Searched refs:new_sender (Results 1 – 3 of 3) sorted by relevance
| /external/webrtc/webrtc/voice_engine/test/auto_test/fakes/ |
| D | conference_transport.cc | 226 const int new_sender = remote_base_->CreateChannel(); in AddStream() local
227 EXPECT_EQ(0, remote_network_->RegisterExternalTransport(new_sender, *this)); in AddStream()
230 EXPECT_EQ(0, remote_rtp_rtcp_->SetLocalSSRC(new_sender, remote_ssrc)); in AddStream()
232 SetSendAudioLevelIndicationStatus(new_sender, true, kAudioLevelHeaderId)); in AddStream()
234 EXPECT_EQ(0, remote_codec_->SetSendCodec(new_sender, kCodecInst)); in AddStream()
235 EXPECT_EQ(0, remote_base_->StartSend(new_sender)); in AddStream()
237 new_sender, file_name.c_str(), true, false, format, 1.0)); in AddStream()
249 streams_[remote_ssrc] = std::make_pair(new_sender, new_receiver); in AddStream()
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| /external/webrtc/talk/app/webrtc/java/src/org/webrtc/ |
| D | PeerConnection.java | 228 RtpSender new_sender = nativeCreateSender(kind, stream_id); in createSender() local
229 if (new_sender != null) { in createSender()
230 senders.add(new_sender); in createSender()
232 return new_sender; in createSender()
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| /external/webrtc/talk/app/webrtc/ |
| D | peerconnection.cc | 762 RtpSenderInterface* new_sender; in CreateSender() local
764 new_sender = new AudioRtpSender(session_.get(), stats_.get()); in CreateSender()
766 new_sender = new VideoRtpSender(session_.get()); in CreateSender()
772 new_sender->set_stream_id(stream_id); in CreateSender()
774 senders_.push_back(new_sender); in CreateSender()
775 return RtpSenderProxy::Create(signaling_thread(), new_sender); in CreateSender()
1358 AudioRtpSender* new_sender = in OnAudioTrackAdded() local
1360 senders_.push_back(new_sender); in OnAudioTrackAdded()
1370 new_sender->SetSsrc(track_info->ssrc); in OnAudioTrackAdded()
1399 VideoRtpSender* new_sender = in OnVideoTrackAdded() local
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