OLD | NEW |
1 /* | 1 /* |
2 * Copyright 2015 The WebRTC project authors. All Rights Reserved. | 2 * Copyright 2015 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 |
(...skipping 262 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
273 VideoTrackInterface* video_track = static_cast<VideoTrackInterface*>(track); | 273 VideoTrackInterface* video_track = static_cast<VideoTrackInterface*>(track); |
274 | 274 |
275 // Detach from old track. | 275 // Detach from old track. |
276 if (track_) { | 276 if (track_) { |
277 track_->UnregisterObserver(this); | 277 track_->UnregisterObserver(this); |
278 } | 278 } |
279 | 279 |
280 // Attach to new track. | 280 // Attach to new track. |
281 bool prev_can_send_track = can_send_track(); | 281 bool prev_can_send_track = can_send_track(); |
282 // Keep a reference to the old track to keep it alive until we call | 282 // Keep a reference to the old track to keep it alive until we call |
283 // SetSource. | 283 // SetVideoSend. |
284 rtc::scoped_refptr<VideoTrackInterface> old_track = track_; | 284 rtc::scoped_refptr<VideoTrackInterface> old_track = track_; |
285 track_ = video_track; | 285 track_ = video_track; |
286 if (track_) { | 286 if (track_) { |
287 cached_track_enabled_ = track_->enabled(); | 287 cached_track_enabled_ = track_->enabled(); |
288 track_->RegisterObserver(this); | 288 track_->RegisterObserver(this); |
289 } | 289 } |
290 | 290 |
291 // Update video provider. | 291 // Update video provider. |
292 if (can_send_track()) { | 292 if (can_send_track()) { |
293 // TODO(deadbeef): If SetTrack is called with a disabled track, and the | |
294 // previous track was enabled, this could cause a frame from the new track | |
295 // to slip out. Really, what we need is for SetSource and SetVideoSend | |
296 // to be combined into one atomic operation, all the way down to | |
297 // WebRtcVideoSendStream. | |
298 | |
299 provider_->SetSource(ssrc_, track_); | |
300 SetVideoSend(); | 293 SetVideoSend(); |
301 } else if (prev_can_send_track) { | 294 } else if (prev_can_send_track) { |
302 provider_->SetSource(ssrc_, nullptr); | 295 provider_->SetVideoSend(ssrc_, false, nullptr, nullptr); |
303 provider_->SetVideoSend(ssrc_, false, nullptr); | |
304 } | 296 } |
305 return true; | 297 return true; |
306 } | 298 } |
307 | 299 |
308 void VideoRtpSender::SetSsrc(uint32_t ssrc) { | 300 void VideoRtpSender::SetSsrc(uint32_t ssrc) { |
309 TRACE_EVENT0("webrtc", "VideoRtpSender::SetSsrc"); | 301 TRACE_EVENT0("webrtc", "VideoRtpSender::SetSsrc"); |
310 if (stopped_ || ssrc == ssrc_) { | 302 if (stopped_ || ssrc == ssrc_) { |
311 return; | 303 return; |
312 } | 304 } |
313 // If we are already sending with a particular SSRC, stop sending. | 305 // If we are already sending with a particular SSRC, stop sending. |
314 if (can_send_track()) { | 306 if (can_send_track()) { |
315 provider_->SetSource(ssrc_, nullptr); | 307 provider_->SetVideoSend(ssrc_, false, nullptr, nullptr); |
316 provider_->SetVideoSend(ssrc_, false, nullptr); | |
317 } | 308 } |
318 ssrc_ = ssrc; | 309 ssrc_ = ssrc; |
319 if (can_send_track()) { | 310 if (can_send_track()) { |
320 provider_->SetSource(ssrc_, track_); | |
321 SetVideoSend(); | 311 SetVideoSend(); |
322 } | 312 } |
323 } | 313 } |
324 | 314 |
325 void VideoRtpSender::Stop() { | 315 void VideoRtpSender::Stop() { |
326 TRACE_EVENT0("webrtc", "VideoRtpSender::Stop"); | 316 TRACE_EVENT0("webrtc", "VideoRtpSender::Stop"); |
327 // TODO(deadbeef): Need to do more here to fully stop sending packets. | 317 // TODO(deadbeef): Need to do more here to fully stop sending packets. |
328 if (stopped_) { | 318 if (stopped_) { |
329 return; | 319 return; |
330 } | 320 } |
331 if (track_) { | 321 if (track_) { |
332 track_->UnregisterObserver(this); | 322 track_->UnregisterObserver(this); |
333 } | 323 } |
334 if (can_send_track()) { | 324 if (can_send_track()) { |
335 provider_->SetSource(ssrc_, nullptr); | 325 provider_->SetVideoSend(ssrc_, false, nullptr, nullptr); |
336 provider_->SetVideoSend(ssrc_, false, nullptr); | |
337 } | 326 } |
338 stopped_ = true; | 327 stopped_ = true; |
339 } | 328 } |
340 | 329 |
341 void VideoRtpSender::SetVideoSend() { | 330 void VideoRtpSender::SetVideoSend() { |
342 RTC_DCHECK(!stopped_ && can_send_track()); | 331 RTC_DCHECK(!stopped_ && can_send_track()); |
343 cricket::VideoOptions options; | 332 cricket::VideoOptions options; |
344 VideoTrackSourceInterface* source = track_->GetSource(); | 333 VideoTrackSourceInterface* source = track_->GetSource(); |
345 if (source) { | 334 if (source) { |
346 options.is_screencast = rtc::Optional<bool>(source->is_screencast()); | 335 options.is_screencast = rtc::Optional<bool>(source->is_screencast()); |
347 options.video_noise_reduction = source->needs_denoising(); | 336 options.video_noise_reduction = source->needs_denoising(); |
348 } | 337 } |
349 provider_->SetVideoSend(ssrc_, track_->enabled(), &options); | 338 provider_->SetVideoSend(ssrc_, track_->enabled(), &options, track_); |
350 } | 339 } |
351 | 340 |
352 RtpParameters VideoRtpSender::GetParameters() const { | 341 RtpParameters VideoRtpSender::GetParameters() const { |
353 return provider_->GetVideoRtpParameters(ssrc_); | 342 return provider_->GetVideoRtpParameters(ssrc_); |
354 } | 343 } |
355 | 344 |
356 bool VideoRtpSender::SetParameters(const RtpParameters& parameters) { | 345 bool VideoRtpSender::SetParameters(const RtpParameters& parameters) { |
357 TRACE_EVENT0("webrtc", "VideoRtpSender::SetParameters"); | 346 TRACE_EVENT0("webrtc", "VideoRtpSender::SetParameters"); |
358 return provider_->SetVideoRtpParameters(ssrc_, parameters); | 347 return provider_->SetVideoRtpParameters(ssrc_, parameters); |
359 } | 348 } |
360 | 349 |
361 } // namespace webrtc | 350 } // namespace webrtc |
OLD | NEW |