/* * Copyright 2003-2019 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 "AlsaOutputPlugin.hxx" #include "lib/alsa/AllowedFormat.hxx" #include "lib/alsa/HwSetup.hxx" #include "lib/alsa/NonBlock.hxx" #include "lib/alsa/PeriodBuffer.hxx" #include "lib/alsa/Version.hxx" #include "../OutputAPI.hxx" #include "mixer/MixerList.hxx" #include "pcm/Export.hxx" #include "thread/Mutex.hxx" #include "thread/Cond.hxx" #include "util/Manual.hxx" #include "util/RuntimeError.hxx" #include "util/Domain.hxx" #include "util/ConstBuffer.hxx" #include "util/ScopeExit.hxx" #include "util/StringView.hxx" #include "event/MultiSocketMonitor.hxx" #include "event/DeferEvent.hxx" #include "event/Call.hxx" #include "Log.hxx" #include #include #include #include static const char default_device[] = "default"; static constexpr unsigned MPD_ALSA_BUFFER_TIME_US = 500000; class AlsaOutput final : AudioOutput, MultiSocketMonitor { DeferEvent defer_invalidate_sockets; Manual pcm_export; /** * The configured name of the ALSA device; empty for the * default device */ const std::string device; #ifdef ENABLE_DSD /** * Enable DSD over PCM according to the DoP standard? * * @see http://dsd-guide.com/dop-open-standard */ bool dop_setting; #endif /** libasound's buffer_time setting (in microseconds) */ const unsigned buffer_time; /** libasound's period_time setting (in microseconds) */ const unsigned period_time; /** the mode flags passed to snd_pcm_open */ int mode = 0; std::forward_list allowed_formats; /** * Protects #dop_setting and #allowed_formats. */ mutable Mutex attributes_mutex; /** the libasound PCM device handle */ snd_pcm_t *pcm; #ifndef NDEBUG /** * The size of one audio frame passed to method play(). */ size_t in_frame_size; #endif /** * 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; /** * If snd_pcm_avail() goes above this value and no more data * is available in the #ring_buffer, we need to play some * silence. */ snd_pcm_sframes_t max_avail_frames; /** * Is this a buggy alsa-lib version, which needs a workaround * for the snd_pcm_drain() bug always returning -EAGAIN? See * alsa-lib commits fdc898d41135 and e4377b16454f for details. * This bug was fixed in alsa-lib version 1.1.4. * * The workaround is to re-enable blocking mode for the * snd_pcm_drain() call. */ bool work_around_drain_bug; /** * After Open(), has this output been activated by a Play() * command? * * Protected by #mutex. */ bool active; /** * 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; /** * Has snd_pcm_writei() been called successfully at least once * since the PCM was prepared? * * This is necessary to work around a kernel bug which causes * snd_pcm_drain() to return -EAGAIN forever in non-blocking * mode if snd_pcm_writei() was never called. */ bool written; bool drain; /** * This buffer gets allocated after opening the ALSA device. * It contains silence samples, enough to fill one period (see * #period_frames). */ uint8_t *silence; AlsaNonBlockPcm non_block; /** * For copying data from OutputThread to IOThread. */ boost::lockfree::spsc_queue *ring_buffer; Alsa::PeriodBuffer period_buffer; /** * Protects #cond, #error, #active, #drain. */ mutable Mutex mutex; /** * Used to wait when #ring_buffer is full. It will be * signalled each time data is popped from the #ring_buffer, * making space for more data. */ Cond cond; std::exception_ptr error; public: AlsaOutput(EventLoop &loop, const ConfigBlock &block); ~AlsaOutput() noexcept { /* free libasound's config cache */ snd_config_update_free_global(); } using MultiSocketMonitor::GetEventLoop; gcc_pure const char *GetDevice() const noexcept { return device.empty() ? default_device : device.c_str(); } static AudioOutput *Create(EventLoop &event_loop, const ConfigBlock &block) { return new AlsaOutput(event_loop, block); } private: const std::map GetAttributes() const noexcept override; void SetAttribute(std::string &&name, std::string &&value) override; void Enable() override; void Disable() noexcept override; void Open(AudioFormat &audio_format) override; void Close() noexcept override; size_t Play(const void *chunk, size_t size) override; void Drain() override; void Cancel() noexcept override; /** * 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 ¶ms); #ifdef ENABLE_DSD void SetupDop(AudioFormat audio_format, PcmExport::Params ¶ms); #endif void SetupOrDop(AudioFormat &audio_format, PcmExport::Params ¶ms #ifdef ENABLE_DSD , bool dop #endif ); gcc_pure bool LockIsActive() const noexcept { const std::lock_guard lock(mutex); return active; } /** * Activate the output by registering the sockets in the * #EventLoop. Before calling this, filling the ring buffer * has no effect; nothing will be played, and no code will be * run on #EventLoop's thread. * * Caller must hold the mutex. * * @return true if Activate() was called, false if the mutex * was never unlocked */ bool Activate() noexcept { if (active) return false; active = true; const ScopeUnlock unlock(mutex); defer_invalidate_sockets.Schedule(); return true; } int Recover(int err) noexcept; /** * Drain all buffers. To be run in #EventLoop's thread. * * Throws on error. * * @return true if draining is complete, false if this method * needs to be called again later */ bool DrainInternal(); /** * Stop playback immediately, dropping all buffers. To be run * in #EventLoop's thread. */ void CancelInternal() noexcept; /** * @return false if no data was moved */ bool CopyRingToPeriodBuffer() noexcept { if (period_buffer.IsFull()) return false; size_t nbytes = ring_buffer->pop(period_buffer.GetTail(), period_buffer.GetSpaceBytes()); if (nbytes == 0) return false; period_buffer.AppendBytes(nbytes); const std::lock_guard lock(mutex); /* notify the OutputThread that there is now room in ring_buffer */ cond.notify_one(); return true; } snd_pcm_sframes_t WriteFromPeriodBuffer() noexcept { assert(!period_buffer.IsEmpty()); auto frames_written = snd_pcm_writei(pcm, period_buffer.GetHead(), period_buffer.GetFrames(out_frame_size)); if (frames_written > 0) { written = true; period_buffer.ConsumeFrames(frames_written, out_frame_size); } return frames_written; } void LockCaughtError() noexcept { period_buffer.Clear(); const std::lock_guard lock(mutex); error = std::current_exception(); active = false; cond.notify_one(); } /* virtual methods from class MultiSocketMonitor */ std::chrono::steady_clock::duration PrepareSockets() noexcept override; void DispatchSockets() noexcept override; }; static constexpr Domain alsa_output_domain("alsa_output"); AlsaOutput::AlsaOutput(EventLoop &_loop, const ConfigBlock &block) :AudioOutput(FLAG_ENABLE_DISABLE), MultiSocketMonitor(_loop), defer_invalidate_sockets(_loop, BIND_THIS_METHOD(InvalidateSockets)), device(block.GetBlockValue("device", "")), #ifdef ENABLE_DSD dop_setting(block.GetBlockValue("dop", false) || /* legacy name from MPD 0.18 and older: */ block.GetBlockValue("dsd_usb", false)), #endif buffer_time(block.GetPositiveValue("buffer_time", MPD_ALSA_BUFFER_TIME_US)), period_time(block.GetPositiveValue("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 const char *allowed_formats_string = block.GetBlockValue("allowed_formats", nullptr); if (allowed_formats_string != nullptr) allowed_formats = Alsa::AllowedFormat::ParseList(allowed_formats_string); } const std::map AlsaOutput::GetAttributes() const noexcept { const std::lock_guard lock(attributes_mutex); return { std::make_pair("allowed_formats", Alsa::ToString(allowed_formats)), #ifdef ENABLE_DSD std::make_pair("dop", dop_setting ? "1" : "0"), #endif }; } void AlsaOutput::SetAttribute(std::string &&name, std::string &&value) { if (name == "allowed_formats") { const std::lock_guard lock(attributes_mutex); allowed_formats = Alsa::AllowedFormat::ParseList({value.data(), value.length()}); #ifdef ENABLE_DSD } else if (name == "dop") { const std::lock_guard lock(attributes_mutex); if (value == "0") dop_setting = false; else if (value == "1") dop_setting = true; else throw std::invalid_argument("Bad 'dop' value"); #endif } else AudioOutput::SetAttribute(std::move(name), std::move(value)); } void AlsaOutput::Enable() { pcm_export.Construct(); } void AlsaOutput::Disable() noexcept { pcm_export.Destruct(); } static bool alsa_test_default_device() { snd_pcm_t *handle; int ret = snd_pcm_open(&handle, default_device, 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; } /** * 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 ¶ms) { const auto hw_result = Alsa::SetupHw(pcm, buffer_time, period_time, audio_format, params); FormatDebug(alsa_output_domain, "format=%s (%s)", snd_pcm_format_name(hw_result.format), snd_pcm_format_description(hw_result.format)); FormatDebug(alsa_output_domain, "buffer_size=%u period_size=%u", (unsigned)hw_result.buffer_size, (unsigned)hw_result.period_size); AlsaSetupSw(pcm, hw_result.buffer_size - hw_result.period_size, hw_result.period_size); auto alsa_period_size = hw_result.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; /* generate silence if there's less than once period of data in the ALSA-PCM buffer */ max_avail_frames = hw_result.buffer_size - hw_result.period_size; silence = new uint8_t[snd_pcm_frames_to_bytes(pcm, alsa_period_size)]; snd_pcm_format_set_silence(hw_result.format, silence, alsa_period_size * audio_format.channels); } #ifdef ENABLE_DSD inline void AlsaOutput::SetupDop(const AudioFormat audio_format, PcmExport::Params ¶ms) { assert(audio_format.format == SampleFormat::DSD); /* 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 ¶ms #ifdef ENABLE_DSD , bool dop #endif ) { #ifdef ENABLE_DSD std::exception_ptr dop_error; if (dop && audio_format.format == SampleFormat::DSD) { try { params.dsd_mode = PcmExport::DsdMode::DOP; SetupDop(audio_format, params); return; } catch (...) { dop_error = std::current_exception(); params.dsd_mode = PcmExport::DsdMode::NONE; } } 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 } static constexpr bool MaybeDmix(snd_pcm_type_t type) { return type == SND_PCM_TYPE_DMIX || type == SND_PCM_TYPE_PLUG; } gcc_pure static bool MaybeDmix(snd_pcm_t *pcm) noexcept { return MaybeDmix(snd_pcm_type(pcm)); } static const Alsa::AllowedFormat & BestMatch(const std::forward_list &haystack, const AudioFormat &needle) { assert(!haystack.empty()); for (const auto &i : haystack) if (needle.MatchMask(i.format)) return i; return haystack.front(); } void AlsaOutput::Open(AudioFormat &audio_format) { #ifdef ENABLE_DSD bool dop; #endif { const std::lock_guard lock(attributes_mutex); #ifdef ENABLE_DSD dop = dop_setting; #endif if (!allowed_formats.empty()) { const auto &a = BestMatch(allowed_formats, audio_format); audio_format.ApplyMask(a.format); #ifdef ENABLE_DSD dop = a.dop; #endif } } 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))); PcmExport::Params params; params.alsa_channel_order = true; try { SetupOrDop(audio_format, params #ifdef ENABLE_DSD , dop #endif ); } catch (...) { snd_pcm_close(pcm); std::throw_with_nested(FormatRuntimeError("Error opening ALSA device \"%s\"", GetDevice())); } work_around_drain_bug = MaybeDmix(pcm) && GetRuntimeAlsaVersion() < MakeAlsaVersion(1, 1, 4); snd_pcm_nonblock(pcm, 1); #ifdef ENABLE_DSD if (params.dsd_mode == PcmExport::DsdMode::DOP) FormatDebug(alsa_output_domain, "DoP (DSD over PCM) enabled"); #endif pcm_export->Open(audio_format.format, audio_format.channels, params); #ifndef NDEBUG in_frame_size = audio_format.GetFrameSize(); #endif out_frame_size = pcm_export->GetFrameSize(); drain = false; size_t period_size = period_frames * out_frame_size; ring_buffer = new boost::lockfree::spsc_queue(period_size * 4); period_buffer.Allocate(period_frames, out_frame_size); active = false; must_prepare = false; written = false; error = {}; } inline int AlsaOutput::Recover(int err) noexcept { 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_buffer.Rewind(); written = false; 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; default: /* this default case is just here to work around -Wswitch due to SND_PCM_STATE_PRIVATE1 (libasound 1.1.6) */ break; } return err; } inline bool AlsaOutput::DrainInternal() { /* drain ring_buffer */ CopyRingToPeriodBuffer(); auto period_position = period_buffer.GetPeriodPosition(out_frame_size); if (period_position > 0) /* generate some silence to finish the partial period */ period_buffer.FillWithSilence(silence, out_frame_size); /* drain period_buffer */ if (!period_buffer.IsEmpty()) { auto frames_written = WriteFromPeriodBuffer(); if (frames_written < 0) { if (frames_written == -EAGAIN) return false; throw FormatRuntimeError("snd_pcm_writei() failed: %s", snd_strerror(-frames_written)); } /* need to call CopyRingToPeriodBuffer() and WriteFromPeriodBuffer() again in the next iteration, so don't finish the drain just yet */ return period_buffer.IsEmpty(); } if (!written) /* if nothing has ever been written to the PCM, we don't need to drain it */ return true; /* .. and finally drain the ALSA hardware buffer */ int result; if (work_around_drain_bug) { snd_pcm_nonblock(pcm, 0); result = snd_pcm_drain(pcm); snd_pcm_nonblock(pcm, 1); } else result = snd_pcm_drain(pcm); if (result == 0) return true; else if (result == -EAGAIN) return false; else throw FormatRuntimeError("snd_pcm_drain() failed: %s", snd_strerror(-result)); } void AlsaOutput::Drain() { std::unique_lock lock(mutex); if (error) std::rethrow_exception(error); drain = true; Activate(); cond.wait(lock, [this]{ return !drain || !active; }); if (error) std::rethrow_exception(error); } inline void AlsaOutput::CancelInternal() noexcept { /* this method doesn't need to lock the mutex because while it runs, the calling thread is blocked inside Cancel() */ must_prepare = true; snd_pcm_drop(pcm); pcm_export->Reset(); period_buffer.Clear(); ring_buffer->reset(); active = false; MultiSocketMonitor::Reset(); defer_invalidate_sockets.Cancel(); } void AlsaOutput::Cancel() noexcept { if (!LockIsActive()) { /* early cancel, quick code path without thread synchronization */ pcm_export->Reset(); assert(period_buffer.IsEmpty()); ring_buffer->reset(); return; } BlockingCall(GetEventLoop(), [this](){ CancelInternal(); }); } void AlsaOutput::Close() noexcept { /* make sure the I/O thread isn't inside DispatchSockets() */ BlockingCall(GetEventLoop(), [this](){ MultiSocketMonitor::Reset(); defer_invalidate_sockets.Cancel(); }); period_buffer.Free(); delete ring_buffer; snd_pcm_close(pcm); delete[] silence; } size_t AlsaOutput::Play(const void *chunk, size_t size) { assert(size > 0); assert(size % in_frame_size == 0); const auto e = pcm_export->Export({chunk, size}); if (e.empty()) return size; std::unique_lock lock(mutex); while (true) { if (error) std::rethrow_exception(error); size_t bytes_written = ring_buffer->push((const uint8_t *)e.data, e.size); if (bytes_written > 0) return pcm_export->CalcInputSize(bytes_written); /* now that the ring_buffer is full, we can activate the socket handlers to trigger the first snd_pcm_writei() */ if (Activate()) /* since everything may have changed while the mutex was unlocked, we need to skip the cond.wait() call below and check the new status */ continue; /* wait for the DispatchSockets() to make room in the ring_buffer */ cond.wait(lock); } } std::chrono::steady_clock::duration AlsaOutput::PrepareSockets() noexcept { if (!LockIsActive()) { ClearSocketList(); return std::chrono::steady_clock::duration(-1); } try { return non_block.PrepareSockets(*this, pcm); } catch (...) { ClearSocketList(); LockCaughtError(); return std::chrono::steady_clock::duration(-1); } } void AlsaOutput::DispatchSockets() noexcept try { non_block.DispatchSockets(*this, pcm); if (must_prepare) { must_prepare = false; written = false; int err = snd_pcm_prepare(pcm); if (err < 0) throw FormatRuntimeError("snd_pcm_prepare() failed: %s", snd_strerror(-err)); } { const std::lock_guard lock(mutex); assert(active); if (drain) { { ScopeUnlock unlock(mutex); if (!DrainInternal()) return; MultiSocketMonitor::InvalidateSockets(); } drain = false; cond.notify_one(); return; } } CopyRingToPeriodBuffer(); if (period_buffer.IsEmpty()) { if (snd_pcm_state(pcm) == SND_PCM_STATE_PREPARED || snd_pcm_avail(pcm) <= max_avail_frames) { /* at SND_PCM_STATE_PREPARED (not yet switched to SND_PCM_STATE_RUNNING), we have no pressure to fill the ALSA buffer, because no xrun can possibly occur; and if no data is available right now, we can easily wait until some is available; so we just stop monitoring the ALSA file descriptor, and let it be reactivated by Play()/Activate() whenever more data arrives */ /* the same applies when there is still enough data in the ALSA-PCM buffer (determined by snd_pcm_avail()); this can happend at the start of playback, when our ring_buffer is smaller than the ALSA-PCM buffer */ { const std::lock_guard lock(mutex); active = false; cond.notify_one(); } /* avoid race condition: see if data has arrived meanwhile before disabling the event (but after clearing the "active" flag) */ if (!CopyRingToPeriodBuffer()) { MultiSocketMonitor::Reset(); defer_invalidate_sockets.Cancel(); } return; } /* insert some silence if the buffer has not enough data yet, to avoid ALSA xrun */ period_buffer.FillWithSilence(silence, out_frame_size); } auto frames_written = WriteFromPeriodBuffer(); if (frames_written < 0) { if (frames_written == -EAGAIN || frames_written == -EINTR) /* try again in the next DispatchSockets() call which is still scheduled */ return; if (Recover(frames_written) < 0) throw FormatRuntimeError("snd_pcm_writei() failed: %s", snd_strerror(-frames_written)); /* recovered; try again in the next DispatchSockets() call */ return; } } catch (...) { MultiSocketMonitor::Reset(); LockCaughtError(); } const struct AudioOutputPlugin alsa_output_plugin = { "alsa", alsa_test_default_device, &AlsaOutput::Create, &alsa_mixer_plugin, };