Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(1)

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

Issue 2666853002: Move DTMF sender to RtpSender (as opposed to WebRtcSession). (Closed)
Patch Set: Merge with master Created 3 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 | « webrtc/pc/dtmfsender_unittest.cc ('k') | webrtc/pc/rtpsender.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 2012 The WebRTC project authors. All Rights Reserved. 2 * Copyright 2012 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 931 matching lines...) Expand 10 before | Expand all | Expand 10 after
942 } 942 }
943 943
944 rtc::scoped_refptr<DtmfSenderInterface> PeerConnection::CreateDtmfSender( 944 rtc::scoped_refptr<DtmfSenderInterface> PeerConnection::CreateDtmfSender(
945 AudioTrackInterface* track) { 945 AudioTrackInterface* track) {
946 TRACE_EVENT0("webrtc", "PeerConnection::CreateDtmfSender"); 946 TRACE_EVENT0("webrtc", "PeerConnection::CreateDtmfSender");
947 if (IsClosed()) { 947 if (IsClosed()) {
948 return nullptr; 948 return nullptr;
949 } 949 }
950 if (!track) { 950 if (!track) {
951 LOG(LS_ERROR) << "CreateDtmfSender - track is NULL."; 951 LOG(LS_ERROR) << "CreateDtmfSender - track is NULL.";
952 return NULL; 952 return nullptr;
953 } 953 }
954 if (!local_streams_->FindAudioTrack(track->id())) { 954 auto it = FindSenderForTrack(track);
955 LOG(LS_ERROR) << "CreateDtmfSender is called with a non local audio track."; 955 if (it == senders_.end()) {
956 return NULL; 956 LOG(LS_ERROR) << "CreateDtmfSender called with a non-added track.";
957 return nullptr;
957 } 958 }
958 959
959 rtc::scoped_refptr<DtmfSenderInterface> sender( 960 return (*it)->GetDtmfSender();
960 DtmfSender::Create(track, signaling_thread(), session_.get()));
961 if (!sender.get()) {
962 LOG(LS_ERROR) << "CreateDtmfSender failed on DtmfSender::Create.";
963 return NULL;
964 }
965 return DtmfSenderProxy::Create(signaling_thread(), sender.get());
966 } 961 }
967 962
968 rtc::scoped_refptr<RtpSenderInterface> PeerConnection::CreateSender( 963 rtc::scoped_refptr<RtpSenderInterface> PeerConnection::CreateSender(
969 const std::string& kind, 964 const std::string& kind,
970 const std::string& stream_id) { 965 const std::string& stream_id) {
971 TRACE_EVENT0("webrtc", "PeerConnection::CreateSender"); 966 TRACE_EVENT0("webrtc", "PeerConnection::CreateSender");
972 if (IsClosed()) { 967 if (IsClosed()) {
973 return nullptr; 968 return nullptr;
974 } 969 }
975 rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>> new_sender; 970 rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>> new_sender;
(...skipping 1590 matching lines...) Expand 10 before | Expand all | Expand 10 after
2566 2561
2567 bool PeerConnection::StartRtcEventLog_w(rtc::PlatformFile file, 2562 bool PeerConnection::StartRtcEventLog_w(rtc::PlatformFile file,
2568 int64_t max_size_bytes) { 2563 int64_t max_size_bytes) {
2569 return event_log_->StartLogging(file, max_size_bytes); 2564 return event_log_->StartLogging(file, max_size_bytes);
2570 } 2565 }
2571 2566
2572 void PeerConnection::StopRtcEventLog_w() { 2567 void PeerConnection::StopRtcEventLog_w() {
2573 event_log_->StopLogging(); 2568 event_log_->StopLogging();
2574 } 2569 }
2575 } // namespace webrtc 2570 } // namespace webrtc
OLDNEW
« no previous file with comments | « webrtc/pc/dtmfsender_unittest.cc ('k') | webrtc/pc/rtpsender.h » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698