Renamed MediaStreamDependencyFactory to PeerConnectionDependencyFactory.

content/renderer/media/media_stream_dependency_factory.cc

Index: content/renderer/media/media_stream_dependency_factory.cc
diff --git a/content/renderer/media/media_stream_dependency_factory.cc b/content/renderer/media/media_stream_dependency_factory.cc
deleted file mode 100644
index 2a781b32faabbf51f97c5de7b06576083e34d1e0..0000000000000000000000000000000000000000
--- a/content/renderer/media/media_stream_dependency_factory.cc
+++ /dev/null
@@ -1,669 +0,0 @@
-// Copyright (c) 2012 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/media_stream_dependency_factory.h"
-
-#include <vector>
-
-#include "base/command_line.h"
-#include "base/strings/utf_string_conversions.h"
-#include "base/synchronization/waitable_event.h"
-#include "content/common/media/media_stream_messages.h"
-#include "content/public/common/content_switches.h"
-#include "content/renderer/media/media_stream.h"
-#include "content/renderer/media/media_stream_audio_processor_options.h"
-#include "content/renderer/media/media_stream_audio_source.h"
-#include "content/renderer/media/media_stream_video_source.h"
-#include "content/renderer/media/media_stream_video_track.h"
-#include "content/renderer/media/peer_connection_identity_service.h"
-#include "content/renderer/media/rtc_media_constraints.h"
-#include "content/renderer/media/rtc_peer_connection_handler.h"
-#include "content/renderer/media/rtc_video_decoder_factory.h"
-#include "content/renderer/media/rtc_video_encoder_factory.h"
-#include "content/renderer/media/webaudio_capturer_source.h"
-#include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h"
-#include "content/renderer/media/webrtc/webrtc_video_capturer_adapter.h"
-#include "content/renderer/media/webrtc_audio_device_impl.h"
-#include "content/renderer/media/webrtc_local_audio_track.h"
-#include "content/renderer/media/webrtc_uma_histograms.h"
-#include "content/renderer/p2p/ipc_network_manager.h"
-#include "content/renderer/p2p/ipc_socket_factory.h"
-#include "content/renderer/p2p/port_allocator.h"
-#include "content/renderer/render_thread_impl.h"
-#include "jingle/glue/thread_wrapper.h"
-#include "media/filters/gpu_video_accelerator_factories.h"
-#include "third_party/WebKit/public/platform/WebMediaConstraints.h"
-#include "third_party/WebKit/public/platform/WebMediaStream.h"
-#include "third_party/WebKit/public/platform/WebMediaStreamSource.h"
-#include "third_party/WebKit/public/platform/WebMediaStreamTrack.h"
-#include "third_party/WebKit/public/platform/WebURL.h"
-#include "third_party/WebKit/public/web/WebDocument.h"
-#include "third_party/WebKit/public/web/WebFrame.h"
-#include "third_party/libjingle/source/talk/app/webrtc/mediaconstraintsinterface.h"
-
-#if defined(USE_OPENSSL)
-#include "third_party/libjingle/source/talk/base/ssladapter.h"
-#else
-#include "net/socket/nss_ssl_util.h"
-#endif
-
-#if defined(OS_ANDROID)
-#include "media/base/android/media_codec_bridge.h"
-#endif
-
-namespace content {
-
-// Map of corresponding media constraints and platform effects.
-struct {
-  const char* constraint;
-  const media::AudioParameters::PlatformEffectsMask effect;
-} const kConstraintEffectMap[] = {
-  { content::kMediaStreamAudioDucking,
-    media::AudioParameters::DUCKING },
-  { webrtc::MediaConstraintsInterface::kEchoCancellation,
-    media::AudioParameters::ECHO_CANCELLER },
-};
-
-// If any platform effects are available, check them against the constraints.
-// Disable effects to match false constraints, but if a constraint is true, set
-// the constraint to false to later disable the software effect.
-//
-// This function may modify both |constraints| and |effects|.
-void HarmonizeConstraintsAndEffects(RTCMediaConstraints* constraints,
-                                    int* effects) {
-  if (*effects != media::AudioParameters::NO_EFFECTS) {
-    for (size_t i = 0; i < ARRAYSIZE_UNSAFE(kConstraintEffectMap); ++i) {
-      bool value;
-      size_t is_mandatory = 0;
-      if (!webrtc::FindConstraint(constraints,
-                                  kConstraintEffectMap[i].constraint,
-                                  &value,
-                                  &is_mandatory) || !value) {
-        // If the constraint is false, or does not exist, disable the platform
-        // effect.
-        *effects &= ~kConstraintEffectMap[i].effect;
-        DVLOG(1) << "Disabling platform effect: "
-                 << kConstraintEffectMap[i].effect;
-      } else if (*effects & kConstraintEffectMap[i].effect) {
-        // If the constraint is true, leave the platform effect enabled, and
-        // set the constraint to false to later disable the software effect.
-        if (is_mandatory) {
-          constraints->AddMandatory(kConstraintEffectMap[i].constraint,
-              webrtc::MediaConstraintsInterface::kValueFalse, true);
-        } else {
-          constraints->AddOptional(kConstraintEffectMap[i].constraint,
-              webrtc::MediaConstraintsInterface::kValueFalse, true);
-        }
-        DVLOG(1) << "Disabling constraint: "
-                 << kConstraintEffectMap[i].constraint;
-      }
-    }
-  }
-}
-
-class P2PPortAllocatorFactory : public webrtc::PortAllocatorFactoryInterface {
- public:
-  P2PPortAllocatorFactory(
-      P2PSocketDispatcher* socket_dispatcher,
-      talk_base::NetworkManager* network_manager,
-      talk_base::PacketSocketFactory* socket_factory,
-      blink::WebFrame* web_frame)
-      : socket_dispatcher_(socket_dispatcher),
-        network_manager_(network_manager),
-        socket_factory_(socket_factory),
-        web_frame_(web_frame) {
-  }
-
-  virtual cricket::PortAllocator* CreatePortAllocator(
-      const std::vector<StunConfiguration>& stun_servers,
-      const std::vector<TurnConfiguration>& turn_configurations) OVERRIDE {
-    CHECK(web_frame_);
-    P2PPortAllocator::Config config;
-    if (stun_servers.size() > 0) {
-      config.stun_server = stun_servers[0].server.hostname();
-      config.stun_server_port = stun_servers[0].server.port();
-    }
-    config.legacy_relay = false;
-    for (size_t i = 0; i < turn_configurations.size(); ++i) {
-      P2PPortAllocator::Config::RelayServerConfig relay_config;
-      relay_config.server_address = turn_configurations[i].server.hostname();
-      relay_config.port = turn_configurations[i].server.port();
-      relay_config.username = turn_configurations[i].username;
-      relay_config.password = turn_configurations[i].password;
-      relay_config.transport_type = turn_configurations[i].transport_type;
-      relay_config.secure = turn_configurations[i].secure;
-      config.relays.push_back(relay_config);
-    }
-
-    // Use first turn server as the stun server.
-    if (turn_configurations.size() > 0) {
-      config.stun_server = config.relays[0].server_address;
-      config.stun_server_port = config.relays[0].port;
-    }
-
-    return new P2PPortAllocator(
-        web_frame_, socket_dispatcher_.get(), network_manager_,
-        socket_factory_, config);
-  }
-
- protected:
-  virtual ~P2PPortAllocatorFactory() {}
-
- private:
-  scoped_refptr<P2PSocketDispatcher> socket_dispatcher_;
-  // |network_manager_| and |socket_factory_| are a weak references, owned by
-  // MediaStreamDependencyFactory.
-  talk_base::NetworkManager* network_manager_;
-  talk_base::PacketSocketFactory* socket_factory_;
-  // Raw ptr to the WebFrame that created the P2PPortAllocatorFactory.
-  blink::WebFrame* web_frame_;
-};
-
-MediaStreamDependencyFactory::MediaStreamDependencyFactory(
-    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") {
-}
-
-MediaStreamDependencyFactory::~MediaStreamDependencyFactory() {
-  CleanupPeerConnectionFactory();
-}
-
-blink::WebRTCPeerConnectionHandler*
-MediaStreamDependencyFactory::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);
-
-  return new RTCPeerConnectionHandler(client, this);
-}
-
-bool MediaStreamDependencyFactory::InitializeMediaStreamAudioSource(
-    int render_view_id,
-    const blink::WebMediaConstraints& audio_constraints,
-    MediaStreamAudioSource* source_data) {
-  DVLOG(1) << "InitializeMediaStreamAudioSources()";
-
-  // Do additional source initialization if the audio source is a valid
-  // microphone or tab audio.
-  RTCMediaConstraints native_audio_constraints(audio_constraints);
-  ApplyFixedAudioConstraints(&native_audio_constraints);
-
-  StreamDeviceInfo device_info = source_data->device_info();
-  RTCMediaConstraints constraints = native_audio_constraints;
-  // May modify both |constraints| and |effects|.
-  HarmonizeConstraintsAndEffects(&constraints,
-                                 &device_info.device.input.effects);
-
-  scoped_refptr<WebRtcAudioCapturer> capturer(
-      CreateAudioCapturer(render_view_id, device_info, audio_constraints,
-                          source_data));
-  if (!capturer.get()) {
-    DLOG(WARNING) << "Failed to create the capturer for device "
-        << device_info.device.id;
-    // TODO(xians): Don't we need to check if source_observer is observing
-    // something? If not, then it looks like we have a leak here.
-    // OTOH, if it _is_ observing something, then the callback might
-    // be called multiple times which is likely also a bug.
-    return false;
-  }
-  source_data->SetAudioCapturer(capturer);
-
-  // Creates a LocalAudioSource object which holds audio options.
-  // TODO(xians): The option should apply to the track instead of the source.
-  // TODO(perkj): Move audio constraints parsing to Chrome.
-  // Currently there are a few constraints that are parsed by libjingle and
-  // the state is set to ended if parsing fails.
-  scoped_refptr<webrtc::AudioSourceInterface> rtc_source(
-      CreateLocalAudioSource(&constraints).get());
-  if (rtc_source->state() != webrtc::MediaSourceInterface::kLive) {
-    DLOG(WARNING) << "Failed to create rtc LocalAudioSource.";
-    return false;
-  }
-  source_data->SetLocalAudioSource(rtc_source);
-  return true;
-}
-
-WebRtcVideoCapturerAdapter* MediaStreamDependencyFactory::CreateVideoCapturer(
-    bool is_screeencast) {
-  // We need to make sure the libjingle thread wrappers have been created
-  // before we can use an instance of a WebRtcVideoCapturerAdapter. This is
-  // since the base class of WebRtcVideoCapturerAdapter is a
-  // cricket::VideoCapturer and it uses the libjingle thread wrappers.
-  if (!GetPcFactory())
-    return NULL;
-  return new WebRtcVideoCapturerAdapter(is_screeencast);
-}
-
-scoped_refptr<webrtc::VideoSourceInterface>
-MediaStreamDependencyFactory::CreateVideoSource(
-    cricket::VideoCapturer* capturer,
-    const blink::WebMediaConstraints& constraints) {
-  RTCMediaConstraints webrtc_constraints(constraints);
-  scoped_refptr<webrtc::VideoSourceInterface> source =
-      GetPcFactory()->CreateVideoSource(capturer, &webrtc_constraints).get();
-  return source;
-}
-
-const scoped_refptr<webrtc::PeerConnectionFactoryInterface>&
-MediaStreamDependencyFactory::GetPcFactory() {
-  if (!pc_factory_)
-    CreatePeerConnectionFactory();
-  CHECK(pc_factory_);
-  return pc_factory_;
-}
-
-void MediaStreamDependencyFactory::CreatePeerConnectionFactory() {
-  DCHECK(!pc_factory_.get());
-  DCHECK(!signaling_thread_);
-  DCHECK(!worker_thread_);
-  DCHECK(!network_manager_);
-  DCHECK(!socket_factory_);
-  DCHECK(!chrome_worker_thread_.IsRunning());
-
-  DVLOG(1) << "MediaStreamDependencyFactory::CreatePeerConnectionFactory()";
-
-  jingle_glue::JingleThreadWrapper::EnsureForCurrentMessageLoop();
-  jingle_glue::JingleThreadWrapper::current()->set_send_allowed(true);
-  signaling_thread_ = jingle_glue::JingleThreadWrapper::current();
-  CHECK(signaling_thread_);
-
-  CHECK(chrome_worker_thread_.Start());
-
-  base::WaitableEvent start_worker_event(true, false);
-  chrome_worker_thread_.message_loop()->PostTask(FROM_HERE, base::Bind(
-      &MediaStreamDependencyFactory::InitializeWorkerThread,
-      base::Unretained(this),
-      &worker_thread_,
-      &start_worker_event));
-  start_worker_event.Wait();
-  CHECK(worker_thread_);
-
-  base::WaitableEvent create_network_manager_event(true, false);
-  chrome_worker_thread_.message_loop()->PostTask(FROM_HERE, base::Bind(
-      &MediaStreamDependencyFactory::CreateIpcNetworkManagerOnWorkerThread,
-      base::Unretained(this),
-      &create_network_manager_event));
-  create_network_manager_event.Wait();
-
-  socket_factory_.reset(
-      new IpcPacketSocketFactory(p2p_socket_dispatcher_.get()));
-
-  // Init SSL, which will be needed by PeerConnection.
-#if defined(USE_OPENSSL)
-  if (!talk_base::InitializeSSL()) {
-    LOG(ERROR) << "Failed on InitializeSSL.";
-    NOTREACHED();
-    return;
-  }
-#else
-  // TODO(ronghuawu): Replace this call with InitializeSSL.
-  net::EnsureNSSSSLInit();
-#endif
-
-  scoped_ptr<cricket::WebRtcVideoDecoderFactory> decoder_factory;
-  scoped_ptr<cricket::WebRtcVideoEncoderFactory> encoder_factory;
-
-  const CommandLine* cmd_line = CommandLine::ForCurrentProcess();
-  scoped_refptr<media::GpuVideoAcceleratorFactories> gpu_factories =
-      RenderThreadImpl::current()->GetGpuFactories();
-  if (!cmd_line->HasSwitch(switches::kDisableWebRtcHWDecoding)) {
-    if (gpu_factories)
-      decoder_factory.reset(new RTCVideoDecoderFactory(gpu_factories));
-  }
-
-  if (!cmd_line->HasSwitch(switches::kDisableWebRtcHWEncoding)) {
-    if (gpu_factories)
-      encoder_factory.reset(new RTCVideoEncoderFactory(gpu_factories));
-  }
-
-#if defined(OS_ANDROID)
-  if (!media::MediaCodecBridge::SupportsSetParameters())
-    encoder_factory.reset();
-#endif
-
-  EnsureWebRtcAudioDeviceImpl();
-
-  scoped_refptr<webrtc::PeerConnectionFactoryInterface> factory(
-      webrtc::CreatePeerConnectionFactory(worker_thread_,
-                                          signaling_thread_,
-                                          audio_device_.get(),
-                                          encoder_factory.release(),
-                                          decoder_factory.release()));
-  CHECK(factory);
-
-  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);
-
-  // |aec_dump_file| will be invalid when dump is not enabled.
-  if (aec_dump_file_.IsValid())
-    StartAecDump(aec_dump_file_.Pass());
-}
-
-bool MediaStreamDependencyFactory::PeerConnectionFactoryCreated() {
-  return pc_factory_.get() != NULL;
-}
-
-scoped_refptr<webrtc::PeerConnectionInterface>
-MediaStreamDependencyFactory::CreatePeerConnection(
-    const webrtc::PeerConnectionInterface::IceServers& ice_servers,
-    const webrtc::MediaConstraintsInterface* constraints,
-    blink::WebFrame* web_frame,
-    webrtc::PeerConnectionObserver* observer) {
-  CHECK(web_frame);
-  CHECK(observer);
-  if (!GetPcFactory())
-    return NULL;
-
-  scoped_refptr<P2PPortAllocatorFactory> pa_factory =
-        new talk_base::RefCountedObject<P2PPortAllocatorFactory>(
-            p2p_socket_dispatcher_.get(),
-            network_manager_,
-            socket_factory_.get(),
-            web_frame);
-
-  PeerConnectionIdentityService* identity_service =
-      new PeerConnectionIdentityService(
-          GURL(web_frame->document().url().spec()).GetOrigin());
-
-  return GetPcFactory()->CreatePeerConnection(ice_servers,
-                                            constraints,
-                                            pa_factory.get(),
-                                            identity_service,
-                                            observer).get();
-}
-
-scoped_refptr<webrtc::MediaStreamInterface>
-MediaStreamDependencyFactory::CreateLocalMediaStream(
-    const std::string& label) {
-  return GetPcFactory()->CreateLocalMediaStream(label).get();
-}
-
-scoped_refptr<webrtc::AudioSourceInterface>
-MediaStreamDependencyFactory::CreateLocalAudioSource(
-    const webrtc::MediaConstraintsInterface* constraints) {
-  scoped_refptr<webrtc::AudioSourceInterface> source =
-      GetPcFactory()->CreateAudioSource(constraints).get();
-  return source;
-}
-
-void MediaStreamDependencyFactory::CreateLocalAudioTrack(
-    const blink::WebMediaStreamTrack& track) {
-  blink::WebMediaStreamSource source = track.source();
-  DCHECK_EQ(source.type(), blink::WebMediaStreamSource::TypeAudio);
-  MediaStreamAudioSource* source_data =
-      static_cast<MediaStreamAudioSource*>(source.extraData());
-
-  scoped_refptr<WebAudioCapturerSource> webaudio_source;
-  if (!source_data) {
-    if (source.requiresAudioConsumer()) {
-      // We're adding a WebAudio MediaStream.
-      // Create a specific capturer for each WebAudio consumer.
-      webaudio_source = CreateWebAudioSource(&source);
-      source_data =
-          static_cast<MediaStreamAudioSource*>(source.extraData());
-    } else {
-      // TODO(perkj): Implement support for sources from
-      // remote MediaStreams.
-      NOTIMPLEMENTED();
-      return;
-    }
-  }
-
-  // Creates an adapter to hold all the libjingle objects.
-  scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter(
-      WebRtcLocalAudioTrackAdapter::Create(track.id().utf8(),
-                                           source_data->local_audio_source()));
-  static_cast<webrtc::AudioTrackInterface*>(adapter.get())->set_enabled(
-      track.isEnabled());
-
-  // TODO(xians): Merge |source| to the capturer(). We can't do this today
-  // because only one capturer() is supported while one |source| is created
-  // for each audio track.
-  scoped_ptr<WebRtcLocalAudioTrack> audio_track(
-      new WebRtcLocalAudioTrack(adapter,
-                                source_data->GetAudioCapturer(),
-                                webaudio_source));
-
-  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 MediaStreamDependencyFactory::StartLocalAudioTrack(
-    WebRtcLocalAudioTrack* audio_track) {
-  // Add the WebRtcAudioDevice as the sink to the local audio track.
-  // TODO(xians): Implement a PeerConnection sink adapter and remove this
-  // AddSink() call.
-  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>
-MediaStreamDependencyFactory::CreateWebAudioSource(
-    blink::WebMediaStreamSource* source) {
-  DVLOG(1) << "MediaStreamDependencyFactory::CreateWebAudioSource()";
-
-  scoped_refptr<WebAudioCapturerSource>
-      webaudio_capturer_source(new WebAudioCapturerSource());
-  MediaStreamAudioSource* source_data = new MediaStreamAudioSource();
-
-  // Use the current default capturer for the WebAudio track so that the
-  // WebAudio track can pass a valid delay value and |need_audio_processing|
-  // flag to PeerConnection.
-  // TODO(xians): Remove this after moving APM to Chrome.
-  if (GetWebRtcAudioDevice()) {
-    source_data->SetAudioCapturer(
-        GetWebRtcAudioDevice()->GetDefaultCapturer());
-  }
-
-  // Create a LocalAudioSource object which holds audio options.
-  // SetLocalAudioSource() affects core audio parts in third_party/Libjingle.
-  source_data->SetLocalAudioSource(CreateLocalAudioSource(NULL).get());
-  source->setExtraData(source_data);
-
-  // Replace the default source with WebAudio as source instead.
-  source->addAudioConsumer(webaudio_capturer_source.get());
-
-  return webaudio_capturer_source;
-}
-
-scoped_refptr<webrtc::VideoTrackInterface>
-MediaStreamDependencyFactory::CreateLocalVideoTrack(
-    const std::string& id,
-    webrtc::VideoSourceInterface* source) {
-  return GetPcFactory()->CreateVideoTrack(id, source).get();
-}
-
-scoped_refptr<webrtc::VideoTrackInterface>
-MediaStreamDependencyFactory::CreateLocalVideoTrack(
-    const std::string& id, cricket::VideoCapturer* capturer) {
-  if (!capturer) {
-    LOG(ERROR) << "CreateLocalVideoTrack called with null VideoCapturer.";
-    return NULL;
-  }
-
-  // Create video source from the |capturer|.
-  scoped_refptr<webrtc::VideoSourceInterface> source =
-      GetPcFactory()->CreateVideoSource(capturer, NULL).get();
-
-  // Create native track from the source.
-  return GetPcFactory()->CreateVideoTrack(id, source.get()).get();
-}
-
-webrtc::SessionDescriptionInterface*
-MediaStreamDependencyFactory::CreateSessionDescription(
-    const std::string& type,
-    const std::string& sdp,
-    webrtc::SdpParseError* error) {
-  return webrtc::CreateSessionDescription(type, sdp, error);
-}
-
-webrtc::IceCandidateInterface* MediaStreamDependencyFactory::CreateIceCandidate(
-    const std::string& sdp_mid,
-    int sdp_mline_index,
-    const std::string& sdp) {
-  return webrtc::CreateIceCandidate(sdp_mid, sdp_mline_index, sdp);
-}
-
-WebRtcAudioDeviceImpl*
-MediaStreamDependencyFactory::GetWebRtcAudioDevice() {
-  return audio_device_.get();
-}
-
-void MediaStreamDependencyFactory::InitializeWorkerThread(
-    talk_base::Thread** thread,
-    base::WaitableEvent* event) {
-  jingle_glue::JingleThreadWrapper::EnsureForCurrentMessageLoop();
-  jingle_glue::JingleThreadWrapper::current()->set_send_allowed(true);
-  *thread = jingle_glue::JingleThreadWrapper::current();
-  event->Signal();
-}
-
-void MediaStreamDependencyFactory::CreateIpcNetworkManagerOnWorkerThread(
-    base::WaitableEvent* event) {
-  DCHECK_EQ(base::MessageLoop::current(), chrome_worker_thread_.message_loop());
-  network_manager_ = new IpcNetworkManager(p2p_socket_dispatcher_.get());
-  event->Signal();
-}
-
-void MediaStreamDependencyFactory::DeleteIpcNetworkManager() {
-  DCHECK_EQ(base::MessageLoop::current(), chrome_worker_thread_.message_loop());
-  delete network_manager_;
-  network_manager_ = NULL;
-}
-
-void MediaStreamDependencyFactory::CleanupPeerConnectionFactory() {
-  pc_factory_ = NULL;
-  if (network_manager_) {
-    // The network manager needs to free its resources on the thread they were
-    // created, which is the worked thread.
-    if (chrome_worker_thread_.IsRunning()) {
-      chrome_worker_thread_.message_loop()->PostTask(FROM_HERE, base::Bind(
-          &MediaStreamDependencyFactory::DeleteIpcNetworkManager,
-          base::Unretained(this)));
-      // Stopping the thread will wait until all tasks have been
-      // processed before returning. We wait for the above task to finish before
-      // letting the the function continue to avoid any potential race issues.
-      chrome_worker_thread_.Stop();
-    } else {
-      NOTREACHED() << "Worker thread not running.";
-    }
-  }
-}
-
-scoped_refptr<WebRtcAudioCapturer>
-MediaStreamDependencyFactory::CreateAudioCapturer(
-    int render_view_id,
-    const StreamDeviceInfo& device_info,
-    const blink::WebMediaConstraints& constraints,
-    MediaStreamAudioSource* audio_source) {
-  // TODO(xians): Handle the cases when gUM is called without a proper render
-  // view, for example, by an extension.
-  DCHECK_GE(render_view_id, 0);
-
-  EnsureWebRtcAudioDeviceImpl();
-  DCHECK(GetWebRtcAudioDevice());
-  return WebRtcAudioCapturer::CreateCapturer(render_view_id, device_info,
-                                             constraints,
-                                             GetWebRtcAudioDevice(),
-                                             audio_source);
-}
-
-void MediaStreamDependencyFactory::AddNativeAudioTrackToBlinkTrack(
-    webrtc::MediaStreamTrackInterface* native_track,
-    const blink::WebMediaStreamTrack& webkit_track,
-    bool is_local_track) {
-  DCHECK(!webkit_track.isNull() && !webkit_track.extraData());
-  DCHECK_EQ(blink::WebMediaStreamSource::TypeAudio,
-            webkit_track.source().type());
-  blink::WebMediaStreamTrack track = webkit_track;
-
-  DVLOG(1) << "AddNativeTrackToBlinkTrack() audio";
-  track.setExtraData(
-      new MediaStreamTrack(
-          static_cast<webrtc::AudioTrackInterface*>(native_track),
-          is_local_track));
-}
-
-scoped_refptr<base::MessageLoopProxy>
-MediaStreamDependencyFactory::GetWebRtcWorkerThread() const {
-  DCHECK(CalledOnValidThread());
-  return chrome_worker_thread_.message_loop_proxy();
-}
-
-bool MediaStreamDependencyFactory::OnControlMessageReceived(
-    const IPC::Message& message) {
-  bool handled = true;
-  IPC_BEGIN_MESSAGE_MAP(MediaStreamDependencyFactory, message)
-    IPC_MESSAGE_HANDLER(MediaStreamMsg_EnableAecDump, OnAecDumpFile)
-    IPC_MESSAGE_HANDLER(MediaStreamMsg_DisableAecDump, OnDisableAecDump)
-    IPC_MESSAGE_UNHANDLED(handled = false)
-  IPC_END_MESSAGE_MAP()
-  return handled;
-}
-
-void MediaStreamDependencyFactory::OnAecDumpFile(
-    IPC::PlatformFileForTransit file_handle) {
-  DCHECK(!aec_dump_file_.IsValid());
-  base::File file = IPC::PlatformFileForTransitToFile(file_handle);
-  DCHECK(file.IsValid());
-
-  if (CommandLine::ForCurrentProcess()->HasSwitch(
-          switches::kEnableAudioTrackProcessing)) {
-    EnsureWebRtcAudioDeviceImpl();
-    GetWebRtcAudioDevice()->EnableAecDump(file.Pass());
-    return;
-  }
-
-  // TODO(xians): Remove the following code after kEnableAudioTrackProcessing
-  // is removed.
-  if (PeerConnectionFactoryCreated())
-    StartAecDump(file.Pass());
-  else
-    aec_dump_file_ = file.Pass();
-}
-
-void MediaStreamDependencyFactory::OnDisableAecDump() {
-  if (CommandLine::ForCurrentProcess()->HasSwitch(
-          switches::kEnableAudioTrackProcessing)) {
-    GetWebRtcAudioDevice()->DisableAecDump();
-    return;
-  }
-
-  // TODO(xians): Remove the following code after kEnableAudioTrackProcessing
-  // is removed.
-  if (aec_dump_file_.IsValid())
-    aec_dump_file_.Close();
-}
-
-void MediaStreamDependencyFactory::StartAecDump(base::File aec_dump_file) {
-  // |pc_factory_| always takes ownership of |aec_dump_file|. If StartAecDump()
-  // fails, |aec_dump_file| will be closed.
-  if (!GetPcFactory()->StartAecDump(aec_dump_file.TakePlatformFile()))
-    VLOG(1) << "Could not start AEC dump.";
-}
-
-void MediaStreamDependencyFactory::EnsureWebRtcAudioDeviceImpl() {
-  if (audio_device_)
-    return;
-
-  audio_device_ = new WebRtcAudioDeviceImpl();
-}
-
-}  // namespace content