@@ -0,0 +1,333 @@
+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;
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