/* the Music Player Daemon (MPD)
* (c)2003-2004 by Warren Dukes (shank@mercury.chem.pitt.edu)
* This project's homepage is: http://www.musicpd.org
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*/
#include "../audioOutput.h"
#include <stdlib.h>
#ifdef HAVE_ALSA
#define ALSA_PCM_NEW_HW_PARAMS_API
#define ALSA_PCM_NEW_SW_PARAMS_API
#define MPD_ALSA_BUFFER_TIME 500000
#define MPD_ALSA_PERIOD_TIME 50000
#include "../conf.h"
#include "../log.h"
#include "../sig_handlers.h"
#include <string.h>
#include <assert.h>
#include <signal.h>
#include <alsa/asoundlib.h>
typedef snd_pcm_sframes_t alsa_writei_t(snd_pcm_t *pcm, const void *buffer,
snd_pcm_uframes_t size);
typedef struct _AlsaData {
char * device;
snd_pcm_t * pcm_handle;
int mmap;
alsa_writei_t * writei;
int sampleSize;
} AlsaData;
static AlsaData * newAlsaData() {
AlsaData * ret = malloc(sizeof(AlsaData));
ret->device = NULL;
ret->pcm_handle = NULL;
ret->writei = snd_pcm_writei;
ret->mmap = 0;
return ret;
}
static void freeAlsaData(AlsaData * ad) {
if(ad->device) free(ad->device);
free(ad);
}
static int alsa_initDriver(AudioOutput * audioOutput, ConfigParam * param) {
BlockParam * bp = getBlockParam(param, "device");
AlsaData * ad = newAlsaData();
audioOutput->data = ad;
ad->device = bp ? strdup(bp->value) : strdup("default");
return 0;
}
static void alsa_finishDriver(AudioOutput * audioOutput) {
AlsaData * ad = audioOutput->data;
freeAlsaData(ad);
}
static int alsa_openDevice(AudioOutput * audioOutput)
{
AlsaData * ad = audioOutput->data;
AudioFormat * audioFormat = &audioOutput->outAudioFormat;
snd_pcm_format_t bitformat;
snd_pcm_hw_params_t * hwparams;
snd_pcm_sw_params_t * swparams;
unsigned int sampleRate = audioFormat->sampleRate;
snd_pcm_uframes_t alsa_buffer_size;
snd_pcm_uframes_t alsa_period_size;
unsigned int alsa_buffer_time = MPD_ALSA_BUFFER_TIME;
unsigned int alsa_period_time = MPD_ALSA_PERIOD_TIME;
int err;
switch(audioFormat->bits) {
case 8:
bitformat = SND_PCM_FORMAT_S8;
break;
case 16:
bitformat = SND_PCM_FORMAT_S16;
break;
case 24:
bitformat = SND_PCM_FORMAT_S16;
break;
case 32:
bitformat = SND_PCM_FORMAT_S16;
break;
default:
ERROR("Alsa device \"%s\" doesn't support %i bit audio\n",
ad->device, audioFormat->bits);
return -1;
}
err = snd_pcm_open(&ad->pcm_handle, ad->device,
SND_PCM_STREAM_PLAYBACK, 0);
if(err < 0) {
ad->pcm_handle = NULL;
goto error;
}
/*err = snd_pcm_nonblock(ad->pcm_handle, 0);
if(err < 0) goto error;*/
/* configure HW params */
snd_pcm_hw_params_alloca(&hwparams);
err = snd_pcm_hw_params_any(ad->pcm_handle, hwparams);
if(err < 0) goto error;
if(ad->mmap) {
err = snd_pcm_hw_params_set_access(ad->pcm_handle, hwparams,
SND_PCM_ACCESS_MMAP_INTERLEAVED);
if(err < 0) {
ERROR("Cannot set mmap'ed mode on alsa device \"%s\": "
" %s\n", ad->device,
snd_strerror(-err));
ERROR("Falling back to direct write mode\n");
ad->mmap = 0;
}
else ad->writei = snd_pcm_mmap_writei;
}
if(!ad->mmap) {
err = snd_pcm_hw_params_set_access(ad->pcm_handle, hwparams,
SND_PCM_ACCESS_RW_INTERLEAVED);
if(err < 0) goto error;
ad->writei = snd_pcm_writei;
}
err = snd_pcm_hw_params_set_format(ad->pcm_handle, hwparams, bitformat);
if(err < 0) {
ERROR("Alsa device \"%s\" does not support %i bit audio: "
"%s\n", ad->device, (int)bitformat,
snd_strerror(-err));
goto fail;
}
err = snd_pcm_hw_params_set_channels(ad->pcm_handle, hwparams,
audioFormat->channels);
if(err < 0) {
ERROR("Alsa device \"%s\" does not support %i channels: "
"%s\n", ad->device, (int)audioFormat->channels,
snd_strerror(-err));
goto fail;
}
err = snd_pcm_hw_params_set_rate_near(ad->pcm_handle, hwparams,
&sampleRate, 0);
if(err < 0 || sampleRate == 0) {
ERROR("Alsa device \"%s\" does not support %i Hz audio\n",
ad->device, (int)audioFormat->sampleRate);
goto fail;
}
err = snd_pcm_hw_params_set_buffer_time_near(ad->pcm_handle, hwparams,
&alsa_buffer_time, 0);
if(err < 0) goto error;
err = snd_pcm_hw_params_set_period_time_near(ad->pcm_handle, hwparams,
&alsa_period_time, 0);
if(err < 0) goto error;
err = snd_pcm_hw_params(ad->pcm_handle, hwparams);
if(err < 0) goto error;
err = snd_pcm_hw_params_get_buffer_size(hwparams, &alsa_buffer_size);
if(err < 0) goto error;
err = snd_pcm_hw_params_get_period_size(hwparams, &alsa_period_size, 0);
if(err < 0) goto error;
/* configure SW params */
snd_pcm_sw_params_alloca(&swparams);
snd_pcm_sw_params_current(ad->pcm_handle, swparams);
err = snd_pcm_sw_params_set_start_threshold(ad->pcm_handle, swparams,
alsa_buffer_size - alsa_period_size);
if(err < 0) goto error;
err = snd_pcm_sw_params(ad->pcm_handle, swparams);
if(err < 0) goto error;
ad->sampleSize = (audioFormat->bits/8)*audioFormat->channels;
audioOutput->open = 1;
return 0;
error:
ERROR("Error opening alsa device \"%s\": %s\n", ad->device,
snd_strerror(-err));
fail:
if(ad->pcm_handle) snd_pcm_close(ad->pcm_handle);
ad->pcm_handle = NULL;
audioOutput->open = 0;
return -1;
}
static void alsa_closeDevice(AudioOutput * audioOutput) {
AlsaData * ad = audioOutput->data;
if(ad->pcm_handle) {
snd_pcm_drain(ad->pcm_handle);
ad->pcm_handle = NULL;
}
audioOutput->open = 0;
}
inline static int alsa_errorRecovery(AlsaData * ad, int err) {
if(err == -EPIPE) {
DEBUG("Underrun on alsa device \"%s\"\n", ad->device);
err = snd_pcm_prepare(ad->pcm_handle);
if(err < 0) return -1;
return 0;
}
return err;
}
static int alsa_playAudio(AudioOutput * audioOutput, char * playChunk,
int size)
{
AlsaData * ad = audioOutput->data;
int ret;
size /= ad->sampleSize;
while (size > 0) {
ret = ad->writei(ad->pcm_handle, playChunk, size);
if(ret == -EAGAIN) continue;
if(ret < 0 && alsa_errorRecovery(ad, ret) < 0) {
ERROR("closing alsa device \"%s\" due to write error:"
" %s\n", ad->device,
snd_strerror(-errno));
alsa_closeDevice(audioOutput);
return -1;
}
playChunk += ret * ad->sampleSize;
size -= ret;
}
return 0;
}
AudioOutputPlugin alsaPlugin =
{
"alsa",
alsa_initDriver,
alsa_finishDriver,
alsa_openDevice,
alsa_playAudio,
alsa_closeDevice,
NULL /* sendMetadataFunc */
};
#else /* HAVE ALSA */
AudioOutputPlugin alsaPlugin =
{
NULL,
NULL,
NULL,
NULL,
NULL,
NULL,
NULL /* sendMetadataFunc */
};
#endif /* HAVE_ALSA */