mpd/src/output/plugins/AlsaOutputPlugin.cxx

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/*
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* Copyright 2003-2017 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"
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#include "AlsaOutputPlugin.hxx"
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#include "../OutputAPI.hxx"
#include "../Wrapper.hxx"
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#include "mixer/MixerList.hxx"
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#include "pcm/PcmExport.hxx"
#include "system/ByteOrder.hxx"
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#include "util/Manual.hxx"
#include "util/RuntimeError.hxx"
#include "util/Domain.hxx"
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#include "util/ConstBuffer.hxx"
#include "Log.hxx"
#include <alsa/asoundlib.h>
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#include <string>
#if SND_LIB_VERSION >= 0x1001c
/* alsa-lib supports DSD since version 1.0.27.1 */
#define HAVE_ALSA_DSD
#endif
#if SND_LIB_VERSION >= 0x1001d
/* alsa-lib supports DSD_U32 since version 1.0.29 */
#define HAVE_ALSA_DSD_U32
#endif
static const char default_device[] = "default";
static constexpr unsigned MPD_ALSA_BUFFER_TIME_US = 500000;
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static constexpr unsigned MPD_ALSA_RETRY_NR = 5;
class AlsaOutput {
friend struct AudioOutputWrapper<AlsaOutput>;
AudioOutput base;
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Manual<PcmExport> pcm_export;
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/**
* The configured name of the ALSA device; empty for the
* default device
*/
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const std::string device;
#ifdef ENABLE_DSD
/**
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* Enable DSD over PCM according to the DoP standard?
*
* @see http://dsd-guide.com/dop-open-standard
*/
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const bool dop;
#endif
/** libasound's buffer_time setting (in microseconds) */
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const unsigned buffer_time;
/** libasound's period_time setting (in microseconds) */
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const unsigned period_time;
/** the mode flags passed to snd_pcm_open */
int mode = 0;
/** the libasound PCM device handle */
snd_pcm_t *pcm;
/**
* 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;
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/**
* Do we need to call snd_pcm_prepare() before the next write?
* It means that we put the device to SND_PCM_STATE_SETUP by
* calling snd_pcm_drop().
*
* Without this flag, we could easily recover after a failed
* optimistic write (returning -EBADFD), but the Raspberry Pi
* audio driver is infamous for generating ugly artefacts from
* this.
*/
bool must_prepare;
/**
* This buffer gets allocated after opening the ALSA device.
* It contains silence samples, enough to fill one period (see
* #period_frames).
*/
uint8_t *silence;
public:
AlsaOutput(const ConfigBlock &block);
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~AlsaOutput() {
/* free libasound's config cache */
snd_config_update_free_global();
}
gcc_pure
const char *GetDevice() {
return device.empty() ? default_device : device.c_str();
}
static AlsaOutput *Create(const ConfigBlock &block);
void Enable();
void Disable();
void Open(AudioFormat &audio_format);
void Close();
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size_t PlayRaw(ConstBuffer<void> data);
size_t Play(const void *chunk, size_t size);
void Drain();
void Cancel();
private:
/**
* Set up the snd_pcm_t object which was opened by the caller.
* Set up the configured settings and the audio format.
*
* Throws #std::runtime_error on error.
*/
void Setup(AudioFormat &audio_format, PcmExport::Params &params);
#ifdef ENABLE_DSD
void SetupDop(AudioFormat audio_format,
PcmExport::Params &params);
#endif
void SetupOrDop(AudioFormat &audio_format, PcmExport::Params &params);
int Recover(int err);
/**
* Write silence to the ALSA device.
*/
void WriteSilence(snd_pcm_uframes_t nframes) {
snd_pcm_writei(pcm, silence, nframes);
}
};
static constexpr Domain alsa_output_domain("alsa_output");
AlsaOutput::AlsaOutput(const ConfigBlock &block)
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:base(alsa_output_plugin, block),
device(block.GetBlockValue("device", "")),
#ifdef ENABLE_DSD
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dop(block.GetBlockValue("dop", false) ||
/* legacy name from MPD 0.18 and older: */
block.GetBlockValue("dsd_usb", false)),
#endif
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buffer_time(block.GetBlockValue("buffer_time",
MPD_ALSA_BUFFER_TIME_US)),
period_time(block.GetBlockValue("period_time", 0u))
{
#ifdef SND_PCM_NO_AUTO_RESAMPLE
if (!block.GetBlockValue("auto_resample", true))
mode |= SND_PCM_NO_AUTO_RESAMPLE;
#endif
#ifdef SND_PCM_NO_AUTO_CHANNELS
if (!block.GetBlockValue("auto_channels", true))
mode |= SND_PCM_NO_AUTO_CHANNELS;
#endif
#ifdef SND_PCM_NO_AUTO_FORMAT
if (!block.GetBlockValue("auto_format", true))
mode |= SND_PCM_NO_AUTO_FORMAT;
#endif
}
inline AlsaOutput *
AlsaOutput::Create(const ConfigBlock &block)
{
return new AlsaOutput(block);
}
inline void
AlsaOutput::Enable()
{
pcm_export.Construct();
}
inline void
AlsaOutput::Disable()
{
pcm_export.Destruct();
}
static bool
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alsa_test_default_device()
{
snd_pcm_t *handle;
int ret = snd_pcm_open(&handle, default_device,
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SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
if (ret) {
FormatError(alsa_output_domain,
"Error opening default ALSA device: %s",
snd_strerror(-ret));
return false;
} else
snd_pcm_close(handle);
return true;
}
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/**
* Convert MPD's #SampleFormat enum to libasound's snd_pcm_format_t
* enum. Returns SND_PCM_FORMAT_UNKNOWN if there is no according ALSA
* PCM format.
*/
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gcc_const
static snd_pcm_format_t
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ToAlsaPcmFormat(SampleFormat sample_format)
{
switch (sample_format) {
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case SampleFormat::UNDEFINED:
return SND_PCM_FORMAT_UNKNOWN;
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case SampleFormat::DSD:
#ifdef HAVE_ALSA_DSD
return SND_PCM_FORMAT_DSD_U8;
#else
return SND_PCM_FORMAT_UNKNOWN;
#endif
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case SampleFormat::S8:
return SND_PCM_FORMAT_S8;
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case SampleFormat::S16:
return SND_PCM_FORMAT_S16;
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case SampleFormat::S24_P32:
return SND_PCM_FORMAT_S24;
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case SampleFormat::S32:
return SND_PCM_FORMAT_S32;
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case SampleFormat::FLOAT:
return SND_PCM_FORMAT_FLOAT;
}
assert(false);
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gcc_unreachable();
}
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/**
* Determine the byte-swapped PCM format. Returns
* SND_PCM_FORMAT_UNKNOWN if the format cannot be byte-swapped.
*/
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static snd_pcm_format_t
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ByteSwapAlsaPcmFormat(snd_pcm_format_t fmt)
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{
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switch (fmt) {
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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;
#ifdef HAVE_ALSA_DSD_U32
case SND_PCM_FORMAT_DSD_U16_LE:
return SND_PCM_FORMAT_DSD_U16_BE;
case SND_PCM_FORMAT_DSD_U16_BE:
return SND_PCM_FORMAT_DSD_U16_LE;
case SND_PCM_FORMAT_DSD_U32_LE:
return SND_PCM_FORMAT_DSD_U32_BE;
case SND_PCM_FORMAT_DSD_U32_BE:
return SND_PCM_FORMAT_DSD_U32_LE;
#endif
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default: return SND_PCM_FORMAT_UNKNOWN;
}
}
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/**
* Check if there is a "packed" version of the give PCM format.
* Returns SND_PCM_FORMAT_UNKNOWN if not.
*/
static snd_pcm_format_t
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PackAlsaPcmFormat(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;
}
}
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/**
* Attempts to configure the specified sample format. On failure,
* fall back to the packed version.
*/
static int
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AlsaTryFormatOrPacked(snd_pcm_t *pcm, snd_pcm_hw_params_t *hwparams,
snd_pcm_format_t fmt, PcmExport::Params &params)
{
int err = snd_pcm_hw_params_set_format(pcm, hwparams, fmt);
if (err == 0)
params.pack24 = false;
if (err != -EINVAL)
return err;
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fmt = PackAlsaPcmFormat(fmt);
if (fmt == SND_PCM_FORMAT_UNKNOWN)
return -EINVAL;
err = snd_pcm_hw_params_set_format(pcm, hwparams, fmt);
if (err == 0)
params.pack24 = true;
return err;
}
/**
* Attempts to configure the specified sample format, and tries the
* reversed host byte order if was not supported.
*/
static int
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AlsaTryFormatOrByteSwap(snd_pcm_t *pcm, snd_pcm_hw_params_t *hwparams,
snd_pcm_format_t fmt,
PcmExport::Params &params)
{
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int err = AlsaTryFormatOrPacked(pcm, hwparams, fmt, params);
if (err == 0)
params.reverse_endian = false;
if (err != -EINVAL)
return err;
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fmt = ByteSwapAlsaPcmFormat(fmt);
if (fmt == SND_PCM_FORMAT_UNKNOWN)
return -EINVAL;
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err = AlsaTryFormatOrPacked(pcm, hwparams, fmt, params);
if (err == 0)
params.reverse_endian = true;
return err;
}
/**
* Attempts to configure the specified sample format. On DSD_U8
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* failure, attempt to switch to DSD_U32 or DSD_U16.
*/
static int
AlsaTryFormatDsd(snd_pcm_t *pcm, snd_pcm_hw_params_t *hwparams,
snd_pcm_format_t fmt, PcmExport::Params &params)
{
int err = AlsaTryFormatOrByteSwap(pcm, hwparams, fmt, params);
#if defined(ENABLE_DSD) && defined(HAVE_ALSA_DSD_U32)
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if (err == 0) {
params.dsd_u16 = false;
params.dsd_u32 = false;
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}
if (err == -EINVAL && fmt == SND_PCM_FORMAT_DSD_U8) {
/* attempt to switch to DSD_U32 */
fmt = IsLittleEndian()
? SND_PCM_FORMAT_DSD_U32_LE
: SND_PCM_FORMAT_DSD_U32_BE;
err = AlsaTryFormatOrByteSwap(pcm, hwparams, fmt, params);
if (err == 0)
params.dsd_u32 = true;
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else
fmt = SND_PCM_FORMAT_DSD_U8;
}
if (err == -EINVAL && fmt == SND_PCM_FORMAT_DSD_U8) {
/* attempt to switch to DSD_U16 */
fmt = IsLittleEndian()
? SND_PCM_FORMAT_DSD_U16_LE
: SND_PCM_FORMAT_DSD_U16_BE;
err = AlsaTryFormatOrByteSwap(pcm, hwparams, fmt, params);
if (err == 0)
params.dsd_u16 = true;
else
fmt = SND_PCM_FORMAT_DSD_U8;
}
#endif
return err;
}
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static int
AlsaTryFormat(snd_pcm_t *pcm, snd_pcm_hw_params_t *hwparams,
SampleFormat sample_format,
PcmExport::Params &params)
{
snd_pcm_format_t alsa_format = ToAlsaPcmFormat(sample_format);
if (alsa_format == SND_PCM_FORMAT_UNKNOWN)
return -EINVAL;
return AlsaTryFormatDsd(pcm, hwparams, alsa_format, params);
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}
/**
* Configure a sample format, and probe other formats if that fails.
*/
static int
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AlsaSetupFormat(snd_pcm_t *pcm, snd_pcm_hw_params_t *hwparams,
AudioFormat &audio_format,
PcmExport::Params &params)
{
/* try the input format first */
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int err = AlsaTryFormat(pcm, hwparams, audio_format.format, params);
/* if unsupported by the hardware, try other formats */
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static constexpr SampleFormat probe_formats[] = {
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SampleFormat::S24_P32,
SampleFormat::S32,
SampleFormat::S16,
SampleFormat::S8,
SampleFormat::UNDEFINED,
};
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for (unsigned i = 0;
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err == -EINVAL && probe_formats[i] != SampleFormat::UNDEFINED;
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++i) {
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const SampleFormat mpd_format = probe_formats[i];
if (mpd_format == audio_format.format)
continue;
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err = AlsaTryFormat(pcm, hwparams, mpd_format, params);
if (err == 0)
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audio_format.format = mpd_format;
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}
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return err;
}
/**
* Wrapper for snd_pcm_hw_params().
*
* @param buffer_time the configured buffer time, or 0 if not configured
* @param period_time the configured period time, or 0 if not configured
* @param audio_format an #AudioFormat to be configured (or modified)
* by this function
* @param params to be modified by this function
*/
static void
AlsaSetupHw(snd_pcm_t *pcm, snd_pcm_hw_params_t *hwparams,
unsigned buffer_time, unsigned period_time,
AudioFormat &audio_format, PcmExport::Params &params)
{
int err;
unsigned retry = MPD_ALSA_RETRY_NR;
unsigned int period_time_ro = period_time;
configure_hw:
/* configure HW params */
err = snd_pcm_hw_params_any(pcm, hwparams);
if (err < 0)
throw FormatRuntimeError("snd_pcm_hw_params_any() failed: %s",
snd_strerror(-err));
err = snd_pcm_hw_params_set_access(pcm, hwparams,
SND_PCM_ACCESS_RW_INTERLEAVED);
if (err < 0)
throw FormatRuntimeError("snd_pcm_hw_params_set_access() failed: %s",
snd_strerror(-err));
err = AlsaSetupFormat(pcm, hwparams, audio_format, params);
if (err < 0)
throw FormatRuntimeError("Failed to configure format %s: %s",
sample_format_to_string(audio_format.format),
snd_strerror(-err));
unsigned int channels = audio_format.channels;
err = snd_pcm_hw_params_set_channels_near(pcm, hwparams,
&channels);
if (err < 0)
throw FormatRuntimeError("Failed to configure %i channels: %s",
(int)audio_format.channels,
snd_strerror(-err));
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audio_format.channels = (int8_t)channels;
const unsigned requested_sample_rate =
params.CalcOutputSampleRate(audio_format.sample_rate);
unsigned output_sample_rate = requested_sample_rate;
err = snd_pcm_hw_params_set_rate_near(pcm, hwparams,
&output_sample_rate, nullptr);
if (err < 0)
throw FormatRuntimeError("Failed to configure sample rate %u Hz: %s",
requested_sample_rate,
snd_strerror(-err));
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if (output_sample_rate == 0)
throw FormatRuntimeError("Failed to configure sample rate %u Hz",
audio_format.sample_rate);
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if (output_sample_rate != requested_sample_rate)
audio_format.sample_rate = params.CalcInputSampleRate(output_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);
FormatDebug(alsa_output_domain, "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);
FormatDebug(alsa_output_domain, "period: size=%u..%u time=%u..%u",
(unsigned)period_size_min, (unsigned)period_size_max,
period_time_min, period_time_max);
if (buffer_time > 0) {
err = snd_pcm_hw_params_set_buffer_time_near(pcm, hwparams,
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&buffer_time, nullptr);
if (err < 0)
throw FormatRuntimeError("snd_pcm_hw_params_set_buffer_time_near() failed: %s",
snd_strerror(-err));
} else {
err = snd_pcm_hw_params_get_buffer_time(hwparams, &buffer_time,
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nullptr);
if (err < 0)
buffer_time = 0;
}
if (period_time_ro == 0 && buffer_time >= 10000) {
period_time_ro = period_time = buffer_time / 4;
FormatDebug(alsa_output_domain,
"default period_time = buffer_time/4 = %u/4 = %u",
buffer_time, period_time);
}
if (period_time_ro > 0) {
period_time = period_time_ro;
err = snd_pcm_hw_params_set_period_time_near(pcm, hwparams,
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&period_time, nullptr);
if (err < 0)
throw FormatRuntimeError("snd_pcm_hw_params_set_period_time_near() failed: %s",
snd_strerror(-err));
}
err = snd_pcm_hw_params(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)
throw FormatRuntimeError("snd_pcm_hw_params() failed: %s",
snd_strerror(-err));
if (retry != MPD_ALSA_RETRY_NR)
FormatDebug(alsa_output_domain,
"ALSA period_time set to %d", period_time);
}
/**
* Wrapper for snd_pcm_sw_params().
*/
static void
AlsaSetupSw(snd_pcm_t *pcm, snd_pcm_uframes_t start_threshold,
snd_pcm_uframes_t avail_min)
{
snd_pcm_sw_params_t *swparams;
snd_pcm_sw_params_alloca(&swparams);
int err = snd_pcm_sw_params_current(pcm, swparams);
if (err < 0)
throw FormatRuntimeError("snd_pcm_sw_params_current() failed: %s",
snd_strerror(-err));
err = snd_pcm_sw_params_set_start_threshold(pcm, swparams,
start_threshold);
if (err < 0)
throw FormatRuntimeError("snd_pcm_sw_params_set_start_threshold() failed: %s",
snd_strerror(-err));
err = snd_pcm_sw_params_set_avail_min(pcm, swparams, avail_min);
if (err < 0)
throw FormatRuntimeError("snd_pcm_sw_params_set_avail_min() failed: %s",
snd_strerror(-err));
err = snd_pcm_sw_params(pcm, swparams);
if (err < 0)
throw FormatRuntimeError("snd_pcm_sw_params() failed: %s",
snd_strerror(-err));
}
inline void
AlsaOutput::Setup(AudioFormat &audio_format,
PcmExport::Params &params)
{
snd_pcm_hw_params_t *hwparams;
snd_pcm_hw_params_alloca(&hwparams);
AlsaSetupHw(pcm, hwparams,
buffer_time, period_time,
audio_format, params);
snd_pcm_format_t format;
if (snd_pcm_hw_params_get_format(hwparams, &format) == 0)
FormatDebug(alsa_output_domain,
"format=%s (%s)", snd_pcm_format_name(format),
snd_pcm_format_description(format));
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snd_pcm_uframes_t alsa_buffer_size;
int err = snd_pcm_hw_params_get_buffer_size(hwparams, &alsa_buffer_size);
if (err < 0)
throw FormatRuntimeError("snd_pcm_hw_params_get_buffer_size() failed: %s",
snd_strerror(-err));
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snd_pcm_uframes_t alsa_period_size;
err = snd_pcm_hw_params_get_period_size(hwparams, &alsa_period_size,
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nullptr);
if (err < 0)
throw FormatRuntimeError("snd_pcm_hw_params_get_period_size() failed: %s",
snd_strerror(-err));
AlsaSetupSw(pcm, alsa_buffer_size - alsa_period_size,
alsa_period_size);
FormatDebug(alsa_output_domain, "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;
period_frames = alsa_period_size;
period_position = 0;
silence = new uint8_t[snd_pcm_frames_to_bytes(pcm, alsa_period_size)];
snd_pcm_format_set_silence(format, silence,
alsa_period_size * audio_format.channels);
}
#ifdef ENABLE_DSD
inline void
AlsaOutput::SetupDop(const AudioFormat audio_format,
PcmExport::Params &params)
{
assert(dop);
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assert(audio_format.format == SampleFormat::DSD);
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/* pass 24 bit to AlsaSetup() */
AudioFormat dop_format = audio_format;
dop_format.format = SampleFormat::S24_P32;
const AudioFormat check = dop_format;
Setup(dop_format, params);
/* if the device allows only 32 bit, shift all DoP
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) */
params.shift8 = dop_format.format == SampleFormat::S32;
if (dop_format.format == SampleFormat::S32)
dop_format.format = SampleFormat::S24_P32;
if (dop_format != check) {
/* no bit-perfect playback, which is required
for DSD over USB */
delete[] silence;
throw std::runtime_error("Failed to configure DSD-over-PCM");
}
}
#endif
inline void
AlsaOutput::SetupOrDop(AudioFormat &audio_format, PcmExport::Params &params)
{
#ifdef ENABLE_DSD
std::exception_ptr dop_error;
if (dop && audio_format.format == SampleFormat::DSD) {
try {
params.dop = true;
SetupDop(audio_format, params);
return;
} catch (...) {
dop_error = std::current_exception();
params.dop = false;
}
}
try {
#endif
Setup(audio_format, params);
#ifdef ENABLE_DSD
} catch (...) {
if (dop_error)
/* if DoP was attempted, prefer returning the
original DoP error instead of the fallback
error */
std::rethrow_exception(dop_error);
else
throw;
}
#endif
}
inline void
AlsaOutput::Open(AudioFormat &audio_format)
{
int err = snd_pcm_open(&pcm, GetDevice(),
SND_PCM_STREAM_PLAYBACK, mode);
if (err < 0)
throw FormatRuntimeError("Failed to open ALSA device \"%s\": %s",
GetDevice(), snd_strerror(err));
FormatDebug(alsa_output_domain, "opened %s type=%s",
snd_pcm_name(pcm),
snd_pcm_type_name(snd_pcm_type(pcm)));
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PcmExport::Params params;
params.alsa_channel_order = true;
try {
SetupOrDop(audio_format, params);
} catch (...) {
snd_pcm_close(pcm);
std::throw_with_nested(FormatRuntimeError("Error opening ALSA device \"%s\"",
GetDevice()));
}
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#ifdef ENABLE_DSD
if (params.dop)
FormatDebug(alsa_output_domain, "DoP (DSD over PCM) enabled");
#endif
pcm_export->Open(audio_format.format,
audio_format.channels,
params);
in_frame_size = audio_format.GetFrameSize();
out_frame_size = pcm_export->GetFrameSize(audio_format);
must_prepare = false;
}
inline int
AlsaOutput::Recover(int err)
{
if (err == -EPIPE) {
FormatDebug(alsa_output_domain,
"Underrun on ALSA device \"%s\"",
GetDevice());
} else if (err == -ESTRPIPE) {
FormatDebug(alsa_output_domain,
"ALSA device \"%s\" was suspended",
GetDevice());
}
switch (snd_pcm_state(pcm)) {
case SND_PCM_STATE_PAUSED:
err = snd_pcm_pause(pcm, /* disable */ 0);
break;
case SND_PCM_STATE_SUSPENDED:
err = snd_pcm_resume(pcm);
if (err == -EAGAIN)
return 0;
/* fall-through to snd_pcm_prepare: */
#if GCC_CHECK_VERSION(7,0)
[[fallthrough]];
#endif
case SND_PCM_STATE_OPEN:
case SND_PCM_STATE_SETUP:
case SND_PCM_STATE_XRUN:
period_position = 0;
err = snd_pcm_prepare(pcm);
break;
case SND_PCM_STATE_DISCONNECTED:
break;
/* this is no error, so just keep running */
case SND_PCM_STATE_PREPARED:
case SND_PCM_STATE_RUNNING:
case SND_PCM_STATE_DRAINING:
err = 0;
break;
}
return err;
}
inline void
AlsaOutput::Drain()
{
if (snd_pcm_state(pcm) != SND_PCM_STATE_RUNNING)
return;
if (period_position > 0) {
/* generate some silence to finish the partial
period */
snd_pcm_uframes_t nframes =
period_frames - period_position;
WriteSilence(nframes);
}
snd_pcm_drain(pcm);
period_position = 0;
}
inline void
AlsaOutput::Cancel()
{
period_position = 0;
must_prepare = true;
snd_pcm_drop(pcm);
pcm_export->Reset();
}
inline void
AlsaOutput::Close()
{
snd_pcm_close(pcm);
delete[] silence;
}
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inline size_t
AlsaOutput::PlayRaw(ConstBuffer<void> data)
{
if (data.IsEmpty())
return 0;
assert(data.size % out_frame_size == 0);
const size_t n_frames = data.size / out_frame_size;
assert(n_frames > 0);
while (true) {
const auto frames_written = snd_pcm_writei(pcm, data.data,
n_frames);
if (frames_written > 0) {
period_position = (period_position + frames_written)
% period_frames;
return frames_written * out_frame_size;
}
if (frames_written < 0 && frames_written != -EAGAIN &&
frames_written != -EINTR &&
Recover(frames_written) < 0)
throw FormatRuntimeError("snd_pcm_writei() failed: %s",
snd_strerror(-frames_written));
}
}
inline size_t
AlsaOutput::Play(const void *chunk, size_t size)
{
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assert(size > 0);
assert(size % in_frame_size == 0);
if (must_prepare) {
must_prepare = false;
int err = snd_pcm_prepare(pcm);
if (err < 0)
throw FormatRuntimeError("snd_pcm_prepare() failed: %s",
snd_strerror(-err));
}
const auto e = pcm_export->Export({chunk, size});
if (e.size == 0)
/* the DoP (DSD over PCM) filter converts two frames
at a time and ignores the last odd frame; if there
was only one frame (e.g. the last frame in the
file), the result is empty; to avoid an endless
loop, bail out here, and pretend the one frame has
been played */
return size;
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const size_t bytes_written = PlayRaw(e);
return pcm_export->CalcSourceSize(bytes_written);
}
typedef AudioOutputWrapper<AlsaOutput> Wrapper;
const struct AudioOutputPlugin alsa_output_plugin = {
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"alsa",
alsa_test_default_device,
&Wrapper::Init,
&Wrapper::Finish,
&Wrapper::Enable,
&Wrapper::Disable,
&Wrapper::Open,
&Wrapper::Close,
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nullptr,
nullptr,
&Wrapper::Play,
&Wrapper::Drain,
&Wrapper::Cancel,
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nullptr,
&alsa_mixer_plugin,
};