mpd/src/output/alsa_plugin.c

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/*
* Copyright (C) 2003-2009 The Music Player Daemon Project
* 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.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
#include "../output_api.h"
#include "mixer_list.h"
#include <glib.h>
#include <alsa/asoundlib.h>
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#undef G_LOG_DOMAIN
#define G_LOG_DOMAIN "alsa"
#define ALSA_PCM_NEW_HW_PARAMS_API
#define ALSA_PCM_NEW_SW_PARAMS_API
static const char default_device[] = "default";
enum {
MPD_ALSA_BUFFER_TIME_US = 500000,
};
#define MPD_ALSA_RETRY_NR 5
typedef snd_pcm_sframes_t alsa_writei_t(snd_pcm_t * pcm, const void *buffer,
snd_pcm_uframes_t size);
struct alsa_data {
/** the configured name of the ALSA device; NULL for the
default device */
char *device;
/** use memory mapped I/O? */
bool use_mmap;
/** libasound's buffer_time setting (in microseconds) */
unsigned int buffer_time;
/** libasound's period_time setting (in microseconds) */
unsigned int period_time;
/** the mode flags passed to snd_pcm_open */
int mode;
/** the libasound PCM device handle */
snd_pcm_t *pcm;
/**
* a pointer to the libasound writei() function, which is
* snd_pcm_writei() or snd_pcm_mmap_writei(), depending on the
* use_mmap configuration
*/
alsa_writei_t *writei;
/** the size of one audio frame */
size_t frame_size;
};
/**
* The quark used for GError.domain.
*/
static inline GQuark
alsa_output_quark(void)
{
return g_quark_from_static_string("alsa_output");
}
static const char *
alsa_device(const struct alsa_data *ad)
{
return ad->device != NULL ? ad->device : default_device;
}
static struct alsa_data *
alsa_data_new(void)
{
struct alsa_data *ret = g_new(struct alsa_data, 1);
ret->mode = 0;
ret->writei = snd_pcm_writei;
return ret;
}
static void
alsa_data_free(struct alsa_data *ad)
{
g_free(ad->device);
g_free(ad);
}
static void
alsa_configure(struct alsa_data *ad, const struct config_param *param)
{
ad->device = config_dup_block_string(param, "device", NULL);
ad->use_mmap = config_get_block_bool(param, "use_mmap", false);
ad->buffer_time = config_get_block_unsigned(param, "buffer_time",
MPD_ALSA_BUFFER_TIME_US);
ad->period_time = config_get_block_unsigned(param, "period_time", 0);
#ifdef SND_PCM_NO_AUTO_RESAMPLE
if (!config_get_block_bool(param, "auto_resample", true))
ad->mode |= SND_PCM_NO_AUTO_RESAMPLE;
#endif
#ifdef SND_PCM_NO_AUTO_CHANNELS
if (!config_get_block_bool(param, "auto_channels", true))
ad->mode |= SND_PCM_NO_AUTO_CHANNELS;
#endif
#ifdef SND_PCM_NO_AUTO_FORMAT
if (!config_get_block_bool(param, "auto_format", true))
ad->mode |= SND_PCM_NO_AUTO_FORMAT;
#endif
}
static void *
alsa_init(G_GNUC_UNUSED const struct audio_format *audio_format,
const struct config_param *param,
G_GNUC_UNUSED GError **error)
{
struct alsa_data *ad = alsa_data_new();
alsa_configure(ad, param);
return ad;
}
static void
alsa_finish(void *data)
{
struct alsa_data *ad = data;
alsa_data_free(ad);
/* free libasound's config cache */
snd_config_update_free_global();
}
static bool
alsa_test_default_device(void)
{
snd_pcm_t *handle;
int ret = snd_pcm_open(&handle, default_device,
SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
if (ret) {
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g_message("Error opening default ALSA device: %s\n",
snd_strerror(-ret));
return false;
} else
snd_pcm_close(handle);
return true;
}
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;
}
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static snd_pcm_format_t
byteswap_bitformat(snd_pcm_format_t fmt)
{
switch(fmt) {
case SND_PCM_FORMAT_S16_LE: return SND_PCM_FORMAT_S16_BE;
case SND_PCM_FORMAT_S24_LE: return SND_PCM_FORMAT_S24_BE;
case SND_PCM_FORMAT_S32_LE: return SND_PCM_FORMAT_S32_BE;
case SND_PCM_FORMAT_S16_BE: return SND_PCM_FORMAT_S16_LE;
case SND_PCM_FORMAT_S24_BE: return SND_PCM_FORMAT_S24_LE;
case SND_PCM_FORMAT_S32_BE: return SND_PCM_FORMAT_S32_LE;
default: return SND_PCM_FORMAT_UNKNOWN;
}
}
/**
* Set up the snd_pcm_t object which was opened by the caller. Set up
* the configured settings and the audio format.
*/
static bool
alsa_setup(struct alsa_data *ad, struct audio_format *audio_format,
snd_pcm_format_t bitformat,
GError **error)
{
snd_pcm_hw_params_t *hwparams;
snd_pcm_sw_params_t *swparams;
unsigned int sample_rate = audio_format->sample_rate;
unsigned int channels = audio_format->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;
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->pcm, hwparams);
if (err < 0)
goto error;
if (ad->use_mmap) {
err = snd_pcm_hw_params_set_access(ad->pcm, hwparams,
SND_PCM_ACCESS_MMAP_INTERLEAVED);
if (err < 0) {
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g_warning("Cannot set mmap'ed mode on ALSA device \"%s\": %s\n",
alsa_device(ad), snd_strerror(-err));
g_warning("Falling back to direct write mode\n");
ad->use_mmap = false;
} else
ad->writei = snd_pcm_mmap_writei;
}
if (!ad->use_mmap) {
cmd = "snd_pcm_hw_params_set_access";
err = snd_pcm_hw_params_set_access(ad->pcm, 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, hwparams, bitformat);
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if (err == -EINVAL &&
byteswap_bitformat(bitformat) != SND_PCM_FORMAT_UNKNOWN) {
err = snd_pcm_hw_params_set_format(ad->pcm, hwparams,
byteswap_bitformat(bitformat));
if (err == 0) {
g_debug("ALSA device \"%s\": converting %u bit to reverse-endian\n",
alsa_device(ad), audio_format->bits);
audio_format->reverse_endian = 1;
}
}
if (err == -EINVAL && (audio_format->bits == 24 ||
audio_format->bits == 16)) {
/* fall back to 32 bit, let pcm_convert.c do the conversion */
err = snd_pcm_hw_params_set_format(ad->pcm, hwparams,
SND_PCM_FORMAT_S32);
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if (err == 0) {
g_debug("ALSA device \"%s\": converting %u bit to 32 bit\n",
alsa_device(ad), audio_format->bits);
audio_format->bits = 32;
}
}
if (err == -EINVAL && (audio_format->bits == 24 ||
audio_format->bits == 16)) {
/* fall back to 32 bit, let pcm_convert.c do the conversion */
err = snd_pcm_hw_params_set_format(ad->pcm, hwparams,
byteswap_bitformat(SND_PCM_FORMAT_S32));
if (err == 0) {
g_debug("ALSA device \"%s\": converting %u bit to 32 bit backward-endian\n",
alsa_device(ad), audio_format->bits);
audio_format->bits = 32;
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audio_format->reverse_endian = 1;
}
}
if (err == -EINVAL && audio_format->bits != 16) {
/* fall back to 16 bit, let pcm_convert.c do the conversion */
err = snd_pcm_hw_params_set_format(ad->pcm, hwparams,
SND_PCM_FORMAT_S16);
if (err == 0) {
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g_debug("ALSA device \"%s\": converting %u bit to 16 bit\n",
alsa_device(ad), audio_format->bits);
audio_format->bits = 16;
}
}
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if (err == -EINVAL && audio_format->bits != 16) {
/* fall back to 16 bit, let pcm_convert.c do the conversion */
err = snd_pcm_hw_params_set_format(ad->pcm, hwparams,
byteswap_bitformat(SND_PCM_FORMAT_S16));
if (err == 0) {
g_debug("ALSA device \"%s\": converting %u bit to 16 bit backward-endian\n",
alsa_device(ad), audio_format->bits);
audio_format->bits = 16;
audio_format->reverse_endian = 1;
}
}
if (err < 0) {
g_set_error(error, alsa_output_quark(), err,
"ALSA device \"%s\" does not support %u bit audio: %s",
alsa_device(ad), audio_format->bits,
snd_strerror(-err));
return false;
}
err = snd_pcm_hw_params_set_channels_near(ad->pcm, hwparams,
&channels);
if (err < 0) {
g_set_error(error, alsa_output_quark(), err,
"ALSA device \"%s\" does not support %i channels: %s",
alsa_device(ad), (int)audio_format->channels,
snd_strerror(-err));
return false;
}
audio_format->channels = (int8_t)channels;
err = snd_pcm_hw_params_set_rate_near(ad->pcm, hwparams,
&sample_rate, NULL);
if (err < 0 || sample_rate == 0) {
g_set_error(error, alsa_output_quark(), err,
"ALSA device \"%s\" does not support %u Hz audio",
alsa_device(ad), audio_format->sample_rate);
return false;
}
audio_format->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->pcm, hwparams,
&buffer_time, NULL);
if (err < 0)
goto error;
} else {
err = snd_pcm_hw_params_get_buffer_time(hwparams, &buffer_time,
NULL);
if (err < 0)
buffer_time = 0;
}
if (period_time_ro == 0 && buffer_time >= 10000) {
period_time_ro = period_time = buffer_time / 4;
g_debug("default period_time = buffer_time/4 = %u/4 = %u",
buffer_time, period_time);
}
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->pcm, hwparams,
&period_time, NULL);
if (err < 0)
goto error;
}
cmd = "snd_pcm_hw_params";
err = snd_pcm_hw_params(ad->pcm, 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)
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g_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->pcm, swparams);
if (err < 0)
goto error;
cmd = "snd_pcm_sw_params_set_start_threshold";
err = snd_pcm_sw_params_set_start_threshold(ad->pcm, 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->pcm, swparams,
alsa_period_size);
if (err < 0)
goto error;
cmd = "snd_pcm_sw_params";
err = snd_pcm_sw_params(ad->pcm, swparams);
if (err < 0)
goto error;
g_debug("buffer_size=%u period_size=%u",
(unsigned)alsa_buffer_size, (unsigned)alsa_period_size);
return true;
error:
g_set_error(error, alsa_output_quark(), err,
"Error opening ALSA device \"%s\" (%s): %s",
alsa_device(ad), cmd, snd_strerror(-err));
return false;
}
static bool
alsa_open(void *data, struct audio_format *audio_format, GError **error)
{
struct alsa_data *ad = data;
snd_pcm_format_t bitformat;
int err;
bool success;
bitformat = get_bitformat(audio_format);
if (bitformat == SND_PCM_FORMAT_UNKNOWN) {
/* sample format is not supported by this plugin -
fall back to 16 bit samples */
audio_format->bits = 16;
bitformat = SND_PCM_FORMAT_S16;
}
err = snd_pcm_open(&ad->pcm, alsa_device(ad),
SND_PCM_STREAM_PLAYBACK, ad->mode);
if (err < 0) {
g_set_error(error, alsa_output_quark(), err,
"Failed to open ALSA device \"%s\": %s",
alsa_device(ad), snd_strerror(err));
return false;
}
success = alsa_setup(ad, audio_format, bitformat, error);
if (!success) {
snd_pcm_close(ad->pcm);
return false;
}
ad->frame_size = audio_format_frame_size(audio_format);
return true;
}
static int
alsa_recover(struct alsa_data *ad, int err)
{
if (err == -EPIPE) {
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g_debug("Underrun on ALSA device \"%s\"\n", alsa_device(ad));
} else if (err == -ESTRPIPE) {
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g_debug("ALSA device \"%s\" was suspended\n", alsa_device(ad));
}
switch (snd_pcm_state(ad->pcm)) {
case SND_PCM_STATE_PAUSED:
err = snd_pcm_pause(ad->pcm, /* disable */ 0);
break;
case SND_PCM_STATE_SUSPENDED:
err = snd_pcm_resume(ad->pcm);
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->pcm);
break;
case SND_PCM_STATE_DISCONNECTED:
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_cancel(void *data)
{
struct alsa_data *ad = data;
alsa_recover(ad, snd_pcm_drop(ad->pcm));
}
static void
alsa_close(void *data)
{
struct alsa_data *ad = data;
if (snd_pcm_state(ad->pcm) == SND_PCM_STATE_RUNNING)
snd_pcm_drain(ad->pcm);
snd_pcm_close(ad->pcm);
}
static size_t
alsa_play(void *data, const void *chunk, size_t size, GError **error)
{
struct alsa_data *ad = data;
size /= ad->frame_size;
while (true) {
snd_pcm_sframes_t ret = ad->writei(ad->pcm, chunk, size);
if (ret > 0)
return ret * ad->frame_size;
if (ret < 0 && ret != -EAGAIN && ret != -EINTR &&
alsa_recover(ad, ret) < 0) {
g_set_error(error, alsa_output_quark(), errno,
"%s", snd_strerror(-errno));
return 0;
}
}
}
const struct audio_output_plugin alsaPlugin = {
.name = "alsa",
.test_default_device = alsa_test_default_device,
.init = alsa_init,
.finish = alsa_finish,
.open = alsa_open,
.play = alsa_play,
.cancel = alsa_cancel,
.close = alsa_close,
.mixer_plugin = &alsa_mixer,
};