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

content/renderer/media/webrtc/peer_connection_remote_audio_source.cc

Index: content/renderer/media/webrtc/peer_connection_remote_audio_source.cc
diff --git a/content/renderer/media/webrtc/peer_connection_remote_audio_source.cc b/content/renderer/media/webrtc/peer_connection_remote_audio_source.cc
deleted file mode 100644
index 46d7318c3b2f5f91b440a4cb82e7a3b0f9245799..0000000000000000000000000000000000000000
--- a/content/renderer/media/webrtc/peer_connection_remote_audio_source.cc
+++ /dev/null
@@ -1,153 +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/peer_connection_remote_audio_source.h"
-
-#include "base/logging.h"
-#include "base/time/time.h"
-#include "media/base/audio_bus.h"
-
-namespace content {
-
-namespace {
-// Used as an identifier for the down-casters.
-void* const kClassIdentifier = const_cast<void**>(&kClassIdentifier);
-}  // namespace
-
-PeerConnectionRemoteAudioTrack::PeerConnectionRemoteAudioTrack(
-    scoped_refptr<webrtc::AudioTrackInterface> track_interface)
-    : MediaStreamAudioTrack(false /* is_local_track */),
-      track_interface_(std::move(track_interface)) {
-  DVLOG(1)
-      << "PeerConnectionRemoteAudioTrack::PeerConnectionRemoteAudioTrack()";
-}
-
-PeerConnectionRemoteAudioTrack::~PeerConnectionRemoteAudioTrack() {
-  DVLOG(1)
-      << "PeerConnectionRemoteAudioTrack::~PeerConnectionRemoteAudioTrack()";
-  // Ensure the track is stopped.
-  MediaStreamAudioTrack::Stop();
-}
-
-// static
-PeerConnectionRemoteAudioTrack* PeerConnectionRemoteAudioTrack::From(
-    MediaStreamAudioTrack* track) {
-  if (track && track->GetClassIdentifier() == kClassIdentifier)
-    return static_cast<PeerConnectionRemoteAudioTrack*>(track);
-  return nullptr;
-}
-
-void PeerConnectionRemoteAudioTrack::SetEnabled(bool enabled) {
-  DCHECK(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. This track's enabled
-  // state and the shared state might not be the same.
-  track_interface_->set_enabled(enabled);
-
-  MediaStreamAudioTrack::SetEnabled(enabled);
-}
-
-void* PeerConnectionRemoteAudioTrack::GetClassIdentifier() const {
-  return kClassIdentifier;
-}
-
-PeerConnectionRemoteAudioSource::PeerConnectionRemoteAudioSource(
-    scoped_refptr<webrtc::AudioTrackInterface> track_interface)
-    : MediaStreamAudioSource(false /* is_local_source */),
-      track_interface_(std::move(track_interface)),
-      is_sink_of_peer_connection_(false) {
-  DCHECK(track_interface_);
-  DVLOG(1)
-      << "PeerConnectionRemoteAudioSource::PeerConnectionRemoteAudioSource()";
-}
-
-PeerConnectionRemoteAudioSource::~PeerConnectionRemoteAudioSource() {
-  DVLOG(1)
-      << "PeerConnectionRemoteAudioSource::~PeerConnectionRemoteAudioSource()";
-  EnsureSourceIsStopped();
-}
-
-std::unique_ptr<MediaStreamAudioTrack>
-PeerConnectionRemoteAudioSource::CreateMediaStreamAudioTrack(
-    const std::string& id) {
-  DCHECK(thread_checker_.CalledOnValidThread());
-  return std::unique_ptr<MediaStreamAudioTrack>(
-      new PeerConnectionRemoteAudioTrack(track_interface_));
-}
-
-bool PeerConnectionRemoteAudioSource::EnsureSourceIsStarted() {
-  DCHECK(thread_checker_.CalledOnValidThread());
-  if (is_sink_of_peer_connection_)
-    return true;
-  VLOG(1) << "Starting PeerConnection remote audio source with id="
-          << track_interface_->id();
-  track_interface_->AddSink(this);
-  is_sink_of_peer_connection_ = true;
-  return true;
-}
-
-void PeerConnectionRemoteAudioSource::EnsureSourceIsStopped() {
-  DCHECK(thread_checker_.CalledOnValidThread());
-  if (is_sink_of_peer_connection_) {
-    track_interface_->RemoveSink(this);
-    is_sink_of_peer_connection_ = false;
-    VLOG(1) << "Stopped PeerConnection remote audio source with id="
-            << track_interface_->id();
-  }
-}
-
-void PeerConnectionRemoteAudioSource::OnData(const void* audio_data,
-                                             int bits_per_sample,
-                                             int sample_rate,
-                                             size_t number_of_channels,
-                                             size_t number_of_frames) {
-  // Debug builds: Note that this lock isn't meant to synchronize anything.
-  // Instead, it is being used as a run-time check to ensure there isn't already
-  // another thread executing this method. The reason we don't use
-  // base::ThreadChecker here is because we shouldn't be making assumptions
-  // about the private threading model of libjingle. For example, it would be
-  // legitimate for libjingle to use a different thread to invoke this method
-  // whenever the audio format changes.
-#ifndef NDEBUG
-  const bool is_only_thread_here = single_audio_thread_guard_.Try();
-  DCHECK(is_only_thread_here);
-#endif
-
-  // TODO(tommi): We should get the timestamp from WebRTC.
-  base::TimeTicks playout_time(base::TimeTicks::Now());
-
-  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);
-
-  media::AudioParameters params = MediaStreamAudioSource::GetAudioParameters();
-  if (!params.IsValid() ||
-      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) {
-    MediaStreamAudioSource::SetFormat(
-        media::AudioParameters(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
-                               media::GuessChannelLayout(number_of_channels),
-                               sample_rate, bits_per_sample, number_of_frames));
-  }
-
-  MediaStreamAudioSource::DeliverDataToTracks(*audio_bus_, playout_time);
-
-#ifndef NDEBUG
-  single_audio_thread_guard_.Release();
-#endif
-}
-
-}  // namespace content