/external/webrtc/webrtc/voice_engine/test/auto_test/ |
D | voe_conference_test.cc | 61 unsigned int id_1 = trans.AddStream(input_file, kInputFormat); in TEST() 62 unsigned int id_2 = trans.AddStream(input_file, kInputFormat); in TEST() 141 unsigned int id_0 = trans.AddStream(silence_file, kInputFormat); in TEST() 142 unsigned int id_1 = trans.AddStream(input_file, kInputFormat); in TEST() 143 unsigned int id_2 = trans.AddStream(input_file, kInputFormat); in TEST() 144 unsigned int id_3 = trans.AddStream(input_file, kInputFormat); in TEST()
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/external/webrtc/talk/media/sctp/ |
D | sctpdataengine_unittest.cc | 245 AddStream(1); in SetupConnectedChannels() 246 AddStream(2); in SetupConnectedChannels() 273 bool AddStream(int ssrc) { in AddStream() function in SctpDataMediaChannelTest 450 AddStream(3); in TEST_F() 472 AddStream(3); in TEST_F() 473 AddStream(4); in TEST_F() 511 EXPECT_TRUE(AddStream(1022)); in TEST_F() 512 EXPECT_FALSE(AddStream(1023)); in TEST_F() 537 AddStream(1); in TEST_F()
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D | sctpdataengine.h | 208 bool AddStream(const StreamParams &sp);
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D | sctpdataengine.cc | 576 return AddStream(stream); in AddSendStream() 726 bool SctpDataMediaChannel::AddStream(const StreamParams& stream) { in AddStream() function in cricket::SctpDataMediaChannel
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/external/webrtc/talk/media/base/ |
D | streamparams.cc | 37 void AddStream(std::vector<StreamParams>* streams, const StreamParams& stream) { in AddStream() function 77 AddStream(&audio_, stream); in AddAudioStream() 81 AddStream(&video_, stream); in AddVideoStream() 85 AddStream(&data_, stream); in AddDataStream()
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/external/webrtc/talk/app/webrtc/ |
D | statscollector_unittest.cc | 839 stats.AddStream(stream_); in TEST_F() 887 stats.AddStream(stream_); in TEST_F() 953 stats.AddStream(stream_); in TEST_F() 989 stats.AddStream(stream_); in TEST_F() 1053 stats.AddStream(stream_); in TEST_F() 1111 stats.AddStream(stream_); in TEST_F() 1137 stats.AddStream(stream_); in TEST_F() 1192 stats.AddStream(stream_); in TEST_F() 1499 stats.AddStream(stream_); in TEST_F() 1534 stats.AddStream(stream_); in TEST_F() [all …]
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D | peerconnectioninterface_unittest.cc | 364 local_collection->AddStream(stream); in CreateStreamCollection() 448 remote_streams_->AddStream(stream); in OnAddStream() 632 EXPECT_TRUE(pc_->AddStream(stream)); in AddVideoStream() 644 EXPECT_TRUE(pc_->AddStream(stream)); in AddVoiceStream() 662 EXPECT_TRUE(pc_->AddStream(stream)); in AddAudioVideoStream() 894 reference_collection_->AddStream(stream); in CreateSessionDescriptionAndReference() 959 EXPECT_TRUE(pc_->AddStream(stream)); in TEST_F() 1785 EXPECT_FALSE(pc_->AddStream(local_stream)); in TEST_F() 2114 pc_->AddStream(reference_collection_->at(0)); in TEST_F() 2127 pc_->AddStream(reference_collection_->at(0)); in TEST_F() [all …]
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D | peerconnectionproxy.h | 42 PROXY_METHOD1(bool, AddStream, MediaStreamInterface*)
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D | statscollector.h | 68 void AddStream(MediaStreamInterface* stream);
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D | streamcollection.h | 94 void AddStream(MediaStreamInterface* stream) { in AddStream() function
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D | peerconnection.cc | 679 bool PeerConnection::AddStream(MediaStreamInterface* local_stream) { in AddStream() function in webrtc::PeerConnection 688 local_streams_->AddStream(local_stream); in AddStream() 707 stats_->AddStream(local_stream); in AddStream() 1118 stats_->AddStream(new_stream); in SetRemoteDescription() 1536 remote_streams_->AddStream(stream); in UpdateRemoteStreamsList() 1537 new_streams->AddStream(stream); in UpdateRemoteStreamsList() 1556 remote_streams_->AddStream(default_stream); in UpdateRemoteStreamsList() 1557 new_streams->AddStream(default_stream); in UpdateRemoteStreamsList()
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D | webrtcsdp_unittest.cc | 570 video->AddStream(video_stream1); in WebRtcSdpTest() 576 video->AddStream(video_stream2); in WebRtcSdpTest() 585 video->AddStream(video_stream3); in WebRtcSdpTest() 727 audio->AddStream(audio_stream1); in CreateAudioContentDescription() 733 audio->AddStream(audio_stream2); in CreateAudioContentDescription() 1130 data_desc_->AddStream(data_stream); in AddRtpDataChannel()
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D | peerconnectioninterface.h | 327 virtual bool AddStream(MediaStreamInterface* stream) = 0;
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D | peerconnection.h | 83 bool AddStream(MediaStreamInterface* local_stream) override;
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D | statscollector.cc | 383 void StatsCollector::AddStream(MediaStreamInterface* stream) { in AddStream() function in webrtc::StatsCollector
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/external/webrtc/talk/session/media/ |
D | channel_unittest.cc | 633 content1.AddStream(stream1); in TestUpdateStreamsInLocalContent() 643 content2.AddStream(stream2); in TestUpdateStreamsInLocalContent() 644 content2.AddStream(stream3); in TestUpdateStreamsInLocalContent() 656 content3.AddStream(stream1); in TestUpdateStreamsInLocalContent() 666 content4.AddStream(stream2); in TestUpdateStreamsInLocalContent() 700 content1.AddStream(stream1); in TestUpdateStreamsInRemoteContent() 711 content2.AddStream(stream2); in TestUpdateStreamsInRemoteContent() 712 content2.AddStream(stream3); in TestUpdateStreamsInRemoteContent() 724 content3.AddStream(stream1); in TestUpdateStreamsInRemoteContent() 734 content4.AddStream(stream2); in TestUpdateStreamsInRemoteContent() [all …]
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D | mediasession.h | 249 void AddStream(const StreamParams& stream) { in AddStream() function
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D | mediasession.cc | 538 content_description->AddStream(stream_param); in AddStreamParams() 544 content_description->AddStream(*param); in AddStreamParams()
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/external/webrtc/webrtc/voice_engine/test/auto_test/fakes/ |
D | conference_transport.h | 67 unsigned int AddStream(std::string file_name, webrtc::FileFormats format);
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D | conference_transport.cc | 224 unsigned int ConferenceTransport::AddStream(std::string file_name, in AddStream() function in voetest::ConferenceTransport
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/external/webrtc/webrtc/modules/remote_bitrate_estimator/ |
D | remote_bitrate_estimator_unittest_helper.cc | 129 void StreamGenerator::AddStream(RtpStream* stream) { in AddStream() function in webrtc::testing::StreamGenerator 202 stream_generator_->AddStream(new testing::RtpStream( in AddDefaultStream() 453 stream_generator_->AddStream(new testing::RtpStream( in CapacityDropTestHelper()
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D | remote_bitrate_estimator_unittest_helper.h | 116 void AddStream(RtpStream* stream);
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/external/webrtc/webrtc/examples/peerconnection/client/ |
D | conductor.cc | 106 peer_connection_->AddStream(streams->at(i)); in ReinitializePeerConnectionForLoopback() 413 if (!peer_connection_->AddStream(stream)) { in AddStreams()
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/external/webrtc/talk/app/webrtc/test/ |
D | peerconnectiontestwrapper.cc | 256 EXPECT_TRUE(peer_connection_->AddStream(stream)); in GetAndAddUserMedia()
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/external/webrtc/talk/app/webrtc/objc/ |
D | RTCPeerConnection.mm | 155 BOOL ret = self.peerConnection->AddStream(stream.mediaStream);
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