ref: ddca83a2f0f45046c7008bb3e19da5294a1bd160
dir: /src/audio/au_alsa.c/
/*************************************************************************/
/* */
/* Language Technologies Institute */
/* Carnegie Mellon University */
/* Copyright (c) 2005 */
/* 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 */
/* */
/* Added snd_config_update_free_global(); after every close to stop */
/* (apparent?) memory leaks */
/*************************************************************************/
#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 const 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(unsigned 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);
snd_config_update_free_global();
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);
snd_config_update_free_global();
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);
snd_config_update_free_global();
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);
snd_config_update_free_global();
cst_errmsg("audio_open_alsa: failed to set format. %s.\n", snd_strerror(err));
return NULL;
}
/* Set sample rate near the desired rate */
real_rate = sps;
err = snd_pcm_hw_params_set_rate(pcm_handle, hwparams, real_rate, 0);
if (err < 0)
{
snd_pcm_close(pcm_handle);
snd_config_update_free_global();
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);
snd_config_update_free_global();
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);
snd_config_update_free_global();
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);
/* snd_pcm_hw_params_free(hwparams); */
/* There doesn't seem to be another way to set the latency -- if done
here, it works, if not, it looses the first 2s or so */
snd_pcm_set_params(pcm_handle,
format,
SND_PCM_ACCESS_RW_INTERLEAVED,
1,
real_rate,
1,
50000);
/* 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;
snd_pcm_drain(pcm_handle); /* wait for current stuff in buffer to finish */
result = snd_pcm_close(pcm_handle);
snd_config_update_free_global();
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;
}