| /external/webrtc/sdk/android/api/org/webrtc/ |
| D | PeerConnection.java | 820 private List<RtpSender> senders = new ArrayList<>();
967 public RtpSender createSender(String kind, String stream_id) { in createSender()
968 RtpSender newSender = nativeCreateSender(kind, stream_id); in createSender()
980 public List<RtpSender> getSenders() { in getSenders()
981 for (RtpSender sender : senders) { in getSenders()
1025 public RtpSender addTrack(MediaStreamTrack track) { in addTrack()
1029 public RtpSender addTrack(MediaStreamTrack track, List<String> streamIds) { in addTrack()
1033 RtpSender newSender = nativeAddTrack(track.getNativeMediaStreamTrack(), streamIds); in addTrack()
1046 public boolean removeTrack(RtpSender sender) { in removeTrack()
1207 for (RtpSender sender : senders) { in dispose()
[all …]
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| D | RtpTransceiver.java | 113 private RtpSender cachedSender;
149 public RtpSender getSender() { in getSender()
235 private static native RtpSender nativeGetSender(long rtpTransceiver); in nativeGetSender()
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| D | RtpSender.java | 17 public class RtpSender { class
25 public RtpSender(long nativeRtpSender) { in RtpSender() method in RtpSender
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| /external/webrtc/modules/rtp_rtcp/mocks/ |
| D | mock_rtp_rtcp.h | 194 MOCK_METHOD(RTPSender*, RtpSender, (), (override));
195 MOCK_METHOD(const RTPSender*, RtpSender, (), (const, override));
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| /external/webrtc/modules/rtp_rtcp/source/ |
| D | rtp_rtcp_impl2.h | 269 RTPSender* RtpSender() override;
270 const RTPSender* RtpSender() const override;
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| D | rtp_rtcp_interface.h | 280 virtual RTPSender* RtpSender() = 0;
281 virtual const RTPSender* RtpSender() const = 0;
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| D | rtp_rtcp_impl.h | 283 RTPSender* RtpSender() override;
284 const RTPSender* RtpSender() const override;
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| D | rtp_sender_audio_unittest.cc | 79 rtp_sender_audio_(&fake_clock_, rtp_module_->RtpSender()) {
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| D | rtp_rtcp_impl2.cc | 733 RTPSender* ModuleRtpRtcpImpl2::RtpSender() { in RtpSender() function in webrtc::ModuleRtpRtcpImpl2
737 const RTPSender* ModuleRtpRtcpImpl2::RtpSender() const { in RtpSender() function in webrtc::ModuleRtpRtcpImpl2
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| D | nack_rtx_unittest.cc | 143 video_config.rtp_sender = rtp_rtcp_module_->RtpSender(); in SetUp()
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| D | rtp_rtcp_impl.cc | 825 RTPSender* ModuleRtpRtcpImpl::RtpSender() { in RtpSender() function in webrtc::ModuleRtpRtcpImpl
829 const RTPSender* ModuleRtpRtcpImpl::RtpSender() const { in RtpSender() function in webrtc::ModuleRtpRtcpImpl
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| D | rtp_sender_video_unittest.cc | 179 rtp_module_->RtpSender(),
721 config.rtp_sender = rtp_module_->RtpSender(); in TEST_P()
924 config.rtp_sender = rtp_module_->RtpSender(); in CreateSenderWithFrameTransformer()
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| D | rtp_rtcp_impl_unittest.cc | 184 video_config.rtp_sender = sender_.impl_->RtpSender(); in SetUp()
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| D | rtp_rtcp_impl2_unittest.cc | 189 video_config.rtp_sender = sender_.impl_->RtpSender(); in SetUp()
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| /external/webrtc/api/ |
| D | rtp_sender_interface.h | 107 BEGIN_SIGNALING_PROXY_MAP(RtpSender)
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| /external/webrtc/sdk/android/instrumentationtests/src/org/webrtc/ |
| D | RtpTransceiverTest.java | 63 RtpSender sender = transceiver.getSender(); in testSetRidInSimulcast()
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| D | RtpSenderTest.java | 54 RtpSender sender = transceiver.getSender(); in testSetDegradationPreference()
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| D | PeerConnectionEndToEndTest.java | 838 RtpSender videoSender = null; in testCompleteSession()
839 RtpSender audioSender = null; in testCompleteSession()
840 for (RtpSender sender : offeringPC.getSenders()) { in testCompleteSession()
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| /external/webrtc/audio/voip/ |
| D | audio_egress.cc | 24 rtp_sender_audio_(clock, rtp_rtcp_->RtpSender()), in AudioEgress()
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| /external/webrtc/examples/androidapp/src/org/appspot/apprtc/ |
| D | PeerConnectionClient.java | 58 import org.webrtc.RtpSender;
166 private RtpSender localVideoSender;
953 for (RtpSender sender : peerConnection.getSenders()) { in findVideoSender()
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| /external/webrtc/call/ |
| D | rtp_video_sender.cc | 268 video_config.rtp_sender = rtp_rtcp->RtpSender(); in CreateRtpStreamSenders()
924 RTPSender* rtp_sender = it->second->RtpSender(); in OnPacketFeedbackVector()
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| /external/webrtc/audio/ |
| D | channel_send.cc | 502 rtp_rtcp_->RtpSender()); in ChannelSend()
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| /external/webrtc/sdk/android/ |
| D | BUILD.gn | 294 "api/org/webrtc/RtpSender.java",
1241 "api/org/webrtc/RtpSender.java",
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