Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(401)

Side by Side Diff: webrtc/media/engine/webrtcvideoengine2.cc

Issue 1803063004: Reset the BWE when the network changes (Closed) Base URL: https://chromium.googlesource.com/external/webrtc@master
Patch Set: Updated comments Created 4 years, 9 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
OLDNEW
1 /* 1 /*
2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2014 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 1410 matching lines...) Expand 10 before | Expand all | Expand 10 after
1421 webrtc_packet_time); 1421 webrtc_packet_time);
1422 } 1422 }
1423 1423
1424 void WebRtcVideoChannel2::OnReadyToSend(bool ready) { 1424 void WebRtcVideoChannel2::OnReadyToSend(bool ready) {
1425 LOG(LS_VERBOSE) << "OnReadyToSend: " << (ready ? "Ready." : "Not ready."); 1425 LOG(LS_VERBOSE) << "OnReadyToSend: " << (ready ? "Ready." : "Not ready.");
1426 call_->SignalChannelNetworkState( 1426 call_->SignalChannelNetworkState(
1427 webrtc::MediaType::VIDEO, 1427 webrtc::MediaType::VIDEO,
1428 ready ? webrtc::kNetworkUp : webrtc::kNetworkDown); 1428 ready ? webrtc::kNetworkUp : webrtc::kNetworkDown);
1429 } 1429 }
1430 1430
1431 void WebRtcVideoChannel2::OnNetworkRouteChanged(
1432 const std::string& transport_name,
1433 const NetworkRoute& network_route) {
1434 call_->OnNetworkRouteChanged(transport_name, network_route);
1435 }
1436
1431 bool WebRtcVideoChannel2::MuteStream(uint32_t ssrc, bool mute) { 1437 bool WebRtcVideoChannel2::MuteStream(uint32_t ssrc, bool mute) {
1432 LOG(LS_VERBOSE) << "MuteStream: " << ssrc << " -> " 1438 LOG(LS_VERBOSE) << "MuteStream: " << ssrc << " -> "
1433 << (mute ? "mute" : "unmute"); 1439 << (mute ? "mute" : "unmute");
1434 RTC_DCHECK(ssrc != 0); 1440 RTC_DCHECK(ssrc != 0);
1435 rtc::CritScope stream_lock(&stream_crit_); 1441 rtc::CritScope stream_lock(&stream_crit_);
1436 const auto& kv = send_streams_.find(ssrc); 1442 const auto& kv = send_streams_.find(ssrc);
1437 if (kv == send_streams_.end()) { 1443 if (kv == send_streams_.end()) {
1438 LOG(LS_ERROR) << "No sending stream on ssrc " << ssrc; 1444 LOG(LS_ERROR) << "No sending stream on ssrc " << ssrc;
1439 return false; 1445 return false;
1440 } 1446 }
(...skipping 1163 matching lines...) Expand 10 before | Expand all | Expand 10 after
2604 rtx_mapping[video_codecs[i].codec.id] != 2610 rtx_mapping[video_codecs[i].codec.id] !=
2605 fec_settings.red_payload_type) { 2611 fec_settings.red_payload_type) {
2606 video_codecs[i].rtx_payload_type = rtx_mapping[video_codecs[i].codec.id]; 2612 video_codecs[i].rtx_payload_type = rtx_mapping[video_codecs[i].codec.id];
2607 } 2613 }
2608 } 2614 }
2609 2615
2610 return video_codecs; 2616 return video_codecs;
2611 } 2617 }
2612 2618
2613 } // namespace cricket 2619 } // namespace cricket
OLDNEW

Powered by Google App Engine
This is Rietveld 408576698