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| 1 // Copyright (c) 2012 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/media_stream_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_options.h" | |
| 16 #include "content/renderer/media/media_stream_audio_source.h" | |
| 17 #include "content/renderer/media/media_stream_video_source.h" | |
| 18 #include "content/renderer/media/media_stream_video_track.h" | |
| 19 #include "content/renderer/media/peer_connection_identity_service.h" | |
| 20 #include "content/renderer/media/rtc_media_constraints.h" | |
| 21 #include "content/renderer/media/rtc_peer_connection_handler.h" | |
| 22 #include "content/renderer/media/rtc_video_decoder_factory.h" | |
| 23 #include "content/renderer/media/rtc_video_encoder_factory.h" | |
| 24 #include "content/renderer/media/webaudio_capturer_source.h" | |
| 25 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h" | |
| 26 #include "content/renderer/media/webrtc/webrtc_video_capturer_adapter.h" | |
| 27 #include "content/renderer/media/webrtc_audio_device_impl.h" | |
| 28 #include "content/renderer/media/webrtc_local_audio_track.h" | |
| 29 #include "content/renderer/media/webrtc_uma_histograms.h" | |
| 30 #include "content/renderer/p2p/ipc_network_manager.h" | |
| 31 #include "content/renderer/p2p/ipc_socket_factory.h" | |
| 32 #include "content/renderer/p2p/port_allocator.h" | |
| 33 #include "content/renderer/render_thread_impl.h" | |
| 34 #include "jingle/glue/thread_wrapper.h" | |
| 35 #include "media/filters/gpu_video_accelerator_factories.h" | |
| 36 #include "third_party/WebKit/public/platform/WebMediaConstraints.h" | |
| 37 #include "third_party/WebKit/public/platform/WebMediaStream.h" | |
| 38 #include "third_party/WebKit/public/platform/WebMediaStreamSource.h" | |
| 39 #include "third_party/WebKit/public/platform/WebMediaStreamTrack.h" | |
| 40 #include "third_party/WebKit/public/platform/WebURL.h" | |
| 41 #include "third_party/WebKit/public/web/WebDocument.h" | |
| 42 #include "third_party/WebKit/public/web/WebFrame.h" | |
| 43 #include "third_party/libjingle/source/talk/app/webrtc/mediaconstraintsinterface
.h" | |
| 44 | |
| 45 #if defined(USE_OPENSSL) | |
| 46 #include "third_party/libjingle/source/talk/base/ssladapter.h" | |
| 47 #else | |
| 48 #include "net/socket/nss_ssl_util.h" | |
| 49 #endif | |
| 50 | |
| 51 #if defined(OS_ANDROID) | |
| 52 #include "media/base/android/media_codec_bridge.h" | |
| 53 #endif | |
| 54 | |
| 55 namespace content { | |
| 56 | |
| 57 // Map of corresponding media constraints and platform effects. | |
| 58 struct { | |
| 59 const char* constraint; | |
| 60 const media::AudioParameters::PlatformEffectsMask effect; | |
| 61 } const kConstraintEffectMap[] = { | |
| 62 { content::kMediaStreamAudioDucking, | |
| 63 media::AudioParameters::DUCKING }, | |
| 64 { webrtc::MediaConstraintsInterface::kEchoCancellation, | |
| 65 media::AudioParameters::ECHO_CANCELLER }, | |
| 66 }; | |
| 67 | |
| 68 // If any platform effects are available, check them against the constraints. | |
| 69 // Disable effects to match false constraints, but if a constraint is true, set | |
| 70 // the constraint to false to later disable the software effect. | |
| 71 // | |
| 72 // This function may modify both |constraints| and |effects|. | |
| 73 void HarmonizeConstraintsAndEffects(RTCMediaConstraints* constraints, | |
| 74 int* effects) { | |
| 75 if (*effects != media::AudioParameters::NO_EFFECTS) { | |
| 76 for (size_t i = 0; i < ARRAYSIZE_UNSAFE(kConstraintEffectMap); ++i) { | |
| 77 bool value; | |
| 78 size_t is_mandatory = 0; | |
| 79 if (!webrtc::FindConstraint(constraints, | |
| 80 kConstraintEffectMap[i].constraint, | |
| 81 &value, | |
| 82 &is_mandatory) || !value) { | |
| 83 // If the constraint is false, or does not exist, disable the platform | |
| 84 // effect. | |
| 85 *effects &= ~kConstraintEffectMap[i].effect; | |
| 86 DVLOG(1) << "Disabling platform effect: " | |
| 87 << kConstraintEffectMap[i].effect; | |
| 88 } else if (*effects & kConstraintEffectMap[i].effect) { | |
| 89 // If the constraint is true, leave the platform effect enabled, and | |
| 90 // set the constraint to false to later disable the software effect. | |
| 91 if (is_mandatory) { | |
| 92 constraints->AddMandatory(kConstraintEffectMap[i].constraint, | |
| 93 webrtc::MediaConstraintsInterface::kValueFalse, true); | |
| 94 } else { | |
| 95 constraints->AddOptional(kConstraintEffectMap[i].constraint, | |
| 96 webrtc::MediaConstraintsInterface::kValueFalse, true); | |
| 97 } | |
| 98 DVLOG(1) << "Disabling constraint: " | |
| 99 << kConstraintEffectMap[i].constraint; | |
| 100 } | |
| 101 } | |
| 102 } | |
| 103 } | |
| 104 | |
| 105 class P2PPortAllocatorFactory : public webrtc::PortAllocatorFactoryInterface { | |
| 106 public: | |
| 107 P2PPortAllocatorFactory( | |
| 108 P2PSocketDispatcher* socket_dispatcher, | |
| 109 talk_base::NetworkManager* network_manager, | |
| 110 talk_base::PacketSocketFactory* socket_factory, | |
| 111 blink::WebFrame* web_frame) | |
| 112 : socket_dispatcher_(socket_dispatcher), | |
| 113 network_manager_(network_manager), | |
| 114 socket_factory_(socket_factory), | |
| 115 web_frame_(web_frame) { | |
| 116 } | |
| 117 | |
| 118 virtual cricket::PortAllocator* CreatePortAllocator( | |
| 119 const std::vector<StunConfiguration>& stun_servers, | |
| 120 const std::vector<TurnConfiguration>& turn_configurations) OVERRIDE { | |
| 121 CHECK(web_frame_); | |
| 122 P2PPortAllocator::Config config; | |
| 123 if (stun_servers.size() > 0) { | |
| 124 config.stun_server = stun_servers[0].server.hostname(); | |
| 125 config.stun_server_port = stun_servers[0].server.port(); | |
| 126 } | |
| 127 config.legacy_relay = false; | |
| 128 for (size_t i = 0; i < turn_configurations.size(); ++i) { | |
| 129 P2PPortAllocator::Config::RelayServerConfig relay_config; | |
| 130 relay_config.server_address = turn_configurations[i].server.hostname(); | |
| 131 relay_config.port = turn_configurations[i].server.port(); | |
| 132 relay_config.username = turn_configurations[i].username; | |
| 133 relay_config.password = turn_configurations[i].password; | |
| 134 relay_config.transport_type = turn_configurations[i].transport_type; | |
| 135 relay_config.secure = turn_configurations[i].secure; | |
| 136 config.relays.push_back(relay_config); | |
| 137 } | |
| 138 | |
| 139 // Use first turn server as the stun server. | |
| 140 if (turn_configurations.size() > 0) { | |
| 141 config.stun_server = config.relays[0].server_address; | |
| 142 config.stun_server_port = config.relays[0].port; | |
| 143 } | |
| 144 | |
| 145 return new P2PPortAllocator( | |
| 146 web_frame_, socket_dispatcher_.get(), network_manager_, | |
| 147 socket_factory_, config); | |
| 148 } | |
| 149 | |
| 150 protected: | |
| 151 virtual ~P2PPortAllocatorFactory() {} | |
| 152 | |
| 153 private: | |
| 154 scoped_refptr<P2PSocketDispatcher> socket_dispatcher_; | |
| 155 // |network_manager_| and |socket_factory_| are a weak references, owned by | |
| 156 // MediaStreamDependencyFactory. | |
| 157 talk_base::NetworkManager* network_manager_; | |
| 158 talk_base::PacketSocketFactory* socket_factory_; | |
| 159 // Raw ptr to the WebFrame that created the P2PPortAllocatorFactory. | |
| 160 blink::WebFrame* web_frame_; | |
| 161 }; | |
| 162 | |
| 163 MediaStreamDependencyFactory::MediaStreamDependencyFactory( | |
| 164 P2PSocketDispatcher* p2p_socket_dispatcher) | |
| 165 : network_manager_(NULL), | |
| 166 p2p_socket_dispatcher_(p2p_socket_dispatcher), | |
| 167 signaling_thread_(NULL), | |
| 168 worker_thread_(NULL), | |
| 169 chrome_worker_thread_("Chrome_libJingle_WorkerThread") { | |
| 170 } | |
| 171 | |
| 172 MediaStreamDependencyFactory::~MediaStreamDependencyFactory() { | |
| 173 CleanupPeerConnectionFactory(); | |
| 174 } | |
| 175 | |
| 176 blink::WebRTCPeerConnectionHandler* | |
| 177 MediaStreamDependencyFactory::CreateRTCPeerConnectionHandler( | |
| 178 blink::WebRTCPeerConnectionHandlerClient* client) { | |
| 179 // Save histogram data so we can see how much PeerConnetion is used. | |
| 180 // The histogram counts the number of calls to the JS API | |
| 181 // webKitRTCPeerConnection. | |
| 182 UpdateWebRTCMethodCount(WEBKIT_RTC_PEER_CONNECTION); | |
| 183 | |
| 184 return new RTCPeerConnectionHandler(client, this); | |
| 185 } | |
| 186 | |
| 187 bool MediaStreamDependencyFactory::InitializeMediaStreamAudioSource( | |
| 188 int render_view_id, | |
| 189 const blink::WebMediaConstraints& audio_constraints, | |
| 190 MediaStreamAudioSource* source_data) { | |
| 191 DVLOG(1) << "InitializeMediaStreamAudioSources()"; | |
| 192 | |
| 193 // Do additional source initialization if the audio source is a valid | |
| 194 // microphone or tab audio. | |
| 195 RTCMediaConstraints native_audio_constraints(audio_constraints); | |
| 196 MediaAudioConstraints::ApplyFixedAudioConstraints(&native_audio_constraints); | |
| 197 | |
| 198 StreamDeviceInfo device_info = source_data->device_info(); | |
| 199 RTCMediaConstraints constraints = native_audio_constraints; | |
| 200 // May modify both |constraints| and |effects|. | |
| 201 HarmonizeConstraintsAndEffects(&constraints, | |
| 202 &device_info.device.input.effects); | |
| 203 | |
| 204 scoped_refptr<WebRtcAudioCapturer> capturer( | |
| 205 CreateAudioCapturer(render_view_id, device_info, audio_constraints, | |
| 206 source_data)); | |
| 207 if (!capturer.get()) { | |
| 208 DLOG(WARNING) << "Failed to create the capturer for device " | |
| 209 << device_info.device.id; | |
| 210 // TODO(xians): Don't we need to check if source_observer is observing | |
| 211 // something? If not, then it looks like we have a leak here. | |
| 212 // OTOH, if it _is_ observing something, then the callback might | |
| 213 // be called multiple times which is likely also a bug. | |
| 214 return false; | |
| 215 } | |
| 216 source_data->SetAudioCapturer(capturer); | |
| 217 | |
| 218 // Creates a LocalAudioSource object which holds audio options. | |
| 219 // TODO(xians): The option should apply to the track instead of the source. | |
| 220 // TODO(perkj): Move audio constraints parsing to Chrome. | |
| 221 // Currently there are a few constraints that are parsed by libjingle and | |
| 222 // the state is set to ended if parsing fails. | |
| 223 scoped_refptr<webrtc::AudioSourceInterface> rtc_source( | |
| 224 CreateLocalAudioSource(&constraints).get()); | |
| 225 if (rtc_source->state() != webrtc::MediaSourceInterface::kLive) { | |
| 226 DLOG(WARNING) << "Failed to create rtc LocalAudioSource."; | |
| 227 return false; | |
| 228 } | |
| 229 source_data->SetLocalAudioSource(rtc_source); | |
| 230 return true; | |
| 231 } | |
| 232 | |
| 233 WebRtcVideoCapturerAdapter* MediaStreamDependencyFactory::CreateVideoCapturer( | |
| 234 bool is_screeencast) { | |
| 235 // We need to make sure the libjingle thread wrappers have been created | |
| 236 // before we can use an instance of a WebRtcVideoCapturerAdapter. This is | |
| 237 // since the base class of WebRtcVideoCapturerAdapter is a | |
| 238 // cricket::VideoCapturer and it uses the libjingle thread wrappers. | |
| 239 if (!GetPcFactory()) | |
| 240 return NULL; | |
| 241 return new WebRtcVideoCapturerAdapter(is_screeencast); | |
| 242 } | |
| 243 | |
| 244 scoped_refptr<webrtc::VideoSourceInterface> | |
| 245 MediaStreamDependencyFactory::CreateVideoSource( | |
| 246 cricket::VideoCapturer* capturer, | |
| 247 const blink::WebMediaConstraints& constraints) { | |
| 248 RTCMediaConstraints webrtc_constraints(constraints); | |
| 249 scoped_refptr<webrtc::VideoSourceInterface> source = | |
| 250 GetPcFactory()->CreateVideoSource(capturer, &webrtc_constraints).get(); | |
| 251 return source; | |
| 252 } | |
| 253 | |
| 254 const scoped_refptr<webrtc::PeerConnectionFactoryInterface>& | |
| 255 MediaStreamDependencyFactory::GetPcFactory() { | |
| 256 if (!pc_factory_) | |
| 257 CreatePeerConnectionFactory(); | |
| 258 CHECK(pc_factory_); | |
| 259 return pc_factory_; | |
| 260 } | |
| 261 | |
| 262 void MediaStreamDependencyFactory::CreatePeerConnectionFactory() { | |
| 263 DCHECK(!pc_factory_.get()); | |
| 264 DCHECK(!signaling_thread_); | |
| 265 DCHECK(!worker_thread_); | |
| 266 DCHECK(!network_manager_); | |
| 267 DCHECK(!socket_factory_); | |
| 268 DCHECK(!chrome_worker_thread_.IsRunning()); | |
| 269 | |
| 270 DVLOG(1) << "MediaStreamDependencyFactory::CreatePeerConnectionFactory()"; | |
| 271 | |
| 272 jingle_glue::JingleThreadWrapper::EnsureForCurrentMessageLoop(); | |
| 273 jingle_glue::JingleThreadWrapper::current()->set_send_allowed(true); | |
| 274 signaling_thread_ = jingle_glue::JingleThreadWrapper::current(); | |
| 275 CHECK(signaling_thread_); | |
| 276 | |
| 277 CHECK(chrome_worker_thread_.Start()); | |
| 278 | |
| 279 base::WaitableEvent start_worker_event(true, false); | |
| 280 chrome_worker_thread_.message_loop()->PostTask(FROM_HERE, base::Bind( | |
| 281 &MediaStreamDependencyFactory::InitializeWorkerThread, | |
| 282 base::Unretained(this), | |
| 283 &worker_thread_, | |
| 284 &start_worker_event)); | |
| 285 start_worker_event.Wait(); | |
| 286 CHECK(worker_thread_); | |
| 287 | |
| 288 base::WaitableEvent create_network_manager_event(true, false); | |
| 289 chrome_worker_thread_.message_loop()->PostTask(FROM_HERE, base::Bind( | |
| 290 &MediaStreamDependencyFactory::CreateIpcNetworkManagerOnWorkerThread, | |
| 291 base::Unretained(this), | |
| 292 &create_network_manager_event)); | |
| 293 create_network_manager_event.Wait(); | |
| 294 | |
| 295 socket_factory_.reset( | |
| 296 new IpcPacketSocketFactory(p2p_socket_dispatcher_.get())); | |
| 297 | |
| 298 // Init SSL, which will be needed by PeerConnection. | |
| 299 #if defined(USE_OPENSSL) | |
| 300 if (!talk_base::InitializeSSL()) { | |
| 301 LOG(ERROR) << "Failed on InitializeSSL."; | |
| 302 NOTREACHED(); | |
| 303 return; | |
| 304 } | |
| 305 #else | |
| 306 // TODO(ronghuawu): Replace this call with InitializeSSL. | |
| 307 net::EnsureNSSSSLInit(); | |
| 308 #endif | |
| 309 | |
| 310 scoped_ptr<cricket::WebRtcVideoDecoderFactory> decoder_factory; | |
| 311 scoped_ptr<cricket::WebRtcVideoEncoderFactory> encoder_factory; | |
| 312 | |
| 313 const CommandLine* cmd_line = CommandLine::ForCurrentProcess(); | |
| 314 scoped_refptr<media::GpuVideoAcceleratorFactories> gpu_factories = | |
| 315 RenderThreadImpl::current()->GetGpuFactories(); | |
| 316 if (!cmd_line->HasSwitch(switches::kDisableWebRtcHWDecoding)) { | |
| 317 if (gpu_factories) | |
| 318 decoder_factory.reset(new RTCVideoDecoderFactory(gpu_factories)); | |
| 319 } | |
| 320 | |
| 321 if (!cmd_line->HasSwitch(switches::kDisableWebRtcHWEncoding)) { | |
| 322 if (gpu_factories) | |
| 323 encoder_factory.reset(new RTCVideoEncoderFactory(gpu_factories)); | |
| 324 } | |
| 325 | |
| 326 #if defined(OS_ANDROID) | |
| 327 if (!media::MediaCodecBridge::SupportsSetParameters()) | |
| 328 encoder_factory.reset(); | |
| 329 #endif | |
| 330 | |
| 331 EnsureWebRtcAudioDeviceImpl(); | |
| 332 | |
| 333 scoped_refptr<webrtc::PeerConnectionFactoryInterface> factory( | |
| 334 webrtc::CreatePeerConnectionFactory(worker_thread_, | |
| 335 signaling_thread_, | |
| 336 audio_device_.get(), | |
| 337 encoder_factory.release(), | |
| 338 decoder_factory.release())); | |
| 339 CHECK(factory); | |
| 340 | |
| 341 pc_factory_ = factory; | |
| 342 webrtc::PeerConnectionFactoryInterface::Options factory_options; | |
| 343 factory_options.disable_sctp_data_channels = false; | |
| 344 factory_options.disable_encryption = | |
| 345 cmd_line->HasSwitch(switches::kDisableWebRtcEncryption); | |
| 346 pc_factory_->SetOptions(factory_options); | |
| 347 | |
| 348 // |aec_dump_file| will be invalid when dump is not enabled. | |
| 349 if (aec_dump_file_.IsValid()) | |
| 350 StartAecDump(aec_dump_file_.Pass()); | |
| 351 } | |
| 352 | |
| 353 bool MediaStreamDependencyFactory::PeerConnectionFactoryCreated() { | |
| 354 return pc_factory_.get() != NULL; | |
| 355 } | |
| 356 | |
| 357 scoped_refptr<webrtc::PeerConnectionInterface> | |
| 358 MediaStreamDependencyFactory::CreatePeerConnection( | |
| 359 const webrtc::PeerConnectionInterface::IceServers& ice_servers, | |
| 360 const webrtc::MediaConstraintsInterface* constraints, | |
| 361 blink::WebFrame* web_frame, | |
| 362 webrtc::PeerConnectionObserver* observer) { | |
| 363 CHECK(web_frame); | |
| 364 CHECK(observer); | |
| 365 if (!GetPcFactory()) | |
| 366 return NULL; | |
| 367 | |
| 368 scoped_refptr<P2PPortAllocatorFactory> pa_factory = | |
| 369 new talk_base::RefCountedObject<P2PPortAllocatorFactory>( | |
| 370 p2p_socket_dispatcher_.get(), | |
| 371 network_manager_, | |
| 372 socket_factory_.get(), | |
| 373 web_frame); | |
| 374 | |
| 375 PeerConnectionIdentityService* identity_service = | |
| 376 new PeerConnectionIdentityService( | |
| 377 GURL(web_frame->document().url().spec()).GetOrigin()); | |
| 378 | |
| 379 return GetPcFactory()->CreatePeerConnection(ice_servers, | |
| 380 constraints, | |
| 381 pa_factory.get(), | |
| 382 identity_service, | |
| 383 observer).get(); | |
| 384 } | |
| 385 | |
| 386 scoped_refptr<webrtc::MediaStreamInterface> | |
| 387 MediaStreamDependencyFactory::CreateLocalMediaStream( | |
| 388 const std::string& label) { | |
| 389 return GetPcFactory()->CreateLocalMediaStream(label).get(); | |
| 390 } | |
| 391 | |
| 392 scoped_refptr<webrtc::AudioSourceInterface> | |
| 393 MediaStreamDependencyFactory::CreateLocalAudioSource( | |
| 394 const webrtc::MediaConstraintsInterface* constraints) { | |
| 395 scoped_refptr<webrtc::AudioSourceInterface> source = | |
| 396 GetPcFactory()->CreateAudioSource(constraints).get(); | |
| 397 return source; | |
| 398 } | |
| 399 | |
| 400 void MediaStreamDependencyFactory::CreateLocalAudioTrack( | |
| 401 const blink::WebMediaStreamTrack& track) { | |
| 402 blink::WebMediaStreamSource source = track.source(); | |
| 403 DCHECK_EQ(source.type(), blink::WebMediaStreamSource::TypeAudio); | |
| 404 MediaStreamAudioSource* source_data = | |
| 405 static_cast<MediaStreamAudioSource*>(source.extraData()); | |
| 406 | |
| 407 scoped_refptr<WebAudioCapturerSource> webaudio_source; | |
| 408 if (!source_data) { | |
| 409 if (source.requiresAudioConsumer()) { | |
| 410 // We're adding a WebAudio MediaStream. | |
| 411 // Create a specific capturer for each WebAudio consumer. | |
| 412 webaudio_source = CreateWebAudioSource(&source); | |
| 413 source_data = | |
| 414 static_cast<MediaStreamAudioSource*>(source.extraData()); | |
| 415 } else { | |
| 416 // TODO(perkj): Implement support for sources from | |
| 417 // remote MediaStreams. | |
| 418 NOTIMPLEMENTED(); | |
| 419 return; | |
| 420 } | |
| 421 } | |
| 422 | |
| 423 // Creates an adapter to hold all the libjingle objects. | |
| 424 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter( | |
| 425 WebRtcLocalAudioTrackAdapter::Create(track.id().utf8(), | |
| 426 source_data->local_audio_source())); | |
| 427 static_cast<webrtc::AudioTrackInterface*>(adapter.get())->set_enabled( | |
| 428 track.isEnabled()); | |
| 429 | |
| 430 // TODO(xians): Merge |source| to the capturer(). We can't do this today | |
| 431 // because only one capturer() is supported while one |source| is created | |
| 432 // for each audio track. | |
| 433 scoped_ptr<WebRtcLocalAudioTrack> audio_track( | |
| 434 new WebRtcLocalAudioTrack(adapter, | |
| 435 source_data->GetAudioCapturer(), | |
| 436 webaudio_source)); | |
| 437 | |
| 438 StartLocalAudioTrack(audio_track.get()); | |
| 439 | |
| 440 // Pass the ownership of the native local audio track to the blink track. | |
| 441 blink::WebMediaStreamTrack writable_track = track; | |
| 442 writable_track.setExtraData(audio_track.release()); | |
| 443 } | |
| 444 | |
| 445 void MediaStreamDependencyFactory::StartLocalAudioTrack( | |
| 446 WebRtcLocalAudioTrack* audio_track) { | |
| 447 // Add the WebRtcAudioDevice as the sink to the local audio track. | |
| 448 // TODO(xians): Implement a PeerConnection sink adapter and remove this | |
| 449 // AddSink() call. | |
| 450 audio_track->AddSink(GetWebRtcAudioDevice()); | |
| 451 // Start the audio track. This will hook the |audio_track| to the capturer | |
| 452 // as the sink of the audio, and only start the source of the capturer if | |
| 453 // it is the first audio track connecting to the capturer. | |
| 454 audio_track->Start(); | |
| 455 } | |
| 456 | |
| 457 scoped_refptr<WebAudioCapturerSource> | |
| 458 MediaStreamDependencyFactory::CreateWebAudioSource( | |
| 459 blink::WebMediaStreamSource* source) { | |
| 460 DVLOG(1) << "MediaStreamDependencyFactory::CreateWebAudioSource()"; | |
| 461 | |
| 462 scoped_refptr<WebAudioCapturerSource> | |
| 463 webaudio_capturer_source(new WebAudioCapturerSource()); | |
| 464 MediaStreamAudioSource* source_data = new MediaStreamAudioSource(); | |
| 465 | |
| 466 // Use the current default capturer for the WebAudio track so that the | |
| 467 // WebAudio track can pass a valid delay value and |need_audio_processing| | |
| 468 // flag to PeerConnection. | |
| 469 // TODO(xians): Remove this after moving APM to Chrome. | |
| 470 if (GetWebRtcAudioDevice()) { | |
| 471 source_data->SetAudioCapturer( | |
| 472 GetWebRtcAudioDevice()->GetDefaultCapturer()); | |
| 473 } | |
| 474 | |
| 475 // Create a LocalAudioSource object which holds audio options. | |
| 476 // SetLocalAudioSource() affects core audio parts in third_party/Libjingle. | |
| 477 source_data->SetLocalAudioSource(CreateLocalAudioSource(NULL).get()); | |
| 478 source->setExtraData(source_data); | |
| 479 | |
| 480 // Replace the default source with WebAudio as source instead. | |
| 481 source->addAudioConsumer(webaudio_capturer_source.get()); | |
| 482 | |
| 483 return webaudio_capturer_source; | |
| 484 } | |
| 485 | |
| 486 scoped_refptr<webrtc::VideoTrackInterface> | |
| 487 MediaStreamDependencyFactory::CreateLocalVideoTrack( | |
| 488 const std::string& id, | |
| 489 webrtc::VideoSourceInterface* source) { | |
| 490 return GetPcFactory()->CreateVideoTrack(id, source).get(); | |
| 491 } | |
| 492 | |
| 493 scoped_refptr<webrtc::VideoTrackInterface> | |
| 494 MediaStreamDependencyFactory::CreateLocalVideoTrack( | |
| 495 const std::string& id, cricket::VideoCapturer* capturer) { | |
| 496 if (!capturer) { | |
| 497 LOG(ERROR) << "CreateLocalVideoTrack called with null VideoCapturer."; | |
| 498 return NULL; | |
| 499 } | |
| 500 | |
| 501 // Create video source from the |capturer|. | |
| 502 scoped_refptr<webrtc::VideoSourceInterface> source = | |
| 503 GetPcFactory()->CreateVideoSource(capturer, NULL).get(); | |
| 504 | |
| 505 // Create native track from the source. | |
| 506 return GetPcFactory()->CreateVideoTrack(id, source.get()).get(); | |
| 507 } | |
| 508 | |
| 509 webrtc::SessionDescriptionInterface* | |
| 510 MediaStreamDependencyFactory::CreateSessionDescription( | |
| 511 const std::string& type, | |
| 512 const std::string& sdp, | |
| 513 webrtc::SdpParseError* error) { | |
| 514 return webrtc::CreateSessionDescription(type, sdp, error); | |
| 515 } | |
| 516 | |
| 517 webrtc::IceCandidateInterface* MediaStreamDependencyFactory::CreateIceCandidate( | |
| 518 const std::string& sdp_mid, | |
| 519 int sdp_mline_index, | |
| 520 const std::string& sdp) { | |
| 521 return webrtc::CreateIceCandidate(sdp_mid, sdp_mline_index, sdp); | |
| 522 } | |
| 523 | |
| 524 WebRtcAudioDeviceImpl* | |
| 525 MediaStreamDependencyFactory::GetWebRtcAudioDevice() { | |
| 526 return audio_device_.get(); | |
| 527 } | |
| 528 | |
| 529 void MediaStreamDependencyFactory::InitializeWorkerThread( | |
| 530 talk_base::Thread** thread, | |
| 531 base::WaitableEvent* event) { | |
| 532 jingle_glue::JingleThreadWrapper::EnsureForCurrentMessageLoop(); | |
| 533 jingle_glue::JingleThreadWrapper::current()->set_send_allowed(true); | |
| 534 *thread = jingle_glue::JingleThreadWrapper::current(); | |
| 535 event->Signal(); | |
| 536 } | |
| 537 | |
| 538 void MediaStreamDependencyFactory::CreateIpcNetworkManagerOnWorkerThread( | |
| 539 base::WaitableEvent* event) { | |
| 540 DCHECK_EQ(base::MessageLoop::current(), chrome_worker_thread_.message_loop()); | |
| 541 network_manager_ = new IpcNetworkManager(p2p_socket_dispatcher_.get()); | |
| 542 event->Signal(); | |
| 543 } | |
| 544 | |
| 545 void MediaStreamDependencyFactory::DeleteIpcNetworkManager() { | |
| 546 DCHECK_EQ(base::MessageLoop::current(), chrome_worker_thread_.message_loop()); | |
| 547 delete network_manager_; | |
| 548 network_manager_ = NULL; | |
| 549 } | |
| 550 | |
| 551 void MediaStreamDependencyFactory::CleanupPeerConnectionFactory() { | |
| 552 pc_factory_ = NULL; | |
| 553 if (network_manager_) { | |
| 554 // The network manager needs to free its resources on the thread they were | |
| 555 // created, which is the worked thread. | |
| 556 if (chrome_worker_thread_.IsRunning()) { | |
| 557 chrome_worker_thread_.message_loop()->PostTask(FROM_HERE, base::Bind( | |
| 558 &MediaStreamDependencyFactory::DeleteIpcNetworkManager, | |
| 559 base::Unretained(this))); | |
| 560 // Stopping the thread will wait until all tasks have been | |
| 561 // processed before returning. We wait for the above task to finish before | |
| 562 // letting the the function continue to avoid any potential race issues. | |
| 563 chrome_worker_thread_.Stop(); | |
| 564 } else { | |
| 565 NOTREACHED() << "Worker thread not running."; | |
| 566 } | |
| 567 } | |
| 568 } | |
| 569 | |
| 570 scoped_refptr<WebRtcAudioCapturer> | |
| 571 MediaStreamDependencyFactory::CreateAudioCapturer( | |
| 572 int render_view_id, | |
| 573 const StreamDeviceInfo& device_info, | |
| 574 const blink::WebMediaConstraints& constraints, | |
| 575 MediaStreamAudioSource* audio_source) { | |
| 576 // TODO(xians): Handle the cases when gUM is called without a proper render | |
| 577 // view, for example, by an extension. | |
| 578 DCHECK_GE(render_view_id, 0); | |
| 579 | |
| 580 EnsureWebRtcAudioDeviceImpl(); | |
| 581 DCHECK(GetWebRtcAudioDevice()); | |
| 582 return WebRtcAudioCapturer::CreateCapturer(render_view_id, device_info, | |
| 583 constraints, | |
| 584 GetWebRtcAudioDevice(), | |
| 585 audio_source); | |
| 586 } | |
| 587 | |
| 588 void MediaStreamDependencyFactory::AddNativeAudioTrackToBlinkTrack( | |
| 589 webrtc::MediaStreamTrackInterface* native_track, | |
| 590 const blink::WebMediaStreamTrack& webkit_track, | |
| 591 bool is_local_track) { | |
| 592 DCHECK(!webkit_track.isNull() && !webkit_track.extraData()); | |
| 593 DCHECK_EQ(blink::WebMediaStreamSource::TypeAudio, | |
| 594 webkit_track.source().type()); | |
| 595 blink::WebMediaStreamTrack track = webkit_track; | |
| 596 | |
| 597 DVLOG(1) << "AddNativeTrackToBlinkTrack() audio"; | |
| 598 track.setExtraData( | |
| 599 new MediaStreamTrack( | |
| 600 static_cast<webrtc::AudioTrackInterface*>(native_track), | |
| 601 is_local_track)); | |
| 602 } | |
| 603 | |
| 604 scoped_refptr<base::MessageLoopProxy> | |
| 605 MediaStreamDependencyFactory::GetWebRtcWorkerThread() const { | |
| 606 DCHECK(CalledOnValidThread()); | |
| 607 return chrome_worker_thread_.message_loop_proxy(); | |
| 608 } | |
| 609 | |
| 610 bool MediaStreamDependencyFactory::OnControlMessageReceived( | |
| 611 const IPC::Message& message) { | |
| 612 bool handled = true; | |
| 613 IPC_BEGIN_MESSAGE_MAP(MediaStreamDependencyFactory, message) | |
| 614 IPC_MESSAGE_HANDLER(MediaStreamMsg_EnableAecDump, OnAecDumpFile) | |
| 615 IPC_MESSAGE_HANDLER(MediaStreamMsg_DisableAecDump, OnDisableAecDump) | |
| 616 IPC_MESSAGE_UNHANDLED(handled = false) | |
| 617 IPC_END_MESSAGE_MAP() | |
| 618 return handled; | |
| 619 } | |
| 620 | |
| 621 void MediaStreamDependencyFactory::OnAecDumpFile( | |
| 622 IPC::PlatformFileForTransit file_handle) { | |
| 623 DCHECK(!aec_dump_file_.IsValid()); | |
| 624 base::File file = IPC::PlatformFileForTransitToFile(file_handle); | |
| 625 DCHECK(file.IsValid()); | |
| 626 | |
| 627 if (CommandLine::ForCurrentProcess()->HasSwitch( | |
| 628 switches::kEnableAudioTrackProcessing)) { | |
| 629 EnsureWebRtcAudioDeviceImpl(); | |
| 630 GetWebRtcAudioDevice()->EnableAecDump(file.Pass()); | |
| 631 return; | |
| 632 } | |
| 633 | |
| 634 // TODO(xians): Remove the following code after kEnableAudioTrackProcessing | |
| 635 // is removed. | |
| 636 if (PeerConnectionFactoryCreated()) | |
| 637 StartAecDump(file.Pass()); | |
| 638 else | |
| 639 aec_dump_file_ = file.Pass(); | |
| 640 } | |
| 641 | |
| 642 void MediaStreamDependencyFactory::OnDisableAecDump() { | |
| 643 if (CommandLine::ForCurrentProcess()->HasSwitch( | |
| 644 switches::kEnableAudioTrackProcessing)) { | |
| 645 GetWebRtcAudioDevice()->DisableAecDump(); | |
| 646 return; | |
| 647 } | |
| 648 | |
| 649 // TODO(xians): Remove the following code after kEnableAudioTrackProcessing | |
| 650 // is removed. | |
| 651 if (aec_dump_file_.IsValid()) | |
| 652 aec_dump_file_.Close(); | |
| 653 } | |
| 654 | |
| 655 void MediaStreamDependencyFactory::StartAecDump(base::File aec_dump_file) { | |
| 656 // |pc_factory_| always takes ownership of |aec_dump_file|. If StartAecDump() | |
| 657 // fails, |aec_dump_file| will be closed. | |
| 658 if (!GetPcFactory()->StartAecDump(aec_dump_file.TakePlatformFile())) | |
| 659 VLOG(1) << "Could not start AEC dump."; | |
| 660 } | |
| 661 | |
| 662 void MediaStreamDependencyFactory::EnsureWebRtcAudioDeviceImpl() { | |
| 663 if (audio_device_) | |
| 664 return; | |
| 665 | |
| 666 audio_device_ = new WebRtcAudioDeviceImpl(); | |
| 667 } | |
| 668 | |
| 669 } // namespace content | |
| OLD | NEW |
|---|
Chromium Code Reviews
Issue 272043003:
Renamed MediaStreamDependencyFactory to PeerConnectionDependencyFactory. (Closed)
Base URL: svn://svn.chromium.org/chrome/trunk/src