diff -uNr flite-1.3-release/configure.in flite-1.3-release-mod/configure.in

--- flite-1.3-release/configure.in	2005-08-13 13:43:21.000000000 +0200

+++ flite-1.3-release-mod/configure.in	2006-11-13 21:16:27.000000000 +0200

@@ -206,10 +206,10 @@

 AC_CHECK_HEADER(sys/audioio.h,

               [AUDIODRIVER="sun"

                AUDIODEFS=-DCST_AUDIO_SUNOS])

-dnl AC_CHECK_HEADER(sys/asoundlib.h,

-dnl              [AUDIODRIVER="alsa"

-dnl	       AUDIODEFS=-DCST_AUDIO_ALSA

-dnl               AUDIOLIBS=-lasound])

+AC_CHECK_HEADER(alsa/asoundlib.h,

+              [AUDIODRIVER="alsa"

+	       AUDIODEFS=-DCST_AUDIO_ALSA

+               AUDIOLIBS=-lasound])

 AC_CHECK_HEADER(mmsystem.h,

 	      [AUDIODRIVER="wince"

 	       AUDIODEFS=-DCST_AUDIO_WINCE

diff -uNr flite-1.3-release/src/audio/au_alsa.c flite-1.3-release-mod/src/audio/au_alsa.c

--- flite-1.3-release/src/audio/au_alsa.c	1970-01-01 02:00:00.000000000 +0200

+++ flite-1.3-release-mod/src/audio/au_alsa.c	2006-11-13 21:16:54.000000000 +0200

@@ -0,0 +1,311 @@

+/*************************************************************************/

+/*                                                                       */

+/*                  Language Technologies Institute                      */

+/*                     Carnegie Mellon University                        */

+/*                        Copyright (c) 2000                             */

+/*                        All Rights Reserved.                           */

+/*                                                                       */

+/*  Permission is hereby granted, free of charge, to use and distribute  */

+/*  this software and its documentation without restriction, including   */

+/*  without limitation the rights to use, copy, modify, merge, publish,  */

+/*  distribute, sublicense, and/or sell copies of this work, and to      */

+/*  permit persons to whom this work is furnished to do so, subject to   */

+/*  the following conditions:                                            */

+/*   1. The code must retain the above copyright notice, this list of    */

+/*      conditions and the following disclaimer.                         */

+/*   2. Any modifications must be clearly marked as such.                */

+/*   3. Original authors' names are not deleted.                         */

+/*   4. The authors' names are not used to endorse or promote products   */

+/*      derived from this software without specific prior written        */

+/*      permission.                                                      */

+/*                                                                       */

+/*  CARNEGIE MELLON UNIVERSITY AND THE CONTRIBUTORS TO THIS WORK         */

+/*  DISCLAIM ALL WARRANTIES WITH REGARD TO THIS SOFTWARE, INCLUDING      */

+/*  ALL IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS, IN NO EVENT   */

+/*  SHALL CARNEGIE MELLON UNIVERSITY NOR THE CONTRIBUTORS BE LIABLE      */

+/*  FOR ANY SPECIAL, INDIRECT OR CONSEQUENTIAL DAMAGES OR ANY DAMAGES    */

+/*  WHATSOEVER RESULTING FROM LOSS OF USE, DATA OR PROFITS, WHETHER IN   */

+/*  AN ACTION OF CONTRACT, NEGLIGENCE OR OTHER TORTIOUS ACTION,          */

+/*  ARISING OUT OF OR IN CONNECTION WITH THE USE OR PERFORMANCE OF       */

+/*  THIS SOFTWARE.                                                       */

+/*                                                                       */

+/*********************************************************************** */

+/*             Author:  Lukas Loehrer (                                  */

+/*               Date:  January 2005                                     */

+/*************************************************************************/

+/*                                                                       */

+/*  Native access to alsa audio devices on Linux                         */

+/*  Tested with libasound version 1.0.10                                 */

+/*************************************************************************/

+

+#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 <alsa/asoundlib.h>

+

+

+/*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)

+{

+  cst_audiodev *ad;

+  unsigned 	int real_rate;

+  int err;

+

+  /* 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);

+

+  /* 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;

+  }

+

+  /* 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;

+  }

+

+  /* 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;

+  }

+

+  /* 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)

+	  format = SND_PCM_FORMAT_S16_LE;

+	else

+	  format = SND_PCM_FORMAT_S16_BE;

+	break;

+  case CST_AUDIO_LINEAR8:

+	format = SND_PCM_FORMAT_U8;

+	break;

+  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;

+  }

+

+  /* 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;

+  }

+

+  /* 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);

+

+  /* 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;

+  }

+

+  /* 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)

+{

+  int result;

+  snd_pcm_t *pcm_handle;

+

+  if (ad == NULL)

+	return 0;

+

+  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;

+}

+

+/* 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)

+{

+  size_t frame_size;

+  ssize_t num_frames, res;

+  snd_pcm_t *pcm_handle;

+  char *buf = (char *) samples;

+

+  /* 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)

+{

+  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)

+{

+  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;

+}