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