| Index: content/renderer/media/webrtc/media_stream_remote_audio_track.cc
|
| diff --git a/content/renderer/media/webrtc/media_stream_remote_audio_track.cc b/content/renderer/media/webrtc/media_stream_remote_audio_track.cc
|
| deleted file mode 100644
|
| index e3940ab72b3bce7149897913d4af49745b673da0..0000000000000000000000000000000000000000
|
| --- a/content/renderer/media/webrtc/media_stream_remote_audio_track.cc
|
| +++ /dev/null
|
| @@ -1,237 +0,0 @@
|
| -// Copyright 2015 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 "content/renderer/media/webrtc/media_stream_remote_audio_track.h"
|
| -
|
| -#include <stddef.h>
|
| -
|
| -#include <list>
|
| -
|
| -#include "base/logging.h"
|
| -#include "content/public/renderer/media_stream_audio_sink.h"
|
| -#include "third_party/webrtc/api/mediastreaminterface.h"
|
| -
|
| -namespace content {
|
| -
|
| -class MediaStreamRemoteAudioSource::AudioSink
|
| - : public webrtc::AudioTrackSinkInterface {
|
| - public:
|
| - AudioSink() {
|
| - }
|
| - ~AudioSink() override {
|
| - DCHECK(sinks_.empty());
|
| - }
|
| -
|
| - void Add(MediaStreamAudioSink* sink, MediaStreamAudioTrack* track,
|
| - bool enabled) {
|
| - DCHECK(thread_checker_.CalledOnValidThread());
|
| - SinkInfo info(sink, track, enabled);
|
| - base::AutoLock lock(lock_);
|
| - sinks_.push_back(info);
|
| - }
|
| -
|
| - void Remove(MediaStreamAudioSink* sink, MediaStreamAudioTrack* track) {
|
| - DCHECK(thread_checker_.CalledOnValidThread());
|
| - base::AutoLock lock(lock_);
|
| - sinks_.remove_if([&sink, &track](const SinkInfo& info) {
|
| - return info.sink == sink && info.track == track;
|
| - });
|
| - }
|
| -
|
| - void SetEnabled(MediaStreamAudioTrack* track, bool enabled) {
|
| - DCHECK(thread_checker_.CalledOnValidThread());
|
| - base::AutoLock lock(lock_);
|
| - for (SinkInfo& info : sinks_) {
|
| - if (info.track == track)
|
| - info.enabled = enabled;
|
| - }
|
| - }
|
| -
|
| - void RemoveAll(MediaStreamAudioTrack* track) {
|
| - base::AutoLock lock(lock_);
|
| - sinks_.remove_if([&track](const SinkInfo& info) {
|
| - return info.track == track;
|
| - });
|
| - }
|
| -
|
| - bool IsNeeded() const {
|
| - DCHECK(thread_checker_.CalledOnValidThread());
|
| - return !sinks_.empty();
|
| - }
|
| -
|
| - private:
|
| - void OnData(const void* audio_data, int bits_per_sample, int sample_rate,
|
| - size_t number_of_channels, size_t number_of_frames) override {
|
| - if (!audio_bus_ ||
|
| - static_cast<size_t>(audio_bus_->channels()) != number_of_channels ||
|
| - static_cast<size_t>(audio_bus_->frames()) != number_of_frames) {
|
| - audio_bus_ = media::AudioBus::Create(number_of_channels,
|
| - number_of_frames);
|
| - }
|
| -
|
| - audio_bus_->FromInterleaved(audio_data, number_of_frames,
|
| - bits_per_sample / 8);
|
| -
|
| - bool format_changed = false;
|
| - if (params_.format() != media::AudioParameters::AUDIO_PCM_LOW_LATENCY ||
|
| - static_cast<size_t>(params_.channels()) != number_of_channels ||
|
| - params_.sample_rate() != sample_rate ||
|
| - static_cast<size_t>(params_.frames_per_buffer()) != number_of_frames) {
|
| - params_ = media::AudioParameters(
|
| - media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
|
| - media::GuessChannelLayout(number_of_channels),
|
| - sample_rate, 16, number_of_frames);
|
| - format_changed = true;
|
| - }
|
| -
|
| - // TODO(tommi): We should get the timestamp from WebRTC.
|
| - base::TimeTicks estimated_capture_time(base::TimeTicks::Now());
|
| -
|
| - base::AutoLock lock(lock_);
|
| - for (const SinkInfo& info : sinks_) {
|
| - if (info.enabled) {
|
| - if (format_changed)
|
| - info.sink->OnSetFormat(params_);
|
| - info.sink->OnData(*audio_bus_.get(), estimated_capture_time);
|
| - }
|
| - }
|
| - }
|
| -
|
| - mutable base::Lock lock_;
|
| - struct SinkInfo {
|
| - SinkInfo(MediaStreamAudioSink* sink, MediaStreamAudioTrack* track,
|
| - bool enabled) : sink(sink), track(track), enabled(enabled) {}
|
| - MediaStreamAudioSink* sink;
|
| - MediaStreamAudioTrack* track;
|
| - bool enabled;
|
| - };
|
| - std::list<SinkInfo> sinks_;
|
| - base::ThreadChecker thread_checker_;
|
| - media::AudioParameters params_; // Only used on the callback thread.
|
| - std::unique_ptr<media::AudioBus>
|
| - audio_bus_; // Only used on the callback thread.
|
| -};
|
| -
|
| -MediaStreamRemoteAudioTrack::MediaStreamRemoteAudioTrack(
|
| - const blink::WebMediaStreamSource& source, bool enabled)
|
| - : MediaStreamAudioTrack(false), source_(source), enabled_(enabled) {
|
| - DCHECK(source.getExtraData()); // Make sure the source has a native source.
|
| -}
|
| -
|
| -MediaStreamRemoteAudioTrack::~MediaStreamRemoteAudioTrack() {
|
| - DCHECK(main_render_thread_checker_.CalledOnValidThread());
|
| - // Ensure the track is stopped.
|
| - MediaStreamAudioTrack::Stop();
|
| -}
|
| -
|
| -void MediaStreamRemoteAudioTrack::SetEnabled(bool enabled) {
|
| - DCHECK(main_render_thread_checker_.CalledOnValidThread());
|
| -
|
| - // This affects the shared state of the source for whether or not it's a part
|
| - // of the mixed audio that's rendered for remote tracks from WebRTC.
|
| - // All tracks from the same source will share this state and thus can step
|
| - // on each other's toes.
|
| - // This is also why we can't check the |enabled_| state for equality with
|
| - // |enabled| before setting the mixing enabled state. |enabled_| and the
|
| - // shared state might not be the same.
|
| - source()->SetEnabledForMixing(enabled);
|
| -
|
| - enabled_ = enabled;
|
| - source()->SetSinksEnabled(this, enabled);
|
| -}
|
| -
|
| -void MediaStreamRemoteAudioTrack::OnStop() {
|
| - DCHECK(main_render_thread_checker_.CalledOnValidThread());
|
| - DVLOG(1) << "MediaStreamRemoteAudioTrack::OnStop()";
|
| -
|
| - source()->RemoveAll(this);
|
| -
|
| - // Stop means that a track should be stopped permanently. But
|
| - // since there is no proper way of doing that on a remote track, we can
|
| - // at least disable the track. Blink will not call down to the content layer
|
| - // after a track has been stopped.
|
| - SetEnabled(false);
|
| -}
|
| -
|
| -void MediaStreamRemoteAudioTrack::AddSink(MediaStreamAudioSink* sink) {
|
| - DCHECK(main_render_thread_checker_.CalledOnValidThread());
|
| - return source()->AddSink(sink, this, enabled_);
|
| -}
|
| -
|
| -void MediaStreamRemoteAudioTrack::RemoveSink(MediaStreamAudioSink* sink) {
|
| - DCHECK(main_render_thread_checker_.CalledOnValidThread());
|
| - return source()->RemoveSink(sink, this);
|
| -}
|
| -
|
| -media::AudioParameters MediaStreamRemoteAudioTrack::GetOutputFormat() const {
|
| - DCHECK(main_render_thread_checker_.CalledOnValidThread());
|
| - // This method is not implemented on purpose and should be removed.
|
| - // TODO(tommi): See comment for GetOutputFormat in MediaStreamAudioTrack.
|
| - NOTIMPLEMENTED();
|
| - return media::AudioParameters();
|
| -}
|
| -
|
| -webrtc::AudioTrackInterface* MediaStreamRemoteAudioTrack::GetAudioAdapter() {
|
| - DCHECK(main_render_thread_checker_.CalledOnValidThread());
|
| - return source()->GetAudioAdapter();
|
| -}
|
| -
|
| -MediaStreamRemoteAudioSource* MediaStreamRemoteAudioTrack::source() const {
|
| - return static_cast<MediaStreamRemoteAudioSource*>(source_.getExtraData());
|
| -}
|
| -
|
| -MediaStreamRemoteAudioSource::MediaStreamRemoteAudioSource(
|
| - const scoped_refptr<webrtc::AudioTrackInterface>& track) : track_(track) {}
|
| -
|
| -MediaStreamRemoteAudioSource::~MediaStreamRemoteAudioSource() {
|
| - DCHECK(thread_checker_.CalledOnValidThread());
|
| -}
|
| -
|
| -void MediaStreamRemoteAudioSource::SetEnabledForMixing(bool enabled) {
|
| - DCHECK(thread_checker_.CalledOnValidThread());
|
| - track_->set_enabled(enabled);
|
| -}
|
| -
|
| -void MediaStreamRemoteAudioSource::AddSink(MediaStreamAudioSink* sink,
|
| - MediaStreamAudioTrack* track,
|
| - bool enabled) {
|
| - DCHECK(thread_checker_.CalledOnValidThread());
|
| - if (!sink_) {
|
| - sink_.reset(new AudioSink());
|
| - track_->AddSink(sink_.get());
|
| - }
|
| -
|
| - sink_->Add(sink, track, enabled);
|
| -}
|
| -
|
| -void MediaStreamRemoteAudioSource::RemoveSink(MediaStreamAudioSink* sink,
|
| - MediaStreamAudioTrack* track) {
|
| - DCHECK(thread_checker_.CalledOnValidThread());
|
| - DCHECK(sink_);
|
| -
|
| - sink_->Remove(sink, track);
|
| -
|
| - if (!sink_->IsNeeded()) {
|
| - track_->RemoveSink(sink_.get());
|
| - sink_.reset();
|
| - }
|
| -}
|
| -
|
| -void MediaStreamRemoteAudioSource::SetSinksEnabled(MediaStreamAudioTrack* track,
|
| - bool enabled) {
|
| - if (sink_)
|
| - sink_->SetEnabled(track, enabled);
|
| -}
|
| -
|
| -void MediaStreamRemoteAudioSource::RemoveAll(MediaStreamAudioTrack* track) {
|
| - if (sink_)
|
| - sink_->RemoveAll(track);
|
| -}
|
| -
|
| -webrtc::AudioTrackInterface* MediaStreamRemoteAudioSource::GetAudioAdapter() {
|
| - DCHECK(thread_checker_.CalledOnValidThread());
|
| - return track_.get();
|
| -}
|
| -
|
| -} // namespace content
|
|
|