| // Copyright (c) 2012 The Chromium Authors. All rights reserved. |
| // Use of this source code is governed by a BSD-style license that can be |
| // found in the LICENSE file. |
| |
| #include "third_party/blink/renderer/modules/mediastream/track_audio_renderer.h" |
| |
| #include <utility> |
| |
| #include "base/location.h" |
| #include "base/logging.h" |
| #include "base/metrics/histogram_macros.h" |
| #include "base/synchronization/lock.h" |
| #include "base/threading/thread_task_runner_handle.h" |
| #include "base/trace_event/trace_event.h" |
| #include "media/audio/audio_sink_parameters.h" |
| #include "media/base/audio_bus.h" |
| #include "media/base/audio_latency.h" |
| #include "media/base/audio_shifter.h" |
| #include "third_party/blink/public/platform/modules/mediastream/web_media_stream_track.h" |
| #include "third_party/blink/public/platform/platform.h" |
| #include "third_party/blink/public/web/web_local_frame.h" |
| #include "third_party/blink/renderer/core/frame/local_frame.h" |
| #include "third_party/blink/renderer/platform/mediastream/media_stream_audio_track.h" |
| #include "third_party/blink/renderer/platform/scheduler/public/post_cross_thread_task.h" |
| #include "third_party/blink/renderer/platform/wtf/cross_thread_functional.h" |
| |
| namespace WTF { |
| |
| template <> |
| struct CrossThreadCopier<media::AudioParameters> { |
| STATIC_ONLY(CrossThreadCopier); |
| using Type = media::AudioParameters; |
| static Type Copy(Type pointer) { return pointer; } |
| }; |
| |
| } // namespace WTF |
| |
| namespace blink { |
| |
| namespace { |
| |
| enum LocalRendererSinkStates { |
| kSinkStarted = 0, |
| kSinkNeverStarted, |
| kSinkStatesMax // Must always be last! |
| }; |
| |
| // Translates |num_samples_rendered| into a TimeDelta duration and adds it to |
| // |prior_elapsed_render_time|. |
| base::TimeDelta ComputeTotalElapsedRenderTime( |
| base::TimeDelta prior_elapsed_render_time, |
| int64_t num_samples_rendered, |
| int sample_rate) { |
| return prior_elapsed_render_time + |
| base::TimeDelta::FromMicroseconds(num_samples_rendered * |
| base::Time::kMicrosecondsPerSecond / |
| sample_rate); |
| } |
| |
| WebLocalFrame* ToWebLocalFrame(LocalFrame* frame) { |
| if (!frame) |
| return nullptr; |
| |
| return static_cast<WebLocalFrame*>(WebFrame::FromCoreFrame(frame)); |
| } |
| |
| } // namespace |
| |
| // media::AudioRendererSink::RenderCallback implementation |
| int TrackAudioRenderer::Render(base::TimeDelta delay, |
| base::TimeTicks delay_timestamp, |
| int prior_frames_skipped, |
| media::AudioBus* audio_bus) { |
| TRACE_EVENT2("audio", "TrackAudioRenderer::Render", "delay (ms)", |
| delay.InMillisecondsF(), "delay_timestamp (ms)", |
| (delay_timestamp - base::TimeTicks()).InMillisecondsF()); |
| base::AutoLock auto_lock(thread_lock_); |
| |
| if (!audio_shifter_) { |
| audio_bus->Zero(); |
| return 0; |
| } |
| |
| // TODO(miu): Plumbing is needed to determine the actual playout timestamp |
| // of the audio, instead of just snapshotting TimeTicks::Now(), for proper |
| // audio/video sync. https://crbug.com/335335 |
| const base::TimeTicks playout_time = base::TimeTicks::Now() + delay; |
| DVLOG(2) << "Pulling audio out of shifter to be played " |
| << delay.InMilliseconds() << " ms from now."; |
| audio_shifter_->Pull(audio_bus, playout_time); |
| num_samples_rendered_ += audio_bus->frames(); |
| return audio_bus->frames(); |
| } |
| |
| void TrackAudioRenderer::OnRenderErrorCrossThread() { |
| DCHECK(task_runner_->BelongsToCurrentThread()); |
| |
| on_render_error_callback_.Run(); |
| } |
| |
| void TrackAudioRenderer::OnRenderError() { |
| DCHECK(on_render_error_callback_); |
| |
| PostCrossThreadTask( |
| *task_runner_, FROM_HERE, |
| CrossThreadBindOnce(&TrackAudioRenderer::OnRenderErrorCrossThread, |
| WrapRefCounted(this))); |
| } |
| |
| // WebMediaStreamAudioSink implementation |
| void TrackAudioRenderer::OnData(const media::AudioBus& audio_bus, |
| base::TimeTicks reference_time) { |
| DCHECK(!reference_time.is_null()); |
| |
| TRACE_EVENT1("audio", "TrackAudioRenderer::OnData", "reference time (ms)", |
| (reference_time - base::TimeTicks()).InMillisecondsF()); |
| |
| base::AutoLock auto_lock(thread_lock_); |
| if (!audio_shifter_) |
| return; |
| |
| std::unique_ptr<media::AudioBus> audio_data( |
| media::AudioBus::Create(audio_bus.channels(), audio_bus.frames())); |
| audio_bus.CopyTo(audio_data.get()); |
| // Note: For remote audio sources, |reference_time| is the local playout time, |
| // the ideal point-in-time at which the first audio sample should be played |
| // out in the future. For local sources, |reference_time| is the |
| // point-in-time at which the first audio sample was captured in the past. In |
| // either case, AudioShifter will auto-detect and do the right thing when |
| // audio is pulled from it. |
| audio_shifter_->Push(std::move(audio_data), reference_time); |
| } |
| |
| void TrackAudioRenderer::OnSetFormat(const media::AudioParameters& params) { |
| DVLOG(1) << "TrackAudioRenderer::OnSetFormat: " |
| << params.AsHumanReadableString(); |
| // If the parameters changed, the audio in the AudioShifter is invalid and |
| // should be dropped. |
| { |
| base::AutoLock auto_lock(thread_lock_); |
| if (audio_shifter_ && |
| (audio_shifter_->sample_rate() != params.sample_rate() || |
| audio_shifter_->channels() != params.channels())) { |
| HaltAudioFlowWhileLockHeld(); |
| } |
| } |
| |
| // Post a task on the main render thread to reconfigure the |sink_| with the |
| // new format. |
| PostCrossThreadTask(*task_runner_, FROM_HERE, |
| CrossThreadBindOnce(&TrackAudioRenderer::ReconfigureSink, |
| WrapRefCounted(this), params)); |
| } |
| |
| TrackAudioRenderer::TrackAudioRenderer( |
| MediaStreamComponent* audio_component, |
| LocalFrame* playout_frame, |
| const base::UnguessableToken& session_id, |
| const String& device_id, |
| base::RepeatingCallback<void()> on_render_error_callback) |
| : audio_component_(audio_component), |
| playout_frame_(playout_frame), |
| session_id_(session_id), |
| task_runner_( |
| playout_frame->GetTaskRunner(blink::TaskType::kInternalMedia)), |
| num_samples_rendered_(0), |
| on_render_error_callback_(std::move(on_render_error_callback)), |
| playing_(false), |
| output_device_id_(device_id), |
| volume_(0.0), |
| sink_started_(false) { |
| DCHECK(MediaStreamAudioTrack::From(audio_component_.Get())); |
| DCHECK(task_runner_->BelongsToCurrentThread()); |
| DVLOG(1) << "TrackAudioRenderer::TrackAudioRenderer()"; |
| } |
| |
| TrackAudioRenderer::~TrackAudioRenderer() { |
| DCHECK(task_runner_->BelongsToCurrentThread()); |
| DCHECK(!sink_); |
| DVLOG(1) << "TrackAudioRenderer::~TrackAudioRenderer()"; |
| } |
| |
| void TrackAudioRenderer::Start() { |
| DVLOG(1) << "TrackAudioRenderer::Start()"; |
| DCHECK(task_runner_->BelongsToCurrentThread()); |
| DCHECK_EQ(playing_, false); |
| |
| // We get audio data from |audio_component_|... |
| WebMediaStreamAudioSink::AddToAudioTrack( |
| this, WebMediaStreamTrack(audio_component_.Get())); |
| // ...and |sink_| will get audio data from us. |
| DCHECK(!sink_); |
| sink_ = Platform::Current()->NewAudioRendererSink( |
| WebAudioDeviceSourceType::kNonRtcAudioTrack, |
| ToWebLocalFrame(playout_frame_), {session_id_, output_device_id_.Utf8()}); |
| |
| base::AutoLock auto_lock(thread_lock_); |
| prior_elapsed_render_time_ = base::TimeDelta(); |
| num_samples_rendered_ = 0; |
| } |
| |
| void TrackAudioRenderer::Stop() { |
| DVLOG(1) << "TrackAudioRenderer::Stop()"; |
| DCHECK(task_runner_->BelongsToCurrentThread()); |
| |
| Pause(); |
| |
| // Stop the output audio stream, i.e, stop asking for data to render. |
| // It is safer to call Stop() on the |sink_| to clean up the resources even |
| // when the |sink_| is never started. |
| if (sink_) { |
| sink_->Stop(); |
| sink_ = nullptr; |
| } |
| |
| if (!sink_started_ && IsLocalRenderer()) { |
| UMA_HISTOGRAM_ENUMERATION("Media.LocalRendererSinkStates", |
| kSinkNeverStarted, kSinkStatesMax); |
| } |
| sink_started_ = false; |
| |
| // Ensure that the capturer stops feeding us with captured audio. |
| WebMediaStreamAudioSink::RemoveFromAudioTrack( |
| this, WebMediaStreamTrack(audio_component_.Get())); |
| } |
| |
| void TrackAudioRenderer::Play() { |
| DVLOG(1) << "TrackAudioRenderer::Play()"; |
| DCHECK(task_runner_->BelongsToCurrentThread()); |
| |
| if (!sink_) |
| return; |
| |
| playing_ = true; |
| |
| MaybeStartSink(); |
| } |
| |
| void TrackAudioRenderer::Pause() { |
| DVLOG(1) << "TrackAudioRenderer::Pause()"; |
| DCHECK(task_runner_->BelongsToCurrentThread()); |
| |
| if (!sink_) |
| return; |
| |
| playing_ = false; |
| |
| base::AutoLock auto_lock(thread_lock_); |
| HaltAudioFlowWhileLockHeld(); |
| } |
| |
| void TrackAudioRenderer::SetVolume(float volume) { |
| DVLOG(1) << "TrackAudioRenderer::SetVolume(" << volume << ")"; |
| DCHECK(task_runner_->BelongsToCurrentThread()); |
| |
| // Cache the volume. Whenever |sink_| is re-created, call SetVolume() with |
| // this cached volume. |
| volume_ = volume; |
| if (sink_) |
| sink_->SetVolume(volume); |
| } |
| |
| base::TimeDelta TrackAudioRenderer::GetCurrentRenderTime() { |
| DCHECK(task_runner_->BelongsToCurrentThread()); |
| base::AutoLock auto_lock(thread_lock_); |
| if (source_params_.IsValid()) { |
| return ComputeTotalElapsedRenderTime(prior_elapsed_render_time_, |
| num_samples_rendered_, |
| source_params_.sample_rate()); |
| } |
| return prior_elapsed_render_time_; |
| } |
| |
| bool TrackAudioRenderer::IsLocalRenderer() { |
| DCHECK(task_runner_->BelongsToCurrentThread()); |
| return MediaStreamAudioTrack::From(audio_component_.Get())->is_local_track(); |
| } |
| |
| void TrackAudioRenderer::SwitchOutputDevice( |
| const std::string& device_id, |
| media::OutputDeviceStatusCB callback) { |
| DVLOG(1) << "TrackAudioRenderer::SwitchOutputDevice()"; |
| DCHECK(task_runner_->BelongsToCurrentThread()); |
| |
| { |
| base::AutoLock auto_lock(thread_lock_); |
| HaltAudioFlowWhileLockHeld(); |
| } |
| |
| scoped_refptr<media::AudioRendererSink> new_sink = |
| Platform::Current()->NewAudioRendererSink( |
| WebAudioDeviceSourceType::kNonRtcAudioTrack, |
| ToWebLocalFrame(playout_frame_), {session_id_, device_id}); |
| |
| media::OutputDeviceStatus new_sink_status = |
| new_sink->GetOutputDeviceInfo().device_status(); |
| UMA_HISTOGRAM_ENUMERATION("Media.Audio.TrackAudioRenderer.SwitchDeviceStatus", |
| new_sink_status, |
| media::OUTPUT_DEVICE_STATUS_MAX + 1); |
| if (new_sink_status != media::OUTPUT_DEVICE_STATUS_OK) { |
| new_sink->Stop(); |
| std::move(callback).Run(new_sink_status); |
| return; |
| } |
| |
| output_device_id_ = String(device_id.data(), device_id.size()); |
| bool was_sink_started = sink_started_; |
| |
| if (sink_) |
| sink_->Stop(); |
| |
| sink_started_ = false; |
| sink_ = new_sink; |
| if (was_sink_started) |
| MaybeStartSink(); |
| |
| std::move(callback).Run(media::OUTPUT_DEVICE_STATUS_OK); |
| } |
| |
| void TrackAudioRenderer::MaybeStartSink() { |
| DCHECK(task_runner_->BelongsToCurrentThread()); |
| DVLOG(1) << "TrackAudioRenderer::MaybeStartSink()"; |
| |
| if (!sink_ || !source_params_.IsValid() || !playing_) { |
| return; |
| } |
| |
| // Re-create the AudioShifter to drop old audio data and reset to a starting |
| // state. MaybeStartSink() is always called in a situation where either the |
| // source or sink has changed somehow and so all of AudioShifter's internal |
| // time-sync state is invalid. |
| CreateAudioShifter(); |
| |
| if (sink_started_) |
| return; |
| |
| const media::OutputDeviceInfo& device_info = sink_->GetOutputDeviceInfo(); |
| UMA_HISTOGRAM_ENUMERATION("Media.Audio.TrackAudioRenderer.DeviceStatus", |
| device_info.device_status(), |
| media::OUTPUT_DEVICE_STATUS_MAX + 1); |
| if (device_info.device_status() != media::OUTPUT_DEVICE_STATUS_OK) |
| return; |
| |
| // Output parameters consist of the same channel layout and sample rate as the |
| // source, but having the buffer duration preferred by the hardware. |
| const media::AudioParameters& hardware_params = device_info.output_params(); |
| media::AudioParameters sink_params( |
| hardware_params.format(), source_params_.channel_layout(), |
| source_params_.sample_rate(), |
| media::AudioLatency::GetRtcBufferSize( |
| source_params_.sample_rate(), hardware_params.frames_per_buffer())); |
| if (sink_params.channel_layout() == media::CHANNEL_LAYOUT_DISCRETE) { |
| DCHECK_LE(source_params_.channels(), 2); |
| sink_params.set_channels_for_discrete(source_params_.channels()); |
| } |
| DVLOG(1) << ("TrackAudioRenderer::MaybeStartSink() -- Starting sink. " |
| "source_params={") |
| << source_params_.AsHumanReadableString() << "}, hardware_params={" |
| << hardware_params.AsHumanReadableString() << "}, sink parameters={" |
| << sink_params.AsHumanReadableString() << '}'; |
| |
| // Specify the latency info to be passed to the browser side. |
| sink_params.set_latency_tag(Platform::Current()->GetAudioSourceLatencyType( |
| WebAudioDeviceSourceType::kNonRtcAudioTrack)); |
| |
| sink_->Initialize(sink_params, this); |
| sink_->Start(); |
| sink_->SetVolume(volume_); |
| sink_->Play(); // Not all the sinks play on start. |
| sink_started_ = true; |
| if (IsLocalRenderer()) { |
| UMA_HISTOGRAM_ENUMERATION("Media.LocalRendererSinkStates", kSinkStarted, |
| kSinkStatesMax); |
| } |
| } |
| |
| void TrackAudioRenderer::ReconfigureSink(const media::AudioParameters& params) { |
| DCHECK(task_runner_->BelongsToCurrentThread()); |
| |
| if (source_params_.Equals(params)) |
| return; |
| source_params_ = params; |
| |
| if (!sink_) |
| return; // TrackAudioRenderer has not yet been started. |
| |
| // Stop |sink_| and re-create a new one to be initialized with different audio |
| // parameters. Then, invoke MaybeStartSink() to restart everything again. |
| sink_->Stop(); |
| sink_started_ = false; |
| sink_ = Platform::Current()->NewAudioRendererSink( |
| WebAudioDeviceSourceType::kNonRtcAudioTrack, |
| ToWebLocalFrame(playout_frame_), {session_id_, output_device_id_.Utf8()}); |
| MaybeStartSink(); |
| } |
| |
| void TrackAudioRenderer::CreateAudioShifter() { |
| DCHECK(task_runner_->BelongsToCurrentThread()); |
| |
| // Note 1: The max buffer is fairly large to cover the case where |
| // remotely-sourced audio is delivered well ahead of its scheduled playout |
| // time (e.g., content streaming with a very large end-to-end |
| // latency). However, there is no penalty for making it large in the |
| // low-latency use cases since AudioShifter will discard data as soon as it is |
| // no longer needed. |
| // |
| // Note 2: The clock accuracy is set to 20ms because clock accuracy is |
| // ~15ms on Windows machines without a working high-resolution clock. See |
| // comments in base/time/time.h for details. |
| media::AudioShifter* const new_shifter = new media::AudioShifter( |
| base::TimeDelta::FromSeconds(5), base::TimeDelta::FromMilliseconds(20), |
| base::TimeDelta::FromSeconds(20), source_params_.sample_rate(), |
| source_params_.channels()); |
| |
| base::AutoLock auto_lock(thread_lock_); |
| audio_shifter_.reset(new_shifter); |
| } |
| |
| void TrackAudioRenderer::HaltAudioFlowWhileLockHeld() { |
| thread_lock_.AssertAcquired(); |
| |
| audio_shifter_.reset(); |
| |
| if (source_params_.IsValid()) { |
| prior_elapsed_render_time_ = ComputeTotalElapsedRenderTime( |
| prior_elapsed_render_time_, num_samples_rendered_, |
| source_params_.sample_rate()); |
| num_samples_rendered_ = 0; |
| } |
| } |
| |
| } // namespace blink |