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Side by Side Diff: talk/app/webrtc/peerconnection.cc

Issue 1538673002: Adding a MediaStream parameter to createSender. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Just take a string as a parameter, for simplicity. Created 5 years ago
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1 /* 1 /*
2 * libjingle 2 * libjingle
3 * Copyright 2012 Google Inc. 3 * Copyright 2012 Google Inc.
4 * 4 *
5 * Redistribution and use in source and binary forms, with or without 5 * Redistribution and use in source and binary forms, with or without
6 * modification, are permitted provided that the following conditions are met: 6 * modification, are permitted provided that the following conditions are met:
7 * 7 *
8 * 1. Redistributions of source code must retain the above copyright notice, 8 * 1. Redistributions of source code must retain the above copyright notice,
9 * this list of conditions and the following disclaimer. 9 * this list of conditions and the following disclaimer.
10 * 2. Redistributions in binary form must reproduce the above copyright notice, 10 * 2. Redistributions in binary form must reproduce the above copyright notice,
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806 rtc::scoped_refptr<DtmfSenderInterface> sender( 806 rtc::scoped_refptr<DtmfSenderInterface> sender(
807 DtmfSender::Create(track, signaling_thread(), session_.get())); 807 DtmfSender::Create(track, signaling_thread(), session_.get()));
808 if (!sender.get()) { 808 if (!sender.get()) {
809 LOG(LS_ERROR) << "CreateDtmfSender failed on DtmfSender::Create."; 809 LOG(LS_ERROR) << "CreateDtmfSender failed on DtmfSender::Create.";
810 return NULL; 810 return NULL;
811 } 811 }
812 return DtmfSenderProxy::Create(signaling_thread(), sender.get()); 812 return DtmfSenderProxy::Create(signaling_thread(), sender.get());
813 } 813 }
814 814
815 rtc::scoped_refptr<RtpSenderInterface> PeerConnection::CreateSender( 815 rtc::scoped_refptr<RtpSenderInterface> PeerConnection::CreateSender(
816 const std::string& kind) { 816 const std::string& kind,
817 const std::string& stream_id) {
817 TRACE_EVENT0("webrtc", "PeerConnection::CreateSender"); 818 TRACE_EVENT0("webrtc", "PeerConnection::CreateSender");
818 RtpSenderInterface* new_sender; 819 RtpSenderInterface* new_sender;
819 if (kind == MediaStreamTrackInterface::kAudioKind) { 820 if (kind == MediaStreamTrackInterface::kAudioKind) {
820 new_sender = new AudioRtpSender(session_.get(), stats_.get()); 821 new_sender = new AudioRtpSender(session_.get(), stats_.get());
821 } else if (kind == MediaStreamTrackInterface::kVideoKind) { 822 } else if (kind == MediaStreamTrackInterface::kVideoKind) {
822 new_sender = new VideoRtpSender(session_.get()); 823 new_sender = new VideoRtpSender(session_.get());
823 } else { 824 } else {
824 LOG(LS_ERROR) << "CreateSender called with invalid kind: " << kind; 825 LOG(LS_ERROR) << "CreateSender called with invalid kind: " << kind;
825 return rtc::scoped_refptr<RtpSenderInterface>(); 826 return rtc::scoped_refptr<RtpSenderInterface>();
826 } 827 }
828 if (!stream_id.empty()) {
829 new_sender->set_stream_id(stream_id);
830 }
827 senders_.push_back(new_sender); 831 senders_.push_back(new_sender);
828 return RtpSenderProxy::Create(signaling_thread(), new_sender); 832 return RtpSenderProxy::Create(signaling_thread(), new_sender);
829 } 833 }
830 834
831 std::vector<rtc::scoped_refptr<RtpSenderInterface>> PeerConnection::GetSenders() 835 std::vector<rtc::scoped_refptr<RtpSenderInterface>> PeerConnection::GetSenders()
832 const { 836 const {
833 std::vector<rtc::scoped_refptr<RtpSenderInterface>> senders; 837 std::vector<rtc::scoped_refptr<RtpSenderInterface>> senders;
834 for (const auto& sender : senders_) { 838 for (const auto& sender : senders_) {
835 senders.push_back(RtpSenderProxy::Create(signaling_thread(), sender.get())); 839 senders.push_back(RtpSenderProxy::Create(signaling_thread(), sender.get()));
836 } 840 }
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2072 DataChannel* PeerConnection::FindDataChannelBySid(int sid) const { 2076 DataChannel* PeerConnection::FindDataChannelBySid(int sid) const {
2073 for (const auto& channel : sctp_data_channels_) { 2077 for (const auto& channel : sctp_data_channels_) {
2074 if (channel->id() == sid) { 2078 if (channel->id() == sid) {
2075 return channel; 2079 return channel;
2076 } 2080 }
2077 } 2081 }
2078 return nullptr; 2082 return nullptr;
2079 } 2083 }
2080 2084
2081 } // namespace webrtc 2085 } // namespace webrtc
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