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
57namespace content {
58
59// Map of corresponding media constraints and platform effects.
60struct {
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|.
75void 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
115class 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
173PeerConnectionDependencyFactory::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
182PeerConnectionDependencyFactory::~PeerConnectionDependencyFactory() {
183  CleanupPeerConnectionFactory();
184  if (aec_dump_message_filter_.get())
185    aec_dump_message_filter_->RemoveDelegate(this);
186}
187
188blink::WebRTCPeerConnectionHandler*
189PeerConnectionDependencyFactory::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
199bool 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
247WebRtcVideoCapturerAdapter*
248PeerConnectionDependencyFactory::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
259scoped_refptr<webrtc::VideoSourceInterface>
260PeerConnectionDependencyFactory::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
269const scoped_refptr<webrtc::PeerConnectionFactoryInterface>&
270PeerConnectionDependencyFactory::GetPcFactory() {
271  if (!pc_factory_.get())
272    CreatePeerConnectionFactory();
273  CHECK(pc_factory_.get());
274  return pc_factory_;
275}
276
277void 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
375bool PeerConnectionDependencyFactory::PeerConnectionFactoryCreated() {
376  return pc_factory_.get() != NULL;
377}
378
379scoped_refptr<webrtc::PeerConnectionInterface>
380PeerConnectionDependencyFactory::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
408scoped_refptr<webrtc::MediaStreamInterface>
409PeerConnectionDependencyFactory::CreateLocalMediaStream(
410    const std::string& label) {
411  return GetPcFactory()->CreateLocalMediaStream(label).get();
412}
413
414scoped_refptr<webrtc::AudioSourceInterface>
415PeerConnectionDependencyFactory::CreateLocalAudioSource(
416    const webrtc::MediaConstraintsInterface* constraints) {
417  scoped_refptr<webrtc::AudioSourceInterface> source =
418      GetPcFactory()->CreateAudioSource(constraints).get();
419  return source;
420}
421
422void 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
465void 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
479scoped_refptr<WebAudioCapturerSource>
480PeerConnectionDependencyFactory::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
508scoped_refptr<webrtc::VideoTrackInterface>
509PeerConnectionDependencyFactory::CreateLocalVideoTrack(
510    const std::string& id,
511    webrtc::VideoSourceInterface* source) {
512  return GetPcFactory()->CreateVideoTrack(id, source).get();
513}
514
515scoped_refptr<webrtc::VideoTrackInterface>
516PeerConnectionDependencyFactory::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
531webrtc::SessionDescriptionInterface*
532PeerConnectionDependencyFactory::CreateSessionDescription(
533    const std::string& type,
534    const std::string& sdp,
535    webrtc::SdpParseError* error) {
536  return webrtc::CreateSessionDescription(type, sdp, error);
537}
538
539webrtc::IceCandidateInterface*
540PeerConnectionDependencyFactory::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
547WebRtcAudioDeviceImpl*
548PeerConnectionDependencyFactory::GetWebRtcAudioDevice() {
549  return audio_device_.get();
550}
551
552void 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
561void 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
568void PeerConnectionDependencyFactory::DeleteIpcNetworkManager() {
569  DCHECK_EQ(base::MessageLoop::current(), chrome_worker_thread_.message_loop());
570  delete network_manager_;
571  network_manager_ = NULL;
572}
573
574void 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
593scoped_refptr<WebRtcAudioCapturer>
594PeerConnectionDependencyFactory::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
611void 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
627scoped_refptr<base::MessageLoopProxy>
628PeerConnectionDependencyFactory::GetWebRtcWorkerThread() const {
629  DCHECK(CalledOnValidThread());
630  return chrome_worker_thread_.message_loop_proxy();
631}
632
633void 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
648void PeerConnectionDependencyFactory::OnDisableAecDump() {
649  DCHECK(CalledOnValidThread());
650  DCHECK(!MediaStreamAudioProcessor::IsAudioTrackProcessingEnabled());
651  // Do nothing. We never disable AEC dump for non-track-processing case.
652}
653
654void PeerConnectionDependencyFactory::OnIpcClosing() {
655  DCHECK(CalledOnValidThread());
656  aec_dump_message_filter_ = NULL;
657}
658
659void PeerConnectionDependencyFactory::EnsureWebRtcAudioDeviceImpl() {
660  if (audio_device_.get())
661    return;
662
663  audio_device_ = new WebRtcAudioDeviceImpl();
664}
665
666}  // namespace content
667