| /external/webrtc/call/ |
| D | rtp_demuxer_unittest.cc | 57 bool AddSinkOnlySsrc(uint32_t ssrc, RtpPacketSinkInterface* sink) { in AddSinkOnlySsrc()
103 uint32_t ssrc, in CreatePacket()
111 std::unique_ptr<RtpPacketReceived> CreatePacketWithSsrc(uint32_t ssrc) { in CreatePacketWithSsrc()
116 uint32_t ssrc, in CreatePacketWithSsrcMid()
127 uint32_t ssrc, in CreatePacketWithSsrcRsid()
138 uint32_t ssrc, in CreatePacketWithSsrcRrid()
149 uint32_t ssrc, in CreatePacketWithSsrcMidRsid()
163 uint32_t ssrc, in CreatePacketWithSsrcRsidRrid()
190 constexpr uint32_t ssrc = 1; in TEST_F() local
262 constexpr uint32_t ssrc = 1; in TEST_F() local
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| D | rtp_demuxer.cc | 61 for (auto ssrc : ssrcs) { in ToString() local
134 for (uint32_t ssrc : criteria.ssrcs) { in AddSink() local
188 for (uint32_t ssrc : criteria.ssrcs) { in CriteriaWouldConflict() local
217 bool RtpDemuxer::AddSink(uint32_t ssrc, RtpPacketSinkInterface* sink) { in AddSink()
267 uint32_t ssrc = packet.Ssrc(); in ResolveSink() local
354 uint32_t ssrc) { in ResolveSinkByMid()
367 uint32_t ssrc) { in ResolveSinkByMidRsid()
378 uint32_t ssrc) { in ResolveSinkByRsid()
390 uint32_t ssrc) { in ResolveSinkByPayloadType()
404 void RtpDemuxer::AddSsrcSinkBinding(uint32_t ssrc, in AddSsrcSinkBinding()
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| D | rtp_stream_receiver_controller.cc | 21 uint32_t ssrc, in Receiver()
49 RtpStreamReceiverController::CreateReceiver(uint32_t ssrc, in CreateReceiver()
59 bool RtpStreamReceiverController::AddSink(uint32_t ssrc, in AddSink()
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| D | rtp_video_sender.cc | 379 for (uint32_t ssrc : rtp_config_.ssrcs) { in RtpVideoSender() local
637 uint32_t ssrc = rtp_config_.ssrcs[i]; in ConfigureSsrcs() local
654 uint32_t ssrc = rtp_config_.rtx.ssrcs[i]; in ConfigureSsrcs() local
702 uint32_t ssrc = rtp_config_.ssrcs[i]; in GetRtpStates() local
712 uint32_t ssrc = rtp_config_.flexfec.ssrc; in GetRtpStates() local
719 uint32_t ssrc = rtp_config_.rtx.ssrcs[i]; in GetRtpStates() local
838 uint32_t ssrc, in GetSentRtpPacketInfos()
921 const uint32_t ssrc = kv.first; in OnPacketFeedbackVector() local
932 const uint32_t ssrc = kv.first; in OnPacketFeedbackVector() local
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| /external/webrtc/media/engine/ |
| D | unhandled_packets_buffer_unittest.cc | 52 buff.BackfillPackets(ssrcs, [&packets](uint32_t ssrc, int64_t packet_time_us, in TEST()
67 buff.BackfillPackets(ssrcs, [&packets](uint32_t ssrc, int64_t packet_time_us, in TEST()
82 buff.BackfillPackets(ssrcs, [](uint32_t ssrc, int64_t packet_time_us, in TEST()
89 buff.BackfillPackets(ssrcs, [](uint32_t ssrc, int64_t packet_time_us, in TEST()
105 buff.BackfillPackets(ssrcs, [&packets](uint32_t ssrc, int64_t packet_time_us, in TEST()
116 buff.BackfillPackets(ssrcs, [](uint32_t ssrc, int64_t packet_time_us, in TEST()
132 buff.BackfillPackets(ssrcs, [&packets](uint32_t ssrc, int64_t packet_time_us, in TEST()
142 buff.BackfillPackets(ssrcs, [](uint32_t ssrc, int64_t packet_time_us, in TEST()
155 buff.BackfillPackets(ssrcs, [](uint32_t ssrc, int64_t packet_time_us, in TEST()
164 buff.BackfillPackets(ssrcs, [](uint32_t ssrc, int64_t packet_time_us, in TEST()
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| D | webrtc_video_engine.cc | 360 uint32_t ssrc = pair.first; in MergeInfoAboutOutboundRtpSubstreams() local
520 uint32_t ssrc) { in OnUnsignalledSsrc()
1011 uint32_t ssrc, in SetRtpSendParameters()
1224 uint32_t ssrc, in SetVideoSend()
1247 for (uint32_t ssrc : sp.ssrcs) { in ValidateSendSsrcAvailability() local
1259 for (uint32_t ssrc : sp.ssrcs) { in ValidateReceiveSsrcAvailability() local
1305 uint32_t ssrc = sp.first_ssrc(); in AddSendStream() local
1324 bool WebRtcVideoChannel::RemoveSendStream(uint32_t ssrc) { in RemoveSendStream()
1387 uint32_t ssrc = sp.first_ssrc(); in AddRecvStream() local
1432 uint32_t ssrc = sp.first_ssrc(); in ConfigureReceiverRtp() local
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| D | webrtc_voice_engine.cc | 754 uint32_t ssrc, in WebRtcAudioSendStream()
1486 uint32_t ssrc, in SetRtpSendParameters()
1873 bool WebRtcVoiceMediaChannel::SetAudioSend(uint32_t ssrc, in SetAudioSend()
1897 uint32_t ssrc = sp.first_ssrc(); in AddSendStream() local
1931 bool WebRtcVoiceMediaChannel::RemoveSendStream(uint32_t ssrc) { in RemoveSendStream()
1973 const uint32_t ssrc = sp.first_ssrc(); in AddRecvStream() local
2003 bool WebRtcVoiceMediaChannel::RemoveRecvStream(uint32_t ssrc) { in RemoveRecvStream()
2029 bool WebRtcVoiceMediaChannel::SetLocalSource(uint32_t ssrc, in SetLocalSource()
2052 bool WebRtcVoiceMediaChannel::SetOutputVolume(uint32_t ssrc, double volume) { in SetOutputVolume()
2068 for (uint32_t ssrc : unsignaled_recv_ssrcs_) { in SetDefaultOutputVolume() local
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| D | unhandled_packets_buffer.cc | 26 void UnhandledPacketsBuffer::AddPacket(uint32_t ssrc, in AddPacket()
57 const uint32_t ssrc = buffer_[pos].ssrc; in BackfillPackets() local
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| /external/webrtc/rtc_tools/rtc_event_log_visualizer/ |
| D | analyzer_common.cc | 18 uint32_t ssrc) { in IsRtxSsrc()
30 uint32_t ssrc) { in IsVideoSsrc()
42 uint32_t ssrc) { in IsAudioSsrc()
54 uint32_t ssrc) { in GetStreamName()
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| /external/webrtc/media/base/ |
| D | media_channel.cc | 37 uint32_t ssrc, in SetFrameEncryptor()
43 uint32_t ssrc, in SetFrameDecryptor()
51 uint32_t ssrc, in SetEncoderToPacketizerFrameTransformer()
54 uint32_t ssrc, in SetDepacketizerToDecoderFrameTransformer()
122 uint32_t ssrc, in SetRtpSendParameters()
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| D | fake_media_engine.cc | 22 FakeVoiceMediaChannel::DtmfInfo::DtmfInfo(uint32_t ssrc, in DtmfInfo()
102 bool FakeVoiceMediaChannel::SetAudioSend(uint32_t ssrc, in SetAudioSend()
127 bool FakeVoiceMediaChannel::RemoveRecvStream(uint32_t ssrc) { in RemoveRecvStream()
144 bool FakeVoiceMediaChannel::InsertDtmf(uint32_t ssrc, in InsertDtmf()
150 bool FakeVoiceMediaChannel::SetOutputVolume(uint32_t ssrc, double volume) { in SetOutputVolume()
163 bool FakeVoiceMediaChannel::GetOutputVolume(uint32_t ssrc, double* volume) { in GetOutputVolume()
169 bool FakeVoiceMediaChannel::SetBaseMinimumPlayoutDelayMs(uint32_t ssrc, in SetBaseMinimumPlayoutDelayMs()
190 uint32_t ssrc, in SetRawAudioSink()
229 bool FakeVoiceMediaChannel::SetLocalSource(uint32_t ssrc, AudioSource* source) { in SetLocalSource()
247 uint32_t ssrc, in CompareDtmfInfo()
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| D | fake_media_engine.h | 113 virtual bool RemoveSendStream(uint32_t ssrc) { in RemoveSendStream()
131 virtual bool RemoveRecvStream(uint32_t ssrc) { in RemoveRecvStream()
139 virtual webrtc::RtpParameters GetRtpSendParameters(uint32_t ssrc) const { in GetRtpSendParameters()
147 uint32_t ssrc, in SetRtpSendParameters()
164 virtual webrtc::RtpParameters GetRtpReceiveParameters(uint32_t ssrc) const { in GetRtpReceiveParameters()
175 bool IsStreamMuted(uint32_t ssrc) const { in IsStreamMuted()
190 bool HasRecvStream(uint32_t ssrc) const { in HasRecvStream()
193 bool HasSendStream(uint32_t ssrc) const { in HasSendStream()
232 bool MuteStream(uint32_t ssrc, bool mute) { in MuteStream()
311 uint32_t ssrc; member
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| D | stream_params.cc | 28 for (uint32_t ssrc : ssrcs) { in AppendSsrcs() local
156 uint32_t ssrc = ssrc_generator->GenerateId(); in GenerateSsrcs() local
167 for (uint32_t ssrc : primary_ssrcs) { in GenerateSsrcs() local
173 for (uint32_t ssrc : primary_ssrcs) { in GenerateSsrcs() local
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| /external/webrtc/modules/pacing/ |
| D | pacing_controller_unittest.cc | 55 uint32_t ssrc, in BuildPacket()
239 uint32_t ssrc, in Send()
248 uint32_t ssrc, in SendAndExpectPacket()
332 const uint32_t ssrc; member
457 bool padding) { wait_end_time = clock_.CurrentTime(); }); in TEST_P()
515 uint32_t ssrc = 12345; in TEST_P() local
606 uint32_t ssrc = 12345; in TEST_P() local
679 constexpr uint32_t ssrc = 333; in TEST_P() local
700 uint32_t ssrc = 12345; in TEST_P() local
721 uint32_t ssrc = 12345; in TEST_P() local
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| /external/webrtc/video/ |
| D | encoder_rtcp_feedback.cc | 46 bool EncoderRtcpFeedback::HasSsrc(uint32_t ssrc) { in HasSsrc()
55 void EncoderRtcpFeedback::OnReceivedIntraFrameRequest(uint32_t ssrc) { in OnReceivedIntraFrameRequest()
71 uint32_t ssrc, in OnReceivedLossNotification()
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| D | send_delay_stats.cc | 58 for (const auto& ssrc : config.rtp.ssrcs) in AddSsrcs() local
62 AvgCounter* SendDelayStats::GetSendDelayCounter(uint32_t ssrc) { in GetSendDelayCounter()
74 uint32_t ssrc) { in OnSendPacket()
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| D | report_block_stats.cc | 34 void ReportBlockStats::Store(uint32_t ssrc, const RtcpStatistics& rtcp_stats) { in Store()
42 void ReportBlockStats::StoreAndAddPacketIncrement(uint32_t ssrc, in StoreAndAddPacketIncrement()
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| /external/webrtc/api/ |
| D | frame_transformer_interface.h | 89 uint32_t ssrc) {} in RegisterTransformedFrameSinkCallback()
91 virtual void UnregisterTransformedFrameSinkCallback(uint32_t ssrc) {} in UnregisterTransformedFrameSinkCallback()
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| /external/webrtc/modules/rtp_rtcp/source/rtcp_packet/ |
| D | tmmb_item.h | 32 void set_ssrc(uint32_t ssrc) { ssrc_ = ssrc; } in set_ssrc()
36 uint32_t ssrc() const { return ssrc_; } in ssrc() function
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| D | fir.h | 28 uint32_t ssrc; member
39 void AddRequestTo(uint32_t ssrc, uint8_t seq_num) { in AddRequestTo()
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| /external/webrtc/pc/ |
| D | track_media_info_map_unittest.cc | 37 for (uint32_t ssrc : ssrcs) { in CreateRtpParametersWithSsrcs() local
123 for (uint32_t ssrc : ssrcs) { in AddRtpSenderWithSsrcs() local
131 for (uint32_t ssrc : ssrcs) { in AddRtpSenderWithSsrcs() local
152 for (uint32_t ssrc : ssrcs) { in AddRtpReceiverWithSsrcs() local
160 for (uint32_t ssrc : ssrcs) { in AddRtpReceiverWithSsrcs() local
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| /external/exoplayer/tree_15dc86382f17a24a3e881e52e31a810c1ea44b49/library/rtsp/src/main/java/com/google/android/exoplayer2/source/rtsp/ |
| D | RtpPacket.java | 67 private int ssrc; field in RtpPacket.Builder
103 public Builder setSsrc(int ssrc) { in setSsrc()
169 public final int ssrc; field in RtpPacket
208 int ssrc = packetBuffer.readInt(); in parse() local
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| /external/exoplayer/tree_8e57d3715f9092d5ec54ebe2e538f34bfcc34479/library/rtsp/src/main/java/com/google/android/exoplayer2/source/rtsp/ |
| D | RtpPacket.java | 67 private int ssrc; field in RtpPacket.Builder
103 public Builder setSsrc(int ssrc) { in setSsrc()
169 public final int ssrc; field in RtpPacket
208 int ssrc = packetBuffer.readInt(); in parse() local
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| /external/webrtc/logging/rtc_event_log/ |
| D | rtc_event_log_unittest.cc | 206 uint32_t ssrc, in SsrcUsed()
218 uint32_t ssrc; in WriteAudioRecvConfigs() local
235 uint32_t ssrc; in WriteAudioSendConfigs() local
260 uint32_t ssrc = prng_.Rand<uint32_t>(); in WriteVideoRecvConfigs() local
289 uint32_t ssrc = prng_.Rand<uint32_t>(); in WriteVideoSendConfigs() local
366 uint32_t ssrc = incoming_extensions_[stream].first; in WriteLog() local
467 uint32_t ssrc = incoming_extensions_[stream].first; in WriteLog() local
479 uint32_t ssrc = outgoing_extensions_[stream].first; in WriteLog() local
576 uint32_t ssrc = kv.first; in ReadAndVerifyLog() local
654 uint32_t ssrc = kv.ssrc; in ReadAndVerifyLog() local
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| /external/openscreen/cast/protocol/castv2/streaming_examples/ |
| D | offer.json | 17 "ssrc": 264890, number
35 "ssrc": 748229, number
54 "ssrc": 748229, number
72 "ssrc": 748230, number
90 "ssrc": 748231, number
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