mpd/src/audioOutputs/audioOutput_alsa.c
Max Kellermann dd7711d86c alsa: don't override libasound's buffer_time and period_time
ALSA does a good job measuring its buffer_time and period_time.  Don't
override its defaults, unless the user demands it.
2008-10-11 12:52:48 +02:00

409 lines
10 KiB
C

/* the Music Player Daemon (MPD)
* Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
* 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 "../output_api.h"
#ifdef HAVE_ALSA
#define ALSA_PCM_NEW_HW_PARAMS_API
#define ALSA_PCM_NEW_SW_PARAMS_API
static const char default_device[] = "default";
#define MPD_ALSA_RETRY_NR 5
#include "../utils.h"
#include "../log.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 {
const char *device;
snd_pcm_t *pcmHandle;
alsa_writei_t *writei;
unsigned int buffer_time;
unsigned int period_time;
int sampleSize;
int useMmap;
} AlsaData;
static AlsaData *newAlsaData(void)
{
AlsaData *ret = xmalloc(sizeof(AlsaData));
ret->device = default_device;
ret->pcmHandle = NULL;
ret->writei = snd_pcm_writei;
ret->useMmap = 0;
ret->buffer_time = 0;
ret->period_time = 0;
return ret;
}
static void freeAlsaData(AlsaData * ad)
{
if (ad->device && ad->device != default_device)
xfree(ad->device);
free(ad);
}
static void *alsa_initDriver(mpd_unused struct audio_output *ao,
mpd_unused const struct audio_format *audio_format,
ConfigParam * param)
{
/* no need for pthread_once thread-safety when reading config */
static int free_global_registered;
AlsaData *ad = newAlsaData();
if (!free_global_registered) {
atexit((void(*)(void))snd_config_update_free_global);
free_global_registered = 1;
}
if (param) {
BlockParam *bp;
if ((bp = getBlockParam(param, "device")))
ad->device = xstrdup(bp->value);
ad->useMmap = getBoolBlockParam(param, "use_mmap", 1);
if (ad->useMmap == CONF_BOOL_UNSET)
ad->useMmap = 0;
if ((bp = getBlockParam(param, "buffer_time")))
ad->buffer_time = atoi(bp->value);
if ((bp = getBlockParam(param, "period_time")))
ad->period_time = atoi(bp->value);
}
return ad;
}
static void alsa_finishDriver(void *data)
{
AlsaData *ad = data;
freeAlsaData(ad);
}
static int alsa_testDefault(void)
{
snd_pcm_t *handle;
int ret = snd_pcm_open(&handle, default_device,
SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
if (ret) {
WARNING("Error opening default ALSA device: %s\n",
snd_strerror(-ret));
return -1;
} else
snd_pcm_close(handle);
return 0;
}
static snd_pcm_format_t get_bitformat(const struct audio_format *af)
{
switch (af->bits) {
case 8: return SND_PCM_FORMAT_S8;
case 16: return SND_PCM_FORMAT_S16;
case 24: return SND_PCM_FORMAT_S24;
case 32: return SND_PCM_FORMAT_S32;
}
return SND_PCM_FORMAT_UNKNOWN;
}
static int alsa_openDevice(void *data, struct audio_format *audioFormat)
{
AlsaData *ad = data;
snd_pcm_format_t bitformat;
snd_pcm_hw_params_t *hwparams;
snd_pcm_sw_params_t *swparams;
unsigned int sample_rate = audioFormat->sample_rate;
unsigned int channels = audioFormat->channels;
snd_pcm_uframes_t alsa_buffer_size;
snd_pcm_uframes_t alsa_period_size;
int err;
const char *cmd = NULL;
int retry = MPD_ALSA_RETRY_NR;
unsigned int period_time, period_time_ro;
unsigned int buffer_time;
if ((bitformat = get_bitformat(audioFormat)) == SND_PCM_FORMAT_UNKNOWN)
ERROR("ALSA device \"%s\" doesn't support %u bit audio\n",
ad->device, audioFormat->bits);
err = snd_pcm_open(&ad->pcmHandle, ad->device,
SND_PCM_STREAM_PLAYBACK, 0);
if (err < 0) {
ad->pcmHandle = NULL;
goto error;
}
period_time_ro = period_time = ad->period_time;
configure_hw:
/* configure HW params */
snd_pcm_hw_params_alloca(&hwparams);
cmd = "snd_pcm_hw_params_any";
err = snd_pcm_hw_params_any(ad->pcmHandle, hwparams);
if (err < 0)
goto error;
if (ad->useMmap) {
err = snd_pcm_hw_params_set_access(ad->pcmHandle, 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->useMmap = 0;
} else
ad->writei = snd_pcm_mmap_writei;
}
if (!ad->useMmap) {
cmd = "snd_pcm_hw_params_set_access";
err = snd_pcm_hw_params_set_access(ad->pcmHandle, hwparams,
SND_PCM_ACCESS_RW_INTERLEAVED);
if (err < 0)
goto error;
ad->writei = snd_pcm_writei;
}
err = snd_pcm_hw_params_set_format(ad->pcmHandle, hwparams, bitformat);
if (err < 0) {
ERROR("ALSA device \"%s\" does not support %u bit audio: "
"%s\n", ad->device, audioFormat->bits, snd_strerror(-err));
goto fail;
}
err = snd_pcm_hw_params_set_channels_near(ad->pcmHandle, hwparams,
&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;
}
audioFormat->channels = (int8_t)channels;
err = snd_pcm_hw_params_set_rate_near(ad->pcmHandle, hwparams,
&sample_rate, NULL);
if (err < 0 || sample_rate == 0) {
ERROR("ALSA device \"%s\" does not support %u Hz audio\n",
ad->device, audioFormat->sample_rate);
goto fail;
}
audioFormat->sample_rate = sample_rate;
if (ad->buffer_time > 0) {
buffer_time = ad->buffer_time;
cmd = "snd_pcm_hw_params_set_buffer_time_near";
err = snd_pcm_hw_params_set_buffer_time_near(ad->pcmHandle, hwparams,
&buffer_time, NULL);
if (err < 0)
goto error;
}
if (period_time_ro > 0) {
period_time = period_time_ro;
cmd = "snd_pcm_hw_params_set_period_time_near";
err = snd_pcm_hw_params_set_period_time_near(ad->pcmHandle, hwparams,
&period_time, NULL);
if (err < 0)
goto error;
}
cmd = "snd_pcm_hw_params";
err = snd_pcm_hw_params(ad->pcmHandle, hwparams);
if (err == -EPIPE && --retry > 0 && period_time_ro > 0) {
period_time_ro = period_time_ro >> 1;
goto configure_hw;
} else if (err < 0)
goto error;
if (retry != MPD_ALSA_RETRY_NR)
DEBUG("ALSA period_time set to %d\n", period_time);
cmd = "snd_pcm_hw_params_get_buffer_size";
err = snd_pcm_hw_params_get_buffer_size(hwparams, &alsa_buffer_size);
if (err < 0)
goto error;
cmd = "snd_pcm_hw_params_get_period_size";
err = snd_pcm_hw_params_get_period_size(hwparams, &alsa_period_size,
NULL);
if (err < 0)
goto error;
/* configure SW params */
snd_pcm_sw_params_alloca(&swparams);
cmd = "snd_pcm_sw_params_current";
err = snd_pcm_sw_params_current(ad->pcmHandle, swparams);
if (err < 0)
goto error;
cmd = "snd_pcm_sw_params_set_start_threshold";
err = snd_pcm_sw_params_set_start_threshold(ad->pcmHandle, swparams,
alsa_buffer_size -
alsa_period_size);
if (err < 0)
goto error;
cmd = "snd_pcm_sw_params_set_avail_min";
err = snd_pcm_sw_params_set_avail_min(ad->pcmHandle, swparams,
alsa_period_size);
if (err < 0)
goto error;
cmd = "snd_pcm_sw_params";
err = snd_pcm_sw_params(ad->pcmHandle, swparams);
if (err < 0)
goto error;
ad->sampleSize = audio_format_frame_size(audioFormat);
DEBUG("ALSA device \"%s\" will be playing %i bit, %u channel audio at "
"%u Hz\n", ad->device, audioFormat->bits,
channels, sample_rate);
return 0;
error:
if (cmd) {
ERROR("Error opening ALSA device \"%s\" (%s): %s\n",
ad->device, cmd, snd_strerror(-err));
} else {
ERROR("Error opening ALSA device \"%s\": %s\n", ad->device,
snd_strerror(-err));
}
fail:
if (ad->pcmHandle)
snd_pcm_close(ad->pcmHandle);
ad->pcmHandle = NULL;
return -1;
}
static int alsa_errorRecovery(AlsaData * ad, int err)
{
if (err == -EPIPE) {
DEBUG("Underrun on ALSA device \"%s\"\n", ad->device);
} else if (err == -ESTRPIPE) {
DEBUG("ALSA device \"%s\" was suspended\n", ad->device);
}
switch (snd_pcm_state(ad->pcmHandle)) {
case SND_PCM_STATE_PAUSED:
err = snd_pcm_pause(ad->pcmHandle, /* disable */ 0);
break;
case SND_PCM_STATE_SUSPENDED:
err = snd_pcm_resume(ad->pcmHandle);
if (err == -EAGAIN)
return 0;
/* fall-through to snd_pcm_prepare: */
case SND_PCM_STATE_SETUP:
case SND_PCM_STATE_XRUN:
err = snd_pcm_prepare(ad->pcmHandle);
break;
case SND_PCM_STATE_DISCONNECTED:
/* so alsa_closeDevice won't try to drain: */
snd_pcm_close(ad->pcmHandle);
ad->pcmHandle = NULL;
break;
/* this is no error, so just keep running */
case SND_PCM_STATE_RUNNING:
err = 0;
break;
default:
/* unknown state, do nothing */
break;
}
return err;
}
static void alsa_dropBufferedAudio(void *data)
{
AlsaData *ad = data;
alsa_errorRecovery(ad, snd_pcm_drop(ad->pcmHandle));
}
static void alsa_closeDevice(void *data)
{
AlsaData *ad = data;
if (ad->pcmHandle) {
if (snd_pcm_state(ad->pcmHandle) == SND_PCM_STATE_RUNNING) {
snd_pcm_drain(ad->pcmHandle);
}
snd_pcm_close(ad->pcmHandle);
ad->pcmHandle = NULL;
}
}
static int alsa_playAudio(void *data, const char *playChunk, size_t size)
{
AlsaData *ad = data;
int ret;
size /= ad->sampleSize;
while (size > 0) {
ret = ad->writei(ad->pcmHandle, playChunk, size);
if (ret == -EAGAIN || ret == -EINTR)
continue;
if (ret < 0) {
if (alsa_errorRecovery(ad, ret) < 0) {
ERROR("closing ALSA device \"%s\" due to write "
"error: %s\n", ad->device,
snd_strerror(-errno));
alsa_closeDevice(ad);
return -1;
}
continue;
}
playChunk += ret * ad->sampleSize;
size -= ret;
}
return 0;
}
const struct audio_output_plugin alsaPlugin = {
.name = "alsa",
.test_default_device = alsa_testDefault,
.init = alsa_initDriver,
.finish = alsa_finishDriver,
.open = alsa_openDevice,
.play = alsa_playAudio,
.cancel = alsa_dropBufferedAudio,
.close = alsa_closeDevice,
};
#else /* HAVE ALSA */
DISABLED_AUDIO_OUTPUT_PLUGIN(alsaPlugin)
#endif /* HAVE_ALSA */