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;
+}