path: root/packages/flite/flite-alsa-1.2/flite-1.2-alsa_support-1.2.diff

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