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1 /* | 1 /* |
2 * Copyright 2004 The WebRTC project authors. All Rights Reserved. | 2 * Copyright 2004 The WebRTC project authors. All Rights Reserved. |
3 * | 3 * |
4 * Use of this source code is governed by a BSD-style license | 4 * Use of this source code is governed by a BSD-style license |
5 * that can be found in the LICENSE file in the root of the source | 5 * that can be found in the LICENSE file in the root of the source |
6 * tree. An additional intellectual property rights grant can be found | 6 * tree. An additional intellectual property rights grant can be found |
7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
9 */ | 9 */ |
10 | 10 |
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211 DestroyVoiceChannel_w(voice_channels_.back()); | 211 DestroyVoiceChannel_w(voice_channels_.back()); |
212 } | 212 } |
213 } | 213 } |
214 | 214 |
215 VoiceChannel* ChannelManager::CreateVoiceChannel( | 215 VoiceChannel* ChannelManager::CreateVoiceChannel( |
216 webrtc::MediaControllerInterface* media_controller, | 216 webrtc::MediaControllerInterface* media_controller, |
217 TransportController* transport_controller, | 217 TransportController* transport_controller, |
218 const std::string& content_name, | 218 const std::string& content_name, |
219 const std::string* bundle_transport_name, | 219 const std::string* bundle_transport_name, |
220 bool rtcp, | 220 bool rtcp, |
| 221 bool srtp_required, |
221 const AudioOptions& options) { | 222 const AudioOptions& options) { |
222 return worker_thread_->Invoke<VoiceChannel*>( | 223 return worker_thread_->Invoke<VoiceChannel*>( |
223 RTC_FROM_HERE, Bind(&ChannelManager::CreateVoiceChannel_w, this, | 224 RTC_FROM_HERE, Bind(&ChannelManager::CreateVoiceChannel_w, this, |
224 media_controller, transport_controller, content_name, | 225 media_controller, transport_controller, content_name, |
225 bundle_transport_name, rtcp, options)); | 226 bundle_transport_name, rtcp, srtp_required, options)); |
226 } | 227 } |
227 | 228 |
228 VoiceChannel* ChannelManager::CreateVoiceChannel_w( | 229 VoiceChannel* ChannelManager::CreateVoiceChannel_w( |
229 webrtc::MediaControllerInterface* media_controller, | 230 webrtc::MediaControllerInterface* media_controller, |
230 TransportController* transport_controller, | 231 TransportController* transport_controller, |
231 const std::string& content_name, | 232 const std::string& content_name, |
232 const std::string* bundle_transport_name, | 233 const std::string* bundle_transport_name, |
233 bool rtcp, | 234 bool rtcp, |
| 235 bool srtp_required, |
234 const AudioOptions& options) { | 236 const AudioOptions& options) { |
235 ASSERT(initialized_); | 237 ASSERT(initialized_); |
236 ASSERT(worker_thread_ == rtc::Thread::Current()); | 238 ASSERT(worker_thread_ == rtc::Thread::Current()); |
237 ASSERT(nullptr != media_controller); | 239 ASSERT(nullptr != media_controller); |
238 VoiceMediaChannel* media_channel = media_engine_->CreateChannel( | 240 VoiceMediaChannel* media_channel = media_engine_->CreateChannel( |
239 media_controller->call_w(), media_controller->config(), options); | 241 media_controller->call_w(), media_controller->config(), options); |
240 if (!media_channel) | 242 if (!media_channel) |
241 return nullptr; | 243 return nullptr; |
242 | 244 |
243 VoiceChannel* voice_channel = | 245 VoiceChannel* voice_channel = new VoiceChannel( |
244 new VoiceChannel(worker_thread_, network_thread_, media_engine_.get(), | 246 worker_thread_, network_thread_, media_engine_.get(), media_channel, |
245 media_channel, transport_controller, content_name, rtcp); | 247 transport_controller, content_name, rtcp, srtp_required); |
246 voice_channel->SetCryptoOptions(crypto_options_); | 248 voice_channel->SetCryptoOptions(crypto_options_); |
247 if (!voice_channel->Init_w(bundle_transport_name)) { | 249 if (!voice_channel->Init_w(bundle_transport_name)) { |
248 delete voice_channel; | 250 delete voice_channel; |
249 return nullptr; | 251 return nullptr; |
250 } | 252 } |
251 voice_channels_.push_back(voice_channel); | 253 voice_channels_.push_back(voice_channel); |
252 return voice_channel; | 254 return voice_channel; |
253 } | 255 } |
254 | 256 |
255 void ChannelManager::DestroyVoiceChannel(VoiceChannel* voice_channel) { | 257 void ChannelManager::DestroyVoiceChannel(VoiceChannel* voice_channel) { |
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274 voice_channels_.erase(it); | 276 voice_channels_.erase(it); |
275 delete voice_channel; | 277 delete voice_channel; |
276 } | 278 } |
277 | 279 |
278 VideoChannel* ChannelManager::CreateVideoChannel( | 280 VideoChannel* ChannelManager::CreateVideoChannel( |
279 webrtc::MediaControllerInterface* media_controller, | 281 webrtc::MediaControllerInterface* media_controller, |
280 TransportController* transport_controller, | 282 TransportController* transport_controller, |
281 const std::string& content_name, | 283 const std::string& content_name, |
282 const std::string* bundle_transport_name, | 284 const std::string* bundle_transport_name, |
283 bool rtcp, | 285 bool rtcp, |
| 286 bool srtp_required, |
284 const VideoOptions& options) { | 287 const VideoOptions& options) { |
285 return worker_thread_->Invoke<VideoChannel*>( | 288 return worker_thread_->Invoke<VideoChannel*>( |
286 RTC_FROM_HERE, Bind(&ChannelManager::CreateVideoChannel_w, this, | 289 RTC_FROM_HERE, Bind(&ChannelManager::CreateVideoChannel_w, this, |
287 media_controller, transport_controller, content_name, | 290 media_controller, transport_controller, content_name, |
288 bundle_transport_name, rtcp, options)); | 291 bundle_transport_name, rtcp, srtp_required, options)); |
289 } | 292 } |
290 | 293 |
291 VideoChannel* ChannelManager::CreateVideoChannel_w( | 294 VideoChannel* ChannelManager::CreateVideoChannel_w( |
292 webrtc::MediaControllerInterface* media_controller, | 295 webrtc::MediaControllerInterface* media_controller, |
293 TransportController* transport_controller, | 296 TransportController* transport_controller, |
294 const std::string& content_name, | 297 const std::string& content_name, |
295 const std::string* bundle_transport_name, | 298 const std::string* bundle_transport_name, |
296 bool rtcp, | 299 bool rtcp, |
| 300 bool srtp_required, |
297 const VideoOptions& options) { | 301 const VideoOptions& options) { |
298 ASSERT(initialized_); | 302 ASSERT(initialized_); |
299 ASSERT(worker_thread_ == rtc::Thread::Current()); | 303 ASSERT(worker_thread_ == rtc::Thread::Current()); |
300 ASSERT(nullptr != media_controller); | 304 ASSERT(nullptr != media_controller); |
301 VideoMediaChannel* media_channel = media_engine_->CreateVideoChannel( | 305 VideoMediaChannel* media_channel = media_engine_->CreateVideoChannel( |
302 media_controller->call_w(), media_controller->config(), options); | 306 media_controller->call_w(), media_controller->config(), options); |
303 if (media_channel == NULL) { | 307 if (media_channel == NULL) { |
304 return NULL; | 308 return NULL; |
305 } | 309 } |
306 | 310 |
307 VideoChannel* video_channel = | 311 VideoChannel* video_channel = |
308 new VideoChannel(worker_thread_, network_thread_, media_channel, | 312 new VideoChannel(worker_thread_, network_thread_, media_channel, |
309 transport_controller, content_name, rtcp); | 313 transport_controller, content_name, rtcp, srtp_required); |
310 video_channel->SetCryptoOptions(crypto_options_); | 314 video_channel->SetCryptoOptions(crypto_options_); |
311 if (!video_channel->Init_w(bundle_transport_name)) { | 315 if (!video_channel->Init_w(bundle_transport_name)) { |
312 delete video_channel; | 316 delete video_channel; |
313 return NULL; | 317 return NULL; |
314 } | 318 } |
315 video_channels_.push_back(video_channel); | 319 video_channels_.push_back(video_channel); |
316 return video_channel; | 320 return video_channel; |
317 } | 321 } |
318 | 322 |
319 void ChannelManager::DestroyVideoChannel(VideoChannel* video_channel) { | 323 void ChannelManager::DestroyVideoChannel(VideoChannel* video_channel) { |
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338 | 342 |
339 video_channels_.erase(it); | 343 video_channels_.erase(it); |
340 delete video_channel; | 344 delete video_channel; |
341 } | 345 } |
342 | 346 |
343 DataChannel* ChannelManager::CreateDataChannel( | 347 DataChannel* ChannelManager::CreateDataChannel( |
344 TransportController* transport_controller, | 348 TransportController* transport_controller, |
345 const std::string& content_name, | 349 const std::string& content_name, |
346 const std::string* bundle_transport_name, | 350 const std::string* bundle_transport_name, |
347 bool rtcp, | 351 bool rtcp, |
| 352 bool srtp_required, |
348 DataChannelType channel_type) { | 353 DataChannelType channel_type) { |
349 return worker_thread_->Invoke<DataChannel*>( | 354 return worker_thread_->Invoke<DataChannel*>( |
350 RTC_FROM_HERE, | 355 RTC_FROM_HERE, |
351 Bind(&ChannelManager::CreateDataChannel_w, this, transport_controller, | 356 Bind(&ChannelManager::CreateDataChannel_w, this, transport_controller, |
352 content_name, bundle_transport_name, rtcp, channel_type)); | 357 content_name, bundle_transport_name, rtcp, srtp_required, |
| 358 channel_type)); |
353 } | 359 } |
354 | 360 |
355 DataChannel* ChannelManager::CreateDataChannel_w( | 361 DataChannel* ChannelManager::CreateDataChannel_w( |
356 TransportController* transport_controller, | 362 TransportController* transport_controller, |
357 const std::string& content_name, | 363 const std::string& content_name, |
358 const std::string* bundle_transport_name, | 364 const std::string* bundle_transport_name, |
359 bool rtcp, | 365 bool rtcp, |
| 366 bool srtp_required, |
360 DataChannelType data_channel_type) { | 367 DataChannelType data_channel_type) { |
361 // This is ok to alloc from a thread other than the worker thread. | 368 // This is ok to alloc from a thread other than the worker thread. |
362 ASSERT(initialized_); | 369 ASSERT(initialized_); |
363 DataMediaChannel* media_channel = data_media_engine_->CreateChannel( | 370 DataMediaChannel* media_channel = data_media_engine_->CreateChannel( |
364 data_channel_type); | 371 data_channel_type); |
365 if (!media_channel) { | 372 if (!media_channel) { |
366 LOG(LS_WARNING) << "Failed to create data channel of type " | 373 LOG(LS_WARNING) << "Failed to create data channel of type " |
367 << data_channel_type; | 374 << data_channel_type; |
368 return NULL; | 375 return NULL; |
369 } | 376 } |
370 | 377 |
| 378 // Only RTP data channels need SRTP. |
| 379 srtp_required = srtp_required && data_channel_type == DCT_RTP; |
371 DataChannel* data_channel = | 380 DataChannel* data_channel = |
372 new DataChannel(worker_thread_, network_thread_, media_channel, | 381 new DataChannel(worker_thread_, network_thread_, media_channel, |
373 transport_controller, content_name, rtcp); | 382 transport_controller, content_name, rtcp, srtp_required); |
374 data_channel->SetCryptoOptions(crypto_options_); | 383 data_channel->SetCryptoOptions(crypto_options_); |
375 if (!data_channel->Init_w(bundle_transport_name)) { | 384 if (!data_channel->Init_w(bundle_transport_name)) { |
376 LOG(LS_WARNING) << "Failed to init data channel."; | 385 LOG(LS_WARNING) << "Failed to init data channel."; |
377 delete data_channel; | 386 delete data_channel; |
378 return NULL; | 387 return NULL; |
379 } | 388 } |
380 data_channels_.push_back(data_channel); | 389 data_channels_.push_back(data_channel); |
381 return data_channel; | 390 return data_channel; |
382 } | 391 } |
383 | 392 |
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411 media_engine_.get(), file, max_size_bytes)); | 420 media_engine_.get(), file, max_size_bytes)); |
412 } | 421 } |
413 | 422 |
414 void ChannelManager::StopAecDump() { | 423 void ChannelManager::StopAecDump() { |
415 worker_thread_->Invoke<void>( | 424 worker_thread_->Invoke<void>( |
416 RTC_FROM_HERE, | 425 RTC_FROM_HERE, |
417 Bind(&MediaEngineInterface::StopAecDump, media_engine_.get())); | 426 Bind(&MediaEngineInterface::StopAecDump, media_engine_.get())); |
418 } | 427 } |
419 | 428 |
420 } // namespace cricket | 429 } // namespace cricket |
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