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.
5 #include "content/renderer/media/webrtc/peer_connection_dependency_factory.h"
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"
46 #if defined(USE_OPENSSL)
47 #include "third_party/webrtc/base/ssladapter.h"
49 #include "net/socket/nss_ssl_util.h"
52 #if defined(OS_ANDROID)
53 #include "media/base/android/media_codec_bridge.h"
58 // Map of corresponding media constraints and platform effects.
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 },
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.
73 // This function may modify both |constraints| and |effects|.
74 void HarmonizeConstraintsAndEffects(RTCMediaConstraints* constraints,
76 if (*effects != media::AudioParameters::NO_EFFECTS) {
77 for (size_t i = 0; i < ARRAYSIZE_UNSAFE(kConstraintEffectMap); ++i) {
79 size_t is_mandatory = 0;
80 if (!webrtc::FindConstraint(constraints,
81 kConstraintEffectMap[i].constraint,
83 &is_mandatory) || !value) {
84 // If the constraint is false, or does not exist, disable the platform
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.
93 constraints->AddMandatory(kConstraintEffectMap[i].constraint,
94 webrtc::MediaConstraintsInterface::kValueFalse, true);
96 constraints->AddOptional(kConstraintEffectMap[i].constraint,
97 webrtc::MediaConstraintsInterface::kValueFalse, true);
99 DVLOG(1) << "Disabling constraint: "
100 << kConstraintEffectMap[i].constraint;
101 } else if (kConstraintEffectMap[i].effect ==
102 media::AudioParameters::DUCKING && value && !is_mandatory) {
103 // Special handling of the DUCKING flag that sets the optional
104 // constraint to |false| to match what the device will support.
105 constraints->AddOptional(kConstraintEffectMap[i].constraint,
106 webrtc::MediaConstraintsInterface::kValueFalse, true);
107 // No need to modify |effects| since the ducking flag is already off.
108 DCHECK((*effects & media::AudioParameters::DUCKING) == 0);
114 class P2PPortAllocatorFactory : public webrtc::PortAllocatorFactoryInterface {
116 P2PPortAllocatorFactory(
117 P2PSocketDispatcher* socket_dispatcher,
118 rtc::NetworkManager* network_manager,
119 rtc::PacketSocketFactory* socket_factory,
120 blink::WebFrame* web_frame)
121 : socket_dispatcher_(socket_dispatcher),
122 network_manager_(network_manager),
123 socket_factory_(socket_factory),
124 web_frame_(web_frame) {
127 virtual cricket::PortAllocator* CreatePortAllocator(
128 const std::vector<StunConfiguration>& stun_servers,
129 const std::vector<TurnConfiguration>& turn_configurations) OVERRIDE {
131 P2PPortAllocator::Config config;
132 for (size_t i = 0; i < stun_servers.size(); ++i) {
133 config.stun_servers.insert(rtc::SocketAddress(
134 stun_servers[i].server.hostname(),
135 stun_servers[i].server.port()));
137 config.legacy_relay = false;
138 for (size_t i = 0; i < turn_configurations.size(); ++i) {
139 P2PPortAllocator::Config::RelayServerConfig relay_config;
140 relay_config.server_address = turn_configurations[i].server.hostname();
141 relay_config.port = turn_configurations[i].server.port();
142 relay_config.username = turn_configurations[i].username;
143 relay_config.password = turn_configurations[i].password;
144 relay_config.transport_type = turn_configurations[i].transport_type;
145 relay_config.secure = turn_configurations[i].secure;
146 config.relays.push_back(relay_config);
148 // Use turn servers as stun servers.
149 config.stun_servers.insert(rtc::SocketAddress(
150 turn_configurations[i].server.hostname(),
151 turn_configurations[i].server.port()));
154 return new P2PPortAllocator(
155 web_frame_, socket_dispatcher_.get(), network_manager_,
156 socket_factory_, config);
160 virtual ~P2PPortAllocatorFactory() {}
163 scoped_refptr<P2PSocketDispatcher> socket_dispatcher_;
164 // |network_manager_| and |socket_factory_| are a weak references, owned by
165 // PeerConnectionDependencyFactory.
166 rtc::NetworkManager* network_manager_;
167 rtc::PacketSocketFactory* socket_factory_;
168 // Raw ptr to the WebFrame that created the P2PPortAllocatorFactory.
169 blink::WebFrame* web_frame_;
172 PeerConnectionDependencyFactory::PeerConnectionDependencyFactory(
173 P2PSocketDispatcher* p2p_socket_dispatcher)
174 : network_manager_(NULL),
175 p2p_socket_dispatcher_(p2p_socket_dispatcher),
176 signaling_thread_(NULL),
177 worker_thread_(NULL),
178 chrome_worker_thread_("Chrome_libJingle_WorkerThread") {
181 PeerConnectionDependencyFactory::~PeerConnectionDependencyFactory() {
182 CleanupPeerConnectionFactory();
183 if (aec_dump_message_filter_)
184 aec_dump_message_filter_->RemoveDelegate(this);
187 blink::WebRTCPeerConnectionHandler*
188 PeerConnectionDependencyFactory::CreateRTCPeerConnectionHandler(
189 blink::WebRTCPeerConnectionHandlerClient* client) {
190 // Save histogram data so we can see how much PeerConnetion is used.
191 // The histogram counts the number of calls to the JS API
192 // webKitRTCPeerConnection.
193 UpdateWebRTCMethodCount(WEBKIT_RTC_PEER_CONNECTION);
195 return new RTCPeerConnectionHandler(client, this);
198 bool PeerConnectionDependencyFactory::InitializeMediaStreamAudioSource(
200 const blink::WebMediaConstraints& audio_constraints,
201 MediaStreamAudioSource* source_data) {
202 DVLOG(1) << "InitializeMediaStreamAudioSources()";
204 // Do additional source initialization if the audio source is a valid
205 // microphone or tab audio.
206 RTCMediaConstraints native_audio_constraints(audio_constraints);
207 MediaAudioConstraints::ApplyFixedAudioConstraints(&native_audio_constraints);
209 StreamDeviceInfo device_info = source_data->device_info();
210 RTCMediaConstraints constraints = native_audio_constraints;
211 // May modify both |constraints| and |effects|.
212 HarmonizeConstraintsAndEffects(&constraints,
213 &device_info.device.input.effects);
215 scoped_refptr<WebRtcAudioCapturer> capturer(
216 CreateAudioCapturer(render_view_id, device_info, audio_constraints,
218 if (!capturer.get()) {
219 DLOG(WARNING) << "Failed to create the capturer for device "
220 << device_info.device.id;
221 // TODO(xians): Don't we need to check if source_observer is observing
222 // something? If not, then it looks like we have a leak here.
223 // OTOH, if it _is_ observing something, then the callback might
224 // be called multiple times which is likely also a bug.
227 source_data->SetAudioCapturer(capturer);
229 // Creates a LocalAudioSource object which holds audio options.
230 // TODO(xians): The option should apply to the track instead of the source.
231 // TODO(perkj): Move audio constraints parsing to Chrome.
232 // Currently there are a few constraints that are parsed by libjingle and
233 // the state is set to ended if parsing fails.
234 scoped_refptr<webrtc::AudioSourceInterface> rtc_source(
235 CreateLocalAudioSource(&constraints).get());
236 if (rtc_source->state() != webrtc::MediaSourceInterface::kLive) {
237 DLOG(WARNING) << "Failed to create rtc LocalAudioSource.";
240 source_data->SetLocalAudioSource(rtc_source);
244 WebRtcVideoCapturerAdapter*
245 PeerConnectionDependencyFactory::CreateVideoCapturer(
246 bool is_screeencast) {
247 // We need to make sure the libjingle thread wrappers have been created
248 // before we can use an instance of a WebRtcVideoCapturerAdapter. This is
249 // since the base class of WebRtcVideoCapturerAdapter is a
250 // cricket::VideoCapturer and it uses the libjingle thread wrappers.
253 return new WebRtcVideoCapturerAdapter(is_screeencast);
256 scoped_refptr<webrtc::VideoSourceInterface>
257 PeerConnectionDependencyFactory::CreateVideoSource(
258 cricket::VideoCapturer* capturer,
259 const blink::WebMediaConstraints& constraints) {
260 RTCMediaConstraints webrtc_constraints(constraints);
261 scoped_refptr<webrtc::VideoSourceInterface> source =
262 GetPcFactory()->CreateVideoSource(capturer, &webrtc_constraints).get();
266 const scoped_refptr<webrtc::PeerConnectionFactoryInterface>&
267 PeerConnectionDependencyFactory::GetPcFactory() {
269 CreatePeerConnectionFactory();
274 void PeerConnectionDependencyFactory::CreatePeerConnectionFactory() {
275 DCHECK(!pc_factory_.get());
276 DCHECK(!signaling_thread_);
277 DCHECK(!worker_thread_);
278 DCHECK(!network_manager_);
279 DCHECK(!socket_factory_);
280 DCHECK(!chrome_worker_thread_.IsRunning());
282 DVLOG(1) << "PeerConnectionDependencyFactory::CreatePeerConnectionFactory()";
284 jingle_glue::JingleThreadWrapper::EnsureForCurrentMessageLoop();
285 jingle_glue::JingleThreadWrapper::current()->set_send_allowed(true);
286 signaling_thread_ = jingle_glue::JingleThreadWrapper::current();
287 CHECK(signaling_thread_);
289 CHECK(chrome_worker_thread_.Start());
291 base::WaitableEvent start_worker_event(true, false);
292 chrome_worker_thread_.message_loop()->PostTask(FROM_HERE, base::Bind(
293 &PeerConnectionDependencyFactory::InitializeWorkerThread,
294 base::Unretained(this),
296 &start_worker_event));
297 start_worker_event.Wait();
298 CHECK(worker_thread_);
300 base::WaitableEvent create_network_manager_event(true, false);
301 chrome_worker_thread_.message_loop()->PostTask(FROM_HERE, base::Bind(
302 &PeerConnectionDependencyFactory::CreateIpcNetworkManagerOnWorkerThread,
303 base::Unretained(this),
304 &create_network_manager_event));
305 create_network_manager_event.Wait();
307 socket_factory_.reset(
308 new IpcPacketSocketFactory(p2p_socket_dispatcher_.get()));
310 // Init SSL, which will be needed by PeerConnection.
311 #if defined(USE_OPENSSL)
312 if (!rtc::InitializeSSL()) {
313 LOG(ERROR) << "Failed on InitializeSSL.";
318 // TODO(ronghuawu): Replace this call with InitializeSSL.
319 net::EnsureNSSSSLInit();
322 scoped_ptr<cricket::WebRtcVideoDecoderFactory> decoder_factory;
323 scoped_ptr<cricket::WebRtcVideoEncoderFactory> encoder_factory;
325 const CommandLine* cmd_line = CommandLine::ForCurrentProcess();
326 scoped_refptr<media::GpuVideoAcceleratorFactories> gpu_factories =
327 RenderThreadImpl::current()->GetGpuFactories();
328 if (!cmd_line->HasSwitch(switches::kDisableWebRtcHWDecoding)) {
330 decoder_factory.reset(new RTCVideoDecoderFactory(gpu_factories));
333 if (!cmd_line->HasSwitch(switches::kDisableWebRtcHWEncoding)) {
335 encoder_factory.reset(new RTCVideoEncoderFactory(gpu_factories));
338 #if defined(OS_ANDROID)
339 if (!media::MediaCodecBridge::SupportsSetParameters())
340 encoder_factory.reset();
343 EnsureWebRtcAudioDeviceImpl();
345 scoped_refptr<webrtc::PeerConnectionFactoryInterface> factory(
346 webrtc::CreatePeerConnectionFactory(worker_thread_,
349 encoder_factory.release(),
350 decoder_factory.release()));
353 pc_factory_ = factory;
354 webrtc::PeerConnectionFactoryInterface::Options factory_options;
355 factory_options.disable_sctp_data_channels = false;
356 factory_options.disable_encryption =
357 cmd_line->HasSwitch(switches::kDisableWebRtcEncryption);
358 pc_factory_->SetOptions(factory_options);
360 // TODO(xians): Remove the following code after kDisableAudioTrackProcessing
362 if (!MediaStreamAudioProcessor::IsAudioTrackProcessingEnabled()) {
363 aec_dump_message_filter_ = AecDumpMessageFilter::Get();
364 // In unit tests not creating a message filter, |aec_dump_message_filter_|
365 // will be NULL. We can just ignore that. Other unit tests and browser tests
366 // ensure that we do get the filter when we should.
367 if (aec_dump_message_filter_)
368 aec_dump_message_filter_->AddDelegate(this);
372 bool PeerConnectionDependencyFactory::PeerConnectionFactoryCreated() {
373 return pc_factory_.get() != NULL;
376 scoped_refptr<webrtc::PeerConnectionInterface>
377 PeerConnectionDependencyFactory::CreatePeerConnection(
378 const webrtc::PeerConnectionInterface::RTCConfiguration& config,
379 const webrtc::MediaConstraintsInterface* constraints,
380 blink::WebFrame* web_frame,
381 webrtc::PeerConnectionObserver* observer) {
387 scoped_refptr<P2PPortAllocatorFactory> pa_factory =
388 new rtc::RefCountedObject<P2PPortAllocatorFactory>(
389 p2p_socket_dispatcher_.get(),
391 socket_factory_.get(),
394 PeerConnectionIdentityService* identity_service =
395 new PeerConnectionIdentityService(
396 GURL(web_frame->document().url().spec()).GetOrigin());
398 return GetPcFactory()->CreatePeerConnection(config,
405 scoped_refptr<webrtc::MediaStreamInterface>
406 PeerConnectionDependencyFactory::CreateLocalMediaStream(
407 const std::string& label) {
408 return GetPcFactory()->CreateLocalMediaStream(label).get();
411 scoped_refptr<webrtc::AudioSourceInterface>
412 PeerConnectionDependencyFactory::CreateLocalAudioSource(
413 const webrtc::MediaConstraintsInterface* constraints) {
414 scoped_refptr<webrtc::AudioSourceInterface> source =
415 GetPcFactory()->CreateAudioSource(constraints).get();
419 void PeerConnectionDependencyFactory::CreateLocalAudioTrack(
420 const blink::WebMediaStreamTrack& track) {
421 blink::WebMediaStreamSource source = track.source();
422 DCHECK_EQ(source.type(), blink::WebMediaStreamSource::TypeAudio);
423 MediaStreamAudioSource* source_data =
424 static_cast<MediaStreamAudioSource*>(source.extraData());
426 scoped_refptr<WebAudioCapturerSource> webaudio_source;
428 if (source.requiresAudioConsumer()) {
429 // We're adding a WebAudio MediaStream.
430 // Create a specific capturer for each WebAudio consumer.
431 webaudio_source = CreateWebAudioSource(&source);
433 static_cast<MediaStreamAudioSource*>(source.extraData());
435 // TODO(perkj): Implement support for sources from
436 // remote MediaStreams.
442 // Creates an adapter to hold all the libjingle objects.
443 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter(
444 WebRtcLocalAudioTrackAdapter::Create(track.id().utf8(),
445 source_data->local_audio_source()));
446 static_cast<webrtc::AudioTrackInterface*>(adapter.get())->set_enabled(
449 // TODO(xians): Merge |source| to the capturer(). We can't do this today
450 // because only one capturer() is supported while one |source| is created
451 // for each audio track.
452 scoped_ptr<WebRtcLocalAudioTrack> audio_track(
453 new WebRtcLocalAudioTrack(adapter,
454 source_data->GetAudioCapturer(),
457 StartLocalAudioTrack(audio_track.get());
459 // Pass the ownership of the native local audio track to the blink track.
460 blink::WebMediaStreamTrack writable_track = track;
461 writable_track.setExtraData(audio_track.release());
464 void PeerConnectionDependencyFactory::StartLocalAudioTrack(
465 WebRtcLocalAudioTrack* audio_track) {
466 // Add the WebRtcAudioDevice as the sink to the local audio track.
467 // TODO(xians): Remove the following line of code after the APM in WebRTC is
468 // completely deprecated. See http://crbug/365672.
469 if (!MediaStreamAudioProcessor::IsAudioTrackProcessingEnabled())
470 audio_track->AddSink(GetWebRtcAudioDevice());
472 // Start the audio track. This will hook the |audio_track| to the capturer
473 // as the sink of the audio, and only start the source of the capturer if
474 // it is the first audio track connecting to the capturer.
475 audio_track->Start();
478 scoped_refptr<WebAudioCapturerSource>
479 PeerConnectionDependencyFactory::CreateWebAudioSource(
480 blink::WebMediaStreamSource* source) {
481 DVLOG(1) << "PeerConnectionDependencyFactory::CreateWebAudioSource()";
483 scoped_refptr<WebAudioCapturerSource>
484 webaudio_capturer_source(new WebAudioCapturerSource());
485 MediaStreamAudioSource* source_data = new MediaStreamAudioSource();
487 // Use the current default capturer for the WebAudio track so that the
488 // WebAudio track can pass a valid delay value and |need_audio_processing|
489 // flag to PeerConnection.
490 // TODO(xians): Remove this after moving APM to Chrome.
491 if (GetWebRtcAudioDevice()) {
492 source_data->SetAudioCapturer(
493 GetWebRtcAudioDevice()->GetDefaultCapturer());
496 // Create a LocalAudioSource object which holds audio options.
497 // SetLocalAudioSource() affects core audio parts in third_party/Libjingle.
498 source_data->SetLocalAudioSource(CreateLocalAudioSource(NULL).get());
499 source->setExtraData(source_data);
501 // Replace the default source with WebAudio as source instead.
502 source->addAudioConsumer(webaudio_capturer_source.get());
504 return webaudio_capturer_source;
507 scoped_refptr<webrtc::VideoTrackInterface>
508 PeerConnectionDependencyFactory::CreateLocalVideoTrack(
509 const std::string& id,
510 webrtc::VideoSourceInterface* source) {
511 return GetPcFactory()->CreateVideoTrack(id, source).get();
514 scoped_refptr<webrtc::VideoTrackInterface>
515 PeerConnectionDependencyFactory::CreateLocalVideoTrack(
516 const std::string& id, cricket::VideoCapturer* capturer) {
518 LOG(ERROR) << "CreateLocalVideoTrack called with null VideoCapturer.";
522 // Create video source from the |capturer|.
523 scoped_refptr<webrtc::VideoSourceInterface> source =
524 GetPcFactory()->CreateVideoSource(capturer, NULL).get();
526 // Create native track from the source.
527 return GetPcFactory()->CreateVideoTrack(id, source.get()).get();
530 webrtc::SessionDescriptionInterface*
531 PeerConnectionDependencyFactory::CreateSessionDescription(
532 const std::string& type,
533 const std::string& sdp,
534 webrtc::SdpParseError* error) {
535 return webrtc::CreateSessionDescription(type, sdp, error);
538 webrtc::IceCandidateInterface*
539 PeerConnectionDependencyFactory::CreateIceCandidate(
540 const std::string& sdp_mid,
542 const std::string& sdp) {
543 return webrtc::CreateIceCandidate(sdp_mid, sdp_mline_index, sdp);
546 WebRtcAudioDeviceImpl*
547 PeerConnectionDependencyFactory::GetWebRtcAudioDevice() {
548 return audio_device_.get();
551 void PeerConnectionDependencyFactory::InitializeWorkerThread(
552 rtc::Thread** thread,
553 base::WaitableEvent* event) {
554 jingle_glue::JingleThreadWrapper::EnsureForCurrentMessageLoop();
555 jingle_glue::JingleThreadWrapper::current()->set_send_allowed(true);
556 *thread = jingle_glue::JingleThreadWrapper::current();
560 void PeerConnectionDependencyFactory::CreateIpcNetworkManagerOnWorkerThread(
561 base::WaitableEvent* event) {
562 DCHECK_EQ(base::MessageLoop::current(), chrome_worker_thread_.message_loop());
563 network_manager_ = new IpcNetworkManager(p2p_socket_dispatcher_.get());
567 void PeerConnectionDependencyFactory::DeleteIpcNetworkManager() {
568 DCHECK_EQ(base::MessageLoop::current(), chrome_worker_thread_.message_loop());
569 delete network_manager_;
570 network_manager_ = NULL;
573 void PeerConnectionDependencyFactory::CleanupPeerConnectionFactory() {
575 if (network_manager_) {
576 // The network manager needs to free its resources on the thread they were
577 // created, which is the worked thread.
578 if (chrome_worker_thread_.IsRunning()) {
579 chrome_worker_thread_.message_loop()->PostTask(FROM_HERE, base::Bind(
580 &PeerConnectionDependencyFactory::DeleteIpcNetworkManager,
581 base::Unretained(this)));
582 // Stopping the thread will wait until all tasks have been
583 // processed before returning. We wait for the above task to finish before
584 // letting the the function continue to avoid any potential race issues.
585 chrome_worker_thread_.Stop();
587 NOTREACHED() << "Worker thread not running.";
592 scoped_refptr<WebRtcAudioCapturer>
593 PeerConnectionDependencyFactory::CreateAudioCapturer(
595 const StreamDeviceInfo& device_info,
596 const blink::WebMediaConstraints& constraints,
597 MediaStreamAudioSource* audio_source) {
598 // TODO(xians): Handle the cases when gUM is called without a proper render
599 // view, for example, by an extension.
600 DCHECK_GE(render_view_id, 0);
602 EnsureWebRtcAudioDeviceImpl();
603 DCHECK(GetWebRtcAudioDevice());
604 return WebRtcAudioCapturer::CreateCapturer(render_view_id, device_info,
606 GetWebRtcAudioDevice(),
610 void PeerConnectionDependencyFactory::AddNativeAudioTrackToBlinkTrack(
611 webrtc::MediaStreamTrackInterface* native_track,
612 const blink::WebMediaStreamTrack& webkit_track,
613 bool is_local_track) {
614 DCHECK(!webkit_track.isNull() && !webkit_track.extraData());
615 DCHECK_EQ(blink::WebMediaStreamSource::TypeAudio,
616 webkit_track.source().type());
617 blink::WebMediaStreamTrack track = webkit_track;
619 DVLOG(1) << "AddNativeTrackToBlinkTrack() audio";
621 new MediaStreamTrack(
622 static_cast<webrtc::AudioTrackInterface*>(native_track),
626 scoped_refptr<base::MessageLoopProxy>
627 PeerConnectionDependencyFactory::GetWebRtcWorkerThread() const {
628 DCHECK(CalledOnValidThread());
629 return chrome_worker_thread_.message_loop_proxy();
632 void PeerConnectionDependencyFactory::OnAecDumpFile(
633 const IPC::PlatformFileForTransit& file_handle) {
634 DCHECK(CalledOnValidThread());
635 DCHECK(!MediaStreamAudioProcessor::IsAudioTrackProcessingEnabled());
636 DCHECK(PeerConnectionFactoryCreated());
638 base::File file = IPC::PlatformFileForTransitToFile(file_handle);
639 DCHECK(file.IsValid());
641 // |pc_factory_| always takes ownership of |aec_dump_file|. If StartAecDump()
642 // fails, |aec_dump_file| will be closed.
643 if (!GetPcFactory()->StartAecDump(file.TakePlatformFile()))
644 VLOG(1) << "Could not start AEC dump.";
647 void PeerConnectionDependencyFactory::OnDisableAecDump() {
648 DCHECK(CalledOnValidThread());
649 DCHECK(!MediaStreamAudioProcessor::IsAudioTrackProcessingEnabled());
650 // Do nothing. We never disable AEC dump for non-track-processing case.
653 void PeerConnectionDependencyFactory::OnIpcClosing() {
654 DCHECK(CalledOnValidThread());
655 aec_dump_message_filter_ = NULL;
658 void PeerConnectionDependencyFactory::EnsureWebRtcAudioDeviceImpl() {
662 audio_device_ = new WebRtcAudioDeviceImpl();
665 } // namespace content