Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(522)

Side by Side Diff: webrtc/video/video_receive_stream.cc

Issue 1693553002: Move simple RtpRtcp calls to VideoSendStream. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: feedback Created 4 years, 10 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 | « no previous file | webrtc/video/video_send_stream.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 (c) 2013 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2013 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 161 matching lines...) Expand 10 before | Expand all | Expand 10 after
172 nullptr, 172 nullptr,
173 congestion_controller_->GetRemoteBitrateEstimator( 173 congestion_controller_->GetRemoteBitrateEstimator(
174 UseSendSideBwe(config_)), 174 UseSendSideBwe(config_)),
175 call_stats_->rtcp_rtt_stats(), 175 call_stats_->rtcp_rtt_stats(),
176 congestion_controller_->pacer(), 176 congestion_controller_->pacer(),
177 congestion_controller_->packet_router(), 177 congestion_controller_->packet_router(),
178 1, 178 1,
179 false), 179 false),
180 vie_receiver_(vie_channel_.vie_receiver()), 180 vie_receiver_(vie_channel_.vie_receiver()),
181 vie_sync_(vcm_.get()), 181 vie_sync_(vcm_.get()),
182 rtp_rtcp_(vie_channel_.rtp_rtcp()) { 182 rtp_rtcp_(vie_channel_.rtp_rtcp().front()) {
183 LOG(LS_INFO) << "VideoReceiveStream: " << config_.ToString(); 183 LOG(LS_INFO) << "VideoReceiveStream: " << config_.ToString();
184 184
185 RTC_DCHECK(process_thread_); 185 RTC_DCHECK(process_thread_);
186 RTC_DCHECK(congestion_controller_); 186 RTC_DCHECK(congestion_controller_);
187 RTC_DCHECK(call_stats_); 187 RTC_DCHECK(call_stats_);
188 RTC_DCHECK(remb_); 188 RTC_DCHECK(remb_);
189 RTC_CHECK(vie_channel_.Init() == 0); 189 RTC_CHECK(vie_channel_.Init() == 0);
190 190
191 // Register the channel to receive stats updates. 191 // Register the channel to receive stats updates.
192 call_stats_->RegisterStatsObserver(vie_channel_.GetStatsObserver()); 192 call_stats_->RegisterStatsObserver(vie_channel_.GetStatsObserver());
(...skipping 235 matching lines...) Expand 10 before | Expand all | Expand 10 after
428 return true; 428 return true;
429 } 429 }
430 430
431 void VideoReceiveStream::Decode() { 431 void VideoReceiveStream::Decode() {
432 static const int kMaxDecodeWaitTimeMs = 50; 432 static const int kMaxDecodeWaitTimeMs = 50;
433 vcm_->Decode(kMaxDecodeWaitTimeMs); 433 vcm_->Decode(kMaxDecodeWaitTimeMs);
434 } 434 }
435 435
436 } // namespace internal 436 } // namespace internal
437 } // namespace webrtc 437 } // namespace webrtc
OLDNEW
« no previous file with comments | « no previous file | webrtc/video/video_send_stream.h » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698