Files
mpd/src/input/plugins/AlsaInputPlugin.cxx
borine 2fb34697c7 input/plugins/Alsa: catch all exceptions
snd_pcm_poll_descriptors_revents() may return any error code; the
ALSA docs do not constrain the permitted values. A 'hw' device
will only ever return an error if the pfd array passed in is
invalid (-EINVAL), but other I/O plugins may return arbitary
errors. For example a network-based device may return -EPIPE etc.
The resulting exception thrown by
AlsaNonBlockPcm::DispatchSockets() must be caught to prevent the
mpd process from being aborted.
2023-12-20 13:27:25 +01:00

489 lines
13 KiB
C++

/*
* Copyright 2003-2021 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.
*/
/*
* ALSA code based on an example by Paul Davis released under GPL here:
* http://equalarea.com/paul/alsa-audio.html
* and one by Matthias Nagorni, also GPL, here:
* http://alsamodular.sourceforge.net/alsa_programming_howto.html
*/
#include "AlsaInputPlugin.hxx"
#include "lib/alsa/NonBlock.hxx"
#include "lib/alsa/Error.hxx"
#include "lib/alsa/Format.hxx"
#include "../AsyncInputStream.hxx"
#include "event/Call.hxx"
#include "config/Block.hxx"
#include "util/Domain.hxx"
#include "util/ASCII.hxx"
#include "util/DivideString.hxx"
#include "pcm/AudioParser.hxx"
#include "pcm/AudioFormat.hxx"
#include "Log.hxx"
#include "event/MultiSocketMonitor.hxx"
#include "event/InjectEvent.hxx"
#include <alsa/asoundlib.h>
#include <cassert>
#include <string.h>
static constexpr Domain alsa_input_domain("alsa");
static constexpr auto ALSA_URI_PREFIX = "alsa://";
static constexpr auto BUILTIN_DEFAULT_DEVICE = "default";
static constexpr auto BUILTIN_DEFAULT_FORMAT = "48000:16:2";
static constexpr auto DEFAULT_BUFFER_TIME = std::chrono::milliseconds(1000);
static constexpr auto DEFAULT_RESUME_TIME = DEFAULT_BUFFER_TIME / 2;
static struct {
EventLoop *event_loop;
const char *default_device;
const char *default_format;
int mode;
} global_config;
class AlsaInputStream final
: public AsyncInputStream,
MultiSocketMonitor {
/**
* The configured name of the ALSA device.
*/
const std::string device;
snd_pcm_t *capture_handle;
const size_t frame_size;
AlsaNonBlockPcm non_block;
InjectEvent defer_invalidate_sockets;
public:
class SourceSpec;
AlsaInputStream(EventLoop &_loop,
Mutex &_mutex,
const SourceSpec &spec);
~AlsaInputStream() override {
BlockingCall(MultiSocketMonitor::GetEventLoop(), [this](){
MultiSocketMonitor::Reset();
defer_invalidate_sockets.Cancel();
});
snd_pcm_close(capture_handle);
}
AlsaInputStream(const AlsaInputStream &) = delete;
AlsaInputStream &operator=(const AlsaInputStream &) = delete;
static InputStreamPtr Create(EventLoop &event_loop, const char *uri,
Mutex &mutex);
protected:
/* virtual methods from AsyncInputStream */
void DoResume() override {
snd_pcm_resume(capture_handle);
InvalidateSockets();
}
void DoSeek([[maybe_unused]] offset_type new_offset) override {
/* unreachable because seekable==false */
SeekDone();
}
private:
void OpenDevice(const SourceSpec &spec);
void ConfigureCapture(AudioFormat audio_format);
void Pause() {
AsyncInputStream::Pause();
InvalidateSockets();
}
int Recover(int err);
/* virtual methods from class MultiSocketMonitor */
Event::Duration PrepareSockets() noexcept override;
void DispatchSockets() noexcept override;
};
class AlsaInputStream::SourceSpec {
const char *uri;
const char *device_name;
const char *format_string;
AudioFormat audio_format;
DivideString components;
public:
explicit SourceSpec(const char *_uri)
: uri(_uri)
, components(uri, '?')
{
if (components.IsDefined()) {
device_name = StringAfterPrefixCaseASCII(components.GetFirst(),
ALSA_URI_PREFIX);
format_string = StringAfterPrefixCaseASCII(components.GetSecond(),
"format=");
}
else {
device_name = StringAfterPrefixCaseASCII(uri, ALSA_URI_PREFIX);
format_string = global_config.default_format;
}
if (IsValidScheme()) {
if (*device_name == 0)
device_name = global_config.default_device;
if (format_string != nullptr)
audio_format = ParseAudioFormat(format_string, false);
}
}
[[nodiscard]] bool IsValidScheme() const noexcept {
return device_name != nullptr;
}
[[nodiscard]] bool IsValid() const noexcept {
return (device_name != nullptr) && (format_string != nullptr);
}
[[nodiscard]] const char *GetURI() const noexcept {
return uri;
}
[[nodiscard]] const char *GetDeviceName() const noexcept {
return device_name;
}
[[nodiscard]] const char *GetFormatString() const noexcept {
return format_string;
}
[[nodiscard]] AudioFormat GetAudioFormat() const noexcept {
return audio_format;
}
};
AlsaInputStream::AlsaInputStream(EventLoop &_loop,
Mutex &_mutex,
const SourceSpec &spec)
:AsyncInputStream(_loop, spec.GetURI(), _mutex,
spec.GetAudioFormat().TimeToSize(DEFAULT_BUFFER_TIME),
spec.GetAudioFormat().TimeToSize(DEFAULT_RESUME_TIME)),
MultiSocketMonitor(_loop),
device(spec.GetDeviceName()),
frame_size(spec.GetAudioFormat().GetFrameSize()),
defer_invalidate_sockets(_loop,
BIND_THIS_METHOD(InvalidateSockets))
{
OpenDevice(spec);
std::string mimestr = "audio/x-mpd-alsa-pcm;format=";
mimestr += spec.GetFormatString();
SetMimeType(mimestr.c_str());
InputStream::SetReady();
snd_pcm_start(capture_handle);
defer_invalidate_sockets.Schedule();
}
inline InputStreamPtr
AlsaInputStream::Create(EventLoop &event_loop, const char *uri,
Mutex &mutex)
{
assert(uri != nullptr);
AlsaInputStream::SourceSpec spec(uri);
if (!spec.IsValidScheme())
return nullptr;
return std::make_unique<AlsaInputStream>(event_loop, mutex, spec);
}
Event::Duration
AlsaInputStream::PrepareSockets() noexcept
{
if (IsPaused()) {
ClearSocketList();
return Event::Duration(-1);
}
return non_block.PrepareSockets(*this, capture_handle);
}
void
AlsaInputStream::DispatchSockets() noexcept
try {
non_block.DispatchSockets(*this, capture_handle);
const std::scoped_lock<Mutex> protect(mutex);
auto w = PrepareWriteBuffer();
const snd_pcm_uframes_t w_frames = w.size / frame_size;
if (w_frames == 0) {
/* buffer is full */
Pause();
return;
}
snd_pcm_sframes_t n_frames;
while ((n_frames = snd_pcm_readi(capture_handle,
w.data, w_frames)) < 0) {
if (n_frames == -EAGAIN)
return;
if (Recover(n_frames) < 0)
throw std::runtime_error("PCM error - stream aborted");
}
size_t nbytes = n_frames * frame_size;
CommitWriteBuffer(nbytes);
}
catch (...) {
postponed_exception = std::current_exception();
InvokeOnAvailable();
}
inline int
AlsaInputStream::Recover(int err)
{
switch(err) {
case -EPIPE:
FmtDebug(alsa_input_domain,
"Overrun on ALSA capture device \"{}\"",
device);
break;
case -ESTRPIPE:
FmtDebug(alsa_input_domain,
"ALSA capture device \"{}\" was suspended",
device);
break;
}
switch (snd_pcm_state(capture_handle)) {
case SND_PCM_STATE_PAUSED:
err = snd_pcm_pause(capture_handle, /* disable */ 0);
break;
case SND_PCM_STATE_SUSPENDED:
err = snd_pcm_resume(capture_handle);
if (err == -EAGAIN)
return 0;
/* fall-through to snd_pcm_prepare: */
#if CLANG_OR_GCC_VERSION(7,0)
[[fallthrough]];
#endif
case SND_PCM_STATE_OPEN:
case SND_PCM_STATE_SETUP:
case SND_PCM_STATE_XRUN:
err = snd_pcm_prepare(capture_handle);
if (err == 0)
err = snd_pcm_start(capture_handle);
break;
case SND_PCM_STATE_DISCONNECTED:
break;
case SND_PCM_STATE_PREPARED:
case SND_PCM_STATE_RUNNING:
case SND_PCM_STATE_DRAINING:
/* this is no error, so just keep running */
err = 0;
break;
default:
/* this default case is just here to work around
-Wswitch due to SND_PCM_STATE_PRIVATE1 (libasound
1.1.6) */
break;
}
return err;
}
void
AlsaInputStream::ConfigureCapture(AudioFormat audio_format)
{
int err;
snd_pcm_hw_params_t *hw_params;
snd_pcm_hw_params_alloca(&hw_params);
if ((err = snd_pcm_hw_params_any(capture_handle, hw_params)) < 0)
throw Alsa::MakeError(err, "snd_pcm_hw_params_any() failed");
if ((err = snd_pcm_hw_params_set_access(capture_handle, hw_params,
SND_PCM_ACCESS_RW_INTERLEAVED)) < 0)
throw Alsa::MakeError(err, "snd_pcm_hw_params_set_access() failed");
if ((err = snd_pcm_hw_params_set_format(capture_handle, hw_params,
ToAlsaPcmFormat(audio_format.format))) < 0)
throw Alsa::MakeError(err, "Cannot set sample format");
if ((err = snd_pcm_hw_params_set_channels(capture_handle,
hw_params, audio_format.channels)) < 0)
throw Alsa::MakeError(err, "Cannot set channels");
if ((err = snd_pcm_hw_params_set_rate(capture_handle,
hw_params, audio_format.sample_rate, 0)) < 0)
throw Alsa::MakeError(err, "Cannot set sample rate");
snd_pcm_uframes_t buffer_size_min, buffer_size_max;
snd_pcm_hw_params_get_buffer_size_min(hw_params, &buffer_size_min);
snd_pcm_hw_params_get_buffer_size_max(hw_params, &buffer_size_max);
unsigned buffer_time_min, buffer_time_max;
snd_pcm_hw_params_get_buffer_time_min(hw_params, &buffer_time_min, nullptr);
snd_pcm_hw_params_get_buffer_time_max(hw_params, &buffer_time_max, nullptr);
FmtDebug(alsa_input_domain, "buffer: size={}..{} time={}..{}",
buffer_size_min, 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(hw_params, &period_size_min, nullptr);
snd_pcm_hw_params_get_period_size_max(hw_params, &period_size_max, nullptr);
unsigned period_time_min, period_time_max;
snd_pcm_hw_params_get_period_time_min(hw_params, &period_time_min, nullptr);
snd_pcm_hw_params_get_period_time_max(hw_params, &period_time_max, nullptr);
FmtDebug(alsa_input_domain, "period: size={}..{} time={}..{}",
period_size_min, period_size_max,
period_time_min, period_time_max);
/* choose the maximum buffer_time up to limit of 2 seconds ... */
unsigned buffer_time = buffer_time_max;
if (buffer_time > 2000000U)
buffer_time = 2000000U;
int direction = -1;
if ((err = snd_pcm_hw_params_set_buffer_time_near(capture_handle,
hw_params, &buffer_time, &direction)) < 0)
throw Alsa::MakeError(err, "Cannot set buffer time");
/* ... and calculate the period_size to have four periods in
one buffer; this way, we get woken up often enough to avoid
buffer overruns, but not too often */
snd_pcm_uframes_t buffer_size;
if (snd_pcm_hw_params_get_buffer_size(hw_params, &buffer_size) == 0) {
snd_pcm_uframes_t period_size = buffer_size / 4;
direction = -1;
if ((err = snd_pcm_hw_params_set_period_size_near(capture_handle,
hw_params, &period_size, &direction)) < 0)
throw Alsa::MakeError(err, "Cannot set period size");
}
if ((err = snd_pcm_hw_params(capture_handle, hw_params)) < 0)
throw Alsa::MakeError(err, "snd_pcm_hw_params() failed");
snd_pcm_uframes_t alsa_buffer_size;
err = snd_pcm_hw_params_get_buffer_size(hw_params, &alsa_buffer_size);
if (err < 0)
throw Alsa::MakeError(err, "snd_pcm_hw_params_get_buffer_size() failed");
snd_pcm_uframes_t alsa_period_size;
err = snd_pcm_hw_params_get_period_size(hw_params, &alsa_period_size,
nullptr);
if (err < 0)
throw Alsa::MakeError(err, "snd_pcm_hw_params_get_period_size() failed");
FmtDebug(alsa_input_domain, "buffer_size={} period_size={}",
alsa_buffer_size, alsa_period_size);
snd_pcm_sw_params_t *sw_params;
snd_pcm_sw_params_alloca(&sw_params);
snd_pcm_sw_params_current(capture_handle, sw_params);
if ((err = snd_pcm_sw_params(capture_handle, sw_params)) < 0)
throw Alsa::MakeError(err, "snd_pcm_sw_params() failed");
}
inline void
AlsaInputStream::OpenDevice(const SourceSpec &spec)
{
int err;
if ((err = snd_pcm_open(&capture_handle, spec.GetDeviceName(),
SND_PCM_STREAM_CAPTURE,
SND_PCM_NONBLOCK | global_config.mode)) < 0)
throw Alsa::MakeError(err,
fmt::format("Failed to open device {}",
spec.GetDeviceName()).c_str());
try {
ConfigureCapture(spec.GetAudioFormat());
} catch (...) {
snd_pcm_close(capture_handle);
throw;
}
snd_pcm_prepare(capture_handle);
}
/*######################### Plugin Functions ##############################*/
static void
alsa_input_init(EventLoop &event_loop, const ConfigBlock &block)
{
global_config.event_loop = &event_loop;
global_config.default_device = block.GetBlockValue("default_device", BUILTIN_DEFAULT_DEVICE);
global_config.default_format = block.GetBlockValue("default_format", BUILTIN_DEFAULT_FORMAT);
global_config.mode = 0;
#ifdef SND_PCM_NO_AUTO_RESAMPLE
if (!block.GetBlockValue("auto_resample", true))
global_config.mode |= SND_PCM_NO_AUTO_RESAMPLE;
#endif
#ifdef SND_PCM_NO_AUTO_CHANNELS
if (!block.GetBlockValue("auto_channels", true))
global_config.mode |= SND_PCM_NO_AUTO_CHANNELS;
#endif
#ifdef SND_PCM_NO_AUTO_FORMAT
if (!block.GetBlockValue("auto_format", true))
global_config.mode |= SND_PCM_NO_AUTO_FORMAT;
#endif
}
static InputStreamPtr
alsa_input_open(const char *uri, Mutex &mutex)
{
return AlsaInputStream::Create(*global_config.event_loop, uri,
mutex);
}
static constexpr const char *alsa_prefixes[] = {
ALSA_URI_PREFIX,
nullptr
};
const struct InputPlugin input_plugin_alsa = {
"alsa",
alsa_prefixes,
alsa_input_init,
nullptr,
alsa_input_open,
nullptr
};