapm_set_endpoint(codec, "Line Jack", 0); ++ reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_L); ++ reset_scoop_gpio(&spitzscoop_device.dev, SPITZ_SCP_MUTE_R); ++ break; ++ } ++ snd_soc_dapm_sync_endpoints(codec); ++} ++ ++static int spitz_startup(struct snd_pcm_substream *substream) ++{ ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ struct snd_soc_codec *codec = rtd->socdev->codec; ++ ++ /* check the jack status at stream startup */ ++ spitz_ext_control(codec); ++ return 0; ++} ++ ++static struct snd_soc_ops spitz_ops = { ++ .startup = spitz_startup, ++}; ++ ++static int spitz_get_jack(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_value *ucontrol) ++{ ++ ucontrol->value.integer.value[0] = spitz_jack_func; ++ return 0; ++} ++ ++static int spitz_set_jack(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_value *ucontrol) ++{ ++ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); ++ ++ if (spitz_jack_func == ucontrol->value.integer.value[0]) ++ return 0; ++ ++ spitz_jack_func = ucontrol->value.integer.value[0]; ++ spitz_ext_control(codec); ++ return 1; ++} ++ ++static int spitz_get_spk(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_value *ucontrol) ++{ ++ ucontrol->value.integer.value[0] = spitz_spk_func; ++ return 0; ++} ++ ++static int spitz_set_spk(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_value *ucontrol) ++{ ++ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); ++ ++ if (spitz_spk_func == ucontrol->value.integer.value[0]) ++ return 0; ++ ++ spitz_spk_func = ucontrol->value.integer.value[0]; ++ spitz_ext_control(codec); ++ return 1; ++} ++ ++static int spitz_mic_bias(struct snd_soc_dapm_widget *w, int event) ++{ ++ if (machine_is_borzoi() || machine_is_spitz()) { ++ if (SND_SOC_DAPM_EVENT_ON(event)) ++ set_scoop_gpio(&spitzscoop2_device.dev, ++ SPITZ_SCP2_MIC_BIAS); ++ else ++ reset_scoop_gpio(&spitzscoop2_device.dev, ++ SPITZ_SCP2_MIC_BIAS); ++ } ++ ++ if (machine_is_akita()) { ++ if (SND_SOC_DAPM_EVENT_ON(event)) ++ akita_set_ioexp(&akitaioexp_device.dev, ++ AKITA_IOEXP_MIC_BIAS); ++ else ++ akita_reset_ioexp(&akitaioexp_device.dev, ++ AKITA_IOEXP_MIC_BIAS); ++ } ++ return 0; ++} ++ ++/* spitz machine dapm widgets */ ++static const struct snd_soc_dapm_widget wm8750_dapm_widgets[] = { ++ SND_SOC_DAPM_HP("Headphone Jack", NULL), ++ SND_SOC_DAPM_MIC("Mic Jack", spitz_mic_bias), ++ SND_SOC_DAPM_SPK("Ext Spk", NULL), ++ SND_SOC_DAPM_LINE("Line Jack", NULL), ++ ++ /* headset is a mic and mono headphone */ ++ SND_SOC_DAPM_HP("Headset Jack", NULL), ++}; ++ ++/* Spitz machine audio_map */ ++static const char *audio_map[][3] = { ++ ++ /* headphone connected to LOUT1, ROUT1 */ ++ {"Headphone Jack", NULL, "LOUT1"}, ++ {"Headphone Jack", NULL, "ROUT1"}, ++ ++ /* headset connected to ROUT1 and LINPUT1 with bias (def below) */ ++ {"Headset Jack", NULL, "ROUT1"}, ++ ++ /* ext speaker connected to LOUT2, ROUT2 */ ++ {"Ext Spk", NULL , "ROUT2"}, ++ {"Ext Spk", NULL , "LOUT2"}, ++ ++ /* mic is connected to input 1 - with bias */ ++ {"LINPUT1", NULL, "Mic Bias"}, ++ {"Mic Bias", NULL, "Mic Jack"}, ++ ++ /* line is connected to input 1 - no bias */ ++ {"LINPUT1", NULL, "Line Jack"}, ++ ++ {NULL, NULL, NULL}, ++}; ++ ++static const char *jack_function[] = {"Headphone", "Mic", "Line", "Headset", ++ "Off"}; ++static const char *spk_function[] = {"On", "Off"}; ++static const struct soc_enum spitz_enum[] = { ++ SOC_ENUM_SINGLE_EXT(5, jack_function), ++ SOC_ENUM_SINGLE_EXT(2, spk_function), ++}; ++ ++static const struct snd_kcontrol_new wm8750_spitz_controls[] = { ++ SOC_ENUM_EXT("Jack Function", spitz_enum[0], spitz_get_jack, ++ spitz_set_jack), ++ SOC_ENUM_EXT("Speaker Function", spitz_enum[1], spitz_get_spk, ++ spitz_set_spk), ++}; ++ ++/* ++ * Logic for a wm8750 as connected on a Sharp SL-Cxx00 Device ++ */ ++static int spitz_wm8750_init(struct snd_soc_codec *codec) ++{ ++ int i, err; ++ ++ /* NC codec pins */ ++ snd_soc_dapm_set_endpoint(codec, "RINPUT1", 0); ++ snd_soc_dapm_set_endpoint(codec, "LINPUT2", 0); ++ snd_soc_dapm_set_endpoint(codec, "RINPUT2", 0); ++ snd_soc_dapm_set_endpoint(codec, "LINPUT3", 0); ++ snd_soc_dapm_set_endpoint(codec, "RINPUT3", 0); ++ snd_soc_dapm_set_endpoint(codec, "OUT3", 0); ++ snd_soc_dapm_set_endpoint(codec, "MONO", 0); ++ ++ /* Add spitz specific controls */ ++ for (i = 0; i < ARRAY_SIZE(wm8750_spitz_controls); i++) { ++ err = snd_ctl_add(codec->card, ++ snd_soc_cnew(&wm8750_spitz_controls[i], codec, NULL)); ++ if (err < 0) ++ return err; ++ } ++ ++ /* Add spitz specific widgets */ ++ for (i = 0; i < ARRAY_SIZE(wm8750_dapm_widgets); i++) { ++ snd_soc_dapm_new_control(codec, &wm8750_dapm_widgets[i]); ++ } ++ ++ /* Set up spitz specific audio path audio_map */ ++ for (i = 0; audio_map[i][0] != NULL; i++) { ++ snd_soc_dapm_connect_input(codec, audio_map[i][0], ++ audio_map[i][1], audio_map[i][2]); ++ } ++ ++ snd_soc_dapm_sync_endpoints(codec); ++ return 0; ++} ++ ++static unsigned int spitz_config_sysclk(struct snd_soc_pcm_runtime *rtd, ++ struct snd_soc_clock_info *info) ++{ ++ if (info->bclk_master & SND_SOC_DAIFMT_CBS_CFS) { ++ /* pxa2xx is i2s master */ ++ switch (info->rate) { ++ case 11025: ++ case 22050: ++ case 44100: ++ case 88200: ++ /* configure codec digital filters ++ * for 11.025, 22.05, 44.1, 88.2 */ ++ rtd->codec_dai->config_sysclk(rtd->codec_dai, info, ++ 11289600); ++ break; ++ default: ++ /* configure codec digital filters for all other rates */ ++ rtd->codec_dai->config_sysclk(rtd->codec_dai, info, ++ SPITZ_AUDIO_CLOCK); ++ break; ++ } ++ /* configure pxa2xx i2s interface clocks as master */ ++ return rtd->cpu_dai->config_sysclk(rtd->cpu_dai, info, ++ SPITZ_AUDIO_CLOCK); ++ } else { ++ /* codec is i2s master - only configure codec DAI clock */ ++ return rtd->codec_dai->config_sysclk(rtd->codec_dai, info, ++ SPITZ_AUDIO_CLOCK); ++ } ++} ++ ++/* spitz digital audio interface glue - connects codec <--> CPU */ ++static struct snd_soc_dai_link spitz_dai = { ++ .name = "wm8750", ++ .stream_name = "WM8750", ++ .cpu_dai = &pxa_i2s_dai, ++ .codec_dai = &wm8750_dai, ++ .init = spitz_wm8750_init, ++ .config_sysclk = spitz_config_sysclk, ++}; ++ ++/* spitz audio machine driver */ ++static struct snd_soc_machine snd_soc_machine_spitz = { ++ .name = "Spitz", ++ .dai_link = &spitz_dai, ++ .num_links = 1, ++ .ops = &spitz_ops, ++}; ++ ++/* spitz audio private data */ ++static struct wm8750_setup_data spitz_wm8750_setup = { ++ .i2c_address = 0x1b, ++}; ++ ++/* spitz audio subsystem */ ++static struct snd_soc_device spitz_snd_devdata = { ++ .machine = &snd_soc_machine_spitz, ++ .platform = &pxa2xx_soc_platform, ++ .codec_dev = &soc_codec_dev_wm8750, ++ .codec_data = &spitz_wm8750_setup, ++}; ++ ++static struct platform_device *spitz_snd_device; ++ ++static int __init spitz_init(void) ++{ ++ int ret; ++ ++ if (!(machine_is_spitz() || machine_is_borzoi() || machine_is_akita())) ++ return -ENODEV; ++ ++ spitz_snd_device = platform_device_alloc("soc-audio", -1); ++ if (!spitz_snd_device) ++ return -ENOMEM; ++ ++ platform_set_drvdata(spitz_snd_device, &spitz_snd_devdata); ++ spitz_snd_devdata.dev = &spitz_snd_device->dev; ++ ret = platform_device_add(spitz_snd_device); ++ ++ if (ret) ++ platform_device_put(spitz_snd_device); ++ ++ return ret; ++} ++ ++static void __exit spitz_exit(void) ++{ ++ platform_device_unregister(spitz_snd_device); ++} ++ ++module_init(spitz_init); ++module_exit(spitz_exit); ++ ++MODULE_AUTHOR("Richard Purdie"); ++MODULE_DESCRIPTION("ALSA SoC Spitz"); ++MODULE_LICENSE("GPL"); +Index: linux-2.6-pxa-new/sound/soc/pxa/tosa.c +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/sound/soc/pxa/tosa.c +@@ -0,0 +1,287 @@ ++/* ++ * tosa.c -- SoC audio for Tosa ++ * ++ * Copyright 2005 Wolfson Microelectronics PLC. ++ * Copyright 2005 Openedhand Ltd. ++ * ++ * Authors: Liam Girdwood <liam.girdwood@wolfsonmicro.com> ++ * Richard Purdie <richard@openedhand.com> ++ * ++ * This program is free software; you can redistribute it and/or modify it ++ * under the terms of the GNU General Public License as published by the ++ * Free Software Foundation; either version 2 of the License, or (at your ++ * option) any later version. ++ * ++ * Revision history ++ * 30th Nov 2005 Initial version. ++ * ++ * GPIO's ++ * 1 - Jack Insertion ++ * 5 - Hookswitch (headset answer/hang up switch) ++ * ++ */ ++ ++#include <linux/module.h> ++#include <linux/moduleparam.h> ++#include <linux/device.h> ++ ++#include <sound/driver.h> ++#include <sound/core.h> ++#include <sound/pcm.h> ++#include <sound/soc.h> ++#include <sound/soc-dapm.h> ++ ++#include <asm/mach-types.h> ++#include <asm/hardware/tmio.h> ++#include <asm/arch/pxa-regs.h> ++#include <asm/arch/hardware.h> ++#include <asm/arch/audio.h> ++#include <asm/arch/tosa.h> ++ ++#include "../codecs/wm9712.h" ++#include "pxa2xx-pcm.h" ++ ++static struct snd_soc_machine tosa; ++ ++#define TOSA_HP 0 ++#define TOSA_MIC_INT 1 ++#define TOSA_HEADSET 2 ++#define TOSA_HP_OFF 3 ++#define TOSA_SPK_ON 0 ++#define TOSA_SPK_OFF 1 ++ ++static int tosa_jack_func; ++static int tosa_spk_func; ++ ++static void tosa_ext_control(struct snd_soc_codec *codec) ++{ ++ int spk = 0, mic_int = 0, hp = 0, hs = 0; ++ ++ /* set up jack connection */ ++ switch (tosa_jack_func) { ++ case TOSA_HP: ++ hp = 1; ++ break; ++ case TOSA_MIC_INT: ++ mic_int = 1; ++ break; ++ case TOSA_HEADSET: ++ hs = 1; ++ break; ++ } ++ ++ if (tosa_spk_func == TOSA_SPK_ON) ++ spk = 1; ++ ++ snd_soc_dapm_set_endpoint(codec, "Speaker", spk); ++ snd_soc_dapm_set_endpoint(codec, "Mic (Internal)", mic_int); ++ snd_soc_dapm_set_endpoint(codec, "Headphone Jack", hp); ++ snd_soc_dapm_set_endpoint(codec, "Headset Jack", hs); ++ snd_soc_dapm_sync_endpoints(codec); ++} ++ ++static int tosa_startup(struct snd_pcm_substream *substream) ++{ ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ struct snd_soc_codec *codec = rtd->socdev->codec; ++ ++ /* check the jack status at stream startup */ ++ tosa_ext_control(codec); ++ return 0; ++} ++ ++static struct snd_soc_ops tosa_ops = { ++ .startup = tosa_startup, ++}; ++ ++static int tosa_get_jack(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_value *ucontrol) ++{ ++ ucontrol->value.integer.value[0] = tosa_jack_func; ++ return 0; ++} ++ ++static int tosa_set_jack(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_value *ucontrol) ++{ ++ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); ++ ++ if (tosa_jack_func == ucontrol->value.integer.value[0]) ++ return 0; ++ ++ tosa_jack_func = ucontrol->value.integer.value[0]; ++ tosa_ext_control(codec); ++ return 1; ++} ++ ++static int tosa_get_spk(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_value *ucontrol) ++{ ++ ucontrol->value.integer.value[0] = tosa_spk_func; ++ return 0; ++} ++ ++static int tosa_set_spk(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_value *ucontrol) ++{ ++ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); ++ ++ if (tosa_spk_func == ucontrol->value.integer.value[0]) ++ return 0; ++ ++ tosa_spk_func = ucontrol->value.integer.value[0]; ++ tosa_ext_control(codec); ++ return 1; ++} ++ ++/* tosa dapm event handlers */ ++static int tosa_hp_event(struct snd_soc_dapm_widget *w, int event) ++{ ++ if (SND_SOC_DAPM_EVENT_ON(event)) ++ set_tc6393_gpio(&tc6393_device.dev,TOSA_TC6393_L_MUTE); ++ else ++ reset_tc6393_gpio(&tc6393_device.dev,TOSA_TC6393_L_MUTE); ++ return 0; ++} ++ ++/* tosa machine dapm widgets */ ++static const struct snd_soc_dapm_widget tosa_dapm_widgets[] = { ++SND_SOC_DAPM_HP("Headphone Jack", tosa_hp_event), ++SND_SOC_DAPM_HP("Headset Jack", NULL), ++SND_SOC_DAPM_MIC("Mic (Internal)", NULL), ++SND_SOC_DAPM_SPK("Speaker", NULL), ++}; ++ ++/* tosa audio map */ ++static const char *audio_map[][3] = { ++ ++ /* headphone connected to HPOUTL, HPOUTR */ ++ {"Headphone Jack", NULL, "HPOUTL"}, ++ {"Headphone Jack", NULL, "HPOUTR"}, ++ ++ /* ext speaker connected to LOUT2, ROUT2 */ ++ {"Speaker", NULL, "LOUT2"}, ++ {"Speaker", NULL, "ROUT2"}, ++ ++ /* internal mic is connected to mic1, mic2 differential - with bias */ ++ {"MIC1", NULL, "Mic Bias"}, ++ {"MIC2", NULL, "Mic Bias"}, ++ {"Mic Bias", NULL, "Mic (Internal)"}, ++ ++ /* headset is connected to HPOUTR, and LINEINR with bias */ ++ {"Headset Jack", NULL, "HPOUTR"}, ++ {"LINEINR", NULL, "Mic Bias"}, ++ {"Mic Bias", NULL, "Headset Jack"}, ++ ++ {NULL, NULL, NULL}, ++}; ++ ++static const char *jack_function[] = {"Headphone", "Mic", "Line", "Headset", ++ "Off"}; ++static const char *spk_function[] = {"On", "Off"}; ++static const struct soc_enum tosa_enum[] = { ++ SOC_ENUM_SINGLE_EXT(5, jack_function), ++ SOC_ENUM_SINGLE_EXT(2, spk_function), ++}; ++ ++static const struct snd_kcontrol_new tosa_controls[] = { ++ SOC_ENUM_EXT("Jack Function", tosa_enum[0], tosa_get_jack, ++ tosa_set_jack), ++ SOC_ENUM_EXT("Speaker Function", tosa_enum[1], tosa_get_spk, ++ tosa_set_spk), ++}; ++ ++static int tosa_ac97_init(struct snd_soc_codec *codec) ++{ ++ int i, err; ++ ++ snd_soc_dapm_set_endpoint(codec, "OUT3", 0); ++ snd_soc_dapm_set_endpoint(codec, "MONOOUT", 0); ++ ++ /* add tosa specific controls */ ++ for (i = 0; i < ARRAY_SIZE(tosa_controls); i++) { ++ err = snd_ctl_add(codec->card, ++ snd_soc_cnew(&tosa_controls[i],codec, NULL)); ++ if (err < 0) ++ return err; ++ } ++ ++ /* add tosa specific widgets */ ++ for (i = 0; i < ARRAY_SIZE(tosa_dapm_widgets); i++) { ++ snd_soc_dapm_new_control(codec, &tosa_dapm_widgets[i]); ++ } ++ ++ /* set up tosa specific audio path audio_map */ ++ for (i = 0; audio_map[i][0] != NULL; i++) { ++ snd_soc_dapm_connect_input(codec, audio_map[i][0], ++ audio_map[i][1], audio_map[i][2]); ++ } ++ ++ snd_soc_dapm_sync_endpoints(codec); ++ return 0; ++} ++ ++static struct snd_soc_dai_link tosa_dai[] = { ++{ ++ .name = "AC97", ++ .stream_name = "AC97 HiFi", ++ .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI], ++ .codec_dai = &wm9712_dai[WM9712_DAI_AC97_HIFI], ++ .init = tosa_ac97_init, ++}, ++{ ++ .name = "AC97 Aux", ++ .stream_name = "AC97 Aux", ++ .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX], ++ .codec_dai = &wm9712_dai[WM9712_DAI_AC97_AUX], ++}, ++}; ++ ++static struct snd_soc_machine tosa = { ++ .name = "Tosa", ++ .dai_link = tosa_dai, ++ .num_links = ARRAY_SIZE(tosa_dai), ++ .ops = &tosa_ops, ++}; ++ ++static struct snd_soc_device tosa_snd_devdata = { ++ .machine = &tosa, ++ .platform = &pxa2xx_soc_platform, ++ .codec_dev = &soc_codec_dev_wm9712, ++}; ++ ++static struct platform_device *tosa_snd_device; ++ ++static int __init tosa_init(void) ++{ ++ int ret; ++ ++ if (!machine_is_tosa()) ++ return -ENODEV; ++ ++ tosa_snd_device = platform_device_alloc("soc-audio", -1); ++ if (!tosa_snd_device) ++ return -ENOMEM; ++ ++ platform_set_drvdata(tosa_snd_device, &tosa_snd_devdata); ++ tosa_snd_devdata.dev = &tosa_snd_device->dev; ++ ret = platform_device_add(tosa_snd_device); ++ ++ if (ret) ++ platform_device_put(tosa_snd_device); ++ ++ return ret; ++} ++ ++static void __exit tosa_exit(void) ++{ ++ platform_device_unregister(tosa_snd_device); ++} ++ ++module_init(tosa_init); ++module_exit(tosa_exit); ++ ++/* Module information */ ++MODULE_AUTHOR("Richard Purdie"); ++MODULE_DESCRIPTION("ALSA SoC Tosa"); ++MODULE_LICENSE("GPL"); +Index: linux-2.6-pxa-new/sound/soc/soc-dapm.c +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/sound/soc/soc-dapm.c +@@ -0,0 +1,1327 @@ ++/* ++ * soc-dapm.c -- ALSA SoC Dynamic Audio Power Management ++ * ++ * Copyright 2005 Wolfson Microelectronics PLC. ++ * Author: Liam Girdwood ++ * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com ++ * ++ * This program is free software; you can redistribute it and/or modify it ++ * under the terms of the GNU General Public License as published by the ++ * Free Software Foundation; either version 2 of the License, or (at your ++ * option) any later version. ++ * ++ * Revision history ++ * 12th Aug 2005 Initial version. ++ * 25th Oct 2005 Implemented path power domain. ++ * 18th Dec 2005 Implemented machine and stream level power domain. ++ * ++ * Features: ++ * o Changes power status of internal codec blocks depending on the ++ * dynamic configuration of codec internal audio paths and active ++ * DAC's/ADC's. ++ * o Platform power domain - can support external components i.e. amps and ++ * mic/meadphone insertion events. ++ * o Automatic Mic Bias support ++ * o Jack insertion power event initiation - e.g. hp insertion will enable ++ * sinks, dacs, etc ++ * o Delayed powerdown of audio susbsytem to reduce pops between a quick ++ * device reopen. ++ * ++ * Todo: ++ * o DAPM power change sequencing - allow for configurable per ++ * codec sequences. ++ * o Support for analogue bias optimisation. ++ * o Support for reduced codec oversampling rates. ++ * o Support for reduced codec bias currents. ++ */ ++ ++#include <linux/module.h> ++#include <linux/moduleparam.h> ++#include <linux/init.h> ++#include <linux/delay.h> ++#include <linux/pm.h> ++#include <linux/bitops.h> ++#include <linux/platform_device.h> ++#include <linux/jiffies.h> ++#include <sound/driver.h> ++#include <sound/core.h> ++#include <sound/pcm.h> ++#include <sound/pcm_params.h> ++#include <sound/soc-dapm.h> ++#include <sound/initval.h> ++ ++/* debug */ ++#define DAPM_DEBUG 0 ++#if DAPM_DEBUG ++#define dump_dapm(codec, action) dbg_dump_dapm(codec, action) ++#define dbg(format, arg...) printk(format, ## arg) ++#else ++#define dump_dapm(codec, action) ++#define dbg(format, arg...) ++#endif ++ ++#define POP_DEBUG 0 ++#if POP_DEBUG ++#define POP_TIME 500 /* 500 msecs - change if pop debug is too fast */ ++#define pop_wait(time) schedule_timeout_interruptible(msecs_to_jiffies(time)) ++#define pop_dbg(format, arg...) printk(format, ## arg); pop_wait(POP_TIME) ++#else ++#define pop_dbg(format, arg...) ++#define pop_wait(time) ++#endif ++ ++/* dapm power sequences - make this per codec in the future */ ++static int dapm_up_seq[] = { ++ snd_soc_dapm_pre, snd_soc_dapm_micbias, snd_soc_dapm_mic, ++ snd_soc_dapm_mux, snd_soc_dapm_dac, snd_soc_dapm_mixer, snd_soc_dapm_pga, ++ snd_soc_dapm_adc, snd_soc_dapm_hp, snd_soc_dapm_spk, snd_soc_dapm_post ++}; ++static int dapm_down_seq[] = { ++ snd_soc_dapm_pre, snd_soc_dapm_adc, snd_soc_dapm_hp, snd_soc_dapm_spk, ++ snd_soc_dapm_pga, snd_soc_dapm_mixer, snd_soc_dapm_dac, snd_soc_dapm_mic, ++ snd_soc_dapm_micbias, snd_soc_dapm_mux, snd_soc_dapm_post ++}; ++ ++static int dapm_status = 1; ++module_param(dapm_status, int, 0); ++MODULE_PARM_DESC(dapm_status, "enable DPM sysfs entries"); ++ ++/* create a new dapm widget */ ++static struct snd_soc_dapm_widget *dapm_cnew_widget( ++ const struct snd_soc_dapm_widget *_widget) ++{ ++ struct snd_soc_dapm_widget* widget; ++ widget = kmalloc(sizeof(struct snd_soc_dapm_widget), GFP_KERNEL); ++ if (!widget) ++ return NULL; ++ ++ memcpy(widget, _widget, sizeof(struct snd_soc_dapm_widget)); ++ return widget; ++} ++ ++/* set up initial codec paths */ ++static void dapm_set_path_status(struct snd_soc_dapm_widget *w, ++ struct snd_soc_dapm_path *p, int i) ++{ ++ switch (w->id) { ++ case snd_soc_dapm_switch: ++ case snd_soc_dapm_mixer: { ++ int val; ++ int reg = w->kcontrols[i].private_value & 0xff; ++ int shift = (w->kcontrols[i].private_value >> 8) & 0x0f; ++ int mask = (w->kcontrols[i].private_value >> 16) & 0xff; ++ int invert = (w->kcontrols[i].private_value >> 24) & 0x01; ++ ++ val = snd_soc_read(w->codec, reg); ++ val = (val >> shift) & mask; ++ ++ if ((invert && !val) || (!invert && val)) ++ p->connect = 1; ++ else ++ p->connect = 0; ++ } ++ break; ++ case snd_soc_dapm_mux: { ++ struct soc_enum *e = (struct soc_enum *)w->kcontrols[i].private_value; ++ int val, item, bitmask; ++ ++ for (bitmask = 1; bitmask < e->mask; bitmask <<= 1) ++ ; ++ val = snd_soc_read(w->codec, e->reg); ++ item = (val >> e->shift_l) & (bitmask - 1); ++ ++ p->connect = 0; ++ for (i = 0; i < e->mask; i++) { ++ if (!(strcmp(p->name, e->texts[i])) && item == i) ++ p->connect = 1; ++ } ++ } ++ break; ++ /* does not effect routing - always connected */ ++ case snd_soc_dapm_pga: ++ case snd_soc_dapm_output: ++ case snd_soc_dapm_adc: ++ case snd_soc_dapm_input: ++ case snd_soc_dapm_dac: ++ case snd_soc_dapm_micbias: ++ case snd_soc_dapm_vmid: ++ p->connect = 1; ++ break; ++ /* does effect routing - dynamically connected */ ++ case snd_soc_dapm_hp: ++ case snd_soc_dapm_mic: ++ case snd_soc_dapm_spk: ++ case snd_soc_dapm_line: ++ case snd_soc_dapm_pre: ++ case snd_soc_dapm_post: ++ p->connect = 0; ++ break; ++ } ++} ++ ++/* connect mux widget to it's interconnecting audio paths */ ++static int dapm_connect_mux(struct snd_soc_codec *codec, ++ struct snd_soc_dapm_widget *src, struct snd_soc_dapm_widget *dest, ++ struct snd_soc_dapm_path *path, const char *control_name, ++ const struct snd_kcontrol_new *kcontrol) ++{ ++ struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; ++ int i; ++ ++ for (i = 0; i < e->mask; i++) { ++ if (!(strcmp(control_name, e->texts[i]))) { ++ list_add(&path->list, &codec->dapm_paths); ++ list_add(&path->list_sink, &dest->sources); ++ list_add(&path->list_source, &src->sinks); ++ path->name = (char*)e->texts[i]; ++ dapm_set_path_status(dest, path, 0); ++ return 0; ++ } ++ } ++ ++ return -ENODEV; ++} ++ ++/* connect mixer widget to it's interconnecting audio paths */ ++static int dapm_connect_mixer(struct snd_soc_codec *codec, ++ struct snd_soc_dapm_widget *src, struct snd_soc_dapm_widget *dest, ++ struct snd_soc_dapm_path *path, const char *control_name) ++{ ++ int i; ++ ++ /* search for mixer kcontrol */ ++ for (i = 0; i < dest->num_kcontrols; i++) { ++ if (!strcmp(control_name, dest->kcontrols[i].name)) { ++ list_add(&path->list, &codec->dapm_paths); ++ list_add(&path->list_sink, &dest->sources); ++ list_add(&path->list_source, &src->sinks); ++ path->name = dest->kcontrols[i].name; ++ dapm_set_path_status(dest, path, i); ++ return 0; ++ } ++ } ++ return -ENODEV; ++} ++ ++/* update dapm codec register bits */ ++static int dapm_update_bits(struct snd_soc_dapm_widget *widget) ++{ ++ int change, power; ++ unsigned short old, new; ++ struct snd_soc_codec *codec = widget->codec; ++ ++ /* check for valid widgets */ ++ if (widget->reg < 0 || widget->id == snd_soc_dapm_input || ++ widget->id == snd_soc_dapm_output || ++ widget->id == snd_soc_dapm_hp || ++ widget->id == snd_soc_dapm_mic || ++ widget->id == snd_soc_dapm_line || ++ widget->id == snd_soc_dapm_spk) ++ return 0; ++ ++ power = widget->power; ++ if (widget->invert) ++ power = (power ? 0:1); ++ ++ old = snd_soc_read(codec, widget->reg); ++ new = (old & ~(0x1 << widget->shift)) | (power << widget->shift); ++ ++ change = old != new; ++ if (change) { ++ pop_dbg("pop test %s : %s in %d ms\n", widget->name, ++ widget->power ? "on" : "off", POP_TIME); ++ snd_soc_write(codec, widget->reg, new); ++ pop_wait(POP_TIME); ++ } ++ dbg("reg old %x new %x change %d\n", old, new, change); ++ return change; ++} ++ ++/* ramps the volume up or down to minimise pops before or after a ++ * DAPM power event */ ++static int dapm_set_pga(struct snd_soc_dapm_widget *widget, int power) ++{ ++ const struct snd_kcontrol_new *k = widget->kcontrols; ++ ++ if (widget->muted && !power) ++ return 0; ++ if (!widget->muted && power) ++ return 0; ++ ++ if (widget->num_kcontrols && k) { ++ int reg = k->private_value & 0xff; ++ int shift = (k->private_value >> 8) & 0x0f; ++ int mask = (k->private_value >> 16) & 0xff; ++ int invert = (k->private_value >> 24) & 0x01; ++ ++ if (power) { ++ int i; ++ /* power up has happended, increase volume to last level */ ++ if (invert) { ++ for (i = mask; i > widget->saved_value; i--) ++ snd_soc_update_bits(widget->codec, reg, mask, i); ++ } else { ++ for (i = 0; i < widget->saved_value; i++) ++ snd_soc_update_bits(widget->codec, reg, mask, i); ++ } ++ widget->muted = 0; ++ } else { ++ /* power down is about to occur, decrease volume to mute */ ++ int val = snd_soc_read(widget->codec, reg); ++ int i = widget->saved_value = (val >> shift) & mask; ++ if (invert) { ++ for (; i < mask; i++) ++ snd_soc_update_bits(widget->codec, reg, mask, i); ++ } else { ++ for (; i > 0; i--) ++ snd_soc_update_bits(widget->codec, reg, mask, i); ++ } ++ widget->muted = 1; ++ } ++ } ++ return 0; ++} ++ ++/* create new dapm mixer control */ ++static int dapm_new_mixer(struct snd_soc_codec *codec, ++ struct snd_soc_dapm_widget *w) ++{ ++ int i, ret = 0; ++ char name[32]; ++ struct snd_soc_dapm_path *path; ++ ++ /* add kcontrol */ ++ for (i = 0; i < w->num_kcontrols; i++) { ++ ++ /* match name */ ++ list_for_each_entry(path, &w->sources, list_sink) { ++ ++ /* mixer/mux paths name must match control name */ ++ if (path->name != (char*)w->kcontrols[i].name) ++ continue; ++ ++ /* add dapm control with long name */ ++ snprintf(name, 32, "%s %s", w->name, w->kcontrols[i].name); ++ path->long_name = kstrdup (name, GFP_KERNEL); ++ if (path->long_name == NULL) ++ return -ENOMEM; ++ ++ path->kcontrol = snd_soc_cnew(&w->kcontrols[i], w, ++ path->long_name); ++ ret = snd_ctl_add(codec->card, path->kcontrol); ++ if (ret < 0) { ++ printk(KERN_ERR "asoc: failed to add dapm kcontrol %s\n", ++ path->long_name); ++ kfree(path->long_name); ++ path->long_name = NULL; ++ return ret; ++ } ++ } ++ } ++ return ret; ++} ++ ++/* create new dapm mux control */ ++static int dapm_new_mux(struct snd_soc_codec *codec, ++ struct snd_soc_dapm_widget *w) ++{ ++ struct snd_soc_dapm_path *path = NULL; ++ struct snd_kcontrol *kcontrol; ++ int ret = 0; ++ ++ if (!w->num_kcontrols) { ++ printk(KERN_ERR "asoc: mux %s has no controls\n", w->name); ++ return -EINVAL; ++ } ++ ++ kcontrol = snd_soc_cnew(&w->kcontrols[0], w, w->name); ++ ret = snd_ctl_add(codec->card, kcontrol); ++ if (ret < 0) ++ goto err; ++ ++ list_for_each_entry(path, &w->sources, list_sink) ++ path->kcontrol = kcontrol; ++ ++ return ret; ++ ++err: ++ printk(KERN_ERR "asoc: failed to add kcontrol %s\n", w->name); ++ return ret; ++} ++ ++/* create new dapm volume control */ ++static int dapm_new_pga(struct snd_soc_codec *codec, ++ struct snd_soc_dapm_widget *w) ++{ ++ struct snd_kcontrol *kcontrol; ++ int ret = 0; ++ ++ if (!w->num_kcontrols) ++ return -EINVAL; ++ ++ kcontrol = snd_soc_cnew(&w->kcontrols[0], w, w->name); ++ ret = snd_ctl_add(codec->card, kcontrol); ++ if (ret < 0) { ++ printk(KERN_ERR "asoc: failed to add kcontrol %s\n", w->name); ++ return ret; ++ } ++ ++ return ret; ++} ++ ++/* reset 'walked' bit for each dapm path */ ++static inline void dapm_clear_walk(struct snd_soc_codec *codec) ++{ ++ struct snd_soc_dapm_path *p; ++ ++ list_for_each_entry(p, &codec->dapm_paths, list) ++ p->walked = 0; ++} ++ ++/* ++ * Recursively check for a completed path to an active or physically connected ++ * output widget. Returns number of complete paths. ++ */ ++static int is_connected_output_ep(struct snd_soc_dapm_widget *widget) ++{ ++ struct snd_soc_dapm_path *path; ++ int con = 0; ++ ++ if (widget->id == snd_soc_dapm_adc && widget->active) ++ return 1; ++ ++ if (widget->connected) { ++ /* connected pin ? */ ++ if (widget->id == snd_soc_dapm_output && !widget->ext) ++ return 1; ++ ++ /* connected jack or spk ? */ ++ if (widget->id == snd_soc_dapm_hp || widget->id == snd_soc_dapm_spk || ++ widget->id == snd_soc_dapm_line) ++ return 1; ++ } ++ ++ list_for_each_entry(path, &widget->sinks, list_source) { ++ if (path->walked) ++ continue; ++ ++ if (path->sink && path->connect) { ++ path->walked = 1; ++ con += is_connected_output_ep(path->sink); ++ } ++ } ++ ++ return con; ++} ++ ++/* ++ * Recursively check for a completed path to an active or physically connected ++ * input widget. Returns number of complete paths. ++ */ ++static int is_connected_input_ep(struct snd_soc_dapm_widget *widget) ++{ ++ struct snd_soc_dapm_path *path; ++ int con = 0; ++ ++ /* active stream ? */ ++ if (widget->id == snd_soc_dapm_dac && widget->active) ++ return 1; ++ ++ if (widget->connected) { ++ /* connected pin ? */ ++ if (widget->id == snd_soc_dapm_input && !widget->ext) ++ return 1; ++ ++ /* connected VMID/Bias for lower pops */ ++ if (widget->id == snd_soc_dapm_vmid) ++ return 1; ++ ++ /* connected jack ? */ ++ if (widget->id == snd_soc_dapm_mic || widget->id == snd_soc_dapm_line) ++ return 1; ++ } ++ ++ list_for_each_entry(path, &widget->sources, list_sink) { ++ if (path->walked) ++ continue; ++ ++ if (path->source && path->connect) { ++ path->walked = 1; ++ con += is_connected_input_ep(path->source); ++ } ++ } ++ ++ return con; ++} ++ ++/* ++ * Scan each dapm widget for complete audio path. ++ * A complete path is a route that has valid endpoints i.e.:- ++ * ++ * o DAC to output pin. ++ * o Input Pin to ADC. ++ * o Input pin to Output pin (bypass, sidetone) ++ * o DAC to ADC (loopback). ++ */ ++int dapm_power_widgets(struct snd_soc_codec *codec, int event) ++{ ++ struct snd_soc_dapm_widget *w; ++ int in, out, i, c = 1, *seq = NULL, ret = 0, power_change, power; ++ ++ /* do we have a sequenced stream event */ ++ if (event == SND_SOC_DAPM_STREAM_START) { ++ c = ARRAY_SIZE(dapm_up_seq); ++ seq = dapm_up_seq; ++ } else if (event == SND_SOC_DAPM_STREAM_STOP) { ++ c = ARRAY_SIZE(dapm_down_seq); ++ seq = dapm_down_seq; ++ } ++ ++ for(i = 0; i < c; i++) { ++ list_for_each_entry(w, &codec->dapm_widgets, list) { ++ ++ /* is widget in stream order */ ++ if (seq && seq[i] && w->id != seq[i]) ++ continue; ++ ++ /* vmid - no action */ ++ if (w->id == snd_soc_dapm_vmid) ++ continue; ++ ++ /* active ADC */ ++ if (w->id == snd_soc_dapm_adc && w->active) { ++ in = is_connected_input_ep(w); ++ dapm_clear_walk(w->codec); ++ w->power = (in != 0) ? 1 : 0; ++ dapm_update_bits(w); ++ continue; ++ } ++ ++ /* active DAC */ ++ if (w->id == snd_soc_dapm_dac && w->active) { ++ out = is_connected_output_ep(w); ++ dapm_clear_walk(w->codec); ++ w->power = (out != 0) ? 1 : 0; ++ dapm_update_bits(w); ++ continue; ++ } ++ ++ /* programmable gain/attenuation */ ++ if (w->id == snd_soc_dapm_pga) { ++ int on; ++ in = is_connected_input_ep(w); ++ dapm_clear_walk(w->codec); ++ out = is_connected_output_ep(w); ++ dapm_clear_walk(w->codec); ++ w->power = on = (out != 0 && in != 0) ? 1 : 0; ++ ++ if (!on) ++ dapm_set_pga(w, on); /* lower volume to reduce pops */ ++ dapm_update_bits(w); ++ if (on) ++ dapm_set_pga(w, on); /* restore volume from zero */ ++ ++ continue; ++ } ++ ++ /* pre and post event widgets */ ++ if (w->id == snd_soc_dapm_pre) { ++ if (!w->event) ++ continue; ++ ++ if (event == SND_SOC_DAPM_STREAM_START) { ++ ret = w->event(w, SND_SOC_DAPM_PRE_PMU); ++ if (ret < 0) ++ return ret; ++ } else if (event == SND_SOC_DAPM_STREAM_STOP) { ++ ret = w->event(w, SND_SOC_DAPM_PRE_PMD); ++ if (ret < 0) ++ return ret; ++ } ++ continue; ++ } ++ if (w->id == snd_soc_dapm_post) { ++ if (!w->event) ++ continue; ++ ++ if (event == SND_SOC_DAPM_STREAM_START) { ++ ret = w->event(w, SND_SOC_DAPM_POST_PMU); ++ if (ret < 0) ++ return ret; ++ } else if (event == SND_SOC_DAPM_STREAM_STOP) { ++ ret = w->event(w, SND_SOC_DAPM_POST_PMD); ++ if (ret < 0) ++ return ret; ++ } ++ continue; ++ } ++ ++ /* all other widgets */ ++ in = is_connected_input_ep(w); ++ dapm_clear_walk(w->codec); ++ out = is_connected_output_ep(w); ++ dapm_clear_walk(w->codec); ++ power = (out != 0 && in != 0) ? 1 : 0; ++ power_change = (w->power == power) ? 0: 1; ++ w->power = power; ++ ++ /* call any power change event handlers */ ++ if (power_change) { ++ if (w->event) { ++ dbg("power %s event for %s flags %x\n", ++ w->power ? "on" : "off", w->name, w->event_flags); ++ if (power) { ++ /* power up event */ ++ if (w->event_flags & SND_SOC_DAPM_PRE_PMU) { ++ ret = w->event(w, SND_SOC_DAPM_PRE_PMU); ++ if (ret < 0) ++ return ret; ++ } ++ dapm_update_bits(w); ++ if (w->event_flags & SND_SOC_DAPM_POST_PMU){ ++ ret = w->event(w, SND_SOC_DAPM_POST_PMU); ++ if (ret < 0) ++ return ret; ++ } ++ } else { ++ /* power down event */ ++ if (w->event_flags & SND_SOC_DAPM_PRE_PMD) { ++ ret = w->event(w, SND_SOC_DAPM_PRE_PMD); ++ if (ret < 0) ++ return ret; ++ } ++ dapm_update_bits(w); ++ if (w->event_flags & SND_SOC_DAPM_POST_PMD) { ++ ret = w->event(w, SND_SOC_DAPM_POST_PMD); ++ if (ret < 0) ++ return ret; ++ } ++ } ++ } else ++ /* no event handler */ ++ dapm_update_bits(w); ++ } ++ } ++ } ++ ++ return ret; ++} ++ ++#if DAPM_DEBUG ++static void dbg_dump_dapm(struct snd_soc_codec* codec, const char *action) ++{ ++ struct snd_soc_dapm_widget *w; ++ struct snd_soc_dapm_path *p = NULL; ++ int in, out; ++ ++ printk("DAPM %s %s\n", codec->name, action); ++ ++ list_for_each_entry(w, &codec->dapm_widgets, list) { ++ ++ /* only display widgets that effect routing */ ++ switch (w->id) { ++ case snd_soc_dapm_pre: ++ case snd_soc_dapm_post: ++ case snd_soc_dapm_vmid: ++ continue; ++ case snd_soc_dapm_mux: ++ case snd_soc_dapm_output: ++ case snd_soc_dapm_input: ++ case snd_soc_dapm_switch: ++ case snd_soc_dapm_hp: ++ case snd_soc_dapm_mic: ++ case snd_soc_dapm_spk: ++ case snd_soc_dapm_line: ++ case snd_soc_dapm_micbias: ++ case snd_soc_dapm_dac: ++ case snd_soc_dapm_adc: ++ case snd_soc_dapm_pga: ++ case snd_soc_dapm_mixer: ++ if (w->name) { ++ in = is_connected_input_ep(w); ++ dapm_clear_walk(w->codec); ++ out = is_connected_output_ep(w); ++ dapm_clear_walk(w->codec); ++ printk("%s: %s in %d out %d\n", w->name, ++ w->power ? "On":"Off",in, out); ++ ++ list_for_each_entry(p, &w->sources, list_sink) { ++ if (p->connect) ++ printk(" in %s %s\n", p->name ? p->name : "static", ++ p->source->name); ++ } ++ list_for_each_entry(p, &w->sinks, list_source) { ++ p = list_entry(lp, struct snd_soc_dapm_path, list_source); ++ if (p->connect) ++ printk(" out %s %s\n", p->name ? p->name : "static", ++ p->sink->name); ++ } ++ } ++ break; ++ } ++ } ++} ++#endif ++ ++/* test and update the power status of a mux widget */ ++int dapm_mux_update_power(struct snd_soc_dapm_widget *widget, ++ struct snd_kcontrol *kcontrol, int mask, int val, struct soc_enum* e) ++{ ++ struct snd_soc_dapm_path *path; ++ int found = 0; ++ ++ if (widget->id != snd_soc_dapm_mux) ++ return -ENODEV; ++ ++ if (!snd_soc_test_bits(widget->codec, e->reg, mask, val)) ++ return 0; ++ ++ /* find dapm widget path assoc with kcontrol */ ++ list_for_each_entry(path, &widget->codec->dapm_paths, list) { ++ if (path->kcontrol != kcontrol) ++ continue; ++ ++ if (!path->name || ! e->texts[val]) ++ continue; ++ ++ found = 1; ++ /* we now need to match the string in the enum to the path */ ++ if (!(strcmp(path->name, e->texts[val]))) ++ path->connect = 1; /* new connection */ ++ else ++ path->connect = 0; /* old connection must be powered down */ ++ } ++ ++ if (found) ++ dapm_power_widgets(widget->codec, SND_SOC_DAPM_STREAM_NOP); ++ ++ return 0; ++} ++EXPORT_SYMBOL_GPL(dapm_mux_update_power); ++ ++/* test and update the power status of a mixer widget */ ++int dapm_mixer_update_power(struct snd_soc_dapm_widget *widget, ++ struct snd_kcontrol *kcontrol, int reg, int val_mask, int val, int invert) ++{ ++ struct snd_soc_dapm_path *path; ++ int found = 0; ++ ++ if (widget->id != snd_soc_dapm_mixer) ++ return -ENODEV; ++ ++ if (!snd_soc_test_bits(widget->codec, reg, val_mask, val)) ++ return 0; ++ ++ /* find dapm widget path assoc with kcontrol */ ++ list_for_each_entry(path, &widget->codec->dapm_paths, list) { ++ if (path->kcontrol != kcontrol) ++ continue; ++ ++ /* found, now check type */ ++ found = 1; ++ if (val) ++ /* new connection */ ++ path->connect = invert ? 0:1; ++ else ++ /* old connection must be powered down */ ++ path->connect = invert ? 1:0; ++ break; ++ } ++ ++ if (found) ++ dapm_power_widgets(widget->codec, SND_SOC_DAPM_STREAM_NOP); ++ ++ return 0; ++} ++EXPORT_SYMBOL_GPL(dapm_mixer_update_power); ++ ++/* show dapm widget status in sys fs */ ++static ssize_t dapm_widget_show(struct device *dev, ++ struct device_attribute *attr, char *buf) ++{ ++ struct snd_soc_device *devdata = dev_get_drvdata(dev); ++ struct snd_soc_codec *codec = devdata->codec; ++ struct snd_soc_dapm_widget *w; ++ int count = 0; ++ char *state = "not set"; ++ ++ list_for_each_entry(w, &codec->dapm_widgets, list) { ++ ++ /* only display widgets that burnm power */ ++ switch (w->id) { ++ case snd_soc_dapm_hp: ++ case snd_soc_dapm_mic: ++ case snd_soc_dapm_spk: ++ case snd_soc_dapm_line: ++ case snd_soc_dapm_micbias: ++ case snd_soc_dapm_dac: ++ case snd_soc_dapm_adc: ++ case snd_soc_dapm_pga: ++ case snd_soc_dapm_mixer: ++ if (w->name) ++ count += sprintf(buf + count, "%s: %s\n", ++ w->name, w->power ? "On":"Off"); ++ break; ++ default: ++ break; ++ } ++ } ++ ++ switch(codec->dapm_state){ ++ case SNDRV_CTL_POWER_D0: ++ state = "D0"; ++ break; ++ case SNDRV_CTL_POWER_D1: ++ state = "D1"; ++ break; ++ case SNDRV_CTL_POWER_D2: ++ state = "D2"; ++ break; ++ case SNDRV_CTL_POWER_D3hot: ++ state = "D3hot"; ++ break; ++ case SNDRV_CTL_POWER_D3cold: ++ state = "D3cold"; ++ break; ++ } ++ count += sprintf(buf + count, "PM State: %s\n", state); ++ ++ return count; ++} ++ ++static DEVICE_ATTR(dapm_widget, 0444, dapm_widget_show, NULL); ++ ++int snd_soc_dapm_sys_add(struct device *dev) ++{ ++ int ret = 0; ++ ++ if (dapm_status) ++ ret = device_create_file(dev, &dev_attr_dapm_widget); ++ ++ return ret; ++} ++ ++static void snd_soc_dapm_sys_remove(struct device *dev) ++{ ++ if (dapm_status) ++ device_remove_file(dev, &dev_attr_dapm_widget); ++} ++ ++/* free all dapm widgets and resources */ ++void dapm_free_widgets(struct snd_soc_codec *codec) ++{ ++ struct snd_soc_dapm_widget *w, *next_w; ++ struct snd_soc_dapm_path *p, *next_p; ++ ++ list_for_each_entry_safe(w, next_w, &codec->dapm_widgets, list) { ++ list_del(&w->list); ++ kfree(w); ++ } ++ ++ list_for_each_entry_safe(p, next_p, &codec->dapm_paths, list) { ++ list_del(&p->list); ++ kfree(p->long_name); ++ kfree(p); ++ } ++} ++ ++/** ++ * snd_soc_dapm_sync_endpoints - scan and power dapm paths ++ * @codec: audio codec ++ * ++ * Walks all dapm audio paths and powers widgets according to their ++ * stream or path usage. ++ * ++ * Returns 0 for success. ++ */ ++int snd_soc_dapm_sync_endpoints(struct snd_soc_codec *codec) ++{ ++ return dapm_power_widgets(codec, SND_SOC_DAPM_STREAM_NOP); ++} ++EXPORT_SYMBOL_GPL(snd_soc_dapm_sync_endpoints); ++ ++/** ++ * snd_soc_dapm_connect_input - connect dapm widgets ++ * @codec: audio codec ++ * @sink: name of target widget ++ * @control: mixer control name ++ * @source: name of source name ++ * ++ * Connects 2 dapm widgets together via a named audio path. The sink is ++ * the widget receiving the audio signal, whilst the source is the sender ++ * of the audio signal. ++ * ++ * Returns 0 for success else error. ++ */ ++int snd_soc_dapm_connect_input(struct snd_soc_codec *codec, const char *sink, ++ const char * control, const char *source) ++{ ++ struct snd_soc_dapm_path *path; ++ struct snd_soc_dapm_widget *wsource = NULL, *wsink = NULL, *w; ++ int ret = 0; ++ ++ /* find src and dest widgets */ ++ list_for_each_entry(w, &codec->dapm_widgets, list) { ++ ++ if (!wsink && !(strcmp(w->name, sink))) { ++ wsink = w; ++ continue; ++ } ++ if (!wsource && !(strcmp(w->name, source))) { ++ wsource = w; ++ } ++ } ++ ++ if (wsource == NULL || wsink == NULL) ++ return -ENODEV; ++ ++ path = kzalloc(sizeof(struct snd_soc_dapm_path), GFP_KERNEL); ++ if (!path) ++ return -ENOMEM; ++ ++ path->source = wsource; ++ path->sink = wsink; ++ INIT_LIST_HEAD(&path->list); ++ INIT_LIST_HEAD(&path->list_source); ++ INIT_LIST_HEAD(&path->list_sink); ++ ++ /* check for external widgets */ ++ if (wsink->id == snd_soc_dapm_input) { ++ if (wsource->id == snd_soc_dapm_micbias || ++ wsource->id == snd_soc_dapm_mic || ++ wsink->id == snd_soc_dapm_line) ++ wsink->ext = 1; ++ } ++ if (wsource->id == snd_soc_dapm_output) { ++ if (wsink->id == snd_soc_dapm_spk || ++ wsink->id == snd_soc_dapm_hp || ++ wsink->id == snd_soc_dapm_line) ++ wsource->ext = 1; ++ } ++ ++ /* connect static paths */ ++ if (control == NULL) { ++ list_add(&path->list, &codec->dapm_paths); ++ list_add(&path->list_sink, &wsink->sources); ++ list_add(&path->list_source, &wsource->sinks); ++ path->connect = 1; ++ return 0; ++ } ++ ++ /* connect dynamic paths */ ++ switch(wsink->id) { ++ case snd_soc_dapm_adc: ++ case snd_soc_dapm_dac: ++ case snd_soc_dapm_pga: ++ case snd_soc_dapm_input: ++ case snd_soc_dapm_output: ++ case snd_soc_dapm_micbias: ++ case snd_soc_dapm_vmid: ++ case snd_soc_dapm_pre: ++ case snd_soc_dapm_post: ++ list_add(&path->list, &codec->dapm_paths); ++ list_add(&path->list_sink, &wsink->sources); ++ list_add(&path->list_source, &wsource->sinks); ++ path->connect = 1; ++ return 0; ++ case snd_soc_dapm_mux: ++ ret = dapm_connect_mux(codec, wsource, wsink, path, control, ++ &wsink->kcontrols[0]); ++ if (ret != 0) ++ goto err; ++ break; ++ case snd_soc_dapm_switch: ++ case snd_soc_dapm_mixer: ++ ret = dapm_connect_mixer(codec, wsource, wsink, path, control); ++ if (ret != 0) ++ goto err; ++ break; ++ case snd_soc_dapm_hp: ++ case snd_soc_dapm_mic: ++ case snd_soc_dapm_line: ++ case snd_soc_dapm_spk: ++ list_add(&path->list, &codec->dapm_paths); ++ list_add(&path->list_sink, &wsink->sources); ++ list_add(&path->list_source, &wsource->sinks); ++ path->connect = 0; ++ return 0; ++ } ++ return 0; ++ ++err: ++ printk(KERN_WARNING "asoc: no dapm match for %s --> %s --> %s\n", source, ++ control, sink); ++ kfree(path); ++ return ret; ++} ++EXPORT_SYMBOL_GPL(snd_soc_dapm_connect_input); ++ ++/** ++ * snd_soc_dapm_new_widgets - add new dapm widgets ++ * @codec: audio codec ++ * ++ * Checks the codec for any new dapm widgets and creates them if found. ++ * ++ * Returns 0 for success. ++ */ ++int snd_soc_dapm_new_widgets(struct snd_soc_codec *codec) ++{ ++ struct snd_soc_dapm_widget *w; ++ ++ mutex_lock(&codec->mutex); ++ list_for_each_entry(w, &codec->dapm_widgets, list) ++ { ++ if (w->new) ++ continue; ++ ++ switch(w->id) { ++ case snd_soc_dapm_switch: ++ case snd_soc_dapm_mixer: ++ dapm_new_mixer(codec, w); ++ break; ++ case snd_soc_dapm_mux: ++ dapm_new_mux(codec, w); ++ break; ++ case snd_soc_dapm_adc: ++ case snd_soc_dapm_dac: ++ case snd_soc_dapm_pga: ++ dapm_new_pga(codec, w); ++ break; ++ case snd_soc_dapm_input: ++ case snd_soc_dapm_output: ++ case snd_soc_dapm_micbias: ++ case snd_soc_dapm_spk: ++ case snd_soc_dapm_hp: ++ case snd_soc_dapm_mic: ++ case snd_soc_dapm_line: ++ case snd_soc_dapm_vmid: ++ case snd_soc_dapm_pre: ++ case snd_soc_dapm_post: ++ break; ++ } ++ w->new = 1; ++ } ++ ++ dapm_power_widgets(codec, SND_SOC_DAPM_STREAM_NOP); ++ mutex_unlock(&codec->mutex); ++ return 0; ++} ++EXPORT_SYMBOL_GPL(snd_soc_dapm_new_widgets); ++ ++/** ++ * snd_soc_dapm_get_volsw - dapm mixer get callback ++ * @kcontrol: mixer control ++ * @uinfo: control element information ++ * ++ * Callback to get the value of a dapm mixer control. ++ * ++ * Returns 0 for success. ++ */ ++int snd_soc_dapm_get_volsw(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_value *ucontrol) ++{ ++ struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); ++ int reg = kcontrol->private_value & 0xff; ++ int shift = (kcontrol->private_value >> 8) & 0x0f; ++ int rshift = (kcontrol->private_value >> 12) & 0x0f; ++ int mask = (kcontrol->private_value >> 16) & 0xff; ++ int invert = (kcontrol->private_value >> 24) & 0x01; ++ ++ /* return the saved value if we are powered down */ ++ if (widget->id == snd_soc_dapm_pga && !widget->power) { ++ ucontrol->value.integer.value[0] = widget->saved_value; ++ return 0; ++ } ++ ++ ucontrol->value.integer.value[0] = ++ (snd_soc_read(widget->codec, reg) >> shift) & mask; ++ if (shift != rshift) ++ ucontrol->value.integer.value[1] = ++ (snd_soc_read(widget->codec, reg) >> rshift) & mask; ++ if (invert) { ++ ucontrol->value.integer.value[0] = ++ mask - ucontrol->value.integer.value[0]; ++ if (shift != rshift) ++ ucontrol->value.integer.value[1] = ++ mask - ucontrol->value.integer.value[1]; ++ } ++ ++ return 0; ++} ++EXPORT_SYMBOL_GPL(snd_soc_dapm_get_volsw); ++ ++/** ++ * snd_soc_dapm_put_volsw - dapm mixer set callback ++ * @kcontrol: mixer control ++ * @uinfo: control element information ++ * ++ * Callback to set the value of a dapm mixer control. ++ * ++ * Returns 0 for success. ++ */ ++int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_value *ucontrol) ++{ ++ struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); ++ int reg = kcontrol->private_value & 0xff; ++ int shift = (kcontrol->private_value >> 8) & 0x0f; ++ int rshift = (kcontrol->private_value >> 12) & 0x0f; ++ int mask = (kcontrol->private_value >> 16) & 0xff; ++ int invert = (kcontrol->private_value >> 24) & 0x01; ++ unsigned short val, val2, val_mask; ++ int ret; ++ ++ val = (ucontrol->value.integer.value[0] & mask); ++ ++ if (invert) ++ val = mask - val; ++ val_mask = mask << shift; ++ val = val << shift; ++ if (shift != rshift) { ++ val2 = (ucontrol->value.integer.value[1] & mask); ++ if (invert) ++ val2 = mask - val2; ++ val_mask |= mask << rshift; ++ val |= val2 << rshift; ++ } ++ ++ mutex_lock(&widget->codec->mutex); ++ widget->value = val; ++ ++ /* save volume value if the widget is powered down */ ++ if (widget->id == snd_soc_dapm_pga && !widget->power) { ++ widget->saved_value = val; ++ mutex_unlock(&widget->codec->mutex); ++ return 1; ++ } ++ ++ dapm_mixer_update_power(widget, kcontrol, reg, val_mask, val, invert); ++ if (widget->event) { ++ if (widget->event_flags & SND_SOC_DAPM_PRE_REG) { ++ ret = widget->event(widget, SND_SOC_DAPM_PRE_REG); ++ if (ret < 0) ++ goto out; ++ } ++ ret = snd_soc_update_bits(widget->codec, reg, val_mask, val); ++ if (widget->event_flags & SND_SOC_DAPM_POST_REG) ++ ret = widget->event(widget, SND_SOC_DAPM_POST_REG); ++ } else ++ ret = snd_soc_update_bits(widget->codec, reg, val_mask, val); ++ ++out: ++ mutex_unlock(&widget->codec->mutex); ++ return ret; ++} ++EXPORT_SYMBOL_GPL(snd_soc_dapm_put_volsw); ++ ++/** ++ * snd_soc_dapm_get_enum_double - dapm enumerated double mixer get callback ++ * @kcontrol: mixer control ++ * @uinfo: control element information ++ * ++ * Callback to get the value of a dapm enumerated double mixer control. ++ * ++ * Returns 0 for success. ++ */ ++int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_value *ucontrol) ++{ ++ struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); ++ struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; ++ unsigned short val, bitmask; ++ ++ for (bitmask = 1; bitmask < e->mask; bitmask <<= 1) ++ ; ++ val = snd_soc_read(widget->codec, e->reg); ++ ucontrol->value.enumerated.item[0] = (val >> e->shift_l) & (bitmask - 1); ++ if (e->shift_l != e->shift_r) ++ ucontrol->value.enumerated.item[1] = ++ (val >> e->shift_r) & (bitmask - 1); ++ ++ return 0; ++} ++EXPORT_SYMBOL_GPL(snd_soc_dapm_get_enum_double); ++ ++/** ++ * snd_soc_dapm_put_enum_double - dapm enumerated double mixer set callback ++ * @kcontrol: mixer control ++ * @uinfo: control element information ++ * ++ * Callback to set the value of a dapm enumerated double mixer control. ++ * ++ * Returns 0 for success. ++ */ ++int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_value *ucontrol) ++{ ++ struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); ++ struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; ++ unsigned short val, mux; ++ unsigned short mask, bitmask; ++ int ret = 0; ++ ++ for (bitmask = 1; bitmask < e->mask; bitmask <<= 1) ++ ; ++ if (ucontrol->value.enumerated.item[0] > e->mask - 1) ++ return -EINVAL; ++ mux = ucontrol->value.enumerated.item[0]; ++ val = mux << e->shift_l; ++ mask = (bitmask - 1) << e->shift_l; ++ if (e->shift_l != e->shift_r) { ++ if (ucontrol->value.enumerated.item[1] > e->mask - 1) ++ return -EINVAL; ++ val |= ucontrol->value.enumerated.item[1] << e->shift_r; ++ mask |= (bitmask - 1) << e->shift_r; ++ } ++ ++ mutex_lock(&widget->codec->mutex); ++ widget->value = val; ++ dapm_mux_update_power(widget, kcontrol, mask, mux, e); ++ if (widget->event) { ++ if (widget->event_flags & SND_SOC_DAPM_PRE_REG) { ++ ret = widget->event(widget, SND_SOC_DAPM_PRE_REG); ++ if (ret < 0) ++ goto out; ++ } ++ ret = snd_soc_update_bits(widget->codec, e->reg, mask, val); ++ if (widget->event_flags & SND_SOC_DAPM_POST_REG) ++ ret = widget->event(widget, SND_SOC_DAPM_POST_REG); ++ } else ++ ret = snd_soc_update_bits(widget->codec, e->reg, mask, val); ++ ++out: ++ mutex_unlock(&widget->codec->mutex); ++ return ret; ++} ++EXPORT_SYMBOL_GPL(snd_soc_dapm_put_enum_double); ++ ++/** ++ * snd_soc_dapm_new_control - create new dapm control ++ * @codec: audio codec ++ * @widget: widget template ++ * ++ * Creates a new dapm control based upon the template. ++ * ++ * Returns 0 for success else error. ++ */ ++int snd_soc_dapm_new_control(struct snd_soc_codec *codec, ++ const struct snd_soc_dapm_widget *widget) ++{ ++ struct snd_soc_dapm_widget *w; ++ ++ if ((w = dapm_cnew_widget(widget)) == NULL) ++ return -ENOMEM; ++ ++ w->codec = codec; ++ INIT_LIST_HEAD(&w->sources); ++ INIT_LIST_HEAD(&w->sinks); ++ INIT_LIST_HEAD(&w->list); ++ list_add(&w->list, &codec->dapm_widgets); ++ ++ /* machine layer set ups unconnected pins and insertions */ ++ w->connected = 1; ++ return 0; ++} ++EXPORT_SYMBOL_GPL(snd_soc_dapm_new_control); ++ ++/** ++ * snd_soc_dapm_stream_event - send a stream event to the dapm core ++ * @codec: audio codec ++ * @stream: stream name ++ * @event: stream event ++ * ++ * Sends a stream event to the dapm core. The core then makes any ++ * necessary widget power changes. ++ * ++ * Returns 0 for success else error. ++ */ ++int snd_soc_dapm_stream_event(struct snd_soc_codec *codec, ++ char *stream, int event) ++{ ++ struct snd_soc_dapm_widget *w; ++ ++ mutex_lock(&codec->mutex); ++ list_for_each_entry(w, &codec->dapm_widgets, list) ++ { ++ if (!w->sname) ++ continue; ++ dbg("widget %s\n %s stream %s event %d\n", w->name, w->sname, ++ stream, event); ++ if (strstr(w->sname, stream)) { ++ switch(event) { ++ case SND_SOC_DAPM_STREAM_START: ++ w->active = 1; ++ break; ++ case SND_SOC_DAPM_STREAM_STOP: ++ w->active = 0; ++ break; ++ case SND_SOC_DAPM_STREAM_SUSPEND: ++ if (w->active) ++ w->suspend = 1; ++ w->active = 0; ++ break; ++ case SND_SOC_DAPM_STREAM_RESUME: ++ if (w->suspend) { ++ w->active = 1; ++ w->suspend = 0; ++ } ++ break; ++ case SND_SOC_DAPM_STREAM_PAUSE_PUSH: ++ break; ++ case SND_SOC_DAPM_STREAM_PAUSE_RELEASE: ++ break; ++ } ++ } ++ } ++ mutex_unlock(&codec->mutex); ++ ++ dapm_power_widgets(codec, event); ++ dump_dapm(codec, __FUNCTION__); ++ return 0; ++} ++EXPORT_SYMBOL_GPL(snd_soc_dapm_stream_event); ++ ++/** ++ * snd_soc_dapm_set_endpoint - set audio endpoint status ++ * @codec: audio codec ++ * @endpoint: audio signal endpoint (or start point) ++ * @status: point status ++ * ++ * Set audio endpoint status - connected or disconnected. ++ * ++ * Returns 0 for success else error. ++ */ ++int snd_soc_dapm_set_endpoint(struct snd_soc_codec *codec, ++ char *endpoint, int status) ++{ ++ struct snd_soc_dapm_widget *w; ++ ++ list_for_each_entry(w, &codec->dapm_widgets, list) { ++ if (!strcmp(w->name, endpoint)) { ++ w->connected = status; ++ } ++ } ++ ++ return 0; ++} ++EXPORT_SYMBOL_GPL(snd_soc_dapm_set_endpoint); ++ ++/** ++ * snd_soc_dapm_free - free dapm resources ++ * @socdev: SoC device ++ * ++ * Free all dapm widgets and resources. ++ */ ++void snd_soc_dapm_free(struct snd_soc_device *socdev) ++{ ++ struct snd_soc_codec *codec = socdev->codec; ++ ++ snd_soc_dapm_sys_remove(socdev->dev); ++ dapm_free_widgets(codec); ++} ++EXPORT_SYMBOL_GPL(snd_soc_dapm_free); ++ ++/* Module information */ ++MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com"); ++MODULE_DESCRIPTION("Dynamic Audio Power Management core for ALSA SoC"); ++MODULE_LICENSE("GPL"); +Index: linux-2.6-pxa-new/sound/soc/soc-core.c +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/sound/soc/soc-core.c +@@ -0,0 +1,2063 @@ ++/* ++ * soc-core.c -- ALSA SoC Audio Layer ++ * ++ * Copyright 2005 Wolfson Microelectronics PLC. ++ * Author: Liam Girdwood ++ * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com ++ * ++ * This program is free software; you can redistribute it and/or modify it ++ * under the terms of the GNU General Public License as published by the ++ * Free Software Foundation; either version 2 of the License, or (at your ++ * option) any later version. ++ * ++ * Revision history ++ * 12th Aug 2005 Initial version. ++ * 25th Oct 2005 Working Codec, Interface and Platform registration. ++ * ++ * TODO: ++ * o Add hw rules to enforce rates, etc. ++ * o More testing with other codecs/machines. ++ * o Add more codecs and platforms to ensure good API coverage. ++ * o Support TDM on PCM and I2S ++ */ ++ ++#include <linux/module.h> ++#include <linux/moduleparam.h> ++#include <linux/init.h> ++#include <linux/delay.h> ++#include <linux/pm.h> ++#include <linux/bitops.h> ++#include <linux/platform_device.h> ++#include <sound/driver.h> ++#include <sound/core.h> ++#include <sound/pcm.h> ++#include <sound/pcm_params.h> ++#include <sound/soc.h> ++#include <sound/soc-dapm.h> ++#include <sound/initval.h> ++ ++/* debug */ ++#define SOC_DEBUG 1 ++#if SOC_DEBUG ++#define dbg(format, arg...) printk(format, ## arg) ++#else ++#define dbg(format, arg...) ++#endif ++/* debug DAI capabilities matching */ ++#define SOC_DEBUG_DAI 1 ++#if SOC_DEBUG_DAI ++#define dbgc(format, arg...) printk(format, ## arg) ++#else ++#define dbgc(format, arg...) ++#endif ++ ++#define CODEC_CPU(codec, cpu) ((codec << 4) | cpu) ++ ++static DEFINE_MUTEX(pcm_mutex); ++static DEFINE_MUTEX(io_mutex); ++static struct workqueue_struct *soc_workq; ++static struct work_struct soc_stream_work; ++static DECLARE_WAIT_QUEUE_HEAD(soc_pm_waitq); ++ ++/* supported sample rates */ ++/* ATTENTION: these values depend on the definition in pcm.h! */ ++static const unsigned int rates[] = { ++ 5512, 8000, 11025, 16000, 22050, 32000, 44100, ++ 48000, 64000, 88200, 96000, 176400, 192000 ++}; ++ ++/* ++ * This is a timeout to do a DAPM powerdown after a stream is closed(). ++ * It can be used to eliminate pops between different playback streams, e.g. ++ * between two audio tracks. ++ */ ++static int pmdown_time = 5000; ++module_param(pmdown_time, int, 0); ++MODULE_PARM_DESC(pmdown_time, "DAPM stream powerdown time (msecs)"); ++ ++#ifdef CONFIG_SND_SOC_AC97_BUS ++/* unregister ac97 codec */ ++static int soc_ac97_dev_unregister(struct snd_soc_codec *codec) ++{ ++ if (codec->ac97->dev.bus) ++ device_unregister(&codec->ac97->dev); ++ return 0; ++} ++ ++/* stop no dev release warning */ ++static void soc_ac97_device_release(struct device *dev){} ++ ++/* register ac97 codec to bus */ ++static int soc_ac97_dev_register(struct snd_soc_codec *codec) ++{ ++ int err; ++ ++ codec->ac97->dev.bus = &ac97_bus_type; ++ codec->ac97->dev.parent = NULL; ++ codec->ac97->dev.release = soc_ac97_device_release; ++ ++ snprintf(codec->ac97->dev.bus_id, BUS_ID_SIZE, "%d-%d:%s", ++ codec->card->number, 0, codec->name); ++ err = device_register(&codec->ac97->dev); ++ if (err < 0) { ++ snd_printk(KERN_ERR "Can't register ac97 bus\n"); ++ codec->ac97->dev.bus = NULL; ++ return err; ++ } ++ return 0; ++} ++#endif ++ ++static inline const char* get_dai_name(int type) ++{ ++ switch(type) { ++ case SND_SOC_DAI_AC97: ++ return "AC97"; ++ case SND_SOC_DAI_I2S: ++ return "I2S"; ++ case SND_SOC_DAI_PCM: ++ return "PCM"; ++ } ++ return NULL; ++} ++ ++/* get rate format from rate */ ++static inline int soc_get_rate_format(int rate) ++{ ++ int i; ++ ++ for (i = 0; i < ARRAY_SIZE(rates); i++) { ++ if (rates[i] == rate) ++ return 1 << i; ++ } ++ return 0; ++} ++ ++/* gets the audio system mclk/sysclk for the given parameters */ ++static unsigned inline int soc_get_mclk(struct snd_soc_pcm_runtime *rtd, ++ struct snd_soc_clock_info *info) ++{ ++ struct snd_soc_device *socdev = rtd->socdev; ++ struct snd_soc_machine *machine = socdev->machine; ++ int i; ++ ++ /* find the matching machine config and get it's mclk for the given ++ * sample rate and hardware format */ ++ for(i = 0; i < machine->num_links; i++) { ++ if (machine->dai_link[i].cpu_dai == rtd->cpu_dai && ++ machine->dai_link[i].config_sysclk) ++ return machine->dai_link[i].config_sysclk(rtd, info); ++ } ++ return 0; ++} ++ ++/* changes a bitclk multiplier mask to a divider mask */ ++static u64 soc_bfs_rcw_to_div(u64 bfs, int rate, unsigned int mclk, ++ unsigned int pcmfmt, unsigned int chn) ++{ ++ int i, j; ++ u64 bfs_ = 0; ++ int size = snd_pcm_format_physical_width(pcmfmt), min = 0; ++ ++ if (size <= 0) ++ return 0; ++ ++ /* the minimum bit clock that has enough bandwidth */ ++ min = size * rate * chn; ++ dbgc("rcw --> div min bclk %d with mclk %d\n", min, mclk); ++ ++ for (i = 0; i < 64; i++) { ++ if ((bfs >> i) & 0x1) { ++ j = min * (i + 1); ++ bfs_ |= SND_SOC_FSBD(mclk/j); ++ dbgc("rcw --> div support mult %d\n", ++ SND_SOC_FSBD_REAL(1<<i)); ++ } ++ } ++ ++ return bfs_; ++} ++ ++/* changes a bitclk divider mask to a multiplier mask */ ++static u64 soc_bfs_div_to_rcw(u64 bfs, int rate, unsigned int mclk, ++ unsigned int pcmfmt, unsigned int chn) ++{ ++ int i, j; ++ u64 bfs_ = 0; ++ ++ int size = snd_pcm_format_physical_width(pcmfmt), min = 0; ++ ++ if (size <= 0) ++ return 0; ++ ++ /* the minimum bit clock that has enough bandwidth */ ++ min = size * rate * chn; ++ dbgc("div to rcw min bclk %d with mclk %d\n", min, mclk); ++ ++ for (i = 0; i < 64; i++) { ++ if ((bfs >> i) & 0x1) { ++ j = mclk / (i + 1); ++ if (j >= min) { ++ bfs_ |= SND_SOC_FSBW(j/min); ++ dbgc("div --> rcw support div %d\n", ++ SND_SOC_FSBW_REAL(1<<i)); ++ } ++ } ++ } ++ ++ return bfs_; ++} ++ ++/* changes a constant bitclk to a multiplier mask */ ++static u64 soc_bfs_rate_to_rcw(u64 bfs, int rate, unsigned int mclk, ++ unsigned int pcmfmt, unsigned int chn) ++{ ++ unsigned int bfs_ = rate * bfs; ++ int size = snd_pcm_format_physical_width(pcmfmt), min = 0; ++ ++ if (size <= 0) ++ return 0; ++ ++ /* the minimum bit clock that has enough bandwidth */ ++ min = size * rate * chn; ++ dbgc("rate --> rcw min bclk %d with mclk %d\n", min, mclk); ++ ++ if (bfs_ < min) ++ return 0; ++ else { ++ bfs_ = SND_SOC_FSBW(bfs_/min); ++ dbgc("rate --> rcw support div %d\n", SND_SOC_FSBW_REAL(bfs_)); ++ return bfs_; ++ } ++} ++ ++/* changes a bitclk multiplier mask to a divider mask */ ++static u64 soc_bfs_rate_to_div(u64 bfs, int rate, unsigned int mclk, ++ unsigned int pcmfmt, unsigned int chn) ++{ ++ unsigned int bfs_ = rate * bfs; ++ int size = snd_pcm_format_physical_width(pcmfmt), min = 0; ++ ++ if (size <= 0) ++ return 0; ++ ++ /* the minimum bit clock that has enough bandwidth */ ++ min = size * rate * chn; ++ dbgc("rate --> div min bclk %d with mclk %d\n", min, mclk); ++ ++ if (bfs_ < min) ++ return 0; ++ else { ++ bfs_ = SND_SOC_FSBW(mclk/bfs_); ++ dbgc("rate --> div support div %d\n", SND_SOC_FSBD_REAL(bfs_)); ++ return bfs_; ++ } ++} ++ ++/* Matches codec DAI and SoC CPU DAI hardware parameters */ ++static int soc_hw_match_params(struct snd_pcm_substream *substream, ++ struct snd_pcm_hw_params *params) ++{ ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ struct snd_soc_dai_mode *codec_dai_mode = NULL; ++ struct snd_soc_dai_mode *cpu_dai_mode = NULL; ++ struct snd_soc_clock_info clk_info; ++ unsigned int fs, mclk, rate = params_rate(params), ++ chn, j, k, cpu_bclk, codec_bclk, pcmrate; ++ u16 fmt = 0; ++ u64 codec_bfs, cpu_bfs; ++ ++ dbg("asoc: match version %s\n", SND_SOC_VERSION); ++ clk_info.rate = rate; ++ pcmrate = soc_get_rate_format(rate); ++ ++ /* try and find a match from the codec and cpu DAI capabilities */ ++ for (j = 0; j < rtd->codec_dai->caps.num_modes; j++) { ++ for (k = 0; k < rtd->cpu_dai->caps.num_modes; k++) { ++ codec_dai_mode = &rtd->codec_dai->caps.mode[j]; ++ cpu_dai_mode = &rtd->cpu_dai->caps.mode[k]; ++ ++ if (!(codec_dai_mode->pcmrate & cpu_dai_mode->pcmrate & ++ pcmrate)) { ++ dbgc("asoc: DAI[%d:%d] failed to match rate\n", j, k); ++ continue; ++ } ++ ++ fmt = codec_dai_mode->fmt & cpu_dai_mode->fmt; ++ if (!(fmt & SND_SOC_DAIFMT_FORMAT_MASK)) { ++ dbgc("asoc: DAI[%d:%d] failed to match format\n", j, k); ++ continue; ++ } ++ ++ if (!(fmt & SND_SOC_DAIFMT_CLOCK_MASK)) { ++ dbgc("asoc: DAI[%d:%d] failed to match clock masters\n", ++ j, k); ++ continue; ++ } ++ ++ if (!(fmt & SND_SOC_DAIFMT_INV_MASK)) { ++ dbgc("asoc: DAI[%d:%d] failed to match invert\n", j, k); ++ continue; ++ } ++ ++ if (!(codec_dai_mode->pcmfmt & cpu_dai_mode->pcmfmt)) { ++ dbgc("asoc: DAI[%d:%d] failed to match pcm format\n", j, k); ++ continue; ++ } ++ ++ if (!(codec_dai_mode->pcmdir & cpu_dai_mode->pcmdir)) { ++ dbgc("asoc: DAI[%d:%d] failed to match direction\n", j, k); ++ continue; ++ } ++ ++ /* todo - still need to add tdm selection */ ++ rtd->cpu_dai->dai_runtime.fmt = ++ rtd->codec_dai->dai_runtime.fmt = ++ 1 << (ffs(fmt & SND_SOC_DAIFMT_FORMAT_MASK) -1) | ++ 1 << (ffs(fmt & SND_SOC_DAIFMT_CLOCK_MASK) - 1) | ++ 1 << (ffs(fmt & SND_SOC_DAIFMT_INV_MASK) - 1); ++ clk_info.bclk_master = ++ rtd->cpu_dai->dai_runtime.fmt & SND_SOC_DAIFMT_CLOCK_MASK; ++ ++ /* make sure the ratio between rate and master ++ * clock is acceptable*/ ++ fs = (cpu_dai_mode->fs & codec_dai_mode->fs); ++ if (fs == 0) { ++ dbgc("asoc: DAI[%d:%d] failed to match FS\n", j, k); ++ continue; ++ } ++ clk_info.fs = rtd->cpu_dai->dai_runtime.fs = ++ rtd->codec_dai->dai_runtime.fs = fs; ++ ++ /* calculate audio system clocking using slowest clocks possible*/ ++ mclk = soc_get_mclk(rtd, &clk_info); ++ if (mclk == 0) { ++ dbgc("asoc: DAI[%d:%d] configuration not clockable\n", j, k); ++ dbgc("asoc: rate %d fs %d master %x\n", rate, fs, ++ clk_info.bclk_master); ++ continue; ++ } ++ ++ /* calculate word size (per channel) and frame size */ ++ rtd->codec_dai->dai_runtime.pcmfmt = ++ rtd->cpu_dai->dai_runtime.pcmfmt = ++ 1 << params_format(params); ++ ++ chn = params_channels(params); ++ /* i2s always has left and right */ ++ if (params_channels(params) == 1 && ++ rtd->cpu_dai->dai_runtime.fmt & (SND_SOC_DAIFMT_I2S | ++ SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_LEFT_J)) ++ chn <<= 1; ++ ++ /* Calculate bfs - the ratio between bitclock and the sample rate ++ * We must take into consideration the dividers and multipliers ++ * used in the codec and cpu DAI modes. We always choose the ++ * lowest possible clocks to reduce power. ++ */ ++ switch (CODEC_CPU(codec_dai_mode->flags, cpu_dai_mode->flags)) { ++ case CODEC_CPU(SND_SOC_DAI_BFS_DIV, SND_SOC_DAI_BFS_DIV): ++ /* cpu & codec bfs dividers */ ++ rtd->cpu_dai->dai_runtime.bfs = ++ rtd->codec_dai->dai_runtime.bfs = ++ 1 << (fls(codec_dai_mode->bfs & cpu_dai_mode->bfs) - 1); ++ break; ++ case CODEC_CPU(SND_SOC_DAI_BFS_DIV, SND_SOC_DAI_BFS_RCW): ++ /* normalise bfs codec divider & cpu rcw mult */ ++ codec_bfs = soc_bfs_div_to_rcw(codec_dai_mode->bfs, rate, ++ mclk, rtd->codec_dai->dai_runtime.pcmfmt, chn); ++ rtd->cpu_dai->dai_runtime.bfs = ++ 1 << (ffs(codec_bfs & cpu_dai_mode->bfs) - 1); ++ cpu_bfs = soc_bfs_rcw_to_div(cpu_dai_mode->bfs, rate, mclk, ++ rtd->codec_dai->dai_runtime.pcmfmt, chn); ++ rtd->codec_dai->dai_runtime.bfs = ++ 1 << (fls(codec_dai_mode->bfs & cpu_bfs) - 1); ++ break; ++ case CODEC_CPU(SND_SOC_DAI_BFS_RCW, SND_SOC_DAI_BFS_DIV): ++ /* normalise bfs codec rcw mult & cpu divider */ ++ codec_bfs = soc_bfs_rcw_to_div(codec_dai_mode->bfs, rate, ++ mclk, rtd->codec_dai->dai_runtime.pcmfmt, chn); ++ rtd->cpu_dai->dai_runtime.bfs = ++ 1 << (fls(codec_bfs & cpu_dai_mode->bfs) -1); ++ cpu_bfs = soc_bfs_div_to_rcw(cpu_dai_mode->bfs, rate, mclk, ++ rtd->codec_dai->dai_runtime.pcmfmt, chn); ++ rtd->codec_dai->dai_runtime.bfs = ++ 1 << (ffs(codec_dai_mode->bfs & cpu_bfs) -1); ++ break; ++ case CODEC_CPU(SND_SOC_DAI_BFS_RCW, SND_SOC_DAI_BFS_RCW): ++ /* codec & cpu bfs rate rcw multipliers */ ++ rtd->cpu_dai->dai_runtime.bfs = ++ rtd->codec_dai->dai_runtime.bfs = ++ 1 << (ffs(codec_dai_mode->bfs & cpu_dai_mode->bfs) -1); ++ break; ++ case CODEC_CPU(SND_SOC_DAI_BFS_DIV, SND_SOC_DAI_BFS_RATE): ++ /* normalise cpu bfs rate const multiplier & codec div */ ++ cpu_bfs = soc_bfs_rate_to_div(cpu_dai_mode->bfs, rate, ++ mclk, rtd->codec_dai->dai_runtime.pcmfmt, chn); ++ if(codec_dai_mode->bfs & cpu_bfs) { ++ rtd->codec_dai->dai_runtime.bfs = cpu_bfs; ++ rtd->cpu_dai->dai_runtime.bfs = cpu_dai_mode->bfs; ++ } else ++ rtd->cpu_dai->dai_runtime.bfs = 0; ++ break; ++ case CODEC_CPU(SND_SOC_DAI_BFS_RCW, SND_SOC_DAI_BFS_RATE): ++ /* normalise cpu bfs rate const multiplier & codec rcw mult */ ++ cpu_bfs = soc_bfs_rate_to_rcw(cpu_dai_mode->bfs, rate, ++ mclk, rtd->codec_dai->dai_runtime.pcmfmt, chn); ++ if(codec_dai_mode->bfs & cpu_bfs) { ++ rtd->codec_dai->dai_runtime.bfs = cpu_bfs; ++ rtd->cpu_dai->dai_runtime.bfs = cpu_dai_mode->bfs; ++ } else ++ rtd->cpu_dai->dai_runtime.bfs = 0; ++ break; ++ case CODEC_CPU(SND_SOC_DAI_BFS_RATE, SND_SOC_DAI_BFS_RCW): ++ /* normalise cpu bfs rate rcw multiplier & codec const mult */ ++ codec_bfs = soc_bfs_rate_to_rcw(codec_dai_mode->bfs, rate, ++ mclk, rtd->codec_dai->dai_runtime.pcmfmt, chn); ++ if(cpu_dai_mode->bfs & codec_bfs) { ++ rtd->cpu_dai->dai_runtime.bfs = codec_bfs; ++ rtd->codec_dai->dai_runtime.bfs = codec_dai_mode->bfs; ++ } else ++ rtd->cpu_dai->dai_runtime.bfs = 0; ++ break; ++ case CODEC_CPU(SND_SOC_DAI_BFS_RATE, SND_SOC_DAI_BFS_DIV): ++ /* normalise cpu bfs div & codec const mult */ ++ codec_bfs = soc_bfs_rate_to_div(codec_dai_mode->bfs, rate, ++ mclk, rtd->codec_dai->dai_runtime.pcmfmt, chn); ++ if(codec_dai_mode->bfs & codec_bfs) { ++ rtd->cpu_dai->dai_runtime.bfs = codec_bfs; ++ rtd->codec_dai->dai_runtime.bfs = codec_dai_mode->bfs; ++ } else ++ rtd->cpu_dai->dai_runtime.bfs = 0; ++ break; ++ case CODEC_CPU(SND_SOC_DAI_BFS_RATE, SND_SOC_DAI_BFS_RATE): ++ /* cpu & codec constant mult */ ++ if(codec_dai_mode->bfs == cpu_dai_mode->bfs) ++ rtd->cpu_dai->dai_runtime.bfs = ++ rtd->codec_dai->dai_runtime.bfs = ++ codec_dai_mode->bfs; ++ else ++ rtd->cpu_dai->dai_runtime.bfs = ++ rtd->codec_dai->dai_runtime.bfs = 0; ++ break; ++ default: ++ if(codec_dai_mode->flags == 0) ++ printk(KERN_ERR "asoc: error missing codec DAI flags\n"); ++ else ++ printk(KERN_ERR "asoc: error missing CPU DAI flags\n"); ++ break; ++ } ++ ++ /* make sure the bit clock speed is acceptable */ ++ if (!rtd->cpu_dai->dai_runtime.bfs || ++ !rtd->codec_dai->dai_runtime.bfs) { ++ dbgc("asoc: DAI[%d:%d] failed to match BFS\n", j, k); ++ dbgc("asoc: cpu_dai %llu codec %llu\n", ++ rtd->cpu_dai->dai_runtime.bfs, ++ rtd->codec_dai->dai_runtime.bfs); ++ dbgc("asoc: mclk %d hwfmt %x\n", mclk, fmt); ++ continue; ++ } ++ ++ goto found; ++ } ++ } ++ printk(KERN_ERR "asoc: no matching DAI found between codec and CPU\n"); ++ return -EINVAL; ++ ++found: ++ /* we have matching DAI's, so complete the runtime info */ ++ rtd->codec_dai->dai_runtime.pcmrate = ++ rtd->cpu_dai->dai_runtime.pcmrate = ++ soc_get_rate_format(rate); ++ ++ rtd->codec_dai->dai_runtime.priv = codec_dai_mode->priv; ++ rtd->cpu_dai->dai_runtime.priv = cpu_dai_mode->priv; ++ rtd->codec_dai->dai_runtime.flags = codec_dai_mode->flags; ++ rtd->cpu_dai->dai_runtime.flags = cpu_dai_mode->flags; ++ ++ /* for debug atm */ ++ dbg("asoc: DAI[%d:%d] Match OK\n", j, k); ++ if (rtd->codec_dai->dai_runtime.flags == SND_SOC_DAI_BFS_DIV) { ++ codec_bclk = (rtd->codec_dai->dai_runtime.fs * params_rate(params)) / ++ SND_SOC_FSBD_REAL(rtd->codec_dai->dai_runtime.bfs); ++ dbg("asoc: codec fs %d mclk %d bfs div %d bclk %d\n", ++ rtd->codec_dai->dai_runtime.fs, mclk, ++ SND_SOC_FSBD_REAL(rtd->codec_dai->dai_runtime.bfs), codec_bclk); ++ } else if(rtd->codec_dai->dai_runtime.flags == SND_SOC_DAI_BFS_RATE) { ++ codec_bclk = params_rate(params) * rtd->codec_dai->dai_runtime.bfs; ++ dbg("asoc: codec fs %d mclk %d bfs rate mult %llu bclk %d\n", ++ rtd->codec_dai->dai_runtime.fs, mclk, ++ rtd->codec_dai->dai_runtime.bfs, codec_bclk); ++ } else if (rtd->cpu_dai->dai_runtime.flags == SND_SOC_DAI_BFS_RCW) { ++ codec_bclk = params_rate(params) * params_channels(params) * ++ snd_pcm_format_physical_width(rtd->codec_dai->dai_runtime.pcmfmt) * ++ SND_SOC_FSBW_REAL(rtd->codec_dai->dai_runtime.bfs); ++ dbg("asoc: codec fs %d mclk %d bfs rcw mult %d bclk %d\n", ++ rtd->codec_dai->dai_runtime.fs, mclk, ++ SND_SOC_FSBW_REAL(rtd->codec_dai->dai_runtime.bfs), codec_bclk); ++ } else ++ codec_bclk = 0; ++ ++ if (rtd->cpu_dai->dai_runtime.flags == SND_SOC_DAI_BFS_DIV) { ++ cpu_bclk = (rtd->cpu_dai->dai_runtime.fs * params_rate(params)) / ++ SND_SOC_FSBD_REAL(rtd->cpu_dai->dai_runtime.bfs); ++ dbg("asoc: cpu fs %d mclk %d bfs div %d bclk %d\n", ++ rtd->cpu_dai->dai_runtime.fs, mclk, ++ SND_SOC_FSBD_REAL(rtd->cpu_dai->dai_runtime.bfs), cpu_bclk); ++ } else if (rtd->cpu_dai->dai_runtime.flags == SND_SOC_DAI_BFS_RATE) { ++ cpu_bclk = params_rate(params) * rtd->cpu_dai->dai_runtime.bfs; ++ dbg("asoc: cpu fs %d mclk %d bfs rate mult %llu bclk %d\n", ++ rtd->cpu_dai->dai_runtime.fs, mclk, ++ rtd->cpu_dai->dai_runtime.bfs, cpu_bclk); ++ } else if (rtd->cpu_dai->dai_runtime.flags == SND_SOC_DAI_BFS_RCW) { ++ cpu_bclk = params_rate(params) * params_channels(params) * ++ snd_pcm_format_physical_width(rtd->cpu_dai->dai_runtime.pcmfmt) * ++ SND_SOC_FSBW_REAL(rtd->cpu_dai->dai_runtime.bfs); ++ dbg("asoc: cpu fs %d mclk %d bfs mult rcw %d bclk %d\n", ++ rtd->cpu_dai->dai_runtime.fs, mclk, ++ SND_SOC_FSBW_REAL(rtd->cpu_dai->dai_runtime.bfs), cpu_bclk); ++ } else ++ cpu_bclk = 0; ++ ++ /* ++ * Check we have matching bitclocks. If we don't then it means the ++ * sysclock returned by either the codec or cpu DAI (selected by the ++ * machine sysclock function) is wrong compared with the supported DAI ++ * modes for the codec or cpu DAI. ++ */ ++ if (cpu_bclk != codec_bclk && cpu_bclk){ ++ printk(KERN_ERR ++ "asoc: codec and cpu bitclocks differ, audio may be wrong speed\n" ++ ); ++ printk(KERN_ERR "asoc: codec %d != cpu %d\n", codec_bclk, cpu_bclk); ++ } ++ ++ switch(rtd->cpu_dai->dai_runtime.fmt & SND_SOC_DAIFMT_CLOCK_MASK) { ++ case SND_SOC_DAIFMT_CBM_CFM: ++ dbg("asoc: DAI codec BCLK master, LRC master\n"); ++ break; ++ case SND_SOC_DAIFMT_CBS_CFM: ++ dbg("asoc: DAI codec BCLK slave, LRC master\n"); ++ break; ++ case SND_SOC_DAIFMT_CBM_CFS: ++ dbg("asoc: DAI codec BCLK master, LRC slave\n"); ++ break; ++ case SND_SOC_DAIFMT_CBS_CFS: ++ dbg("asoc: DAI codec BCLK slave, LRC slave\n"); ++ break; ++ } ++ dbg("asoc: mode %x, invert %x\n", ++ rtd->cpu_dai->dai_runtime.fmt & SND_SOC_DAIFMT_FORMAT_MASK, ++ rtd->cpu_dai->dai_runtime.fmt & SND_SOC_DAIFMT_INV_MASK); ++ dbg("asoc: audio rate %d chn %d fmt %x\n", params_rate(params), ++ params_channels(params), params_format(params)); ++ ++ return 0; ++} ++ ++static inline u32 get_rates(struct snd_soc_dai_mode *modes, int nmodes) ++{ ++ int i; ++ u32 rates = 0; ++ ++ for(i = 0; i < nmodes; i++) ++ rates |= modes[i].pcmrate; ++ ++ return rates; ++} ++ ++static inline u64 get_formats(struct snd_soc_dai_mode *modes, int nmodes) ++{ ++ int i; ++ u64 formats = 0; ++ ++ for(i = 0; i < nmodes; i++) ++ formats |= modes[i].pcmfmt; ++ ++ return formats; ++} ++ ++/* ++ * Called by ALSA when a PCM substream is opened, the runtime->hw record is ++ * then initialized and any private data can be allocated. This also calls ++ * startup for the cpu DAI, platform, machine and codec DAI. ++ */ ++static int soc_pcm_open(struct snd_pcm_substream *substream) ++{ ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ struct snd_soc_device *socdev = rtd->socdev; ++ struct snd_pcm_runtime *runtime = substream->runtime; ++ struct snd_soc_machine *machine = socdev->machine; ++ struct snd_soc_platform *platform = socdev->platform; ++ struct snd_soc_codec_dai *codec_dai = rtd->codec_dai; ++ struct snd_soc_cpu_dai *cpu_dai = rtd->cpu_dai; ++ int ret = 0; ++ ++ mutex_lock(&pcm_mutex); ++ ++ /* startup the audio subsystem */ ++ if (rtd->cpu_dai->ops.startup) { ++ ret = rtd->cpu_dai->ops.startup(substream); ++ if (ret < 0) { ++ printk(KERN_ERR "asoc: can't open interface %s\n", ++ rtd->cpu_dai->name); ++ goto out; ++ } ++ } ++ ++ if (platform->pcm_ops->open) { ++ ret = platform->pcm_ops->open(substream); ++ if (ret < 0) { ++ printk(KERN_ERR "asoc: can't open platform %s\n", platform->name); ++ goto platform_err; ++ } ++ } ++ ++ if (machine->ops && machine->ops->startup) { ++ ret = machine->ops->startup(substream); ++ if (ret < 0) { ++ printk(KERN_ERR "asoc: %s startup failed\n", machine->name); ++ goto machine_err; ++ } ++ } ++ ++ if (rtd->codec_dai->ops.startup) { ++ ret = rtd->codec_dai->ops.startup(substream); ++ if (ret < 0) { ++ printk(KERN_ERR "asoc: can't open codec %s\n", ++ rtd->codec_dai->name); ++ goto codec_dai_err; ++ } ++ } ++ ++ /* create runtime params from DMA, codec and cpu DAI */ ++ if (runtime->hw.rates) ++ runtime->hw.rates &= ++ get_rates(codec_dai->caps.mode, codec_dai->caps.num_modes) & ++ get_rates(cpu_dai->caps.mode, cpu_dai->caps.num_modes); ++ else ++ runtime->hw.rates = ++ get_rates(codec_dai->caps.mode, codec_dai->caps.num_modes) & ++ get_rates(cpu_dai->caps.mode, cpu_dai->caps.num_modes); ++ if (runtime->hw.formats) ++ runtime->hw.formats &= ++ get_formats(codec_dai->caps.mode, codec_dai->caps.num_modes) & ++ get_formats(cpu_dai->caps.mode, cpu_dai->caps.num_modes); ++ else ++ runtime->hw.formats = ++ get_formats(codec_dai->caps.mode, codec_dai->caps.num_modes) & ++ get_formats(cpu_dai->caps.mode, cpu_dai->caps.num_modes); ++ ++ /* Check that the codec and cpu DAI's are compatible */ ++ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { ++ runtime->hw.rate_min = ++ max(rtd->codec_dai->playback.rate_min, ++ rtd->cpu_dai->playback.rate_min); ++ runtime->hw.rate_max = ++ min(rtd->codec_dai->playback.rate_max, ++ rtd->cpu_dai->playback.rate_max); ++ runtime->hw.channels_min = ++ max(rtd->codec_dai->playback.channels_min, ++ rtd->cpu_dai->playback.channels_min); ++ runtime->hw.channels_max = ++ min(rtd->codec_dai->playback.channels_max, ++ rtd->cpu_dai->playback.channels_max); ++ } else { ++ runtime->hw.rate_min = ++ max(rtd->codec_dai->capture.rate_min, ++ rtd->cpu_dai->capture.rate_min); ++ runtime->hw.rate_max = ++ min(rtd->codec_dai->capture.rate_max, ++ rtd->cpu_dai->capture.rate_max); ++ runtime->hw.channels_min = ++ max(rtd->codec_dai->capture.channels_min, ++ rtd->cpu_dai->capture.channels_min); ++ runtime->hw.channels_max = ++ min(rtd->codec_dai->capture.channels_max, ++ rtd->cpu_dai->capture.channels_max); ++ } ++ ++ snd_pcm_limit_hw_rates(runtime); ++ if (!runtime->hw.rates) { ++ printk(KERN_ERR "asoc: %s <-> %s No matching rates\n", ++ rtd->codec_dai->name, rtd->cpu_dai->name); ++ goto codec_dai_err; ++ } ++ if (!runtime->hw.formats) { ++ printk(KERN_ERR "asoc: %s <-> %s No matching formats\n", ++ rtd->codec_dai->name, rtd->cpu_dai->name); ++ goto codec_dai_err; ++ } ++ if (!runtime->hw.channels_min || !runtime->hw.channels_max) { ++ printk(KERN_ERR "asoc: %s <-> %s No matching channels\n", ++ rtd->codec_dai->name, rtd->cpu_dai->name); ++ goto codec_dai_err; ++ } ++ ++ dbg("asoc: %s <-> %s info:\n", rtd->codec_dai->name, rtd->cpu_dai->name); ++ dbg("asoc: rate mask 0x%x\n", runtime->hw.rates); ++ dbg("asoc: min ch %d max ch %d\n", runtime->hw.channels_min, ++ runtime->hw.channels_max); ++ dbg("asoc: min rate %d max rate %d\n", runtime->hw.rate_min, ++ runtime->hw.rate_max); ++ ++ ++ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ++ rtd->cpu_dai->playback.active = rtd->codec_dai->playback.active = 1; ++ else ++ rtd->cpu_dai->capture.active = rtd->codec_dai->capture.active = 1; ++ rtd->cpu_dai->active = rtd->codec_dai->active = 1; ++ rtd->cpu_dai->runtime = runtime; ++ socdev->codec->active++; ++ mutex_unlock(&pcm_mutex); ++ return 0; ++ ++codec_dai_err: ++ if (machine->ops && machine->ops->shutdown) ++ machine->ops->shutdown(substream); ++ ++machine_err: ++ if (platform->pcm_ops->close) ++ platform->pcm_ops->close(substream); ++ ++platform_err: ++ if (rtd->cpu_dai->ops.shutdown) ++ rtd->cpu_dai->ops.shutdown(substream); ++out: ++ mutex_unlock(&pcm_mutex); ++ return ret; ++} ++ ++/* ++ * Power down the audio subsytem pmdown_time msecs after close is called. ++ * This is to ensure there are no pops or clicks in between any music tracks ++ * due to DAPM power cycling. ++ */ ++static void close_delayed_work(void *data) ++{ ++ struct snd_soc_device *socdev = data; ++ struct snd_soc_codec *codec = socdev->codec; ++ struct snd_soc_codec_dai *codec_dai; ++ int i; ++ ++ mutex_lock(&pcm_mutex); ++ for(i = 0; i < codec->num_dai; i++) { ++ codec_dai = &codec->dai[i]; ++ ++ dbg("pop wq checking: %s status: %s waiting: %s\n", ++ codec_dai->playback.stream_name, ++ codec_dai->playback.active ? "active" : "inactive", ++ codec_dai->pop_wait ? "yes" : "no"); ++ ++ /* are we waiting on this codec DAI stream */ ++ if (codec_dai->pop_wait == 1) { ++ ++ codec_dai->pop_wait = 0; ++ snd_soc_dapm_stream_event(codec, codec_dai->playback.stream_name, ++ SND_SOC_DAPM_STREAM_STOP); ++ ++ /* power down the codec power domain if no longer active */ ++ if (codec->active == 0) { ++ dbg("pop wq D3 %s %s\n", codec->name, ++ codec_dai->playback.stream_name); ++ if (codec->dapm_event) ++ codec->dapm_event(codec, SNDRV_CTL_POWER_D3hot); ++ } ++ } ++ } ++ mutex_unlock(&pcm_mutex); ++} ++ ++/* ++ * Called by ALSA when a PCM substream is closed. Private data can be ++ * freed here. The cpu DAI, codec DAI, machine and platform are also ++ * shutdown. ++ */ ++static int soc_codec_close(struct snd_pcm_substream *substream) ++{ ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ struct snd_soc_device *socdev = rtd->socdev; ++ struct snd_soc_machine *machine = socdev->machine; ++ struct snd_soc_platform *platform = socdev->platform; ++ struct snd_soc_codec *codec = socdev->codec; ++ ++ mutex_lock(&pcm_mutex); ++ ++ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ++ rtd->cpu_dai->playback.active = rtd->codec_dai->playback.active = 0; ++ else ++ rtd->cpu_dai->capture.active = rtd->codec_dai->capture.active = 0; ++ ++ if (rtd->codec_dai->playback.active == 0 && ++ rtd->codec_dai->capture.active == 0) { ++ rtd->cpu_dai->active = rtd->codec_dai->active = 0; ++ } ++ codec->active--; ++ ++ if (rtd->cpu_dai->ops.shutdown) ++ rtd->cpu_dai->ops.shutdown(substream); ++ ++ if (rtd->codec_dai->ops.shutdown) ++ rtd->codec_dai->ops.shutdown(substream); ++ ++ if (machine->ops && machine->ops->shutdown) ++ machine->ops->shutdown(substream); ++ ++ if (platform->pcm_ops->close) ++ platform->pcm_ops->close(substream); ++ rtd->cpu_dai->runtime = NULL; ++ ++ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { ++ /* start delayed pop wq here for playback streams */ ++ rtd->codec_dai->pop_wait = 1; ++ queue_delayed_work(soc_workq, &soc_stream_work, ++ msecs_to_jiffies(pmdown_time)); ++ } else { ++ /* capture streams can be powered down now */ ++ snd_soc_dapm_stream_event(codec, rtd->codec_dai->capture.stream_name, ++ SND_SOC_DAPM_STREAM_STOP); ++ ++ if (codec->active == 0 && rtd->codec_dai->pop_wait == 0){ ++ if (codec->dapm_event) ++ codec->dapm_event(codec, SNDRV_CTL_POWER_D3hot); ++ } ++ } ++ ++ mutex_unlock(&pcm_mutex); ++ return 0; ++} ++ ++/* ++ * Called by ALSA when the PCM substream is prepared, can set format, sample ++ * rate, etc. This function is non atomic and can be called multiple times, ++ * it can refer to the runtime info. ++ */ ++static int soc_pcm_prepare(struct snd_pcm_substream *substream) ++{ ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ struct snd_soc_device *socdev = rtd->socdev; ++ struct snd_soc_platform *platform = socdev->platform; ++ struct snd_soc_codec *codec = socdev->codec; ++ int ret = 0; ++ ++ mutex_lock(&pcm_mutex); ++ if (platform->pcm_ops->prepare) { ++ ret = platform->pcm_ops->prepare(substream); ++ if (ret < 0) { ++ printk(KERN_ERR "asoc: platform prepare error\n"); ++ goto out; ++ } ++ } ++ ++ if (rtd->codec_dai->ops.prepare) { ++ ret = rtd->codec_dai->ops.prepare(substream); ++ if (ret < 0) { ++ printk(KERN_ERR "asoc: codec DAI prepare error\n"); ++ goto out; ++ } ++ } ++ ++ if (rtd->cpu_dai->ops.prepare) ++ ret = rtd->cpu_dai->ops.prepare(substream); ++ ++ /* we only want to start a DAPM playback stream if we are not waiting ++ * on an existing one stopping */ ++ if (rtd->codec_dai->pop_wait) { ++ /* we are waiting for the delayed work to start */ ++ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) ++ snd_soc_dapm_stream_event(codec, ++ rtd->codec_dai->capture.stream_name, ++ SND_SOC_DAPM_STREAM_START); ++ else { ++ rtd->codec_dai->pop_wait = 0; ++ cancel_delayed_work(&soc_stream_work); ++ if (rtd->codec_dai->digital_mute) ++ rtd->codec_dai->digital_mute(codec, rtd->codec_dai, 0); ++ } ++ } else { ++ /* no delayed work - do we need to power up codec */ ++ if (codec->dapm_state != SNDRV_CTL_POWER_D0) { ++ ++ if (codec->dapm_event) ++ codec->dapm_event(codec, SNDRV_CTL_POWER_D1); ++ ++ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ++ snd_soc_dapm_stream_event(codec, ++ rtd->codec_dai->playback.stream_name, ++ SND_SOC_DAPM_STREAM_START); ++ else ++ snd_soc_dapm_stream_event(codec, ++ rtd->codec_dai->capture.stream_name, ++ SND_SOC_DAPM_STREAM_START); ++ ++ if (codec->dapm_event) ++ codec->dapm_event(codec, SNDRV_CTL_POWER_D0); ++ if (rtd->codec_dai->digital_mute) ++ rtd->codec_dai->digital_mute(codec, rtd->codec_dai, 0); ++ ++ } else { ++ /* codec already powered - power on widgets */ ++ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ++ snd_soc_dapm_stream_event(codec, ++ rtd->codec_dai->playback.stream_name, ++ SND_SOC_DAPM_STREAM_START); ++ else ++ snd_soc_dapm_stream_event(codec, ++ rtd->codec_dai->capture.stream_name, ++ SND_SOC_DAPM_STREAM_START); ++ if (rtd->codec_dai->digital_mute) ++ rtd->codec_dai->digital_mute(codec, rtd->codec_dai, 0); ++ } ++ } ++ ++out: ++ mutex_unlock(&pcm_mutex); ++ return ret; ++} ++ ++/* ++ * Called by ALSA when the hardware params are set by application. This ++ * function can also be called multiple times and can allocate buffers ++ * (using snd_pcm_lib_* ). It's non-atomic. ++ */ ++static int soc_pcm_hw_params(struct snd_pcm_substream *substream, ++ struct snd_pcm_hw_params *params) ++{ ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ struct snd_soc_device *socdev = rtd->socdev; ++ struct snd_soc_platform *platform = socdev->platform; ++ struct snd_soc_machine *machine = socdev->machine; ++ int ret = 0; ++ ++ mutex_lock(&pcm_mutex); ++ ++ /* we don't need to match any AC97 params */ ++ if (rtd->cpu_dai->type != SND_SOC_DAI_AC97) { ++ ret = soc_hw_match_params(substream, params); ++ if (ret < 0) ++ goto out; ++ } else { ++ struct snd_soc_clock_info clk_info; ++ clk_info.rate = params_rate(params); ++ ret = soc_get_mclk(rtd, &clk_info); ++ if (ret < 0) ++ goto out; ++ } ++ ++ if (rtd->codec_dai->ops.hw_params) { ++ ret = rtd->codec_dai->ops.hw_params(substream, params); ++ if (ret < 0) { ++ printk(KERN_ERR "asoc: can't set codec %s hw params\n", ++ rtd->codec_dai->name); ++ goto out; ++ } ++ } ++ ++ if (rtd->cpu_dai->ops.hw_params) { ++ ret = rtd->cpu_dai->ops.hw_params(substream, params); ++ if (ret < 0) { ++ printk(KERN_ERR "asoc: can't set interface %s hw params\n", ++ rtd->cpu_dai->name); ++ goto interface_err; ++ } ++ } ++ ++ if (platform->pcm_ops->hw_params) { ++ ret = platform->pcm_ops->hw_params(substream, params); ++ if (ret < 0) { ++ printk(KERN_ERR "asoc: can't set platform %s hw params\n", ++ platform->name); ++ goto platform_err; ++ } ++ } ++ ++ if (machine->ops && machine->ops->hw_params) { ++ ret = machine->ops->hw_params(substream, params); ++ if (ret < 0) { ++ printk(KERN_ERR "asoc: machine hw_params failed\n"); ++ goto machine_err; ++ } ++ } ++ ++out: ++ mutex_unlock(&pcm_mutex); ++ return ret; ++ ++machine_err: ++ if (platform->pcm_ops->hw_free) ++ platform->pcm_ops->hw_free(substream); ++ ++platform_err: ++ if (rtd->cpu_dai->ops.hw_free) ++ rtd->cpu_dai->ops.hw_free(substream); ++ ++interface_err: ++ if (rtd->codec_dai->ops.hw_free) ++ rtd->codec_dai->ops.hw_free(substream); ++ ++ mutex_unlock(&pcm_mutex); ++ return ret; ++} ++ ++/* ++ * Free's resources allocated by hw_params, can be called multiple times ++ */ ++static int soc_pcm_hw_free(struct snd_pcm_substream *substream) ++{ ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ struct snd_soc_device *socdev = rtd->socdev; ++ struct snd_soc_platform *platform = socdev->platform; ++ struct snd_soc_codec *codec = socdev->codec; ++ struct snd_soc_machine *machine = socdev->machine; ++ ++ mutex_lock(&pcm_mutex); ++ ++ /* apply codec digital mute */ ++ if (!codec->active && rtd->codec_dai->digital_mute) ++ rtd->codec_dai->digital_mute(codec, rtd->codec_dai, 1); ++ ++ /* free any machine hw params */ ++ if (machine->ops && machine->ops->hw_free) ++ machine->ops->hw_free(substream); ++ ++ /* free any DMA resources */ ++ if (platform->pcm_ops->hw_free) ++ platform->pcm_ops->hw_free(substream); ++ ++ /* now free hw params for the DAI's */ ++ if (rtd->codec_dai->ops.hw_free) ++ rtd->codec_dai->ops.hw_free(substream); ++ ++ if (rtd->cpu_dai->ops.hw_free) ++ rtd->cpu_dai->ops.hw_free(substream); ++ ++ mutex_unlock(&pcm_mutex); ++ return 0; ++} ++ ++static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) ++{ ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ struct snd_soc_device *socdev = rtd->socdev; ++ struct snd_soc_platform *platform = socdev->platform; ++ int ret; ++ ++ if (rtd->codec_dai->ops.trigger) { ++ ret = rtd->codec_dai->ops.trigger(substream, cmd); ++ if (ret < 0) ++ return ret; ++ } ++ ++ if (platform->pcm_ops->trigger) { ++ ret = platform->pcm_ops->trigger(substream, cmd); ++ if (ret < 0) ++ return ret; ++ } ++ ++ if (rtd->cpu_dai->ops.trigger) { ++ ret = rtd->cpu_dai->ops.trigger(substream, cmd); ++ if (ret < 0) ++ return ret; ++ } ++ return 0; ++} ++ ++/* ASoC PCM operations */ ++static struct snd_pcm_ops soc_pcm_ops = { ++ .open = soc_pcm_open, ++ .close = soc_codec_close, ++ .hw_params = soc_pcm_hw_params, ++ .hw_free = soc_pcm_hw_free, ++ .prepare = soc_pcm_prepare, ++ .trigger = soc_pcm_trigger, ++}; ++ ++#ifdef CONFIG_PM ++/* powers down audio subsystem for suspend */ ++static int soc_suspend(struct platform_device *pdev, pm_message_t state) ++{ ++ struct snd_soc_device *socdev = platform_get_drvdata(pdev); ++ struct snd_soc_machine *machine = socdev->machine; ++ struct snd_soc_platform *platform = socdev->platform; ++ struct snd_soc_codec_device *codec_dev = socdev->codec_dev; ++ struct snd_soc_codec *codec = socdev->codec; ++ int i; ++ ++ /* mute any active DAC's */ ++ for(i = 0; i < machine->num_links; i++) { ++ struct snd_soc_codec_dai *dai = machine->dai_link[i].codec_dai; ++ if (dai->digital_mute && dai->playback.active) ++ dai->digital_mute(codec, dai, 1); ++ } ++ ++ if (machine->suspend_pre) ++ machine->suspend_pre(pdev, state); ++ ++ for(i = 0; i < machine->num_links; i++) { ++ struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai; ++ if (cpu_dai->suspend && cpu_dai->type != SND_SOC_DAI_AC97) ++ cpu_dai->suspend(pdev, cpu_dai); ++ if (platform->suspend) ++ platform->suspend(pdev, cpu_dai); ++ } ++ ++ /* close any waiting streams and save state */ ++ flush_workqueue(soc_workq); ++ codec->suspend_dapm_state = codec->dapm_state; ++ ++ for(i = 0; i < codec->num_dai; i++) { ++ char *stream = codec->dai[i].playback.stream_name; ++ if (stream != NULL) ++ snd_soc_dapm_stream_event(codec, stream, ++ SND_SOC_DAPM_STREAM_SUSPEND); ++ stream = codec->dai[i].capture.stream_name; ++ if (stream != NULL) ++ snd_soc_dapm_stream_event(codec, stream, ++ SND_SOC_DAPM_STREAM_SUSPEND); ++ } ++ ++ if (codec_dev->suspend) ++ codec_dev->suspend(pdev, state); ++ ++ for(i = 0; i < machine->num_links; i++) { ++ struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai; ++ if (cpu_dai->suspend && cpu_dai->type == SND_SOC_DAI_AC97) ++ cpu_dai->suspend(pdev, cpu_dai); ++ } ++ ++ if (machine->suspend_post) ++ machine->suspend_post(pdev, state); ++ ++ return 0; ++} ++ ++/* powers up audio subsystem after a suspend */ ++static int soc_resume(struct platform_device *pdev) ++{ ++ struct snd_soc_device *socdev = platform_get_drvdata(pdev); ++ struct snd_soc_machine *machine = socdev->machine; ++ struct snd_soc_platform *platform = socdev->platform; ++ struct snd_soc_codec_device *codec_dev = socdev->codec_dev; ++ struct snd_soc_codec *codec = socdev->codec; ++ int i; ++ ++ if (machine->resume_pre) ++ machine->resume_pre(pdev); ++ ++ for(i = 0; i < machine->num_links; i++) { ++ struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai; ++ if (cpu_dai->resume && cpu_dai->type == SND_SOC_DAI_AC97) ++ cpu_dai->resume(pdev, cpu_dai); ++ } ++ ++ if (codec_dev->resume) ++ codec_dev->resume(pdev); ++ ++ for(i = 0; i < codec->num_dai; i++) { ++ char* stream = codec->dai[i].playback.stream_name; ++ if (stream != NULL) ++ snd_soc_dapm_stream_event(codec, stream, ++ SND_SOC_DAPM_STREAM_RESUME); ++ stream = codec->dai[i].capture.stream_name; ++ if (stream != NULL) ++ snd_soc_dapm_stream_event(codec, stream, ++ SND_SOC_DAPM_STREAM_RESUME); ++ } ++ ++ /* unmute any active DAC's */ ++ for(i = 0; i < machine->num_links; i++) { ++ struct snd_soc_codec_dai *dai = machine->dai_link[i].codec_dai; ++ if (dai->digital_mute && dai->playback.active) ++ dai->digital_mute(codec, dai, 0); ++ } ++ ++ for(i = 0; i < machine->num_links; i++) { ++ struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai; ++ if (cpu_dai->resume && cpu_dai->type != SND_SOC_DAI_AC97) ++ cpu_dai->resume(pdev, cpu_dai); ++ if (platform->resume) ++ platform->resume(pdev, cpu_dai); ++ } ++ ++ if (machine->resume_post) ++ machine->resume_post(pdev); ++ ++ return 0; ++} ++ ++#else ++#define soc_suspend NULL ++#define soc_resume NULL ++#endif ++ ++/* probes a new socdev */ ++static int soc_probe(struct platform_device *pdev) ++{ ++ int ret = 0, i; ++ struct snd_soc_device *socdev = platform_get_drvdata(pdev); ++ struct snd_soc_machine *machine = socdev->machine; ++ struct snd_soc_platform *platform = socdev->platform; ++ struct snd_soc_codec_device *codec_dev = socdev->codec_dev; ++ ++ if (machine->probe) { ++ ret = machine->probe(pdev); ++ if(ret < 0) ++ return ret; ++ } ++ ++ for (i = 0; i < machine->num_links; i++) { ++ struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai; ++ if (cpu_dai->probe) { ++ ret = cpu_dai->probe(pdev); ++ if(ret < 0) ++ goto cpu_dai_err; ++ } ++ } ++ ++ if (codec_dev->probe) { ++ ret = codec_dev->probe(pdev); ++ if(ret < 0) ++ goto cpu_dai_err; ++ } ++ ++ if (platform->probe) { ++ ret = platform->probe(pdev); ++ if(ret < 0) ++ goto platform_err; ++ } ++ ++ /* DAPM stream work */ ++ soc_workq = create_workqueue("kdapm"); ++ if (soc_workq == NULL) ++ goto work_err; ++ INIT_WORK(&soc_stream_work, close_delayed_work, socdev); ++ return 0; ++ ++work_err: ++ if (platform->remove) ++ platform->remove(pdev); ++ ++platform_err: ++ if (codec_dev->remove) ++ codec_dev->remove(pdev); ++ ++cpu_dai_err: ++ for (i--; i > 0; i--) { ++ struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai; ++ if (cpu_dai->remove) ++ cpu_dai->remove(pdev); ++ } ++ ++ if (machine->remove) ++ machine->remove(pdev); ++ ++ return ret; ++} ++ ++/* removes a socdev */ ++static int soc_remove(struct platform_device *pdev) ++{ ++ int i; ++ struct snd_soc_device *socdev = platform_get_drvdata(pdev); ++ struct snd_soc_machine *machine = socdev->machine; ++ struct snd_soc_platform *platform = socdev->platform; ++ struct snd_soc_codec_device *codec_dev = socdev->codec_dev; ++ ++ if (soc_workq) ++ destroy_workqueue(soc_workq); ++ ++ if (platform->remove) ++ platform->remove(pdev); ++ ++ if (codec_dev->remove) ++ codec_dev->remove(pdev); ++ ++ for (i = 0; i < machine->num_links; i++) { ++ struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[i].cpu_dai; ++ if (cpu_dai->remove) ++ cpu_dai->remove(pdev); ++ } ++ ++ if (machine->remove) ++ machine->remove(pdev); ++ ++ return 0; ++} ++ ++/* ASoC platform driver */ ++static struct platform_driver soc_driver = { ++ .driver = { ++ .name = "soc-audio", ++ }, ++ .probe = soc_probe, ++ .remove = soc_remove, ++ .suspend = soc_suspend, ++ .resume = soc_resume, ++}; ++ ++/* create a new pcm */ ++static int soc_new_pcm(struct snd_soc_device *socdev, ++ struct snd_soc_dai_link *dai_link, int num) ++{ ++ struct snd_soc_codec *codec = socdev->codec; ++ struct snd_soc_codec_dai *codec_dai = dai_link->codec_dai; ++ struct snd_soc_cpu_dai *cpu_dai = dai_link->cpu_dai; ++ struct snd_soc_pcm_runtime *rtd; ++ struct snd_pcm *pcm; ++ char new_name[64]; ++ int ret = 0, playback = 0, capture = 0; ++ ++ rtd = kzalloc(sizeof(struct snd_soc_pcm_runtime), GFP_KERNEL); ++ if (rtd == NULL) ++ return -ENOMEM; ++ rtd->cpu_dai = cpu_dai; ++ rtd->codec_dai = codec_dai; ++ rtd->socdev = socdev; ++ ++ /* check client and interface hw capabilities */ ++ sprintf(new_name, "%s %s-%s-%d",dai_link->stream_name, codec_dai->name, ++ get_dai_name(cpu_dai->type), num); ++ ++ if (codec_dai->playback.channels_min) ++ playback = 1; ++ if (codec_dai->capture.channels_min) ++ capture = 1; ++ ++ ret = snd_pcm_new(codec->card, new_name, codec->pcm_devs++, playback, ++ capture, &pcm); ++ if (ret < 0) { ++ printk(KERN_ERR "asoc: can't create pcm for codec %s\n", codec->name); ++ kfree(rtd); ++ return ret; ++ } ++ ++ pcm->private_data = rtd; ++ soc_pcm_ops.mmap = socdev->platform->pcm_ops->mmap; ++ soc_pcm_ops.pointer = socdev->platform->pcm_ops->pointer; ++ soc_pcm_ops.ioctl = socdev->platform->pcm_ops->ioctl; ++ soc_pcm_ops.copy = socdev->platform->pcm_ops->copy; ++ soc_pcm_ops.silence = socdev->platform->pcm_ops->silence; ++ soc_pcm_ops.ack = socdev->platform->pcm_ops->ack; ++ soc_pcm_ops.page = socdev->platform->pcm_ops->page; ++ ++ if (playback) ++ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &soc_pcm_ops); ++ ++ if (capture) ++ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &soc_pcm_ops); ++ ++ ret = socdev->platform->pcm_new(codec->card, codec_dai, pcm); ++ if (ret < 0) { ++ printk(KERN_ERR "asoc: platform pcm constructor failed\n"); ++ kfree(rtd); ++ return ret; ++ } ++ ++ pcm->private_free = socdev->platform->pcm_free; ++ printk(KERN_INFO "asoc: %s <-> %s mapping ok\n", codec_dai->name, ++ cpu_dai->name); ++ return ret; ++} ++ ++/* codec register dump */ ++static ssize_t codec_reg_show(struct device *dev, ++ struct device_attribute *attr, char *buf) ++{ ++ struct snd_soc_device *devdata = dev_get_drvdata(dev); ++ struct snd_soc_codec *codec = devdata->codec; ++ int i, step = 1, count = 0; ++ ++ if (!codec->reg_cache_size) ++ return 0; ++ ++ if (codec->reg_cache_step) ++ step = codec->reg_cache_step; ++ ++ count += sprintf(buf, "%s registers\n", codec->name); ++ for(i = 0; i < codec->reg_cache_size; i += step) ++ count += sprintf(buf + count, "%2x: %4x\n", i, codec->read(codec, i)); ++ ++ return count; ++} ++static DEVICE_ATTR(codec_reg, 0444, codec_reg_show, NULL); ++ ++/** ++ * snd_soc_new_ac97_codec - initailise AC97 device ++ * @codec: audio codec ++ * @ops: AC97 bus operations ++ * @num: AC97 codec number ++ * ++ * Initialises AC97 codec resources for use by ad-hoc devices only. ++ */ ++int snd_soc_new_ac97_codec(struct snd_soc_codec *codec, ++ struct snd_ac97_bus_ops *ops, int num) ++{ ++ mutex_lock(&codec->mutex); ++ ++ codec->ac97 = kzalloc(sizeof(struct snd_ac97), GFP_KERNEL); ++ if (codec->ac97 == NULL) { ++ mutex_unlock(&codec->mutex); ++ return -ENOMEM; ++ } ++ ++ codec->ac97->bus = kzalloc(sizeof(struct snd_ac97_bus), GFP_KERNEL); ++ if (codec->ac97->bus == NULL) { ++ kfree(codec->ac97); ++ codec->ac97 = NULL; ++ mutex_unlock(&codec->mutex); ++ return -ENOMEM; ++ } ++ ++ codec->ac97->bus->ops = ops; ++ codec->ac97->num = num; ++ mutex_unlock(&codec->mutex); ++ return 0; ++} ++EXPORT_SYMBOL_GPL(snd_soc_new_ac97_codec); ++ ++/** ++ * snd_soc_free_ac97_codec - free AC97 codec device ++ * @codec: audio codec ++ * ++ * Frees AC97 codec device resources. ++ */ ++void snd_soc_free_ac97_codec(struct snd_soc_codec *codec) ++{ ++ mutex_lock(&codec->mutex); ++ kfree(codec->ac97->bus); ++ kfree(codec->ac97); ++ codec->ac97 = NULL; ++ mutex_unlock(&codec->mutex); ++} ++EXPORT_SYMBOL_GPL(snd_soc_free_ac97_codec); ++ ++/** ++ * snd_soc_update_bits - update codec register bits ++ * @codec: audio codec ++ * @reg: codec register ++ * @mask: register mask ++ * @value: new value ++ * ++ * Writes new register value. ++ * ++ * Returns 1 for change else 0. ++ */ ++int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned short reg, ++ unsigned short mask, unsigned short value) ++{ ++ int change; ++ unsigned short old, new; ++ ++ mutex_lock(&io_mutex); ++ old = snd_soc_read(codec, reg); ++ new = (old & ~mask) | value; ++ change = old != new; ++ if (change) ++ snd_soc_write(codec, reg, new); ++ ++ mutex_unlock(&io_mutex); ++ return change; ++} ++EXPORT_SYMBOL_GPL(snd_soc_update_bits); ++ ++/** ++ * snd_soc_test_bits - test register for change ++ * @codec: audio codec ++ * @reg: codec register ++ * @mask: register mask ++ * @value: new value ++ * ++ * Tests a register with a new value and checks if the new value is ++ * different from the old value. ++ * ++ * Returns 1 for change else 0. ++ */ ++int snd_soc_test_bits(struct snd_soc_codec *codec, unsigned short reg, ++ unsigned short mask, unsigned short value) ++{ ++ int change; ++ unsigned short old, new; ++ ++ mutex_lock(&io_mutex); ++ old = snd_soc_read(codec, reg); ++ new = (old & ~mask) | value; ++ change = old != new; ++ mutex_unlock(&io_mutex); ++ ++ return change; ++} ++EXPORT_SYMBOL_GPL(snd_soc_test_bits); ++ ++/** ++ * snd_soc_get_rate - get int sample rate ++ * @hwpcmrate: the hardware pcm rate ++ * ++ * Returns the audio rate integaer value, else 0. ++ */ ++int snd_soc_get_rate(int hwpcmrate) ++{ ++ int rate = ffs(hwpcmrate) - 1; ++ ++ if (rate > ARRAY_SIZE(rates)) ++ return 0; ++ return rates[rate]; ++} ++EXPORT_SYMBOL_GPL(snd_soc_get_rate); ++ ++/** ++ * snd_soc_new_pcms - create new sound card and pcms ++ * @socdev: the SoC audio device ++ * ++ * Create a new sound card based upon the codec and interface pcms. ++ * ++ * Returns 0 for success, else error. ++ */ ++int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char * xid) ++{ ++ struct snd_soc_codec *codec = socdev->codec; ++ struct snd_soc_machine *machine = socdev->machine; ++ int ret = 0, i; ++ ++ mutex_lock(&codec->mutex); ++ ++ /* register a sound card */ ++ codec->card = snd_card_new(idx, xid, codec->owner, 0); ++ if (!codec->card) { ++ printk(KERN_ERR "asoc: can't create sound card for codec %s\n", ++ codec->name); ++ mutex_unlock(&codec->mutex); ++ return -ENODEV; ++ } ++ ++ codec->card->dev = socdev->dev; ++ codec->card->private_data = codec; ++ strncpy(codec->card->driver, codec->name, sizeof(codec->card->driver)); ++ ++ /* create the pcms */ ++ for(i = 0; i < machine->num_links; i++) { ++ ret = soc_new_pcm(socdev, &machine->dai_link[i], i); ++ if (ret < 0) { ++ printk(KERN_ERR "asoc: can't create pcm %s\n", ++ machine->dai_link[i].stream_name); ++ mutex_unlock(&codec->mutex); ++ return ret; ++ } ++ } ++ ++ mutex_unlock(&codec->mutex); ++ return ret; ++} ++EXPORT_SYMBOL_GPL(snd_soc_new_pcms); ++ ++/** ++ * snd_soc_register_card - register sound card ++ * @socdev: the SoC audio device ++ * ++ * Register a SoC sound card. Also registers an AC97 device if the ++ * codec is AC97 for ad hoc devices. ++ * ++ * Returns 0 for success, else error. ++ */ ++int snd_soc_register_card(struct snd_soc_device *socdev) ++{ ++ struct snd_soc_codec *codec = socdev->codec; ++ struct snd_soc_machine *machine = socdev->machine; ++ int ret = 0, i, ac97 = 0, err = 0; ++ ++ mutex_lock(&codec->mutex); ++ for(i = 0; i < machine->num_links; i++) { ++ if (socdev->machine->dai_link[i].init) { ++ err = socdev->machine->dai_link[i].init(codec); ++ if (err < 0) { ++ printk(KERN_ERR "asoc: failed to init %s\n", ++ socdev->machine->dai_link[i].stream_name); ++ continue; ++ } ++ } ++ if (socdev->machine->dai_link[i].cpu_dai->type == SND_SOC_DAI_AC97) ++ ac97 = 1; ++ } ++ snprintf(codec->card->shortname, sizeof(codec->card->shortname), ++ "%s", machine->name); ++ snprintf(codec->card->longname, sizeof(codec->card->longname), ++ "%s (%s)", machine->name, codec->name); ++ ++ ret = snd_card_register(codec->card); ++ if (ret < 0) { ++ printk(KERN_ERR "asoc: failed to register soundcard for codec %s\n", ++ codec->name); ++ goto out; ++ } ++ ++#ifdef CONFIG_SND_SOC_AC97_BUS ++ if (ac97) { ++ ret = soc_ac97_dev_register(codec); ++ if (ret < 0) { ++ printk(KERN_ERR "asoc: AC97 device register failed\n"); ++ snd_card_free(codec->card); ++ goto out; ++ } ++ } ++#endif ++ ++ err = snd_soc_dapm_sys_add(socdev->dev); ++ if (err < 0) ++ printk(KERN_WARNING "asoc: failed to add dapm sysfs entries\n"); ++ ++ err = device_create_file(socdev->dev, &dev_attr_codec_reg); ++ if (err < 0) ++ printk(KERN_WARNING "asoc: failed to add codec sysfs entries\n"); ++out: ++ mutex_unlock(&codec->mutex); ++ return ret; ++} ++EXPORT_SYMBOL_GPL(snd_soc_register_card); ++ ++/** ++ * snd_soc_free_pcms - free sound card and pcms ++ * @socdev: the SoC audio device ++ * ++ * Frees sound card and pcms associated with the socdev. ++ * Also unregister the codec if it is an AC97 device. ++ */ ++void snd_soc_free_pcms(struct snd_soc_device *socdev) ++{ ++ struct snd_soc_codec *codec = socdev->codec; ++ ++ mutex_lock(&codec->mutex); ++#ifdef CONFIG_SND_SOC_AC97_BUS ++ if (codec->ac97) ++ soc_ac97_dev_unregister(codec); ++#endif ++ ++ if (codec->card) ++ snd_card_free(codec->card); ++ device_remove_file(socdev->dev, &dev_attr_codec_reg); ++ mutex_unlock(&codec->mutex); ++} ++EXPORT_SYMBOL_GPL(snd_soc_free_pcms); ++ ++/** ++ * snd_soc_set_runtime_hwparams - set the runtime hardware parameters ++ * @substream: the pcm substream ++ * @hw: the hardware parameters ++ * ++ * Sets the substream runtime hardware parameters. ++ */ ++int snd_soc_set_runtime_hwparams(struct snd_pcm_substream *substream, ++ const struct snd_pcm_hardware *hw) ++{ ++ struct snd_pcm_runtime *runtime = substream->runtime; ++ runtime->hw.info = hw->info; ++ runtime->hw.formats = hw->formats; ++ runtime->hw.period_bytes_min = hw->period_bytes_min; ++ runtime->hw.period_bytes_max = hw->period_bytes_max; ++ runtime->hw.periods_min = hw->periods_min; ++ runtime->hw.periods_max = hw->periods_max; ++ runtime->hw.buffer_bytes_max = hw->buffer_bytes_max; ++ runtime->hw.fifo_size = hw->fifo_size; ++ return 0; ++} ++EXPORT_SYMBOL_GPL(snd_soc_set_runtime_hwparams); ++ ++/** ++ * snd_soc_cnew - create new control ++ * @_template: control template ++ * @data: control private data ++ * @lnng_name: control long name ++ * ++ * Create a new mixer control from a template control. ++ * ++ * Returns 0 for success, else error. ++ */ ++struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template, ++ void *data, char *long_name) ++{ ++ struct snd_kcontrol_new template; ++ ++ memcpy(&template, _template, sizeof(template)); ++ if (long_name) ++ template.name = long_name; ++ template.access = SNDRV_CTL_ELEM_ACCESS_READWRITE; ++ template.index = 0; ++ ++ return snd_ctl_new1(&template, data); ++} ++EXPORT_SYMBOL_GPL(snd_soc_cnew); ++ ++/** ++ * snd_soc_info_enum_double - enumerated double mixer info callback ++ * @kcontrol: mixer control ++ * @uinfo: control element information ++ * ++ * Callback to provide information about a double enumerated ++ * mixer control. ++ * ++ * Returns 0 for success. ++ */ ++int snd_soc_info_enum_double(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_info *uinfo) ++{ ++ struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; ++ ++ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; ++ uinfo->count = e->shift_l == e->shift_r ? 1 : 2; ++ uinfo->value.enumerated.items = e->mask; ++ ++ if (uinfo->value.enumerated.item > e->mask - 1) ++ uinfo->value.enumerated.item = e->mask - 1; ++ strcpy(uinfo->value.enumerated.name, ++ e->texts[uinfo->value.enumerated.item]); ++ return 0; ++} ++EXPORT_SYMBOL_GPL(snd_soc_info_enum_double); ++ ++/** ++ * snd_soc_get_enum_double - enumerated double mixer get callback ++ * @kcontrol: mixer control ++ * @uinfo: control element information ++ * ++ * Callback to get the value of a double enumerated mixer. ++ * ++ * Returns 0 for success. ++ */ ++int snd_soc_get_enum_double(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_value *ucontrol) ++{ ++ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); ++ struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; ++ unsigned short val, bitmask; ++ ++ for (bitmask = 1; bitmask < e->mask; bitmask <<= 1) ++ ; ++ val = snd_soc_read(codec, e->reg); ++ ucontrol->value.enumerated.item[0] = (val >> e->shift_l) & (bitmask - 1); ++ if (e->shift_l != e->shift_r) ++ ucontrol->value.enumerated.item[1] = ++ (val >> e->shift_r) & (bitmask - 1); ++ ++ return 0; ++} ++EXPORT_SYMBOL_GPL(snd_soc_get_enum_double); ++ ++/** ++ * snd_soc_put_enum_double - enumerated double mixer put callback ++ * @kcontrol: mixer control ++ * @uinfo: control element information ++ * ++ * Callback to set the value of a double enumerated mixer. ++ * ++ * Returns 0 for success. ++ */ ++int snd_soc_put_enum_double(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_value *ucontrol) ++{ ++ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); ++ struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; ++ unsigned short val; ++ unsigned short mask, bitmask; ++ ++ for (bitmask = 1; bitmask < e->mask; bitmask <<= 1) ++ ; ++ if (ucontrol->value.enumerated.item[0] > e->mask - 1) ++ return -EINVAL; ++ val = ucontrol->value.enumerated.item[0] << e->shift_l; ++ mask = (bitmask - 1) << e->shift_l; ++ if (e->shift_l != e->shift_r) { ++ if (ucontrol->value.enumerated.item[1] > e->mask - 1) ++ return -EINVAL; ++ val |= ucontrol->value.enumerated.item[1] << e->shift_r; ++ mask |= (bitmask - 1) << e->shift_r; ++ } ++ ++ return snd_soc_update_bits(codec, e->reg, mask, val); ++} ++EXPORT_SYMBOL_GPL(snd_soc_put_enum_double); ++ ++/** ++ * snd_soc_info_enum_ext - external enumerated single mixer info callback ++ * @kcontrol: mixer control ++ * @uinfo: control element information ++ * ++ * Callback to provide information about an external enumerated ++ * single mixer. ++ * ++ * Returns 0 for success. ++ */ ++int snd_soc_info_enum_ext(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_info *uinfo) ++{ ++ struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; ++ ++ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; ++ uinfo->count = 1; ++ uinfo->value.enumerated.items = e->mask; ++ ++ if (uinfo->value.enumerated.item > e->mask - 1) ++ uinfo->value.enumerated.item = e->mask - 1; ++ strcpy(uinfo->value.enumerated.name, ++ e->texts[uinfo->value.enumerated.item]); ++ return 0; ++} ++EXPORT_SYMBOL_GPL(snd_soc_info_enum_ext); ++ ++/** ++ * snd_soc_info_volsw_ext - external single mixer info callback ++ * @kcontrol: mixer control ++ * @uinfo: control element information ++ * ++ * Callback to provide information about a single external mixer control. ++ * ++ * Returns 0 for success. ++ */ ++int snd_soc_info_volsw_ext(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_info *uinfo) ++{ ++ int mask = kcontrol->private_value; ++ ++ uinfo->type = ++ mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER; ++ uinfo->count = 1; ++ uinfo->value.integer.min = 0; ++ uinfo->value.integer.max = mask; ++ return 0; ++} ++EXPORT_SYMBOL_GPL(snd_soc_info_volsw_ext); ++ ++/** ++ * snd_soc_info_bool_ext - external single boolean mixer info callback ++ * @kcontrol: mixer control ++ * @uinfo: control element information ++ * ++ * Callback to provide information about a single boolean external mixer control. ++ * ++ * Returns 0 for success. ++ */ ++int snd_soc_info_bool_ext(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_info *uinfo) ++{ ++ uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; ++ uinfo->count = 1; ++ uinfo->value.integer.min = 0; ++ uinfo->value.integer.max = 1; ++ return 0; ++} ++EXPORT_SYMBOL_GPL(snd_soc_info_bool_ext); ++ ++/** ++ * snd_soc_info_volsw - single mixer info callback ++ * @kcontrol: mixer control ++ * @uinfo: control element information ++ * ++ * Callback to provide information about a single mixer control. ++ * ++ * Returns 0 for success. ++ */ ++int snd_soc_info_volsw(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_info *uinfo) ++{ ++ int mask = (kcontrol->private_value >> 16) & 0xff; ++ int shift = (kcontrol->private_value >> 8) & 0x0f; ++ int rshift = (kcontrol->private_value >> 12) & 0x0f; ++ ++ uinfo->type = ++ mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER; ++ uinfo->count = shift == rshift ? 1 : 2; ++ uinfo->value.integer.min = 0; ++ uinfo->value.integer.max = mask; ++ return 0; ++} ++EXPORT_SYMBOL_GPL(snd_soc_info_volsw); ++ ++/** ++ * snd_soc_get_volsw - single mixer get callback ++ * @kcontrol: mixer control ++ * @uinfo: control element information ++ * ++ * Callback to get the value of a single mixer control. ++ * ++ * Returns 0 for success. ++ */ ++int snd_soc_get_volsw(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_value *ucontrol) ++{ ++ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); ++ int reg = kcontrol->private_value & 0xff; ++ int shift = (kcontrol->private_value >> 8) & 0x0f; ++ int rshift = (kcontrol->private_value >> 12) & 0x0f; ++ int mask = (kcontrol->private_value >> 16) & 0xff; ++ int invert = (kcontrol->private_value >> 24) & 0x01; ++ ++ ucontrol->value.integer.value[0] = ++ (snd_soc_read(codec, reg) >> shift) & mask; ++ if (shift != rshift) ++ ucontrol->value.integer.value[1] = ++ (snd_soc_read(codec, reg) >> rshift) & mask; ++ if (invert) { ++ ucontrol->value.integer.value[0] = ++ mask - ucontrol->value.integer.value[0]; ++ if (shift != rshift) ++ ucontrol->value.integer.value[1] = ++ mask - ucontrol->value.integer.value[1]; ++ } ++ ++ return 0; ++} ++EXPORT_SYMBOL_GPL(snd_soc_get_volsw); ++ ++/** ++ * snd_soc_put_volsw - single mixer put callback ++ * @kcontrol: mixer control ++ * @uinfo: control element information ++ * ++ * Callback to set the value of a single mixer control. ++ * ++ * Returns 0 for success. ++ */ ++int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_value *ucontrol) ++{ ++ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); ++ int reg = kcontrol->private_value & 0xff; ++ int shift = (kcontrol->private_value >> 8) & 0x0f; ++ int rshift = (kcontrol->private_value >> 12) & 0x0f; ++ int mask = (kcontrol->private_value >> 16) & 0xff; ++ int invert = (kcontrol->private_value >> 24) & 0x01; ++ int err; ++ unsigned short val, val2, val_mask; ++ ++ val = (ucontrol->value.integer.value[0] & mask); ++ if (invert) ++ val = mask - val; ++ val_mask = mask << shift; ++ val = val << shift; ++ if (shift != rshift) { ++ val2 = (ucontrol->value.integer.value[1] & mask); ++ if (invert) ++ val2 = mask - val2; ++ val_mask |= mask << rshift; ++ val |= val2 << rshift; ++ } ++ err = snd_soc_update_bits(codec, reg, val_mask, val); ++ return err; ++} ++EXPORT_SYMBOL_GPL(snd_soc_put_volsw); ++ ++/** ++ * snd_soc_info_volsw_2r - double mixer info callback ++ * @kcontrol: mixer control ++ * @uinfo: control element information ++ * ++ * Callback to provide information about a double mixer control that ++ * spans 2 codec registers. ++ * ++ * Returns 0 for success. ++ */ ++int snd_soc_info_volsw_2r(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_info *uinfo) ++{ ++ int mask = (kcontrol->private_value >> 12) & 0xff; ++ ++ uinfo->type = ++ mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER; ++ uinfo->count = 2; ++ uinfo->value.integer.min = 0; ++ uinfo->value.integer.max = mask; ++ return 0; ++} ++EXPORT_SYMBOL_GPL(snd_soc_info_volsw_2r); ++ ++/** ++ * snd_soc_get_volsw_2r - double mixer get callback ++ * @kcontrol: mixer control ++ * @uinfo: control element information ++ * ++ * Callback to get the value of a double mixer control that spans 2 registers. ++ * ++ * Returns 0 for success. ++ */ ++int snd_soc_get_volsw_2r(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_value *ucontrol) ++{ ++ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); ++ int reg = kcontrol->private_value & 0xff; ++ int reg2 = (kcontrol->private_value >> 24) & 0xff; ++ int shift = (kcontrol->private_value >> 8) & 0x0f; ++ int mask = (kcontrol->private_value >> 12) & 0xff; ++ int invert = (kcontrol->private_value >> 20) & 0x01; ++ ++ ucontrol->value.integer.value[0] = ++ (snd_soc_read(codec, reg) >> shift) & mask; ++ ucontrol->value.integer.value[1] = ++ (snd_soc_read(codec, reg2) >> shift) & mask; ++ if (invert) { ++ ucontrol->value.integer.value[0] = ++ mask - ucontrol->value.integer.value[0]; ++ ucontrol->value.integer.value[1] = ++ mask - ucontrol->value.integer.value[1]; ++ } ++ ++ return 0; ++} ++EXPORT_SYMBOL_GPL(snd_soc_get_volsw_2r); ++ ++/** ++ * snd_soc_put_volsw_2r - double mixer set callback ++ * @kcontrol: mixer control ++ * @uinfo: control element information ++ * ++ * Callback to set the value of a double mixer control that spans 2 registers. ++ * ++ * Returns 0 for success. ++ */ ++int snd_soc_put_volsw_2r(struct snd_kcontrol *kcontrol, ++ struct snd_ctl_elem_value *ucontrol) ++{ ++ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); ++ int reg = kcontrol->private_value & 0xff; ++ int reg2 = (kcontrol->private_value >> 24) & 0xff; ++ int shift = (kcontrol->private_value >> 8) & 0x0f; ++ int mask = (kcontrol->private_value >> 12) & 0xff; ++ int invert = (kcontrol->private_value >> 20) & 0x01; ++ int err; ++ unsigned short val, val2, val_mask; ++ ++ val_mask = mask << shift; ++ val = (ucontrol->value.integer.value[0] & mask); ++ val2 = (ucontrol->value.integer.value[1] & mask); ++ ++ if (invert) { ++ val = mask - val; ++ val2 = mask - val2; ++ } ++ ++ val = val << shift; ++ val2 = val2 << shift; ++ ++ if ((err = snd_soc_update_bits(codec, reg, val_mask, val)) < 0) ++ return err; ++ ++ err = snd_soc_update_bits(codec, reg2, val_mask, val2); ++ return err; ++} ++EXPORT_SYMBOL_GPL(snd_soc_put_volsw_2r); ++ ++static int __devinit snd_soc_init(void) ++{ ++ printk(KERN_INFO "ASoC version %s\n", SND_SOC_VERSION); ++ return platform_driver_register(&soc_driver); ++} ++ ++static void snd_soc_exit(void) ++{ ++ platform_driver_unregister(&soc_driver); ++} ++ ++module_init(snd_soc_init); ++module_exit(snd_soc_exit); ++ ++/* Module information */ ++MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com"); ++MODULE_DESCRIPTION("ALSA SoC Core"); ++MODULE_LICENSE("GPL"); +Index: linux-2.6-pxa-new/sound/soc/at91/Kconfig +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/sound/soc/at91/Kconfig +@@ -0,0 +1,24 @@ ++menu "SoC Audio for the Atmel AT91" ++ ++config SND_AT91_SOC ++ tristate "SoC Audio for the Atmel AT91 System-on-Chip" ++ depends on ARCH_AT91 && SND ++ select SND_PCM ++ help ++ Say Y or M if you want to add support for codecs attached to ++ the AT91 SSC interface. You will also need ++ to select the audio interfaces to support below. ++ ++config SND_AT91_SOC_I2S ++ tristate ++ ++config SND_AT91_SOC_ETI_B1_WM8731 ++ tristate "SoC I2S Audio support for Endrelia ETI-B1 board" ++ depends on SND_AT91_SOC && MACH_ETI_B1 ++ select SND_AT91_SOC_I2S ++ select SND_SOC_WM8731 ++ help ++ Say Y if you want to add support for SoC audio on Endrelia ++ ETI-B1 board. ++ ++endmenu +Index: linux-2.6-pxa-new/sound/soc/at91/Makefile +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/sound/soc/at91/Makefile +@@ -0,0 +1,11 @@ ++# AT91 Platform Support ++snd-soc-at91-objs := at91rm9200-pcm.o ++snd-soc-at91-i2s-objs := at91rm9200-i2s.o ++ ++obj-$(CONFIG_SND_AT91_SOC) += snd-soc-at91.o ++obj-$(CONFIG_SND_AT91_SOC_I2S) += snd-soc-at91-i2s.o ++ ++# AT91 Machine Support ++snd-soc-eti-b1-wm8731-objs := eti_b1_wm8731.o ++ ++obj-$(CONFIG_SND_AT91_SOC_ETI_B1_WM8731) += snd-soc-eti-b1-wm8731.o +Index: linux-2.6-pxa-new/sound/soc/at91/at91rm9200-i2s.c +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/sound/soc/at91/at91rm9200-i2s.c +@@ -0,0 +1,715 @@ ++/* ++ * at91rm9200-i2s.c -- ALSA Soc Audio Layer Platform driver and DMA engine ++ * ++ * Author: Frank Mandarino <fmandarino@endrelia.com> ++ * Endrelia Technologies Inc. ++ * ++ * Based on pxa2xx Platform drivers by ++ * Liam Girdwood <liam.girdwood@wolfsonmicro.com> ++ * ++ * This program is free software; you can redistribute it and/or modify it ++ * under the terms of the GNU General Public License as published by the ++ * Free Software Foundation; either version 2 of the License, or (at your ++ * option) any later version. ++ * ++ * Revision history ++ * 3rd Mar 2006 Initial version. ++ */ ++ ++#include <linux/init.h> ++#include <linux/module.h> ++#include <linux/interrupt.h> ++#include <linux/device.h> ++#include <linux/delay.h> ++#include <linux/clk.h> ++#include <sound/driver.h> ++#include <sound/core.h> ++#include <sound/pcm.h> ++#include <sound/initval.h> ++#include <sound/soc.h> ++ ++#include <asm/arch/at91rm9200.h> ++#include <asm/arch/at91rm9200_ssc.h> ++#include <asm/arch/at91rm9200_pdc.h> ++#include <asm/arch/hardware.h> ++ ++#include "at91rm9200-pcm.h" ++ ++#if 0 ++#define DBG(x...) printk(KERN_DEBUG "at91rm9200-i2s:" x) ++#else ++#define DBG(x...) ++#endif ++ ++#define AT91RM9200_I2S_DAIFMT \ ++ (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS | SND_SOC_DAIFMT_NB_NF) ++ ++#define AT91RM9200_I2S_DIR \ ++ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE) ++ ++/* priv is (SSC_CMR.DIV << 16 | SSC_TCMR.PERIOD ) */ ++static struct snd_soc_dai_mode at91rm9200_i2s[] = { ++ ++ /* 8k: BCLK = (MCLK/10) = (60MHz/50) = 1.2MHz */ ++ { ++ .fmt = AT91RM9200_I2S_DAIFMT, ++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, ++ .pcmrate = SNDRV_PCM_RATE_8000, ++ .pcmdir = AT91RM9200_I2S_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 1500, ++ .bfs = SND_SOC_FSBD(10), ++ .priv = (25 << 16 | 74), ++ }, ++ ++ /* 16k: BCLK = (MCLK/3) ~= (60MHz/14) = 4.285714MHz */ ++ { ++ .fmt = AT91RM9200_I2S_DAIFMT, ++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, ++ .pcmrate = SNDRV_PCM_RATE_16000, ++ .pcmdir = AT91RM9200_I2S_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 750, ++ .bfs = SND_SOC_FSBD(3), ++ .priv = (7 << 16 | 133), ++ }, ++ ++ /* 32k: BCLK = (MCLK/3) ~= (60MHz/14) = 4.285714MHz */ ++ { ++ .fmt = AT91RM9200_I2S_DAIFMT, ++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, ++ .pcmrate = SNDRV_PCM_RATE_32000, ++ .pcmdir = AT91RM9200_I2S_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 375, ++ .bfs = SND_SOC_FSBD(3), ++ .priv = (7 << 16 | 66), ++ }, ++ ++ /* 48k: BCLK = (MCLK/5) ~= (60MHz/26) = 2.3076923MHz */ ++ { ++ .fmt = AT91RM9200_I2S_DAIFMT, ++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, ++ .pcmrate = SNDRV_PCM_RATE_48000, ++ .pcmdir = AT91RM9200_I2S_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 250, ++ .bfs SND_SOC_FSBD(5), ++ .priv = (13 << 16 | 23), ++ }, ++}; ++ ++ ++/* ++ * SSC registers required by the PCM DMA engine. ++ */ ++static struct at91rm9200_ssc_regs ssc_reg[3] = { ++ { ++ .cr = (void __iomem *) (AT91_VA_BASE_SSC0 + AT91_SSC_CR), ++ .ier = (void __iomem *) (AT91_VA_BASE_SSC0 + AT91_SSC_IER), ++ .idr = (void __iomem *) (AT91_VA_BASE_SSC0 + AT91_SSC_IDR), ++ }, ++ { ++ .cr = (void __iomem *) (AT91_VA_BASE_SSC1 + AT91_SSC_CR), ++ .ier = (void __iomem *) (AT91_VA_BASE_SSC1 + AT91_SSC_IER), ++ .idr = (void __iomem *) (AT91_VA_BASE_SSC1 + AT91_SSC_IDR), ++ }, ++ { ++ .cr = (void __iomem *) (AT91_VA_BASE_SSC2 + AT91_SSC_CR), ++ .ier = (void __iomem *) (AT91_VA_BASE_SSC2 + AT91_SSC_IER), ++ .idr = (void __iomem *) (AT91_VA_BASE_SSC2 + AT91_SSC_IDR), ++ }, ++}; ++ ++static struct at91rm9200_pdc_regs pdc_tx_reg[3] = { ++ { ++ .xpr = (void __iomem *) (AT91_VA_BASE_SSC0 + AT91_PDC_TPR), ++ .xcr = (void __iomem *) (AT91_VA_BASE_SSC0 + AT91_PDC_TCR), ++ .xnpr = (void __iomem *) (AT91_VA_BASE_SSC0 + AT91_PDC_TNPR), ++ .xncr = (void __iomem *) (AT91_VA_BASE_SSC0 + AT91_PDC_TNCR), ++ .ptcr = (void __iomem *) (AT91_VA_BASE_SSC0 + AT91_PDC_PTCR), ++ }, ++ { ++ .xpr = (void __iomem *) (AT91_VA_BASE_SSC1 + AT91_PDC_TPR), ++ .xcr = (void __iomem *) (AT91_VA_BASE_SSC1 + AT91_PDC_TCR), ++ .xnpr = (void __iomem *) (AT91_VA_BASE_SSC1 + AT91_PDC_TNPR), ++ .xncr = (void __iomem *) (AT91_VA_BASE_SSC1 + AT91_PDC_TNCR), ++ .ptcr = (void __iomem *) (AT91_VA_BASE_SSC1 + AT91_PDC_PTCR), ++ }, ++ { ++ .xpr = (void __iomem *) (AT91_VA_BASE_SSC2 + AT91_PDC_TPR), ++ .xcr = (void __iomem *) (AT91_VA_BASE_SSC2 + AT91_PDC_TCR), ++ .xnpr = (void __iomem *) (AT91_VA_BASE_SSC2 + AT91_PDC_TNPR), ++ .xncr = (void __iomem *) (AT91_VA_BASE_SSC2 + AT91_PDC_TNCR), ++ .ptcr = (void __iomem *) (AT91_VA_BASE_SSC2 + AT91_PDC_PTCR), ++ }, ++}; ++ ++static struct at91rm9200_pdc_regs pdc_rx_reg[3] = { ++ { ++ .xpr = (void __iomem *) (AT91_VA_BASE_SSC0 + AT91_PDC_RPR), ++ .xcr = (void __iomem *) (AT91_VA_BASE_SSC0 + AT91_PDC_RCR), ++ .xnpr = (void __iomem *) (AT91_VA_BASE_SSC0 + AT91_PDC_RNPR), ++ .xncr = (void __iomem *) (AT91_VA_BASE_SSC0 + AT91_PDC_RNCR), ++ .ptcr = (void __iomem *) (AT91_VA_BASE_SSC0 + AT91_PDC_PTCR), ++ }, ++ { ++ .xpr = (void __iomem *) (AT91_VA_BASE_SSC1 + AT91_PDC_RPR), ++ .xcr = (void __iomem *) (AT91_VA_BASE_SSC1 + AT91_PDC_RCR), ++ .xnpr = (void __iomem *) (AT91_VA_BASE_SSC1 + AT91_PDC_RNPR), ++ .xncr = (void __iomem *) (AT91_VA_BASE_SSC1 + AT91_PDC_RNCR), ++ .ptcr = (void __iomem *) (AT91_VA_BASE_SSC1 + AT91_PDC_PTCR), ++ }, ++ { ++ .xpr = (void __iomem *) (AT91_VA_BASE_SSC2 + AT91_PDC_RPR), ++ .xcr = (void __iomem *) (AT91_VA_BASE_SSC2 + AT91_PDC_RCR), ++ .xnpr = (void __iomem *) (AT91_VA_BASE_SSC2 + AT91_PDC_RNPR), ++ .xncr = (void __iomem *) (AT91_VA_BASE_SSC2 + AT91_PDC_RNCR), ++ .ptcr = (void __iomem *) (AT91_VA_BASE_SSC2 + AT91_PDC_PTCR), ++ }, ++}; ++ ++/* ++ * SSC & PDC status bits for transmit and receive. ++ */ ++static struct at91rm9200_ssc_mask ssc_tx_mask = { ++ .ssc_enable = AT91_SSC_TXEN, ++ .ssc_disable = AT91_SSC_TXDIS, ++ .ssc_endx = AT91_SSC_ENDTX, ++ .ssc_endbuf = AT91_SSC_TXBUFE, ++ .pdc_enable = AT91_PDC_TXTEN, ++ .pdc_disable = AT91_PDC_TXTDIS, ++}; ++ ++static struct at91rm9200_ssc_mask ssc_rx_mask = { ++ .ssc_enable = AT91_SSC_RXEN, ++ .ssc_disable = AT91_SSC_RXDIS, ++ .ssc_endx = AT91_SSC_ENDRX, ++ .ssc_endbuf = AT91_SSC_RXBUFF, ++ .pdc_enable = AT91_PDC_RXTEN, ++ .pdc_disable = AT91_PDC_RXTDIS, ++}; ++ ++/* ++ * A MUTEX is used to protect an SSC initialzed flag which allows ++ * the substream hw_params() call to initialize the SSC only if ++ * there are no other substreams open. If there are other ++ * substreams open, the hw_param() call can only check that ++ * it is using the same format and rate. ++ */ ++static DECLARE_MUTEX(ssc0_mutex); ++static DECLARE_MUTEX(ssc1_mutex); ++static DECLARE_MUTEX(ssc2_mutex); ++ ++/* ++ * DMA parameters. ++ */ ++static at91rm9200_pcm_dma_params_t ssc_dma_params[3][2] = { ++ {{ ++ .name = "SSC0/I2S PCM Stereo out", ++ .ssc = &ssc_reg[0], ++ .pdc = &pdc_tx_reg[0], ++ .mask = &ssc_tx_mask, ++ }, ++ { ++ .name = "SSC0/I2S PCM Stereo in", ++ .ssc = &ssc_reg[0], ++ .pdc = &pdc_rx_reg[0], ++ .mask = &ssc_rx_mask, ++ }}, ++ {{ ++ .name = "SSC1/I2S PCM Stereo out", ++ .ssc = &ssc_reg[1], ++ .pdc = &pdc_tx_reg[1], ++ .mask = &ssc_tx_mask, ++ }, ++ { ++ .name = "SSC1/I2S PCM Stereo in", ++ .ssc = &ssc_reg[1], ++ .pdc = &pdc_rx_reg[1], ++ .mask = &ssc_rx_mask, ++ }}, ++ {{ ++ .name = "SSC2/I2S PCM Stereo out", ++ .ssc = &ssc_reg[2], ++ .pdc = &pdc_tx_reg[2], ++ .mask = &ssc_tx_mask, ++ }, ++ { ++ .name = "SSC1/I2S PCM Stereo in", ++ .ssc = &ssc_reg[2], ++ .pdc = &pdc_rx_reg[2], ++ .mask = &ssc_rx_mask, ++ }}, ++}; ++ ++ ++struct at91rm9200_ssc_state { ++ u32 ssc_cmr; ++ u32 ssc_rcmr; ++ u32 ssc_rfmr; ++ u32 ssc_tcmr; ++ u32 ssc_tfmr; ++ u32 ssc_sr; ++ u32 ssc_imr; ++}; ++ ++static struct at91rm9200_ssc_info { ++ char *name; ++ void __iomem *ssc_base; ++ u32 pid; ++ spinlock_t lock; /* lock for dir_mask */ ++ int dir_mask; /* 0=unused, 1=playback, 2=capture */ ++ struct semaphore *mutex; ++ int initialized; ++ int pcmfmt; ++ int rate; ++ at91rm9200_pcm_dma_params_t *dma_params[2]; ++ struct at91rm9200_ssc_state ssc_state; ++ ++} ssc_info[3] = { ++ { ++ .name = "ssc0", ++ .ssc_base = (void __iomem *) AT91_VA_BASE_SSC0, ++ .pid = AT91_ID_SSC0, ++ .lock = SPIN_LOCK_UNLOCKED, ++ .dir_mask = 0, ++ .mutex = &ssc0_mutex, ++ .initialized = 0, ++ }, ++ { ++ .name = "ssc1", ++ .ssc_base = (void __iomem *) AT91_VA_BASE_SSC1, ++ .pid = AT91_ID_SSC1, ++ .lock = SPIN_LOCK_UNLOCKED, ++ .dir_mask = 0, ++ .mutex = &ssc1_mutex, ++ .initialized = 0, ++ }, ++ { ++ .name = "ssc2", ++ .ssc_base = (void __iomem *) AT91_VA_BASE_SSC2, ++ .pid = AT91_ID_SSC2, ++ .lock = SPIN_LOCK_UNLOCKED, ++ .dir_mask = 0, ++ .mutex = &ssc2_mutex, ++ .initialized = 0, ++ }, ++}; ++ ++ ++static irqreturn_t at91rm9200_i2s_interrupt(int irq, void *dev_id) ++{ ++ struct at91rm9200_ssc_info *ssc_p = dev_id; ++ at91rm9200_pcm_dma_params_t *dma_params; ++ u32 ssc_sr; ++ int i; ++ ++ ssc_sr = at91_ssc_read(ssc_p->ssc_base + AT91_SSC_SR) ++ & at91_ssc_read(ssc_p->ssc_base + AT91_SSC_IMR); ++ ++ /* ++ * Loop through the substreams attached to this SSC. If ++ * a DMA-related interrupt occurred on that substream, call ++ * the DMA interrupt handler function, if one has been ++ * registered in the dma_params structure by the PCM driver. ++ */ ++ for (i = 0; i < ARRAY_SIZE(ssc_p->dma_params); i++) { ++ dma_params = ssc_p->dma_params[i]; ++ ++ if (dma_params != NULL && dma_params->dma_intr_handler != NULL && ++ (ssc_sr & ++ (dma_params->mask->ssc_endx | dma_params->mask->ssc_endbuf))) ++ ++ dma_params->dma_intr_handler(ssc_sr, dma_params->substream); ++ } ++ ++ return IRQ_HANDLED; ++} ++ ++static int at91rm9200_i2s_startup(struct snd_pcm_substream *substream) ++{ ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ struct at91rm9200_ssc_info *ssc_p = &ssc_info[rtd->cpu_dai->id]; ++ int dir_mask; ++ ++ DBG("i2s_startup: SSC_SR=0x%08lx\n", ++ at91_ssc_read(ssc_p->ssc_base + AT91_SSC_SR)); ++ dir_mask = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0x1 : 0x2; ++ ++ spin_lock_irq(&ssc_p->lock); ++ if (ssc_p->dir_mask & dir_mask) { ++ spin_unlock_irq(&ssc_p->lock); ++ return -EBUSY; ++ } ++ ssc_p->dir_mask |= dir_mask; ++ spin_unlock_irq(&ssc_p->lock); ++ ++ return 0; ++} ++ ++static void at91rm9200_i2s_shutdown(struct snd_pcm_substream *substream) ++{ ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ struct at91rm9200_ssc_info *ssc_p = &ssc_info[rtd->cpu_dai->id]; ++ at91rm9200_pcm_dma_params_t *dma_params = rtd->cpu_dai->dma_data; ++ int dir, dir_mask; ++ ++ dir = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1; ++ ++ if (dma_params != NULL) { ++ at91_ssc_write(dma_params->ssc->cr, dma_params->mask->ssc_disable); ++ DBG("%s disabled SSC_SR=0x%08lx\n", (dir ? "receive" : "transmit"), ++ at91_ssc_read(ssc_p->ssc_base + AT91_SSC_SR)); ++ ++ dma_params->substream = NULL; ++ ssc_p->dma_params[dir] = NULL; ++ } ++ ++ dir_mask = 1 << dir; ++ ++ spin_lock_irq(&ssc_p->lock); ++ ssc_p->dir_mask &= ~dir_mask; ++ if (!ssc_p->dir_mask) { ++ /* Shutdown the SSC clock. */ ++ DBG("Stopping pid %d clock\n", ssc_p->pid); ++ at91_sys_write(AT91_PMC_PCDR, 1<<ssc_p->pid); ++ ++ if (ssc_p->initialized) ++ free_irq(ssc_p->pid, ssc_p); ++ ++ /* Reset the SSC */ ++ at91_ssc_write(ssc_p->ssc_base + AT91_SSC_CR, AT91_SSC_SWRST); ++ ++ /* Force a re-init on the next hw_params() call. */ ++ ssc_p->initialized = 0; ++ } ++ spin_unlock_irq(&ssc_p->lock); ++} ++ ++#ifdef CONFIG_PM ++static int at91rm9200_i2s_suspend(struct platform_device *pdev, ++ struct snd_soc_cpu_dai *dai) ++{ ++ struct at91rm9200_ssc_info *ssc_p; ++ ++ if(!dai->active) ++ return 0; ++ ++ ssc_p = &ssc_info[dai->id]; ++ ++ /* Save the status register before disabling transmit and receive. */ ++ ssc_p->state->ssc_sr = at91_ssc_read(ssc_p->ssc_base + AT91_SSC_SR); ++ at91_ssc_write(ssc_p->ssc_base + ++ AT91_SSC_CR, AT91_SSC_TXDIS | AT91_SSC_RXDIS); ++ ++ /* Save the current interrupt mask, then disable unmasked interrupts. */ ++ ssc_p->state->ssc_imr = at91_ssc_read(ssc_p->ssc_base + AT91_SSC_IMR); ++ at91_ssc_write(ssc_p->ssc_base + AT91_SSC_IDR, ssc_p->state->ssc_imr); ++ ++ ssc_p->state->ssc_cmr = at91_ssc_read(ssc_p->ssc_base + AT91_SSC_CMR); ++ ssc_p->state->ssc_rcmr = at91_ssc_read(ssc_p->ssc_base + AT91_SSC_RCMR); ++ ssc_p->state->ssc_rfmr = at91_ssc_read(ssc_p->ssc_base + AT91_SSC_RCMR); ++ ssc_p->state->ssc_tcmr = at91_ssc_read(ssc_p->ssc_base + AT91_SSC_RCMR); ++ ssc_p->state->ssc_tfmr = at91_ssc_read(ssc_p->ssc_base + AT91_SSC_RCMR); ++ ++ return 0; ++} ++ ++static int at91rm9200_i2s_resume(struct platform_device *pdev, ++ struct snd_soc_cpu_dai *dai) ++{ ++ struct at91rm9200_ssc_info *ssc_p; ++ u32 cr_mask; ++ ++ if(!dai->active) ++ return 0; ++ ++ ssc_p = &ssc_info[dai->id]; ++ ++ at91_ssc_write(ssc_p->ssc_base + AT91_SSC_RCMR, ssc_p->state->ssc_tfmr); ++ at91_ssc_write(ssc_p->ssc_base + AT91_SSC_RCMR, ssc_p->state->ssc_tcmr); ++ at91_ssc_write(ssc_p->ssc_base + AT91_SSC_RCMR, ssc_p->state->ssc_rfmr); ++ at91_ssc_write(ssc_p->ssc_base + AT91_SSC_RCMR, ssc_p->state->ssc_rcmr); ++ at91_ssc_write(ssc_p->ssc_base + AT91_SSC_CMR, ssc_p->state->ssc_cmr); ++ ++ at91_ssc_write(ssc_p->ssc_base + AT91_SSC_IER, ssc_p->state->ssc_imr); ++ ++ at91_ssc_write(ssc_p->ssc_base + AT91_SSC_CR, ++ ((ssc_p->state->ssc_sr & AT91_SSC_RXENA) ? AT91_SSC_RXEN : 0) | ++ ((ssc_p->state->ssc_sr & AT91_SSC_TXENA) ? AT91_SSC_TXEN : 0)); ++ ++ return 0; ++} ++ ++#else ++#define at91rm9200_i2s_suspend NULL ++#define at91rm9200_i2s_resume NULL ++#endif ++ ++static unsigned int at91rm9200_i2s_config_sysclk( ++ struct snd_soc_cpu_dai *iface, struct snd_soc_clock_info *info, ++ unsigned int clk) ++{ ++ /* Currently, there is only support for USB (12Mhz) mode */ ++ if (clk != 12000000) ++ return 0; ++ return 12000000; ++} ++ ++static int at91rm9200_i2s_hw_params(struct snd_pcm_substream *substream, ++ struct snd_pcm_hw_params *params) ++{ ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ int id = rtd->cpu_dai->id; ++ struct at91rm9200_ssc_info *ssc_p = &ssc_info[id]; ++ at91rm9200_pcm_dma_params_t *dma_params; ++ unsigned int pcmfmt, rate; ++ int dir, channels, bits; ++ struct clk *mck_clk; ++ unsigned long bclk; ++ u32 div, period, tfmr, rfmr, tcmr, rcmr; ++ int ret; ++ ++ /* ++ * Currently, there is only one set of dma params for ++ * each direction. If more are added, this code will ++ * have to be changed to select the proper set. ++ */ ++ dir = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1; ++ ++ dma_params = &ssc_dma_params[id][dir]; ++ dma_params->substream = substream; ++ ++ ssc_p->dma_params[dir] = dma_params; ++ rtd->cpu_dai->dma_data = dma_params; ++ ++ rate = params_rate(params); ++ channels = params_channels(params); ++ ++ pcmfmt = rtd->cpu_dai->dai_runtime.pcmfmt; ++ switch (pcmfmt) { ++ case SNDRV_PCM_FMTBIT_S16_LE: ++ /* likely this is all we'll ever support, but ... */ ++ bits = 16; ++ dma_params->pdc_xfer_size = 2; ++ break; ++ default: ++ printk(KERN_WARNING "at91rm9200-i2s: unsupported format %x\n", ++ pcmfmt); ++ return -EINVAL; ++ } ++ ++ /* Don't allow both SSC substreams to initialize at the same time. */ ++ down(ssc_p->mutex); ++ ++ /* ++ * If this SSC is alreadly initialized, then this substream must use ++ * the same format and rate. ++ */ ++ if (ssc_p->initialized) { ++ if (pcmfmt != ssc_p->pcmfmt || rate != ssc_p->rate) { ++ printk(KERN_WARNING "at91rm9200-i2s: " ++ "incompatible substream in other direction\n"); ++ up(ssc_p->mutex); ++ return -EINVAL; ++ } ++ } else { ++ /* Enable PMC peripheral clock for this SSC */ ++ DBG("Starting pid %d clock\n", ssc_p->pid); ++ at91_sys_write(AT91_PMC_PCER, 1<<ssc_p->pid); ++ ++ /* Reset the SSC */ ++ at91_ssc_write(ssc_p->ssc_base + AT91_SSC_CR, AT91_SSC_SWRST); ++ ++ at91_ssc_write(ssc_p->ssc_base + AT91_PDC_RPR, 0); ++ at91_ssc_write(ssc_p->ssc_base + AT91_PDC_RCR, 0); ++ at91_ssc_write(ssc_p->ssc_base + AT91_PDC_RNPR, 0); ++ at91_ssc_write(ssc_p->ssc_base + AT91_PDC_RNCR, 0); ++ at91_ssc_write(ssc_p->ssc_base + AT91_PDC_TPR, 0); ++ at91_ssc_write(ssc_p->ssc_base + AT91_PDC_TCR, 0); ++ at91_ssc_write(ssc_p->ssc_base + AT91_PDC_TNPR, 0); ++ at91_ssc_write(ssc_p->ssc_base + AT91_PDC_TNCR, 0); ++ ++ mck_clk = clk_get(NULL, "mck"); ++ ++ div = rtd->cpu_dai->dai_runtime.priv >> 16; ++ period = rtd->cpu_dai->dai_runtime.priv & 0xffff; ++ bclk = 60000000 / (2 * div); ++ ++ DBG("mck %ld fsbd %d bfs %d bfs_real %d bclk %ld div %d period %d\n", ++ clk_get_rate(mck_clk), ++ SND_SOC_FSBD(6), ++ rtd->cpu_dai->dai_runtime.bfs, ++ SND_SOC_FSBD_REAL(rtd->cpu_dai->dai_runtime.bfs), ++ bclk, ++ div, ++ period); ++ ++ clk_put(mck_clk); ++ ++ at91_ssc_write(ssc_p->ssc_base + AT91_SSC_CMR, div); ++ ++ /* ++ * Setup the TFMR and RFMR for the proper data format. ++ */ ++ tfmr = ++ (( AT91_SSC_FSEDGE_POSITIVE ) & AT91_SSC_FSEDGE) ++ | (( 0 << 23) & AT91_SSC_FSDEN) ++ | (( AT91_SSC_FSOS_NEGATIVE ) & AT91_SSC_FSOS) ++ | (((bits - 1) << 16) & AT91_SSC_FSLEN) ++ | (((channels - 1) << 8) & AT91_SSC_DATNB) ++ | (( 1 << 7) & AT91_SSC_MSBF) ++ | (( 0 << 5) & AT91_SSC_DATDEF) ++ | (((bits - 1) << 0) & AT91_SSC_DATALEN); ++ DBG("SSC_TFMR=0x%08x\n", tfmr); ++ at91_ssc_write(ssc_p->ssc_base + AT91_SSC_TFMR, tfmr); ++ ++ rfmr = ++ (( AT91_SSC_FSEDGE_POSITIVE ) & AT91_SSC_FSEDGE) ++ | (( AT91_SSC_FSOS_NONE ) & AT91_SSC_FSOS) ++ | (( 0 << 16) & AT91_SSC_FSLEN) ++ | (((channels - 1) << 8) & AT91_SSC_DATNB) ++ | (( 1 << 7) & AT91_SSC_MSBF) ++ | (( 0 << 5) & AT91_SSC_LOOP) ++ | (((bits - 1) << 0) & AT91_SSC_DATALEN); ++ ++ DBG("SSC_RFMR=0x%08x\n", rfmr); ++ at91_ssc_write(ssc_p->ssc_base + AT91_SSC_RFMR, rfmr); ++ ++ /* ++ * Setup the TCMR and RCMR to generate the proper BCLK ++ * and LRC signals. ++ */ ++ tcmr = ++ (( period << 24) & AT91_SSC_PERIOD) ++ | (( 1 << 16) & AT91_SSC_STTDLY) ++ | (( AT91_SSC_START_FALLING_RF ) & AT91_SSC_START) ++ | (( AT91_SSC_CKI_FALLING ) & AT91_SSC_CKI) ++ | (( AT91_SSC_CKO_CONTINUOUS ) & AT91_SSC_CKO) ++ | (( AT91_SSC_CKS_DIV ) & AT91_SSC_CKS); ++ ++ DBG("SSC_TCMR=0x%08x\n", tcmr); ++ at91_ssc_write(ssc_p->ssc_base + AT91_SSC_TCMR, tcmr); ++ ++ rcmr = ++ (( 0 << 24) & AT91_SSC_PERIOD) ++ | (( 1 << 16) & AT91_SSC_STTDLY) ++ | (( AT91_SSC_START_TX_RX ) & AT91_SSC_START) ++ | (( AT91_SSC_CK_RISING ) & AT91_SSC_CKI) ++ | (( AT91_SSC_CKO_NONE ) & AT91_SSC_CKO) ++ | (( AT91_SSC_CKS_CLOCK ) & AT91_SSC_CKS); ++ ++ DBG("SSC_RCMR=0x%08x\n", rcmr); ++ at91_ssc_write(ssc_p->ssc_base + AT91_SSC_RCMR, rcmr); ++ ++ if ((ret = request_irq(ssc_p->pid, at91rm9200_i2s_interrupt, ++ 0, ssc_p->name, ssc_p)) < 0) { ++ printk(KERN_WARNING "at91rm9200-i2s: request_irq failure\n"); ++ return ret; ++ } ++ ++ /* ++ * Save the current substream parameters in order to check ++ * that the substream in the opposite direction uses the ++ * same parameters. ++ */ ++ ssc_p->pcmfmt = pcmfmt; ++ ssc_p->rate = rate; ++ ssc_p->initialized = 1; ++ ++ DBG("hw_params: SSC initialized\n"); ++ } ++ ++ up(ssc_p->mutex); ++ ++ return 0; ++} ++ ++ ++static int at91rm9200_i2s_prepare(struct snd_pcm_substream *substream) ++{ ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ at91rm9200_pcm_dma_params_t *dma_params = rtd->cpu_dai->dma_data; ++ ++ at91_ssc_write(dma_params->ssc->cr, dma_params->mask->ssc_enable); ++ ++ DBG("%s enabled SSC_SR=0x%08lx\n", ++ substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? "transmit" : "receive", ++ at91_ssc_read(ssc_info[rtd->cpu_dai->id].ssc_base + AT91_SSC_SR)); ++ return 0; ++} ++ ++ ++struct snd_soc_cpu_dai at91rm9200_i2s_dai[] = { ++ { .name = "at91rm9200-ssc0/i2s", ++ .id = 0, ++ .type = SND_SOC_DAI_I2S, ++ .suspend = at91rm9200_i2s_suspend, ++ .resume = at91rm9200_i2s_resume, ++ .config_sysclk = at91rm9200_i2s_config_sysclk, ++ .playback = { ++ .channels_min = 1, ++ .channels_max = 2,}, ++ .capture = { ++ .channels_min = 1, ++ .channels_max = 2,}, ++ .ops = { ++ .startup = at91rm9200_i2s_startup, ++ .shutdown = at91rm9200_i2s_shutdown, ++ .prepare = at91rm9200_i2s_prepare, ++ .hw_params = at91rm9200_i2s_hw_params,}, ++ .caps = { ++ .mode = &at91rm9200_i2s[0], ++ .num_modes = ARRAY_SIZE(at91rm9200_i2s),}, ++ }, ++ { .name = "at91rm9200-ssc1/i2s", ++ .id = 1, ++ .type = SND_SOC_DAI_I2S, ++ .suspend = at91rm9200_i2s_suspend, ++ .resume = at91rm9200_i2s_resume, ++ .config_sysclk = at91rm9200_i2s_config_sysclk, ++ .playback = { ++ .channels_min = 1, ++ .channels_max = 2,}, ++ .capture = { ++ .channels_min = 1, ++ .channels_max = 2,}, ++ .ops = { ++ .startup = at91rm9200_i2s_startup, ++ .shutdown = at91rm9200_i2s_shutdown, ++ .prepare = at91rm9200_i2s_prepare, ++ .hw_params = at91rm9200_i2s_hw_params,}, ++ .caps = { ++ .mode = &at91rm9200_i2s[0], ++ .num_modes = ARRAY_SIZE(at91rm9200_i2s),}, ++ }, ++ { .name = "at91rm9200-ssc2/i2s", ++ .id = 2, ++ .type = SND_SOC_DAI_I2S, ++ .suspend = at91rm9200_i2s_suspend, ++ .resume = at91rm9200_i2s_resume, ++ .config_sysclk = at91rm9200_i2s_config_sysclk, ++ .playback = { ++ .channels_min = 1, ++ .channels_max = 2,}, ++ .capture = { ++ .channels_min = 1, ++ .channels_max = 2,}, ++ .ops = { ++ .startup = at91rm9200_i2s_startup, ++ .shutdown = at91rm9200_i2s_shutdown, ++ .prepare = at91rm9200_i2s_prepare, ++ .hw_params = at91rm9200_i2s_hw_params,}, ++ .caps = { ++ .mode = &at91rm9200_i2s[0], ++ .num_modes = ARRAY_SIZE(at91rm9200_i2s),}, ++ }, ++}; ++ ++EXPORT_SYMBOL_GPL(at91rm9200_i2s_dai); ++ ++/* Module information */ ++MODULE_AUTHOR("Frank Mandarino, fmandarino@endrelia.com, www.endrelia.com"); ++MODULE_DESCRIPTION("AT91RM9200 I2S ASoC Interface"); ++MODULE_LICENSE("GPL"); +Index: linux-2.6-pxa-new/sound/soc/at91/at91rm9200-pcm.c +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/sound/soc/at91/at91rm9200-pcm.c +@@ -0,0 +1,428 @@ ++/* ++ * at91rm9200-pcm.c -- ALSA PCM interface for the Atmel AT91RM9200 chip. ++ * ++ * Author: Frank Mandarino <fmandarino@endrelia.com> ++ * Endrelia Technologies Inc. ++ * Created: Mar 3, 2006 ++ * ++ * Based on pxa2xx-pcm.c by: ++ * ++ * Author: Nicolas Pitre ++ * Created: Nov 30, 2004 ++ * Copyright: (C) 2004 MontaVista Software, Inc. ++ * ++ * This program is free software; you can redistribute it and/or modify ++ * it under the terms of the GNU General Public License version 2 as ++ * published by the Free Software Foundation. ++ */ ++ ++#include <linux/module.h> ++#include <linux/init.h> ++#include <linux/platform_device.h> ++#include <linux/slab.h> ++#include <linux/dma-mapping.h> ++ ++#include <sound/driver.h> ++#include <sound/core.h> ++#include <sound/pcm.h> ++#include <sound/pcm_params.h> ++#include <sound/soc.h> ++ ++#include <asm/arch/at91rm9200.h> ++#include <asm/arch/at91rm9200_ssc.h> ++#include <asm/arch/at91rm9200_pdc.h> ++#include <asm/arch/hardware.h> ++ ++#include "at91rm9200-pcm.h" ++ ++#if 0 ++#define DBG(x...) printk(KERN_INFO "at91rm9200-pcm: " x) ++#else ++#define DBG(x...) ++#endif ++ ++static const snd_pcm_hardware_t at91rm9200_pcm_hardware = { ++ .info = SNDRV_PCM_INFO_MMAP | ++ SNDRV_PCM_INFO_MMAP_VALID | ++ SNDRV_PCM_INFO_INTERLEAVED | ++ SNDRV_PCM_INFO_PAUSE, ++ .formats = SNDRV_PCM_FMTBIT_S16_LE, ++ .period_bytes_min = 32, ++ .period_bytes_max = 8192, ++ .periods_min = 2, ++ .periods_max = 1024, ++ .buffer_bytes_max = 32 * 1024, ++}; ++ ++struct at91rm9200_runtime_data { ++ at91rm9200_pcm_dma_params_t *params; ++ dma_addr_t dma_buffer; /* physical address of dma buffer */ ++ dma_addr_t dma_buffer_end; /* first address beyond DMA buffer */ ++ size_t period_size; ++ dma_addr_t period_ptr; /* physical address of next period */ ++ u32 pdc_xpr_save; /* PDC register save */ ++ u32 pdc_xcr_save; ++ u32 pdc_xnpr_save; ++ u32 pdc_xncr_save; ++}; ++ ++static void at91rm9200_pcm_dma_irq(u32 ssc_sr, ++ struct snd_pcm_substream *substream) ++{ ++ struct at91rm9200_runtime_data *prtd = substream->runtime->private_data; ++ at91rm9200_pcm_dma_params_t *params = prtd->params; ++ static int count = 0; ++ ++ count++; ++ ++ if (ssc_sr & params->mask->ssc_endbuf) { ++ ++ printk(KERN_WARNING ++ "at91rm9200-pcm: buffer %s on %s (SSC_SR=%#x, count=%d)\n", ++ substream->stream == SNDRV_PCM_STREAM_PLAYBACK ++ ? "underrun" : "overrun", ++ params->name, ssc_sr, count); ++ ++ /* re-start the PDC */ ++ at91_ssc_write(params->pdc->ptcr, params->mask->pdc_disable); ++ ++ prtd->period_ptr += prtd->period_size; ++ if (prtd->period_ptr >= prtd->dma_buffer_end) { ++ prtd->period_ptr = prtd->dma_buffer; ++ } ++ ++ at91_ssc_write(params->pdc->xpr, prtd->period_ptr); ++ at91_ssc_write(params->pdc->xcr, ++ prtd->period_size / params->pdc_xfer_size); ++ ++ at91_ssc_write(params->pdc->ptcr, params->mask->pdc_enable); ++ } ++ ++ if (ssc_sr & params->mask->ssc_endx) { ++ ++ /* Load the PDC next pointer and counter registers */ ++ prtd->period_ptr += prtd->period_size; ++ if (prtd->period_ptr >= prtd->dma_buffer_end) { ++ prtd->period_ptr = prtd->dma_buffer; ++ } ++ at91_ssc_write(params->pdc->xnpr, prtd->period_ptr); ++ at91_ssc_write(params->pdc->xncr, ++ prtd->period_size / params->pdc_xfer_size); ++ } ++ ++ snd_pcm_period_elapsed(substream); ++} ++ ++static int at91rm9200_pcm_hw_params(struct snd_pcm_substream *substream, ++ struct snd_pcm_hw_params *params) ++{ ++ snd_pcm_runtime_t *runtime = substream->runtime; ++ struct at91rm9200_runtime_data *prtd = runtime->private_data; ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ ++ /* this may get called several times by oss emulation ++ * with different params */ ++ ++ snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); ++ runtime->dma_bytes = params_buffer_bytes(params); ++ ++ prtd->params = rtd->cpu_dai->dma_data; ++ prtd->params->dma_intr_handler = at91rm9200_pcm_dma_irq; ++ ++ prtd->dma_buffer = runtime->dma_addr; ++ prtd->dma_buffer_end = runtime->dma_addr + runtime->dma_bytes; ++ prtd->period_size = params_period_bytes(params); ++ ++ DBG("hw_params: DMA for %s initialized (dma_bytes=%d, period_size=%d)\n", ++ prtd->params->name, runtime->dma_bytes, prtd->period_size); ++ return 0; ++} ++ ++static int at91rm9200_pcm_hw_free(struct snd_pcm_substream *substream) ++{ ++ struct at91rm9200_runtime_data *prtd = substream->runtime->private_data; ++ at91rm9200_pcm_dma_params_t *params = prtd->params; ++ ++ if (params != NULL) { ++ at91_ssc_write(params->pdc->ptcr, params->mask->pdc_disable); ++ prtd->params->dma_intr_handler = NULL; ++ } ++ ++ return 0; ++} ++ ++static int at91rm9200_pcm_prepare(struct snd_pcm_substream *substream) ++{ ++ struct at91rm9200_runtime_data *prtd = substream->runtime->private_data; ++ at91rm9200_pcm_dma_params_t *params = prtd->params; ++ ++ at91_ssc_write(params->ssc->idr, ++ params->mask->ssc_endx | params->mask->ssc_endbuf); ++ ++ at91_ssc_write(params->pdc->ptcr, params->mask->pdc_disable); ++ return 0; ++} ++ ++static int at91rm9200_pcm_trigger(struct snd_pcm_substream *substream, ++ int cmd) ++{ ++ struct at91rm9200_runtime_data *prtd = substream->runtime->private_data; ++ at91rm9200_pcm_dma_params_t *params = prtd->params; ++ int ret = 0; ++ ++ switch (cmd) { ++ case SNDRV_PCM_TRIGGER_START: ++ prtd->period_ptr = prtd->dma_buffer; ++ ++ at91_ssc_write(params->pdc->xpr, prtd->period_ptr); ++ at91_ssc_write(params->pdc->xcr, ++ prtd->period_size / params->pdc_xfer_size); ++ ++ prtd->period_ptr += prtd->period_size; ++ at91_ssc_write(params->pdc->xnpr, prtd->period_ptr); ++ at91_ssc_write(params->pdc->xncr, ++ prtd->period_size / params->pdc_xfer_size); ++ ++ DBG("trigger: period_ptr=%lx, xpr=%lx, xcr=%ld, xnpr=%lx, xncr=%ld\n", ++ (unsigned long) prtd->period_ptr, ++ at91_ssc_read(params->pdc->xpr), ++ at91_ssc_read(params->pdc->xcr), ++ at91_ssc_read(params->pdc->xnpr), ++ at91_ssc_read(params->pdc->xncr)); ++ ++ at91_ssc_write(params->ssc->ier, ++ params->mask->ssc_endx | params->mask->ssc_endbuf); ++ ++ at91_ssc_write(params->pdc->ptcr, params->mask->pdc_enable); ++ ++ DBG("sr=%lx imr=%lx\n", at91_ssc_read(params->ssc->ier - 4), ++ at91_ssc_read(params->ssc->ier + 8)); ++ break; ++ ++ case SNDRV_PCM_TRIGGER_STOP: ++ case SNDRV_PCM_TRIGGER_SUSPEND: ++ case SNDRV_PCM_TRIGGER_PAUSE_PUSH: ++ at91_ssc_write(params->pdc->ptcr, params->mask->pdc_disable); ++ break; ++ ++ case SNDRV_PCM_TRIGGER_RESUME: ++ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: ++ at91_ssc_write(params->pdc->ptcr, params->mask->pdc_enable); ++ break; ++ ++ default: ++ ret = -EINVAL; ++ } ++ ++ return ret; ++} ++ ++static snd_pcm_uframes_t at91rm9200_pcm_pointer( ++ struct snd_pcm_substream *substream) ++{ ++ struct snd_pcm_runtime *runtime = substream->runtime; ++ struct at91rm9200_runtime_data *prtd = runtime->private_data; ++ at91rm9200_pcm_dma_params_t *params = prtd->params; ++ dma_addr_t ptr; ++ snd_pcm_uframes_t x; ++ ++ ptr = (dma_addr_t) at91_ssc_read(params->pdc->xpr); ++ x = bytes_to_frames(runtime, ptr - prtd->dma_buffer); ++ ++ if (x == runtime->buffer_size) ++ x = 0; ++ return x; ++} ++ ++static int at91rm9200_pcm_open(struct snd_pcm_substream *substream) ++{ ++ struct snd_pcm_runtime *runtime = substream->runtime; ++ struct at91rm9200_runtime_data *prtd; ++ int ret = 0; ++ ++ snd_soc_set_runtime_hwparams(substream, &at91rm9200_pcm_hardware); ++ ++ /* ensure that buffer size is a multiple of period size */ ++ ret = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); ++ if (ret < 0) ++ goto out; ++ ++ prtd = kzalloc(sizeof(struct at91rm9200_runtime_data), GFP_KERNEL); ++ if (prtd == NULL) { ++ ret = -ENOMEM; ++ goto out; ++ } ++ runtime->private_data = prtd; ++ ++ out: ++ return ret; ++} ++ ++static int at91rm9200_pcm_close(struct snd_pcm_substream *substream) ++{ ++ struct at91rm9200_runtime_data *prtd = substream->runtime->private_data; ++ ++ kfree(prtd); ++ return 0; ++} ++ ++static int at91rm9200_pcm_mmap(struct snd_pcm_substream *substream, ++ struct vm_area_struct *vma) ++{ ++ struct snd_pcm_runtime *runtime = substream->runtime; ++ ++ return dma_mmap_writecombine(substream->pcm->card->dev, vma, ++ runtime->dma_area, ++ runtime->dma_addr, ++ runtime->dma_bytes); ++} ++ ++struct snd_pcm_ops at91rm9200_pcm_ops = { ++ .open = at91rm9200_pcm_open, ++ .close = at91rm9200_pcm_close, ++ .ioctl = snd_pcm_lib_ioctl, ++ .hw_params = at91rm9200_pcm_hw_params, ++ .hw_free = at91rm9200_pcm_hw_free, ++ .prepare = at91rm9200_pcm_prepare, ++ .trigger = at91rm9200_pcm_trigger, ++ .pointer = at91rm9200_pcm_pointer, ++ .mmap = at91rm9200_pcm_mmap, ++}; ++ ++static int at91rm9200_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, ++ int stream) ++{ ++ struct snd_pcm_substream *substream = pcm->streams[stream].substream; ++ struct snd_dma_buffer *buf = &substream->dma_buffer; ++ size_t size = at91rm9200_pcm_hardware.buffer_bytes_max; ++ ++ buf->dev.type = SNDRV_DMA_TYPE_DEV; ++ buf->dev.dev = pcm->card->dev; ++ buf->private_data = NULL; ++ buf->area = dma_alloc_writecombine(pcm->card->dev, size, ++ &buf->addr, GFP_KERNEL); ++ ++ DBG("preallocate_dma_buffer: area=%p, addr=%p, size=%d\n", ++ (void *) buf->area, ++ (void *) buf->addr, ++ size); ++ ++ if (!buf->area) ++ return -ENOMEM; ++ ++ buf->bytes = size; ++ return 0; ++} ++ ++static u64 at91rm9200_pcm_dmamask = 0xffffffff; ++ ++static int at91rm9200_pcm_new(struct snd_card *card, ++ struct snd_soc_codec_dai *dai, struct snd_pcm *pcm) ++{ ++ int ret = 0; ++ ++ if (!card->dev->dma_mask) ++ card->dev->dma_mask = &at91rm9200_pcm_dmamask; ++ if (!card->dev->coherent_dma_mask) ++ card->dev->coherent_dma_mask = 0xffffffff; ++ ++ if (dai->playback.channels_min) { ++ ret = at91rm9200_pcm_preallocate_dma_buffer(pcm, ++ SNDRV_PCM_STREAM_PLAYBACK); ++ if (ret) ++ goto out; ++ } ++ ++ if (dai->capture.channels_min) { ++ ret = at91rm9200_pcm_preallocate_dma_buffer(pcm, ++ SNDRV_PCM_STREAM_CAPTURE); ++ if (ret) ++ goto out; ++ } ++ out: ++ return ret; ++} ++ ++static void at91rm9200_pcm_free_dma_buffers(struct snd_pcm *pcm) ++{ ++ struct snd_pcm_substream *substream; ++ struct snd_dma_buffer *buf; ++ int stream; ++ ++ for (stream = 0; stream < 2; stream++) { ++ substream = pcm->streams[stream].substream; ++ if (!substream) ++ continue; ++ ++ buf = &substream->dma_buffer; ++ if (!buf->area) ++ continue; ++ ++ dma_free_writecombine(pcm->card->dev, buf->bytes, ++ buf->area, buf->addr); ++ buf->area = NULL; ++ } ++} ++ ++static int at91rm9200_pcm_suspend(struct platform_device *pdev, ++ struct snd_soc_cpu_dai *dai) ++{ ++ struct snd_pcm_runtime *runtime = dai->runtime; ++ struct at91rm9200_runtime_data *prtd; ++ at91rm9200_pcm_dma_params_t *params; ++ ++ if (!runtime) ++ return 0; ++ ++ prtd = runtime->private_data; ++ params = prtd->params; ++ ++ /* disable the PDC and save the PDC registers */ ++ ++ at91_ssc_write(params->pdc->ptcr, params->mask->pdc_disable); ++ ++ prtd->pdc_xpr_save = at91_ssc_read(params->pdc->xpr); ++ prtd->pdc_xcr_save = at91_ssc_read(params->pdc->xcr); ++ prtd->pdc_xnpr_save = at91_ssc_read(params->pdc->xnpr); ++ prtd->pdc_xncr_save = at91_ssc_read(params->pdc->xncr); ++ ++ return 0; ++} ++ ++static int at91rm9200_pcm_resume(struct platform_device *pdev, ++ struct snd_soc_cpu_dai *dai) ++{ ++ struct snd_pcm_runtime *runtime = dai->runtime; ++ struct at91rm9200_runtime_data *prtd; ++ at91rm9200_pcm_dma_params_t *params; ++ ++ if (!runtime) ++ return 0; ++ ++ prtd = runtime->private_data; ++ params = prtd->params; ++ ++ /* restore the PDC registers and enable the PDC */ ++ at91_ssc_write(params->pdc->xpr, prtd->pdc_xpr_save); ++ at91_ssc_write(params->pdc->xcr, prtd->pdc_xcr_save); ++ at91_ssc_write(params->pdc->xnpr, prtd->pdc_xnpr_save); ++ at91_ssc_write(params->pdc->xncr, prtd->pdc_xncr_save); ++ ++ at91_ssc_write(params->pdc->ptcr, params->mask->pdc_enable); ++ return 0; ++} ++ ++struct snd_soc_platform at91rm9200_soc_platform = { ++ .name = "at91rm9200-audio", ++ .pcm_ops = &at91rm9200_pcm_ops, ++ .pcm_new = at91rm9200_pcm_new, ++ .pcm_free = at91rm9200_pcm_free_dma_buffers, ++ .suspend = at91rm9200_pcm_suspend, ++ .resume = at91rm9200_pcm_resume, ++}; ++ ++EXPORT_SYMBOL_GPL(at91rm9200_soc_platform); ++ ++MODULE_AUTHOR("Frank Mandarino <fmandarino@endrelia.com>"); ++MODULE_DESCRIPTION("Atmel AT91RM9200 PCM module"); ++MODULE_LICENSE("GPL"); +Index: linux-2.6-pxa-new/sound/soc/at91/at91rm9200-pcm.h +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/sound/soc/at91/at91rm9200-pcm.h +@@ -0,0 +1,75 @@ ++/* ++ * at91rm9200-pcm.h - ALSA PCM interface for the Atmel AT91RM9200 chip ++ * ++ * Author: Frank Mandarino <fmandarino@endrelia.com> ++ * Endrelia Technologies Inc. ++ * Created: Mar 3, 2006 ++ * ++ * Based on pxa2xx-pcm.h by: ++ * ++ * Author: Nicolas Pitre ++ * Created: Nov 30, 2004 ++ * Copyright: MontaVista Software, Inc. ++ * ++ * This program is free software; you can redistribute it and/or modify ++ * it under the terms of the GNU General Public License version 2 as ++ * published by the Free Software Foundation. ++ */ ++ ++/* ++ * Registers and status bits that are required by the PCM driver. ++ */ ++struct at91rm9200_ssc_regs { ++ void __iomem *cr; /* SSC control */ ++ void __iomem *ier; /* SSC interrupt enable */ ++ void __iomem *idr; /* SSC interrupt disable */ ++}; ++ ++struct at91rm9200_pdc_regs { ++ void __iomem *xpr; /* PDC recv/trans pointer */ ++ void __iomem *xcr; /* PDC recv/trans counter */ ++ void __iomem *xnpr; /* PDC next recv/trans pointer */ ++ void __iomem *xncr; /* PDC next recv/trans counter */ ++ void __iomem *ptcr; /* PDC transfer control */ ++}; ++ ++struct at91rm9200_ssc_mask { ++ u32 ssc_enable; /* SSC recv/trans enable */ ++ u32 ssc_disable; /* SSC recv/trans disable */ ++ u32 ssc_endx; /* SSC ENDTX or ENDRX */ ++ u32 ssc_endbuf; /* SSC TXBUFE or RXBUFF */ ++ u32 pdc_enable; /* PDC recv/trans enable */ ++ u32 pdc_disable; /* PDC recv/trans disable */ ++}; ++ ++ ++/* ++ * This structure, shared between the PCM driver and the interface, ++ * contains all information required by the PCM driver to perform the ++ * PDC DMA operation. All fields except dma_intr_handler() are initialized ++ * by the interface. The dms_intr_handler() pointer is set by the PCM ++ * driver and called by the interface SSC interrupt handler if it is ++ * non-NULL. ++ */ ++typedef struct { ++ char *name; /* stream identifier */ ++ int pdc_xfer_size; /* PDC counter increment in bytes */ ++ struct at91rm9200_ssc_regs *ssc; /* SSC register addresses */ ++ struct at91rm9200_pdc_regs *pdc; /* PDC receive/transmit registers */ ++ struct at91rm9200_ssc_mask *mask;/* SSC & PDC status bits */ ++ snd_pcm_substream_t *substream; ++ void (*dma_intr_handler)(u32, snd_pcm_substream_t *); ++} at91rm9200_pcm_dma_params_t; ++ ++extern struct snd_soc_cpu_dai at91rm9200_i2s_dai[3]; ++extern struct snd_soc_platform at91rm9200_soc_platform; ++ ++ ++/* ++ * SSC I/O helpers. ++ * E.g., at91_ssc_write(AT91_SSC(1) + AT91_SSC_CR, AT91_SSC_RXEN); ++ */ ++#define AT91_SSC(x) (((x)==0) ? AT91_VA_BASE_SSC0 :\ ++ ((x)==1) ? AT91_VA_BASE_SSC1 : ((x)==2) ? AT91_VA_BASE_SSC2 : NULL) ++#define at91_ssc_read(a) ((unsigned long) __raw_readl(a)) ++#define at91_ssc_write(a,v) __raw_writel((v),(a)) +Index: linux-2.6-pxa-new/sound/soc/imx/imx-ssi.c +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/sound/soc/imx/imx-ssi.c +@@ -0,0 +1,452 @@ ++/* ++ * imx-ssi.c -- SSI driver for Freescale IMX ++ * ++ * Copyright 2006 Wolfson Microelectronics PLC. ++ * Author: Liam Girdwood ++ * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com ++ * ++ * Based on mxc-alsa-mc13783 (C) 2006 Freescale. ++ * ++ * This program is free software; you can redistribute it and/or modify it ++ * under the terms of the GNU General Public License as published by the ++ * Free Software Foundation; either version 2 of the License, or (at your ++ * option) any later version. ++ * ++ * Revision history ++ * 29th Aug 2006 Initial version. ++ * ++ */ ++ ++#define IMX_DSP_DAIFMT \ ++ ( SND_SOC_DAIFMT_DSP__A |SND_SOC_DAIFMT_DSP_B | \ ++ SND_SOC_DAIFMT_CBS_CFS |SND_SOC_DAIFMT_CBM_CFS | \ ++ SND_SOC_DAIFMT_CBS_CFM |SND_SOC_DAIFMT_NB_NF |\ ++ SND_SOC_DAIFMT_NB_IF) ++ ++#define IMX_DSP_DIR \ ++ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE) ++ ++#define IMX_DSP_RATES \ ++ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | \ ++ SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | \ ++ SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ ++ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | \ ++ SNDRV_PCM_RATE_96000) ++ ++#define IMX_DSP_BITS \ ++ (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ ++ SNDRV_PCM_FMTBIT_S24_LE) ++ ++static struct snd_soc_dai_mode imx_dsp_pcm_modes[] = { ++ ++ /* frame master and clock slave mode */ ++ {IMX_DSP_DAIFMT | SND_SOC_DAIFMT_CBM_CFS, ++ SND_SOC_DAITDM_LRDW(0,0), IMX_DSP_BITS, IMX_DSP_RATES, ++ IMX_DSP_DIR, 0, SND_SOC_FS_ALL, ++ SND_SOC_FSB(32) | SND_SOC_FSB(32) | SND_SOC_FSB(16)}, ++ ++}; ++ ++static imx_pcm_dma_params_t imx_ssi1_pcm_stereo_out = { ++ .name = "SSI1 PCM Stereo out", ++ .params = { ++ .bd_number = 1, ++ .transfer_type = emi_2_per, ++ .watermark_level = SDMA_TXFIFO_WATERMARK, ++ .word_size = TRANSFER_16BIT, // maybe add this in setup func ++ .per_address = SSI1_STX0, ++ .event_id = DMA_REQ_SSI1_TX1, ++ .peripheral_type = SSI, ++ }, ++}; ++ ++static imx_pcm_dma_params_t imx_ssi1_pcm_stereo_in = { ++ .name = "SSI1 PCM Stereo in", ++ .params = { ++ .bd_number = 1, ++ .transfer_type = per_2_emi, ++ .watermark_level = SDMA_RXFIFO_WATERMARK, ++ .word_size = TRANSFER_16BIT, // maybe add this in setup func ++ .per_address = SSI1_SRX0, ++ .event_id = DMA_REQ_SSI1_RX1, ++ .peripheral_type = SSI, ++ }, ++}; ++ ++static imx_pcm_dma_params_t imx_ssi2_pcm_stereo_out = { ++ .name = "SSI2 PCM Stereo out", ++ .params = { ++ .bd_number = 1, ++ .transfer_type = per_2_emi, ++ .watermark_level = SDMA_TXFIFO_WATERMARK, ++ .word_size = TRANSFER_16BIT, // maybe add this in setup func ++ .per_address = SSI2_STX0, ++ .event_id = DMA_REQ_SSI2_TX1, ++ .peripheral_type = SSI, ++ }, ++}; ++ ++static imx_pcm_dma_params_t imx_ssi2_pcm_stereo_in = { ++ .name = "SSI2 PCM Stereo in", ++ .params = { ++ .bd_number = 1, ++ .transfer_type = per_2_emi, ++ .watermark_level = SDMA_RXFIFO_WATERMARK, ++ .word_size = TRANSFER_16BIT, // maybe add this in setup func ++ .per_address = SSI2_SRX0, ++ .event_id = DMA_REQ_SSI2_RX1, ++ .peripheral_type = SSI, ++ }, ++}; ++ ++static int imx_dsp_startup(struct snd_pcm_substream *substream) ++{ ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ ++ if (!rtd->cpu_dai->active) { ++ ++ } ++ ++ return 0; ++} ++ ++static int imx_ssi1_hw_tx_params(struct snd_pcm_substream *substream, ++ struct snd_pcm_hw_params *params) ++{ ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ struct snd_soc_device *socdev = rtd->socdev; ++ struct snd_soc_codec *codec = socdev->codec; ++ u16 bfs, div; ++ ++ bfs = SND_SOC_FSBD_REAL(rtd->cpu_dai->dai_runtime.bfs); ++ ++ SSI1_STCR = 0; ++ SSI1_STCCR = 0; ++ ++ /* DAI mode */ ++ switch(rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_FORMAT_MASK) { ++ case SND_SOC_DAIFMT_DSP_B: ++ SSI1_STCR |= SSI_STCR_TEFS; // data 1 bit after sync ++ case SND_SOC_DAIFMT_DSP_A: ++ SSI1_STCR |= SSI_STCR_TFSL; // frame is 1 bclk long ++ break; ++ } ++ ++ /* DAI clock inversion */ ++ switch(rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_INV_MASK) { ++ case SND_SOC_DAIFMT_IB_IF: ++ SSI1_STCR |= SSI_STCR_TFSI | SSI_STCR_TSCKP; ++ break; ++ case SND_SOC_DAIFMT_IB_NF: ++ SSI1_STCR |= SSI_STCR_TSCKP; ++ break; ++ case SND_SOC_DAIFMT_NB_IF: ++ SSI1_STCR |= SSI_STCR_TFSI; ++ break; ++ } ++ ++ /* DAI data (word) size */ ++ switch(rtd->codec_dai->dai_runtime.pcmfmt) { ++ case SNDRV_PCM_FMTBIT_S16_LE: ++ SSI1_STCCR |= SSI_STCCR_WL(16); ++ break; ++ case SNDRV_PCM_FMTBIT_S20_3LE: ++ SSI1_STCCR |= SSI_STCCR_WL(20); ++ break; ++ case SNDRV_PCM_FMTBIT_S24_LE: ++ SSI1_STCCR |= SSI_STCCR_WL(24); ++ break; ++ } ++ ++ /* DAI clock master masks */ ++ switch(rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_CLOCK_MASK){ ++ case SND_SOC_DAIFMT_CBM_CFM: ++ SSI1_STCR |= SSI_STCR_TFDIR | SSI_STCR_TXDIR; ++ break; ++ case SND_SOC_DAIFMT_CBS_CFM: ++ SSI1_STCR |= SSI_STCR_TFDIR; ++ break; ++ case SND_SOC_DAIFMT_CBM_CFS: ++ SSI1_STCR |= SSI_STCR_TXDIR; ++ break; ++ } ++ ++ /* DAI BCLK ratio to SYSCLK / MCLK */ ++ /* prescaler modulus - todo */ ++ switch (bfs) { ++ case 2: ++ break; ++ case 4: ++ break; ++ case 8: ++ break; ++ case 16: ++ break; ++ } ++ ++ /* TDM - todo, only fifo 0 atm */ ++ SSI1_STCR |= SSI_STCR_TFEN0; ++ SSI1_STCCR |= SSI_STCCR_DC(params_channels(params)); ++ ++ return 0; ++} ++ ++static int imx_ssi1_hw_rx_params(struct snd_pcm_substream *substream, ++ struct snd_pcm_hw_params *params) ++{ ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ struct snd_soc_device *socdev = rtd->socdev; ++ struct snd_soc_codec *codec = socdev->codec; ++ u16 bfs, div; ++ ++ bfs = SND_SOC_FSBD_REAL(rtd->cpu_dai->dai_runtime.bfs); ++ ++ SSI1_SRCR = 0; ++ SSI1_SRCCR = 0; ++ ++ /* DAI mode */ ++ switch(rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_FORMAT_MASK) { ++ case SND_SOC_DAIFMT_DSP_B: ++ SSI1_SRCR |= SSI_SRCR_REFS; // data 1 bit after sync ++ case SND_SOC_DAIFMT_DSP_A: ++ SSI1_SRCR |= SSI_SRCR_RFSL; // frame is 1 bclk long ++ break; ++ } ++ ++ /* DAI clock inversion */ ++ switch(rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_INV_MASK) { ++ case SND_SOC_DAIFMT_IB_IF: ++ SSI1_SRCR |= SSI_SRCR_TFSI | SSI_SRCR_TSCKP; ++ break; ++ case SND_SOC_DAIFMT_IB_NF: ++ SSI1_SRCR |= SSI_SRCR_RSCKP; ++ break; ++ case SND_SOC_DAIFMT_NB_IF: ++ SSI1_SRCR |= SSI_SRCR_RFSI; ++ break; ++ } ++ ++ /* DAI data (word) size */ ++ switch(rtd->codec_dai->dai_runtime.pcmfmt) { ++ case SNDRV_PCM_FMTBIT_S16_LE: ++ SSI1_SRCCR |= SSI_SRCCR_WL(16); ++ break; ++ case SNDRV_PCM_FMTBIT_S20_3LE: ++ SSI1_SRCCR |= SSI_SRCCR_WL(20); ++ break; ++ case SNDRV_PCM_FMTBIT_S24_LE: ++ SSI1_SRCCR |= SSI_SRCCR_WL(24); ++ break; ++ } ++ ++ /* DAI clock master masks */ ++ switch(rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_CLOCK_MASK){ ++ case SND_SOC_DAIFMT_CBM_CFM: ++ SSI1_SRCR |= SSI_SRCR_RFDIR | SSI_SRCR_RXDIR; ++ break; ++ case SND_SOC_DAIFMT_CBS_CFM: ++ SSI1_SRCR |= SSI_SRCR_RFDIR; ++ break; ++ case SND_SOC_DAIFMT_CBM_CFS: ++ SSI1_SRCR |= SSI_SRCR_RXDIR; ++ break; ++ } ++ ++ /* DAI BCLK ratio to SYSCLK / MCLK */ ++ /* prescaler modulus - todo */ ++ switch (bfs) { ++ case 2: ++ break; ++ case 4: ++ break; ++ case 8: ++ break; ++ case 16: ++ break; ++ } ++ ++ /* TDM - todo, only fifo 0 atm */ ++ SSI1_SRCR |= SSI_SRCR_RFEN0; ++ SSI1_SRCCR |= SSI_SRCCR_DC(params_channels(params)); ++ ++ return 0; ++} ++ ++static int imx_ssi_dsp_hw_params(struct snd_pcm_substream *substream, ++ struct snd_pcm_hw_params *params) ++{ ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ ++ /* clear register if not enabled */ ++ if(!(SSI1_SCR & SSI_SCR_SSIEN)) ++ SSI1_SCR = 0; ++ ++ /* async */ ++ if (rtd->cpu_dai->flags & SND_SOC_DAI_ASYNC) ++ SSI1_SCR |= SSI_SCR_SYN; ++ ++ /* DAI mode */ ++ switch(rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_FORMAT_MASK) { ++ case SND_SOC_DAIFMT_I2S: ++ case SND_SOC_DAIFMT_LEFT_J: ++ SSI1_SCR |= SSI_SCR_NET; ++ break; ++ } ++ ++ /* TDM - to complete */ ++ ++ /* Tx/Rx config */ ++ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { ++ return imx_ssi1_dsp_hw_tx_params(substream, params); ++ } else { ++ return imx_ssi1_dsp_hw_rx_params(substream, params); ++ } ++} ++ ++ ++ ++static int imx_ssi_dsp_trigger(struct snd_pcm_substream *substream, int cmd) ++{ ++ int ret = 0; ++ ++ switch (cmd) { ++ case SNDRV_PCM_TRIGGER_START: ++ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { ++ SSI1_SCR |= SSI_SCR_TE; ++ SSI1_SIER |= SSI_SIER_TDMAE; ++ } else { ++ SSI1_SCR |= SSI_SCR_RE; ++ SSI1_SIER |= SSI_SIER_RDMAE; ++ } ++ SSI1_SCR |= SSI_SCR_SSIEN; ++ ++ break; ++ case SNDRV_PCM_TRIGGER_RESUME: ++ break; ++ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: ++ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ++ SSI1_SCR |= SSI_SCR_TE; ++ else ++ SSI1_SCR |= SSI_SCR_RE; ++ break ++ case SNDRV_PCM_TRIGGER_STOP: ++ case SNDRV_PCM_TRIGGER_SUSPEND: ++ break; ++ case SNDRV_PCM_TRIGGER_PAUSE_PUSH: ++ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ++ SSI1_SCR &= ~SSI_SCR_TE; ++ else ++ SSI1_SCR &= ~SSI_SCR_RE; ++ break; ++ default: ++ ret = -EINVAL; ++ } ++ ++ return ret; ++} ++ ++static void imx_ssi_dsp_shutdown(struct snd_pcm_substream *substream) ++{ ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ ++ /* shutdown SSI */ ++ if (!rtd->cpu_dai->active) { ++ if(rtd->cpu_dai->id == 0) ++ SSI1_SCR &= ~SSI_SCR_SSIEN; ++ else ++ SSI2_SCR &= ~SSI_SCR_SSIEN; ++ } ++} ++ ++#ifdef CONFIG_PM ++static int imx_ssi_dsp_suspend(struct platform_device *dev, ++ struct snd_soc_cpu_dai *dai) ++{ ++ if(!dai->active) ++ return 0; ++ ++ if(rtd->cpu_dai->id == 0) ++ SSI1_SCR &= ~SSI_SCR_SSIEN; ++ else ++ SSI2_SCR &= ~SSI_SCR_SSIEN; ++ ++ return 0; ++} ++ ++static int imx_ssi_dsp_resume(struct platform_device *pdev, ++ struct snd_soc_cpu_dai *dai) ++{ ++ if(!dai->active) ++ return 0; ++ ++ if(rtd->cpu_dai->id == 0) ++ SSI1_SCR |= SSI_SCR_SSIEN; ++ else ++ SSI2_SCR |= SSI_SCR_SSIEN; ++ ++ return 0; ++} ++ ++#else ++#define imx_ssi_dsp_suspend NULL ++#define imx_ssi_dsp_resume NULL ++#endif ++ ++static unsigned int imx_ssi_config_dsp_sysclk(struct snd_soc_cpu_dai *iface, ++ struct snd_soc_clock_info *info, unsigned int clk) ++{ ++ return clk; ++} ++ ++struct snd_soc_cpu_dai imx_ssi_dsp_dai = { ++ .name = "imx-dsp-1", ++ .id = 0, ++ .type = SND_SOC_DAI_PCM, ++ .suspend = imx_ssi_dsp_suspend, ++ .resume = imx_ssi_dsp_resume, ++ .config_sysclk = imx_ssi_config_dsp_sysclk, ++ .playback = { ++ .channels_min = 1, ++ .channels_max = 2,}, ++ .capture = { ++ .channels_min = 1, ++ .channels_max = 2,}, ++ .ops = { ++ .startup = imx_ssi_dsp_startup, ++ .shutdown = imx_ssi_dsp_shutdown, ++ .trigger = imx_ssi_trigger, ++ .hw_params = imx_ssi_dsp_hw_params,}, ++ .caps = { ++ .num_modes = ARRAY_SIZE(imx_dsp_modes), ++ .mode = imx_dsp_modes,}, ++}, ++{ ++ .name = "imx-dsp-2", ++ .id = 1, ++ .type = SND_SOC_DAI_PCM, ++ .suspend = imx_ssi_dsp_suspend, ++ .resume = imx_ssi_dsp_resume, ++ .config_sysclk = imx_ssi_config_dsp_sysclk, ++ .playback = { ++ .channels_min = 1, ++ .channels_max = 2,}, ++ .capture = { ++ .channels_min = 1, ++ .channels_max = 2,}, ++ .ops = { ++ .startup = imx_dsp_startup, ++ .shutdown = imx_dsp_shutdown, ++ .trigger = imx_ssi1_trigger, ++ .hw_params = imx_ssi1_pcm_hw_params,}, ++ .caps = { ++ .num_modes = ARRAY_SIZE(imx_dsp_modes), ++ .mode = imx_dsp_modes,}, ++}; ++ ++ ++EXPORT_SYMBOL_GPL(imx_ssi_dsp_dai); ++ ++/* Module information */ ++MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com"); ++MODULE_DESCRIPTION("i.MX ASoC SSI driver"); ++MODULE_LICENSE("GPL"); +Index: linux-2.6-pxa-new/sound/soc/imx/Kconfig +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/sound/soc/imx/Kconfig +@@ -0,0 +1,31 @@ ++menu "SoC Audio for the Freescale i.MX" ++ ++config SND_MXC_SOC ++ tristate "SoC Audio for the Freescale i.MX CPU" ++ depends on ARCH_MXC && SND ++ select SND_PCM ++ help ++ Say Y or M if you want to add support for codecs attached to ++ the MXC AC97, I2S or SSP interface. You will also need ++ to select the audio interfaces to support below. ++ ++config SND_MXC_AC97 ++ tristate ++ select SND_AC97_CODEC ++ ++config SND_MXC_SOC_AC97 ++ tristate ++ select SND_AC97_BUS ++ ++config SND_MXC_SOC_SSI ++ tristate ++ ++config SND_MXC_SOC_MX3_WM8753 ++ tristate "SoC Audio support for MX31 - WM8753" ++ depends on SND_MXC_SOC && ARCH_MX3 ++ select SND_MXC_SOC_SSI ++ help ++ Say Y if you want to add support for SoC audio on MX31ADS ++ with the WM8753. ++ ++endmenu +Index: linux-2.6-pxa-new/sound/soc/imx/Makefile +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/sound/soc/imx/Makefile +@@ -0,0 +1,18 @@ ++# i.MX Platform Support ++snd-soc-imx21-objs := imx21-pcm.o ++snd-soc-imx31-objs := imx31-pcm.o ++snd-soc-imx-ac97-objs := imx-ac97.o ++snd-soc-imx-i2s-objs := imx-i2s.o ++ ++obj-$(CONFIG_SND_MXC_SOC) += snd-soc-imx.o ++obj-$(CONFIG_SND_MXC_SOC_AC97) += snd-soc-imx-ac97.o ++obj-$(CONFIG_SND_MXC_SOC_I2S) += snd-soc-imx-i2s.o ++ ++# i.MX Machine Support ++snd-soc-mx31ads-wm8753-objs := mx31ads_wm8753.o ++obj-$(CONFIG_SND_SOC_MX31ADS_WM8753) += snd-soc-mx31ads-wm8753.o ++snd-soc-mx21ads-wm8753-objs := mx21ads_wm8753.o ++obj-$(CONFIG_SND_SOC_MX21ADS_WM8753) += snd-soc-mx21ads-wm8753.o ++snd-soc-mx21ads-wm8731-objs := mx21ads_wm8731.o ++obj-$(CONFIG_SND_SOC_MX21ADS_WM8731) += snd-soc-mx21ads-wm8731.o ++ +Index: linux-2.6-pxa-new/sound/Makefile +=================================================================== +--- linux-2.6-pxa-new.orig/sound/Makefile ++++ linux-2.6-pxa-new/sound/Makefile +@@ -5,7 +5,7 @@ obj-$(CONFIG_SOUND) += soundcore.o + obj-$(CONFIG_SOUND_PRIME) += sound_firmware.o + obj-$(CONFIG_SOUND_PRIME) += oss/ + obj-$(CONFIG_DMASOUND) += oss/ +-obj-$(CONFIG_SND) += core/ i2c/ drivers/ isa/ pci/ ppc/ arm/ synth/ usb/ sparc/ parisc/ pcmcia/ mips/ ++obj-$(CONFIG_SND) += core/ i2c/ drivers/ isa/ pci/ ppc/ arm/ synth/ usb/ sparc/ parisc/ pcmcia/ mips/ soc/ + obj-$(CONFIG_SND_AOA) += aoa/ + + ifeq ($(CONFIG_SND),y) +Index: linux-2.6-pxa-new/sound/soc/codecs/wm8711.c +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/sound/soc/codecs/wm8711.c +@@ -0,0 +1,843 @@ ++/* ++ * wm8711.c -- WM8711 ALSA SoC Audio driver ++ * ++ * Copyright 2006 Wolfson Microelectronics ++ * ++ * Author: Mike Arthur <linux@wolfsonmicro.com> ++ * ++ * Based on wm8711.c by Richard Purdie ++ * ++ * This program is free software; you can redistribute it and/or modify ++ * it under the terms of the GNU General Public License version 2 as ++ * published by the Free Software Foundation. ++ */ ++ ++#include <linux/module.h> ++#include <linux/moduleparam.h> ++#include <linux/init.h> ++#include <linux/delay.h> ++#include <linux/pm.h> ++#include <linux/i2c.h> ++#include <linux/platform_device.h> ++#include <sound/driver.h> ++#include <sound/core.h> ++#include <sound/pcm.h> ++#include <sound/pcm_params.h> ++#include <sound/soc.h> ++#include <sound/soc-dapm.h> ++#include <sound/initval.h> ++ ++#include "wm8711.h" ++ ++#define AUDIO_NAME "wm8711" ++#define WM8711_VERSION "0.2" ++ ++/* ++ * Debug ++ */ ++ ++#define WM8711_DEBUG 0 ++ ++#ifdef WM8711_DEBUG ++#define dbg(format, arg...) \ ++ printk(KERN_DEBUG AUDIO_NAME ": " format "\n" , ## arg) ++#else ++#define dbg(format, arg...) do {} while (0) ++#endif ++#define err(format, arg...) \ ++ printk(KERN_ERR AUDIO_NAME ": " format "\n" , ## arg) ++#define info(format, arg...) \ ++ printk(KERN_INFO AUDIO_NAME ": " format "\n" , ## arg) ++#define warn(format, arg...) \ ++ printk(KERN_WARNING AUDIO_NAME ": " format "\n" , ## arg) ++ ++struct snd_soc_codec_device soc_codec_dev_wm8711; ++ ++/* ++ * wm8711 register cache ++ * We can't read the WM8711 register space when we are ++ * using 2 wire for device control, so we cache them instead. ++ * There is no point in caching the reset register ++ */ ++static const u16 wm8711_reg[WM8711_CACHEREGNUM] = { ++ 0x0079, 0x0079, 0x000a, 0x0008, ++ 0x009f, 0x000a, 0x0000, 0x0000 ++}; ++ ++#define WM8711_DAIFMT \ ++ (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_RIGHT_J | \ ++ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_NB_IF | SND_SOC_DAIFMT_IB_NF | \ ++ SND_SOC_DAIFMT_IB_IF) ++ ++#define WM8711_DIR \ ++ (SND_SOC_DAIDIR_PLAYBACK) ++ ++#define WM8711_RATES \ ++ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ ++ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ ++ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) ++ ++#define WM8711_HIFI_BITS \ ++ (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ ++ SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_LE | \ ++ SNDRV_PCM_FMTBIT_S32_LE) ++ ++static struct snd_soc_dai_mode wm8711_modes[] = { ++ /* codec frame and clock master modes */ ++ /* 8k */ ++ { ++ .fmt = WM8711_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8711_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_8000, ++ .pcmdir = WM8711_DIR, ++ .flags = SND_SOC_DAI_BFS_RATE, ++ .fs = 1536, ++ .bfs = 64, ++ }, ++ { ++ .fmt = WM8711_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8711_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_8000, ++ .pcmdir = WM8711_DIR, ++ .flags = SND_SOC_DAI_BFS_RATE, ++ .fs = 2304, ++ .bfs = 64, ++ }, ++ { ++ .fmt = WM8711_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8711_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_8000, ++ .pcmdir = WM8711_DIR, ++ .flags = SND_SOC_DAI_BFS_RATE, ++ .fs = 1408, ++ .bfs = 64, ++ }, ++ { ++ .fmt = WM8711_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8711_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_8000, ++ .pcmdir = WM8711_DIR, ++ .flags = SND_SOC_DAI_BFS_RATE, ++ .fs = 2112, ++ .bfs = 64, ++ }, ++ ++ /* 32k */ ++ { ++ .fmt = WM8711_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8711_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_32000, ++ .pcmdir = WM8711_DIR, ++ .flags = SND_SOC_DAI_BFS_RATE, ++ .fs = 384, ++ .bfs = 64, ++ }, ++ { ++ .fmt = WM8711_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8711_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_32000, ++ .pcmdir = WM8711_DIR, ++ .flags = SND_SOC_DAI_BFS_RATE, ++ .fs = 576, ++ .bfs = 64, ++ }, ++ ++ /* 44.1k & 48k */ ++ { ++ .fmt = WM8711_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8711_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000, ++ .pcmdir = WM8711_DIR, ++ .flags = SND_SOC_DAI_BFS_RATE, ++ .fs = 256, ++ .bfs = 64, ++ }, ++ { ++ .fmt = WM8711_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8711_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000, ++ .pcmdir = WM8711_DIR, ++ .flags = SND_SOC_DAI_BFS_RATE, ++ .fs = 384, ++ .bfs = 64, ++ }, ++ ++ /* 88.2 & 96k */ ++ { ++ .fmt = WM8711_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8711_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000, ++ .pcmdir = WM8711_DIR, ++ .flags = SND_SOC_DAI_BFS_RATE, ++ .fs = 128, ++ .bfs = 64, ++ ++ }, ++ { ++ .fmt = WM8711_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8711_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000, ++ .pcmdir = WM8711_DIR, ++ .flags = SND_SOC_DAI_BFS_RATE, ++ .fs = 192, ++ .bfs = 64, ++ }, ++ ++ /* USB codec frame and clock master modes */ ++ /* 8k */ ++ { ++ .fmt = WM8711_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8711_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_8000, ++ .pcmdir = WM8711_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 1500, ++ .bfs = SND_SOC_FSBD(1), ++ }, ++ ++ /* 44.1k */ ++ { ++ .fmt = WM8711_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8711_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_44100, ++ .pcmdir = WM8711_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 272, ++ .bfs = SND_SOC_FSBD(1), ++ }, ++ ++ /* 48k */ ++ { ++ .fmt = WM8711_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8711_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_48000, ++ .pcmdir = WM8711_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 250, ++ .bfs = SND_SOC_FSBD(1), ++ }, ++ ++ /* 88.2k */ ++ { ++ .fmt = WM8711_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8711_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_88200, ++ .pcmdir = WM8711_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 136, ++ .bfs = SND_SOC_FSBD(1), ++ }, ++ ++ /* 96k */ ++ { ++ .fmt = WM8711_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8711_HIFI_BITS, ++ .pcmrate = SNDRV_PCM_RATE_96000, ++ .pcmdir = WM8711_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 125, ++ .bfs = SND_SOC_FSBD(1), ++ }, ++ ++ /* codec frame and clock slave modes */ ++ { ++ .fmt = WM8711_DAIFMT | SND_SOC_DAIFMT_CBS_CFS, ++ .pcmfmt = WM8711_HIFI_BITS, ++ .pcmrate = WM8711_RATES, ++ .pcmdir = WM8711_DIR, ++ .fs = SND_SOC_FS_ALL, ++ .bfs = SND_SOC_FSB_ALL, ++ }, ++}; ++ ++/* ++ * read wm8711 register cache ++ */ ++static inline unsigned int wm8711_read_reg_cache(struct snd_soc_codec * codec, ++ unsigned int reg) ++{ ++ u16 *cache = codec->reg_cache; ++ if (reg == WM8711_RESET) ++ return 0; ++ if (reg >= WM8711_CACHEREGNUM) ++ return -1; ++ return cache[reg]; ++} ++ ++/* ++ * write wm8711 register cache ++ */ ++static inline void wm8711_write_reg_cache(struct snd_soc_codec *codec, ++ u16 reg, unsigned int value) ++{ ++ u16 *cache = codec->reg_cache; ++ if (reg >= WM8711_CACHEREGNUM) ++ return; ++ cache[reg] = value; ++} ++ ++/* ++ * write to the WM8711 register space ++ */ ++static int wm8711_write(struct snd_soc_codec * codec, unsigned int reg, ++ unsigned int value) ++{ ++ u8 data[2]; ++ ++ /* data is ++ * D15..D9 WM8753 register offset ++ * D8...D0 register data ++ */ ++ data[0] = (reg << 1) | ((value >> 8) & 0x0001); ++ data[1] = value & 0x00ff; ++ ++ wm8711_write_reg_cache (codec, reg, value); ++ if (codec->hw_write(codec->control_data, data, 2) == 2) ++ return 0; ++ else ++ return -EIO; ++} ++ ++#define wm8711_reset(c) wm8711_write(c, WM8711_RESET, 0) ++ ++static const struct snd_kcontrol_new wm8711_snd_controls[] = { ++ ++SOC_DOUBLE_R("Master Playback Volume", WM8711_LOUT1V, WM8711_ROUT1V, ++ 0, 127, 0), ++SOC_DOUBLE_R("Master Playback ZC Switch", WM8711_LOUT1V, WM8711_ROUT1V, ++ 7, 1, 0), ++ ++}; ++ ++/* add non dapm controls */ ++static int wm8711_add_controls(struct snd_soc_codec *codec) ++{ ++ int err, i; ++ ++ for (i = 0; i < ARRAY_SIZE(wm8711_snd_controls); i++) { ++ err = snd_ctl_add(codec->card, ++ snd_soc_cnew(&wm8711_snd_controls[i],codec, NULL)); ++ if (err < 0) ++ return err; ++ } ++ ++ return 0; ++} ++ ++/* Output Mixer */ ++static const snd_kcontrol_new_t wm8711_output_mixer_controls[] = { ++SOC_DAPM_SINGLE("Line Bypass Switch", WM8711_APANA, 3, 1, 0), ++SOC_DAPM_SINGLE("HiFi Playback Switch", WM8711_APANA, 4, 1, 0), ++}; ++ ++static const struct snd_soc_dapm_widget wm8711_dapm_widgets[] = { ++SND_SOC_DAPM_MIXER("Output Mixer", WM8711_PWR, 4, 1, ++ &wm8711_output_mixer_controls[0], ++ ARRAY_SIZE(wm8711_output_mixer_controls)), ++SND_SOC_DAPM_DAC("DAC", "HiFi Playback", WM8711_PWR, 3, 1), ++SND_SOC_DAPM_OUTPUT("LOUT"), ++SND_SOC_DAPM_OUTPUT("LHPOUT"), ++SND_SOC_DAPM_OUTPUT("ROUT"), ++SND_SOC_DAPM_OUTPUT("RHPOUT"), ++}; ++ ++static const char *intercon[][3] = { ++ /* output mixer */ ++ {"Output Mixer", "Line Bypass Switch", "Line Input"}, ++ {"Output Mixer", "HiFi Playback Switch", "DAC"}, ++ ++ /* outputs */ ++ {"RHPOUT", NULL, "Output Mixer"}, ++ {"ROUT", NULL, "Output Mixer"}, ++ {"LHPOUT", NULL, "Output Mixer"}, ++ {"LOUT", NULL, "Output Mixer"}, ++ ++ /* terminator */ ++ {NULL, NULL, NULL}, ++}; ++ ++static int wm8711_add_widgets(struct snd_soc_codec *codec) ++{ ++ int i; ++ ++ for(i = 0; i < ARRAY_SIZE(wm8711_dapm_widgets); i++) { ++ snd_soc_dapm_new_control(codec, &wm8711_dapm_widgets[i]); ++ } ++ ++ /* set up audio path interconnects */ ++ for(i = 0; intercon[i][0] != NULL; i++) { ++ snd_soc_dapm_connect_input(codec, intercon[i][0], intercon[i][1], ++ intercon[i][2]); ++ } ++ ++ snd_soc_dapm_new_widgets(codec); ++ return 0; ++} ++ ++struct _coeff_div { ++ u32 mclk; ++ u32 rate; ++ u16 fs; ++ u8 sr:4; ++ u8 bosr:1; ++ u8 usb:1; ++}; ++ ++/* codec mclk clock divider coefficients */ ++static const struct _coeff_div coeff_div[] = { ++ /* 48k */ ++ {12288000, 48000, 256, 0x0, 0x0, 0x0}, ++ {18432000, 48000, 384, 0x0, 0x1, 0x0}, ++ {12000000, 48000, 250, 0x0, 0x0, 0x1}, ++ ++ /* 32k */ ++ {12288000, 32000, 384, 0x6, 0x0, 0x0}, ++ {18432000, 32000, 576, 0x6, 0x1, 0x0}, ++ ++ /* 8k */ ++ {12288000, 8000, 1536, 0x3, 0x0, 0x0}, ++ {18432000, 8000, 2304, 0x3, 0x1, 0x0}, ++ {11289600, 8000, 1408, 0xb, 0x0, 0x0}, ++ {16934400, 8000, 2112, 0xb, 0x1, 0x0}, ++ {12000000, 8000, 1500, 0x3, 0x0, 0x1}, ++ ++ /* 96k */ ++ {12288000, 96000, 128, 0x7, 0x0, 0x0}, ++ {18432000, 96000, 192, 0x7, 0x1, 0x0}, ++ {12000000, 96000, 125, 0x7, 0x0, 0x1}, ++ ++ /* 44.1k */ ++ {11289600, 44100, 256, 0x8, 0x0, 0x0}, ++ {16934400, 44100, 384, 0x8, 0x1, 0x0}, ++ {12000000, 44100, 272, 0x8, 0x1, 0x1}, ++ ++ /* 88.2k */ ++ {11289600, 88200, 128, 0xf, 0x0, 0x0}, ++ {16934400, 88200, 192, 0xf, 0x1, 0x0}, ++ {12000000, 88200, 136, 0xf, 0x1, 0x1}, ++}; ++ ++static inline int get_coeff(int mclk, int rate) ++{ ++ int i; ++ ++ for (i = 0; i < ARRAY_SIZE(coeff_div); i++) { ++ if (coeff_div[i].rate == rate && coeff_div[i].mclk == mclk) ++ return i; ++ } ++ return 0; ++} ++ ++/* WM8711 supports numerous clocks per sample rate */ ++static unsigned int wm8711_config_sysclk(struct snd_soc_codec_dai *dai, ++ struct snd_soc_clock_info *info, unsigned int clk) ++{ ++ dai->mclk = 0; ++ ++ /* check that the calculated FS and rate actually match a clock from ++ * the machine driver */ ++ if (info->fs * info->rate == clk) ++ dai->mclk = clk; ++ ++ return dai->mclk; ++} ++ ++static int wm8711_pcm_prepare(struct snd_pcm_substream *substream) ++{ ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ struct snd_soc_device *socdev = rtd->socdev; ++ struct snd_soc_codec *codec = socdev->codec; ++ u16 iface = 0, srate; ++ int i = get_coeff(rtd->codec_dai->mclk, ++ snd_soc_get_rate(rtd->codec_dai->dai_runtime.pcmrate)); ++ ++ /* set master/slave audio interface */ ++ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_CLOCK_MASK) { ++ case SND_SOC_DAIFMT_CBM_CFM: ++ iface |= 0x0040; ++ break; ++ case SND_SOC_DAIFMT_CBS_CFS: ++ break; ++ } ++ srate = (coeff_div[i].sr << 2) | (coeff_div[i].bosr << 1) | ++ coeff_div[i].usb; ++ wm8711_write(codec, WM8711_SRATE, srate); ++ ++ /* interface format */ ++ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_FORMAT_MASK) { ++ case SND_SOC_DAIFMT_I2S: ++ iface |= 0x0002; ++ break; ++ case SND_SOC_DAIFMT_RIGHT_J: ++ break; ++ case SND_SOC_DAIFMT_LEFT_J: ++ iface |= 0x0001; ++ break; ++ case SND_SOC_DAIFMT_DSP_A: ++ iface |= 0x0003; ++ break; ++ case SND_SOC_DAIFMT_DSP_B: ++ iface |= 0x0013; ++ break; ++ } ++ ++ /* bit size */ ++ switch (rtd->codec_dai->dai_runtime.pcmfmt) { ++ case SNDRV_PCM_FORMAT_S16_LE: ++ break; ++ case SNDRV_PCM_FORMAT_S20_3LE: ++ iface |= 0x0004; ++ break; ++ case SNDRV_PCM_FORMAT_S24_LE: ++ iface |= 0x0008; ++ break; ++ case SNDRV_PCM_FORMAT_S32_LE: ++ iface |= 0x000c; ++ break; ++ } ++ ++ /* clock inversion */ ++ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_INV_MASK) { ++ case SND_SOC_DAIFMT_NB_NF: ++ break; ++ case SND_SOC_DAIFMT_IB_IF: ++ iface |= 0x0090; ++ break; ++ case SND_SOC_DAIFMT_IB_NF: ++ iface |= 0x0080; ++ break; ++ case SND_SOC_DAIFMT_NB_IF: ++ iface |= 0x0010; ++ break; ++ } ++ ++ /* set iface */ ++ wm8711_write(codec, WM8711_IFACE, iface); ++ ++ /* set active */ ++ wm8711_write(codec, WM8711_ACTIVE, 0x0001); ++ return 0; ++} ++ ++static void wm8711_shutdown(struct snd_pcm_substream *substream) ++{ ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ struct snd_soc_device *socdev = rtd->socdev; ++ struct snd_soc_codec *codec = socdev->codec; ++ ++ /* deactivate */ ++ if (!codec->active) { ++ udelay(50); ++ wm8711_write(codec, WM8711_ACTIVE, 0x0); ++ } ++} ++ ++static int wm8711_mute(struct snd_soc_codec *codec, ++ struct snd_soc_codec_dai *dai, int mute) ++{ ++ u16 mute_reg = wm8711_read_reg_cache(codec, WM8711_APDIGI) & 0xfff7; ++ if (mute) ++ wm8711_write(codec, WM8711_APDIGI, mute_reg | 0x8); ++ else ++ wm8711_write(codec, WM8711_APDIGI, mute_reg); ++ ++ return 0; ++} ++ ++static int wm8711_dapm_event(struct snd_soc_codec *codec, int event) ++{ ++ u16 reg = wm8711_read_reg_cache(codec, WM8711_PWR) & 0xff7f; ++ ++ switch (event) { ++ case SNDRV_CTL_POWER_D0: /* full On */ ++ /* vref/mid, osc on, dac unmute */ ++ wm8711_write(codec, WM8711_PWR, reg); ++ break; ++ case SNDRV_CTL_POWER_D1: /* partial On */ ++ case SNDRV_CTL_POWER_D2: /* partial On */ ++ break; ++ case SNDRV_CTL_POWER_D3hot: /* Off, with power */ ++ /* everything off except vref/vmid, */ ++ wm8711_write(codec, WM8711_PWR, reg | 0x0040); ++ break; ++ case SNDRV_CTL_POWER_D3cold: /* Off, without power */ ++ /* everything off, dac mute, inactive */ ++ wm8711_write(codec, WM8711_ACTIVE, 0x0); ++ wm8711_write(codec, WM8711_PWR, 0xffff); ++ break; ++ } ++ codec->dapm_state = event; ++ return 0; ++} ++ ++struct snd_soc_codec_dai wm8711_dai = { ++ .name = "WM8711", ++ .playback = { ++ .stream_name = "Playback", ++ .channels_min = 1, ++ .channels_max = 2, ++ }, ++ .config_sysclk = wm8711_config_sysclk, ++ .digital_mute = wm8711_mute, ++ .ops = { ++ .prepare = wm8711_pcm_prepare, ++ .shutdown = wm8711_shutdown, ++ }, ++ .caps = { ++ .num_modes = ARRAY_SIZE(wm8711_modes), ++ .mode = wm8711_modes, ++ }, ++}; ++EXPORT_SYMBOL_GPL(wm8711_dai); ++ ++static int wm8711_suspend(struct platform_device *pdev, pm_message_t state) ++{ ++ struct snd_soc_device *socdev = platform_get_drvdata(pdev); ++ struct snd_soc_codec *codec = socdev->codec; ++ ++ wm8711_write(codec, WM8711_ACTIVE, 0x0); ++ wm8711_dapm_event(codec, SNDRV_CTL_POWER_D3cold); ++ return 0; ++} ++ ++static int wm8711_resume(struct platform_device *pdev) ++{ ++ struct snd_soc_device *socdev = platform_get_drvdata(pdev); ++ struct snd_soc_codec *codec = socdev->codec; ++ int i; ++ u8 data[2]; ++ u16 *cache = codec->reg_cache; ++ ++ /* Sync reg_cache with the hardware */ ++ for (i = 0; i < ARRAY_SIZE(wm8711_reg); i++) { ++ data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); ++ data[1] = cache[i] & 0x00ff; ++ codec->hw_write(codec->control_data, data, 2); ++ } ++ wm8711_dapm_event(codec, SNDRV_CTL_POWER_D3hot); ++ wm8711_dapm_event(codec, codec->suspend_dapm_state); ++ return 0; ++} ++ ++/* ++ * initialise the WM8711 driver ++ * register the mixer and dsp interfaces with the kernel ++ */ ++static int wm8711_init(struct snd_soc_device* socdev) ++{ ++ struct snd_soc_codec* codec = socdev->codec; ++ int reg, ret = 0; ++ ++ codec->name = "WM8711"; ++ codec->owner = THIS_MODULE; ++ codec->read = wm8711_read_reg_cache; ++ codec->write = wm8711_write; ++ codec->dapm_event = wm8711_dapm_event; ++ codec->dai = &wm8711_dai; ++ codec->num_dai = 1; ++ codec->reg_cache_size = ARRAY_SIZE(wm8711_reg); ++ codec->reg_cache = ++ kzalloc(sizeof(u16) * ARRAY_SIZE(wm8711_reg), GFP_KERNEL); ++ if (codec->reg_cache == NULL) ++ return -ENOMEM; ++ memcpy(codec->reg_cache, wm8711_reg, ++ sizeof(u16) * ARRAY_SIZE(wm8711_reg)); ++ codec->reg_cache_size = sizeof(u16) * ARRAY_SIZE(wm8711_reg); ++ ++ wm8711_reset(codec); ++ ++ /* register pcms */ ++ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); ++ if (ret < 0) { ++ kfree(codec->reg_cache); ++ return ret; ++ } ++ ++ /* power on device */ ++ wm8711_dapm_event(codec, SNDRV_CTL_POWER_D3hot); ++ ++ /* set the update bits */ ++ reg = wm8711_read_reg_cache(codec, WM8711_LOUT1V); ++ wm8711_write(codec, WM8711_LOUT1V, reg | 0x0100); ++ reg = wm8711_read_reg_cache(codec, WM8711_ROUT1V); ++ wm8711_write(codec, WM8711_ROUT1V, reg | 0x0100); ++ ++ wm8711_add_controls(codec); ++ wm8711_add_widgets(codec); ++ ret = snd_soc_register_card(socdev); ++ if (ret < 0) { ++ snd_soc_free_pcms(socdev); ++ snd_soc_dapm_free(socdev); ++ } ++ ++ return ret; ++} ++ ++static struct snd_soc_device *wm8711_socdev; ++ ++#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) ++ ++/* ++ * WM8711 2 wire address is determined by GPIO5 ++ * state during powerup. ++ * low = 0x1a ++ * high = 0x1b ++ */ ++#define I2C_DRIVERID_WM8711 0xfefe /* liam - need a proper id */ ++ ++static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END }; ++ ++/* Magic definition of all other variables and things */ ++I2C_CLIENT_INSMOD; ++ ++static struct i2c_driver wm8711_i2c_driver; ++static struct i2c_client client_template; ++ ++/* If the i2c layer weren't so broken, we could pass this kind of data ++ around */ ++ ++static int wm8711_codec_probe(struct i2c_adapter *adap, int addr, int kind) ++{ ++ struct snd_soc_device *socdev = wm8711_socdev; ++ struct wm8711_setup_data *setup = socdev->codec_data; ++ struct snd_soc_codec *codec = socdev->codec; ++ struct i2c_client *i2c; ++ int ret; ++ ++ if (addr != setup->i2c_address) ++ return -ENODEV; ++ ++ client_template.adapter = adap; ++ client_template.addr = addr; ++ ++ i2c = kzalloc(sizeof(struct i2c_client), GFP_KERNEL); ++ if (i2c == NULL){ ++ kfree(codec); ++ return -ENOMEM; ++ } ++ memcpy(i2c, &client_template, sizeof(struct i2c_client)); ++ ++ i2c_set_clientdata(i2c, codec); ++ ++ codec->control_data = i2c; ++ ++ ret = i2c_attach_client(i2c); ++ if (ret < 0) { ++ err("failed to attach codec at addr %x\n", addr); ++ goto err; ++ } ++ ++ ret = wm8711_init(socdev); ++ if (ret < 0) { ++ err("failed to initialise WM8711\n"); ++ goto err; ++ } ++ return ret; ++ ++err: ++ kfree(codec); ++ kfree(i2c); ++ return ret; ++ ++} ++ ++static int wm8711_i2c_detach(struct i2c_client *client) ++{ ++ struct snd_soc_codec* codec = i2c_get_clientdata(client); ++ ++ i2c_detach_client(client); ++ ++ kfree(codec->reg_cache); ++ kfree(client); ++ ++ return 0; ++} ++ ++static int wm8711_i2c_attach(struct i2c_adapter *adap) ++{ ++ return i2c_probe(adap, &addr_data, wm8711_codec_probe); ++} ++ ++/* corgi i2c codec control layer */ ++static struct i2c_driver wm8711_i2c_driver = { ++ .driver = { ++ .name = "WM8711 I2C Codec", ++ .owner = THIS_MODULE, ++ }, ++ .id = I2C_DRIVERID_WM8711, ++ .attach_adapter = wm8711_i2c_attach, ++ .detach_client = wm8711_i2c_detach, ++ .command = NULL, ++}; ++ ++static struct i2c_client client_template = { ++ .name = "WM8711", ++ .driver = &wm8711_i2c_driver, ++}; ++#endif ++ ++static int wm8711_probe(struct platform_device *pdev) ++{ ++ struct snd_soc_device *socdev = platform_get_drvdata(pdev); ++ struct wm8711_setup_data *setup; ++ struct snd_soc_codec* codec; ++ int ret = 0; ++ ++ info("WM8711 Audio Codec %s", WM8711_VERSION); ++ ++ setup = socdev->codec_data; ++ codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); ++ if (codec == NULL) ++ return -ENOMEM; ++ ++ socdev->codec = codec; ++ mutex_init(&codec->mutex); ++ INIT_LIST_HEAD(&codec->dapm_widgets); ++ INIT_LIST_HEAD(&codec->dapm_paths); ++ ++ wm8711_socdev = socdev; ++#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) ++ if (setup->i2c_address) { ++ normal_i2c[0] = setup->i2c_address; ++ codec->hw_write = (hw_write_t)i2c_master_send; ++ ret = i2c_add_driver(&wm8711_i2c_driver); ++ if (ret != 0) ++ printk(KERN_ERR "can't add i2c driver"); ++ } ++#else ++ /* Add other interfaces here */ ++#endif ++ return ret; ++} ++ ++/* power down chip */ ++static int wm8711_remove(struct platform_device *pdev) ++{ ++ struct snd_soc_device *socdev = platform_get_drvdata(pdev); ++ struct snd_soc_codec *codec = socdev->codec; ++ ++ if (codec->control_data) ++ wm8711_dapm_event(codec, SNDRV_CTL_POWER_D3cold); ++ ++ snd_soc_free_pcms(socdev); ++ snd_soc_dapm_free(socdev); ++#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) ++ i2c_del_driver(&wm8711_i2c_driver); ++#endif ++ kfree(codec); ++ ++ return 0; ++} ++ ++struct snd_soc_codec_device soc_codec_dev_wm8711 = { ++ .probe = wm8711_probe, ++ .remove = wm8711_remove, ++ .suspend = wm8711_suspend, ++ .resume = wm8711_resume, ++}; ++ ++EXPORT_SYMBOL_GPL(soc_codec_dev_wm8711); ++ ++MODULE_DESCRIPTION("ASoC WM8711 driver"); ++MODULE_AUTHOR("Mike Arthur"); ++MODULE_LICENSE("GPL"); +Index: linux-2.6-pxa-new/sound/soc/codecs/wm8711.h +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/sound/soc/codecs/wm8711.h +@@ -0,0 +1,39 @@ ++/* ++ * wm8711.h -- WM8711 Soc Audio driver ++ * ++ * Copyright 2006 Wolfson Microelectronics ++ * ++ * Author: Mike Arthur <linux@wolfsonmicro.com> ++ * ++ * Based on wm8731.h ++ * ++ * This program is free software; you can redistribute it and/or modify ++ * it under the terms of the GNU General Public License version 2 as ++ * published by the Free Software Foundation. ++ */ ++ ++#ifndef _WM8711_H ++#define _WM8711_H ++ ++/* WM8711 register space */ ++ ++#define WM8711_LOUT1V 0x02 ++#define WM8711_ROUT1V 0x03 ++#define WM8711_APANA 0x04 ++#define WM8711_APDIGI 0x05 ++#define WM8711_PWR 0x06 ++#define WM8711_IFACE 0x07 ++#define WM8711_SRATE 0x08 ++#define WM8711_ACTIVE 0x09 ++#define WM8711_RESET 0x0f ++ ++#define WM8711_CACHEREGNUM 8 ++ ++struct wm8711_setup_data { ++ unsigned short i2c_address; ++}; ++ ++extern struct snd_soc_codec_dai wm8711_dai; ++extern struct snd_soc_codec_device soc_codec_dev_wm8711; ++ ++#endif +Index: linux-2.6-pxa-new/sound/soc/codecs/wm8980.c +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/sound/soc/codecs/wm8980.c +@@ -0,0 +1,991 @@ ++/* ++ * wm8980.c -- WM8980 ALSA Soc Audio driver ++ * ++ * Copyright 2006 Wolfson Microelectronics PLC. ++ * ++ * Authors: ++ * Mike Arthur <linux@wolfsonmicro.com> ++ * ++ * This program is free software; you can redistribute it and/or modify ++ * it under the terms of the GNU General Public License version 2 as ++ * published by the Free Software Foundation. ++ */ ++ ++#include <linux/module.h> ++#include <linux/moduleparam.h> ++#include <linux/version.h> ++#include <linux/kernel.h> ++#include <linux/init.h> ++#include <linux/delay.h> ++#include <linux/pm.h> ++#include <linux/i2c.h> ++#include <linux/platform_device.h> ++#include <sound/driver.h> ++#include <sound/core.h> ++#include <sound/pcm.h> ++#include <sound/pcm_params.h> ++#include <sound/soc.h> ++#include <sound/soc-dapm.h> ++#include <sound/initval.h> ++ ++#include "wm8980.h" ++ ++#define AUDIO_NAME "wm8980" ++#define WM8980_VERSION "0.2" ++ ++/* ++ * Debug ++ */ ++ ++#define WM8980_DEBUG 0 ++ ++#ifdef WM8980_DEBUG ++#define dbg(format, arg...) \ ++ printk(KERN_DEBUG AUDIO_NAME ": " format "\n" , ## arg) ++#else ++#define dbg(format, arg...) do {} while (0) ++#endif ++#define err(format, arg...) \ ++ printk(KERN_ERR AUDIO_NAME ": " format "\n" , ## arg) ++#define info(format, arg...) \ ++ printk(KERN_INFO AUDIO_NAME ": " format "\n" , ## arg) ++#define warn(format, arg...) \ ++ printk(KERN_WARNING AUDIO_NAME ": " format "\n" , ## arg) ++ ++struct snd_soc_codec_device soc_codec_dev_wm8980; ++ ++/* ++ * wm8980 register cache ++ * We can't read the WM8980 register space when we are ++ * using 2 wire for device control, so we cache them instead. ++ */ ++static const u16 wm8980_reg[WM8980_CACHEREGNUM] = { ++ 0x0000, 0x0000, 0x0000, 0x0000, ++ 0x0050, 0x0000, 0x0140, 0x0000, ++ 0x0000, 0x0000, 0x0000, 0x00ff, ++ 0x00ff, 0x0000, 0x0100, 0x00ff, ++ 0x00ff, 0x0000, 0x012c, 0x002c, ++ 0x002c, 0x002c, 0x002c, 0x0000, ++ 0x0032, 0x0000, 0x0000, 0x0000, ++ 0x0000, 0x0000, 0x0000, 0x0000, ++ 0x0038, 0x000b, 0x0032, 0x0000, ++ 0x0008, 0x000c, 0x0093, 0x00e9, ++ 0x0000, 0x0000, 0x0000, 0x0000, ++ 0x0033, 0x0010, 0x0010, 0x0100, ++ 0x0100, 0x0002, 0x0001, 0x0001, ++ 0x0039, 0x0039, 0x0039, 0x0039, ++ 0x0001, 0x0001, ++}; ++ ++#define WM8980_DAIFMT \ ++ (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_RIGHT_J | \ ++ SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF | \ ++ SND_SOC_DAIFMT_NB_IF | SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_IB_IF) ++ ++#define WM8980_DIR \ ++ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE) ++ ++#define WM8980_RATES \ ++ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ ++ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ ++ SNDRV_PCM_RATE_48000) ++ ++#define WM8980_PCM_FORMATS \ ++ (SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S20_3LE | \ ++ SNDRV_PCM_FORMAT_S24_3LE | SNDRV_PCM_FORMAT_S24_LE | \ ++ SNDRV_PCM_FORMAT_S32_LE) ++ ++#define WM8980_BCLK \ ++ (SND_SOC_FSBD(1) | SND_SOC_FSBD(2) | SND_SOC_FSBD(4) | SND_SOC_FSBD(8) |\ ++ SND_SOC_FSBD(16) | SND_SOC_FSBD(32)) ++ ++static struct snd_soc_dai_mode wm8980_modes[] = { ++ /* codec frame and clock master modes */ ++ { ++ .fmt = WM8980_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8980_PCM_FORMATS, ++ .pcmrate = WM8980_RATES, ++ .pcmdir = WM8980_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 256, ++ .bfs = WM8980_BCLK, ++ }, ++ ++ /* codec frame and clock slave modes */ ++ { ++ .fmt = WM8980_DAIFMT | SND_SOC_DAIFMT_CBS_CFS, ++ .pcmfmt = WM8980_PCM_FORMATS, ++ .pcmrate = WM8980_RATES, ++ .pcmdir = WM8980_DIR, ++ .fs = SND_SOC_FS_ALL, ++ .bfs = SND_SOC_FSB_ALL, ++ }, ++}; ++ ++/* ++ * read wm8980 register cache ++ */ ++static inline unsigned int wm8980_read_reg_cache(struct snd_soc_codec *codec, ++ unsigned int reg) ++{ ++ u16 *cache = codec->reg_cache; ++ if (reg == WM8980_RESET) ++ return 0; ++ if (reg >= WM8980_CACHEREGNUM) ++ return -1; ++ return cache[reg]; ++} ++ ++/* ++ * write wm8980 register cache ++ */ ++static inline void wm8980_write_reg_cache(struct snd_soc_codec *codec, ++ u16 reg, unsigned int value) ++{ ++ u16 *cache = codec->reg_cache; ++ if (reg >= WM8980_CACHEREGNUM) ++ return; ++ cache[reg] = value; ++} ++ ++/* ++ * write to the WM8980 register space ++ */ ++static int wm8980_write(struct snd_soc_codec *codec, unsigned int reg, ++ unsigned int value) ++{ ++ u8 data[2]; ++ ++ /* data is ++ * D15..D9 WM8980 register offset ++ * D8...D0 register data ++ */ ++ data[0] = (reg << 1) | ((value >> 8) & 0x0001); ++ data[1] = value & 0x00ff; ++ ++ wm8980_write_reg_cache (codec, reg, value); ++ if (codec->hw_write(codec->control_data, data, 2) == 2) ++ return 0; ++ else ++ return -1; ++} ++ ++#define wm8980_reset(c) wm8980_write(c, WM8980_RESET, 0) ++ ++static const char *wm8980_companding[] = {"Off", "NC", "u-law", "A-law" }; ++static const char *wm8980_deemp[] = {"None", "32kHz", "44.1kHz", "48kHz" }; ++static const char *wm8980_eqmode[] = {"Capture", "Playback" }; ++static const char *wm8980_bw[] = {"Narrow", "Wide" }; ++static const char *wm8980_eq1[] = {"80Hz", "105Hz", "135Hz", "175Hz" }; ++static const char *wm8980_eq2[] = {"230Hz", "300Hz", "385Hz", "500Hz" }; ++static const char *wm8980_eq3[] = {"650Hz", "850Hz", "1.1kHz", "1.4kHz" }; ++static const char *wm8980_eq4[] = {"1.8kHz", "2.4kHz", "3.2kHz", "4.1kHz" }; ++static const char *wm8980_eq5[] = {"5.3kHz", "6.9kHz", "9kHz", "11.7kHz" }; ++static const char *wm8980_alc[] = ++ {"ALC both on", "ALC left only", "ALC right only", "Limiter" }; ++ ++static const struct soc_enum wm8980_enum[] = { ++ SOC_ENUM_SINGLE(WM8980_COMP, 1, 4, wm8980_companding), /* adc */ ++ SOC_ENUM_SINGLE(WM8980_COMP, 3, 4, wm8980_companding), /* dac */ ++ SOC_ENUM_SINGLE(WM8980_DAC, 4, 4, wm8980_deemp), ++ SOC_ENUM_SINGLE(WM8980_EQ1, 8, 2, wm8980_eqmode), ++ ++ SOC_ENUM_SINGLE(WM8980_EQ1, 5, 4, wm8980_eq1), ++ SOC_ENUM_SINGLE(WM8980_EQ2, 8, 2, wm8980_bw), ++ SOC_ENUM_SINGLE(WM8980_EQ2, 5, 4, wm8980_eq2), ++ SOC_ENUM_SINGLE(WM8980_EQ3, 8, 2, wm8980_bw), ++ ++ SOC_ENUM_SINGLE(WM8980_EQ3, 5, 4, wm8980_eq3), ++ SOC_ENUM_SINGLE(WM8980_EQ4, 8, 2, wm8980_bw), ++ SOC_ENUM_SINGLE(WM8980_EQ4, 5, 4, wm8980_eq4), ++ SOC_ENUM_SINGLE(WM8980_EQ5, 8, 2, wm8980_bw), ++ ++ SOC_ENUM_SINGLE(WM8980_EQ5, 5, 4, wm8980_eq5), ++ SOC_ENUM_SINGLE(WM8980_ALC3, 8, 2, wm8980_alc), ++}; ++ ++static const struct snd_kcontrol_new wm8980_snd_controls[] = { ++SOC_SINGLE("Digital Loopback Switch", WM8980_COMP, 0, 1, 0), ++ ++SOC_ENUM("ADC Companding", wm8980_enum[0]), ++SOC_ENUM("DAC Companding", wm8980_enum[1]), ++ ++SOC_SINGLE("Jack Detection Enable", WM8980_JACK1, 6, 1, 0), ++ ++SOC_SINGLE("DAC Right Inversion Switch", WM8980_DAC, 1, 1, 0), ++SOC_SINGLE("DAC Left Inversion Switch", WM8980_DAC, 0, 1, 0), ++ ++SOC_SINGLE("Left Playback Volume", WM8980_DACVOLL, 0, 127, 0), ++SOC_SINGLE("Right Playback Volume", WM8980_DACVOLR, 0, 127, 0), ++ ++SOC_SINGLE("High Pass Filter Switch", WM8980_ADC, 8, 1, 0), ++SOC_SINGLE("High Pass Filter Switch", WM8980_ADC, 8, 1, 0), ++SOC_SINGLE("High Pass Cut Off", WM8980_ADC, 4, 7, 0), ++SOC_SINGLE("Right ADC Inversion Switch", WM8980_ADC, 1, 1, 0), ++SOC_SINGLE("Left ADC Inversion Switch", WM8980_ADC, 0, 1, 0), ++ ++SOC_SINGLE("Left Capture Volume", WM8980_ADCVOLL, 0, 127, 0), ++SOC_SINGLE("Right Capture Volume", WM8980_ADCVOLR, 0, 127, 0), ++ ++SOC_ENUM("Equaliser Function", wm8980_enum[3]), ++SOC_ENUM("EQ1 Cut Off", wm8980_enum[4]), ++SOC_SINGLE("EQ1 Volume", WM8980_EQ1, 0, 31, 1), ++ ++SOC_ENUM("Equaliser EQ2 Bandwith", wm8980_enum[5]), ++SOC_ENUM("EQ2 Cut Off", wm8980_enum[6]), ++SOC_SINGLE("EQ2 Volume", WM8980_EQ2, 0, 31, 1), ++ ++SOC_ENUM("Equaliser EQ3 Bandwith", wm8980_enum[7]), ++SOC_ENUM("EQ3 Cut Off", wm8980_enum[8]), ++SOC_SINGLE("EQ3 Volume", WM8980_EQ3, 0, 31, 1), ++ ++SOC_ENUM("Equaliser EQ4 Bandwith", wm8980_enum[9]), ++SOC_ENUM("EQ4 Cut Off", wm8980_enum[10]), ++SOC_SINGLE("EQ4 Volume", WM8980_EQ4, 0, 31, 1), ++ ++SOC_ENUM("Equaliser EQ5 Bandwith", wm8980_enum[11]), ++SOC_ENUM("EQ5 Cut Off", wm8980_enum[12]), ++SOC_SINGLE("EQ5 Volume", WM8980_EQ5, 0, 31, 1), ++ ++SOC_SINGLE("DAC Playback Limiter Switch", WM8980_DACLIM1, 8, 1, 0), ++SOC_SINGLE("DAC Playback Limiter Decay", WM8980_DACLIM1, 4, 15, 0), ++SOC_SINGLE("DAC Playback Limiter Attack", WM8980_DACLIM1, 0, 15, 0), ++ ++SOC_SINGLE("DAC Playback Limiter Threshold", WM8980_DACLIM2, 4, 7, 0), ++SOC_SINGLE("DAC Playback Limiter Boost", WM8980_DACLIM2, 0, 15, 0), ++ ++SOC_SINGLE("ALC Enable Switch", WM8980_ALC1, 8, 1, 0), ++SOC_SINGLE("ALC Capture Max Gain", WM8980_ALC1, 3, 7, 0), ++SOC_SINGLE("ALC Capture Min Gain", WM8980_ALC1, 0, 7, 0), ++ ++SOC_SINGLE("ALC Capture ZC Switch", WM8980_ALC2, 8, 1, 0), ++SOC_SINGLE("ALC Capture Hold", WM8980_ALC2, 4, 7, 0), ++SOC_SINGLE("ALC Capture Target", WM8980_ALC2, 0, 15, 0), ++ ++SOC_ENUM("ALC Capture Mode", wm8980_enum[13]), ++SOC_SINGLE("ALC Capture Decay", WM8980_ALC3, 4, 15, 0), ++SOC_SINGLE("ALC Capture Attack", WM8980_ALC3, 0, 15, 0), ++ ++SOC_SINGLE("ALC Capture Noise Gate Switch", WM8980_NGATE, 3, 1, 0), ++SOC_SINGLE("ALC Capture Noise Gate Threshold", WM8980_NGATE, 0, 7, 0), ++ ++SOC_SINGLE("Left Capture PGA ZC Switch", WM8980_INPPGAL, 7, 1, 0), ++SOC_SINGLE("Left Capture PGA Volume", WM8980_INPPGAL, 0, 63, 0), ++ ++SOC_SINGLE("Right Capture PGA ZC Switch", WM8980_INPPGAR, 7, 1, 0), ++SOC_SINGLE("Right Capture PGA Volume", WM8980_INPPGAR, 0, 63, 0), ++ ++SOC_SINGLE("Left Headphone Playback ZC Switch", WM8980_HPVOLL, 7, 1, 0), ++SOC_SINGLE("Left Headphone Playback Switch", WM8980_HPVOLL, 6, 1, 1), ++SOC_SINGLE("Left Headphone Playback Volume", WM8980_HPVOLL, 0, 63, 0), ++ ++SOC_SINGLE("Right Headphone Playback ZC Switch", WM8980_HPVOLR, 7, 1, 0), ++SOC_SINGLE("Right Headphone Playback Switch", WM8980_HPVOLR, 6, 1, 1), ++SOC_SINGLE("Right Headphone Playback Volume", WM8980_HPVOLR, 0, 63, 0), ++ ++SOC_SINGLE("Left Speaker Playback ZC Switch", WM8980_SPKVOLL, 7, 1, 0), ++SOC_SINGLE("Left Speaker Playback Switch", WM8980_SPKVOLL, 6, 1, 1), ++SOC_SINGLE("Left Speaker Playback Volume", WM8980_SPKVOLL, 0, 63, 0), ++ ++SOC_SINGLE("Right Speaker Playback ZC Switch", WM8980_SPKVOLR, 7, 1, 0), ++SOC_SINGLE("Right Speaker Playback Switch", WM8980_SPKVOLR, 6, 1, 1), ++SOC_SINGLE("Right Speaker Playback Volume", WM8980_SPKVOLR, 0, 63, 0), ++ ++SOC_DOUBLE_R("Capture Boost(+20dB)", WM8980_ADCBOOSTL, WM8980_ADCBOOSTR, ++ 8, 1, 0), ++}; ++ ++/* add non dapm controls */ ++static int wm8980_add_controls(struct snd_soc_codec *codec) ++{ ++ int err, i; ++ ++ for (i = 0; i < ARRAY_SIZE(wm8980_snd_controls); i++) { ++ err = snd_ctl_add(codec->card, ++ snd_soc_cnew(&wm8980_snd_controls[i],codec, NULL)); ++ if (err < 0) ++ return err; ++ } ++ ++ return 0; ++} ++ ++/* Left Output Mixer */ ++static const snd_kcontrol_new_t wm8980_left_mixer_controls[] = { ++SOC_DAPM_SINGLE("Right PCM Playback Switch", WM8980_OUTPUT, 6, 1, 1), ++SOC_DAPM_SINGLE("Left PCM Playback Switch", WM8980_MIXL, 0, 1, 1), ++SOC_DAPM_SINGLE("Line Bypass Switch", WM8980_MIXL, 1, 1, 0), ++SOC_DAPM_SINGLE("Aux Playback Switch", WM8980_MIXL, 5, 1, 0), ++}; ++ ++/* Right Output Mixer */ ++static const snd_kcontrol_new_t wm8980_right_mixer_controls[] = { ++SOC_DAPM_SINGLE("Left PCM Playback Switch", WM8980_OUTPUT, 5, 1, 1), ++SOC_DAPM_SINGLE("Right PCM Playback Switch", WM8980_MIXR, 0, 1, 1), ++SOC_DAPM_SINGLE("Line Bypass Switch", WM8980_MIXR, 1, 1, 0), ++SOC_DAPM_SINGLE("Aux Playback Switch", WM8980_MIXR, 5, 1, 0), ++}; ++ ++/* Left AUX Input boost vol */ ++static const snd_kcontrol_new_t wm8980_laux_boost_controls = ++SOC_DAPM_SINGLE("Left Aux Volume", WM8980_ADCBOOSTL, 0, 3, 0); ++ ++/* Right AUX Input boost vol */ ++static const snd_kcontrol_new_t wm8980_raux_boost_controls = ++SOC_DAPM_SINGLE("Right Aux Volume", WM8980_ADCBOOSTR, 0, 3, 0); ++ ++/* Left Input boost vol */ ++static const snd_kcontrol_new_t wm8980_lmic_boost_controls = ++SOC_DAPM_SINGLE("Left Input Volume", WM8980_ADCBOOSTL, 4, 3, 0); ++ ++/* Right Input boost vol */ ++static const snd_kcontrol_new_t wm8980_rmic_boost_controls = ++SOC_DAPM_SINGLE("Right Input Volume", WM8980_ADCBOOSTR, 4, 3, 0); ++ ++/* Left Aux In to PGA */ ++static const snd_kcontrol_new_t wm8980_laux_capture_boost_controls = ++SOC_DAPM_SINGLE("Left Capture Switch", WM8980_ADCBOOSTL, 8, 1, 0); ++ ++/* Right Aux In to PGA */ ++static const snd_kcontrol_new_t wm8980_raux_capture_boost_controls = ++SOC_DAPM_SINGLE("Right Capture Switch", WM8980_ADCBOOSTR, 8, 1, 0); ++ ++/* Left Input P In to PGA */ ++static const snd_kcontrol_new_t wm8980_lmicp_capture_boost_controls = ++SOC_DAPM_SINGLE("Left Input P Capture Boost Switch", WM8980_INPUT, 0, 1, 0); ++ ++/* Right Input P In to PGA */ ++static const snd_kcontrol_new_t wm8980_rmicp_capture_boost_controls = ++SOC_DAPM_SINGLE("Right Input P Capture Boost Switch", WM8980_INPUT, 4, 1, 0); ++ ++/* Left Input N In to PGA */ ++static const snd_kcontrol_new_t wm8980_lmicn_capture_boost_controls = ++SOC_DAPM_SINGLE("Left Input N Capture Boost Switch", WM8980_INPUT, 1, 1, 0); ++ ++/* Right Input N In to PGA */ ++static const snd_kcontrol_new_t wm8980_rmicn_capture_boost_controls = ++SOC_DAPM_SINGLE("Right Input N Capture Boost Switch", WM8980_INPUT, 5, 1, 0); ++ ++// TODO Widgets ++static const struct snd_soc_dapm_widget wm8980_dapm_widgets[] = { ++#if 0 ++//SND_SOC_DAPM_MUTE("Mono Mute", WM8980_MONOMIX, 6, 0), ++//SND_SOC_DAPM_MUTE("Speaker Mute", WM8980_SPKMIX, 6, 0), ++ ++SND_SOC_DAPM_MIXER("Speaker Mixer", WM8980_POWER3, 2, 0, ++ &wm8980_speaker_mixer_controls[0], ++ ARRAY_SIZE(wm8980_speaker_mixer_controls)), ++SND_SOC_DAPM_MIXER("Mono Mixer", WM8980_POWER3, 3, 0, ++ &wm8980_mono_mixer_controls[0], ++ ARRAY_SIZE(wm8980_mono_mixer_controls)), ++SND_SOC_DAPM_DAC("DAC", "HiFi Playback", WM8980_POWER3, 0, 0), ++SND_SOC_DAPM_ADC("ADC", "HiFi Capture", WM8980_POWER3, 0, 0), ++SND_SOC_DAPM_PGA("Aux Input", WM8980_POWER1, 6, 0, NULL, 0), ++SND_SOC_DAPM_PGA("SpkN Out", WM8980_POWER3, 5, 0, NULL, 0), ++SND_SOC_DAPM_PGA("SpkP Out", WM8980_POWER3, 6, 0, NULL, 0), ++SND_SOC_DAPM_PGA("Mono Out", WM8980_POWER3, 7, 0, NULL, 0), ++SND_SOC_DAPM_PGA("Mic PGA", WM8980_POWER2, 2, 0, NULL, 0), ++ ++SND_SOC_DAPM_PGA("Aux Boost", SND_SOC_NOPM, 0, 0, ++ &wm8980_aux_boost_controls, 1), ++SND_SOC_DAPM_PGA("Mic Boost", SND_SOC_NOPM, 0, 0, ++ &wm8980_mic_boost_controls, 1), ++SND_SOC_DAPM_SWITCH("Capture Boost", SND_SOC_NOPM, 0, 0, ++ &wm8980_capture_boost_controls), ++ ++SND_SOC_DAPM_MIXER("Boost Mixer", WM8980_POWER2, 4, 0, NULL, 0), ++ ++SND_SOC_DAPM_MICBIAS("Mic Bias", WM8980_POWER1, 4, 0), ++ ++SND_SOC_DAPM_INPUT("MICN"), ++SND_SOC_DAPM_INPUT("MICP"), ++SND_SOC_DAPM_INPUT("AUX"), ++SND_SOC_DAPM_OUTPUT("MONOOUT"), ++SND_SOC_DAPM_OUTPUT("SPKOUTP"), ++SND_SOC_DAPM_OUTPUT("SPKOUTN"), ++#endif ++}; ++ ++static const char *audio_map[][3] = { ++ /* Mono output mixer */ ++ {"Mono Mixer", "PCM Playback Switch", "DAC"}, ++ {"Mono Mixer", "Aux Playback Switch", "Aux Input"}, ++ {"Mono Mixer", "Line Bypass Switch", "Boost Mixer"}, ++ ++ /* Speaker output mixer */ ++ {"Speaker Mixer", "PCM Playback Switch", "DAC"}, ++ {"Speaker Mixer", "Aux Playback Switch", "Aux Input"}, ++ {"Speaker Mixer", "Line Bypass Switch", "Boost Mixer"}, ++ ++ /* Outputs */ ++ {"Mono Out", NULL, "Mono Mixer"}, ++ {"MONOOUT", NULL, "Mono Out"}, ++ {"SpkN Out", NULL, "Speaker Mixer"}, ++ {"SpkP Out", NULL, "Speaker Mixer"}, ++ {"SPKOUTN", NULL, "SpkN Out"}, ++ {"SPKOUTP", NULL, "SpkP Out"}, ++ ++ /* Boost Mixer */ ++ {"Boost Mixer", NULL, "ADC"}, ++ {"Capture Boost Switch", "Aux Capture Boost Switch", "AUX"}, ++ {"Aux Boost", "Aux Volume", "Boost Mixer"}, ++ {"Capture Boost", "Capture Switch", "Boost Mixer"}, ++ {"Mic Boost", "Mic Volume", "Boost Mixer"}, ++ ++ /* Inputs */ ++ {"MICP", NULL, "Mic Boost"}, ++ {"MICN", NULL, "Mic PGA"}, ++ {"Mic PGA", NULL, "Capture Boost"}, ++ {"AUX", NULL, "Aux Input"}, ++ ++ /* */ ++ ++ /* terminator */ ++ {NULL, NULL, NULL}, ++}; ++ ++static int wm8980_add_widgets(struct snd_soc_codec *codec) ++{ ++ int i; ++ ++ for(i = 0; i < ARRAY_SIZE(wm8980_dapm_widgets); i++) { ++ snd_soc_dapm_new_control(codec, &wm8980_dapm_widgets[i]); ++ } ++ ++ /* set up audio path map */ ++ for(i = 0; audio_map[i][0] != NULL; i++) { ++ snd_soc_dapm_connect_input(codec, audio_map[i][0], audio_map[i][1], ++ audio_map[i][2]); ++ } ++ ++ snd_soc_dapm_new_widgets(codec); ++ return 0; ++} ++ ++struct pll_ { ++ unsigned int in_hz, out_hz; ++ unsigned int pre:4; /* prescale - 1 */ ++ unsigned int n:4; ++ unsigned int k; ++}; ++ ++struct pll_ pll[] = { ++ {12000000, 11289600, 0, 7, 0x86c220}, ++ {12000000, 12288000, 0, 8, 0x3126e8}, ++ {13000000, 11289600, 0, 6, 0xf28bd4}, ++ {13000000, 12288000, 0, 7, 0x8fd525}, ++ {12288000, 11289600, 0, 7, 0x59999a}, ++ {11289600, 12288000, 0, 8, 0x80dee9}, ++ /* TODO: liam - add more entries */ ++}; ++ ++static int set_pll(struct snd_soc_codec *codec, unsigned int in, ++ unsigned int out) ++{ ++ int i; ++ u16 reg; ++ ++ if(out == 0) { ++ reg = wm8980_read_reg_cache(codec, WM8980_POWER1); ++ wm8980_write(codec, WM8980_POWER1, reg & 0x1df); ++ return 0; ++ } ++ ++ for(i = 0; i < ARRAY_SIZE(pll); i++) { ++ if (in == pll[i].in_hz && out == pll[i].out_hz) { ++ wm8980_write(codec, WM8980_PLLN, (pll[i].pre << 4) | pll[i].n); ++ wm8980_write(codec, WM8980_PLLK1, pll[i].k >> 18); ++ wm8980_write(codec, WM8980_PLLK1, (pll[i].k >> 9) && 0x1ff); ++ wm8980_write(codec, WM8980_PLLK1, pll[i].k && 0x1ff); ++ reg = wm8980_read_reg_cache(codec, WM8980_POWER1); ++ wm8980_write(codec, WM8980_POWER1, reg | 0x020); ++ return 0; ++ } ++ } ++ return -EINVAL; ++} ++ ++/* mclk dividers * 2 */ ++static unsigned char mclk_div[] = {2, 3, 4, 6, 8, 12, 16, 24}; ++ ++/* we need 256FS to drive the DAC's and ADC's */ ++static unsigned int wm8980_config_sysclk(struct snd_soc_codec_dai *dai, ++ struct snd_soc_clock_info *info, unsigned int clk) ++{ ++ int i, j, best_clk = info->fs * info->rate; ++ ++ /* can we run at this clk without the PLL ? */ ++ for (i = 0; i < ARRAY_SIZE(mclk_div); i++) { ++ if ((best_clk >> 1) * mclk_div[i] == clk) { ++ dai->pll_in = 0; ++ dai->clk_div = mclk_div[i]; ++ dai->mclk = best_clk; ++ return dai->mclk; ++ } ++ } ++ ++ /* now check for PLL support */ ++ for (i = 0; i < ARRAY_SIZE(pll); i++) { ++ if (pll[i].in_hz == clk) { ++ for (j = 0; j < ARRAY_SIZE(mclk_div); j++) { ++ if (pll[i].out_hz == mclk_div[j] * (best_clk >> 1)) { ++ dai->pll_in = clk; ++ dai->pll_out = pll[i].out_hz; ++ dai->clk_div = mclk_div[j]; ++ dai->mclk = best_clk; ++ return dai->mclk; ++ } ++ } ++ } ++ } ++ ++ /* this clk is not supported */ ++ return 0; ++} ++ ++static int wm8980_pcm_prepare(snd_pcm_substream_t *substream) ++{ ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ struct snd_soc_device *socdev = rtd->socdev; ++ struct snd_soc_codec *codec = socdev->codec; ++ struct snd_soc_codec_dai *dai = rtd->codec_dai; ++ u16 iface = 0, bfs, clk = 0, adn; ++ int fs = 48000 << 7, i; ++ ++ bfs = SND_SOC_FSBD_REAL(rtd->codec_dai->dai_runtime.bfs); ++ switch (bfs) { ++ case 2: ++ clk |= 0x1 << 2; ++ break; ++ case 4: ++ clk |= 0x2 << 2; ++ break; ++ case 8: ++ clk |= 0x3 << 2; ++ break; ++ case 16: ++ clk |= 0x4 << 2; ++ break; ++ case 32: ++ clk |= 0x5 << 2; ++ break; ++ } ++ ++ /* set master/slave audio interface */ ++ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_CLOCK_MASK) { ++ case SND_SOC_DAIFMT_CBM_CFM: ++ clk |= 0x0001; ++ break; ++ case SND_SOC_DAIFMT_CBS_CFS: ++ break; ++ } ++ ++ /* interface format */ ++ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_FORMAT_MASK) { ++ case SND_SOC_DAIFMT_I2S: ++ iface |= 0x0010; ++ break; ++ case SND_SOC_DAIFMT_RIGHT_J: ++ break; ++ case SND_SOC_DAIFMT_LEFT_J: ++ iface |= 0x0008; ++ break; ++ case SND_SOC_DAIFMT_DSP_A: ++ iface |= 0x00018; ++ break; ++ } ++ ++ /* bit size */ ++ switch (rtd->codec_dai->dai_runtime.pcmfmt) { ++ case SNDRV_PCM_FMTBIT_S16_LE: ++ break; ++ case SNDRV_PCM_FMTBIT_S20_3LE: ++ iface |= 0x0020; ++ break; ++ case SNDRV_PCM_FMTBIT_S24_LE: ++ iface |= 0x0040; ++ break; ++ case SNDRV_PCM_FMTBIT_S32_LE: ++ iface |= 0x0060; ++ break; ++ } ++ ++ /* clock inversion */ ++ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_INV_MASK) { ++ case SND_SOC_DAIFMT_NB_NF: ++ break; ++ case SND_SOC_DAIFMT_IB_IF: ++ iface |= 0x0180; ++ break; ++ case SND_SOC_DAIFMT_IB_NF: ++ iface |= 0x0100; ++ break; ++ case SND_SOC_DAIFMT_NB_IF: ++ iface |= 0x0080; ++ break; ++ } ++ ++ /* filter coefficient */ ++ adn = wm8980_read_reg_cache(codec, WM8980_ADD) & 0x1f1; ++ switch (rtd->codec_dai->dai_runtime.pcmrate) { ++ case SNDRV_PCM_RATE_8000: ++ adn |= 0x5 << 1; ++ fs = 8000 << 7; ++ break; ++ case SNDRV_PCM_RATE_11025: ++ adn |= 0x4 << 1; ++ fs = 11025 << 7; ++ break; ++ case SNDRV_PCM_RATE_16000: ++ adn |= 0x3 << 1; ++ fs = 16000 << 7; ++ break; ++ case SNDRV_PCM_RATE_22050: ++ adn |= 0x2 << 1; ++ fs = 22050 << 7; ++ break; ++ case SNDRV_PCM_RATE_32000: ++ adn |= 0x1 << 1; ++ fs = 32000 << 7; ++ break; ++ case SNDRV_PCM_RATE_44100: ++ fs = 44100 << 7; ++ break; ++ } ++ ++ /* do we need to enable the PLL */ ++ if(dai->pll_in) ++ set_pll(codec, dai->pll_in, dai->pll_out); ++ ++ /* divide the clock to 256 fs */ ++ for(i = 0; i < ARRAY_SIZE(mclk_div); i++) { ++ if (dai->clk_div == mclk_div[i]) { ++ clk |= i << 5; ++ clk &= 0xff; ++ goto set; ++ } ++ } ++ ++set: ++ /* set iface */ ++ wm8980_write(codec, WM8980_IFACE, iface); ++ wm8980_write(codec, WM8980_CLOCK, clk); ++ ++ return 0; ++} ++ ++static int wm8980_hw_free(struct snd_pcm_substream *substream) ++{ ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ struct snd_soc_device *socdev = rtd->socdev; ++ struct snd_soc_codec *codec = socdev->codec; ++ set_pll(codec, 0, 0); ++ return 0; ++} ++ ++static int wm8980_mute(struct snd_soc_codec *codec, ++ struct snd_soc_codec_dai *dai, int mute) ++{ ++ u16 mute_reg = wm8980_read_reg_cache(codec, WM8980_DAC) & 0xffbf; ++ if(mute) ++ wm8980_write(codec, WM8980_DAC, mute_reg | 0x40); ++ else ++ wm8980_write(codec, WM8980_DAC, mute_reg); ++ ++ return 0; ++} ++ ++/* TODO: liam need to make this lower power with dapm */ ++static int wm8980_dapm_event(struct snd_soc_codec *codec, int event) ++{ ++ ++ switch (event) { ++ case SNDRV_CTL_POWER_D0: /* full On */ ++ /* vref/mid, clk and osc on, dac unmute, active */ ++ wm8980_write(codec, WM8980_POWER1, 0x1ff); ++ wm8980_write(codec, WM8980_POWER2, 0x1ff); ++ wm8980_write(codec, WM8980_POWER3, 0x1ff); ++ break; ++ case SNDRV_CTL_POWER_D1: /* partial On */ ++ case SNDRV_CTL_POWER_D2: /* partial On */ ++ break; ++ case SNDRV_CTL_POWER_D3hot: /* Off, with power */ ++ /* everything off except vref/vmid, dac mute, inactive */ ++ ++ break; ++ case SNDRV_CTL_POWER_D3cold: /* Off, without power */ ++ /* everything off, dac mute, inactive */ ++ wm8980_write(codec, WM8980_POWER1, 0x0); ++ wm8980_write(codec, WM8980_POWER2, 0x0); ++ wm8980_write(codec, WM8980_POWER3, 0x0); ++ break; ++ } ++ codec->dapm_state = event; ++ return 0; ++} ++ ++struct snd_soc_codec_dai wm8980_dai = { ++ .name = "WM8980 HiFi", ++ .playback = { ++ .stream_name = "Playback", ++ .channels_min = 1, ++ .channels_max = 2, ++ }, ++ .capture = { ++ .stream_name = "Capture", ++ .channels_min = 1, ++ .channels_max = 2, ++ }, ++ .config_sysclk = wm8980_config_sysclk, ++ .digital_mute = wm8980_mute, ++ .ops = { ++ .prepare = wm8980_pcm_prepare, ++ .hw_free = wm8980_hw_free, ++ }, ++ .caps = { ++ .num_modes = ARRAY_SIZE(wm8980_modes), ++ .mode = wm8980_modes, ++ }, ++}; ++EXPORT_SYMBOL_GPL(wm8980_dai); ++ ++static int wm8980_suspend(struct platform_device *pdev, pm_message_t state) ++{ ++ struct snd_soc_device *socdev = platform_get_drvdata(pdev); ++ struct snd_soc_codec *codec = socdev->codec; ++ ++ wm8980_dapm_event(codec, SNDRV_CTL_POWER_D3cold); ++ return 0; ++} ++ ++static int wm8980_resume(struct platform_device *pdev) ++{ ++ struct snd_soc_device *socdev = platform_get_drvdata(pdev); ++ struct snd_soc_codec *codec = socdev->codec; ++ int i; ++ u8 data[2]; ++ u16 *cache = codec->reg_cache; ++ ++ /* Sync reg_cache with the hardware */ ++ for (i = 0; i < ARRAY_SIZE(wm8980_reg); i++) { ++ data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); ++ data[1] = cache[i] & 0x00ff; ++ codec->hw_write(codec->control_data, data, 2); ++ } ++ wm8980_dapm_event(codec, SNDRV_CTL_POWER_D3hot); ++ wm8980_dapm_event(codec, codec->suspend_dapm_state); ++ return 0; ++} ++ ++/* ++ * initialise the WM8980 driver ++ * register the mixer and dsp interfaces with the kernel ++ */ ++static int wm8980_init(struct snd_soc_device* socdev) ++{ ++ struct snd_soc_codec *codec = socdev->codec; ++ int ret = 0; ++ ++ codec->name = "WM8980"; ++ codec->owner = THIS_MODULE; ++ codec->read = wm8980_read_reg_cache; ++ codec->write = wm8980_write; ++ codec->dapm_event = wm8980_dapm_event; ++ codec->dai = &wm8980_dai; ++ codec->num_dai = 1; ++ codec->reg_cache_size = ARRAY_SIZE(wm8980_reg); ++ codec->reg_cache = ++ kzalloc(sizeof(u16) * ARRAY_SIZE(wm8980_reg), GFP_KERNEL); ++ if (codec->reg_cache == NULL) ++ return -ENOMEM; ++ memcpy(codec->reg_cache, wm8980_reg, ++ sizeof(u16) * ARRAY_SIZE(wm8980_reg)); ++ codec->reg_cache_size = sizeof(u16) * ARRAY_SIZE(wm8980_reg); ++ ++ wm8980_reset(codec); ++ ++ /* register pcms */ ++ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); ++ if(ret < 0) { ++ kfree(codec->reg_cache); ++ return ret; ++ } ++ ++ /* power on device */ ++ wm8980_dapm_event(codec, SNDRV_CTL_POWER_D3hot); ++ wm8980_add_controls(codec); ++ wm8980_add_widgets(codec); ++ ret = snd_soc_register_card(socdev); ++ if(ret < 0) { ++ snd_soc_free_pcms(socdev); ++ snd_soc_dapm_free(socdev); ++ } ++ ++ return ret; ++} ++ ++static struct snd_soc_device *wm8980_socdev; ++ ++#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) ++ ++/* ++ * WM8980 2 wire address is 0x1a ++ */ ++#define I2C_DRIVERID_WM8980 0xfefe /* liam - need a proper id */ ++ ++static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END }; ++ ++/* Magic definition of all other variables and things */ ++I2C_CLIENT_INSMOD; ++ ++static struct i2c_driver wm8980_i2c_driver; ++static struct i2c_client client_template; ++ ++/* If the i2c layer weren't so broken, we could pass this kind of data ++ around */ ++ ++static int wm8980_codec_probe(struct i2c_adapter *adap, int addr, int kind) ++{ ++ struct snd_soc_device *socdev = wm8980_socdev; ++ struct wm8980_setup_data *setup = socdev->codec_data; ++ struct snd_soc_codec *codec = socdev->codec; ++ struct i2c_client *i2c; ++ int ret; ++ ++ if (addr != setup->i2c_address) ++ return -ENODEV; ++ ++ client_template.adapter = adap; ++ client_template.addr = addr; ++ ++ i2c = kzalloc(sizeof(struct i2c_client), GFP_KERNEL); ++ if (i2c == NULL){ ++ kfree(codec); ++ return -ENOMEM; ++ } ++ memcpy(i2c, &client_template, sizeof(struct i2c_client)); ++ ++ i2c_set_clientdata(i2c, codec); ++ ++ codec->control_data = i2c; ++ ++ ret = i2c_attach_client(i2c); ++ if(ret < 0) { ++ err("failed to attach codec at addr %x\n", addr); ++ goto err; ++ } ++ ++ ret = wm8980_init(socdev); ++ if(ret < 0) { ++ err("failed to initialise WM8980\n"); ++ goto err; ++ } ++ return ret; ++ ++err: ++ kfree(codec); ++ kfree(i2c); ++ return ret; ++ ++} ++ ++static int wm8980_i2c_detach(struct i2c_client *client) ++{ ++ struct snd_soc_codec *codec = i2c_get_clientdata(client); ++ ++ i2c_detach_client(client); ++ ++ kfree(codec->reg_cache); ++ kfree(client); ++ ++ return 0; ++} ++ ++static int wm8980_i2c_attach(struct i2c_adapter *adap) ++{ ++ return i2c_probe(adap, &addr_data, wm8980_codec_probe); ++} ++ ++/* corgi i2c codec control layer */ ++static struct i2c_driver wm8980_i2c_driver = { ++ .driver = { ++ .name = "WM8980 I2C Codec", ++ .owner = THIS_MODULE, ++ }, ++ .id = I2C_DRIVERID_WM8980, ++ .attach_adapter = wm8980_i2c_attach, ++ .detach_client = wm8980_i2c_detach, ++ .command = NULL, ++}; ++ ++static struct i2c_client client_template = { ++ .name = "WM8980", ++ .driver = &wm8980_i2c_driver, ++}; ++#endif ++ ++static int wm8980_probe(struct platform_device *pdev) ++{ ++ struct snd_soc_device *socdev = platform_get_drvdata(pdev); ++ struct wm8980_setup_data *setup; ++ struct snd_soc_codec *codec; ++ int ret = 0; ++ ++ info("WM8980 Audio Codec %s", WM8980_VERSION); ++ ++ setup = socdev->codec_data; ++ codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); ++ if (codec == NULL) ++ return -ENOMEM; ++ ++ socdev->codec = codec; ++ mutex_init(&codec->mutex); ++ INIT_LIST_HEAD(&codec->dapm_widgets); ++ INIT_LIST_HEAD(&codec->dapm_paths); ++ ++ wm8980_socdev = socdev; ++#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) ++ if (setup->i2c_address) { ++ normal_i2c[0] = setup->i2c_address; ++ codec->hw_write = (hw_write_t)i2c_master_send; ++ ret = i2c_add_driver(&wm8980_i2c_driver); ++ if (ret != 0) ++ printk(KERN_ERR "can't add i2c driver"); ++ } ++#else ++ /* Add other interfaces here */ ++#endif ++ return ret; ++} ++ ++/* power down chip */ ++static int wm8980_remove(struct platform_device *pdev) ++{ ++ struct snd_soc_device *socdev = platform_get_drvdata(pdev); ++ struct snd_soc_codec *codec = socdev->codec; ++ ++ if (codec->control_data) ++ wm8980_dapm_event(codec, SNDRV_CTL_POWER_D3cold); ++ ++ snd_soc_free_pcms(socdev); ++ snd_soc_dapm_free(socdev); ++#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) ++ i2c_del_driver(&wm8980_i2c_driver); ++#endif ++ kfree(codec); ++ ++ return 0; ++} ++ ++struct snd_soc_codec_device soc_codec_dev_wm8980 = { ++ .probe = wm8980_probe, ++ .remove = wm8980_remove, ++ .suspend = wm8980_suspend, ++ .resume = wm8980_resume, ++}; ++ ++EXPORT_SYMBOL_GPL(soc_codec_dev_wm8980); ++ ++MODULE_DESCRIPTION("ASoC WM8980 driver"); ++MODULE_AUTHOR("Mike Arthur"); ++MODULE_LICENSE("GPL"); +Index: linux-2.6-pxa-new/sound/soc/codecs/wm8980.h +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/sound/soc/codecs/wm8980.h +@@ -0,0 +1,77 @@ ++/* ++ * wm8980.h -- WM8980 Soc Audio driver ++ * ++ * This program is free software; you can redistribute it and/or modify ++ * it under the terms of the GNU General Public License version 2 as ++ * published by the Free Software Foundation. ++ */ ++ ++#ifndef _WM8980_H ++#define _WM8980_H ++ ++/* WM8980 register space */ ++ ++#define WM8980_RESET 0x0 ++#define WM8980_POWER1 0x1 ++#define WM8980_POWER2 0x2 ++#define WM8980_POWER3 0x3 ++#define WM8980_IFACE 0x4 ++#define WM8980_COMP 0x5 ++#define WM8980_CLOCK 0x6 ++#define WM8980_ADD 0x7 ++#define WM8980_GPIO 0x8 ++#define WM8980_JACK1 0x9 ++#define WM8980_DAC 0xa ++#define WM8980_DACVOLL 0xb ++#define WM8980_DACVOLR 0xc ++#define WM8980_JACK2 0xd ++#define WM8980_ADC 0xe ++#define WM8980_ADCVOLL 0xf ++#define WM8980_ADCVOLR 0x10 ++#define WM8980_EQ1 0x12 ++#define WM8980_EQ2 0x13 ++#define WM8980_EQ3 0x14 ++#define WM8980_EQ4 0x15 ++#define WM8980_EQ5 0x16 ++#define WM8980_DACLIM1 0x18 ++#define WM8980_DACLIM2 0x19 ++#define WM8980_NOTCH1 0x1b ++#define WM8980_NOTCH2 0x1c ++#define WM8980_NOTCH3 0x1d ++#define WM8980_NOTCH4 0x1e ++#define WM8980_ALC1 0x20 ++#define WM8980_ALC2 0x21 ++#define WM8980_ALC3 0x22 ++#define WM8980_NGATE 0x23 ++#define WM8980_PLLN 0x24 ++#define WM8980_PLLK1 0x25 ++#define WM8980_PLLK2 0x26 ++#define WM8980_PLLK3 0x27 ++#define WM8980_VIDEO 0x28 ++#define WM8980_3D 0x29 ++#define WM8980_BEEP 0x2b ++#define WM8980_INPUT 0x2c ++#define WM8980_INPPGAL 0x2d ++#define WM8980_INPPGAR 0x2e ++#define WM8980_ADCBOOSTL 0x2f ++#define WM8980_ADCBOOSTR 0x30 ++#define WM8980_OUTPUT 0x31 ++#define WM8980_MIXL 0x32 ++#define WM8980_MIXR 0x33 ++#define WM8980_HPVOLL 0x34 ++#define WM8980_HPVOLR 0x35 ++#define WM8980_SPKVOLL 0x36 ++#define WM8980_SPKVOLR 0x37 ++#define WM8980_OUT3MIX 0x38 ++#define WM8980_MONOMIX 0x39 ++ ++#define WM8980_CACHEREGNUM 58 ++ ++struct wm8980_setup_data { ++ unsigned short i2c_address; ++}; ++ ++extern struct snd_soc_codec_dai wm8980_dai; ++extern struct snd_soc_codec_device soc_codec_dev_wm8980; ++ ++#endif +Index: linux-2.6-pxa-new/sound/soc/at91/eti_b1_wm8731.c +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/sound/soc/at91/eti_b1_wm8731.c +@@ -0,0 +1,230 @@ ++/* ++ * eti_b1_wm8731 -- SoC audio for Endrelia ETI_B1. ++ * ++ * Author: Frank Mandarino <fmandarino@endrelia.com> ++ * Endrelia Technologies Inc. ++ * Created: Mar 29, 2006 ++ * ++ * Based on corgi.c by: ++ * ++ * Copyright 2005 Wolfson Microelectronics PLC. ++ * Copyright 2005 Openedhand Ltd. ++ * ++ * Authors: Liam Girdwood <liam.girdwood@wolfsonmicro.com> ++ * Richard Purdie <richard@openedhand.com> ++ * ++ * This program is free software; you can redistribute it and/or modify it ++ * under the terms of the GNU General Public License as published by the ++ * Free Software Foundation; either version 2 of the License, or (at your ++ * option) any later version. ++ * ++ * Revision history ++ * 30th Nov 2005 Initial version. ++ * ++ */ ++ ++#include <linux/module.h> ++#include <linux/moduleparam.h> ++#include <linux/version.h> ++#include <linux/kernel.h> ++#include <linux/clk.h> ++#include <linux/timer.h> ++#include <linux/interrupt.h> ++#include <linux/platform_device.h> ++#include <sound/driver.h> ++#include <sound/core.h> ++#include <sound/pcm.h> ++#include <sound/soc.h> ++#include <sound/soc-dapm.h> ++ ++#include <asm/arch/at91rm9200.h> ++#include <asm/arch/gpio.h> ++#include <asm/arch/hardware.h> ++ ++#include "../codecs/wm8731.h" ++#include "at91rm9200-pcm.h" ++ ++#if 0 ++#define DBG(x...) printk(KERN_INFO "eti_b1_wm8731:" x) ++#else ++#define DBG(x...) ++#endif ++ ++static struct clk *pck1_clk; ++static struct clk *pllb_clk; ++ ++static int eti_b1_startup(snd_pcm_substream_t *substream) ++{ ++ /* Start PCK1 clock. */ ++ clk_enable(pck1_clk); ++ DBG("pck1 started\n"); ++ ++ return 0; ++} ++ ++static void eti_b1_shutdown(snd_pcm_substream_t *substream) ++{ ++ /* Stop PCK1 clock. */ ++ clk_disable(pck1_clk); ++ DBG("pck1 stopped\n"); ++} ++ ++static struct snd_soc_ops eti_b1_ops = { ++ .startup = eti_b1_startup, ++ .shutdown = eti_b1_shutdown, ++}; ++ ++ ++static const struct snd_soc_dapm_widget eti_b1_dapm_widgets[] = { ++ SND_SOC_DAPM_MIC("Int Mic", NULL), ++ SND_SOC_DAPM_SPK("Ext Spk", NULL), ++}; ++ ++static const char *intercon[][3] = { ++ ++ /* speaker connected to LHPOUT */ ++ {"Ext Spk", NULL, "LHPOUT"}, ++ ++ /* mic is connected to Mic Jack, with WM8731 Mic Bias */ ++ {"MICIN", NULL, "Mic Bias"}, ++ {"Mic Bias", NULL, "Int Mic"}, ++ ++ /* terminator */ ++ {NULL, NULL, NULL}, ++}; ++ ++/* ++ * Logic for a wm8731 as connected on a Endrelia ETI-B1 board. ++ */ ++static int eti_b1_wm8731_init(struct snd_soc_codec *codec) ++{ ++ int i; ++ ++ DBG("eti_b1_wm8731_init() called\n"); ++ ++ /* Add specific widgets */ ++ for(i = 0; i < ARRAY_SIZE(eti_b1_dapm_widgets); i++) { ++ snd_soc_dapm_new_control(codec, &eti_b1_dapm_widgets[i]); ++ } ++ ++ /* Set up specific audio path interconnects */ ++ for(i = 0; intercon[i][0] != NULL; i++) { ++ snd_soc_dapm_connect_input(codec, intercon[i][0], ++ intercon[i][1], intercon[i][2]); ++ } ++ ++ /* not connected */ ++ snd_soc_dapm_set_endpoint(codec, "RLINEIN", 0); ++ snd_soc_dapm_set_endpoint(codec, "LLINEIN", 0); ++ ++ /* always connected */ ++ snd_soc_dapm_set_endpoint(codec, "Int Mic", 1); ++ snd_soc_dapm_set_endpoint(codec, "Ext Spk", 1); ++ ++ snd_soc_dapm_sync_endpoints(codec); ++ ++ return 0; ++} ++ ++unsigned int eti_b1_config_sysclk(struct snd_soc_pcm_runtime *rtd, ++ struct snd_soc_clock_info *info) ++{ ++ if(info->bclk_master & SND_SOC_DAIFMT_CBS_CFS) { ++ return rtd->codec_dai->config_sysclk(rtd->codec_dai, info, 12000000); ++ } ++ return 0; ++} ++ ++static struct snd_soc_dai_link eti_b1_dai = { ++ .name = "WM8731", ++ .stream_name = "WM8731", ++ .cpu_dai = &at91rm9200_i2s_dai[1], ++ .codec_dai = &wm8731_dai, ++ .init = eti_b1_wm8731_init, ++ .config_sysclk = eti_b1_config_sysclk, ++}; ++ ++static struct snd_soc_machine snd_soc_machine_eti_b1 = { ++ .name = "ETI_B1", ++ .dai_link = &eti_b1_dai, ++ .num_links = 1, ++ .ops = &eti_b1_ops, ++}; ++ ++static struct wm8731_setup_data eti_b1_wm8731_setup = { ++ .i2c_address = 0x1a, ++}; ++ ++static struct snd_soc_device eti_b1_snd_devdata = { ++ .machine = &snd_soc_machine_eti_b1, ++ .platform = &at91rm9200_soc_platform, ++ .codec_dev = &soc_codec_dev_wm8731, ++ .codec_data = &eti_b1_wm8731_setup, ++}; ++ ++static struct platform_device *eti_b1_snd_device; ++ ++static int __init eti_b1_init(void) ++{ ++ int ret; ++ u32 ssc_pio_lines; ++ ++ eti_b1_snd_device = platform_device_alloc("soc-audio", -1); ++ if (!eti_b1_snd_device) ++ return -ENOMEM; ++ ++ platform_set_drvdata(eti_b1_snd_device, &eti_b1_snd_devdata); ++ eti_b1_snd_devdata.dev = &eti_b1_snd_device->dev; ++ ++ ret = platform_device_add(eti_b1_snd_device); ++ if (ret) { ++ platform_device_put(eti_b1_snd_device); ++ return ret; ++ } ++ ++ ssc_pio_lines = AT91_PB6_TF1 | AT91_PB7_TK1 | AT91_PB8_TD1 ++ | AT91_PB9_RD1 /* | AT91_PB10_RK1 | AT91_PB11_RF1 */; ++ ++ /* Reset all PIO registers and assign lines to peripheral A */ ++ at91_sys_write(AT91_PIOB + PIO_PDR, ssc_pio_lines); ++ at91_sys_write(AT91_PIOB + PIO_ODR, ssc_pio_lines); ++ at91_sys_write(AT91_PIOB + PIO_IFDR, ssc_pio_lines); ++ at91_sys_write(AT91_PIOB + PIO_CODR, ssc_pio_lines); ++ at91_sys_write(AT91_PIOB + PIO_IDR, ssc_pio_lines); ++ at91_sys_write(AT91_PIOB + PIO_MDDR, ssc_pio_lines); ++ at91_sys_write(AT91_PIOB + PIO_PUDR, ssc_pio_lines); ++ at91_sys_write(AT91_PIOB + PIO_ASR, ssc_pio_lines); ++ at91_sys_write(AT91_PIOB + PIO_OWDR, ssc_pio_lines); ++ ++ /* ++ * Set PCK1 parent to PLLB and its rate to 12 Mhz. ++ */ ++ pllb_clk = clk_get(NULL, "pllb"); ++ pck1_clk = clk_get(NULL, "pck1"); ++ ++ clk_set_parent(pck1_clk, pllb_clk); ++ clk_set_rate(pck1_clk, 12000000); ++ ++ DBG("MCLK rate %luHz\n", clk_get_rate(pck1_clk)); ++ ++ /* assign the GPIO pin to PCK1 */ ++ at91_set_B_periph(AT91_PIN_PA24, 0); ++ ++ return ret; ++} ++ ++static void __exit eti_b1_exit(void) ++{ ++ clk_put(pck1_clk); ++ clk_put(pllb_clk); ++ ++ platform_device_unregister(eti_b1_snd_device); ++} ++ ++module_init(eti_b1_init); ++module_exit(eti_b1_exit); ++ ++/* Module information */ ++MODULE_AUTHOR("Frank Mandarino <fmandarino@endrelia.com>"); ++MODULE_DESCRIPTION("ALSA SoC ETI-B1-WM8731"); ++MODULE_LICENSE("GPL"); +Index: linux-2.6-pxa-new/sound/soc/codecs/wm8510.c +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/sound/soc/codecs/wm8510.c +@@ -0,0 +1,895 @@ ++/* ++ * wm8510.c -- WM8510 ALSA Soc Audio driver ++ * ++ * Copyright 2006 Wolfson Microelectronics PLC. ++ * ++ * Author: Liam Girdwood <liam.girdwood@wolfsonmicro.com> ++ * ++ * This program is free software; you can redistribute it and/or modify ++ * it under the terms of the GNU General Public License version 2 as ++ * published by the Free Software Foundation. ++ */ ++ ++#include <linux/module.h> ++#include <linux/moduleparam.h> ++#include <linux/version.h> ++#include <linux/kernel.h> ++#include <linux/init.h> ++#include <linux/delay.h> ++#include <linux/pm.h> ++#include <linux/i2c.h> ++#include <linux/platform_device.h> ++#include <sound/driver.h> ++#include <sound/core.h> ++#include <sound/pcm.h> ++#include <sound/pcm_params.h> ++#include <sound/soc.h> ++#include <sound/soc-dapm.h> ++#include <sound/initval.h> ++ ++#include "wm8510.h" ++ ++#define AUDIO_NAME "wm8510" ++#define WM8510_VERSION "0.5" ++ ++/* ++ * Debug ++ */ ++ ++#define WM8510_DEBUG 0 ++ ++#ifdef WM8510_DEBUG ++#define dbg(format, arg...) \ ++ printk(KERN_DEBUG AUDIO_NAME ": " format "\n" , ## arg) ++#else ++#define dbg(format, arg...) do {} while (0) ++#endif ++#define err(format, arg...) \ ++ printk(KERN_ERR AUDIO_NAME ": " format "\n" , ## arg) ++#define info(format, arg...) \ ++ printk(KERN_INFO AUDIO_NAME ": " format "\n" , ## arg) ++#define warn(format, arg...) \ ++ printk(KERN_WARNING AUDIO_NAME ": " format "\n" , ## arg) ++ ++struct snd_soc_codec_device soc_codec_dev_wm8510; ++ ++/* ++ * wm8510 register cache ++ * We can't read the WM8510 register space when we are ++ * using 2 wire for device control, so we cache them instead. ++ */ ++static const u16 wm8510_reg[WM8510_CACHEREGNUM] = { ++ 0x0000, 0x0000, 0x0000, 0x0000, ++ 0x0050, 0x0000, 0x0140, 0x0000, ++ 0x0000, 0x0000, 0x0000, 0x00ff, ++ 0x0000, 0x0000, 0x0100, 0x00ff, ++ 0x0000, 0x0000, 0x012c, 0x002c, ++ 0x002c, 0x002c, 0x002c, 0x0000, ++ 0x0032, 0x0000, 0x0000, 0x0000, ++ 0x0000, 0x0000, 0x0000, 0x0000, ++ 0x0038, 0x000b, 0x0032, 0x0000, ++ 0x0008, 0x000c, 0x0093, 0x00e9, ++ 0x0000, 0x0000, 0x0000, 0x0000, ++ 0x0003, 0x0010, 0x0000, 0x0000, ++ 0x0000, 0x0002, 0x0000, 0x0000, ++ 0x0000, 0x0000, 0x0039, 0x0000, ++ 0x0000, ++}; ++ ++#define WM8510_DAIFMT \ ++ (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_RIGHT_J | \ ++ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_NB_IF | SND_SOC_DAIFMT_IB_NF | \ ++ SND_SOC_DAIFMT_IB_IF) ++ ++#define WM8510_DIR \ ++ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE) ++ ++#define WM8510_RATES \ ++ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ ++ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ ++ SNDRV_PCM_RATE_48000) ++ ++#define WM8794_BCLK \ ++ (SND_SOC_FSBD(1) | SND_SOC_FSBD(2) | SND_SOC_FSBD(4) | SND_SOC_FSBD(8) |\ ++ SND_SOC_FSBD(16) | SND_SOC_FSBD(32)) ++ ++#define WM8794_HIFI_BITS \ ++ (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ ++ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) ++ ++static struct snd_soc_dai_mode wm8510_modes[] = { ++ /* codec frame and clock master modes */ ++ { ++ .fmt = WM8510_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8794_HIFI_BITS, ++ .pcmrate = WM8510_RATES, ++ .pcmdir = WM8510_DIR, ++ .fs = 256, ++ .bfs = WM8794_BCLK, ++ }, ++ ++ /* codec frame and clock slave modes */ ++ { ++ .fmt = WM8510_DAIFMT | SND_SOC_DAIFMT_CBS_CFS, ++ .pcmfmt = WM8794_HIFI_BITS, ++ .pcmrate = WM8510_RATES, ++ .pcmdir = WM8510_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = SND_SOC_FS_ALL, ++ .bfs = SND_SOC_FSB_ALL, ++ }, ++}; ++ ++/* ++ * read wm8510 register cache ++ */ ++static inline unsigned int wm8510_read_reg_cache(struct snd_soc_codec * codec, ++ unsigned int reg) ++{ ++ u16 *cache = codec->reg_cache; ++ if (reg == WM8510_RESET) ++ return 0; ++ if (reg >= WM8510_CACHEREGNUM) ++ return -1; ++ return cache[reg]; ++} ++ ++/* ++ * write wm8510 register cache ++ */ ++static inline void wm8510_write_reg_cache(struct snd_soc_codec *codec, ++ u16 reg, unsigned int value) ++{ ++ u16 *cache = codec->reg_cache; ++ if (reg >= WM8510_CACHEREGNUM) ++ return; ++ cache[reg] = value; ++} ++ ++/* ++ * write to the WM8510 register space ++ */ ++static int wm8510_write(struct snd_soc_codec *codec, unsigned int reg, ++ unsigned int value) ++{ ++ u8 data[2]; ++ ++ /* data is ++ * D15..D9 WM8510 register offset ++ * D8...D0 register data ++ */ ++ data[0] = (reg << 1) | ((value >> 8) & 0x0001); ++ data[1] = value & 0x00ff; ++ ++ wm8510_write_reg_cache (codec, reg, value); ++ if (codec->hw_write(codec->control_data, data, 2) == 2) ++ return 0; ++ else ++ return -EIO; ++} ++ ++#define wm8510_reset(c) wm8510_write(c, WM8510_RESET, 0) ++ ++static const char *wm8510_companding[] = {"Off", "NC", "u-law", "A-law" }; ++static const char *wm8510_deemp[] = {"None", "32kHz", "44.1kHz", "48kHz" }; ++static const char *wm8510_alc[] = {"ALC", "Limiter" }; ++ ++static const struct soc_enum wm8510_enum[] = { ++ SOC_ENUM_SINGLE(WM8510_COMP, 1, 4, wm8510_companding), /* adc */ ++ SOC_ENUM_SINGLE(WM8510_COMP, 3, 4, wm8510_companding), /* dac */ ++ SOC_ENUM_SINGLE(WM8510_DAC, 4, 4, wm8510_deemp), ++ SOC_ENUM_SINGLE(WM8510_ALC3, 8, 2, wm8510_alc), ++}; ++ ++static const struct snd_kcontrol_new wm8510_snd_controls[] = { ++ ++SOC_SINGLE("Digital Loopback Switch", WM8510_COMP, 0, 1, 0), ++ ++SOC_ENUM("DAC Companding", wm8510_enum[1]), ++SOC_ENUM("ADC Companding", wm8510_enum[0]), ++ ++SOC_ENUM("Playback De-emphasis", wm8510_enum[2]), ++SOC_SINGLE("DAC Inversion Switch", WM8510_DAC, 0, 1, 0), ++ ++SOC_SINGLE("Master Playback Volume", WM8510_DACVOL, 0, 127, 0), ++ ++SOC_SINGLE("High Pass Filter Switch", WM8510_ADC, 8, 1, 0), ++SOC_SINGLE("High Pass Cut Off", WM8510_ADC, 4, 7, 0), ++SOC_SINGLE("ADC Inversion Switch", WM8510_COMP, 0, 1, 0), ++ ++SOC_SINGLE("Capture Volume", WM8510_ADCVOL, 0, 127, 0), ++ ++SOC_SINGLE("DAC Playback Limiter Switch", WM8510_DACLIM1, 8, 1, 0), ++SOC_SINGLE("DAC Playback Limiter Decay", WM8510_DACLIM1, 4, 15, 0), ++SOC_SINGLE("DAC Playback Limiter Attack", WM8510_DACLIM1, 0, 15, 0), ++ ++SOC_SINGLE("DAC Playback Limiter Threshold", WM8510_DACLIM2, 4, 7, 0), ++SOC_SINGLE("DAC Playback Limiter Boost", WM8510_DACLIM2, 0, 15, 0), ++ ++SOC_SINGLE("ALC Enable Switch", WM8510_ALC1, 8, 1, 0), ++SOC_SINGLE("ALC Capture Max Gain", WM8510_ALC1, 3, 7, 0), ++SOC_SINGLE("ALC Capture Min Gain", WM8510_ALC1, 0, 7, 0), ++ ++SOC_SINGLE("ALC Capture ZC Switch", WM8510_ALC2, 8, 1, 0), ++SOC_SINGLE("ALC Capture Hold", WM8510_ALC2, 4, 7, 0), ++SOC_SINGLE("ALC Capture Target", WM8510_ALC2, 0, 15, 0), ++ ++SOC_ENUM("ALC Capture Mode", wm8510_enum[3]), ++SOC_SINGLE("ALC Capture Decay", WM8510_ALC3, 4, 15, 0), ++SOC_SINGLE("ALC Capture Attack", WM8510_ALC3, 0, 15, 0), ++ ++SOC_SINGLE("ALC Capture Noise Gate Switch", WM8510_NGATE, 3, 1, 0), ++SOC_SINGLE("ALC Capture Noise Gate Threshold", WM8510_NGATE, 0, 7, 0), ++ ++SOC_SINGLE("Capture PGA ZC Switch", WM8510_INPPGA, 7, 1, 0), ++SOC_SINGLE("Capture PGA Volume", WM8510_INPPGA, 0, 63, 0), ++ ++SOC_SINGLE("Speaker Playback ZC Switch", WM8510_SPKVOL, 7, 1, 0), ++SOC_SINGLE("Speaker Playback Switch", WM8510_SPKVOL, 6, 1, 1), ++SOC_SINGLE("Speaker Playback Volume", WM8510_SPKVOL, 0, 63, 0), ++ ++SOC_SINGLE("Capture Boost(+20dB)", WM8510_ADCBOOST, 8, 1, 0), ++SOC_SINGLE("Mono Playback Switch", WM8510_MONOMIX, 6, 1, 0), ++}; ++ ++/* add non dapm controls */ ++static int wm8510_add_controls(struct snd_soc_codec *codec) ++{ ++ int err, i; ++ ++ for (i = 0; i < ARRAY_SIZE(wm8510_snd_controls); i++) { ++ err = snd_ctl_add(codec->card, ++ snd_soc_cnew(&wm8510_snd_controls[i],codec, NULL)); ++ if (err < 0) ++ return err; ++ } ++ ++ return 0; ++} ++ ++/* Speaker Output Mixer */ ++static const struct snd_kcontrol_new wm8510_speaker_mixer_controls[] = { ++SOC_DAPM_SINGLE("Line Bypass Switch", WM8510_SPKMIX, 1, 1, 0), ++SOC_DAPM_SINGLE("Aux Playback Switch", WM8510_SPKMIX, 5, 1, 0), ++SOC_DAPM_SINGLE("PCM Playback Switch", WM8510_SPKMIX, 0, 1, 1), ++}; ++ ++/* Mono Output Mixer */ ++static const struct snd_kcontrol_new wm8510_mono_mixer_controls[] = { ++SOC_DAPM_SINGLE("Line Bypass Switch", WM8510_MONOMIX, 1, 1, 0), ++SOC_DAPM_SINGLE("Aux Playback Switch", WM8510_MONOMIX, 2, 1, 0), ++SOC_DAPM_SINGLE("PCM Playback Switch", WM8510_MONOMIX, 0, 1, 1), ++}; ++ ++/* AUX Input boost vol */ ++static const struct snd_kcontrol_new wm8510_aux_boost_controls = ++SOC_DAPM_SINGLE("Aux Volume", WM8510_ADCBOOST, 0, 7, 0); ++ ++/* Mic Input boost vol */ ++static const struct snd_kcontrol_new wm8510_mic_boost_controls = ++SOC_DAPM_SINGLE("Mic Volume", WM8510_ADCBOOST, 4, 7, 0); ++ ++/* Capture boost switch */ ++static const struct snd_kcontrol_new wm8510_capture_boost_controls = ++SOC_DAPM_SINGLE("Capture Boost Switch", WM8510_INPPGA, 6, 1, 0); ++ ++/* Aux In to PGA */ ++static const struct snd_kcontrol_new wm8510_aux_capture_boost_controls = ++SOC_DAPM_SINGLE("Aux Capture Boost Switch", WM8510_INPPGA, 2, 1, 0); ++ ++/* Mic P In to PGA */ ++static const struct snd_kcontrol_new wm8510_micp_capture_boost_controls = ++SOC_DAPM_SINGLE("Mic P Capture Boost Switch", WM8510_INPPGA, 0, 1, 0); ++ ++/* Mic N In to PGA */ ++static const struct snd_kcontrol_new wm8510_micn_capture_boost_controls = ++SOC_DAPM_SINGLE("Mic N Capture Boost Switch", WM8510_INPPGA, 1, 1, 0); ++ ++static const struct snd_soc_dapm_widget wm8510_dapm_widgets[] = { ++SND_SOC_DAPM_MIXER("Speaker Mixer", WM8510_POWER3, 2, 0, ++ &wm8510_speaker_mixer_controls[0], ++ ARRAY_SIZE(wm8510_speaker_mixer_controls)), ++SND_SOC_DAPM_MIXER("Mono Mixer", WM8510_POWER3, 3, 0, ++ &wm8510_mono_mixer_controls[0], ++ ARRAY_SIZE(wm8510_mono_mixer_controls)), ++SND_SOC_DAPM_DAC("DAC", "HiFi Playback", WM8510_POWER3, 0, 0), ++SND_SOC_DAPM_ADC("ADC", "HiFi Capture", WM8510_POWER3, 0, 0), ++SND_SOC_DAPM_PGA("Aux Input", WM8510_POWER1, 6, 0, NULL, 0), ++SND_SOC_DAPM_PGA("SpkN Out", WM8510_POWER3, 5, 0, NULL, 0), ++SND_SOC_DAPM_PGA("SpkP Out", WM8510_POWER3, 6, 0, NULL, 0), ++SND_SOC_DAPM_PGA("Mono Out", WM8510_POWER3, 7, 0, NULL, 0), ++SND_SOC_DAPM_PGA("Mic PGA", WM8510_POWER2, 2, 0, NULL, 0), ++ ++SND_SOC_DAPM_PGA("Aux Boost", SND_SOC_NOPM, 0, 0, ++ &wm8510_aux_boost_controls, 1), ++SND_SOC_DAPM_PGA("Mic Boost", SND_SOC_NOPM, 0, 0, ++ &wm8510_mic_boost_controls, 1), ++SND_SOC_DAPM_SWITCH("Capture Boost", SND_SOC_NOPM, 0, 0, ++ &wm8510_capture_boost_controls), ++ ++SND_SOC_DAPM_MIXER("Boost Mixer", WM8510_POWER2, 4, 0, NULL, 0), ++ ++SND_SOC_DAPM_MICBIAS("Mic Bias", WM8510_POWER1, 4, 0), ++ ++SND_SOC_DAPM_INPUT("MICN"), ++SND_SOC_DAPM_INPUT("MICP"), ++SND_SOC_DAPM_INPUT("AUX"), ++SND_SOC_DAPM_OUTPUT("MONOOUT"), ++SND_SOC_DAPM_OUTPUT("SPKOUTP"), ++SND_SOC_DAPM_OUTPUT("SPKOUTN"), ++}; ++ ++static const char *audio_map[][3] = { ++ /* Mono output mixer */ ++ {"Mono Mixer", "PCM Playback Switch", "DAC"}, ++ {"Mono Mixer", "Aux Playback Switch", "Aux Input"}, ++ {"Mono Mixer", "Line Bypass Switch", "Boost Mixer"}, ++ ++ /* Speaker output mixer */ ++ {"Speaker Mixer", "PCM Playback Switch", "DAC"}, ++ {"Speaker Mixer", "Aux Playback Switch", "Aux Input"}, ++ {"Speaker Mixer", "Line Bypass Switch", "Boost Mixer"}, ++ ++ /* Outputs */ ++ {"Mono Out", NULL, "Mono Mixer"}, ++ {"MONOOUT", NULL, "Mono Out"}, ++ {"SpkN Out", NULL, "Speaker Mixer"}, ++ {"SpkP Out", NULL, "Speaker Mixer"}, ++ {"SPKOUTN", NULL, "SpkN Out"}, ++ {"SPKOUTP", NULL, "SpkP Out"}, ++ ++ /* Boost Mixer */ ++ {"Boost Mixer", NULL, "ADC"}, ++ {"Capture Boost Switch", "Aux Capture Boost Switch", "AUX"}, ++ {"Aux Boost", "Aux Volume", "Boost Mixer"}, ++ {"Capture Boost", "Capture Switch", "Boost Mixer"}, ++ {"Mic Boost", "Mic Volume", "Boost Mixer"}, ++ ++ /* Inputs */ ++ {"MICP", NULL, "Mic Boost"}, ++ {"MICN", NULL, "Mic PGA"}, ++ {"Mic PGA", NULL, "Capture Boost"}, ++ {"AUX", NULL, "Aux Input"}, ++ ++ /* terminator */ ++ {NULL, NULL, NULL}, ++}; ++ ++static int wm8510_add_widgets(struct snd_soc_codec *codec) ++{ ++ int i; ++ ++ for(i = 0; i < ARRAY_SIZE(wm8510_dapm_widgets); i++) { ++ snd_soc_dapm_new_control(codec, &wm8510_dapm_widgets[i]); ++ } ++ ++ /* set up audio path audio_mapnects */ ++ for(i = 0; audio_map[i][0] != NULL; i++) { ++ snd_soc_dapm_connect_input(codec, audio_map[i][0], ++ audio_map[i][1], audio_map[i][2]); ++ } ++ ++ snd_soc_dapm_new_widgets(codec); ++ return 0; ++} ++ ++struct pll_ { ++ unsigned int in_hz, out_hz; ++ unsigned int pre:4; /* prescale - 1 */ ++ unsigned int n:4; ++ unsigned int k; ++}; ++ ++struct pll_ pll[] = { ++ {12000000, 11289600, 0, 7, 0x86c220}, ++ {12000000, 12288000, 0, 8, 0x3126e8}, ++ {13000000, 11289600, 0, 6, 0xf28bd4}, ++ {13000000, 12288000, 0, 7, 0x8fd525}, ++ {12288000, 11289600, 0, 7, 0x59999a}, ++ {11289600, 12288000, 0, 8, 0x80dee9}, ++ /* liam - add more entries */ ++}; ++ ++static int set_pll(struct snd_soc_codec *codec, unsigned int in, ++ unsigned int out) ++{ ++ int i; ++ u16 reg; ++ ++ if(out == 0) { ++ reg = wm8510_read_reg_cache(codec, WM8510_POWER1); ++ wm8510_write(codec, WM8510_POWER1, reg & 0x1df); ++ return 0; ++ } ++ ++ for(i = 0; i < ARRAY_SIZE(pll); i++) { ++ if (in == pll[i].in_hz && out == pll[i].out_hz) { ++ wm8510_write(codec, WM8510_PLLN, (pll[i].pre << 4) | pll[i].n); ++ wm8510_write(codec, WM8510_PLLK1, pll[i].k >> 18); ++ wm8510_write(codec, WM8510_PLLK1, (pll[i].k >> 9) && 0x1ff); ++ wm8510_write(codec, WM8510_PLLK1, pll[i].k && 0x1ff); ++ reg = wm8510_read_reg_cache(codec, WM8510_POWER1); ++ wm8510_write(codec, WM8510_POWER1, reg | 0x020); ++ return 0; ++ } ++ } ++ return -EINVAL; ++} ++ ++/* mclk dividers * 2 */ ++static unsigned char mclk_div[] = {2, 3, 4, 6, 8, 12, 16, 24}; ++ ++/* we need 256FS to drive the DAC's and ADC's */ ++static unsigned int wm8510_config_sysclk(struct snd_soc_codec_dai *dai, ++ struct snd_soc_clock_info *info, unsigned int clk) ++{ ++ int i, j, best_clk = info->fs * info->rate; ++ ++ /* can we run at this clk without the PLL ? */ ++ for (i = 0; i < ARRAY_SIZE(mclk_div); i++) { ++ if ((best_clk >> 1) * mclk_div[i] == clk) { ++ dai->pll_in = 0; ++ dai->clk_div = mclk_div[i]; ++ dai->mclk = best_clk; ++ return dai->mclk; ++ } ++ } ++ ++ /* now check for PLL support */ ++ for (i = 0; i < ARRAY_SIZE(pll); i++) { ++ if (pll[i].in_hz == clk) { ++ for (j = 0; j < ARRAY_SIZE(mclk_div); j++) { ++ if (pll[i].out_hz == mclk_div[j] * (best_clk >> 1)) { ++ dai->pll_in = clk; ++ dai->pll_out = pll[i].out_hz; ++ dai->clk_div = mclk_div[j]; ++ dai->mclk = best_clk; ++ return dai->mclk; ++ } ++ } ++ } ++ } ++ ++ /* this clk is not supported */ ++ return 0; ++} ++ ++static int wm8510_pcm_prepare(struct snd_pcm_substream *substream) ++{ ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ struct snd_soc_device *socdev = rtd->socdev; ++ struct snd_soc_codec *codec = socdev->codec; ++ struct snd_soc_codec_dai *dai = rtd->codec_dai; ++ u16 iface = 0, bfs, clk = 0, adn; ++ int fs = 48000 << 7, i; ++ ++ bfs = SND_SOC_FSBD_REAL(rtd->codec_dai->dai_runtime.bfs); ++ switch (bfs) { ++ case 2: ++ clk |= 0x1 << 2; ++ break; ++ case 4: ++ clk |= 0x2 << 2; ++ break; ++ case 8: ++ clk |= 0x3 << 2; ++ break; ++ case 16: ++ clk |= 0x4 << 2; ++ break; ++ case 32: ++ clk |= 0x5 << 2; ++ break; ++ } ++ ++ /* set master/slave audio interface */ ++ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_CLOCK_MASK) { ++ case SND_SOC_DAIFMT_CBM_CFM: ++ clk |= 0x0001; ++ break; ++ case SND_SOC_DAIFMT_CBS_CFS: ++ break; ++ } ++ ++ /* interface format */ ++ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_FORMAT_MASK) { ++ case SND_SOC_DAIFMT_I2S: ++ iface |= 0x0010; ++ break; ++ case SND_SOC_DAIFMT_RIGHT_J: ++ break; ++ case SND_SOC_DAIFMT_LEFT_J: ++ iface |= 0x0008; ++ break; ++ case SND_SOC_DAIFMT_DSP_A: ++ iface |= 0x00018; ++ break; ++ } ++ ++ /* bit size */ ++ switch (rtd->codec_dai->dai_runtime.pcmfmt) { ++ case SNDRV_PCM_FMTBIT_S16_LE: ++ break; ++ case SNDRV_PCM_FMTBIT_S20_3LE: ++ iface |= 0x0020; ++ break; ++ case SNDRV_PCM_FMTBIT_S24_LE: ++ iface |= 0x0040; ++ break; ++ case SNDRV_PCM_FMTBIT_S32_LE: ++ iface |= 0x0060; ++ break; ++ } ++ ++ /* clock inversion */ ++ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_INV_MASK) { ++ case SND_SOC_DAIFMT_NB_NF: ++ break; ++ case SND_SOC_DAIFMT_IB_IF: ++ iface |= 0x0180; ++ break; ++ case SND_SOC_DAIFMT_IB_NF: ++ iface |= 0x0100; ++ break; ++ case SND_SOC_DAIFMT_NB_IF: ++ iface |= 0x0080; ++ break; ++ } ++ ++ /* filter coefficient */ ++ adn = wm8510_read_reg_cache(codec, WM8510_ADD) & 0x1f1; ++ switch (rtd->codec_dai->dai_runtime.pcmrate) { ++ case SNDRV_PCM_RATE_8000: ++ adn |= 0x5 << 1; ++ fs = 8000 << 7; ++ break; ++ case SNDRV_PCM_RATE_11025: ++ adn |= 0x4 << 1; ++ fs = 11025 << 7; ++ break; ++ case SNDRV_PCM_RATE_16000: ++ adn |= 0x3 << 1; ++ fs = 16000 << 7; ++ break; ++ case SNDRV_PCM_RATE_22050: ++ adn |= 0x2 << 1; ++ fs = 22050 << 7; ++ break; ++ case SNDRV_PCM_RATE_32000: ++ adn |= 0x1 << 1; ++ fs = 32000 << 7; ++ break; ++ case SNDRV_PCM_RATE_44100: ++ fs = 44100 << 7; ++ break; ++ } ++ ++ /* do we need to enable the PLL */ ++ if(dai->pll_in) ++ set_pll(codec, dai->pll_in, dai->pll_out); ++ ++ /* divide the clock to 256 fs */ ++ for(i = 0; i < ARRAY_SIZE(mclk_div); i++) { ++ if (dai->clk_div == mclk_div[i]) { ++ clk |= i << 5; ++ clk &= 0xff; ++ goto set; ++ } ++ } ++ ++set: ++ /* set iface */ ++ wm8510_write(codec, WM8510_IFACE, iface); ++ wm8510_write(codec, WM8510_CLOCK, clk); ++ ++ return 0; ++} ++ ++static int wm8510_hw_free(struct snd_pcm_substream *substream) ++{ ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ struct snd_soc_device *socdev = rtd->socdev; ++ struct snd_soc_codec *codec = socdev->codec; ++ set_pll(codec, 0, 0); ++ return 0; ++} ++ ++static int wm8510_mute(struct snd_soc_codec *codec, ++ struct snd_soc_codec_dai *dai, int mute) ++{ ++ u16 mute_reg = wm8510_read_reg_cache(codec, WM8510_DAC) & 0xffbf; ++ if(mute) ++ wm8510_write(codec, WM8510_DAC, mute_reg | 0x40); ++ else ++ wm8510_write(codec, WM8510_DAC, mute_reg); ++ return 0; ++} ++ ++/* liam need to make this lower power with dapm */ ++static int wm8510_dapm_event(struct snd_soc_codec *codec, int event) ++{ ++ ++ switch (event) { ++ case SNDRV_CTL_POWER_D0: /* full On */ ++ /* vref/mid, clk and osc on, dac unmute, active */ ++ wm8510_write(codec, WM8510_POWER1, 0x1ff); ++ wm8510_write(codec, WM8510_POWER2, 0x1ff); ++ wm8510_write(codec, WM8510_POWER3, 0x1ff); ++ break; ++ case SNDRV_CTL_POWER_D1: /* partial On */ ++ case SNDRV_CTL_POWER_D2: /* partial On */ ++ break; ++ case SNDRV_CTL_POWER_D3hot: /* Off, with power */ ++ /* everything off except vref/vmid, dac mute, inactive */ ++ ++ break; ++ case SNDRV_CTL_POWER_D3cold: /* Off, without power */ ++ /* everything off, dac mute, inactive */ ++ wm8510_write(codec, WM8510_POWER1, 0x0); ++ wm8510_write(codec, WM8510_POWER2, 0x0); ++ wm8510_write(codec, WM8510_POWER3, 0x0); ++ break; ++ } ++ codec->dapm_state = event; ++ return 0; ++} ++ ++struct snd_soc_codec_dai wm8510_dai = { ++ .name = "WM8510 HiFi", ++ .playback = { ++ .stream_name = "Playback", ++ .channels_min = 1, ++ .channels_max = 1, ++ }, ++ .capture = { ++ .stream_name = "Capture", ++ .channels_min = 1, ++ .channels_max = 1, ++ }, ++ .config_sysclk = wm8510_config_sysclk, ++ .digital_mute = wm8510_mute, ++ .ops = { ++ .prepare = wm8510_pcm_prepare, ++ .hw_free = wm8510_hw_free, ++ }, ++ .caps = { ++ .num_modes = ARRAY_SIZE(wm8510_modes), ++ .mode = wm8510_modes, ++ }, ++}; ++EXPORT_SYMBOL_GPL(wm8510_dai); ++ ++static int wm8510_suspend(struct platform_device *pdev, pm_message_t state) ++{ ++ struct snd_soc_device *socdev = platform_get_drvdata(pdev); ++ struct snd_soc_codec *codec = socdev->codec; ++ ++ wm8510_dapm_event(codec, SNDRV_CTL_POWER_D3cold); ++ return 0; ++} ++ ++static int wm8510_resume(struct platform_device *pdev) ++{ ++ struct snd_soc_device *socdev = platform_get_drvdata(pdev); ++ struct snd_soc_codec *codec = socdev->codec; ++ int i; ++ u8 data[2]; ++ u16 *cache = codec->reg_cache; ++ ++ /* Sync reg_cache with the hardware */ ++ for (i = 0; i < ARRAY_SIZE(wm8510_reg); i++) { ++ data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); ++ data[1] = cache[i] & 0x00ff; ++ codec->hw_write(codec->control_data, data, 2); ++ } ++ wm8510_dapm_event(codec, SNDRV_CTL_POWER_D3hot); ++ wm8510_dapm_event(codec, codec->suspend_dapm_state); ++ return 0; ++} ++ ++/* ++ * initialise the WM8510 driver ++ * register the mixer and dsp interfaces with the kernel ++ */ ++static int wm8510_init(struct snd_soc_device *socdev) ++{ ++ struct snd_soc_codec *codec = socdev->codec; ++ int ret = 0; ++ ++ codec->name = "WM8510"; ++ codec->owner = THIS_MODULE; ++ codec->read = wm8510_read_reg_cache; ++ codec->write = wm8510_write; ++ codec->dapm_event = wm8510_dapm_event; ++ codec->dai = &wm8510_dai; ++ codec->num_dai = 1; ++ codec->reg_cache_size = ARRAY_SIZE(wm8510_reg); ++ codec->reg_cache = ++ kzalloc(sizeof(u16) * ARRAY_SIZE(wm8510_reg), GFP_KERNEL); ++ if (codec->reg_cache == NULL) ++ return -ENOMEM; ++ memcpy(codec->reg_cache, wm8510_reg, ++ sizeof(u16) * ARRAY_SIZE(wm8510_reg)); ++ codec->reg_cache_size = sizeof(u16) * ARRAY_SIZE(wm8510_reg); ++ ++ wm8510_reset(codec); ++ ++ /* register pcms */ ++ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); ++ if(ret < 0) { ++ kfree(codec->reg_cache); ++ return ret; ++ } ++ ++ /* power on device */ ++ wm8510_dapm_event(codec, SNDRV_CTL_POWER_D3hot); ++ wm8510_add_controls(codec); ++ wm8510_add_widgets(codec); ++ ret = snd_soc_register_card(socdev); ++ if(ret < 0) { ++ snd_soc_free_pcms(socdev); ++ snd_soc_dapm_free(socdev); ++ } ++ ++ return ret; ++} ++ ++static struct snd_soc_device *wm8510_socdev; ++ ++#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) ++ ++/* ++ * WM8510 2 wire address is 0x1a ++ */ ++#define I2C_DRIVERID_WM8510 0xfefe /* liam - need a proper id */ ++ ++static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END }; ++ ++/* Magic definition of all other variables and things */ ++I2C_CLIENT_INSMOD; ++ ++static struct i2c_driver wm8510_i2c_driver; ++static struct i2c_client client_template; ++ ++/* If the i2c layer weren't so broken, we could pass this kind of data ++ around */ ++ ++static int wm8510_codec_probe(struct i2c_adapter *adap, int addr, int kind) ++{ ++ struct snd_soc_device *socdev = wm8510_socdev; ++ struct wm8510_setup_data *setup = socdev->codec_data; ++ struct snd_soc_codec *codec = socdev->codec; ++ struct i2c_client *i2c; ++ int ret; ++ ++ if (addr != setup->i2c_address) ++ return -ENODEV; ++ ++ client_template.adapter = adap; ++ client_template.addr = addr; ++ ++ i2c = kzalloc(sizeof(struct i2c_client), GFP_KERNEL); ++ if (i2c == NULL){ ++ kfree(codec); ++ return -ENOMEM; ++ } ++ memcpy(i2c, &client_template, sizeof(struct i2c_client)); ++ i2c_set_clientdata(i2c, codec); ++ codec->control_data = i2c; ++ ++ ret = i2c_attach_client(i2c); ++ if(ret < 0) { ++ err("failed to attach codec at addr %x\n", addr); ++ goto err; ++ } ++ ++ ret = wm8510_init(socdev); ++ if(ret < 0) { ++ err("failed to initialise WM8510\n"); ++ goto err; ++ } ++ return ret; ++ ++err: ++ kfree(codec); ++ kfree(i2c); ++ return ret; ++} ++ ++static int wm8510_i2c_detach(struct i2c_client *client) ++{ ++ struct snd_soc_codec *codec = i2c_get_clientdata(client); ++ i2c_detach_client(client); ++ kfree(codec->reg_cache); ++ kfree(client); ++ return 0; ++} ++ ++static int wm8510_i2c_attach(struct i2c_adapter *adap) ++{ ++ return i2c_probe(adap, &addr_data, wm8510_codec_probe); ++} ++ ++/* corgi i2c codec control layer */ ++static struct i2c_driver wm8510_i2c_driver = { ++ .driver = { ++ .name = "WM8510 I2C Codec", ++ .owner = THIS_MODULE, ++ }, ++ .id = I2C_DRIVERID_WM8510, ++ .attach_adapter = wm8510_i2c_attach, ++ .detach_client = wm8510_i2c_detach, ++ .command = NULL, ++}; ++ ++static struct i2c_client client_template = { ++ .name = "WM8510", ++ .driver = &wm8510_i2c_driver, ++}; ++#endif ++ ++static int wm8510_probe(struct platform_device *pdev) ++{ ++ struct snd_soc_device *socdev = platform_get_drvdata(pdev); ++ struct wm8510_setup_data *setup; ++ struct snd_soc_codec *codec; ++ int ret = 0; ++ ++ info("WM8510 Audio Codec %s", WM8510_VERSION); ++ ++ setup = socdev->codec_data; ++ codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); ++ if (codec == NULL) ++ return -ENOMEM; ++ ++ socdev->codec = codec; ++ mutex_init(&codec->mutex); ++ INIT_LIST_HEAD(&codec->dapm_widgets); ++ INIT_LIST_HEAD(&codec->dapm_paths); ++ ++ wm8510_socdev = socdev; ++#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) ++ if (setup->i2c_address) { ++ normal_i2c[0] = setup->i2c_address; ++ codec->hw_write = (hw_write_t)i2c_master_send; ++ ret = i2c_add_driver(&wm8510_i2c_driver); ++ if (ret != 0) ++ printk(KERN_ERR "can't add i2c driver"); ++ } ++#else ++ /* Add other interfaces here */ ++#endif ++ return ret; ++} ++ ++/* power down chip */ ++static int wm8510_remove(struct platform_device *pdev) ++{ ++ struct snd_soc_device *socdev = platform_get_drvdata(pdev); ++ struct snd_soc_codec *codec = socdev->codec; ++ ++ if (codec->control_data) ++ wm8510_dapm_event(codec, SNDRV_CTL_POWER_D3cold); ++ ++ snd_soc_free_pcms(socdev); ++ snd_soc_dapm_free(socdev); ++#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) ++ i2c_del_driver(&wm8510_i2c_driver); ++#endif ++ kfree(codec); ++ ++ return 0; ++} ++ ++struct snd_soc_codec_device soc_codec_dev_wm8510 = { ++ .probe = wm8510_probe, ++ .remove = wm8510_remove, ++ .suspend = wm8510_suspend, ++ .resume = wm8510_resume, ++}; ++ ++EXPORT_SYMBOL_GPL(soc_codec_dev_wm8510); ++ ++MODULE_DESCRIPTION("ASoC WM8510 driver"); ++MODULE_AUTHOR("Liam Girdwood"); ++MODULE_LICENSE("GPL"); +Index: linux-2.6-pxa-new/sound/soc/codecs/wm8510.h +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/sound/soc/codecs/wm8510.h +@@ -0,0 +1,64 @@ ++/* ++ * wm8510.h -- WM8510 Soc Audio driver ++ * ++ * This program is free software; you can redistribute it and/or modify ++ * it under the terms of the GNU General Public License version 2 as ++ * published by the Free Software Foundation. ++ */ ++ ++#ifndef _WM8510_H ++#define _WM8510_H ++ ++/* WM8510 register space */ ++ ++#define WM8510_RESET 0x0 ++#define WM8510_POWER1 0x1 ++#define WM8510_POWER2 0x2 ++#define WM8510_POWER3 0x3 ++#define WM8510_IFACE 0x4 ++#define WM8510_COMP 0x5 ++#define WM8510_CLOCK 0x6 ++#define WM8510_ADD 0x7 ++#define WM8510_GPIO 0x8 ++#define WM8510_DAC 0xa ++#define WM8510_DACVOL 0xb ++#define WM8510_ADC 0xe ++#define WM8510_ADCVOL 0xf ++#define WM8510_EQ1 0x12 ++#define WM8510_EQ2 0x13 ++#define WM8510_EQ3 0x14 ++#define WM8510_EQ4 0x15 ++#define WM8510_EQ5 0x16 ++#define WM8510_DACLIM1 0x18 ++#define WM8510_DACLIM2 0x19 ++#define WM8510_NOTCH1 0x1b ++#define WM8510_NOTCH2 0x1c ++#define WM8510_NOTCH3 0x1d ++#define WM8510_NOTCH4 0x1e ++#define WM8510_ALC1 0x20 ++#define WM8510_ALC2 0x21 ++#define WM8510_ALC3 0x22 ++#define WM8510_NGATE 0x23 ++#define WM8510_PLLN 0x24 ++#define WM8510_PLLK1 0x25 ++#define WM8510_PLLK2 0x26 ++#define WM8510_PLLK3 0x27 ++#define WM8510_ATTEN 0x28 ++#define WM8510_INPUT 0x2c ++#define WM8510_INPPGA 0x2d ++#define WM8510_ADCBOOST 0x2f ++#define WM8510_OUTPUT 0x31 ++#define WM8510_SPKMIX 0x32 ++#define WM8510_SPKVOL 0x36 ++#define WM8510_MONOMIX 0x38 ++ ++#define WM8510_CACHEREGNUM 57 ++ ++struct wm8510_setup_data { ++ unsigned short i2c_address; ++}; ++ ++extern struct snd_soc_codec_dai wm8510_dai; ++extern struct snd_soc_codec_device soc_codec_dev_wm8510; ++ ++#endif +Index: linux-2.6-pxa-new/sound/soc/imx/imx-ac97.c +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/sound/soc/imx/imx-ac97.c +@@ -0,0 +1,281 @@ ++/* ++ * imx-ssi.c -- SSI driver for Freescale IMX ++ * ++ * Copyright 2006 Wolfson Microelectronics PLC. ++ * Author: Liam Girdwood ++ * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com ++ * ++ * Based on mxc-alsa-mc13783 (C) 2006 Freescale. ++ * ++ * This program is free software; you can redistribute it and/or modify it ++ * under the terms of the GNU General Public License as published by the ++ * Free Software Foundation; either version 2 of the License, or (at your ++ * option) any later version. ++ * ++ * Revision history ++ * 29th Aug 2006 Initial version. ++ * ++ */ ++ ++#define IMX_AC97_RATES \ ++ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ ++ SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ ++ SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ ++ SNDRV_PCM_RATE_48000) ++ ++/* may need to expand this */ ++static struct snd_soc_dai_mode imx_ssi_ac97_modes[] = { ++ {0, 0, SNDRV_PCM_FMTBIT_S16_LE, IMX_AC97_RATES}, ++ {0, 0, SNDRV_PCM_FMTBIT_S18_3LE, IMX_AC97_RATES}, ++ {0, 0, SNDRV_PCM_FMTBIT_S20_3LE, IMX_AC97_RATES}, ++}; ++ ++static imx_pcm_dma_params_t imx_ssi1_pcm_stereo_out = { ++ .name = "SSI1 PCM Stereo out", ++ .params = { ++ .bd_number = 1, ++ .transfer_type = emi_2_per, ++ .watermark_level = SDMA_TXFIFO_WATERMARK, ++ .word_size = TRANSFER_16BIT, // maybe add this in setup func ++ .per_address = SSI1_STX0, ++ .event_id = DMA_REQ_SSI1_TX1, ++ .peripheral_type = SSI, ++ }, ++}; ++ ++static imx_pcm_dma_params_t imx_ssi1_pcm_stereo_in = { ++ .name = "SSI1 PCM Stereo in", ++ .params = { ++ .bd_number = 1, ++ .transfer_type = per_2_emi, ++ .watermark_level = SDMA_RXFIFO_WATERMARK, ++ .word_size = TRANSFER_16BIT, // maybe add this in setup func ++ .per_address = SSI1_SRX0, ++ .event_id = DMA_REQ_SSI1_RX1, ++ .peripheral_type = SSI, ++ }, ++}; ++ ++static imx_pcm_dma_params_t imx_ssi2_pcm_stereo_out = { ++ .name = "SSI2 PCM Stereo out", ++ .params = { ++ .bd_number = 1, ++ .transfer_type = per_2_emi, ++ .watermark_level = SDMA_TXFIFO_WATERMARK, ++ .word_size = TRANSFER_16BIT, // maybe add this in setup func ++ .per_address = SSI2_STX0, ++ .event_id = DMA_REQ_SSI2_TX1, ++ .peripheral_type = SSI, ++ }, ++}; ++ ++static imx_pcm_dma_params_t imx_ssi2_pcm_stereo_in = { ++ .name = "SSI2 PCM Stereo in", ++ .params = { ++ .bd_number = 1, ++ .transfer_type = per_2_emi, ++ .watermark_level = SDMA_RXFIFO_WATERMARK, ++ .word_size = TRANSFER_16BIT, // maybe add this in setup func ++ .per_address = SSI2_SRX0, ++ .event_id = DMA_REQ_SSI2_RX1, ++ .peripheral_type = SSI, ++ }, ++}; ++ ++static unsigned short imx_ssi_ac97_read(struct snd_ac97 *ac97, unsigned short reg) ++{ ++} ++ ++static void imx_ssi_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short val) ++{ ++} ++ ++static void imx_ssi_ac97_warm_reset(struct snd_ac97 *ac97) ++{ ++} ++ ++static void imx_ssi_ac97_cold_reset(struct snd_ac97 *ac97) ++{ ++} ++ ++struct snd_ac97_bus_ops soc_ac97_ops = { ++ .read = imx_ssi_ac97_read, ++ .write = imx_ssi_ac97_write, ++ .warm_reset = imx_ssi_ac97_warm_reset, ++ .reset = imx_ssi_ac97_cold_reset, ++}; ++ ++ ++static intimx_ssi1_ac97_probe(struct platform_device *pdev) ++{ ++ int ret; ++ ++ ++ return ret; ++} ++ ++static void imx_ssi1_ac97_remove(struct platform_device *pdev) ++{ ++ /* shutdown SSI */ ++ if(rtd->cpu_dai->id == 0) ++ SSI1_SCR &= ~SSI_SCR_SSIEN; ++ else ++ SSI2_SCR &= ~SSI_SCR_SSIEN; ++ } ++ ++} ++ ++static int imx_ssi1_ac97_prepare(struct snd_pcm_substream *substream) ++{ ++ // set vra ++} ++ ++static int imx_ssi_startup(struct snd_pcm_substream *substream) ++{ ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ ++ if (!rtd->cpu_dai->active) { ++ ++ } ++ ++ return 0; ++} ++ ++static int imx_ssi1_trigger(struct snd_pcm_substream *substream, int cmd) ++{ ++ int ret = 0; ++ ++ switch (cmd) { ++ case SNDRV_PCM_TRIGGER_START: ++ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { ++ SSI1_SCR |= SSI_SCR_TE; ++ SSI1_SIER |= SSI_SIER_TDMAE; ++ } else { ++ SSI1_SCR |= SSI_SCR_RE; ++ SSI1_SIER |= SSI_SIER_RDMAE; ++ } ++ SSI1_SCR |= SSI_SCR_SSIEN; ++ ++ break; ++ case SNDRV_PCM_TRIGGER_RESUME: ++ break; ++ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: ++ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ++ SSI1_SCR |= SSI_SCR_TE; ++ else ++ SSI1_SCR |= SSI_SCR_RE; ++ break ++ case SNDRV_PCM_TRIGGER_STOP: ++ case SNDRV_PCM_TRIGGER_SUSPEND: ++ break; ++ case SNDRV_PCM_TRIGGER_PAUSE_PUSH: ++ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ++ SSI1_SCR &= ~SSI_SCR_TE; ++ else ++ SSI1_SCR &= ~SSI_SCR_RE; ++ break; ++ default: ++ ret = -EINVAL; ++ } ++ ++ return ret; ++} ++ ++static void imx_ssi_shutdown(struct snd_pcm_substream *substream) ++{ ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ ++ ++} ++ ++#ifdef CONFIG_PM ++static int imx_ssi_suspend(struct platform_device *dev, ++ struct snd_soc_cpu_dai *dai) ++{ ++ if(!dai->active) ++ return 0; ++ ++ if(rtd->cpu_dai->id == 0) ++ SSI1_SCR &= ~SSI_SCR_SSIEN; ++ else ++ SSI2_SCR &= ~SSI_SCR_SSIEN; ++ ++ return 0; ++} ++ ++static int imx_ssi_resume(struct platform_device *pdev, ++ struct snd_soc_cpu_dai *dai) ++{ ++ if(!dai->active) ++ return 0; ++ ++ if(rtd->cpu_dai->id == 0) ++ SSI1_SCR |= SSI_SCR_SSIEN; ++ else ++ SSI2_SCR |= SSI_SCR_SSIEN; ++ ++ return 0; ++} ++ ++#else ++#define imx_ssi_suspend NULL ++#define imx_ssi_resume NULL ++#endif ++ ++static unsigned int imx_ssi_config_ac97_sysclk(struct snd_soc_cpu_dai *iface, ++ struct snd_soc_clock_info *info, unsigned int clk) ++{ ++ return clk; ++} ++ ++struct snd_soc_cpu_dai imx_ssi_ac97_dai = { ++ .name = "imx-ac97-1", ++ .id = 0, ++ .type = SND_SOC_DAI_AC97, ++ .suspend = imx_ssi_suspend, ++ .resume = imx_ssi_resume, ++ .config_sysclk = imx_ssi_ac97_config_sysclk, ++ .playback = { ++ .channels_min = 2, ++ .channels_max = 2,}, ++ .capture = { ++ .channels_min = 2, ++ .channels_max = 2,}, ++ .ops = { ++ .probe = imx_ac97_probe, ++ .remove = imx_ac97_shutdown, ++ .trigger = imx_ssi1_trigger, ++ .prepare = imx_ssi_ac97_prepare,}, ++ .caps = { ++ .num_modes = ARRAY_SIZE(imx_ssi_ac97_modes), ++ .mode = imx_ssi_ac97_modes,}, ++}, ++{ ++ .name = "imx-ac97-2", ++ .id = 1, ++ .type = SND_SOC_DAI_AC97, ++ .suspend = imx_ssi_suspend, ++ .resume = imx_ssi_resume, ++ .config_sysclk = imx_ssi_ac97_config_sysclk, ++ .playback = { ++ .channels_min = 2, ++ .channels_max = 2,}, ++ .capture = { ++ .channels_min = 2, ++ .channels_max = 2,}, ++ .ops = { ++ .probe = imx_ac97_probe, ++ .remove = imx_ac97_shutdown, ++ .trigger = imx_ssi1_trigger, ++ .prepare = imx_ssi_ac97_prepare,}, ++ .caps = { ++ .num_modes = ARRAY_SIZE(imx_ssi_ac97_modes), ++ .mode = imx_ssi_ac97_modes,}, ++}; ++ ++EXPORT_SYMBOL_GPL(imx_ssi_ac97_dai); ++ ++/* Module information */ ++MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com"); ++MODULE_DESCRIPTION("i.MX ASoC AC97 driver"); ++MODULE_LICENSE("GPL"); +Index: linux-2.6-pxa-new/sound/soc/imx/imx-i2s.c +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/sound/soc/imx/imx-i2s.c +@@ -0,0 +1,473 @@ ++/* ++ * imx-ssi.c -- SSI driver for Freescale IMX ++ * ++ * Copyright 2006 Wolfson Microelectronics PLC. ++ * Author: Liam Girdwood ++ * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com ++ * ++ * Based on mxc-alsa-mc13783 (C) 2006 Freescale. ++ * ++ * This program is free software; you can redistribute it and/or modify it ++ * under the terms of the GNU General Public License as published by the ++ * Free Software Foundation; either version 2 of the License, or (at your ++ * option) any later version. ++ * ++ * Revision history ++ * 29th Aug 2006 Initial version. ++ * ++ */ ++ ++#define IMX_SSI_DAIFMT \ ++ (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J |\ ++ SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_DSP__A |\ ++ SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBS_CFS |\ ++ SND_SOC_DAIFMT_CBM_CFS | SND_SOC_DAIFMT_CBS_CFM |\ ++ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_NB_IF) ++ ++#define IMX_SSI_DIR \ ++ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE) ++ ++#define IMX_SSI_RATES \ ++ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | \ ++ SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | \ ++ SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ ++ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | \ ++ SNDRV_PCM_RATE_96000) ++ ++#define IMX_SSI_BITS \ ++ (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ ++ SNDRV_PCM_FMTBIT_S24_LE) ++ ++static struct snd_soc_dai_mode imx_ssi_pcm_modes[] = { ++ ++ /* frame master and clock slave mode */ ++ { ++ .fmt = IMX_SSI_DAIFMT | SND_SOC_DAIFMT_CBM_CFS, ++ .tdm = SND_SOC_DAITDM_LRDW(0,0), ++ .pcmfmt = IMX_SSI_BITS, ++ .pcmrate = IMX_SSI_RATES, ++ .pcmdir = IMX_SSI_DIR, ++ .flags = SND_SOC_DAI_BFS_RCW, ++ .fs = SND_SOC_FS_ALL, ++ .bfs = SND_SOC_FSBW(1) | SND_SOC_FSBW(2), ++ }, ++}; ++ ++static imx_pcm_dma_params_t imx_ssi1_pcm_stereo_out = { ++ .name = "SSI1 PCM Stereo out", ++ .params = { ++ .bd_number = 1, ++ .transfer_type = emi_2_per, ++ .watermark_level = SDMA_TXFIFO_WATERMARK, ++ .word_size = TRANSFER_16BIT, // maybe add this in setup func ++ .per_address = SSI1_STX0, ++ .event_id = DMA_REQ_SSI1_TX1, ++ .peripheral_type = SSI, ++ }, ++}; ++ ++static imx_pcm_dma_params_t imx_ssi1_pcm_stereo_in = { ++ .name = "SSI1 PCM Stereo in", ++ .params = { ++ .bd_number = 1, ++ .transfer_type = per_2_emi, ++ .watermark_level = SDMA_RXFIFO_WATERMARK, ++ .word_size = TRANSFER_16BIT, // maybe add this in setup func ++ .per_address = SSI1_SRX0, ++ .event_id = DMA_REQ_SSI1_RX1, ++ .peripheral_type = SSI, ++ }, ++}; ++ ++static imx_pcm_dma_params_t imx_ssi2_pcm_stereo_out = { ++ .name = "SSI2 PCM Stereo out", ++ .params = { ++ .bd_number = 1, ++ .transfer_type = per_2_emi, ++ .watermark_level = SDMA_TXFIFO_WATERMARK, ++ .word_size = TRANSFER_16BIT, // maybe add this in setup func ++ .per_address = SSI2_STX0, ++ .event_id = DMA_REQ_SSI2_TX1, ++ .peripheral_type = SSI, ++ }, ++}; ++ ++static imx_pcm_dma_params_t imx_ssi2_pcm_stereo_in = { ++ .name = "SSI2 PCM Stereo in", ++ .params = { ++ .bd_number = 1, ++ .transfer_type = per_2_emi, ++ .watermark_level = SDMA_RXFIFO_WATERMARK, ++ .word_size = TRANSFER_16BIT, // maybe add this in setup func ++ .per_address = SSI2_SRX0, ++ .event_id = DMA_REQ_SSI2_RX1, ++ .peripheral_type = SSI, ++ }, ++}; ++ ++ ++static int imx_ssi_startup(struct snd_pcm_substream *substream) ++{ ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ ++ if (!rtd->cpu_dai->active) { ++ ++ } ++ ++ return 0; ++} ++ ++static int imx_ssi1_hw_tx_params(struct snd_pcm_substream *substream, ++ struct snd_pcm_hw_params *params) ++{ ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ struct snd_soc_device *socdev = rtd->socdev; ++ struct snd_soc_codec *codec = socdev->codec; ++ u16 bfs, div; ++ ++ bfs = SND_SOC_FSBD_REAL(rtd->cpu_dai->dai_runtime.bfs); ++ ++ SSI1_STCR = 0; ++ SSI1_STCCR = 0; ++ ++ /* DAI mode */ ++ switch(rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_FORMAT_MASK) { ++ case SND_SOC_DAIFMT_I2S: ++ SSI1_STCR |= SSI_STCR_TSCKP | SSI_STCR_TFSI | ++ SSI_STCR_TEFS | SSI_STCR_TXBIT0; ++ break; ++ case SND_SOC_DAIFMT_LEFT_J: ++ SSI1_STCR |= SSI_STCR_TSCKP | SSI_STCR_TFSI | SSI_STCR_TXBIT0; ++ break; ++ case SND_SOC_DAIFMT_DSP_B: ++ SSI1_STCR |= SSI_STCR_TEFS; // data 1 bit after sync ++ case SND_SOC_DAIFMT_DSP_A: ++ SSI1_STCR |= SSI_STCR_TFSL; // frame is 1 bclk long ++ ++ /* DAI clock inversion */ ++ switch(rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_INV_MASK) { ++ case SND_SOC_DAIFMT_IB_IF: ++ SSI1_STCR |= SSI_STCR_TFSI | SSI_STCR_TSCKP; ++ break; ++ case SND_SOC_DAIFMT_IB_NF: ++ SSI1_STCR |= SSI_STCR_TSCKP; ++ break; ++ case SND_SOC_DAIFMT_NB_IF: ++ SSI1_STCR |= SSI_STCR_TFSI; ++ break; ++ } ++ break; ++ } ++ ++ /* DAI data (word) size */ ++ switch(rtd->codec_dai->dai_runtime.pcmfmt) { ++ case SNDRV_PCM_FMTBIT_S16_LE: ++ SSI1_STCCR |= SSI_STCCR_WL(16); ++ break; ++ case SNDRV_PCM_FMTBIT_S20_3LE: ++ SSI1_STCCR |= SSI_STCCR_WL(20); ++ break; ++ case SNDRV_PCM_FMTBIT_S24_LE: ++ SSI1_STCCR |= SSI_STCCR_WL(24); ++ break; ++ } ++ ++ /* DAI clock master masks */ ++ switch(rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_CLOCK_MASK){ ++ case SND_SOC_DAIFMT_CBM_CFM: ++ SSI1_STCR |= SSI_STCR_TFDIR | SSI_STCR_TXDIR; ++ break; ++ case SND_SOC_DAIFMT_CBS_CFM: ++ SSI1_STCR |= SSI_STCR_TFDIR; ++ break; ++ case SND_SOC_DAIFMT_CBM_CFS: ++ SSI1_STCR |= SSI_STCR_TXDIR; ++ break; ++ } ++ ++ /* DAI BCLK ratio to SYSCLK / MCLK */ ++ /* prescaler modulus - todo */ ++ switch (bfs) { ++ case 2: ++ break; ++ case 4: ++ break; ++ case 8: ++ break; ++ case 16: ++ break; ++ } ++ ++ /* TDM - todo, only fifo 0 atm */ ++ SSI1_STCR |= SSI_STCR_TFEN0; ++ SSI1_STCCR |= SSI_STCCR_DC(params_channels(params)); ++ ++ return 0; ++} ++ ++static int imx_ssi1_hw_rx_params(struct snd_pcm_substream *substream, ++ struct snd_pcm_hw_params *params) ++{ ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ struct snd_soc_device *socdev = rtd->socdev; ++ struct snd_soc_codec *codec = socdev->codec; ++ u16 bfs, div; ++ ++ bfs = SND_SOC_FSBD_REAL(rtd->cpu_dai->dai_runtime.bfs); ++ ++ SSI1_SRCR = 0; ++ SSI1_SRCCR = 0; ++ ++ /* DAI mode */ ++ switch(rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_FORMAT_MASK) { ++ case SND_SOC_DAIFMT_I2S: ++ SSI1_SRCR |= SSI_SRCR_RSCKP | SSI_SRCR_RFSI | ++ SSI_STCR_REFS | SSI_STCR_RXBIT0; ++ break; ++ case SND_SOC_DAIFMT_LEFT_J: ++ SSI1_SRCR |= SSI_SRCR_RSCKP | SSI_SRCR_RFSI | SSI_SRCR_RXBIT0; ++ break; ++ case SND_SOC_DAIFMT_DSP_B: ++ SSI1_SRCR |= SSI_SRCR_REFS; // data 1 bit after sync ++ case SND_SOC_DAIFMT_DSP_A: ++ SSI1_SRCR |= SSI_SRCR_RFSL; // frame is 1 bclk long ++ ++ /* DAI clock inversion */ ++ switch(rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_INV_MASK) { ++ case SND_SOC_DAIFMT_IB_IF: ++ SSI1_SRCR |= SSI_SRCR_TFSI | SSI_SRCR_TSCKP; ++ break; ++ case SND_SOC_DAIFMT_IB_NF: ++ SSI1_SRCR |= SSI_SRCR_RSCKP; ++ break; ++ case SND_SOC_DAIFMT_NB_IF: ++ SSI1_SRCR |= SSI_SRCR_RFSI; ++ break; ++ } ++ break; ++ } ++ ++ /* DAI data (word) size */ ++ switch(rtd->codec_dai->dai_runtime.pcmfmt) { ++ case SNDRV_PCM_FMTBIT_S16_LE: ++ SSI1_SRCCR |= SSI_SRCCR_WL(16); ++ break; ++ case SNDRV_PCM_FMTBIT_S20_3LE: ++ SSI1_SRCCR |= SSI_SRCCR_WL(20); ++ break; ++ case SNDRV_PCM_FMTBIT_S24_LE: ++ SSI1_SRCCR |= SSI_SRCCR_WL(24); ++ break; ++ } ++ ++ /* DAI clock master masks */ ++ switch(rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_CLOCK_MASK){ ++ case SND_SOC_DAIFMT_CBM_CFM: ++ SSI1_SRCR |= SSI_SRCR_RFDIR | SSI_SRCR_RXDIR; ++ break; ++ case SND_SOC_DAIFMT_CBS_CFM: ++ SSI1_SRCR |= SSI_SRCR_RFDIR; ++ break; ++ case SND_SOC_DAIFMT_CBM_CFS: ++ SSI1_SRCR |= SSI_SRCR_RXDIR; ++ break; ++ } ++ ++ /* DAI BCLK ratio to SYSCLK / MCLK */ ++ /* prescaler modulus - todo */ ++ switch (bfs) { ++ case 2: ++ break; ++ case 4: ++ break; ++ case 8: ++ break; ++ case 16: ++ break; ++ } ++ ++ /* TDM - todo, only fifo 0 atm */ ++ SSI1_SRCR |= SSI_SRCR_RFEN0; ++ SSI1_SRCCR |= SSI_SRCCR_DC(params_channels(params)); ++ ++ return 0; ++} ++ ++static int imx_ssi1_hw_params(struct snd_pcm_substream *substream, ++ struct snd_pcm_hw_params *params) ++{ ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ ++ /* clear register if not enabled */ ++ if(!(SSI1_SCR & SSI_SCR_SSIEN)) ++ SSI1_SCR = 0; ++ ++ /* async */ ++ if (rtd->cpu_dai->flags & SND_SOC_DAI_ASYNC) ++ SSI1_SCR |= SSI_SCR_SYN; ++ ++ /* DAI mode */ ++ switch(rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_FORMAT_MASK) { ++ case SND_SOC_DAIFMT_I2S: ++ case SND_SOC_DAIFMT_LEFT_J: ++ SSI1_SCR |= SSI_SCR_NET; ++ break; ++ } ++ ++ /* TDM - to complete */ ++ ++ /* Tx/Rx config */ ++ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { ++ return imx_ssi1_hw_tx_params(substream, params); ++ } else { ++ return imx_ssi1_hw_rx_params(substream, params); ++ } ++} ++ ++ ++ ++static int imx_ssi1_trigger(struct snd_pcm_substream *substream, int cmd) ++{ ++ int ret = 0; ++ ++ switch (cmd) { ++ case SNDRV_PCM_TRIGGER_START: ++ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { ++ SSI1_SCR |= SSI_SCR_TE; ++ SSI1_SIER |= SSI_SIER_TDMAE; ++ } else { ++ SSI1_SCR |= SSI_SCR_RE; ++ SSI1_SIER |= SSI_SIER_RDMAE; ++ } ++ SSI1_SCR |= SSI_SCR_SSIEN; ++ ++ break; ++ case SNDRV_PCM_TRIGGER_RESUME: ++ break; ++ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: ++ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ++ SSI1_SCR |= SSI_SCR_TE; ++ else ++ SSI1_SCR |= SSI_SCR_RE; ++ break ++ case SNDRV_PCM_TRIGGER_STOP: ++ case SNDRV_PCM_TRIGGER_SUSPEND: ++ break; ++ case SNDRV_PCM_TRIGGER_PAUSE_PUSH: ++ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ++ SSI1_SCR &= ~SSI_SCR_TE; ++ else ++ SSI1_SCR &= ~SSI_SCR_RE; ++ break; ++ default: ++ ret = -EINVAL; ++ } ++ ++ return ret; ++} ++ ++static void imx_ssi_shutdown(struct snd_pcm_substream *substream) ++{ ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ ++ /* shutdown SSI */ ++ if (!rtd->cpu_dai->active) { ++ if(rtd->cpu_dai->id == 0) ++ SSI1_SCR &= ~SSI_SCR_SSIEN; ++ else ++ SSI2_SCR &= ~SSI_SCR_SSIEN; ++ } ++} ++ ++#ifdef CONFIG_PM ++static int imx_ssi_suspend(struct platform_device *dev, ++ struct snd_soc_cpu_dai *dai) ++{ ++ if(!dai->active) ++ return 0; ++ ++ if(rtd->cpu_dai->id == 0) ++ SSI1_SCR &= ~SSI_SCR_SSIEN; ++ else ++ SSI2_SCR &= ~SSI_SCR_SSIEN; ++ ++ return 0; ++} ++ ++static int imx_ssi_resume(struct platform_device *pdev, ++ struct snd_soc_cpu_dai *dai) ++{ ++ if(!dai->active) ++ return 0; ++ ++ if(rtd->cpu_dai->id == 0) ++ SSI1_SCR |= SSI_SCR_SSIEN; ++ else ++ SSI2_SCR |= SSI_SCR_SSIEN; ++ ++ return 0; ++} ++ ++#else ++#define imx_ssi_suspend NULL ++#define imx_ssi_resume NULL ++#endif ++ ++static unsigned int imx_ssi_config_pcm_sysclk(struct snd_soc_cpu_dai *iface, ++ struct snd_soc_clock_info *info, unsigned int clk) ++{ ++ return clk; ++} ++ ++struct snd_soc_cpu_dai imx_ssi_pcm_dai = { ++ .name = "imx-i2s-1", ++ .id = 0, ++ .type = SND_SOC_DAI_I2S, ++ .suspend = imx_ssi_suspend, ++ .resume = imx_ssi_resume, ++ .config_sysclk = imx_ssi_config_pcm_sysclk, ++ .playback = { ++ .channels_min = 1, ++ .channels_max = 2,}, ++ .capture = { ++ .channels_min = 1, ++ .channels_max = 2,}, ++ .ops = { ++ .startup = imx_ssi_startup, ++ .shutdown = imx_ssi_shutdown, ++ .trigger = imx_ssi1_trigger, ++ .hw_params = imx_ssi1_pcm_hw_params,}, ++ .caps = { ++ .num_modes = ARRAY_SIZE(imx_ssi_modes), ++ .mode = imx_ssi_modes,}, ++}, ++{ ++ .name = "imx-i2s-2", ++ .id = 1, ++ .type = SND_SOC_DAI_I2S, ++ .suspend = imx_ssi_suspend, ++ .resume = imx_ssi_resume, ++ .config_sysclk = imx_ssi_config_pcm_sysclk, ++ .playback = { ++ .channels_min = 1, ++ .channels_max = 2,}, ++ .capture = { ++ .channels_min = 1, ++ .channels_max = 2,}, ++ .ops = { ++ .startup = imx_ssi_startup, ++ .shutdown = imx_ssi_shutdown, ++ .trigger = imx_ssi1_trigger, ++ .hw_params = imx_ssi1_pcm_hw_params,}, ++ .caps = { ++ .num_modes = ARRAY_SIZE(imx_ssi_modes), ++ .mode = imx_ssi_modes,}, ++}; ++ ++ ++EXPORT_SYMBOL_GPL(imx_ssi_i2s_dai); ++ ++/* Module information */ ++MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com"); ++MODULE_DESCRIPTION("i.MX ASoC I2S driver"); ++MODULE_LICENSE("GPL"); +Index: linux-2.6-pxa-new/include/linux/i2c-id.h +=================================================================== +--- linux-2.6-pxa-new.orig/include/linux/i2c-id.h ++++ linux-2.6-pxa-new/include/linux/i2c-id.h +@@ -116,6 +116,8 @@ + #define I2C_DRIVERID_KS0127 86 /* Samsung ks0127 video decoder */ + #define I2C_DRIVERID_TLV320AIC23B 87 /* TI TLV320AIC23B audio codec */ + #define I2C_DRIVERID_ISL1208 88 /* Intersil ISL1208 RTC */ ++#define I2C_DRIVERID_WM8731 89 /* Wolfson WM8731 audio codec */ ++#define I2C_DRIVERID_WM8750 90 /* Wolfson WM8750 audio codec */ + + #define I2C_DRIVERID_I2CDEV 900 + #define I2C_DRIVERID_ARP 902 /* SMBus ARP Client */ +Index: linux-2.6-pxa-new/sound/soc/codecs/wm8976.c +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/sound/soc/codecs/wm8976.c +@@ -0,0 +1,953 @@ ++/* ++ * wm8976.c -- WM8976 ALSA Soc Audio driver ++ * ++ * Copyright 2006 Wolfson Microelectronics PLC. ++ * ++ * This program is free software; you can redistribute it and/or modify ++ * it under the terms of the GNU General Public License version 2 as ++ * published by the Free Software Foundation. ++ */ ++ ++#include <linux/module.h> ++#include <linux/moduleparam.h> ++#include <linux/version.h> ++#include <linux/kernel.h> ++#include <linux/init.h> ++#include <linux/delay.h> ++#include <linux/pm.h> ++#include <linux/i2c.h> ++#include <linux/platform_device.h> ++#include <sound/driver.h> ++#include <sound/core.h> ++#include <sound/pcm.h> ++#include <sound/pcm_params.h> ++#include <sound/soc.h> ++#include <sound/soc-dapm.h> ++#include <sound/initval.h> ++ ++#include "wm8976.h" ++ ++#define AUDIO_NAME "wm8976" ++#define WM8976_VERSION "0.2" ++ ++/* ++ * Debug ++ */ ++ ++#define WM8976_DEBUG 0 ++ ++#ifdef WM8976_DEBUG ++#define dbg(format, arg...) \ ++ printk(KERN_DEBUG AUDIO_NAME ": " format "\n" , ## arg) ++#else ++#define dbg(format, arg...) do {} while (0) ++#endif ++#define err(format, arg...) \ ++ printk(KERN_ERR AUDIO_NAME ": " format "\n" , ## arg) ++#define info(format, arg...) \ ++ printk(KERN_INFO AUDIO_NAME ": " format "\n" , ## arg) ++#define warn(format, arg...) \ ++ printk(KERN_WARNING AUDIO_NAME ": " format "\n" , ## arg) ++ ++struct snd_soc_codec_device soc_codec_dev_wm8976; ++ ++/* ++ * wm8976 register cache ++ * We can't read the WM8976 register space when we are ++ * using 2 wire for device control, so we cache them instead. ++ */ ++static const u16 wm8976_reg[WM8976_CACHEREGNUM] = { ++ 0x0000, 0x0000, 0x0000, 0x0000, ++ 0x0050, 0x0000, 0x0140, 0x0000, ++ 0x0000, 0x0000, 0x0000, 0x00ff, ++ 0x00ff, 0x0000, 0x0100, 0x00ff, ++ 0x00ff, 0x0000, 0x012c, 0x002c, ++ 0x002c, 0x002c, 0x002c, 0x0000, ++ 0x0032, 0x0000, 0x0000, 0x0000, ++ 0x0000, 0x0000, 0x0000, 0x0000, ++ 0x0038, 0x000b, 0x0032, 0x0000, ++ 0x0008, 0x000c, 0x0093, 0x00e9, ++ 0x0000, 0x0000, 0x0000, 0x0000, ++ 0x0033, 0x0010, 0x0010, 0x0100, ++ 0x0100, 0x0002, 0x0001, 0x0001, ++ 0x0039, 0x0039, 0x0039, 0x0039, ++ 0x0001, 0x0001, ++}; ++ ++#define WM8976_DAIFMT \ ++ (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_RIGHT_J | \ ++ SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF | \ ++ SND_SOC_DAIFMT_NB_IF | SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_IB_IF) ++ ++#define WM8976_DIR \ ++ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE) ++ ++#define WM8976_RATES \ ++ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ ++ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ ++ SNDRV_PCM_RATE_48000) ++ ++#define WM8976_PCM_FORMATS \ ++ (SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S20_3LE | \ ++ SNDRV_PCM_FORMAT_S24_3LE | SNDRV_PCM_FORMAT_S24_LE | \ ++ SNDRV_PCM_FORMAT_S32_LE) ++ ++#define WM8976_BCLK \ ++ (SND_SOC_FSBD(1) | SND_SOC_FSBD(2) | SND_SOC_FSBD(4) | SND_SOC_FSBD(8) |\ ++ SND_SOC_FSBD(16) | SND_SOC_FSBD(32)) ++ ++static struct snd_soc_dai_mode wm8976_modes[] = { ++ /* codec frame and clock master modes */ ++ { ++ .fmt = WM8976_DAIFMT | SND_SOC_DAIFMT_CBM_CFM, ++ .pcmfmt = WM8976_PCM_FORMATS, ++ .pcmrate = WM8976_RATES, ++ .pcmdir = WM8976_DIR, ++ .flags = SND_SOC_DAI_BFS_DIV, ++ .fs = 256, ++ .bfs = WM8976_BCLK, ++ }, ++ ++ /* codec frame and clock slave modes */ ++ { ++ .fmt = WM8976_DAIFMT | SND_SOC_DAIFMT_CBS_CFS, ++ .pcmfmt = WM8976_PCM_FORMATS, ++ .pcmrate = WM8976_RATES, ++ .pcmdir = WM8976_DIR, ++ .fs = SND_SOC_FS_ALL, ++ .bfs = SND_SOC_FSB_ALL, ++ }, ++}; ++ ++/* ++ * read wm8976 register cache ++ */ ++static inline unsigned int wm8976_read_reg_cache(struct snd_soc_codec *codec, ++ unsigned int reg) ++{ ++ u16 *cache = codec->reg_cache; ++ if (reg == WM8976_RESET) ++ return 0; ++ if (reg >= WM8976_CACHEREGNUM) ++ return -1; ++ return cache[reg]; ++} ++ ++/* ++ * write wm8976 register cache ++ */ ++static inline void wm8976_write_reg_cache(struct snd_soc_codec *codec, ++ u16 reg, unsigned int value) ++{ ++ u16 *cache = codec->reg_cache; ++ if (reg >= WM8976_CACHEREGNUM) ++ return; ++ cache[reg] = value; ++} ++ ++/* ++ * write to the WM8976 register space ++ */ ++static int wm8976_write(struct snd_soc_codec *codec, unsigned int reg, ++ unsigned int value) ++{ ++ u8 data[2]; ++ ++ /* data is ++ * D15..D9 WM8976 register offset ++ * D8...D0 register data ++ */ ++ data[0] = (reg << 1) | ((value >> 8) & 0x0001); ++ data[1] = value & 0x00ff; ++ ++ wm8976_write_reg_cache (codec, reg, value); ++ if (codec->hw_write(codec->control_data, data, 2) == 2) ++ return 0; ++ else ++ return -1; ++} ++ ++#define wm8976_reset(c) wm8976_write(c, WM8976_RESET, 0) ++ ++static const char *wm8976_companding[] = {"Off", "NC", "u-law", "A-law" }; ++static const char *wm8976_deemp[] = {"None", "32kHz", "44.1kHz", "48kHz" }; ++static const char *wm8976_eqmode[] = {"Capture", "Playback" }; ++static const char *wm8976_bw[] = {"Narrow", "Wide" }; ++static const char *wm8976_eq1[] = {"80Hz", "105Hz", "135Hz", "175Hz" }; ++static const char *wm8976_eq2[] = {"230Hz", "300Hz", "385Hz", "500Hz" }; ++static const char *wm8976_eq3[] = {"650Hz", "850Hz", "1.1kHz", "1.4kHz" }; ++static const char *wm8976_eq4[] = {"1.8kHz", "2.4kHz", "3.2kHz", "4.1kHz" }; ++static const char *wm8976_eq5[] = {"5.3kHz", "6.9kHz", "9kHz", "11.7kHz" }; ++static const char *wm8976_alc[] = ++ {"ALC both on", "ALC left only", "ALC right only", "Limiter" }; ++ ++static const struct soc_enum wm8976_enum[] = { ++ SOC_ENUM_SINGLE(WM8976_COMP, 1, 4, wm8976_companding), /* adc */ ++ SOC_ENUM_SINGLE(WM8976_COMP, 3, 4, wm8976_companding), /* dac */ ++ SOC_ENUM_SINGLE(WM8976_DAC, 4, 4, wm8976_deemp), ++ SOC_ENUM_SINGLE(WM8976_EQ1, 8, 2, wm8976_eqmode), ++ ++ SOC_ENUM_SINGLE(WM8976_EQ1, 5, 4, wm8976_eq1), ++ SOC_ENUM_SINGLE(WM8976_EQ2, 8, 2, wm8976_bw), ++ SOC_ENUM_SINGLE(WM8976_EQ2, 5, 4, wm8976_eq2), ++ SOC_ENUM_SINGLE(WM8976_EQ3, 8, 2, wm8976_bw), ++ ++ SOC_ENUM_SINGLE(WM8976_EQ3, 5, 4, wm8976_eq3), ++ SOC_ENUM_SINGLE(WM8976_EQ4, 8, 2, wm8976_bw), ++ SOC_ENUM_SINGLE(WM8976_EQ4, 5, 4, wm8976_eq4), ++ SOC_ENUM_SINGLE(WM8976_EQ5, 8, 2, wm8976_bw), ++ ++ SOC_ENUM_SINGLE(WM8976_EQ5, 5, 4, wm8976_eq5), ++ SOC_ENUM_SINGLE(WM8976_ALC3, 8, 2, wm8976_alc), ++}; ++ ++static const struct snd_kcontrol_new wm8976_snd_controls[] = { ++SOC_SINGLE("Digital Loopback Switch", WM8976_COMP, 0, 1, 0), ++ ++SOC_ENUM("ADC Companding", wm8976_enum[0]), ++SOC_ENUM("DAC Companding", wm8976_enum[1]), ++ ++SOC_SINGLE("Jack Detection Enable", WM8976_JACK1, 6, 1, 0), ++ ++SOC_DOUBLE("DAC Inversion Switch", WM8976_DAC, 0, 1, 1, 0), ++ ++SOC_DOUBLE_R("Headphone Playback Volume", WM8976_DACVOLL, WM8976_DACVOLR, 0, 127, 0), ++ ++SOC_SINGLE("High Pass Filter Switch", WM8976_ADC, 8, 1, 0), ++SOC_SINGLE("High Pass Filter Switch", WM8976_ADC, 8, 1, 0), ++SOC_SINGLE("High Pass Cut Off", WM8976_ADC, 4, 7, 0), ++ ++SOC_DOUBLE("ADC Inversion Switch", WM8976_ADC, 0, 1, 1, 0), ++ ++SOC_SINGLE("Capture Volume", WM8976_ADCVOL, 0, 127, 0), ++ ++SOC_ENUM("Equaliser Function", wm8976_enum[3]), ++SOC_ENUM("EQ1 Cut Off", wm8976_enum[4]), ++SOC_SINGLE("EQ1 Volume", WM8976_EQ1, 0, 31, 1), ++ ++SOC_ENUM("Equaliser EQ2 Bandwith", wm8976_enum[5]), ++SOC_ENUM("EQ2 Cut Off", wm8976_enum[6]), ++SOC_SINGLE("EQ2 Volume", WM8976_EQ2, 0, 31, 1), ++ ++SOC_ENUM("Equaliser EQ3 Bandwith", wm8976_enum[7]), ++SOC_ENUM("EQ3 Cut Off", wm8976_enum[8]), ++SOC_SINGLE("EQ3 Volume", WM8976_EQ3, 0, 31, 1), ++ ++SOC_ENUM("Equaliser EQ4 Bandwith", wm8976_enum[9]), ++SOC_ENUM("EQ4 Cut Off", wm8976_enum[10]), ++SOC_SINGLE("EQ4 Volume", WM8976_EQ4, 0, 31, 1), ++ ++SOC_ENUM("Equaliser EQ5 Bandwith", wm8976_enum[11]), ++SOC_ENUM("EQ5 Cut Off", wm8976_enum[12]), ++SOC_SINGLE("EQ5 Volume", WM8976_EQ5, 0, 31, 1), ++ ++SOC_SINGLE("DAC Playback Limiter Switch", WM8976_DACLIM1, 8, 1, 0), ++SOC_SINGLE("DAC Playback Limiter Decay", WM8976_DACLIM1, 4, 15, 0), ++SOC_SINGLE("DAC Playback Limiter Attack", WM8976_DACLIM1, 0, 15, 0), ++ ++SOC_SINGLE("DAC Playback Limiter Threshold", WM8976_DACLIM2, 4, 7, 0), ++SOC_SINGLE("DAC Playback Limiter Boost", WM8976_DACLIM2, 0, 15, 0), ++ ++SOC_SINGLE("ALC Enable Switch", WM8976_ALC1, 8, 1, 0), ++SOC_SINGLE("ALC Capture Max Gain", WM8976_ALC1, 3, 7, 0), ++SOC_SINGLE("ALC Capture Min Gain", WM8976_ALC1, 0, 7, 0), ++ ++SOC_SINGLE("ALC Capture ZC Switch", WM8976_ALC2, 8, 1, 0), ++SOC_SINGLE("ALC Capture Hold", WM8976_ALC2, 4, 7, 0), ++SOC_SINGLE("ALC Capture Target", WM8976_ALC2, 0, 15, 0), ++ ++SOC_ENUM("ALC Capture Mode", wm8976_enum[13]), ++SOC_SINGLE("ALC Capture Decay", WM8976_ALC3, 4, 15, 0), ++SOC_SINGLE("ALC Capture Attack", WM8976_ALC3, 0, 15, 0), ++ ++SOC_SINGLE("ALC Capture Noise Gate Switch", WM8976_NGATE, 3, 1, 0), ++SOC_SINGLE("ALC Capture Noise Gate Threshold", WM8976_NGATE, 0, 7, 0), ++ ++SOC_SINGLE("Capture PGA ZC Switch", WM8976_INPPGA, 7, 1, 0), ++SOC_SINGLE("Capture PGA Volume", WM8976_INPPGA, 0, 63, 0), ++ ++SOC_DOUBLE_R("Headphone Playback ZC Switch", WM8976_HPVOLL, WM8976_HPVOLR, 7, 1, 0), ++SOC_DOUBLE_R("Headphone Playback Switch", WM8976_HPVOLL, WM8976_HPVOLR, 6, 1, 1), ++SOC_DOUBLE_R("Headphone Playback Volume", WM8976_HPVOLL, WM8976_HPVOLR, 0, 63, 0), ++ ++SOC_DOUBLE_R("Speaker Playback ZC Switch", WM8976_SPKVOLL, WM8976_SPKVOLR, 7, 1, 0), ++SOC_DOUBLE_R("Speaker Playback Switch", WM8976_SPKVOLL, WM8976_SPKVOLR, 6, 1, 1), ++SOC_DOUBLE_R("Speaker Playback Volume", WM8976_SPKVOLL, WM8976_SPKVOLR, 0, 63, 0), ++ ++SOC_SINGLE("Capture Boost(+20dB)", WM8976_ADCBOOST, 8, 1, 0), ++}; ++ ++/* add non dapm controls */ ++static int wm8976_add_controls(struct snd_soc_codec *codec) ++{ ++ int err, i; ++ ++ for (i = 0; i < ARRAY_SIZE(wm8976_snd_controls); i++) { ++ err = snd_ctl_add(codec->card, ++ snd_soc_cnew(&wm8976_snd_controls[i],codec, NULL)); ++ if (err < 0) ++ return err; ++ } ++ ++ return 0; ++} ++ ++/* Left Output Mixer */ ++static const snd_kcontrol_new_t wm8976_left_mixer_controls[] = { ++SOC_DAPM_SINGLE("Right PCM Playback Switch", WM8976_OUTPUT, 6, 1, 1), ++SOC_DAPM_SINGLE("Left PCM Playback Switch", WM8976_MIXL, 0, 1, 1), ++SOC_DAPM_SINGLE("Line Bypass Switch", WM8976_MIXL, 1, 1, 0), ++SOC_DAPM_SINGLE("Aux Playback Switch", WM8976_MIXL, 5, 1, 0), ++}; ++ ++/* Right Output Mixer */ ++static const snd_kcontrol_new_t wm8976_right_mixer_controls[] = { ++SOC_DAPM_SINGLE("Left PCM Playback Switch", WM8976_OUTPUT, 5, 1, 1), ++SOC_DAPM_SINGLE("Right PCM Playback Switch", WM8976_MIXR, 0, 1, 1), ++SOC_DAPM_SINGLE("Line Bypass Switch", WM8976_MIXR, 1, 1, 0), ++SOC_DAPM_SINGLE("Aux Playback Switch", WM8976_MIXR, 5, 1, 0), ++}; ++ ++/* Left AUX Input boost vol */ ++static const snd_kcontrol_new_t wm8976_laux_boost_controls = ++SOC_DAPM_SINGLE("Aux Volume", WM8976_ADCBOOST, 0, 3, 0); ++ ++/* Left Input boost vol */ ++static const snd_kcontrol_new_t wm8976_lmic_boost_controls = ++SOC_DAPM_SINGLE("Input Volume", WM8976_ADCBOOST, 4, 3, 0); ++ ++/* Left Aux In to PGA */ ++static const snd_kcontrol_new_t wm8976_laux_capture_boost_controls = ++SOC_DAPM_SINGLE("Capture Switch", WM8976_ADCBOOST, 8, 1, 0); ++ ++/* Left Input P In to PGA */ ++static const snd_kcontrol_new_t wm8976_lmicp_capture_boost_controls = ++SOC_DAPM_SINGLE("Input P Capture Boost Switch", WM8976_INPUT, 0, 1, 0); ++ ++/* Left Input N In to PGA */ ++static const snd_kcontrol_new_t wm8976_lmicn_capture_boost_controls = ++SOC_DAPM_SINGLE("Input N Capture Boost Switch", WM8976_INPUT, 1, 1, 0); ++ ++// TODO Widgets ++static const struct snd_soc_dapm_widget wm8976_dapm_widgets[] = { ++#if 0 ++//SND_SOC_DAPM_MUTE("Mono Mute", WM8976_MONOMIX, 6, 0), ++//SND_SOC_DAPM_MUTE("Speaker Mute", WM8976_SPKMIX, 6, 0), ++ ++SND_SOC_DAPM_MIXER("Speaker Mixer", WM8976_POWER3, 2, 0, ++ &wm8976_speaker_mixer_controls[0], ++ ARRAY_SIZE(wm8976_speaker_mixer_controls)), ++SND_SOC_DAPM_MIXER("Mono Mixer", WM8976_POWER3, 3, 0, ++ &wm8976_mono_mixer_controls[0], ++ ARRAY_SIZE(wm8976_mono_mixer_controls)), ++SND_SOC_DAPM_DAC("DAC", "HiFi Playback", WM8976_POWER3, 0, 0), ++SND_SOC_DAPM_ADC("ADC", "HiFi Capture", WM8976_POWER3, 0, 0), ++SND_SOC_DAPM_PGA("Aux Input", WM8976_POWER1, 6, 0, NULL, 0), ++SND_SOC_DAPM_PGA("SpkN Out", WM8976_POWER3, 5, 0, NULL, 0), ++SND_SOC_DAPM_PGA("SpkP Out", WM8976_POWER3, 6, 0, NULL, 0), ++SND_SOC_DAPM_PGA("Mono Out", WM8976_POWER3, 7, 0, NULL, 0), ++SND_SOC_DAPM_PGA("Mic PGA", WM8976_POWER2, 2, 0, NULL, 0), ++ ++SND_SOC_DAPM_PGA("Aux Boost", SND_SOC_NOPM, 0, 0, ++ &wm8976_aux_boost_controls, 1), ++SND_SOC_DAPM_PGA("Mic Boost", SND_SOC_NOPM, 0, 0, ++ &wm8976_mic_boost_controls, 1), ++SND_SOC_DAPM_SWITCH("Capture Boost", SND_SOC_NOPM, 0, 0, ++ &wm8976_capture_boost_controls), ++ ++SND_SOC_DAPM_MIXER("Boost Mixer", WM8976_POWER2, 4, 0, NULL, 0), ++ ++SND_SOC_DAPM_MICBIAS("Mic Bias", WM8976_POWER1, 4, 0), ++ ++SND_SOC_DAPM_INPUT("MICN"), ++SND_SOC_DAPM_INPUT("MICP"), ++SND_SOC_DAPM_INPUT("AUX"), ++SND_SOC_DAPM_OUTPUT("MONOOUT"), ++SND_SOC_DAPM_OUTPUT("SPKOUTP"), ++SND_SOC_DAPM_OUTPUT("SPKOUTN"), ++#endif ++}; ++ ++static const char *audio_map[][3] = { ++ /* Mono output mixer */ ++ {"Mono Mixer", "PCM Playback Switch", "DAC"}, ++ {"Mono Mixer", "Aux Playback Switch", "Aux Input"}, ++ {"Mono Mixer", "Line Bypass Switch", "Boost Mixer"}, ++ ++ /* Speaker output mixer */ ++ {"Speaker Mixer", "PCM Playback Switch", "DAC"}, ++ {"Speaker Mixer", "Aux Playback Switch", "Aux Input"}, ++ {"Speaker Mixer", "Line Bypass Switch", "Boost Mixer"}, ++ ++ /* Outputs */ ++ {"Mono Out", NULL, "Mono Mixer"}, ++ {"MONOOUT", NULL, "Mono Out"}, ++ {"SpkN Out", NULL, "Speaker Mixer"}, ++ {"SpkP Out", NULL, "Speaker Mixer"}, ++ {"SPKOUTN", NULL, "SpkN Out"}, ++ {"SPKOUTP", NULL, "SpkP Out"}, ++ ++ /* Boost Mixer */ ++ {"Boost Mixer", NULL, "ADC"}, ++ {"Capture Boost Switch", "Aux Capture Boost Switch", "AUX"}, ++ {"Aux Boost", "Aux Volume", "Boost Mixer"}, ++ {"Capture Boost", "Capture Switch", "Boost Mixer"}, ++ {"Mic Boost", "Mic Volume", "Boost Mixer"}, ++ ++ /* Inputs */ ++ {"MICP", NULL, "Mic Boost"}, ++ {"MICN", NULL, "Mic PGA"}, ++ {"Mic PGA", NULL, "Capture Boost"}, ++ {"AUX", NULL, "Aux Input"}, ++ ++ /* */ ++ ++ /* terminator */ ++ {NULL, NULL, NULL}, ++}; ++ ++static int wm8976_add_widgets(struct snd_soc_codec *codec) ++{ ++ int i; ++ ++ for(i = 0; i < ARRAY_SIZE(wm8976_dapm_widgets); i++) { ++ snd_soc_dapm_new_control(codec, &wm8976_dapm_widgets[i]); ++ } ++ ++ /* set up audio path map */ ++ for(i = 0; audio_map[i][0] != NULL; i++) { ++ snd_soc_dapm_connect_input(codec, audio_map[i][0], audio_map[i][1], ++ audio_map[i][2]); ++ } ++ ++ snd_soc_dapm_new_widgets(codec); ++ return 0; ++} ++ ++struct pll_ { ++ unsigned int in_hz, out_hz; ++ unsigned int pre:4; /* prescale - 1 */ ++ unsigned int n:4; ++ unsigned int k; ++}; ++ ++struct pll_ pll[] = { ++ {12000000, 11289600, 0, 7, 0x86c220}, ++ {12000000, 12288000, 0, 8, 0x3126e8}, ++ {13000000, 11289600, 0, 6, 0xf28bd4}, ++ {13000000, 12288000, 0, 7, 0x8fd525}, ++ {12288000, 11289600, 0, 7, 0x59999a}, ++ {11289600, 12288000, 0, 8, 0x80dee9}, ++ /* TODO: liam - add more entries */ ++}; ++ ++static int set_pll(struct snd_soc_codec *codec, unsigned int in, ++ unsigned int out) ++{ ++ int i; ++ u16 reg; ++ ++ if(out == 0) { ++ reg = wm8976_read_reg_cache(codec, WM8976_POWER1); ++ wm8976_write(codec, WM8976_POWER1, reg & 0x1df); ++ return 0; ++ } ++ ++ for(i = 0; i < ARRAY_SIZE(pll); i++) { ++ if (in == pll[i].in_hz && out == pll[i].out_hz) { ++ wm8976_write(codec, WM8976_PLLN, (pll[i].pre << 4) | pll[i].n); ++ wm8976_write(codec, WM8976_PLLK1, pll[i].k >> 18); ++ wm8976_write(codec, WM8976_PLLK1, (pll[i].k >> 9) && 0x1ff); ++ wm8976_write(codec, WM8976_PLLK1, pll[i].k && 0x1ff); ++ reg = wm8976_read_reg_cache(codec, WM8976_POWER1); ++ wm8976_write(codec, WM8976_POWER1, reg | 0x020); ++ return 0; ++ } ++ } ++ return -EINVAL; ++} ++ ++/* mclk dividers * 2 */ ++static unsigned char mclk_div[] = {2, 3, 4, 6, 8, 12, 16, 24}; ++ ++/* we need 256FS to drive the DAC's and ADC's */ ++static unsigned int wm8976_config_sysclk(struct snd_soc_codec_dai *dai, ++ struct snd_soc_clock_info *info, unsigned int clk) ++{ ++ int i, j, best_clk = info->fs * info->rate; ++ ++ /* can we run at this clk without the PLL ? */ ++ for (i = 0; i < ARRAY_SIZE(mclk_div); i++) { ++ if ((best_clk >> 1) * mclk_div[i] == clk) { ++ dai->pll_in = 0; ++ dai->clk_div = mclk_div[i]; ++ dai->mclk = best_clk; ++ return dai->mclk; ++ } ++ } ++ ++ /* now check for PLL support */ ++ for (i = 0; i < ARRAY_SIZE(pll); i++) { ++ if (pll[i].in_hz == clk) { ++ for (j = 0; j < ARRAY_SIZE(mclk_div); j++) { ++ if (pll[i].out_hz == mclk_div[j] * (best_clk >> 1)) { ++ dai->pll_in = clk; ++ dai->pll_out = pll[i].out_hz; ++ dai->clk_div = mclk_div[j]; ++ dai->mclk = best_clk; ++ return dai->mclk; ++ } ++ } ++ } ++ } ++ ++ /* this clk is not supported */ ++ return 0; ++} ++ ++static int wm8976_pcm_prepare(snd_pcm_substream_t *substream) ++{ ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ struct snd_soc_device *socdev = rtd->socdev; ++ struct snd_soc_codec *codec = socdev->codec; ++ struct snd_soc_codec_dai *dai = rtd->codec_dai; ++ u16 iface = 0, bfs, clk = 0, adn; ++ int fs = 48000 << 7, i; ++ ++ bfs = SND_SOC_FSBD_REAL(rtd->codec_dai->dai_runtime.bfs); ++ switch (bfs) { ++ case 2: ++ clk |= 0x1 << 2; ++ break; ++ case 4: ++ clk |= 0x2 << 2; ++ break; ++ case 8: ++ clk |= 0x3 << 2; ++ break; ++ case 16: ++ clk |= 0x4 << 2; ++ break; ++ case 32: ++ clk |= 0x5 << 2; ++ break; ++ } ++ ++ /* set master/slave audio interface */ ++ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_CLOCK_MASK) { ++ case SND_SOC_DAIFMT_CBM_CFM: ++ clk |= 0x0001; ++ break; ++ case SND_SOC_DAIFMT_CBS_CFS: ++ break; ++ } ++ ++ /* interface format */ ++ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_FORMAT_MASK) { ++ case SND_SOC_DAIFMT_I2S: ++ iface |= 0x0010; ++ break; ++ case SND_SOC_DAIFMT_RIGHT_J: ++ break; ++ case SND_SOC_DAIFMT_LEFT_J: ++ iface |= 0x0008; ++ break; ++ case SND_SOC_DAIFMT_DSP_A: ++ iface |= 0x00018; ++ break; ++ } ++ ++ /* bit size */ ++ switch (rtd->codec_dai->dai_runtime.pcmfmt) { ++ case SNDRV_PCM_FMTBIT_S16_LE: ++ break; ++ case SNDRV_PCM_FMTBIT_S20_3LE: ++ iface |= 0x0020; ++ break; ++ case SNDRV_PCM_FMTBIT_S24_LE: ++ iface |= 0x0040; ++ break; ++ case SNDRV_PCM_FMTBIT_S32_LE: ++ iface |= 0x0060; ++ break; ++ } ++ ++ /* clock inversion */ ++ switch (rtd->codec_dai->dai_runtime.fmt & SND_SOC_DAIFMT_INV_MASK) { ++ case SND_SOC_DAIFMT_NB_NF: ++ break; ++ case SND_SOC_DAIFMT_IB_IF: ++ iface |= 0x0180; ++ break; ++ case SND_SOC_DAIFMT_IB_NF: ++ iface |= 0x0100; ++ break; ++ case SND_SOC_DAIFMT_NB_IF: ++ iface |= 0x0080; ++ break; ++ } ++ ++ /* filter coefficient */ ++ adn = wm8976_read_reg_cache(codec, WM8976_ADD) & 0x1f1; ++ switch (rtd->codec_dai->dai_runtime.pcmrate) { ++ case SNDRV_PCM_RATE_8000: ++ adn |= 0x5 << 1; ++ fs = 8000 << 7; ++ break; ++ case SNDRV_PCM_RATE_11025: ++ adn |= 0x4 << 1; ++ fs = 11025 << 7; ++ break; ++ case SNDRV_PCM_RATE_16000: ++ adn |= 0x3 << 1; ++ fs = 16000 << 7; ++ break; ++ case SNDRV_PCM_RATE_22050: ++ adn |= 0x2 << 1; ++ fs = 22050 << 7; ++ break; ++ case SNDRV_PCM_RATE_32000: ++ adn |= 0x1 << 1; ++ fs = 32000 << 7; ++ break; ++ case SNDRV_PCM_RATE_44100: ++ fs = 44100 << 7; ++ break; ++ } ++ ++ /* do we need to enable the PLL */ ++ if(dai->pll_in) ++ set_pll(codec, dai->pll_in, dai->pll_out); ++ ++ /* divide the clock to 256 fs */ ++ for(i = 0; i < ARRAY_SIZE(mclk_div); i++) { ++ if (dai->clk_div == mclk_div[i]) { ++ clk |= i << 5; ++ clk &= 0xff; ++ goto set; ++ } ++ } ++ ++set: ++ /* set iface */ ++ wm8976_write(codec, WM8976_IFACE, iface); ++ wm8976_write(codec, WM8976_CLOCK, clk); ++ ++ return 0; ++} ++ ++static int wm8976_hw_free(struct snd_pcm_substream *substream) ++{ ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ struct snd_soc_device *socdev = rtd->socdev; ++ struct snd_soc_codec *codec = socdev->codec; ++ set_pll(codec, 0, 0); ++ return 0; ++} ++ ++static int wm8976_mute(struct snd_soc_codec *codec, ++ struct snd_soc_codec_dai *dai, int mute) ++{ ++ u16 mute_reg = wm8976_read_reg_cache(codec, WM8976_DAC) & 0xffbf; ++ if(mute) ++ wm8976_write(codec, WM8976_DAC, mute_reg | 0x40); ++ else ++ wm8976_write(codec, WM8976_DAC, mute_reg); ++ ++ return 0; ++} ++ ++/* TODO: liam need to make this lower power with dapm */ ++static int wm8976_dapm_event(struct snd_soc_codec *codec, int event) ++{ ++ ++ switch (event) { ++ case SNDRV_CTL_POWER_D0: /* full On */ ++ /* vref/mid, clk and osc on, dac unmute, active */ ++ wm8976_write(codec, WM8976_POWER1, 0x1ff); ++ wm8976_write(codec, WM8976_POWER2, 0x1ff); ++ wm8976_write(codec, WM8976_POWER3, 0x1ff); ++ break; ++ case SNDRV_CTL_POWER_D1: /* partial On */ ++ case SNDRV_CTL_POWER_D2: /* partial On */ ++ break; ++ case SNDRV_CTL_POWER_D3hot: /* Off, with power */ ++ /* everything off except vref/vmid, dac mute, inactive */ ++ ++ break; ++ case SNDRV_CTL_POWER_D3cold: /* Off, without power */ ++ /* everything off, dac mute, inactive */ ++ wm8976_write(codec, WM8976_POWER1, 0x0); ++ wm8976_write(codec, WM8976_POWER2, 0x0); ++ wm8976_write(codec, WM8976_POWER3, 0x0); ++ break; ++ } ++ codec->dapm_state = event; ++ return 0; ++} ++ ++struct snd_soc_codec_dai wm8976_dai = { ++ .name = "WM8976 HiFi", ++ .playback = { ++ .stream_name = "Playback", ++ .channels_min = 1, ++ .channels_max = 2, ++ }, ++ .capture = { ++ .stream_name = "Capture", ++ .channels_min = 1, ++ .channels_max = 1, ++ }, ++ .config_sysclk = wm8976_config_sysclk, ++ .digital_mute = wm8976_mute, ++ .ops = { ++ .prepare = wm8976_pcm_prepare, ++ .hw_free = wm8976_hw_free, ++ }, ++ .caps = { ++ .num_modes = ARRAY_SIZE(wm8976_modes), ++ .mode = wm8976_modes, ++ }, ++}; ++EXPORT_SYMBOL_GPL(wm8976_dai); ++ ++static int wm8976_suspend(struct platform_device *pdev, pm_message_t state) ++{ ++ struct snd_soc_device *socdev = platform_get_drvdata(pdev); ++ struct snd_soc_codec *codec = socdev->codec; ++ ++ wm8976_dapm_event(codec, SNDRV_CTL_POWER_D3cold); ++ return 0; ++} ++ ++static int wm8976_resume(struct platform_device *pdev) ++{ ++ struct snd_soc_device *socdev = platform_get_drvdata(pdev); ++ struct snd_soc_codec *codec = socdev->codec; ++ int i; ++ u8 data[2]; ++ u16 *cache = codec->reg_cache; ++ ++ /* Sync reg_cache with the hardware */ ++ for (i = 0; i < ARRAY_SIZE(wm8976_reg); i++) { ++ data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); ++ data[1] = cache[i] & 0x00ff; ++ codec->hw_write(codec->control_data, data, 2); ++ } ++ wm8976_dapm_event(codec, SNDRV_CTL_POWER_D3hot); ++ wm8976_dapm_event(codec, codec->suspend_dapm_state); ++ return 0; ++} ++ ++/* ++ * initialise the WM8976 driver ++ * register the mixer and dsp interfaces with the kernel ++ */ ++static int wm8976_init(struct snd_soc_device* socdev) ++{ ++ struct snd_soc_codec *codec = socdev->codec; ++ int ret = 0; ++ ++ codec->name = "WM8976"; ++ codec->owner = THIS_MODULE; ++ codec->read = wm8976_read_reg_cache; ++ codec->write = wm8976_write; ++ codec->dapm_event = wm8976_dapm_event; ++ codec->dai = &wm8976_dai; ++ codec->num_dai = 1; ++ codec->reg_cache_size = ARRAY_SIZE(wm8976_reg); ++ codec->reg_cache = ++ kzalloc(sizeof(u16) * ARRAY_SIZE(wm8976_reg), GFP_KERNEL); ++ if (codec->reg_cache == NULL) ++ return -ENOMEM; ++ memcpy(codec->reg_cache, wm8976_reg, ++ sizeof(u16) * ARRAY_SIZE(wm8976_reg)); ++ codec->reg_cache_size = sizeof(u16) * ARRAY_SIZE(wm8976_reg); ++ ++ wm8976_reset(codec); ++ ++ /* register pcms */ ++ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); ++ if(ret < 0) { ++ kfree(codec->reg_cache); ++ return ret; ++ } ++ ++ /* power on device */ ++ wm8976_dapm_event(codec, SNDRV_CTL_POWER_D3hot); ++ wm8976_add_controls(codec); ++ wm8976_add_widgets(codec); ++ ret = snd_soc_register_card(socdev); ++ if(ret < 0) { ++ snd_soc_free_pcms(socdev); ++ snd_soc_dapm_free(socdev); ++ } ++ ++ return ret; ++} ++ ++static struct snd_soc_device *wm8976_socdev; ++ ++#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) ++ ++/* ++ * WM8976 2 wire address is 0x1a ++ */ ++#define I2C_DRIVERID_WM8976 0xfefe /* liam - need a proper id */ ++ ++static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END }; ++ ++/* Magic definition of all other variables and things */ ++I2C_CLIENT_INSMOD; ++ ++static struct i2c_driver wm8976_i2c_driver; ++static struct i2c_client client_template; ++ ++/* If the i2c layer weren't so broken, we could pass this kind of data ++ around */ ++ ++static int wm8976_codec_probe(struct i2c_adapter *adap, int addr, int kind) ++{ ++ struct snd_soc_device *socdev = wm8976_socdev; ++ struct wm8976_setup_data *setup = socdev->codec_data; ++ struct snd_soc_codec *codec = socdev->codec; ++ struct i2c_client *i2c; ++ int ret; ++ ++ if (addr != setup->i2c_address) ++ return -ENODEV; ++ ++ client_template.adapter = adap; ++ client_template.addr = addr; ++ ++ i2c = kzalloc(sizeof(struct i2c_client), GFP_KERNEL); ++ if (i2c == NULL){ ++ kfree(codec); ++ return -ENOMEM; ++ } ++ memcpy(i2c, &client_template, sizeof(struct i2c_client)); ++ ++ i2c_set_clientdata(i2c, codec); ++ ++ codec->control_data = i2c; ++ ++ ret = i2c_attach_client(i2c); ++ if(ret < 0) { ++ err("failed to attach codec at addr %x\n", addr); ++ goto err; ++ } ++ ++ ret = wm8976_init(socdev); ++ if(ret < 0) { ++ err("failed to initialise WM8976\n"); ++ goto err; ++ } ++ return ret; ++ ++err: ++ kfree(codec); ++ kfree(i2c); ++ return ret; ++ ++} ++ ++static int wm8976_i2c_detach(struct i2c_client *client) ++{ ++ struct snd_soc_codec *codec = i2c_get_clientdata(client); ++ ++ i2c_detach_client(client); ++ ++ kfree(codec->reg_cache); ++ kfree(client); ++ ++ return 0; ++} ++ ++static int wm8976_i2c_attach(struct i2c_adapter *adap) ++{ ++ return i2c_probe(adap, &addr_data, wm8976_codec_probe); ++} ++ ++/* corgi i2c codec control layer */ ++static struct i2c_driver wm8976_i2c_driver = { ++ .driver = { ++ .name = "WM8976 I2C Codec", ++ .owner = THIS_MODULE, ++ }, ++ .id = I2C_DRIVERID_WM8976, ++ .attach_adapter = wm8976_i2c_attach, ++ .detach_client = wm8976_i2c_detach, ++ .command = NULL, ++}; ++ ++static struct i2c_client client_template = { ++ .name = "WM8976", ++ .driver = &wm8976_i2c_driver, ++}; ++#endif ++ ++static int wm8976_probe(struct platform_device *pdev) ++{ ++ struct snd_soc_device *socdev = platform_get_drvdata(pdev); ++ struct wm8976_setup_data *setup; ++ struct snd_soc_codec *codec; ++ int ret = 0; ++ ++ info("WM8976 Audio Codec %s", WM8976_VERSION); ++ ++ setup = socdev->codec_data; ++ codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); ++ if (codec == NULL) ++ return -ENOMEM; ++ ++ socdev->codec = codec; ++ mutex_init(&codec->mutex); ++ INIT_LIST_HEAD(&codec->dapm_widgets); ++ INIT_LIST_HEAD(&codec->dapm_paths); ++ ++ wm8976_socdev = socdev; ++#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) ++ if (setup->i2c_address) { ++ normal_i2c[0] = setup->i2c_address; ++ codec->hw_write = (hw_write_t)i2c_master_send; ++ ret = i2c_add_driver(&wm8976_i2c_driver); ++ if (ret != 0) ++ printk(KERN_ERR "can't add i2c driver"); ++ } ++#else ++ /* Add other interfaces here */ ++#endif ++ return ret; ++} ++ ++/* power down chip */ ++static int wm8976_remove(struct platform_device *pdev) ++{ ++ struct snd_soc_device *socdev = platform_get_drvdata(pdev); ++ struct snd_soc_codec *codec = socdev->codec; ++ ++ if (codec->control_data) ++ wm8976_dapm_event(codec, SNDRV_CTL_POWER_D3cold); ++ ++ snd_soc_free_pcms(socdev); ++ snd_soc_dapm_free(socdev); ++#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) ++ i2c_del_driver(&wm8976_i2c_driver); ++#endif ++ kfree(codec); ++ ++ return 0; ++} ++ ++struct snd_soc_codec_device soc_codec_dev_wm8976 = { ++ .probe = wm8976_probe, ++ .remove = wm8976_remove, ++ .suspend = wm8976_suspend, ++ .resume = wm8976_resume, ++}; ++ ++EXPORT_SYMBOL_GPL(soc_codec_dev_wm8976); ++ ++MODULE_DESCRIPTION("ASoC WM8976 driver"); ++MODULE_AUTHOR("Graeme Gregory"); ++MODULE_LICENSE("GPL"); +Index: linux-2.6-pxa-new/sound/soc/codecs/wm8976.h +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/sound/soc/codecs/wm8976.h +@@ -0,0 +1,73 @@ ++/* ++ * wm8976.h -- WM8976 Soc Audio driver ++ * ++ * This program is free software; you can redistribute it and/or modify ++ * it under the terms of the GNU General Public License version 2 as ++ * published by the Free Software Foundation. ++ */ ++ ++#ifndef _WM8976_H ++#define _WM8976_H ++ ++/* WM8976 register space */ ++ ++#define WM8976_RESET 0x0 ++#define WM8976_POWER1 0x1 ++#define WM8976_POWER2 0x2 ++#define WM8976_POWER3 0x3 ++#define WM8976_IFACE 0x4 ++#define WM8976_COMP 0x5 ++#define WM8976_CLOCK 0x6 ++#define WM8976_ADD 0x7 ++#define WM8976_GPIO 0x8 ++#define WM8976_JACK1 0x9 ++#define WM8976_DAC 0xa ++#define WM8976_DACVOLL 0xb ++#define WM8976_DACVOLR 0xc ++#define WM8976_JACK2 0xd ++#define WM8976_ADC 0xe ++#define WM8976_ADCVOL 0xf ++#define WM8976_EQ1 0x12 ++#define WM8976_EQ2 0x13 ++#define WM8976_EQ3 0x14 ++#define WM8976_EQ4 0x15 ++#define WM8976_EQ5 0x16 ++#define WM8976_DACLIM1 0x18 ++#define WM8976_DACLIM2 0x19 ++#define WM8976_NOTCH1 0x1b ++#define WM8976_NOTCH2 0x1c ++#define WM8976_NOTCH3 0x1d ++#define WM8976_NOTCH4 0x1e ++#define WM8976_ALC1 0x20 ++#define WM8976_ALC2 0x21 ++#define WM8976_ALC3 0x22 ++#define WM8976_NGATE 0x23 ++#define WM8976_PLLN 0x24 ++#define WM8976_PLLK1 0x25 ++#define WM8976_PLLK2 0x26 ++#define WM8976_PLLK3 0x27 ++#define WM8976_3D 0x29 ++#define WM8976_BEEP 0x2b ++#define WM8976_INPUT 0x2c ++#define WM8976_INPPGA 0x2d ++#define WM8976_ADCBOOST 0x2f ++#define WM8976_OUTPUT 0x31 ++#define WM8976_MIXL 0x32 ++#define WM8976_MIXR 0x33 ++#define WM8976_HPVOLL 0x34 ++#define WM8976_HPVOLR 0x35 ++#define WM8976_SPKVOLL 0x36 ++#define WM8976_SPKVOLR 0x37 ++#define WM8976_OUT3MIX 0x38 ++#define WM8976_MONOMIX 0x39 ++ ++#define WM8976_CACHEREGNUM 58 ++ ++struct wm8976_setup_data { ++ unsigned short i2c_address; ++}; ++ ++extern struct snd_soc_codec_dai wm8976_dai; ++extern struct snd_soc_codec_device soc_codec_dev_wm8976; ++ ++#endif +Index: linux-2.6-pxa-new/sound/soc/imx/imx21-pcm.c +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/sound/soc/imx/imx21-pcm.c +@@ -0,0 +1,454 @@ ++/* ++ * linux/sound/arm/mxc-pcm.c -- ALSA SoC interface for the Freescale i.MX CPU's ++ * ++ * Copyright 2006 Wolfson Microelectronics PLC. ++ * Author: Liam Girdwood ++ * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com ++ * ++ * Based on pxa2xx-pcm.c by Nicolas Pitre, (C) 2004 MontaVista Software, Inc. ++ * and on mxc-alsa-mc13783 (C) 2006 Freescale. ++ * ++ * This program is free software; you can redistribute it and/or modify ++ * it under the terms of the GNU General Public License version 2 as ++ * published by the Free Software Foundation. ++ * ++ * Revision history ++ * 29th Aug 2006 Initial version. ++ * ++ */ ++ ++#include <linux/module.h> ++#include <linux/init.h> ++#include <linux/platform_device.h> ++#include <linux/slab.h> ++#include <linux/dma-mapping.h> ++#include <sound/driver.h> ++#include <sound/core.h> ++#include <sound/pcm.h> ++#include <sound/pcm_params.h> ++#include <sound/soc.h> ++#include <asm/dma.h> ++#include <asm/hardware.h> ++ ++#include "imx-pcm.h" ++ ++/* debug */ ++#define IMX_DEBUG 0 ++#if IMX_DEBUG ++#define dbg(format, arg...) printk(format, ## arg) ++#else ++#define dbg(format, arg...) ++#endif ++ ++static const struct snd_pcm_hardware mxc_pcm_hardware = { ++ .info = (SNDRV_PCM_INFO_INTERLEAVED | ++ SNDRV_PCM_INFO_BLOCK_TRANSFER | ++ SNDRV_PCM_INFO_MMAP | ++ SNDRV_PCM_INFO_MMAP_VALID | ++ SNDRV_PCM_INFO_PAUSE | ++ SNDRV_PCM_INFO_RESUME), ++ .formats = SNDRV_PCM_FMTBIT_S16_LE | ++ SNDRV_PCM_FMTBIT_S24_LE, ++ .buffer_bytes_max = 32 * 1024, ++ .period_bytes_min = 64, ++ .period_bytes_max = 8 * 1024, ++ .periods_min = 2, ++ .periods_max = 255, ++ .fifo_size = 0, ++}; ++ ++struct mxc_runtime_data { ++ int dma_ch; ++ struct mxc_pcm_dma_param *dma_params; ++}; ++ ++/*! ++ * This function stops the current dma transfert for playback ++ * and clears the dma pointers. ++ * ++ * @param substream pointer to the structure of the current stream. ++ * ++ */ ++static void audio_stop_dma(struct snd_pcm_substream *substream) ++{ ++ struct snd_pcm_runtime *runtime = substream->runtime; ++ struct mxc_runtime_data *prtd = runtime->private_data; ++ unsigned int dma_size = frames_to_bytes(runtime, runtime->period_size); ++ unsigned int offset dma_size * s->periods; ++ unsigned long flags; ++ ++ spin_lock_irqsave(&prtd->dma_lock, flags); ++ ++ dbg("MXC : audio_stop_dma active = 0\n"); ++ prtd->active = 0; ++ prtd->period = 0; ++ prtd->periods = 0; ++ ++ /* this stops the dma channel and clears the buffer ptrs */ ++ mxc_dma_stop(prtd->dma_wchannel); ++ if(substream == SNDRV_PCM_STREAM_PLAYBACK) ++ dma_unmap_single(NULL, runtime->dma_addr + offset, dma_size, ++ DMA_TO_DEVICE); ++ else ++ dma_unmap_single(NULL, runtime->dma_addr + offset, dma_size, ++ DMA_FROM_DEVICE); ++ ++ spin_unlock_irqrestore(&prtd->dma_lock, flags); ++} ++ ++/*! ++ * This function is called whenever a new audio block needs to be ++ * transferred to mc13783. The function receives the address and the size ++ * of the new block and start a new DMA transfer. ++ * ++ * @param substream pointer to the structure of the current stream. ++ * ++ */ ++static int dma_new_period(struct snd_pcm_substream *substream) ++{ ++ struct snd_pcm_runtime *runtime = substream->runtime; ++ struct mxc_runtime_data *prtd = runtime->private_data; ++ unsigned int dma_size; ++ unsigned int offset; ++ int ret=0; ++ dma_request_t sdma_request; ++ ++ if (prtd->active){ ++ memset(&sdma_request, 0, sizeof(dma_request_t)); ++ dma_size = frames_to_bytes(runtime, runtime->period_size); ++ dbg("s->period (%x) runtime->periods (%d)\n", ++ s->period,runtime->periods); ++ dbg("runtime->period_size (%d) dma_size (%d)\n", ++ (unsigned int)runtime->period_size, ++ runtime->dma_bytes); ++ ++ offset = dma_size * prtd->period; ++ snd_assert(dma_size <= DMA_BUF_SIZE, ); ++ if(substream == SNDRV_PCM_STREAM_PLAYBACK) ++ sdma_request.sourceAddr = (char*)(dma_map_single(NULL, ++ runtime->dma_area + offset, dma_size, DMA_TO_DEVICE)); ++ else ++ sdma_request.destAddr = (char*)(dma_map_single(NULL, ++ runtime->dma_area + offset, dma_size, DMA_FROM_DEVICE)); ++ sdma_request.count = dma_size; ++ ++ dbg("MXC: Start DMA offset (%d) size (%d)\n", offset, ++ runtime->dma_bytes); ++ ++ mxc_dma_set_config(prtd->dma_wchannel, &sdma_request, 0); ++ if((ret = mxc_dma_start(prtd->dma_wchannel)) < 0) { ++ dbg("audio_process_dma: cannot queue DMA buffer\ ++ (%i)\n", ret); ++ return err; ++ } ++ prtd->tx_spin = 1; /* FGA little trick to retrieve DMA pos */ ++ prtd->period++; ++ prtd->period %= runtime->periods; ++ } ++ return ret; ++} ++ ++ ++/*! ++ * This is a callback which will be called ++ * when a TX transfer finishes. The call occurs ++ * in interrupt context. ++ * ++ * @param dat pointer to the structure of the current stream. ++ * ++ */ ++static void audio_dma_irq(void *data) ++{ ++ struct snd_pcm_substream *substream; ++ struct snd_pcm_runtime *runtime; ++ struct mxc_runtime_data *prtd; ++ unsigned int dma_size; ++ unsigned int previous_period; ++ unsigned int offset; ++ ++ substream = data; ++ runtime = substream->runtime; ++ prtd = runtime->private_data; ++ previous_period = prtd->periods; ++ dma_size = frames_to_bytes(runtime, runtime->period_size); ++ offset = dma_size * previous_period; ++ ++ prtd->tx_spin = 0; ++ prtd->periods++; ++ prtd->periods %= runtime->periods; ++ ++ /* ++ * Give back to the CPU the access to the non cached memory ++ */ ++ if(substream == SNDRV_PCM_STREAM_PLAYBACK) ++ dma_unmap_single(NULL, runtime->dma_addr + offset, dma_size, ++ DMA_TO_DEVICE); ++ else ++ dma_unmap_single(NULL, runtime->dma_addr + offset, dma_size, ++ DMA_FROM_DEVICE); ++ /* ++ * If we are getting a callback for an active stream then we inform ++ * the PCM middle layer we've finished a period ++ */ ++ if (prtd->active) ++ snd_pcm_period_elapsed(substream); ++ ++ /* ++ * Trig next DMA transfer ++ */ ++ dma_new_period(substream); ++} ++ ++/*! ++ * This function configures the hardware to allow audio ++ * playback operations. It is called by ALSA framework. ++ * ++ * @param substream pointer to the structure of the current stream. ++ * ++ * @return 0 on success, -1 otherwise. ++ */ ++static int ++snd_mxc_prepare(struct snd_pcm_substream *substream) ++{ ++ struct mxc_runtime_data *prtd = runtime->private_data; ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ int ret = 0; ++ prtd->period = 0; ++ prtd->periods = 0; ++ ++ dma_channel_params params; ++ int channel = 0; // passed in ? ++ ++ if ((ret = mxc_request_dma(&channel, "ALSA TX SDMA") < 0)){ ++ dbg("error requesting a write dma channel\n"); ++ return ret; ++ } ++ ++ /* configure DMA params */ ++ memset(¶ms, 0, sizeof(dma_channel_params)); ++ params.bd_number = 1; ++ params.arg = s; ++ params.callback = callback; ++ params.transfer_type = emi_2_per; ++ params.watermark_level = SDMA_TXFIFO_WATERMARK; ++ params.word_size = TRANSFER_16BIT; ++ //dbg(KERN_ERR "activating connection SSI1 - SDMA\n"); ++ params.per_address = SSI1_BASE_ADDR; ++ params.event_id = DMA_REQ_SSI1_TX1; ++ params.peripheral_type = SSI; ++ ++ /* set up chn with params */ ++ mxc_dma_setup_channel(channel, ¶ms); ++ s->dma_wchannel = channel; ++ ++ return ret; ++} ++ ++static int mxc_pcm_hw_params(struct snd_pcm_substream *substream, ++ struct snd_pcm_hw_params *params) ++{ ++ struct snd_pcm_runtime *runtime = substream->runtime; ++ int ret; ++ ++ if((ret=snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params))) < 0) ++ return ret; ++ runtime->dma_addr = virt_to_phys(runtime->dma_area); ++ ++ return ret; ++} ++ ++static int mxc_pcm_hw_free(struct snd_pcm_substream *substream) ++{ ++ return snd_pcm_lib_free_pages(substream); ++} ++ ++static int mxc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) ++{ ++ struct mxc_runtime_data *prtd = substream->runtime->private_data; ++ int ret = 0; ++ ++ switch (cmd) { ++ case SNDRV_PCM_TRIGGER_START: ++ prtd->tx_spin = 0; ++ /* requested stream startup */ ++ prtd->active = 1; ++ ret = dma_new_period(substream); ++ break; ++ case SNDRV_PCM_TRIGGER_STOP: ++ /* requested stream shutdown */ ++ ret = audio_stop_dma(substream); ++ break; ++ case SNDRV_PCM_TRIGGER_SUSPEND: ++ prtd->active = 0; ++ prtd->periods = 0; ++ break; ++ case SNDRV_PCM_TRIGGER_RESUME: ++ prtd->active = 1; ++ prtd->tx_spin = 0; ++ ret = dma_new_period(substream); ++ break; ++ case SNDRV_PCM_TRIGGER_PAUSE_PUSH: ++ prtd->active = 0; ++ break; ++ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: ++ prtd->active = 1; ++ if (prtd->old_offset) { ++ prtd->tx_spin = 0; ++ ret = dma_new_period(substream); ++ } ++ break; ++ default: ++ ret = -EINVAL; ++ break; ++ } ++ ++ return ret; ++} ++ ++static snd_pcm_uframes_t mxc_pcm_pointer(struct snd_pcm_substream *substream) ++{ ++ struct snd_pcm_runtime *runtime = substream->runtime; ++ struct mxc_runtime_data *prtd = runtime->private_data; ++ unsigned int offset = 0; ++ ++ /* tx_spin value is used here to check if a transfert is active */ ++ if (prtd->tx_spin){ ++ offset = (runtime->period_size * (prtd->periods)) + ++ (runtime->period_size >> 1); ++ if (offset >= runtime->buffer_size) ++ offset = runtime->period_size >> 1; ++ } else { ++ offset = (runtime->period_size * (s->periods)); ++ if (offset >= runtime->buffer_size) ++ offset = 0; ++ } ++ ++ return offset; ++} ++ ++ ++static int mxc_pcm_open(struct snd_pcm_substream *substream) ++{ ++ struct snd_pcm_runtime *runtime = substream->runtime; ++ struct mxc_runtime_data *prtd; ++ int ret; ++ ++ snd_soc_set_runtime_hwparams(substream, &mxc_pcm_hardware); ++ ++ if ((err = snd_pcm_hw_constraint_integer(runtime, ++ SNDRV_PCM_HW_PARAM_PERIODS)) < 0) ++ return err; ++ if ((err = snd_pcm_hw_constraint_list(runtime, 0, ++ SNDRV_PCM_HW_PARAM_RATE, &hw_playback_rates)) < 0) ++ return err; ++ msleep(10); // liam - why ++ ++ /* setup DMA controller for playback */ ++ if((err = configure_write_channel(&mxc_mc13783->s[SNDRV_PCM_STREAM_PLAYBACK], ++ audio_dma_irq)) < 0 ) ++ return err; ++ ++ if((prtd = kzalloc(sizeof(struct mxc_runtime_data), GFP_KERNEL)) == NULL) { ++ ret = -ENOMEM; ++ goto out; ++ } ++ ++ runtime->private_data = prtd; ++ return 0; ++ ++ err1: ++ kfree(prtd); ++ out: ++ return ret; ++} ++ ++static int mxc_pcm_close(struct snd_pcm_substream *substream) ++{ ++ struct snd_pcm_runtime *runtime = substream->runtime; ++ struct mxc_runtime_data *prtd = runtime->private_data; ++ ++// mxc_mc13783_t *chip; ++ audio_stream_t *s; ++ device_data_t* device; ++ int ssi; ++ ++ //chip = snd_pcm_substream_chip(substream); ++ s = &chip->s[substream->pstr->stream]; ++ device = &s->stream_device; ++ ssi = device->ssi; ++ ++ //disable_stereodac(); ++ ++ ssi_transmit_enable(ssi, false); ++ ssi_interrupt_disable(ssi, ssi_tx_dma_interrupt_enable); ++ ssi_tx_fifo_enable(ssi, ssi_fifo_0, false); ++ ssi_enable(ssi, false); ++ ++ chip->s[substream->pstr->stream].stream = NULL; ++ ++ return 0; ++} ++ ++static int ++mxc_pcm_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *vma) ++{ ++ struct snd_pcm_runtime *runtime = substream->runtime; ++ return dma_mmap_writecombine(substream->pcm->card->dev, vma, ++ runtime->dma_area, ++ runtime->dma_addr, ++ runtime->dma_bytes); ++} ++ ++struct snd_pcm_ops mxc_pcm_ops = { ++ .open = mxc_pcm_open, ++ .close = mxc_pcm_close, ++ .ioctl = snd_pcm_lib_ioctl, ++ .hw_params = mxc_pcm_hw_params, ++ .hw_free = mxc_pcm_hw_free, ++ .prepare = mxc_pcm_prepare, ++ .trigger = mxc_pcm_trigger, ++ .pointer = mxc_pcm_pointer, ++ .mmap = mxc_pcm_mmap, ++}; ++ ++static u64 mxc_pcm_dmamask = 0xffffffff; ++ ++int mxc_pcm_new(struct snd_card *card, struct snd_soc_codec_dai *dai, ++ struct snd_pcm *pcm) ++{ ++ int ret = 0; ++ ++ if (!card->dev->dma_mask) ++ card->dev->dma_mask = &mxc_pcm_dmamask; ++ if (!card->dev->coherent_dma_mask) ++ card->dev->coherent_dma_mask = 0xffffffff; ++ ++ if (dai->playback.channels_min) { ++ ret = mxc_pcm_preallocate_dma_buffer(pcm, ++ SNDRV_PCM_STREAM_PLAYBACK); ++ if (ret) ++ goto out; ++ } ++ ++ if (dai->capture.channels_min) { ++ ret = mxc_pcm_preallocate_dma_buffer(pcm, ++ SNDRV_PCM_STREAM_CAPTURE); ++ if (ret) ++ goto out; ++ } ++ out: ++ return ret; ++} ++ ++struct snd_soc_platform mxc_soc_platform = { ++ .name = "mxc-audio", ++ .pcm_ops = &mxc_pcm_ops, ++ .pcm_new = mxc_pcm_new, ++ .pcm_free = mxc_pcm_free_dma_buffers, ++}; ++ ++EXPORT_SYMBOL_GPL(mxc_soc_platform); ++ ++MODULE_AUTHOR("Liam Girdwood"); ++MODULE_DESCRIPTION("Freescale i.MX PCM DMA module"); ++MODULE_LICENSE("GPL"); +Index: linux-2.6-pxa-new/sound/soc/imx/imx21-pcm.h +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/sound/soc/imx/imx21-pcm.h +@@ -0,0 +1,237 @@ ++/* ++ * mxc-pcm.h :- ASoC platform header for Freescale i.MX ++ * ++ * This program is free software; you can redistribute it and/or modify ++ * it under the terms of the GNU General Public License version 2 as ++ * published by the Free Software Foundation. ++ */ ++ ++#ifndef _MXC_PCM_H ++#define _MXC_PCM_H ++ ++struct { ++ char *name; /* stream identifier */ ++ dma_channel_params dma_params; ++} mxc_pcm_dma_param; ++ ++extern struct snd_soc_cpu_dai mxc_ssi_dai[3]; ++ ++/* platform data */ ++extern struct snd_soc_platform mxc_soc_platform; ++extern struct snd_ac97_bus_ops mxc_ac97_ops; ++ ++/* temp until imx-regs.h is up2date */ ++#define SSI1_STX0 __REG(IMX_SSI1_BASE + 0x00) ++#define SSI1_STX0_PHYS __PHYS_REG(IMX_SSI1_BASE + 0x00) ++#define SSI1_STX1 __REG(IMX_SSI1_BASE + 0x04) ++#define SSI1_STX1_PHYS __PHYS_REG(IMX_SSI1_BASE + 0x04) ++#define SSI1_SRX0 __REG(IMX_SSI1_BASE + 0x08) ++#define SSI1_SRX0_PHYS __PHYS_REG(IMX_SSI1_BASE + 0x08) ++#define SSI1_SRX1 __REG(IMX_SSI1_BASE + 0x0c) ++#define SSI1_SRX1_PHYS __PHYS_REG(IMX_SSI1_BASE + 0x0c) ++#define SSI1_SCR __REG(IMX_SSI1_BASE + 0x10) ++#define SSI1_SISR __REG(IMX_SSI1_BASE + 0x14) ++#define SSI1_SIER __REG(IMX_SSI1_BASE + 0x18) ++#define SSI1_STCR __REG(IMX_SSI1_BASE + 0x1c) ++#define SSI1_SRCR __REG(IMX_SSI1_BASE + 0x20) ++#define SSI1_STCCR __REG(IMX_SSI1_BASE + 0x24) ++#define SSI1_SRCCR __REG(IMX_SSI1_BASE + 0x28) ++#define SSI1_SFCSR __REG(IMX_SSI1_BASE + 0x2c) ++#define SSI1_STR __REG(IMX_SSI1_BASE + 0x30) ++#define SSI1_SOR __REG(IMX_SSI1_BASE + 0x34) ++#define SSI1_SACNT __REG(IMX_SSI1_BASE + 0x38) ++#define SSI1_SACADD __REG(IMX_SSI1_BASE + 0x3c) ++#define SSI1_SACDAT __REG(IMX_SSI1_BASE + 0x40) ++#define SSI1_SATAG __REG(IMX_SSI1_BASE + 0x44) ++#define SSI1_STMSK __REG(IMX_SSI1_BASE + 0x48) ++#define SSI1_SRMSK __REG(IMX_SSI1_BASE + 0x4c) ++ ++#define SSI2_STX0 __REG(IMX_SSI2_BASE + 0x00) ++#define SSI2_STX0_PHYS __PHYS_REG(IMX_SSI2_BASE + 0x00) ++#define SSI2_STX1 __REG(IMX_SSI2_BASE + 0x04) ++#define SSI2_STX1_PHYS __PHYS_REG(IMX_SSI2_BASE + 0x04) ++#define SSI2_SRX0 __REG(IMX_SSI2_BASE + 0x08) ++#define SSI2_SRX0_PHYS __PHYS_REG(IMX_SSI2_BASE + 0x08) ++#define SSI2_SRX1 __REG(IMX_SSI2_BASE + 0x0c) ++#define SSI2_SRX1_PHYS __PHYS_REG(IMX_SSI2_BASE + 0x0c) ++#define SSI2_SCR __REG(IMX_SSI2_BASE + 0x10) ++#define SSI2_SISR __REG(IMX_SSI2_BASE + 0x14) ++#define SSI2_SIER __REG(IMX_SSI2_BASE + 0x18) ++#define SSI2_STCR __REG(IMX_SSI2_BASE + 0x1c) ++#define SSI2_SRCR __REG(IMX_SSI2_BASE + 0x20) ++#define SSI2_STCCR __REG(IMX_SSI2_BASE + 0x24) ++#define SSI2_SRCCR __REG(IMX_SSI2_BASE + 0x28) ++#define SSI2_SFCSR __REG(IMX_SSI2_BASE + 0x2c) ++#define SSI2_STR __REG(IMX_SSI2_BASE + 0x30) ++#define SSI2_SOR __REG(IMX_SSI2_BASE + 0x34) ++#define SSI2_SACNT __REG(IMX_SSI2_BASE + 0x38) ++#define SSI2_SACADD __REG(IMX_SSI2_BASE + 0x3c) ++#define SSI2_SACDAT __REG(IMX_SSI2_BASE + 0x40) ++#define SSI2_SATAG __REG(IMX_SSI2_BASE + 0x44) ++#define SSI2_STMSK __REG(IMX_SSI2_BASE + 0x48) ++#define SSI2_SRMSK __REG(IMX_SSI2_BASE + 0x4c) ++ ++#define SSI_SCR_CLK_IST (1 << 9) ++#define SSI_SCR_TCH_EN (1 << 8) ++#define SSI_SCR_SYS_CLK_EN (1 << 7) ++#define SSI_SCR_I2S_MODE_NORM (0 << 5) ++#define SSI_SCR_I2S_MODE_MSTR (1 << 5) ++#define SSI_SCR_I2S_MODE_SLAVE (2 << 5) ++#define SSI_SCR_SYN (1 << 4) ++#define SSI_SCR_NET (1 << 3) ++#define SSI_SCR_RE (1 << 2) ++#define SSI_SCR_TE (1 << 1) ++#define SSI_SCR_SSIEN (1 << 0) ++ ++#define SSI_SISR_CMDAU (1 << 18) ++#define SSI_SISR_CMDDU (1 << 17) ++#define SSI_SISR_RXT (1 << 16) ++#define SSI_SISR_RDR1 (1 << 15) ++#define SSI_SISR_RDR0 (1 << 14) ++#define SSI_SISR_TDE1 (1 << 13) ++#define SSI_SISR_TDE0 (1 << 12) ++#define SSI_SISR_ROE1 (1 << 11) ++#define SSI_SISR_ROE0 (1 << 10) ++#define SSI_SISR_TUE1 (1 << 9) ++#define SSI_SISR_TUE0 (1 << 8) ++#define SSI_SISR_TFS (1 << 7) ++#define SSI_SISR_RFS (1 << 6) ++#define SSI_SISR_TLS (1 << 5) ++#define SSI_SISR_RLS (1 << 4) ++#define SSI_SISR_RFF1 (1 << 3) ++#define SSI_SISR_RFF0 (1 << 2) ++#define SSI_SISR_TFE1 (1 << 1) ++#define SSI_SISR_TFE0 (1 << 0) ++ ++#define SSI_SIER_RDMAE (1 << 22) ++#define SSI_SIER_RIE (1 << 21) ++#define SSI_SIER_TDMAE (1 << 20) ++#define SSI_SIER_TIE (1 << 19) ++#define SSI_SIER_CMDAU_EN (1 << 18) ++#define SSI_SIER_CMDDU_EN (1 << 17) ++#define SSI_SIER_RXT_EN (1 << 16) ++#define SSI_SIER_RDR1_EN (1 << 15) ++#define SSI_SIER_RDR0_EN (1 << 14) ++#define SSI_SIER_TDE1_EN (1 << 13) ++#define SSI_SIER_TDE0_EN (1 << 12) ++#define SSI_SIER_ROE1_EN (1 << 11) ++#define SSI_SIER_ROE0_EN (1 << 10) ++#define SSI_SIER_TUE1_EN (1 << 9) ++#define SSI_SIER_TUE0_EN (1 << 8) ++#define SSI_SIER_TFS_EN (1 << 7) ++#define SSI_SIER_RFS_EN (1 << 6) ++#define SSI_SIER_TLS_EN (1 << 5) ++#define SSI_SIER_RLS_EN (1 << 4) ++#define SSI_SIER_RFF1_EN (1 << 3) ++#define SSI_SIER_RFF0_EN (1 << 2) ++#define SSI_SIER_TFE1_EN (1 << 1) ++#define SSI_SIER_TFE0_EN (1 << 0) ++ ++#define SSI_STCR_TXBIT0 (1 << 9) ++#define SSI_STCR_TFEN1 (1 << 8) ++#define SSI_STCR_TFEN0 (1 << 7) ++#define SSI_STCR_TFDIR (1 << 6) ++#define SSI_STCR_TXDIR (1 << 5) ++#define SSI_STCR_TSHFD (1 << 4) ++#define SSI_STCR_TSCKP (1 << 3) ++#define SSI_STCR_TFSI (1 << 2) ++#define SSI_STCR_TFSL (1 << 1) ++#define SSI_STCR_TEFS (1 << 0) ++ ++#define SSI_SRCR_RXBIT0 (1 << 9) ++#define SSI_SRCR_RFEN1 (1 << 8) ++#define SSI_SRCR_RFEN0 (1 << 7) ++#define SSI_SRCR_RFDIR (1 << 6) ++#define SSI_SRCR_RXDIR (1 << 5) ++#define SSI_SRCR_RSHFD (1 << 4) ++#define SSI_SRCR_RSCKP (1 << 3) ++#define SSI_SRCR_RFSI (1 << 2) ++#define SSI_SRCR_RFSL (1 << 1) ++#define SSI_SRCR_REFS (1 << 0) ++ ++#define SSI_STCCR_DIV2 (1 << 18) ++#define SSI_STCCR_PSR (1 << 15) ++#define SSI_STCCR_WL(x) ((((x) - 2) >> 1) << 13) ++#define SSI_STCCR_DC(x) (((x) & 0x1f) << 8) ++#define SSI_STCCR_PM(x) (((x) & 0xff) << 0) ++ ++#define SSI_SRCCR_DIV2 (1 << 18) ++#define SSI_SRCCR_PSR (1 << 15) ++#define SSI_SRCCR_WL(x) ((((x) - 2) >> 1) << 13) ++#define SSI_SRCCR_DC(x) (((x) & 0x1f) << 8) ++#define SSI_SRCCR_PM(x) (((x) & 0xff) << 0) ++ ++ ++#define SSI_SFCSR_RFCNT1(x) (((x) & 0xf) << 28) ++#define SSI_SFCSR_TFCNT1(x) (((x) & 0xf) << 24) ++#define SSI_SFCSR_RFWM1(x) (((x) & 0xf) << 20) ++#define SSI_SFCSR_TFWM1(x) (((x) & 0xf) << 16) ++#define SSI_SFCSR_RFCNT0(x) (((x) & 0xf) << 12) ++#define SSI_SFCSR_TFCNT0(x) (((x) & 0xf) << 8) ++#define SSI_SFCSR_RFWM0(x) (((x) & 0xf) << 4) ++#define SSI_SFCSR_TFWM0(x) (((x) & 0xf) << 0) ++ ++#define SSI_STR_TEST (1 << 15) ++#define SSI_STR_RCK2TCK (1 << 14) ++#define SSI_STR_RFS2TFS (1 << 13) ++#define SSI_STR_RXSTATE(x) (((x) & 0xf) << 8) ++#define SSI_STR_TXD2RXD (1 << 7) ++#define SSI_STR_TCK2RCK (1 << 6) ++#define SSI_STR_TFS2RFS (1 << 5) ++#define SSI_STR_TXSTATE(x) (((x) & 0xf) << 0) ++ ++#define SSI_SOR_CLKOFF (1 << 6) ++#define SSI_SOR_RX_CLR (1 << 5) ++#define SSI_SOR_TX_CLR (1 << 4) ++#define SSI_SOR_INIT (1 << 3) ++#define SSI_SOR_WAIT(x) (((x) & 0x3) << 1) ++#define SSI_SOR_SYNRST (1 << 0) ++ ++#define SSI_SACNT_FRDIV(x) (((x) & 0x3f) << 5) ++#define SSI_SACNT_WR (x << 4) ++#define SSI_SACNT_RD (x << 3) ++#define SSI_SACNT_TIF (x << 2) ++#define SSI_SACNT_FV (x << 1) ++#define SSI_SACNT_A97EN (x << 0) ++ ++ ++/* AUDMUX registers */ ++#define AUDMUX_HPCR1 __REG(IMX_AUDMUX_BASE + 0x00) ++#define AUDMUX_HPCR2 __REG(IMX_AUDMUX_BASE + 0x04) ++#define AUDMUX_HPCR3 __REG(IMX_AUDMUX_BASE + 0x08) ++#define AUDMUX_PPCR1 __REG(IMX_AUDMUX_BASE + 0x10) ++#define AUDMUX_PPCR2 __REG(IMX_AUDMUX_BASE + 0x14) ++#define AUDMUX_PPCR3 __REG(IMX_AUDMUX_BASE + 0x18) ++ ++#define AUDMUX_HPCR_TFSDIR (1 << 31) ++#define AUDMUX_HPCR_TCLKDIR (1 << 30) ++#define AUDMUX_HPCR_TFCSEL_TX (0 << 26) ++#define AUDMUX_HPCR_TFCSEL_RX (8 << 26) ++#define AUDMUX_HPCR_TFCSEL(x) (((x) & 0x7) << 26) ++#define AUDMUX_HPCR_RFSDIR (1 << 25) ++#define AUDMUX_HPCR_RCLKDIR (1 << 24) ++#define AUDMUX_HPCR_RFCSEL_TX (0 << 20) ++#define AUDMUX_HPCR_RFCSEL_RX (8 << 20) ++#define AUDMUX_HPCR_RFCSEL(x) (((x) & 0x7) << 20) ++#define AUDMUX_HPCR_RXDSEL(x) (((x) & 0x7) << 13) ++#define AUDMUX_HPCR_SYN (1 << 12) ++#define AUDMUX_HPCR_TXRXEN (1 << 10) ++#define AUDMUX_HPCR_INMEN (1 << 8) ++#define AUDMUX_HPCR_INMMASK(x) (((x) & 0xff) << 0) ++ ++#define AUDMUX_PPCR_TFSDIR (1 << 31) ++#define AUDMUX_PPCR_TCLKDIR (1 << 30) ++#define AUDMUX_PPCR_TFCSEL_TX (0 << 26) ++#define AUDMUX_PPCR_TFCSEL_RX (8 << 26) ++#define AUDMUX_PPCR_TFCSEL(x) (((x) & 0x7) << 26) ++#define AUDMUX_PPCR_RFSDIR (1 << 25) ++#define AUDMUX_PPCR_RCLKDIR (1 << 24) ++#define AUDMUX_PPCR_RFCSEL_TX (0 << 20) ++#define AUDMUX_PPCR_RFCSEL_RX (8 << 20) ++#define AUDMUX_PPCR_RFCSEL(x) (((x) & 0x7) << 20) ++#define AUDMUX_PPCR_RXDSEL(x) (((x) & 0x7) << 13) ++#define AUDMUX_PPCR_SYN (1 << 12) ++#define AUDMUX_PPCR_TXRXEN (1 << 10) ++ ++ ++#endif +Index: linux-2.6-pxa-new/sound/soc/imx/imx31-pcm.c +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/sound/soc/imx/imx31-pcm.c +@@ -0,0 +1,454 @@ ++/* ++ * linux/sound/arm/mxc-pcm.c -- ALSA SoC interface for the Freescale i.MX CPU's ++ * ++ * Copyright 2006 Wolfson Microelectronics PLC. ++ * Author: Liam Girdwood ++ * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com ++ * ++ * Based on pxa2xx-pcm.c by Nicolas Pitre, (C) 2004 MontaVista Software, Inc. ++ * and on mxc-alsa-mc13783 (C) 2006 Freescale. ++ * ++ * This program is free software; you can redistribute it and/or modify ++ * it under the terms of the GNU General Public License version 2 as ++ * published by the Free Software Foundation. ++ * ++ * Revision history ++ * 29th Aug 2006 Initial version. ++ * ++ */ ++ ++#include <linux/module.h> ++#include <linux/init.h> ++#include <linux/platform_device.h> ++#include <linux/slab.h> ++#include <linux/dma-mapping.h> ++#include <sound/driver.h> ++#include <sound/core.h> ++#include <sound/pcm.h> ++#include <sound/pcm_params.h> ++#include <sound/soc.h> ++#include <asm/dma.h> ++#include <asm/hardware.h> ++ ++#include "imx-pcm.h" ++ ++/* debug */ ++#define IMX_DEBUG 0 ++#if IMX_DEBUG ++#define dbg(format, arg...) printk(format, ## arg) ++#else ++#define dbg(format, arg...) ++#endif ++ ++static const struct snd_pcm_hardware mxc_pcm_hardware = { ++ .info = (SNDRV_PCM_INFO_INTERLEAVED | ++ SNDRV_PCM_INFO_BLOCK_TRANSFER | ++ SNDRV_PCM_INFO_MMAP | ++ SNDRV_PCM_INFO_MMAP_VALID | ++ SNDRV_PCM_INFO_PAUSE | ++ SNDRV_PCM_INFO_RESUME), ++ .formats = SNDRV_PCM_FMTBIT_S16_LE | ++ SNDRV_PCM_FMTBIT_S24_LE, ++ .buffer_bytes_max = 32 * 1024, ++ .period_bytes_min = 64, ++ .period_bytes_max = 8 * 1024, ++ .periods_min = 2, ++ .periods_max = 255, ++ .fifo_size = 0, ++}; ++ ++struct mxc_runtime_data { ++ int dma_ch; ++ struct mxc_pcm_dma_param *dma_params; ++}; ++ ++/*! ++ * This function stops the current dma transfert for playback ++ * and clears the dma pointers. ++ * ++ * @param substream pointer to the structure of the current stream. ++ * ++ */ ++static void audio_stop_dma(struct snd_pcm_substream *substream) ++{ ++ struct snd_pcm_runtime *runtime = substream->runtime; ++ struct mxc_runtime_data *prtd = runtime->private_data; ++ unsigned int dma_size = frames_to_bytes(runtime, runtime->period_size); ++ unsigned int offset dma_size * s->periods; ++ unsigned long flags; ++ ++ spin_lock_irqsave(&prtd->dma_lock, flags); ++ ++ dbg("MXC : audio_stop_dma active = 0\n"); ++ prtd->active = 0; ++ prtd->period = 0; ++ prtd->periods = 0; ++ ++ /* this stops the dma channel and clears the buffer ptrs */ ++ mxc_dma_stop(prtd->dma_wchannel); ++ if(substream == SNDRV_PCM_STREAM_PLAYBACK) ++ dma_unmap_single(NULL, runtime->dma_addr + offset, dma_size, ++ DMA_TO_DEVICE); ++ else ++ dma_unmap_single(NULL, runtime->dma_addr + offset, dma_size, ++ DMA_FROM_DEVICE); ++ ++ spin_unlock_irqrestore(&prtd->dma_lock, flags); ++} ++ ++/*! ++ * This function is called whenever a new audio block needs to be ++ * transferred to mc13783. The function receives the address and the size ++ * of the new block and start a new DMA transfer. ++ * ++ * @param substream pointer to the structure of the current stream. ++ * ++ */ ++static int dma_new_period(struct snd_pcm_substream *substream) ++{ ++ struct snd_pcm_runtime *runtime = substream->runtime; ++ struct mxc_runtime_data *prtd = runtime->private_data; ++ unsigned int dma_size; ++ unsigned int offset; ++ int ret=0; ++ dma_request_t sdma_request; ++ ++ if (prtd->active){ ++ memset(&sdma_request, 0, sizeof(dma_request_t)); ++ dma_size = frames_to_bytes(runtime, runtime->period_size); ++ dbg("s->period (%x) runtime->periods (%d)\n", ++ s->period,runtime->periods); ++ dbg("runtime->period_size (%d) dma_size (%d)\n", ++ (unsigned int)runtime->period_size, ++ runtime->dma_bytes); ++ ++ offset = dma_size * prtd->period; ++ snd_assert(dma_size <= DMA_BUF_SIZE, ); ++ if(substream == SNDRV_PCM_STREAM_PLAYBACK) ++ sdma_request.sourceAddr = (char*)(dma_map_single(NULL, ++ runtime->dma_area + offset, dma_size, DMA_TO_DEVICE)); ++ else ++ sdma_request.destAddr = (char*)(dma_map_single(NULL, ++ runtime->dma_area + offset, dma_size, DMA_FROM_DEVICE)); ++ sdma_request.count = dma_size; ++ ++ dbg("MXC: Start DMA offset (%d) size (%d)\n", offset, ++ runtime->dma_bytes); ++ ++ mxc_dma_set_config(prtd->dma_wchannel, &sdma_request, 0); ++ if((ret = mxc_dma_start(prtd->dma_wchannel)) < 0) { ++ dbg("audio_process_dma: cannot queue DMA buffer\ ++ (%i)\n", ret); ++ return err; ++ } ++ prtd->tx_spin = 1; /* FGA little trick to retrieve DMA pos */ ++ prtd->period++; ++ prtd->period %= runtime->periods; ++ } ++ return ret; ++} ++ ++ ++/*! ++ * This is a callback which will be called ++ * when a TX transfer finishes. The call occurs ++ * in interrupt context. ++ * ++ * @param dat pointer to the structure of the current stream. ++ * ++ */ ++static void audio_dma_irq(void *data) ++{ ++ struct snd_pcm_substream *substream; ++ struct snd_pcm_runtime *runtime; ++ struct mxc_runtime_data *prtd; ++ unsigned int dma_size; ++ unsigned int previous_period; ++ unsigned int offset; ++ ++ substream = data; ++ runtime = substream->runtime; ++ prtd = runtime->private_data; ++ previous_period = prtd->periods; ++ dma_size = frames_to_bytes(runtime, runtime->period_size); ++ offset = dma_size * previous_period; ++ ++ prtd->tx_spin = 0; ++ prtd->periods++; ++ prtd->periods %= runtime->periods; ++ ++ /* ++ * Give back to the CPU the access to the non cached memory ++ */ ++ if(substream == SNDRV_PCM_STREAM_PLAYBACK) ++ dma_unmap_single(NULL, runtime->dma_addr + offset, dma_size, ++ DMA_TO_DEVICE); ++ else ++ dma_unmap_single(NULL, runtime->dma_addr + offset, dma_size, ++ DMA_FROM_DEVICE); ++ /* ++ * If we are getting a callback for an active stream then we inform ++ * the PCM middle layer we've finished a period ++ */ ++ if (prtd->active) ++ snd_pcm_period_elapsed(substream); ++ ++ /* ++ * Trig next DMA transfer ++ */ ++ dma_new_period(substream); ++} ++ ++/*! ++ * This function configures the hardware to allow audio ++ * playback operations. It is called by ALSA framework. ++ * ++ * @param substream pointer to the structure of the current stream. ++ * ++ * @return 0 on success, -1 otherwise. ++ */ ++static int ++snd_mxc_prepare(struct snd_pcm_substream *substream) ++{ ++ struct mxc_runtime_data *prtd = runtime->private_data; ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ int ret = 0; ++ prtd->period = 0; ++ prtd->periods = 0; ++ ++ dma_channel_params params; ++ int channel = 0; // passed in ? ++ ++ if ((ret = mxc_request_dma(&channel, "ALSA TX SDMA") < 0)){ ++ dbg("error requesting a write dma channel\n"); ++ return ret; ++ } ++ ++ /* configure DMA params */ ++ memset(¶ms, 0, sizeof(dma_channel_params)); ++ params.bd_number = 1; ++ params.arg = s; ++ params.callback = callback; ++ params.transfer_type = emi_2_per; ++ params.watermark_level = SDMA_TXFIFO_WATERMARK; ++ params.word_size = TRANSFER_16BIT; ++ //dbg(KERN_ERR "activating connection SSI1 - SDMA\n"); ++ params.per_address = SSI1_BASE_ADDR; ++ params.event_id = DMA_REQ_SSI1_TX1; ++ params.peripheral_type = SSI; ++ ++ /* set up chn with params */ ++ mxc_dma_setup_channel(channel, ¶ms); ++ s->dma_wchannel = channel; ++ ++ return ret; ++} ++ ++static int mxc_pcm_hw_params(struct snd_pcm_substream *substream, ++ struct snd_pcm_hw_params *params) ++{ ++ struct snd_pcm_runtime *runtime = substream->runtime; ++ int ret; ++ ++ if((ret=snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params))) < 0) ++ return ret; ++ runtime->dma_addr = virt_to_phys(runtime->dma_area); ++ ++ return ret; ++} ++ ++static int mxc_pcm_hw_free(struct snd_pcm_substream *substream) ++{ ++ return snd_pcm_lib_free_pages(substream); ++} ++ ++static int mxc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) ++{ ++ struct mxc_runtime_data *prtd = substream->runtime->private_data; ++ int ret = 0; ++ ++ switch (cmd) { ++ case SNDRV_PCM_TRIGGER_START: ++ prtd->tx_spin = 0; ++ /* requested stream startup */ ++ prtd->active = 1; ++ ret = dma_new_period(substream); ++ break; ++ case SNDRV_PCM_TRIGGER_STOP: ++ /* requested stream shutdown */ ++ ret = audio_stop_dma(substream); ++ break; ++ case SNDRV_PCM_TRIGGER_SUSPEND: ++ prtd->active = 0; ++ prtd->periods = 0; ++ break; ++ case SNDRV_PCM_TRIGGER_RESUME: ++ prtd->active = 1; ++ prtd->tx_spin = 0; ++ ret = dma_new_period(substream); ++ break; ++ case SNDRV_PCM_TRIGGER_PAUSE_PUSH: ++ prtd->active = 0; ++ break; ++ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: ++ prtd->active = 1; ++ if (prtd->old_offset) { ++ prtd->tx_spin = 0; ++ ret = dma_new_period(substream); ++ } ++ break; ++ default: ++ ret = -EINVAL; ++ break; ++ } ++ ++ return ret; ++} ++ ++static snd_pcm_uframes_t mxc_pcm_pointer(struct snd_pcm_substream *substream) ++{ ++ struct snd_pcm_runtime *runtime = substream->runtime; ++ struct mxc_runtime_data *prtd = runtime->private_data; ++ unsigned int offset = 0; ++ ++ /* tx_spin value is used here to check if a transfert is active */ ++ if (prtd->tx_spin){ ++ offset = (runtime->period_size * (prtd->periods)) + ++ (runtime->period_size >> 1); ++ if (offset >= runtime->buffer_size) ++ offset = runtime->period_size >> 1; ++ } else { ++ offset = (runtime->period_size * (s->periods)); ++ if (offset >= runtime->buffer_size) ++ offset = 0; ++ } ++ ++ return offset; ++} ++ ++ ++static int mxc_pcm_open(struct snd_pcm_substream *substream) ++{ ++ struct snd_pcm_runtime *runtime = substream->runtime; ++ struct mxc_runtime_data *prtd; ++ int ret; ++ ++ snd_soc_set_runtime_hwparams(substream, &mxc_pcm_hardware); ++ ++ if ((err = snd_pcm_hw_constraint_integer(runtime, ++ SNDRV_PCM_HW_PARAM_PERIODS)) < 0) ++ return err; ++ if ((err = snd_pcm_hw_constraint_list(runtime, 0, ++ SNDRV_PCM_HW_PARAM_RATE, &hw_playback_rates)) < 0) ++ return err; ++ msleep(10); // liam - why ++ ++ /* setup DMA controller for playback */ ++ if((err = configure_write_channel(&mxc_mc13783->s[SNDRV_PCM_STREAM_PLAYBACK], ++ audio_dma_irq)) < 0 ) ++ return err; ++ ++ if((prtd = kzalloc(sizeof(struct mxc_runtime_data), GFP_KERNEL)) == NULL) { ++ ret = -ENOMEM; ++ goto out; ++ } ++ ++ runtime->private_data = prtd; ++ return 0; ++ ++ err1: ++ kfree(prtd); ++ out: ++ return ret; ++} ++ ++static int mxc_pcm_close(struct snd_pcm_substream *substream) ++{ ++ struct snd_pcm_runtime *runtime = substream->runtime; ++ struct mxc_runtime_data *prtd = runtime->private_data; ++ ++// mxc_mc13783_t *chip; ++ audio_stream_t *s; ++ device_data_t* device; ++ int ssi; ++ ++ //chip = snd_pcm_substream_chip(substream); ++ s = &chip->s[substream->pstr->stream]; ++ device = &s->stream_device; ++ ssi = device->ssi; ++ ++ //disable_stereodac(); ++ ++ ssi_transmit_enable(ssi, false); ++ ssi_interrupt_disable(ssi, ssi_tx_dma_interrupt_enable); ++ ssi_tx_fifo_enable(ssi, ssi_fifo_0, false); ++ ssi_enable(ssi, false); ++ ++ chip->s[substream->pstr->stream].stream = NULL; ++ ++ return 0; ++} ++ ++static int ++mxc_pcm_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *vma) ++{ ++ struct snd_pcm_runtime *runtime = substream->runtime; ++ return dma_mmap_writecombine(substream->pcm->card->dev, vma, ++ runtime->dma_area, ++ runtime->dma_addr, ++ runtime->dma_bytes); ++} ++ ++struct snd_pcm_ops mxc_pcm_ops = { ++ .open = mxc_pcm_open, ++ .close = mxc_pcm_close, ++ .ioctl = snd_pcm_lib_ioctl, ++ .hw_params = mxc_pcm_hw_params, ++ .hw_free = mxc_pcm_hw_free, ++ .prepare = mxc_pcm_prepare, ++ .trigger = mxc_pcm_trigger, ++ .pointer = mxc_pcm_pointer, ++ .mmap = mxc_pcm_mmap, ++}; ++ ++static u64 mxc_pcm_dmamask = 0xffffffff; ++ ++int mxc_pcm_new(struct snd_card *card, struct snd_soc_codec_dai *dai, ++ struct snd_pcm *pcm) ++{ ++ int ret = 0; ++ ++ if (!card->dev->dma_mask) ++ card->dev->dma_mask = &mxc_pcm_dmamask; ++ if (!card->dev->coherent_dma_mask) ++ card->dev->coherent_dma_mask = 0xffffffff; ++ ++ if (dai->playback.channels_min) { ++ ret = mxc_pcm_preallocate_dma_buffer(pcm, ++ SNDRV_PCM_STREAM_PLAYBACK); ++ if (ret) ++ goto out; ++ } ++ ++ if (dai->capture.channels_min) { ++ ret = mxc_pcm_preallocate_dma_buffer(pcm, ++ SNDRV_PCM_STREAM_CAPTURE); ++ if (ret) ++ goto out; ++ } ++ out: ++ return ret; ++} ++ ++struct snd_soc_platform mxc_soc_platform = { ++ .name = "mxc-audio", ++ .pcm_ops = &mxc_pcm_ops, ++ .pcm_new = mxc_pcm_new, ++ .pcm_free = mxc_pcm_free_dma_buffers, ++}; ++ ++EXPORT_SYMBOL_GPL(mxc_soc_platform); ++ ++MODULE_AUTHOR("Liam Girdwood"); ++MODULE_DESCRIPTION("Freescale i.MX PCM DMA module"); ++MODULE_LICENSE("GPL"); +Index: linux-2.6-pxa-new/sound/soc/imx/imx31-pcm.h +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/sound/soc/imx/imx31-pcm.h +@@ -0,0 +1,237 @@ ++/* ++ * mxc-pcm.h :- ASoC platform header for Freescale i.MX ++ * ++ * This program is free software; you can redistribute it and/or modify ++ * it under the terms of the GNU General Public License version 2 as ++ * published by the Free Software Foundation. ++ */ ++ ++#ifndef _MXC_PCM_H ++#define _MXC_PCM_H ++ ++struct { ++ char *name; /* stream identifier */ ++ dma_channel_params dma_params; ++} mxc_pcm_dma_param; ++ ++extern struct snd_soc_cpu_dai mxc_ssi_dai[3]; ++ ++/* platform data */ ++extern struct snd_soc_platform mxc_soc_platform; ++extern struct snd_ac97_bus_ops mxc_ac97_ops; ++ ++/* temp until imx-regs.h is up2date */ ++#define SSI1_STX0 __REG(IMX_SSI1_BASE + 0x00) ++#define SSI1_STX0_PHYS __PHYS_REG(IMX_SSI1_BASE + 0x00) ++#define SSI1_STX1 __REG(IMX_SSI1_BASE + 0x04) ++#define SSI1_STX1_PHYS __PHYS_REG(IMX_SSI1_BASE + 0x04) ++#define SSI1_SRX0 __REG(IMX_SSI1_BASE + 0x08) ++#define SSI1_SRX0_PHYS __PHYS_REG(IMX_SSI1_BASE + 0x08) ++#define SSI1_SRX1 __REG(IMX_SSI1_BASE + 0x0c) ++#define SSI1_SRX1_PHYS __PHYS_REG(IMX_SSI1_BASE + 0x0c) ++#define SSI1_SCR __REG(IMX_SSI1_BASE + 0x10) ++#define SSI1_SISR __REG(IMX_SSI1_BASE + 0x14) ++#define SSI1_SIER __REG(IMX_SSI1_BASE + 0x18) ++#define SSI1_STCR __REG(IMX_SSI1_BASE + 0x1c) ++#define SSI1_SRCR __REG(IMX_SSI1_BASE + 0x20) ++#define SSI1_STCCR __REG(IMX_SSI1_BASE + 0x24) ++#define SSI1_SRCCR __REG(IMX_SSI1_BASE + 0x28) ++#define SSI1_SFCSR __REG(IMX_SSI1_BASE + 0x2c) ++#define SSI1_STR __REG(IMX_SSI1_BASE + 0x30) ++#define SSI1_SOR __REG(IMX_SSI1_BASE + 0x34) ++#define SSI1_SACNT __REG(IMX_SSI1_BASE + 0x38) ++#define SSI1_SACADD __REG(IMX_SSI1_BASE + 0x3c) ++#define SSI1_SACDAT __REG(IMX_SSI1_BASE + 0x40) ++#define SSI1_SATAG __REG(IMX_SSI1_BASE + 0x44) ++#define SSI1_STMSK __REG(IMX_SSI1_BASE + 0x48) ++#define SSI1_SRMSK __REG(IMX_SSI1_BASE + 0x4c) ++ ++#define SSI2_STX0 __REG(IMX_SSI2_BASE + 0x00) ++#define SSI2_STX0_PHYS __PHYS_REG(IMX_SSI2_BASE + 0x00) ++#define SSI2_STX1 __REG(IMX_SSI2_BASE + 0x04) ++#define SSI2_STX1_PHYS __PHYS_REG(IMX_SSI2_BASE + 0x04) ++#define SSI2_SRX0 __REG(IMX_SSI2_BASE + 0x08) ++#define SSI2_SRX0_PHYS __PHYS_REG(IMX_SSI2_BASE + 0x08) ++#define SSI2_SRX1 __REG(IMX_SSI2_BASE + 0x0c) ++#define SSI2_SRX1_PHYS __PHYS_REG(IMX_SSI2_BASE + 0x0c) ++#define SSI2_SCR __REG(IMX_SSI2_BASE + 0x10) ++#define SSI2_SISR __REG(IMX_SSI2_BASE + 0x14) ++#define SSI2_SIER __REG(IMX_SSI2_BASE + 0x18) ++#define SSI2_STCR __REG(IMX_SSI2_BASE + 0x1c) ++#define SSI2_SRCR __REG(IMX_SSI2_BASE + 0x20) ++#define SSI2_STCCR __REG(IMX_SSI2_BASE + 0x24) ++#define SSI2_SRCCR __REG(IMX_SSI2_BASE + 0x28) ++#define SSI2_SFCSR __REG(IMX_SSI2_BASE + 0x2c) ++#define SSI2_STR __REG(IMX_SSI2_BASE + 0x30) ++#define SSI2_SOR __REG(IMX_SSI2_BASE + 0x34) ++#define SSI2_SACNT __REG(IMX_SSI2_BASE + 0x38) ++#define SSI2_SACADD __REG(IMX_SSI2_BASE + 0x3c) ++#define SSI2_SACDAT __REG(IMX_SSI2_BASE + 0x40) ++#define SSI2_SATAG __REG(IMX_SSI2_BASE + 0x44) ++#define SSI2_STMSK __REG(IMX_SSI2_BASE + 0x48) ++#define SSI2_SRMSK __REG(IMX_SSI2_BASE + 0x4c) ++ ++#define SSI_SCR_CLK_IST (1 << 9) ++#define SSI_SCR_TCH_EN (1 << 8) ++#define SSI_SCR_SYS_CLK_EN (1 << 7) ++#define SSI_SCR_I2S_MODE_NORM (0 << 5) ++#define SSI_SCR_I2S_MODE_MSTR (1 << 5) ++#define SSI_SCR_I2S_MODE_SLAVE (2 << 5) ++#define SSI_SCR_SYN (1 << 4) ++#define SSI_SCR_NET (1 << 3) ++#define SSI_SCR_RE (1 << 2) ++#define SSI_SCR_TE (1 << 1) ++#define SSI_SCR_SSIEN (1 << 0) ++ ++#define SSI_SISR_CMDAU (1 << 18) ++#define SSI_SISR_CMDDU (1 << 17) ++#define SSI_SISR_RXT (1 << 16) ++#define SSI_SISR_RDR1 (1 << 15) ++#define SSI_SISR_RDR0 (1 << 14) ++#define SSI_SISR_TDE1 (1 << 13) ++#define SSI_SISR_TDE0 (1 << 12) ++#define SSI_SISR_ROE1 (1 << 11) ++#define SSI_SISR_ROE0 (1 << 10) ++#define SSI_SISR_TUE1 (1 << 9) ++#define SSI_SISR_TUE0 (1 << 8) ++#define SSI_SISR_TFS (1 << 7) ++#define SSI_SISR_RFS (1 << 6) ++#define SSI_SISR_TLS (1 << 5) ++#define SSI_SISR_RLS (1 << 4) ++#define SSI_SISR_RFF1 (1 << 3) ++#define SSI_SISR_RFF0 (1 << 2) ++#define SSI_SISR_TFE1 (1 << 1) ++#define SSI_SISR_TFE0 (1 << 0) ++ ++#define SSI_SIER_RDMAE (1 << 22) ++#define SSI_SIER_RIE (1 << 21) ++#define SSI_SIER_TDMAE (1 << 20) ++#define SSI_SIER_TIE (1 << 19) ++#define SSI_SIER_CMDAU_EN (1 << 18) ++#define SSI_SIER_CMDDU_EN (1 << 17) ++#define SSI_SIER_RXT_EN (1 << 16) ++#define SSI_SIER_RDR1_EN (1 << 15) ++#define SSI_SIER_RDR0_EN (1 << 14) ++#define SSI_SIER_TDE1_EN (1 << 13) ++#define SSI_SIER_TDE0_EN (1 << 12) ++#define SSI_SIER_ROE1_EN (1 << 11) ++#define SSI_SIER_ROE0_EN (1 << 10) ++#define SSI_SIER_TUE1_EN (1 << 9) ++#define SSI_SIER_TUE0_EN (1 << 8) ++#define SSI_SIER_TFS_EN (1 << 7) ++#define SSI_SIER_RFS_EN (1 << 6) ++#define SSI_SIER_TLS_EN (1 << 5) ++#define SSI_SIER_RLS_EN (1 << 4) ++#define SSI_SIER_RFF1_EN (1 << 3) ++#define SSI_SIER_RFF0_EN (1 << 2) ++#define SSI_SIER_TFE1_EN (1 << 1) ++#define SSI_SIER_TFE0_EN (1 << 0) ++ ++#define SSI_STCR_TXBIT0 (1 << 9) ++#define SSI_STCR_TFEN1 (1 << 8) ++#define SSI_STCR_TFEN0 (1 << 7) ++#define SSI_STCR_TFDIR (1 << 6) ++#define SSI_STCR_TXDIR (1 << 5) ++#define SSI_STCR_TSHFD (1 << 4) ++#define SSI_STCR_TSCKP (1 << 3) ++#define SSI_STCR_TFSI (1 << 2) ++#define SSI_STCR_TFSL (1 << 1) ++#define SSI_STCR_TEFS (1 << 0) ++ ++#define SSI_SRCR_RXBIT0 (1 << 9) ++#define SSI_SRCR_RFEN1 (1 << 8) ++#define SSI_SRCR_RFEN0 (1 << 7) ++#define SSI_SRCR_RFDIR (1 << 6) ++#define SSI_SRCR_RXDIR (1 << 5) ++#define SSI_SRCR_RSHFD (1 << 4) ++#define SSI_SRCR_RSCKP (1 << 3) ++#define SSI_SRCR_RFSI (1 << 2) ++#define SSI_SRCR_RFSL (1 << 1) ++#define SSI_SRCR_REFS (1 << 0) ++ ++#define SSI_STCCR_DIV2 (1 << 18) ++#define SSI_STCCR_PSR (1 << 15) ++#define SSI_STCCR_WL(x) ((((x) - 2) >> 1) << 13) ++#define SSI_STCCR_DC(x) (((x) & 0x1f) << 8) ++#define SSI_STCCR_PM(x) (((x) & 0xff) << 0) ++ ++#define SSI_SRCCR_DIV2 (1 << 18) ++#define SSI_SRCCR_PSR (1 << 15) ++#define SSI_SRCCR_WL(x) ((((x) - 2) >> 1) << 13) ++#define SSI_SRCCR_DC(x) (((x) & 0x1f) << 8) ++#define SSI_SRCCR_PM(x) (((x) & 0xff) << 0) ++ ++ ++#define SSI_SFCSR_RFCNT1(x) (((x) & 0xf) << 28) ++#define SSI_SFCSR_TFCNT1(x) (((x) & 0xf) << 24) ++#define SSI_SFCSR_RFWM1(x) (((x) & 0xf) << 20) ++#define SSI_SFCSR_TFWM1(x) (((x) & 0xf) << 16) ++#define SSI_SFCSR_RFCNT0(x) (((x) & 0xf) << 12) ++#define SSI_SFCSR_TFCNT0(x) (((x) & 0xf) << 8) ++#define SSI_SFCSR_RFWM0(x) (((x) & 0xf) << 4) ++#define SSI_SFCSR_TFWM0(x) (((x) & 0xf) << 0) ++ ++#define SSI_STR_TEST (1 << 15) ++#define SSI_STR_RCK2TCK (1 << 14) ++#define SSI_STR_RFS2TFS (1 << 13) ++#define SSI_STR_RXSTATE(x) (((x) & 0xf) << 8) ++#define SSI_STR_TXD2RXD (1 << 7) ++#define SSI_STR_TCK2RCK (1 << 6) ++#define SSI_STR_TFS2RFS (1 << 5) ++#define SSI_STR_TXSTATE(x) (((x) & 0xf) << 0) ++ ++#define SSI_SOR_CLKOFF (1 << 6) ++#define SSI_SOR_RX_CLR (1 << 5) ++#define SSI_SOR_TX_CLR (1 << 4) ++#define SSI_SOR_INIT (1 << 3) ++#define SSI_SOR_WAIT(x) (((x) & 0x3) << 1) ++#define SSI_SOR_SYNRST (1 << 0) ++ ++#define SSI_SACNT_FRDIV(x) (((x) & 0x3f) << 5) ++#define SSI_SACNT_WR (x << 4) ++#define SSI_SACNT_RD (x << 3) ++#define SSI_SACNT_TIF (x << 2) ++#define SSI_SACNT_FV (x << 1) ++#define SSI_SACNT_A97EN (x << 0) ++ ++ ++/* AUDMUX registers */ ++#define AUDMUX_HPCR1 __REG(IMX_AUDMUX_BASE + 0x00) ++#define AUDMUX_HPCR2 __REG(IMX_AUDMUX_BASE + 0x04) ++#define AUDMUX_HPCR3 __REG(IMX_AUDMUX_BASE + 0x08) ++#define AUDMUX_PPCR1 __REG(IMX_AUDMUX_BASE + 0x10) ++#define AUDMUX_PPCR2 __REG(IMX_AUDMUX_BASE + 0x14) ++#define AUDMUX_PPCR3 __REG(IMX_AUDMUX_BASE + 0x18) ++ ++#define AUDMUX_HPCR_TFSDIR (1 << 31) ++#define AUDMUX_HPCR_TCLKDIR (1 << 30) ++#define AUDMUX_HPCR_TFCSEL_TX (0 << 26) ++#define AUDMUX_HPCR_TFCSEL_RX (8 << 26) ++#define AUDMUX_HPCR_TFCSEL(x) (((x) & 0x7) << 26) ++#define AUDMUX_HPCR_RFSDIR (1 << 25) ++#define AUDMUX_HPCR_RCLKDIR (1 << 24) ++#define AUDMUX_HPCR_RFCSEL_TX (0 << 20) ++#define AUDMUX_HPCR_RFCSEL_RX (8 << 20) ++#define AUDMUX_HPCR_RFCSEL(x) (((x) & 0x7) << 20) ++#define AUDMUX_HPCR_RXDSEL(x) (((x) & 0x7) << 13) ++#define AUDMUX_HPCR_SYN (1 << 12) ++#define AUDMUX_HPCR_TXRXEN (1 << 10) ++#define AUDMUX_HPCR_INMEN (1 << 8) ++#define AUDMUX_HPCR_INMMASK(x) (((x) & 0xff) << 0) ++ ++#define AUDMUX_PPCR_TFSDIR (1 << 31) ++#define AUDMUX_PPCR_TCLKDIR (1 << 30) ++#define AUDMUX_PPCR_TFCSEL_TX (0 << 26) ++#define AUDMUX_PPCR_TFCSEL_RX (8 << 26) ++#define AUDMUX_PPCR_TFCSEL(x) (((x) & 0x7) << 26) ++#define AUDMUX_PPCR_RFSDIR (1 << 25) ++#define AUDMUX_PPCR_RCLKDIR (1 << 24) ++#define AUDMUX_PPCR_RFCSEL_TX (0 << 20) ++#define AUDMUX_PPCR_RFCSEL_RX (8 << 20) ++#define AUDMUX_PPCR_RFCSEL(x) (((x) & 0x7) << 20) ++#define AUDMUX_PPCR_RXDSEL(x) (((x) & 0x7) << 13) ++#define AUDMUX_PPCR_SYN (1 << 12) ++#define AUDMUX_PPCR_TXRXEN (1 << 10) ++ ++ ++#endif +Index: linux-2.6-pxa-new/sound/soc/s3c24xx/s3c24xx-i2s.c +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/sound/soc/s3c24xx/s3c24xx-i2s.c +@@ -0,0 +1,271 @@ ++/* ++ * s3c24xx-i2s.c -- ALSA Soc Audio Layer ++ * ++ * Copyright 2005 Wolfson Microelectronics PLC. ++ * Author: Graeme Gregory ++ * graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.com ++ * ++ * This program is free software; you can redistribute it and/or modify it ++ * under the terms of the GNU General Public License as published by the ++ * Free Software Foundation; either version 2 of the License, or (at your ++ * option) any later version. ++ * ++ * Revision history ++ * 10th Nov 2006 Initial version. ++ */ ++ ++#include <linux/init.h> ++#include <linux/module.h> ++#include <linux/device.h> ++#include <linux/delay.h> ++#include <linux/clk.h> ++#include <sound/driver.h> ++#include <sound/core.h> ++#include <sound/pcm.h> ++#include <sound/initval.h> ++#include <sound/soc.h> ++ ++#include <asm/hardware.h> ++#include <asm/io.h> ++#include <asm/arch/regs-iis.h> ++#include <asm/arch/regs-gpio.h> ++#include <asm/arch/regs-clock.h> ++#include <asm/arch/audio.h> ++#include <asm/dma.h> ++#include <asm/arch/dma.h> ++ ++#include "s3c24xx-pcm.h" ++ ++/* used to disable sysclk if external crystal is used */ ++static int extclk = 0; ++module_param(extclk, int, 0); ++MODULE_PARM_DESC(extclk, "set to 1 to disable s3c24XX i2s sysclk"); ++ ++#define S3C24XX_I2S_DAIFMT \ ++ (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF) ++ ++#define S3C24XX_I2S_DIR \ ++ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE) ++ ++#define S3C24XX_I2S_RATES \ ++ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ ++ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ ++ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) ++ ++/* priv is divider */ ++static struct snd_soc_dai_mode s3c24xx_i2s_modes[] = ++{ ++ /* s3c24xx I2S frame and clock master modes */ ++ { ++ .fmt = S3C24XX_I2S_DAIFMT | SND_SOC_DAIFMT_CBS_CFS, ++ .pcmfmt = SNDRV_PCM_FMTBIT_S16_LE, ++ .pcmrate = SNDRV_PCM_RATE_44100, ++ .pcmdir = S3C24XX_I2S_DIR, ++ .flags = SND_SOC_DAI_BFS_RATE, ++ .fs = 384, ++ .bfs = 32, ++ .priv = 0x00 ++ }, ++}; ++ ++static struct s3c2410_dma_client s3c24xx_dma_client_out = { ++ .name = "I2S PCM Stereo out" ++}; ++ ++static struct s3c2410_dma_client s3c24xx_dma_client_in = { ++ .name = "I2S PCM Stereo in" ++}; ++ ++static s3c24xx_pcm_dma_params_t s3c24xx_i2s_pcm_stereo_out = { ++ .client = &s3c24xx_dma_client_out, ++ .channel = DMACH_I2S_OUT, ++ .dma_addr = S3C2410_PA_IIS+S3C2410_IISFIFO ++}; ++ ++static s3c24xx_pcm_dma_params_t s3c24xx_i2s_pcm_stereo_in = { ++ .client = &s3c24xx_dma_client_in, ++ .channel = DMACH_I2S_IN, ++ .dma_addr = S3C2410_PA_IIS+S3C2410_IISFIFO ++}; ++ ++ ++struct s3c24xx_i2s_port { ++ int master; ++}; ++static struct s3c24xx_i2s_port s3c24xx_i2s; ++ ++/* Empty for the s3c24xx platforms */ ++static int s3c24xx_i2s_startup(struct snd_pcm_substream *substream) ++{ ++ return 0; ++} ++ ++static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream, ++ struct snd_pcm_hw_params *params) ++{ ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ unsigned long iiscon; ++ unsigned long iismod; ++ unsigned long iisfcon; ++ ++ s3c24xx_i2s.master = 0; ++ if(rtd->cpu_dai->dai_runtime.fmt & SND_SOC_DAIFMT_CBS_CFS) ++ s3c24xx_i2s.master = 1; ++ ++ /* Configure the I2S pins in correct mode */ ++ s3c2410_gpio_cfgpin(S3C2410_GPE0,S3C2410_GPE0_I2SLRCK); ++ if (s3c24xx_i2s.master && !extclk){ ++ printk("Setting Clock Output as we are Master\n"); ++ s3c2410_gpio_cfgpin(S3C2410_GPE1,S3C2410_GPE1_I2SSCLK); ++ } ++ s3c2410_gpio_cfgpin(S3C2410_GPE2,S3C2410_GPE2_CDCLK); ++ s3c2410_gpio_cfgpin(S3C2410_GPE3,S3C2410_GPE3_I2SSDI); ++ s3c2410_gpio_cfgpin(S3C2410_GPE4,S3C2410_GPE4_I2SSDO); ++ ++ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ++ { ++ rtd->cpu_dai->dma_data = &s3c24xx_i2s_pcm_stereo_out; ++ } ++ else ++ { ++ rtd->cpu_dai->dma_data = &s3c24xx_i2s_pcm_stereo_in; ++ } ++ ++ /* Working copies of registers */ ++ iiscon=readl(S3C24XX_VA_IIS+S3C2410_IISCON); ++ iismod=readl(S3C24XX_VA_IIS+S3C2410_IISMOD); ++ iisfcon=readl(S3C24XX_VA_IIS+S3C2410_IISFCON); ++ /* is port used by another stream */ ++ if (!(iiscon & S3C2410_IISCON_IISEN)) { ++ ++ /* Clear the registers */ ++ ++ iismod |= S3C2410_IISMOD_32FS | S3C2410_IISMOD_384FS; ++ ++ if (!s3c24xx_i2s.master) ++ iismod |= S3C2410_IISMOD_SLAVE; ++ ++ if (rtd->cpu_dai->dai_runtime.fmt & SND_SOC_DAIFMT_LEFT_J) ++ iismod |= S3C2410_IISMOD_MSB; ++ } ++ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ++ { ++ iismod |= S3C2410_IISMOD_TXMODE; ++ iiscon |= S3C2410_IISCON_TXDMAEN; ++ iisfcon |= S3C2410_IISFCON_TXDMA | S3C2410_IISFCON_TXENABLE; ++ } ++ else ++ { ++ iismod |= S3C2410_IISMOD_RXMODE; ++ iiscon |= S3C2410_IISCON_RXDMAEN; ++ iisfcon |= S3C2410_IISFCON_RXDMA | S3C2410_IISFCON_RXENABLE; ++ } ++ ++ writel(iiscon, S3C24XX_VA_IIS+S3C2410_IISCON); ++ writel(iismod, S3C24XX_VA_IIS+S3C2410_IISMOD); ++ writel(iisfcon, S3C24XX_VA_IIS+S3C2410_IISFCON); ++ ++ printk("IISCON: %lx IISMOD: %lx", iiscon, iismod); ++ ++ return 0; ++} ++ ++static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd) ++{ ++ int ret = 0; ++ unsigned long iiscon; ++ ++ switch (cmd) { ++ case SNDRV_PCM_TRIGGER_START: ++ /* Enable the IIS unit */ ++ iiscon = readl(S3C24XX_VA_IIS+S3C2410_IISCON); ++ iiscon |= S3C2410_IISCON_IISEN; ++ writel(iiscon, S3C24XX_VA_IIS+S3C2410_IISCON); ++ break; ++ case SNDRV_PCM_TRIGGER_RESUME: ++ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: ++ case SNDRV_PCM_TRIGGER_STOP: ++ case SNDRV_PCM_TRIGGER_SUSPEND: ++ case SNDRV_PCM_TRIGGER_PAUSE_PUSH: ++ break; ++ default: ++ ret = -EINVAL; ++ } ++ ++ return ret; ++} ++ ++static void s3c24xx_i2s_shutdown(struct snd_pcm_substream *substream) ++{ ++ unsigned long iismod, iiscon; ++ ++ iismod=readl(S3C24XX_VA_IIS+S3C2410_IISMOD); ++ ++ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { ++ iismod &= ~S3C2410_IISMOD_TXMODE; ++ } else { ++ iismod &= ~S3C2410_IISMOD_RXMODE; ++ } ++ ++ writel(iismod,S3C24XX_VA_IIS+S3C2410_IISMOD); ++ ++ iiscon=readl(S3C24XX_VA_IIS+S3C2410_IISCON); ++ ++ if (iismod & ( S3C2410_IISMOD_TXMODE | S3C2410_IISMOD_RXMODE )) { ++ iiscon &= ! S3C2410_IISCON_IISEN; ++ writel(iiscon,S3C24XX_VA_IIS+S3C2410_IISCON); ++ } ++} ++ ++#ifdef CONFIG_PM ++static int s3c24xx_i2s_suspend(struct platform_device *dev, ++ struct snd_soc_cpu_dai *dai) ++{ ++} ++ ++static int s3c24xx_i2s_resume(struct platform_device *pdev, ++ struct snd_soc_cpu_dai *dai) ++{ ++} ++ ++#else ++#define s3c24xx_i2s_suspend NULL ++#define s3c24xx_i2s_resume NULL ++#endif ++ ++/* s3c24xx I2S sysclock is always 384 FS */ ++static unsigned int s3c24xx_i2s_config_sysclk(struct snd_soc_cpu_dai *iface, ++ struct snd_soc_clock_info *info, unsigned int clk) ++{ ++ return info->rate * 384; ++} ++ ++struct snd_soc_cpu_dai s3c24xx_i2s_dai = { ++ .name = "s3c24xx-i2s", ++ .id = 0, ++ .type = SND_SOC_DAI_I2S, ++ .suspend = s3c24xx_i2s_suspend, ++ .resume = s3c24xx_i2s_resume, ++ .config_sysclk = s3c24xx_i2s_config_sysclk, ++ .playback = { ++ .channels_min = 2, ++ .channels_max = 2,}, ++ .capture = { ++ .channels_min = 2, ++ .channels_max = 2,}, ++ .ops = { ++ .startup = s3c24xx_i2s_startup, ++ .shutdown = s3c24xx_i2s_shutdown, ++ .trigger = s3c24xx_i2s_trigger, ++ .hw_params = s3c24xx_i2s_hw_params,}, ++ .caps = { ++ .num_modes = ARRAY_SIZE(s3c24xx_i2s_modes), ++ .mode = s3c24xx_i2s_modes,}, ++}; ++ ++EXPORT_SYMBOL_GPL(s3c24xx_i2s_dai); ++ ++/* Module information */ ++MODULE_AUTHOR("Graeme Gregory, graeme.gregory@wolfsonmicro.com, www.wolfsonmicro.com"); ++MODULE_DESCRIPTION("s3c24xx I2S SoC Interface"); ++MODULE_LICENSE("GPL"); +Index: linux-2.6-pxa-new/sound/soc/s3c24xx/s3c24xx-pcm.c +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/sound/soc/s3c24xx/s3c24xx-pcm.c +@@ -0,0 +1,362 @@ ++/* ++ * s3c24xx-pcm.c -- ALSA Soc Audio Layer ++ * ++ * Copyright 2005 Wolfson Microelectronics PLC. ++ * Author: Graeme Gregory ++ * graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.com ++ * ++ * This program is free software; you can redistribute it and/or modify it ++ * under the terms of the GNU General Public License as published by the ++ * Free Software Foundation; either version 2 of the License, or (at your ++ * option) any later version. ++ * ++ * Revision history ++ * 10th Nov 2006 Initial version. ++ */ ++ ++#include <linux/module.h> ++#include <linux/init.h> ++#include <linux/platform_device.h> ++#include <linux/slab.h> ++#include <linux/dma-mapping.h> ++ ++#include <sound/driver.h> ++#include <sound/core.h> ++#include <sound/pcm.h> ++#include <sound/pcm_params.h> ++#include <sound/soc.h> ++ ++#include <asm/dma.h> ++#include <asm/io.h> ++#include <asm/hardware.h> ++#include <asm/arch/dma.h> ++#include <asm/arch/audio.h> ++ ++#include "s3c24xx-pcm.h" ++ ++static const struct snd_pcm_hardware s3c24xx_pcm_hardware = { ++ .info = SNDRV_PCM_INFO_MMAP | ++ SNDRV_PCM_INFO_MMAP_VALID | ++ SNDRV_PCM_INFO_INTERLEAVED | ++ SNDRV_PCM_INFO_PAUSE | ++ SNDRV_PCM_INFO_RESUME, ++ .formats = SNDRV_PCM_FMTBIT_S16_LE, ++ .period_bytes_min = 32, ++ .period_bytes_max = 8192, ++ .periods_min = 1, ++ .periods_max = 8192, ++ .buffer_bytes_max = 256 * 1024, ++ .fifo_size = 32, ++}; ++ ++struct s3c24xx_runtime_data { ++ dma_addr_t dma_buffer; ++ dma_addr_t dma_buffer_end; ++ size_t period_size; ++ dma_addr_t period_ptr; ++ s3c24xx_pcm_dma_params_t *params; ++}; ++ ++/* Move the pointer onto the next period, dealing with wrap around. ++ */ ++void static next_period(struct s3c24xx_runtime_data *prtd) ++{ ++ prtd->period_ptr+=prtd->period_size; ++ if(prtd->period_ptr>=prtd->dma_buffer_end) ++ { ++ prtd->period_ptr=prtd->dma_buffer; ++ } ++} ++ ++void s3c24xx_audio_buffdone(struct s3c2410_dma_chan *channel, ++ void *dev_id, int size, ++ enum s3c2410_dma_buffresult result) ++{ ++ struct snd_pcm_substream *substream = dev_id; ++ struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; ++ ++ if(result==S3C2410_RES_OK) ++ { ++ next_period(prtd); ++ s3c2410_dma_enqueue(prtd->params->channel, substream, prtd->period_ptr, prtd->period_size); ++ } ++ snd_pcm_period_elapsed(substream); ++ ++} ++ ++static int s3c24xx_pcm_hw_params(struct snd_pcm_substream *substream, ++ struct snd_pcm_hw_params *params) ++{ ++ struct snd_pcm_runtime *runtime = substream->runtime; ++ struct s3c24xx_runtime_data *prtd = runtime->private_data; ++ struct snd_soc_pcm_runtime *rtd = substream->private_data; ++ s3c24xx_pcm_dma_params_t *dma = rtd->cpu_dai->dma_data; ++ int ret; ++ ++ printk("Entered s3c24xx hw_params\n"); ++ ++ snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); ++ runtime->dma_bytes = params_buffer_bytes(params); ++ ++ prtd->params=dma; ++ if(ret=s3c2410_dma_request(prtd->params->channel, ++ prtd->params->client,NULL)) ++ { ++ printk("Failed to get dma channel %d for %s\n",prtd->params->channel, ++ prtd->params->client->name); ++ return ret; ++ } ++ ++ //s3c2410_dma_setflags(prtd->params->channel,S3C2410_DMAF_AUTOSTART); ++ if(substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ++ { ++ s3c2410_dma_devconfig(prtd->params->channel, S3C2410_DMASRC_MEM, ++ S3C2410_DISRCC_INC | S3C2410_DISRCC_APB, ++ prtd->params->dma_addr); ++ } ++ else ++ { ++ s3c2410_dma_devconfig(prtd->params->channel, S3C2410_DMASRC_HW, ++ S3C2410_DISRCC_INC | S3C2410_DISRCC_APB, ++ prtd->params->dma_addr); ++ } ++ ++ s3c2410_dma_config(prtd->params->channel,2,S3C2410_DCON_HANDSHAKE); ++ ++ s3c2410_dma_set_buffdone_fn(prtd->params->channel, s3c24xx_audio_buffdone); ++ ++ prtd->dma_buffer = runtime->dma_addr; ++ prtd->dma_buffer_end = runtime->dma_addr + runtime->dma_bytes; ++ prtd->period_size = params_period_bytes(params); ++ ++ return 0; ++} ++ ++static int s3c24xx_pcm_hw_free(struct snd_pcm_substream *substream) ++{ ++ struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; ++ ++ printk("Entered s3c24xx hw_free\n"); ++ ++ return 0; ++} ++ ++static int s3c24xx_pcm_prepare(struct snd_pcm_substream *substream) ++{ ++ struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; ++ ++ printk("Entered s3c24xx prepare\n"); ++ ++ /* Set the period that is to be queued in DMA */ ++ prtd->period_ptr = prtd->dma_buffer; ++ ++ /* queue the first period */ ++ s3c2410_dma_enqueue(prtd->params->channel, substream, prtd->period_ptr, prtd->period_size); ++ ++ /* Move to next period to be queued */ ++ next_period(prtd); ++ ++ /* queue the second buffer */ ++ s3c2410_dma_enqueue(prtd->params->channel, substream, prtd->period_ptr, prtd->period_size); ++ ++ ++ return 0; ++} ++ ++static int s3c24xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) ++{ ++ struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; ++ int ret = 0; ++ ++ printk("Entered s3c24xx trigger\n"); ++ switch (cmd) { ++ case SNDRV_PCM_TRIGGER_START: ++ case SNDRV_PCM_TRIGGER_RESUME: ++ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: ++ s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_START); ++ break; ++ ++ case SNDRV_PCM_TRIGGER_STOP: ++ case SNDRV_PCM_TRIGGER_SUSPEND: ++ case SNDRV_PCM_TRIGGER_PAUSE_PUSH: ++ s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_STOP); ++ break; ++ ++ default: ++ ret = -EINVAL; ++ } ++ ++ return ret; ++} ++ ++static snd_pcm_uframes_t s3c24xx_pcm_pointer(struct snd_pcm_substream *substream) ++{ ++ struct snd_pcm_runtime *runtime = substream->runtime; ++ struct s3c24xx_runtime_data *prtd = runtime->private_data; ++ dma_addr_t dst,src; ++ snd_pcm_uframes_t x; ++ ++ printk("Entered s3c24xx pointer\n"); ++ ++ s3c2410_dma_getposition(prtd->params->channel, &src, &dst); ++ ++ printk("DMA Position: %lx, %lx\n", src, dst); ++ ++ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ++ { ++ x = bytes_to_frames(runtime, src - prtd->dma_buffer); ++ } ++ else ++ { ++ x = bytes_to_frames(runtime, dst - prtd->dma_buffer); ++ } ++ ++ if (x == runtime->buffer_size) ++ x=0; ++ return x; ++ ++} ++ ++static int s3c24xx_pcm_open(struct snd_pcm_substream *substream) ++{ ++ struct snd_pcm_runtime *runtime = substream->runtime; ++ struct s3c24xx_runtime_data *prtd; ++ int ret; ++ ++ printk("Entered s3c24xx open\n"); ++ ++ snd_soc_set_runtime_hwparams(substream, &s3c24xx_pcm_hardware); ++ ++ if((prtd = kzalloc(sizeof(struct s3c24xx_runtime_data), GFP_KERNEL)) == NULL) ++ { ++ ret = -ENOMEM; ++ goto out; ++ } ++ ++ runtime->private_data = prtd; ++ return 0; ++ ++out: ++ return ret; ++} ++ ++static int s3c24xx_pcm_close(struct snd_pcm_substream *substream) ++{ ++ struct snd_pcm_runtime *runtime = substream->runtime; ++ struct s3c24xx_runtime_data *prtd = runtime->private_data; ++ ++ printk("Entered s3c24xx close\n"); ++ ++ s3c2410_dma_free(prtd->params->channel, prtd->params->client); ++ ++ return 0; ++} ++ ++static int ++s3c24xx_pcm_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *vma) ++{ ++ struct snd_pcm_runtime *runtime = substream->runtime; ++ ++ printk("Entered s3c24xx mmap\n"); ++ ++ return dma_mmap_writecombine(substream->pcm->card->dev, vma, ++ runtime->dma_area, ++ runtime->dma_addr, ++ runtime->dma_bytes); ++} ++ ++struct snd_pcm_ops s3c24xx_pcm_ops = { ++ .open = s3c24xx_pcm_open, ++ .close = s3c24xx_pcm_close, ++ .ioctl = snd_pcm_lib_ioctl, ++ .hw_params = s3c24xx_pcm_hw_params, ++ .hw_free = s3c24xx_pcm_hw_free, ++ .prepare = s3c24xx_pcm_prepare, ++ .trigger = s3c24xx_pcm_trigger, ++ .pointer = s3c24xx_pcm_pointer, ++ .mmap = s3c24xx_pcm_mmap, ++}; ++ ++static int s3c24xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) ++{ ++ struct snd_pcm_substream *substream = pcm->streams[stream].substream; ++ struct snd_dma_buffer *buf = &substream->dma_buffer; ++ size_t size = s3c24xx_pcm_hardware.buffer_bytes_max; ++ ++ printk("Entered s3c24xx preaccolate_dma_buffer\n"); ++ ++ buf->dev.type = SNDRV_DMA_TYPE_DEV; ++ buf->dev.dev = pcm->card->dev; ++ buf->private_data = NULL; ++ buf->area = dma_alloc_writecombine(pcm->card->dev, size, ++ &buf->addr, GFP_KERNEL); ++ if (!buf->area) ++ return -ENOMEM; ++ buf->bytes = size; ++ return 0; ++} ++ ++static void s3c24xx_pcm_free_dma_buffers(struct snd_pcm *pcm) ++{ ++ struct snd_pcm_substream *substream; ++ struct snd_dma_buffer *buf; ++ int stream; ++ ++ printk("Entered s3c24xx free_dma_buffers\n"); ++ ++ for (stream = 0; stream < 2; stream++) { ++ substream = pcm->streams[stream].substream; ++ if (!substream) ++ continue; ++ ++ buf = &substream->dma_buffer; ++ if (!buf->area) ++ continue; ++ ++ dma_free_writecombine(pcm->card->dev, buf->bytes, ++ buf->area, buf->addr); ++ buf->area = NULL; ++ } ++} ++ ++static u64 s3c24xx_pcm_dmamask = 0xffffffff; ++ ++int s3c24xx_pcm_new(struct snd_card *card, struct snd_soc_codec_dai *dai, ++ struct snd_pcm *pcm) ++{ ++ int ret = 0; ++ ++ printk("Entered s3c24xx new\n"); ++ ++ if (!card->dev->dma_mask) ++ card->dev->dma_mask = &s3c24xx_pcm_dmamask; ++ if (!card->dev->coherent_dma_mask) ++ card->dev->coherent_dma_mask = 0xffffffff; ++ ++ if (dai->playback.channels_min) { ++ ret = s3c24xx_pcm_preallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_PLAYBACK); ++ if (ret) ++ goto out; ++ } ++ ++ if (dai->capture.channels_min) { ++ ret = s3c24xx_pcm_preallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_CAPTURE); ++ if (ret) ++ goto out; ++ } ++ out: ++ return ret; ++} ++ ++struct snd_soc_platform s3c24xx_soc_platform = { ++ .name = "s3c24xx-audio", ++ .pcm_ops = &s3c24xx_pcm_ops, ++ .pcm_new = s3c24xx_pcm_new, ++ .pcm_free = s3c24xx_pcm_free_dma_buffers, ++}; ++ ++EXPORT_SYMBOL_GPL(s3c24xx_soc_platform); ++ ++MODULE_AUTHOR("Graeme Gregory"); ++MODULE_DESCRIPTION("Samsung S3C24XX PCM DMA module"); ++MODULE_LICENSE("GPL"); +Index: linux-2.6-pxa-new/sound/soc/s3c24xx/Kconfig +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/sound/soc/s3c24xx/Kconfig +@@ -0,0 +1,26 @@ ++menu "SoC Audio for the Atmel AT91" ++ ++config SND_S3C24XX_SOC ++ tristate "SoC Audio for the Samsung S3C24xx System-on-Chip" ++ depends on ARCH_S3C2410 && SND ++ select SND_PCM ++ help ++ Say Y or M if you want to add support for codecs attached to ++ the Samsung S3C24xx. ++ ++config SND_S3C24XX_SOC_I2S ++ tristate ++ ++config SND_S3C24XX_SOC_AC97 ++ tristate ++ ++# graeme - add mach dep ++config SND_S3C24XX_SOC_SMDK2440 ++ tristate "SoC I2S Audio support for SMDK2440" ++ depends on SND_S3C24XX_SOC ++ select SND_S3C24XX_SOC_I2S ++ select SND_SOC_UDA1380 ++ help ++ Say Y if you want to add support for SoC audio on ++ ++endmenu +Index: linux-2.6-pxa-new/sound/soc/s3c24xx/Makefile +=================================================================== +--- /dev/null ++++ linux-2.6-pxa-new/sound/soc/s3c24xx/Makefile +@@ -0,0 +1,11 @@ ++# S3C24xx Platform Support ++snd-soc-s3c24xx-objs := s3c24xx-pcm.o ++snd-soc-at91-i2s-objs := s3c24xx-i2s.o ++ ++obj-$(CONFIG_SND_S3C24XX_SOC) += snd-soc-s3c24xx.o ++obj-$(CONFIG_SND_S3C24XX_SOC_I2S) += snd-soc-s3c24xx-i2s.o ++ ++# S3C24xx Machine Support ++snd-soc-smdk2440-uda1380-objs := smdk2440_uda1380.o ++ ++obj-$(CONFIG_SND_S3C24XX_SOC_SMDK2440) += snd-soc-smdk2440-uda1380.o |