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      1 // Copyright 2014 The Chromium Authors. All rights reserved.
      2 // Use of this source code is governed by a BSD-style license that can be
      3 // found in the LICENSE file.
      4 
      5 #include "content/renderer/media/webrtc/peer_connection_dependency_factory.h"
      6 
      7 #include <vector>
      8 
      9 #include "base/command_line.h"
     10 #include "base/strings/utf_string_conversions.h"
     11 #include "base/synchronization/waitable_event.h"
     12 #include "content/common/media/media_stream_messages.h"
     13 #include "content/public/common/content_switches.h"
     14 #include "content/renderer/media/media_stream.h"
     15 #include "content/renderer/media/media_stream_audio_processor.h"
     16 #include "content/renderer/media/media_stream_audio_processor_options.h"
     17 #include "content/renderer/media/media_stream_audio_source.h"
     18 #include "content/renderer/media/media_stream_video_source.h"
     19 #include "content/renderer/media/media_stream_video_track.h"
     20 #include "content/renderer/media/peer_connection_identity_service.h"
     21 #include "content/renderer/media/rtc_media_constraints.h"
     22 #include "content/renderer/media/rtc_peer_connection_handler.h"
     23 #include "content/renderer/media/rtc_video_decoder_factory.h"
     24 #include "content/renderer/media/rtc_video_encoder_factory.h"
     25 #include "content/renderer/media/webaudio_capturer_source.h"
     26 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h"
     27 #include "content/renderer/media/webrtc/webrtc_video_capturer_adapter.h"
     28 #include "content/renderer/media/webrtc_audio_device_impl.h"
     29 #include "content/renderer/media/webrtc_local_audio_track.h"
     30 #include "content/renderer/media/webrtc_logging.h"
     31 #include "content/renderer/media/webrtc_uma_histograms.h"
     32 #include "content/renderer/p2p/ipc_network_manager.h"
     33 #include "content/renderer/p2p/ipc_socket_factory.h"
     34 #include "content/renderer/p2p/port_allocator.h"
     35 #include "content/renderer/render_thread_impl.h"
     36 #include "jingle/glue/thread_wrapper.h"
     37 #include "media/filters/gpu_video_accelerator_factories.h"
     38 #include "third_party/WebKit/public/platform/WebMediaConstraints.h"
     39 #include "third_party/WebKit/public/platform/WebMediaStream.h"
     40 #include "third_party/WebKit/public/platform/WebMediaStreamSource.h"
     41 #include "third_party/WebKit/public/platform/WebMediaStreamTrack.h"
     42 #include "third_party/WebKit/public/platform/WebURL.h"
     43 #include "third_party/WebKit/public/web/WebDocument.h"
     44 #include "third_party/WebKit/public/web/WebFrame.h"
     45 #include "third_party/libjingle/source/talk/app/webrtc/mediaconstraintsinterface.h"
     46 
     47 #if defined(USE_OPENSSL)
     48 #include "third_party/webrtc/base/ssladapter.h"
     49 #else
     50 #include "net/socket/nss_ssl_util.h"
     51 #endif
     52 
     53 #if defined(OS_ANDROID)
     54 #include "media/base/android/media_codec_bridge.h"
     55 #endif
     56 
     57 namespace content {
     58 
     59 // Map of corresponding media constraints and platform effects.
     60 struct {
     61   const char* constraint;
     62   const media::AudioParameters::PlatformEffectsMask effect;
     63 } const kConstraintEffectMap[] = {
     64   { content::kMediaStreamAudioDucking,
     65     media::AudioParameters::DUCKING },
     66   { webrtc::MediaConstraintsInterface::kEchoCancellation,
     67     media::AudioParameters::ECHO_CANCELLER },
     68 };
     69 
     70 // If any platform effects are available, check them against the constraints.
     71 // Disable effects to match false constraints, but if a constraint is true, set
     72 // the constraint to false to later disable the software effect.
     73 //
     74 // This function may modify both |constraints| and |effects|.
     75 void HarmonizeConstraintsAndEffects(RTCMediaConstraints* constraints,
     76                                     int* effects) {
     77   if (*effects != media::AudioParameters::NO_EFFECTS) {
     78     for (size_t i = 0; i < ARRAYSIZE_UNSAFE(kConstraintEffectMap); ++i) {
     79       bool value;
     80       size_t is_mandatory = 0;
     81       if (!webrtc::FindConstraint(constraints,
     82                                   kConstraintEffectMap[i].constraint,
     83                                   &value,
     84                                   &is_mandatory) || !value) {
     85         // If the constraint is false, or does not exist, disable the platform
     86         // effect.
     87         *effects &= ~kConstraintEffectMap[i].effect;
     88         DVLOG(1) << "Disabling platform effect: "
     89                  << kConstraintEffectMap[i].effect;
     90       } else if (*effects & kConstraintEffectMap[i].effect) {
     91         // If the constraint is true, leave the platform effect enabled, and
     92         // set the constraint to false to later disable the software effect.
     93         if (is_mandatory) {
     94           constraints->AddMandatory(kConstraintEffectMap[i].constraint,
     95               webrtc::MediaConstraintsInterface::kValueFalse, true);
     96         } else {
     97           constraints->AddOptional(kConstraintEffectMap[i].constraint,
     98               webrtc::MediaConstraintsInterface::kValueFalse, true);
     99         }
    100         DVLOG(1) << "Disabling constraint: "
    101                  << kConstraintEffectMap[i].constraint;
    102       } else if (kConstraintEffectMap[i].effect ==
    103                  media::AudioParameters::DUCKING && value && !is_mandatory) {
    104         // Special handling of the DUCKING flag that sets the optional
    105         // constraint to |false| to match what the device will support.
    106         constraints->AddOptional(kConstraintEffectMap[i].constraint,
    107             webrtc::MediaConstraintsInterface::kValueFalse, true);
    108         // No need to modify |effects| since the ducking flag is already off.
    109         DCHECK((*effects & media::AudioParameters::DUCKING) == 0);
    110       }
    111     }
    112   }
    113 }
    114 
    115 class P2PPortAllocatorFactory : public webrtc::PortAllocatorFactoryInterface {
    116  public:
    117   P2PPortAllocatorFactory(
    118       P2PSocketDispatcher* socket_dispatcher,
    119       rtc::NetworkManager* network_manager,
    120       rtc::PacketSocketFactory* socket_factory,
    121       blink::WebFrame* web_frame)
    122       : socket_dispatcher_(socket_dispatcher),
    123         network_manager_(network_manager),
    124         socket_factory_(socket_factory),
    125         web_frame_(web_frame) {
    126   }
    127 
    128   virtual cricket::PortAllocator* CreatePortAllocator(
    129       const std::vector<StunConfiguration>& stun_servers,
    130       const std::vector<TurnConfiguration>& turn_configurations) OVERRIDE {
    131     CHECK(web_frame_);
    132     P2PPortAllocator::Config config;
    133     for (size_t i = 0; i < stun_servers.size(); ++i) {
    134       config.stun_servers.insert(rtc::SocketAddress(
    135           stun_servers[i].server.hostname(),
    136           stun_servers[i].server.port()));
    137     }
    138     config.legacy_relay = false;
    139     for (size_t i = 0; i < turn_configurations.size(); ++i) {
    140       P2PPortAllocator::Config::RelayServerConfig relay_config;
    141       relay_config.server_address = turn_configurations[i].server.hostname();
    142       relay_config.port = turn_configurations[i].server.port();
    143       relay_config.username = turn_configurations[i].username;
    144       relay_config.password = turn_configurations[i].password;
    145       relay_config.transport_type = turn_configurations[i].transport_type;
    146       relay_config.secure = turn_configurations[i].secure;
    147       config.relays.push_back(relay_config);
    148 
    149       // Use turn servers as stun servers.
    150       config.stun_servers.insert(rtc::SocketAddress(
    151           turn_configurations[i].server.hostname(),
    152           turn_configurations[i].server.port()));
    153     }
    154 
    155     return new P2PPortAllocator(
    156         web_frame_, socket_dispatcher_.get(), network_manager_,
    157         socket_factory_, config);
    158   }
    159 
    160  protected:
    161   virtual ~P2PPortAllocatorFactory() {}
    162 
    163  private:
    164   scoped_refptr<P2PSocketDispatcher> socket_dispatcher_;
    165   // |network_manager_| and |socket_factory_| are a weak references, owned by
    166   // PeerConnectionDependencyFactory.
    167   rtc::NetworkManager* network_manager_;
    168   rtc::PacketSocketFactory* socket_factory_;
    169   // Raw ptr to the WebFrame that created the P2PPortAllocatorFactory.
    170   blink::WebFrame* web_frame_;
    171 };
    172 
    173 PeerConnectionDependencyFactory::PeerConnectionDependencyFactory(
    174     P2PSocketDispatcher* p2p_socket_dispatcher)
    175     : network_manager_(NULL),
    176       p2p_socket_dispatcher_(p2p_socket_dispatcher),
    177       signaling_thread_(NULL),
    178       worker_thread_(NULL),
    179       chrome_worker_thread_("Chrome_libJingle_WorkerThread") {
    180 }
    181 
    182 PeerConnectionDependencyFactory::~PeerConnectionDependencyFactory() {
    183   CleanupPeerConnectionFactory();
    184   if (aec_dump_message_filter_.get())
    185     aec_dump_message_filter_->RemoveDelegate(this);
    186 }
    187 
    188 blink::WebRTCPeerConnectionHandler*
    189 PeerConnectionDependencyFactory::CreateRTCPeerConnectionHandler(
    190     blink::WebRTCPeerConnectionHandlerClient* client) {
    191   // Save histogram data so we can see how much PeerConnetion is used.
    192   // The histogram counts the number of calls to the JS API
    193   // webKitRTCPeerConnection.
    194   UpdateWebRTCMethodCount(WEBKIT_RTC_PEER_CONNECTION);
    195 
    196   return new RTCPeerConnectionHandler(client, this);
    197 }
    198 
    199 bool PeerConnectionDependencyFactory::InitializeMediaStreamAudioSource(
    200     int render_view_id,
    201     const blink::WebMediaConstraints& audio_constraints,
    202     MediaStreamAudioSource* source_data) {
    203   DVLOG(1) << "InitializeMediaStreamAudioSources()";
    204 
    205   // Do additional source initialization if the audio source is a valid
    206   // microphone or tab audio.
    207   RTCMediaConstraints native_audio_constraints(audio_constraints);
    208   MediaAudioConstraints::ApplyFixedAudioConstraints(&native_audio_constraints);
    209 
    210   StreamDeviceInfo device_info = source_data->device_info();
    211   RTCMediaConstraints constraints = native_audio_constraints;
    212   // May modify both |constraints| and |effects|.
    213   HarmonizeConstraintsAndEffects(&constraints,
    214                                  &device_info.device.input.effects);
    215 
    216   scoped_refptr<WebRtcAudioCapturer> capturer(
    217       CreateAudioCapturer(render_view_id, device_info, audio_constraints,
    218                           source_data));
    219   if (!capturer.get()) {
    220     const std::string log_string =
    221         "PCDF::InitializeMediaStreamAudioSource: fails to create capturer";
    222     WebRtcLogMessage(log_string);
    223     DVLOG(1) << log_string;
    224     // TODO(xians): Don't we need to check if source_observer is observing
    225     // something? If not, then it looks like we have a leak here.
    226     // OTOH, if it _is_ observing something, then the callback might
    227     // be called multiple times which is likely also a bug.
    228     return false;
    229   }
    230   source_data->SetAudioCapturer(capturer.get());
    231 
    232   // Creates a LocalAudioSource object which holds audio options.
    233   // TODO(xians): The option should apply to the track instead of the source.
    234   // TODO(perkj): Move audio constraints parsing to Chrome.
    235   // Currently there are a few constraints that are parsed by libjingle and
    236   // the state is set to ended if parsing fails.
    237   scoped_refptr<webrtc::AudioSourceInterface> rtc_source(
    238       CreateLocalAudioSource(&constraints).get());
    239   if (rtc_source->state() != webrtc::MediaSourceInterface::kLive) {
    240     DLOG(WARNING) << "Failed to create rtc LocalAudioSource.";
    241     return false;
    242   }
    243   source_data->SetLocalAudioSource(rtc_source.get());
    244   return true;
    245 }
    246 
    247 WebRtcVideoCapturerAdapter*
    248 PeerConnectionDependencyFactory::CreateVideoCapturer(
    249     bool is_screeencast) {
    250   // We need to make sure the libjingle thread wrappers have been created
    251   // before we can use an instance of a WebRtcVideoCapturerAdapter. This is
    252   // since the base class of WebRtcVideoCapturerAdapter is a
    253   // cricket::VideoCapturer and it uses the libjingle thread wrappers.
    254   if (!GetPcFactory().get())
    255     return NULL;
    256   return new WebRtcVideoCapturerAdapter(is_screeencast);
    257 }
    258 
    259 scoped_refptr<webrtc::VideoSourceInterface>
    260 PeerConnectionDependencyFactory::CreateVideoSource(
    261     cricket::VideoCapturer* capturer,
    262     const blink::WebMediaConstraints& constraints) {
    263   RTCMediaConstraints webrtc_constraints(constraints);
    264   scoped_refptr<webrtc::VideoSourceInterface> source =
    265       GetPcFactory()->CreateVideoSource(capturer, &webrtc_constraints).get();
    266   return source;
    267 }
    268 
    269 const scoped_refptr<webrtc::PeerConnectionFactoryInterface>&
    270 PeerConnectionDependencyFactory::GetPcFactory() {
    271   if (!pc_factory_.get())
    272     CreatePeerConnectionFactory();
    273   CHECK(pc_factory_.get());
    274   return pc_factory_;
    275 }
    276 
    277 void PeerConnectionDependencyFactory::CreatePeerConnectionFactory() {
    278   DCHECK(!pc_factory_.get());
    279   DCHECK(!signaling_thread_);
    280   DCHECK(!worker_thread_);
    281   DCHECK(!network_manager_);
    282   DCHECK(!socket_factory_);
    283   DCHECK(!chrome_worker_thread_.IsRunning());
    284 
    285   DVLOG(1) << "PeerConnectionDependencyFactory::CreatePeerConnectionFactory()";
    286 
    287   jingle_glue::JingleThreadWrapper::EnsureForCurrentMessageLoop();
    288   jingle_glue::JingleThreadWrapper::current()->set_send_allowed(true);
    289   signaling_thread_ = jingle_glue::JingleThreadWrapper::current();
    290   CHECK(signaling_thread_);
    291 
    292   CHECK(chrome_worker_thread_.Start());
    293 
    294   base::WaitableEvent start_worker_event(true, false);
    295   chrome_worker_thread_.message_loop()->PostTask(FROM_HERE, base::Bind(
    296       &PeerConnectionDependencyFactory::InitializeWorkerThread,
    297       base::Unretained(this),
    298       &worker_thread_,
    299       &start_worker_event));
    300   start_worker_event.Wait();
    301   CHECK(worker_thread_);
    302 
    303   base::WaitableEvent create_network_manager_event(true, false);
    304   chrome_worker_thread_.message_loop()->PostTask(FROM_HERE, base::Bind(
    305       &PeerConnectionDependencyFactory::CreateIpcNetworkManagerOnWorkerThread,
    306       base::Unretained(this),
    307       &create_network_manager_event));
    308   create_network_manager_event.Wait();
    309 
    310   socket_factory_.reset(
    311       new IpcPacketSocketFactory(p2p_socket_dispatcher_.get()));
    312 
    313   // Init SSL, which will be needed by PeerConnection.
    314 #if defined(USE_OPENSSL)
    315   if (!rtc::InitializeSSL()) {
    316     LOG(ERROR) << "Failed on InitializeSSL.";
    317     NOTREACHED();
    318     return;
    319   }
    320 #else
    321   // TODO(ronghuawu): Replace this call with InitializeSSL.
    322   net::EnsureNSSSSLInit();
    323 #endif
    324 
    325   scoped_ptr<cricket::WebRtcVideoDecoderFactory> decoder_factory;
    326   scoped_ptr<cricket::WebRtcVideoEncoderFactory> encoder_factory;
    327 
    328   const CommandLine* cmd_line = CommandLine::ForCurrentProcess();
    329   scoped_refptr<media::GpuVideoAcceleratorFactories> gpu_factories =
    330       RenderThreadImpl::current()->GetGpuFactories();
    331   if (!cmd_line->HasSwitch(switches::kDisableWebRtcHWDecoding)) {
    332     if (gpu_factories.get())
    333       decoder_factory.reset(new RTCVideoDecoderFactory(gpu_factories));
    334   }
    335 
    336   if (!cmd_line->HasSwitch(switches::kDisableWebRtcHWEncoding)) {
    337     if (gpu_factories.get())
    338       encoder_factory.reset(new RTCVideoEncoderFactory(gpu_factories));
    339   }
    340 
    341 #if defined(OS_ANDROID)
    342   if (!media::MediaCodecBridge::SupportsSetParameters())
    343     encoder_factory.reset();
    344 #endif
    345 
    346   EnsureWebRtcAudioDeviceImpl();
    347 
    348   scoped_refptr<webrtc::PeerConnectionFactoryInterface> factory(
    349       webrtc::CreatePeerConnectionFactory(worker_thread_,
    350                                           signaling_thread_,
    351                                           audio_device_.get(),
    352                                           encoder_factory.release(),
    353                                           decoder_factory.release()));
    354   CHECK(factory.get());
    355 
    356   pc_factory_ = factory;
    357   webrtc::PeerConnectionFactoryInterface::Options factory_options;
    358   factory_options.disable_sctp_data_channels = false;
    359   factory_options.disable_encryption =
    360       cmd_line->HasSwitch(switches::kDisableWebRtcEncryption);
    361   pc_factory_->SetOptions(factory_options);
    362 
    363   // TODO(xians): Remove the following code after kDisableAudioTrackProcessing
    364   // is removed.
    365   if (!MediaStreamAudioProcessor::IsAudioTrackProcessingEnabled()) {
    366     aec_dump_message_filter_ = AecDumpMessageFilter::Get();
    367     // In unit tests not creating a message filter, |aec_dump_message_filter_|
    368     // will be NULL. We can just ignore that. Other unit tests and browser tests
    369     // ensure that we do get the filter when we should.
    370     if (aec_dump_message_filter_.get())
    371       aec_dump_message_filter_->AddDelegate(this);
    372   }
    373 }
    374 
    375 bool PeerConnectionDependencyFactory::PeerConnectionFactoryCreated() {
    376   return pc_factory_.get() != NULL;
    377 }
    378 
    379 scoped_refptr<webrtc::PeerConnectionInterface>
    380 PeerConnectionDependencyFactory::CreatePeerConnection(
    381     const webrtc::PeerConnectionInterface::RTCConfiguration& config,
    382     const webrtc::MediaConstraintsInterface* constraints,
    383     blink::WebFrame* web_frame,
    384     webrtc::PeerConnectionObserver* observer) {
    385   CHECK(web_frame);
    386   CHECK(observer);
    387   if (!GetPcFactory().get())
    388     return NULL;
    389 
    390   scoped_refptr<P2PPortAllocatorFactory> pa_factory =
    391         new rtc::RefCountedObject<P2PPortAllocatorFactory>(
    392             p2p_socket_dispatcher_.get(),
    393             network_manager_,
    394             socket_factory_.get(),
    395             web_frame);
    396 
    397   PeerConnectionIdentityService* identity_service =
    398       new PeerConnectionIdentityService(
    399           GURL(web_frame->document().url().spec()).GetOrigin());
    400 
    401   return GetPcFactory()->CreatePeerConnection(config,
    402                                               constraints,
    403                                               pa_factory.get(),
    404                                               identity_service,
    405                                               observer).get();
    406 }
    407 
    408 scoped_refptr<webrtc::MediaStreamInterface>
    409 PeerConnectionDependencyFactory::CreateLocalMediaStream(
    410     const std::string& label) {
    411   return GetPcFactory()->CreateLocalMediaStream(label).get();
    412 }
    413 
    414 scoped_refptr<webrtc::AudioSourceInterface>
    415 PeerConnectionDependencyFactory::CreateLocalAudioSource(
    416     const webrtc::MediaConstraintsInterface* constraints) {
    417   scoped_refptr<webrtc::AudioSourceInterface> source =
    418       GetPcFactory()->CreateAudioSource(constraints).get();
    419   return source;
    420 }
    421 
    422 void PeerConnectionDependencyFactory::CreateLocalAudioTrack(
    423     const blink::WebMediaStreamTrack& track) {
    424   blink::WebMediaStreamSource source = track.source();
    425   DCHECK_EQ(source.type(), blink::WebMediaStreamSource::TypeAudio);
    426   MediaStreamAudioSource* source_data =
    427       static_cast<MediaStreamAudioSource*>(source.extraData());
    428 
    429   scoped_refptr<WebAudioCapturerSource> webaudio_source;
    430   if (!source_data) {
    431     if (source.requiresAudioConsumer()) {
    432       // We're adding a WebAudio MediaStream.
    433       // Create a specific capturer for each WebAudio consumer.
    434       webaudio_source = CreateWebAudioSource(&source);
    435       source_data =
    436           static_cast<MediaStreamAudioSource*>(source.extraData());
    437     } else {
    438       // TODO(perkj): Implement support for sources from
    439       // remote MediaStreams.
    440       NOTIMPLEMENTED();
    441       return;
    442     }
    443   }
    444 
    445   // Creates an adapter to hold all the libjingle objects.
    446   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter(
    447       WebRtcLocalAudioTrackAdapter::Create(track.id().utf8(),
    448                                            source_data->local_audio_source()));
    449   static_cast<webrtc::AudioTrackInterface*>(adapter.get())->set_enabled(
    450       track.isEnabled());
    451 
    452   // TODO(xians): Merge |source| to the capturer(). We can't do this today
    453   // because only one capturer() is supported while one |source| is created
    454   // for each audio track.
    455   scoped_ptr<WebRtcLocalAudioTrack> audio_track(new WebRtcLocalAudioTrack(
    456       adapter.get(), source_data->GetAudioCapturer(), webaudio_source.get()));
    457 
    458   StartLocalAudioTrack(audio_track.get());
    459 
    460   // Pass the ownership of the native local audio track to the blink track.
    461   blink::WebMediaStreamTrack writable_track = track;
    462   writable_track.setExtraData(audio_track.release());
    463 }
    464 
    465 void PeerConnectionDependencyFactory::StartLocalAudioTrack(
    466     WebRtcLocalAudioTrack* audio_track) {
    467   // Add the WebRtcAudioDevice as the sink to the local audio track.
    468   // TODO(xians): Remove the following line of code after the APM in WebRTC is
    469   // completely deprecated. See http://crbug/365672.
    470   if (!MediaStreamAudioProcessor::IsAudioTrackProcessingEnabled())
    471     audio_track->AddSink(GetWebRtcAudioDevice());
    472 
    473   // Start the audio track. This will hook the |audio_track| to the capturer
    474   // as the sink of the audio, and only start the source of the capturer if
    475   // it is the first audio track connecting to the capturer.
    476   audio_track->Start();
    477 }
    478 
    479 scoped_refptr<WebAudioCapturerSource>
    480 PeerConnectionDependencyFactory::CreateWebAudioSource(
    481     blink::WebMediaStreamSource* source) {
    482   DVLOG(1) << "PeerConnectionDependencyFactory::CreateWebAudioSource()";
    483 
    484   scoped_refptr<WebAudioCapturerSource>
    485       webaudio_capturer_source(new WebAudioCapturerSource());
    486   MediaStreamAudioSource* source_data = new MediaStreamAudioSource();
    487 
    488   // Use the current default capturer for the WebAudio track so that the
    489   // WebAudio track can pass a valid delay value and |need_audio_processing|
    490   // flag to PeerConnection.
    491   // TODO(xians): Remove this after moving APM to Chrome.
    492   if (GetWebRtcAudioDevice()) {
    493     source_data->SetAudioCapturer(
    494         GetWebRtcAudioDevice()->GetDefaultCapturer());
    495   }
    496 
    497   // Create a LocalAudioSource object which holds audio options.
    498   // SetLocalAudioSource() affects core audio parts in third_party/Libjingle.
    499   source_data->SetLocalAudioSource(CreateLocalAudioSource(NULL).get());
    500   source->setExtraData(source_data);
    501 
    502   // Replace the default source with WebAudio as source instead.
    503   source->addAudioConsumer(webaudio_capturer_source.get());
    504 
    505   return webaudio_capturer_source;
    506 }
    507 
    508 scoped_refptr<webrtc::VideoTrackInterface>
    509 PeerConnectionDependencyFactory::CreateLocalVideoTrack(
    510     const std::string& id,
    511     webrtc::VideoSourceInterface* source) {
    512   return GetPcFactory()->CreateVideoTrack(id, source).get();
    513 }
    514 
    515 scoped_refptr<webrtc::VideoTrackInterface>
    516 PeerConnectionDependencyFactory::CreateLocalVideoTrack(
    517     const std::string& id, cricket::VideoCapturer* capturer) {
    518   if (!capturer) {
    519     LOG(ERROR) << "CreateLocalVideoTrack called with null VideoCapturer.";
    520     return NULL;
    521   }
    522 
    523   // Create video source from the |capturer|.
    524   scoped_refptr<webrtc::VideoSourceInterface> source =
    525       GetPcFactory()->CreateVideoSource(capturer, NULL).get();
    526 
    527   // Create native track from the source.
    528   return GetPcFactory()->CreateVideoTrack(id, source.get()).get();
    529 }
    530 
    531 webrtc::SessionDescriptionInterface*
    532 PeerConnectionDependencyFactory::CreateSessionDescription(
    533     const std::string& type,
    534     const std::string& sdp,
    535     webrtc::SdpParseError* error) {
    536   return webrtc::CreateSessionDescription(type, sdp, error);
    537 }
    538 
    539 webrtc::IceCandidateInterface*
    540 PeerConnectionDependencyFactory::CreateIceCandidate(
    541     const std::string& sdp_mid,
    542     int sdp_mline_index,
    543     const std::string& sdp) {
    544   return webrtc::CreateIceCandidate(sdp_mid, sdp_mline_index, sdp);
    545 }
    546 
    547 WebRtcAudioDeviceImpl*
    548 PeerConnectionDependencyFactory::GetWebRtcAudioDevice() {
    549   return audio_device_.get();
    550 }
    551 
    552 void PeerConnectionDependencyFactory::InitializeWorkerThread(
    553     rtc::Thread** thread,
    554     base::WaitableEvent* event) {
    555   jingle_glue::JingleThreadWrapper::EnsureForCurrentMessageLoop();
    556   jingle_glue::JingleThreadWrapper::current()->set_send_allowed(true);
    557   *thread = jingle_glue::JingleThreadWrapper::current();
    558   event->Signal();
    559 }
    560 
    561 void PeerConnectionDependencyFactory::CreateIpcNetworkManagerOnWorkerThread(
    562     base::WaitableEvent* event) {
    563   DCHECK_EQ(base::MessageLoop::current(), chrome_worker_thread_.message_loop());
    564   network_manager_ = new IpcNetworkManager(p2p_socket_dispatcher_.get());
    565   event->Signal();
    566 }
    567 
    568 void PeerConnectionDependencyFactory::DeleteIpcNetworkManager() {
    569   DCHECK_EQ(base::MessageLoop::current(), chrome_worker_thread_.message_loop());
    570   delete network_manager_;
    571   network_manager_ = NULL;
    572 }
    573 
    574 void PeerConnectionDependencyFactory::CleanupPeerConnectionFactory() {
    575   pc_factory_ = NULL;
    576   if (network_manager_) {
    577     // The network manager needs to free its resources on the thread they were
    578     // created, which is the worked thread.
    579     if (chrome_worker_thread_.IsRunning()) {
    580       chrome_worker_thread_.message_loop()->PostTask(FROM_HERE, base::Bind(
    581           &PeerConnectionDependencyFactory::DeleteIpcNetworkManager,
    582           base::Unretained(this)));
    583       // Stopping the thread will wait until all tasks have been
    584       // processed before returning. We wait for the above task to finish before
    585       // letting the the function continue to avoid any potential race issues.
    586       chrome_worker_thread_.Stop();
    587     } else {
    588       NOTREACHED() << "Worker thread not running.";
    589     }
    590   }
    591 }
    592 
    593 scoped_refptr<WebRtcAudioCapturer>
    594 PeerConnectionDependencyFactory::CreateAudioCapturer(
    595     int render_view_id,
    596     const StreamDeviceInfo& device_info,
    597     const blink::WebMediaConstraints& constraints,
    598     MediaStreamAudioSource* audio_source) {
    599   // TODO(xians): Handle the cases when gUM is called without a proper render
    600   // view, for example, by an extension.
    601   DCHECK_GE(render_view_id, 0);
    602 
    603   EnsureWebRtcAudioDeviceImpl();
    604   DCHECK(GetWebRtcAudioDevice());
    605   return WebRtcAudioCapturer::CreateCapturer(render_view_id, device_info,
    606                                              constraints,
    607                                              GetWebRtcAudioDevice(),
    608                                              audio_source);
    609 }
    610 
    611 void PeerConnectionDependencyFactory::AddNativeAudioTrackToBlinkTrack(
    612     webrtc::MediaStreamTrackInterface* native_track,
    613     const blink::WebMediaStreamTrack& webkit_track,
    614     bool is_local_track) {
    615   DCHECK(!webkit_track.isNull() && !webkit_track.extraData());
    616   DCHECK_EQ(blink::WebMediaStreamSource::TypeAudio,
    617             webkit_track.source().type());
    618   blink::WebMediaStreamTrack track = webkit_track;
    619 
    620   DVLOG(1) << "AddNativeTrackToBlinkTrack() audio";
    621   track.setExtraData(
    622       new MediaStreamTrack(
    623           static_cast<webrtc::AudioTrackInterface*>(native_track),
    624           is_local_track));
    625 }
    626 
    627 scoped_refptr<base::MessageLoopProxy>
    628 PeerConnectionDependencyFactory::GetWebRtcWorkerThread() const {
    629   DCHECK(CalledOnValidThread());
    630   return chrome_worker_thread_.message_loop_proxy();
    631 }
    632 
    633 void PeerConnectionDependencyFactory::OnAecDumpFile(
    634     const IPC::PlatformFileForTransit& file_handle) {
    635   DCHECK(CalledOnValidThread());
    636   DCHECK(!MediaStreamAudioProcessor::IsAudioTrackProcessingEnabled());
    637   DCHECK(PeerConnectionFactoryCreated());
    638 
    639   base::File file = IPC::PlatformFileForTransitToFile(file_handle);
    640   DCHECK(file.IsValid());
    641 
    642   // |pc_factory_| always takes ownership of |aec_dump_file|. If StartAecDump()
    643   // fails, |aec_dump_file| will be closed.
    644   if (!GetPcFactory()->StartAecDump(file.TakePlatformFile()))
    645     VLOG(1) << "Could not start AEC dump.";
    646 }
    647 
    648 void PeerConnectionDependencyFactory::OnDisableAecDump() {
    649   DCHECK(CalledOnValidThread());
    650   DCHECK(!MediaStreamAudioProcessor::IsAudioTrackProcessingEnabled());
    651   // Do nothing. We never disable AEC dump for non-track-processing case.
    652 }
    653 
    654 void PeerConnectionDependencyFactory::OnIpcClosing() {
    655   DCHECK(CalledOnValidThread());
    656   aec_dump_message_filter_ = NULL;
    657 }
    658 
    659 void PeerConnectionDependencyFactory::EnsureWebRtcAudioDeviceImpl() {
    660   if (audio_device_.get())
    661     return;
    662 
    663   audio_device_ = new WebRtcAudioDeviceImpl();
    664 }
    665 
    666 }  // namespace content
    667