Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(118)

Side by Side Diff: webrtc/call/call.cc

Issue 2125523004: Fix bug where a connection switch causes BWE to be set to zero. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Comments addressed. Created 4 years, 5 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_tests.cc » ('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 561 matching lines...) Expand 10 before | Expand all | Expand 10 after
572 if (config_.bitrate_config.min_bitrate_bps == 572 if (config_.bitrate_config.min_bitrate_bps ==
573 bitrate_config.min_bitrate_bps && 573 bitrate_config.min_bitrate_bps &&
574 (bitrate_config.start_bitrate_bps <= 0 || 574 (bitrate_config.start_bitrate_bps <= 0 ||
575 config_.bitrate_config.start_bitrate_bps == 575 config_.bitrate_config.start_bitrate_bps ==
576 bitrate_config.start_bitrate_bps) && 576 bitrate_config.start_bitrate_bps) &&
577 config_.bitrate_config.max_bitrate_bps == 577 config_.bitrate_config.max_bitrate_bps ==
578 bitrate_config.max_bitrate_bps) { 578 bitrate_config.max_bitrate_bps) {
579 // Nothing new to set, early abort to avoid encoder reconfigurations. 579 // Nothing new to set, early abort to avoid encoder reconfigurations.
580 return; 580 return;
581 } 581 }
582 config_.bitrate_config = bitrate_config; 582 config_.bitrate_config.min_bitrate_bps = bitrate_config.min_bitrate_bps;
583 // Start bitrate of -1 means we should keep the old bitrate, which there is
584 // no point in remembering for the future.
585 if (bitrate_config.start_bitrate_bps > 0)
586 config_.bitrate_config.start_bitrate_bps = bitrate_config.start_bitrate_bps;
587 config_.bitrate_config.max_bitrate_bps = bitrate_config.max_bitrate_bps;
583 congestion_controller_->SetBweBitrates(bitrate_config.min_bitrate_bps, 588 congestion_controller_->SetBweBitrates(bitrate_config.min_bitrate_bps,
584 bitrate_config.start_bitrate_bps, 589 bitrate_config.start_bitrate_bps,
585 bitrate_config.max_bitrate_bps); 590 bitrate_config.max_bitrate_bps);
586 } 591 }
587 592
588 void Call::SignalChannelNetworkState(MediaType media, NetworkState state) { 593 void Call::SignalChannelNetworkState(MediaType media, NetworkState state) {
589 RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread()); 594 RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
590 switch (media) { 595 switch (media) {
591 case MediaType::AUDIO: 596 case MediaType::AUDIO:
592 audio_network_state_ = state; 597 audio_network_state_ = state;
(...skipping 275 matching lines...) Expand 10 before | Expand all | Expand 10 after
868 // thread. Then this check can be enabled. 873 // thread. Then this check can be enabled.
869 // RTC_DCHECK(!configuration_thread_checker_.CalledOnValidThread()); 874 // RTC_DCHECK(!configuration_thread_checker_.CalledOnValidThread());
870 if (RtpHeaderParser::IsRtcp(packet, length)) 875 if (RtpHeaderParser::IsRtcp(packet, length))
871 return DeliverRtcp(media_type, packet, length); 876 return DeliverRtcp(media_type, packet, length);
872 877
873 return DeliverRtp(media_type, packet, length, packet_time); 878 return DeliverRtp(media_type, packet, length, packet_time);
874 } 879 }
875 880
876 } // namespace internal 881 } // namespace internal
877 } // namespace webrtc 882 } // namespace webrtc
OLDNEW
« no previous file with comments | « no previous file | webrtc/video/video_send_stream_tests.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698