Clean up the media stream audio track code

content/renderer/media/webrtc/peer_connection_dependency_factory.cc

Index: content/renderer/media/webrtc/peer_connection_dependency_factory.cc
diff --git a/content/renderer/media/webrtc/peer_connection_dependency_factory.cc b/content/renderer/media/webrtc/peer_connection_dependency_factory.cc
index 33406f213753dde260dfe8df97fc86a47d36a671..74f354ed241195d2d102563b266fd1310d45bf27 100644
--- a/content/renderer/media/webrtc/peer_connection_dependency_factory.cc
+++ b/content/renderer/media/webrtc/peer_connection_dependency_factory.cc
@@ -174,8 +174,6 @@ PeerConnectionDependencyFactory::PeerConnectionDependencyFactory(
 
 PeerConnectionDependencyFactory::~PeerConnectionDependencyFactory() {
   CleanupPeerConnectionFactory();
-  if (aec_dump_message_filter_.get())
-    aec_dump_message_filter_->RemoveDelegate(this);
 }
 
 blink::WebRTCPeerConnectionHandler*
@@ -352,17 +350,6 @@ void PeerConnectionDependencyFactory::CreatePeerConnectionFactory() {
   factory_options.disable_encryption =
       cmd_line->HasSwitch(switches::kDisableWebRtcEncryption);
   pc_factory_->SetOptions(factory_options);
-
-  // TODO(xians): Remove the following code after kDisableAudioTrackProcessing
-  // is removed.
-  if (!MediaStreamAudioProcessor::IsAudioTrackProcessingEnabled()) {
-    aec_dump_message_filter_ = AecDumpMessageFilter::Get();
-    // In unit tests not creating a message filter, |aec_dump_message_filter_|
-    // will be NULL. We can just ignore that. Other unit tests and browser tests
-    // ensure that we do get the filter when we should.
-    if (aec_dump_message_filter_.get())
-      aec_dump_message_filter_->AddDelegate(this);
-  }
 }
 
 bool PeerConnectionDependencyFactory::PeerConnectionFactoryCreated() {
@@ -456,12 +443,6 @@ void PeerConnectionDependencyFactory::CreateLocalAudioTrack(
 
 void PeerConnectionDependencyFactory::StartLocalAudioTrack(
     WebRtcLocalAudioTrack* audio_track) {
-  // Add the WebRtcAudioDevice as the sink to the local audio track.
-  // TODO(xians): Remove the following line of code after the APM in WebRTC is
-  // completely deprecated. See http://crbug/365672.
-  if (!MediaStreamAudioProcessor::IsAudioTrackProcessingEnabled())
-    audio_track->AddSink(GetWebRtcAudioDevice());
-
   // Start the audio track. This will hook the |audio_track| to the capturer
   // as the sink of the audio, and only start the source of the capturer if
   // it is the first audio track connecting to the capturer.
@@ -622,32 +603,6 @@ PeerConnectionDependencyFactory::GetWebRtcWorkerThread() const {
   return chrome_worker_thread_.message_loop_proxy();
 }
 
-void PeerConnectionDependencyFactory::OnAecDumpFile(
-    const IPC::PlatformFileForTransit& file_handle) {
-  DCHECK(CalledOnValidThread());
-  DCHECK(!MediaStreamAudioProcessor::IsAudioTrackProcessingEnabled());
-  DCHECK(PeerConnectionFactoryCreated());
-
-  base::File file = IPC::PlatformFileForTransitToFile(file_handle);
-  DCHECK(file.IsValid());
-
-  // |pc_factory_| always takes ownership of |aec_dump_file|. If StartAecDump()
-  // fails, |aec_dump_file| will be closed.
-  if (!GetPcFactory()->StartAecDump(file.TakePlatformFile()))
-    VLOG(1) << "Could not start AEC dump.";
-}
-
-void PeerConnectionDependencyFactory::OnDisableAecDump() {
-  DCHECK(CalledOnValidThread());
-  DCHECK(!MediaStreamAudioProcessor::IsAudioTrackProcessingEnabled());
-  // Do nothing. We never disable AEC dump for non-track-processing case.
-}
-
-void PeerConnectionDependencyFactory::OnIpcClosing() {
-  DCHECK(CalledOnValidThread());
-  aec_dump_message_filter_ = NULL;
-}
-
 void PeerConnectionDependencyFactory::EnsureWebRtcAudioDeviceImpl() {
   if (audio_device_.get())
     return;