Clean up the media stream audio track code

content/renderer/media/webrtc/peer_connection_dependency_factory.cc

 // Copyright 2014 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_dependency_factory.h"
 
 #include <vector>
 
 #include "base/command_line.h"
 #include "base/strings/utf_string_conversions.h"
@@ skipping 156 matching lines @@
     P2PSocketDispatcher* p2p_socket_dispatcher)
     : network_manager_(NULL),
       p2p_socket_dispatcher_(p2p_socket_dispatcher),
       signaling_thread_(NULL),
       worker_thread_(NULL),
       chrome_worker_thread_("Chrome_libJingle_WorkerThread") {
 }
 
 PeerConnectionDependencyFactory::~PeerConnectionDependencyFactory() {
   CleanupPeerConnectionFactory();
-  if (aec_dump_message_filter_.get())
-    aec_dump_message_filter_->RemoveDelegate(this);
 }
 
 blink::WebRTCPeerConnectionHandler*
 PeerConnectionDependencyFactory::CreateRTCPeerConnectionHandler(
     blink::WebRTCPeerConnectionHandlerClient* client) {
   // Save histogram data so we can see how much PeerConnetion is used.
   // The histogram counts the number of calls to the JS API
   // webKitRTCPeerConnection.
   UpdateWebRTCMethodCount(WEBKIT_RTC_PEER_CONNECTION);
 
@@ skipping 156 matching lines @@
                                           encoder_factory.release(),
                                           decoder_factory.release()));
   CHECK(factory.get());
 
   pc_factory_ = factory;
   webrtc::PeerConnectionFactoryInterface::Options factory_options;
   factory_options.disable_sctp_data_channels = false;
   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() {
   return pc_factory_.get() != NULL;
 }
 
 scoped_refptr<webrtc::PeerConnectionInterface>
 PeerConnectionDependencyFactory::CreatePeerConnection(
     const webrtc::PeerConnectionInterface::RTCConfiguration& config,
     const webrtc::MediaConstraintsInterface* constraints,
@@ skipping 73 matching lines @@
 
   StartLocalAudioTrack(audio_track.get());
 
   // Pass the ownership of the native local audio track to the blink track.
   blink::WebMediaStreamTrack writable_track = track;
   writable_track.setExtraData(audio_track.release());
 }
 
 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.
   audio_track->Start();
 }
 
 scoped_refptr<WebAudioCapturerSource>
 PeerConnectionDependencyFactory::CreateWebAudioSource(
     blink::WebMediaStreamSource* source) {
   DVLOG(1) << "PeerConnectionDependencyFactory::CreateWebAudioSource()";
@@ skipping 140 matching lines @@
           static_cast<webrtc::AudioTrackInterface*>(native_track),
           is_local_track));
 }
 
 scoped_refptr<base::MessageLoopProxy>
 PeerConnectionDependencyFactory::GetWebRtcWorkerThread() const {
   DCHECK(CalledOnValidThread());
   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;
 
   audio_device_ = new WebRtcAudioDeviceImpl();
 }
 
 }  // namespace content