mpd/src/output/alsa_output_plugin.c

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/*
2011-01-29 10:13:54 +01:00
* Copyright (C) 2003-2011 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 "config.h"
#include "alsa_output_plugin.h"
#include "output_api.h"
#include "mixer_list.h"
#include "pcm_export.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 {
struct audio_output base;
struct pcm_export_state export;
/** the configured name of the ALSA device; NULL for the
default device */
char *device;
/** use memory mapped I/O? */
bool use_mmap;
/**
* Enable DSD over USB according to the dCS suggested
* standard?
*
* @see http://www.dcsltd.co.uk/page/assets/DSDoverUSB.pdf
*/
bool dsd_usb;
/** 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 passed to method play().
*/
size_t in_frame_size;
/**
* The size of one audio frame passed to libasound.
*/
size_t out_frame_size;
/**
* The size of one period, in number of frames.
*/
snd_pcm_uframes_t period_frames;
/**
* The number of frames written in the current period.
*/
snd_pcm_uframes_t period_position;
};
/**
* 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_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->dsd_usb = config_get_block_bool(param, "dsd_usb", 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 struct audio_output *
alsa_init(const struct config_param *param, GError **error_r)
{
struct alsa_data *ad = alsa_data_new();
if (!ao_base_init(&ad->base, &alsa_output_plugin, param, error_r)) {
g_free(ad);
return NULL;
}
alsa_configure(ad, param);
return &ad->base;
}
static void
alsa_finish(struct audio_output *ao)
{
struct alsa_data *ad = (struct alsa_data *)ao;
ao_base_finish(&ad->base);
g_free(ad->device);
g_free(ad);
/* free libasound's config cache */
snd_config_update_free_global();
}
static bool
alsa_output_enable(struct audio_output *ao, G_GNUC_UNUSED GError **error_r)
{
struct alsa_data *ad = (struct alsa_data *)ao;
pcm_export_init(&ad->export);
return true;
}
static void
alsa_output_disable(struct audio_output *ao)
{
struct alsa_data *ad = (struct alsa_data *)ao;
pcm_export_deinit(&ad->export);
}
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(enum sample_format sample_format)
{
switch (sample_format) {
case SAMPLE_FORMAT_UNDEFINED:
case SAMPLE_FORMAT_DSD:
return SND_PCM_FORMAT_UNKNOWN;
case SAMPLE_FORMAT_S8:
return SND_PCM_FORMAT_S8;
case SAMPLE_FORMAT_S16:
return SND_PCM_FORMAT_S16;
case SAMPLE_FORMAT_S24_P32:
return SND_PCM_FORMAT_S24;
case SAMPLE_FORMAT_S32:
return SND_PCM_FORMAT_S32;
case SAMPLE_FORMAT_FLOAT:
return SND_PCM_FORMAT_FLOAT;
}
assert(false);
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_S24_3BE:
return SND_PCM_FORMAT_S24_3LE;
case SND_PCM_FORMAT_S24_3LE:
return SND_PCM_FORMAT_S24_3BE;
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case SND_PCM_FORMAT_S32_BE: return SND_PCM_FORMAT_S32_LE;
default: return SND_PCM_FORMAT_UNKNOWN;
}
}
static snd_pcm_format_t
alsa_to_packed_format(snd_pcm_format_t fmt)
{
switch (fmt) {
case SND_PCM_FORMAT_S24_LE:
return SND_PCM_FORMAT_S24_3LE;
case SND_PCM_FORMAT_S24_BE:
return SND_PCM_FORMAT_S24_3BE;
default:
return SND_PCM_FORMAT_UNKNOWN;
}
}
static int
alsa_try_format_or_packed(snd_pcm_t *pcm, snd_pcm_hw_params_t *hwparams,
snd_pcm_format_t fmt, bool *packed_r)
{
int err = snd_pcm_hw_params_set_format(pcm, hwparams, fmt);
if (err == 0)
*packed_r = false;
if (err != -EINVAL)
return err;
fmt = alsa_to_packed_format(fmt);
if (fmt == SND_PCM_FORMAT_UNKNOWN)
return -EINVAL;
err = snd_pcm_hw_params_set_format(pcm, hwparams, fmt);
if (err == 0)
*packed_r = true;
return err;
}
/**
* Attempts to configure the specified sample format, and tries the
* reversed host byte order if was not supported.
*/
static int
alsa_output_try_format(snd_pcm_t *pcm, snd_pcm_hw_params_t *hwparams,
enum sample_format sample_format,
bool *packed_r, bool *reverse_endian_r)
{
snd_pcm_format_t alsa_format = get_bitformat(sample_format);
if (alsa_format == SND_PCM_FORMAT_UNKNOWN)
return -EINVAL;
int err = alsa_try_format_or_packed(pcm, hwparams, alsa_format,
packed_r);
if (err == 0)
*reverse_endian_r = false;
if (err != -EINVAL)
return err;
alsa_format = byteswap_bitformat(alsa_format);
if (alsa_format == SND_PCM_FORMAT_UNKNOWN)
return -EINVAL;
err = alsa_try_format_or_packed(pcm, hwparams, alsa_format, packed_r);
if (err == 0)
*reverse_endian_r = true;
return err;
}
/**
* Configure a sample format, and probe other formats if that fails.
*/
static int
alsa_output_setup_format(snd_pcm_t *pcm, snd_pcm_hw_params_t *hwparams,
struct audio_format *audio_format,
bool *packed_r, bool *reverse_endian_r)
{
/* try the input format first */
int err = alsa_output_try_format(pcm, hwparams, audio_format->format,
packed_r, reverse_endian_r);
/* if unsupported by the hardware, try other formats */
static const enum sample_format probe_formats[] = {
SAMPLE_FORMAT_S24_P32,
SAMPLE_FORMAT_S32,
SAMPLE_FORMAT_S16,
SAMPLE_FORMAT_S8,
SAMPLE_FORMAT_UNDEFINED,
};
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for (unsigned i = 0;
err == -EINVAL && probe_formats[i] != SAMPLE_FORMAT_UNDEFINED;
++i) {
const enum sample_format mpd_format = probe_formats[i];
if (mpd_format == audio_format->format)
continue;
err = alsa_output_try_format(pcm, hwparams, mpd_format,
packed_r, reverse_endian_r);
if (err == 0)
audio_format->format = mpd_format;
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}
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return err;
}
/**
* 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,
bool *packed_r, bool *reverse_endian_r, 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) {
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 = alsa_output_setup_format(ad->pcm, hwparams, audio_format,
packed_r, reverse_endian_r);
if (err < 0) {
g_set_error(error, alsa_output_quark(), err,
"ALSA device \"%s\" does not support format %s: %s",
alsa_device(ad),
sample_format_to_string(audio_format->format),
snd_strerror(-err));
return false;
}
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snd_pcm_format_t format;
if (snd_pcm_hw_params_get_format(hwparams, &format) == 0)
g_debug("format=%s (%s)", snd_pcm_format_name(format),
snd_pcm_format_description(format));
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;
snd_pcm_uframes_t buffer_size_min, buffer_size_max;
snd_pcm_hw_params_get_buffer_size_min(hwparams, &buffer_size_min);
snd_pcm_hw_params_get_buffer_size_max(hwparams, &buffer_size_max);
unsigned buffer_time_min, buffer_time_max;
snd_pcm_hw_params_get_buffer_time_min(hwparams, &buffer_time_min, 0);
snd_pcm_hw_params_get_buffer_time_max(hwparams, &buffer_time_max, 0);
g_debug("buffer: size=%u..%u time=%u..%u",
(unsigned)buffer_size_min, (unsigned)buffer_size_max,
buffer_time_min, buffer_time_max);
snd_pcm_uframes_t period_size_min, period_size_max;
snd_pcm_hw_params_get_period_size_min(hwparams, &period_size_min, 0);
snd_pcm_hw_params_get_period_size_max(hwparams, &period_size_max, 0);
unsigned period_time_min, period_time_max;
snd_pcm_hw_params_get_period_time_min(hwparams, &period_time_min, 0);
snd_pcm_hw_params_get_period_time_max(hwparams, &period_time_max, 0);
g_debug("period: size=%u..%u time=%u..%u",
(unsigned)period_size_min, (unsigned)period_size_max,
period_time_min, period_time_max);
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);
if (alsa_period_size == 0)
/* this works around a SIGFPE bug that occurred when
an ALSA driver indicated period_size==0; this
caused a division by zero in alsa_play(). By using
the fallback "1", we make sure that this won't
happen again. */
alsa_period_size = 1;
ad->period_frames = alsa_period_size;
ad->period_position = 0;
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_setup_dsd(struct alsa_data *ad, struct audio_format *audio_format,
bool *shift8_r, bool *packed_r, bool *reverse_endian_r,
GError **error_r)
{
assert(ad->dsd_usb);
assert(audio_format->format == SAMPLE_FORMAT_DSD);
/* pass 24 bit to alsa_setup() */
struct audio_format usb_format = *audio_format;
usb_format.format = SAMPLE_FORMAT_S24_P32;
usb_format.sample_rate /= 2;
const struct audio_format check = usb_format;
if (!alsa_setup(ad, &usb_format, packed_r, reverse_endian_r, error_r))
return false;
/* if the device allows only 32 bit, shift all DSD-over-USB
samples left by 8 bit and leave the lower 8 bit cleared;
the DSD-over-USB documentation does not specify whether
this is legal, but there is anecdotical evidence that this
is possible (and the only option for some devices) */
*shift8_r = usb_format.format == SAMPLE_FORMAT_S32;
if (usb_format.format == SAMPLE_FORMAT_S32)
usb_format.format = SAMPLE_FORMAT_S24_P32;
if (!audio_format_equals(&usb_format, &check)) {
/* no bit-perfect playback, which is required
for DSD over USB */
g_set_error(error_r, alsa_output_quark(), 0,
"Failed to configure DSD-over-USB on ALSA device \"%s\"",
alsa_device(ad));
return false;
}
return true;
}
static bool
alsa_setup_or_dsd(struct alsa_data *ad, struct audio_format *audio_format,
GError **error_r)
{
bool shift8 = false, packed, reverse_endian;
const bool dsd_usb = ad->dsd_usb &&
audio_format->format == SAMPLE_FORMAT_DSD;
const bool success = dsd_usb
? alsa_setup_dsd(ad, audio_format,
&shift8, &packed, &reverse_endian,
error_r)
: alsa_setup(ad, audio_format, &packed, &reverse_endian,
error_r);
if (!success)
return false;
pcm_export_open(&ad->export,
audio_format->format, audio_format->channels,
dsd_usb, shift8, packed, reverse_endian);
return true;
}
static bool
alsa_open(struct audio_output *ao, struct audio_format *audio_format, GError **error)
{
struct alsa_data *ad = (struct alsa_data *)ao;
int err;
bool success;
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;
}
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g_debug("opened %s type=%s", snd_pcm_name(ad->pcm),
snd_pcm_type_name(snd_pcm_type(ad->pcm)));
success = alsa_setup_or_dsd(ad, audio_format, error);
if (!success) {
snd_pcm_close(ad->pcm);
return false;
}
ad->in_frame_size = audio_format_frame_size(audio_format);
ad->out_frame_size = ad->export.pack24 ? 3 : ad->in_frame_size;
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:
ad->period_position = 0;
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_drain(struct audio_output *ao)
{
struct alsa_data *ad = (struct alsa_data *)ao;
if (snd_pcm_state(ad->pcm) != SND_PCM_STATE_RUNNING)
return;
if (ad->period_position > 0) {
/* generate some silence to finish the partial
period */
snd_pcm_uframes_t nframes =
ad->period_frames - ad->period_position;
size_t nbytes = nframes * ad->out_frame_size;
void *buffer = g_malloc(nbytes);
snd_pcm_hw_params_t *params;
snd_pcm_format_t format;
unsigned channels;
snd_pcm_hw_params_alloca(&params);
snd_pcm_hw_params_current(ad->pcm, params);
snd_pcm_hw_params_get_format(params, &format);
snd_pcm_hw_params_get_channels(params, &channels);
snd_pcm_format_set_silence(format, buffer, nframes * channels);
ad->writei(ad->pcm, buffer, nframes);
g_free(buffer);
}
snd_pcm_drain(ad->pcm);
ad->period_position = 0;
}
static void
alsa_cancel(struct audio_output *ao)
{
struct alsa_data *ad = (struct alsa_data *)ao;
ad->period_position = 0;
snd_pcm_drop(ad->pcm);
}
static void
alsa_close(struct audio_output *ao)
{
struct alsa_data *ad = (struct alsa_data *)ao;
snd_pcm_close(ad->pcm);
}
static size_t
alsa_play(struct audio_output *ao, const void *chunk, size_t size,
GError **error)
{
struct alsa_data *ad = (struct alsa_data *)ao;
assert(size % ad->in_frame_size == 0);
chunk = pcm_export(&ad->export, chunk, size, &size);
assert(size % ad->out_frame_size == 0);
size /= ad->out_frame_size;
while (true) {
snd_pcm_sframes_t ret = ad->writei(ad->pcm, chunk, size);
if (ret > 0) {
ad->period_position = (ad->period_position + ret)
% ad->period_frames;
return pcm_export_source_size(&ad->export,
ret * ad->in_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 alsa_output_plugin = {
.name = "alsa",
.test_default_device = alsa_test_default_device,
.init = alsa_init,
.finish = alsa_finish,
.enable = alsa_output_enable,
.disable = alsa_output_disable,
.open = alsa_open,
.play = alsa_play,
.drain = alsa_drain,
.cancel = alsa_cancel,
.close = alsa_close,
.mixer_plugin = &alsa_mixer_plugin,
};