Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(1511)

Side by Side Diff: webrtc/pc/rtpsender.cc

Issue 2988153003: Replace CHECK(x && y) with two separate CHECK() calls (Closed)
Patch Set: fix mistakes Created 3 years, 4 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
« no previous file with comments | « webrtc/pc/rtpreceiver.cc ('k') | webrtc/pc/test/fakedatachannelprovider.h » ('j') | no next file with comments »
Toggle Intra-line Diffs ('i') | Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
OLDNEW
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 225 matching lines...) Expand 10 before | Expand all | Expand 10 after
236 if (can_send_track()) { 236 if (can_send_track()) {
237 ClearAudioSend(); 237 ClearAudioSend();
238 if (stats_) { 238 if (stats_) {
239 stats_->RemoveLocalAudioTrack(track_.get(), ssrc_); 239 stats_->RemoveLocalAudioTrack(track_.get(), ssrc_);
240 } 240 }
241 } 241 }
242 stopped_ = true; 242 stopped_ = true;
243 } 243 }
244 244
245 void AudioRtpSender::SetAudioSend() { 245 void AudioRtpSender::SetAudioSend() {
246 RTC_DCHECK(!stopped_ && can_send_track()); 246 RTC_DCHECK(!stopped_);
247 RTC_DCHECK(can_send_track());
247 if (!channel_) { 248 if (!channel_) {
248 LOG(LS_ERROR) << "SetAudioSend: No audio channel exists."; 249 LOG(LS_ERROR) << "SetAudioSend: No audio channel exists.";
249 return; 250 return;
250 } 251 }
251 cricket::AudioOptions options; 252 cricket::AudioOptions options;
252 #if !defined(WEBRTC_CHROMIUM_BUILD) && !defined(WEBRTC_WEBKIT_BUILD) 253 #if !defined(WEBRTC_CHROMIUM_BUILD) && !defined(WEBRTC_WEBKIT_BUILD)
253 // TODO(tommi): Remove this hack when we move CreateAudioSource out of 254 // TODO(tommi): Remove this hack when we move CreateAudioSource out of
254 // PeerConnection. This is a bit of a strange way to apply local audio 255 // PeerConnection. This is a bit of a strange way to apply local audio
255 // options since it is also applied to all streams/channels, local or remote. 256 // options since it is also applied to all streams/channels, local or remote.
256 if (track_->enabled() && track_->GetSource() && 257 if (track_->enabled() && track_->GetSource() &&
(...skipping 166 matching lines...) Expand 10 before | Expand all | Expand 10 after
423 if (track_) { 424 if (track_) {
424 track_->UnregisterObserver(this); 425 track_->UnregisterObserver(this);
425 } 426 }
426 if (can_send_track()) { 427 if (can_send_track()) {
427 ClearVideoSend(); 428 ClearVideoSend();
428 } 429 }
429 stopped_ = true; 430 stopped_ = true;
430 } 431 }
431 432
432 void VideoRtpSender::SetVideoSend() { 433 void VideoRtpSender::SetVideoSend() {
433 RTC_DCHECK(!stopped_ && can_send_track()); 434 RTC_DCHECK(!stopped_);
435 RTC_DCHECK(can_send_track());
434 if (!channel_) { 436 if (!channel_) {
435 LOG(LS_ERROR) << "SetVideoSend: No video channel exists."; 437 LOG(LS_ERROR) << "SetVideoSend: No video channel exists.";
436 return; 438 return;
437 } 439 }
438 cricket::VideoOptions options; 440 cricket::VideoOptions options;
439 VideoTrackSourceInterface* source = track_->GetSource(); 441 VideoTrackSourceInterface* source = track_->GetSource();
440 if (source) { 442 if (source) {
441 options.is_screencast = rtc::Optional<bool>(source->is_screencast()); 443 options.is_screencast = rtc::Optional<bool>(source->is_screencast());
442 options.video_noise_reduction = source->needs_denoising(); 444 options.video_noise_reduction = source->needs_denoising();
443 } 445 }
(...skipping 19 matching lines...) Expand all
463 LOG(LS_WARNING) << "SetVideoSend: No video channel exists."; 465 LOG(LS_WARNING) << "SetVideoSend: No video channel exists.";
464 return; 466 return;
465 } 467 }
466 // Allow SetVideoSend to fail since |enable| is false and |source| is null. 468 // Allow SetVideoSend to fail since |enable| is false and |source| is null.
467 // This the normal case when the underlying media channel has already been 469 // This the normal case when the underlying media channel has already been
468 // deleted. 470 // deleted.
469 channel_->SetVideoSend(ssrc_, false, nullptr, nullptr); 471 channel_->SetVideoSend(ssrc_, false, nullptr, nullptr);
470 } 472 }
471 473
472 } // namespace webrtc 474 } // namespace webrtc
OLDNEW
« no previous file with comments | « webrtc/pc/rtpreceiver.cc ('k') | webrtc/pc/test/fakedatachannelprovider.h » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698