Revert of MediaStream audio: Refactor 3 separate "glue" implementations into one.

content/renderer/media/webrtc/media_stream_remote_audio_track.cc

+// 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