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>

2 files changed, 524 insertions, 0 deletions

diff --git a/recipes/flite/flite-alsa-1.2/flite-1.2-alsa_support-1.2.diff b/recipes/flite/flite-alsa-1.2/flite-1.2-alsa_support-1.2.diff
new file mode 100644
index 0000000000..3d2753a01f
--- /dev/null
+++ b/recipes/flite/flite-alsa-1.2/flite-1.2-alsa_support-1.2.diff
@@ -0,0 +1,512 @@
+Index: configure
+===================================================================
+--- flite-1.2-release/configure	(.../flite-1.2-orig)	(revision 10)
++++ flite-1.2-release/configure	(.../release-v1.2)	(revision 10)
+@@ -1415,16 +1415,16 @@
+   echo "$ac_t""no" 1>&6
+ fi
+ 
+-ac_safe=`echo "sys/asoundlib.h" | sed 'y%./+-%__p_%'`
+-echo $ac_n "checking for sys/asoundlib.h""... $ac_c" 1>&6
+-echo "configure:1421: checking for sys/asoundlib.h" >&5
++ac_safe=`echo "alsa/asoundlib.h" | sed 'y%./+-%__p_%'`
++echo $ac_n "checking for alsa/asoundlib.h""... $ac_c" 1>&6
++echo "configure:1421: checking for alsa/asoundlib.h" >&5
+ if eval "test \"`echo '$''{'ac_cv_header_$ac_safe'+set}'`\" = set"; then
+   echo $ac_n "(cached) $ac_c" 1>&6
+ else
+   cat > conftest.$ac_ext <<EOF
+ #line 1426 "configure"
+ #include "confdefs.h"
+-#include <sys/asoundlib.h>
++#include <alsa/asoundlib.h>
+ EOF
+ ac_try="$ac_cpp conftest.$ac_ext >/dev/null 2>conftest.out"
+ { (eval echo configure:1431: \"$ac_try\") 1>&5; (eval $ac_try) 2>&5; }
+@@ -1445,23 +1445,24 @@
+   echo "$ac_t""yes" 1>&6
+   AUDIODRIVER="alsa"
+ 	       AUDIODEFS=-DCST_AUDIO_ALSA
++		   AUDIOLIBS=-lasound
+ else
+   echo "$ac_t""no" 1>&6
+ fi
+ 
+ ac_safe=`echo "mmsystem.h" | sed 'y%./+-%__p_%'`
+ echo $ac_n "checking for mmsystem.h""... $ac_c" 1>&6
+-echo "configure:1455: checking for mmsystem.h" >&5
++echo "configure:1456: checking for mmsystem.h" >&5
+ if eval "test \"`echo '$''{'ac_cv_header_$ac_safe'+set}'`\" = set"; then
+   echo $ac_n "(cached) $ac_c" 1>&6
+ else
+   cat > conftest.$ac_ext <<EOF
+-#line 1460 "configure"
++#line 1461 "configure"
+ #include "confdefs.h"
+ #include <mmsystem.h>
+ EOF
+ ac_try="$ac_cpp conftest.$ac_ext >/dev/null 2>conftest.out"
+-{ (eval echo configure:1465: \"$ac_try\") 1>&5; (eval $ac_try) 2>&5; }
++{ (eval echo configure:1466: \"$ac_try\") 1>&5; (eval $ac_try) 2>&5; }
+ ac_err=`grep -v '^ *+' conftest.out | grep -v "^conftest.${ac_ext}\$"`
+ if test -z "$ac_err"; then
+   rm -rf conftest*
+Index: include/cst_sts.h
+===================================================================
+--- flite-1.2-release/include/cst_sts.h	(.../flite-1.2-orig)	(revision 10)
++++ flite-1.2-release/include/cst_sts.h	(.../release-v1.2)	(revision 10)
+@@ -47,9 +47,9 @@
+ /* else where, this information plus the indexes in the Unit relation */
+ /* allow reconstruction of the signal itself                          */
+ struct cst_sts_struct {
+-    const unsigned short *frame;  
+-    const int size;      /* in samples */
+-    const unsigned char *residual;
++    unsigned short *frame;  
++    int size;      /* in samples */
++    unsigned char *residual;
+ };
+ typedef struct cst_sts_struct cst_sts;
+ 
+Index: configure.in
+===================================================================
+--- flite-1.2-release/configure.in	(.../flite-1.2-orig)	(revision 10)
++++ flite-1.2-release/configure.in	(.../release-v1.2)	(revision 10)
+@@ -131,9 +131,10 @@
+ AC_CHECK_HEADER(sys/audioio.h,
+               [AUDIODRIVER="sun"
+                AUDIODEFS=-DCST_AUDIO_SUNOS])
+-AC_CHECK_HEADER(sys/asoundlib.h,
++AC_CHECK_HEADER(alsa/asoundlib.h,
+               [AUDIODRIVER="alsa"
+-	       AUDIODEFS=-DCST_AUDIO_ALSA])
++	       AUDIODEFS=-DCST_AUDIO_ALSA
++		   AUDIOLIBS=-lasound])
+ AC_CHECK_HEADER(mmsystem.h,
+ 	      [AUDIODRIVER="wince"
+ 	       AUDIODEFS=-DCST_AUDIO_WINCE
+Index: src/audio/au_alsa.c
+===================================================================
+--- flite-1.2-release/src/audio/au_alsa.c	(.../flite-1.2-orig)	(revision 10)
++++ flite-1.2-release/src/audio/au_alsa.c	(.../release-v1.2)	(revision 10)
+@@ -2,7 +2,7 @@
+ /*                                                                       */
+ /*                  Language Technologies Institute                      */
+ /*                     Carnegie Mellon University                        */
+-/*                        Copyright (c) 2001                             */
++/*                        Copyright (c) 2000                             */
+ /*                        All Rights Reserved.                           */
+ /*                                                                       */
+ /*  Permission is hereby granted, free of charge, to use and distribute  */
+@@ -29,158 +29,283 @@
+ /*  ARISING OUT OF OR IN CONNECTION WITH THE USE OR PERFORMANCE OF       */
+ /*  THIS SOFTWARE.                                                       */
+ /*                                                                       */
++/*********************************************************************** */
++/*             Author:  Lukas Loehrer (                                  */
++/*               Date:  January 2005                                     */
+ /*************************************************************************/
+-/*             Author:  Geoff Harrison (mandrake@cepstral.com)           */
+-/*               Date:  Sepetember 2001                                  */
+-/*************************************************************************/
+ /*                                                                       */
+-/*  Access to ALSA audio devices                                          */
+-/*                                                                       */
++/*  Native access to alsa audio devices on Linux                         */
++/*  Tested with libasound version 1.0.10                                 */
+ /*************************************************************************/
+ 
+-#include <stdio.h>
+ #include <stdlib.h>
+ #include <unistd.h>
+ #include <sys/types.h>
++#include <assert.h>
++#include <errno.h>
++
+ #include "cst_string.h"
+ #include "cst_wave.h"
+ #include "cst_audio.h"
+ 
+-#include <sys/asoundlib.h>
++#include <alsa/asoundlib.h>
+ 
+-#include <sys/stat.h>
+-#include <fcntl.h>
+ 
+-static int alsa_card = 0, alsa_device = 0;
++/*static char *pcm_dev_name = "hw:0,0"; */
++static char *pcm_dev_name ="default";
+ 
++static inline void print_pcm_state(snd_pcm_t *handle, char *msg)
++{
++  fprintf(stderr, "PCM state at %s = %s\n", msg,
++		  snd_pcm_state_name(snd_pcm_state(handle)));
++}
++
+ cst_audiodev *audio_open_alsa(int sps, int channels, cst_audiofmt fmt)
+ {
+-    snd_pcm_channel_info_t pinfo;
+-    snd_pcm_channel_params_t params;
+-    snd_pcm_channel_setup_t setup;
+-    snd_pcm_t *pcm;
+-    cst_audiodev *ad;
+-    int err;
++  cst_audiodev *ad;
++  unsigned 	int real_rate;
++  int err;
+ 
+-#ifdef __QNXNTO__
+-    if (snd_pcm_open_preferred(&pcm,&alsa_card,&alsa_device,SND_PCM_OPEN_PLAYBACK) < 0)
+-    {
+-	cst_errmsg("alsa_audio: failed to open audio device\n");
+-	cst_error();
+-    }
+-    if (snd_pcm_plugin_set_disable(pcm,PLUGIN_DISABLE_MMAP) < 0)
+-    {
+-	cst_errmsg("alsa_audio: failed to disable mmap\n");
+-	snd_pcm_close(pcm);
+-	cst_error();
+-    }
+-#else
+-    if (snd_pcm_open(&pcm,alsa_card,alsa_device,SND_PCM_OPEN_PLAYBACK) < 0)
+-    {
+-	cst_errmsg("alsa_audio: failed to open audio device\n");
+-	cst_error();
+-    }
+-#endif
++  /* alsa specific stuff */
++  snd_pcm_t *pcm_handle;          
++  snd_pcm_stream_t stream = SND_PCM_STREAM_PLAYBACK;
++  snd_pcm_hw_params_t *hwparams;
++  snd_pcm_format_t format;
++  snd_pcm_access_t access = SND_PCM_ACCESS_RW_INTERLEAVED;
+ 
++  /* Allocate the snd_pcm_hw_params_t structure on the stack. */
++  snd_pcm_hw_params_alloca(&hwparams);
+ 
+-    memset(&pinfo, 0, sizeof(pinfo));
+-    memset(&params, 0, sizeof(params));
+-    memset(&setup, 0, sizeof(setup));
++  /* Open pcm device */
++  err = snd_pcm_open(&pcm_handle, pcm_dev_name, stream, 0);
++  if (err < 0) 
++  {
++	cst_errmsg("audio_open_alsa: failed to open audio device %s. %s\n",
++			   pcm_dev_name, snd_strerror(err));
++	return NULL;
++  }
+ 
+-    pinfo.channel = SND_PCM_CHANNEL_PLAYBACK;
+-    snd_pcm_plugin_info(pcm,&pinfo);
++  /* Init hwparams with full configuration space */
++  err = snd_pcm_hw_params_any(pcm_handle, hwparams);
++  if (err < 0) 
++  {
++	snd_pcm_close(pcm_handle);
++	cst_errmsg("audio_open_alsa: failed to get hardware parameters from audio device. %s\n", snd_strerror(err));
++	return NULL;
++  }
+ 
+-    params.mode = SND_PCM_MODE_BLOCK;
+-    params.channel = SND_PCM_CHANNEL_PLAYBACK;
+-    params.start_mode = SND_PCM_START_DATA;
+-    params.stop_mode = SND_PCM_STOP_STOP;
++  /* Set access mode */
++  err = snd_pcm_hw_params_set_access(pcm_handle, hwparams, access);
++  if (err < 0) 
++  {
++	snd_pcm_close(pcm_handle);
++	cst_errmsg("audio_open_alsa: failed to set access mode. %s.\n", snd_strerror(err));
++	return NULL;
++  }
+ 
+-    params.buf.block.frag_size = pinfo.max_fragment_size;
+-    params.buf.block.frags_max = 1;
+-    params.buf.block.frags_min = 1;
+-    
+-    params.format.interleave = 1;
+-    params.format.rate = sps;
+-    params.format.voices = channels;
+-
+-    switch (fmt)
+-    {
+-    case CST_AUDIO_LINEAR16:
++  /* Determine matching alsa sample format */
++  /* This could be implemented in a more */
++  /* flexible way (byte order conversion). */
++  switch (fmt)
++  {
++  case CST_AUDIO_LINEAR16:
+ 	if (CST_LITTLE_ENDIAN)
+-	    params.format.format = SND_PCM_SFMT_S16_LE;
++	  format = SND_PCM_FORMAT_S16_LE;
+ 	else
+-	    params.format.format = SND_PCM_SFMT_S16_BE;
++	  format = SND_PCM_FORMAT_S16_BE;
+ 	break;
+-    case CST_AUDIO_LINEAR8:
+-	params.format.format = SND_PCM_SFMT_U8;
++  case CST_AUDIO_LINEAR8:
++	format = SND_PCM_FORMAT_U8;
+ 	break;
+-    case CST_AUDIO_MULAW:
+-	params.format.format = SND_PCM_SFMT_MU_LAW;
++  case CST_AUDIO_MULAW:
++	format = SND_PCM_FORMAT_MU_LAW;
+ 	break;
+-    }
++  default:
++	snd_pcm_close(pcm_handle);
++	cst_errmsg("audio_open_alsa: failed to find suitable format.\n");
++	return NULL;
++	break;
++  }
+ 
+-    if((err = snd_pcm_plugin_params(pcm,&params)) < 0)
+-    {
+-	cst_errmsg("alsa_audio params setting failed: %s\n",snd_strerror(err));
+-	snd_pcm_close(pcm);	
+-	cst_error();
+-    }
+-    if((err = snd_pcm_plugin_setup(pcm,SND_PCM_CHANNEL_PLAYBACK)) > 0) {
+-	cst_errmsg("alsa_audio sound prepare setting failed: %s\n",snd_strerror(err));
+-	snd_pcm_close(pcm);
+-	cst_error();
+-    }
+-    if((err = snd_pcm_plugin_prepare(pcm,SND_PCM_CHANNEL_PLAYBACK)) > 0) {
+-	cst_errmsg("alsa_audio sound prepare setting failed: %s\n",snd_strerror(err));
+-	snd_pcm_close(pcm);
+-	cst_error();
+-    }
++  /* Set samble format */
++  err = snd_pcm_hw_params_set_format(pcm_handle, hwparams, format);
++  if (err <0) 
++  {
++	snd_pcm_close(pcm_handle);
++	cst_errmsg("audio_open_alsa: failed to set format. %s.\n", snd_strerror(err));
++	return NULL;
++  }
+ 
+-    pinfo.channel = SND_PCM_CHANNEL_PLAYBACK;
+-    snd_pcm_plugin_info(pcm,&pinfo);
++  /* Set sample rate near the disired rate */
++  real_rate = sps;
++  err = snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams, &real_rate, 0);
++  if (err < 0)   
++  {
++	snd_pcm_close(pcm_handle);
++	cst_errmsg("audio_open_alsa: failed to set sample rate near %d. %s.\n", sps, snd_strerror(err));
++	return NULL;
++  }
++  /*FIXME:  This is probably too strict */
++  assert(sps == real_rate);
+ 
+-    ad = cst_alloc(cst_audiodev, 1);
+-    ad->platform_data = pcm;
+-    ad->sps = ad->real_sps = sps;
+-    ad->channels = ad->real_channels = channels;
+-    ad->fmt = ad->real_fmt = fmt;
++  /* Set number of channels */
++  assert(channels >0);
++  err = snd_pcm_hw_params_set_channels(pcm_handle, hwparams, channels);
++  if (err < 0) 
++  {
++	snd_pcm_close(pcm_handle);
++	cst_errmsg("audio_open_alsa: failed to set number of channels to %d. %s.\n", channels, snd_strerror(err));
++	return NULL;
++  }
+ 
+-    return ad;
++  /* Commit hardware parameters */
++  err = snd_pcm_hw_params(pcm_handle, hwparams);
++  if (err < 0) 
++  {
++	snd_pcm_close(pcm_handle);
++	cst_errmsg("audio_open_alsa: failed to set hw parameters. %s.\n", snd_strerror(err));
++	return NULL;
++  }
++
++  /* Make sure the device is ready to accept data */
++  assert(snd_pcm_state(pcm_handle) == SND_PCM_STATE_PREPARED);
++
++  /* Write hardware parameters to flite audio device data structure */
++  ad = cst_alloc(cst_audiodev, 1);
++  assert(ad != NULL);
++  ad->real_sps = ad->sps = sps;
++  ad->real_channels = ad->channels = channels;
++  ad->real_fmt = ad->fmt = fmt;
++  ad->platform_data = (void *) pcm_handle;
++
++  return ad;
+ }
+ 
+ int audio_close_alsa(cst_audiodev *ad)
+ {
+-    snd_pcm_t *pcm;
++  int result;
++  snd_pcm_t *pcm_handle;
+ 
+-    if (ad == NULL)
+-	    return 0;
++  if (ad == NULL)
++	return 0;
+ 
+-    pcm = ad->platform_data;
+-    snd_pcm_plugin_flush(pcm,0);
+-    snd_pcm_close(pcm);
+-    cst_free(ad);
++  pcm_handle = (snd_pcm_t *) ad->platform_data;
++  result = snd_pcm_close(pcm_handle);
++  if (result < 0)
++  {
++	cst_errmsg("audio_close_alsa: Error: %s.\n", snd_strerror(result));
++  }
++  cst_free(ad);
++  return result;
++}
+ 
+-    return 0;
++/* Returns zero if recovery was successful. */
++static int recover_from_error(snd_pcm_t *pcm_handle, ssize_t res)
++{
++  if (res == -EPIPE) /* xrun */
++  {
++	res = snd_pcm_prepare(pcm_handle);
++	if (res < 0) 
++	{
++	  /* Failed to recover from xrun */
++	  cst_errmsg("recover_from_write_error: failed to recover from xrun. %s\n.", snd_strerror(res));
++	  return res;
++	}
++  } 
++  else if (res == -ESTRPIPE) /* Suspend */
++  {
++	while ((res = snd_pcm_resume(pcm_handle)) == -EAGAIN) 
++	{
++	  snd_pcm_wait(pcm_handle, 1000);
++	}
++	if (res < 0) 
++	{
++	  res = snd_pcm_prepare(pcm_handle);
++	  if (res <0) 
++	  {
++		/* Resume failed */
++		cst_errmsg("audio_recover_from_write_error: failed to resume after suspend. %s\n.", snd_strerror(res));
++		return res;
++	  }
++	}
++  } 
++  else if (res < 0) 
++  {
++	/* Unknown failure */
++	cst_errmsg("audio_recover_from_write_error: %s.\n", snd_strerror(res));
++	return res;
++  }
++  return 0;
+ }
+ 
+ int audio_write_alsa(cst_audiodev *ad, void *samples, int num_bytes)
+ {
+-    snd_pcm_t *pcm = ad->platform_data;
++  size_t frame_size;
++  ssize_t num_frames, res;
++  snd_pcm_t *pcm_handle;
++  char *buf = (char *) samples;
+ 
+-    return snd_pcm_plugin_write(pcm,samples,num_bytes);
++  /* Determine frame size in bytes */
++  frame_size  = audio_bps(ad->real_fmt) * ad->real_channels;
++  /* Require that only complete frames are handed in */
++  assert((num_bytes % frame_size) == 0);
++  num_frames = num_bytes / frame_size;
++  pcm_handle = (snd_pcm_t *) ad->platform_data;
++
++  while (num_frames > 0) 
++  {
++	res = snd_pcm_writei(pcm_handle, buf, num_frames);
++	if (res != num_frames) 
++	{
++	  if (res == -EAGAIN || (res > 0 && res < num_frames)) 
++	  {
++		snd_pcm_wait(pcm_handle, 100);
++	  }
++	  else if (recover_from_error(pcm_handle, res) < 0) 
++	  {
++		return -1;
++	  }
++	}
++
++	if (res >0) 
++	{
++	  num_frames -= res;
++	  buf += res * frame_size;
++	}
++  }
++  return num_bytes;
+ }
+ 
+ int audio_flush_alsa(cst_audiodev *ad)
+ {
+-    snd_pcm_t *pcm = ad->platform_data;
+-
+-    return snd_pcm_plugin_flush(pcm,0);
++  int result;
++  result = snd_pcm_drain((snd_pcm_t *) ad->platform_data);
++  if (result < 0)
++  {
++	cst_errmsg("audio_flush_alsa: Error: %s.\n", snd_strerror(result));
++  }
++	/* Prepare device for more data */
++  result = snd_pcm_prepare((snd_pcm_t *) ad->platform_data);
++if (result < 0)
++  {
++	cst_errmsg("audio_flush_alsa: Error: %s.\n", snd_strerror(result));
++  }
++  return result;
+ }
+ 
+ int audio_drain_alsa(cst_audiodev *ad)
+ {
+-    snd_pcm_t *pcm = ad->platform_data;
+-
+-    return snd_pcm_plugin_playback_drain(pcm);
++  int result;
++  result = snd_pcm_drop((snd_pcm_t *) ad->platform_data);
++  if (result < 0)
++  {
++	cst_errmsg("audio_drain_alsa: Error: %s.\n", snd_strerror(result));
++  }
++/* Prepare device for more data */
++  result = snd_pcm_prepare((snd_pcm_t *) ad->platform_data);
++if (result < 0)
++  {
++	cst_errmsg("audio_drain_alsa: Error: %s.\n", snd_strerror(result));
++  }
++  return result;
+ }
+-
+Index: doc/Makefile
+===================================================================
+--- flite-1.2-release/doc/Makefile	(.../flite-1.2-orig)	(revision 10)
++++ flite-1.2-release/doc/Makefile	(.../release-v1.2)	(revision 10)
+@@ -53,6 +53,7 @@
+ 	@ if [ ! -d html ] ; \
+           then mkdir -p html ; fi
+ 	(cd html; texi2html -number -split_chapter ../flite.texi)
++	mv html/flite/*.html html/ && rmdir html/flite
+ 	@ for i in html/*.html ; \
+ 	  do \
+ 	    sed 's/<BODY>/<BODY bgcolor="#ffffff">/' $$i >ttt.html; \
+Index: config/common_make_rules
+===================================================================
+--- flite-1.2-release/config/common_make_rules	(.../flite-1.2-orig)	(revision 10)
++++ flite-1.2-release/config/common_make_rules	(.../release-v1.2)	(revision 10)
+@@ -88,7 +88,7 @@
+ 	@ rm -rf shared_os && mkdir shared_os
+ 	@ rm -f $@ $(LIBDIR)/$@.${PROJECT_VERSION} $(LIBDIR)/$@.${PROJECT_SHLIB_VERSION} 
+ 	@ (cd shared_os && ar x ../$<)
+-	@ (cd shared_os && $(CC) -shared -Wl,-soname,$@.${PROJECT_SHLIB_VERSION} -o ../$@.${PROJECT_VERSION} *.os)
++	@ (cd shared_os && $(CC) -shared -Wl,-soname,$@.${PROJECT_SHLIB_VERSION} -o ../$@.${PROJECT_VERSION} *.os $(AUDIOLIBS))
+ 	@ ln -s $(LIBDIR)/$@.${PROJECT_VERSION} $(LIBDIR)/$@.${PROJECT_SHLIB_VERSION}
+ 	@ ln -s $(LIBDIR)/$@.${PROJECT_SHLIB_VERSION} $(LIBDIR)/$@
+ 	@ rm -rf shared_os
diff --git a/recipes/flite/flite-alsa-1.2/flite-alsa-1.2-configure-with-audio.patch b/recipes/flite/flite-alsa-1.2/flite-alsa-1.2-configure-with-audio.patch
new file mode 100644
index 0000000000..b344877d04
--- /dev/null
+++ b/recipes/flite/flite-alsa-1.2/flite-alsa-1.2-configure-with-audio.patch
@@ -0,0 +1,12 @@
+--- flite-1.2-release/configure.in.old	2008-01-06 02:30:57.000000000 -0600
++++ flite-1.2-release/configure.in	2008-01-06 02:31:32.000000000 -0600
+@@ -145,7 +145,8 @@ dnl allow the user to override the one d
+ dnl
+ AC_ARG_WITH( audio,
+ 	[  --with-audio          with specific audio support (none linux freebsd etc) ],
+-        AUDIODRIVER=$with_audio )
++        [AUDIODRIVER=$with_audio
++         AUDIODEFS=])
+ 
+ if test "x$AUDIODEFS" = x; then
+     case "$AUDIODRIVER" in