| /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 | 372 uint32_t ssrc = pair.first; in MergeInfoAboutOutboundRtpSubstreams() local
420 absl::optional<uint32_t> ssrc, in IsActiveFromEncodings()
549 uint32_t ssrc, in OnUnsignalledSsrc()
1044 uint32_t ssrc, in SetRtpSendParameters()
1259 void WebRtcVideoChannel::SetReceiverReportSsrc(uint32_t ssrc) { in SetReceiverReportSsrc()
1295 uint32_t ssrc, in SetVideoSend()
1318 for (uint32_t ssrc : sp.ssrcs) { in ValidateSendSsrcAvailability() local
1330 for (uint32_t ssrc : sp.ssrcs) { in ValidateReceiveSsrcAvailability() local
1376 uint32_t ssrc = sp.first_ssrc(); in AddSendStream() local
1391 bool WebRtcVideoChannel::RemoveSendStream(uint32_t ssrc) { in RemoveSendStream()
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| D | webrtc_voice_engine.cc | 728 uint32_t ssrc, in WebRtcAudioSendStream()
1387 uint32_t ssrc, in SetRtpSendParameters()
1814 bool WebRtcVoiceMediaChannel::SetAudioSend(uint32_t ssrc, in SetAudioSend()
1838 uint32_t ssrc = sp.first_ssrc(); in AddSendStream() local
1870 bool WebRtcVoiceMediaChannel::RemoveSendStream(uint32_t ssrc) { in RemoveSendStream()
1912 const uint32_t ssrc = sp.first_ssrc(); in AddRecvStream() local
1946 bool WebRtcVoiceMediaChannel::RemoveRecvStream(uint32_t ssrc) { in RemoveRecvStream()
1972 for (uint32_t ssrc : to_remove) { in ResetUnsignaledRecvStream() local
1984 bool WebRtcVoiceMediaChannel::SetLocalSource(uint32_t ssrc, in SetLocalSource()
2007 bool WebRtcVoiceMediaChannel::SetOutputVolume(uint32_t ssrc, double volume) { in SetOutputVolume()
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| D | unhandled_packets_buffer.cc | 26 void UnhandledPacketsBuffer::AddPacket(uint32_t ssrc, in AddPacket()
56 const uint32_t ssrc = buffer_[pos].ssrc; in BackfillPackets() local
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| /external/webrtc/call/ |
| D | rtp_demuxer_unittest.cc | 58 bool AddSinkOnlySsrc(uint32_t ssrc, RtpPacketSinkInterface* sink) { in AddSinkOnlySsrc()
100 uint32_t ssrc, in CreatePacket()
108 std::unique_ptr<RtpPacketReceived> CreatePacketWithSsrc(uint32_t ssrc) { in CreatePacketWithSsrc()
113 uint32_t ssrc, in CreatePacketWithSsrcMid()
124 uint32_t ssrc, in CreatePacketWithSsrcRsid()
135 uint32_t ssrc, in CreatePacketWithSsrcRrid()
146 uint32_t ssrc, in CreatePacketWithSsrcMidRsid()
160 uint32_t ssrc, in CreatePacketWithSsrcRsidRrid()
187 constexpr uint32_t ssrc = 1; in TEST_F() local
257 constexpr uint32_t ssrc = 1; in TEST_F() local
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| D | rtp_demuxer.cc | 80 for (auto ssrc : ssrcs_) { in ToString() local
153 for (uint32_t ssrc : criteria.ssrcs()) { in AddSink() local
207 for (uint32_t ssrc : criteria.ssrcs()) { in CriteriaWouldConflict() local
236 bool RtpDemuxer::AddSink(uint32_t ssrc, RtpPacketSinkInterface* sink) { in AddSink()
280 uint32_t ssrc = packet.Ssrc(); in ResolveSink() local
367 uint32_t ssrc) { in ResolveSinkByMid()
379 uint32_t ssrc) { in ResolveSinkByMidRsid()
391 uint32_t ssrc) { in ResolveSinkByRsid()
403 uint32_t ssrc) { in ResolveSinkByPayloadType()
417 void RtpDemuxer::AddSsrcSinkBinding(uint32_t ssrc, in AddSsrcSinkBinding()
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| D | rtp_stream_receiver_controller.cc | 21 uint32_t ssrc, in Receiver()
43 RtpStreamReceiverController::CreateReceiver(uint32_t ssrc, in CreateReceiver()
53 bool RtpStreamReceiverController::AddSink(uint32_t ssrc, in AddSink()
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| /external/webrtc/media/base/ |
| D | fake_media_engine.cc | 23 FakeVoiceMediaChannel::DtmfInfo::DtmfInfo(uint32_t ssrc, in DtmfInfo()
106 bool FakeVoiceMediaChannel::SetAudioSend(uint32_t ssrc, in SetAudioSend()
131 bool FakeVoiceMediaChannel::RemoveRecvStream(uint32_t ssrc) { in RemoveRecvStream()
148 bool FakeVoiceMediaChannel::InsertDtmf(uint32_t ssrc, in InsertDtmf()
154 bool FakeVoiceMediaChannel::SetOutputVolume(uint32_t ssrc, double volume) { in SetOutputVolume()
167 bool FakeVoiceMediaChannel::GetOutputVolume(uint32_t ssrc, double* volume) { in GetOutputVolume()
173 bool FakeVoiceMediaChannel::SetBaseMinimumPlayoutDelayMs(uint32_t ssrc, in SetBaseMinimumPlayoutDelayMs()
195 uint32_t ssrc, in SetRawAudioSink()
234 bool FakeVoiceMediaChannel::SetLocalSource(uint32_t ssrc, AudioSource* source) { in SetLocalSource()
252 uint32_t ssrc, in CompareDtmfInfo()
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| D | fake_media_engine.h | 116 virtual bool RemoveSendStream(uint32_t ssrc) { in RemoveSendStream()
136 virtual bool RemoveRecvStream(uint32_t ssrc) { in RemoveRecvStream()
144 virtual webrtc::RtpParameters GetRtpSendParameters(uint32_t ssrc) const { in GetRtpSendParameters()
152 uint32_t ssrc, in SetRtpSendParameters()
174 virtual webrtc::RtpParameters GetRtpReceiveParameters(uint32_t ssrc) const { in GetRtpReceiveParameters()
185 bool IsStreamMuted(uint32_t ssrc) const { in IsStreamMuted()
200 bool HasRecvStream(uint32_t ssrc) const { in HasRecvStream()
203 bool HasSendStream(uint32_t ssrc) const { in HasSendStream()
242 bool MuteStream(uint32_t ssrc, bool mute) { in MuteStream()
324 uint32_t ssrc; member
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| D | fake_network_interface.h | 65 int NumRtpBytes(uint32_t ssrc) RTC_LOCKS_EXCLUDED(mutex_) { in NumRtpBytes()
77 int NumRtpPackets(uint32_t ssrc) RTC_LOCKS_EXCLUDED(mutex_) { in NumRtpPackets()
176 void SetRtpSsrc(uint32_t ssrc, rtc::CopyOnWriteBuffer& buffer) { in SetRtpSsrc()
181 void GetNumRtpBytesAndPackets(uint32_t ssrc, int* bytes, int* packets) { in GetNumRtpBytesAndPackets()
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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/modules/rtp_rtcp/source/ |
| D | receive_statistics_impl.cc | 37 StreamStatisticianImpl::StreamStatisticianImpl(uint32_t ssrc, Clock* clock, in StreamStatisticianImpl()
340 clock, [](uint32_t ssrc, Clock* clock, int max_reordering_threshold) { in Create()
349 clock, [](uint32_t ssrc, Clock* clock, int max_reordering_threshold) { in CreateThreadCompatible()
355 ReceiveStatisticsImpl::ReceiveStatisticsImpl( in ReceiveStatisticsImpl()
383 uint32_t ssrc) { in GetOrCreateStatistician()
402 uint32_t ssrc, in SetMaxReorderingThreshold()
408 void ReceiveStatisticsImpl::EnableRetransmitDetection(uint32_t ssrc, in EnableRetransmitDetection()
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| D | receive_statistics_impl.h | 120 StreamStatisticianLocked(uint32_t ssrc, in StreamStatisticianLocked()
210 explicit ReceiveStatisticsLocked( in ReceiveStatisticsLocked()
226 StreamStatistician* GetStatistician(uint32_t ssrc) const override { in GetStatistician()
234 void SetMaxReorderingThreshold(uint32_t ssrc, in SetMaxReorderingThreshold()
239 void EnableRetransmitDetection(uint32_t ssrc, bool enable) override { in EnableRetransmitDetection()
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| /external/webrtc/pc/ |
| D | video_rtp_receiver.cc | 115 void VideoRtpReceiver::RestartMediaChannel(absl::optional<uint32_t> ssrc) { in RestartMediaChannel()
127 absl::optional<uint32_t> ssrc, in RestartMediaChannel_w()
177 void VideoRtpReceiver::SetupMediaChannel(uint32_t ssrc) { in SetupMediaChannel()
187 uint32_t VideoRtpReceiver::ssrc() const { in ssrc() function in webrtc::VideoRtpReceiver
313 void VideoRtpReceiver::SetupMediaChannel(absl::optional<uint32_t> ssrc, in SetupMediaChannel()
355 const auto ssrc = ssrc_.value_or(0); in SetEncodedSinkEnabled() local
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| D | track_media_info_map_unittest.cc | 45 for (uint32_t ssrc : ssrcs) { in CreateRtpParametersWithSsrcs() local
130 for (uint32_t ssrc : ssrcs) { in AddRtpSenderWithSsrcs() local
138 for (uint32_t ssrc : ssrcs) { in AddRtpSenderWithSsrcs() local
159 for (uint32_t ssrc : ssrcs) { in AddRtpReceiverWithSsrcs() local
167 for (uint32_t ssrc : ssrcs) { in AddRtpReceiverWithSsrcs() local
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| D | audio_rtp_receiver.cc | 166 void AudioRtpReceiver::RestartMediaChannel(absl::optional<uint32_t> ssrc) { in RestartMediaChannel()
178 absl::optional<uint32_t> ssrc, in RestartMediaChannel_w()
205 void AudioRtpReceiver::SetupMediaChannel(uint32_t ssrc) { in SetupMediaChannel()
215 uint32_t AudioRtpReceiver::ssrc() const { in ssrc() function in webrtc::AudioRtpReceiver
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| /external/webrtc/video/ |
| D | encoder_rtcp_feedback.cc | 27 EncoderRtcpFeedback::EncoderRtcpFeedback( in EncoderRtcpFeedback()
48 void EncoderRtcpFeedback::OnReceivedIntraFrameRequest(uint32_t ssrc) { in OnReceivedIntraFrameRequest()
63 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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| /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/api/ |
| D | frame_transformer_interface.h | 102 uint32_t ssrc) {} in RegisterTransformedFrameSinkCallback()
104 virtual void UnregisterTransformedFrameSinkCallback(uint32_t ssrc) {} in UnregisterTransformedFrameSinkCallback()
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| /external/webrtc/modules/pacing/ |
| D | pacing_controller_unittest.cc | 57 uint32_t ssrc, in BuildPacket()
74 uint32_t ssrc, in MediaStream()
269 uint32_t ssrc, in SendAndExpectPacket()
513 uint32_t ssrc = 12345; in TEST_F() local
583 constexpr uint32_t ssrc = 333; in TEST_F() local
602 uint32_t ssrc = 12345; in TEST_F() local
622 uint32_t ssrc = 12345; in TEST_F() local
707 uint32_t ssrc = 12345; in TEST_F() local
726 uint32_t ssrc = 12345; in TEST_F() local
763 uint32_t ssrc = 12346; in TEST_F() 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 | 213 uint32_t ssrc, in SsrcUsed()
225 uint32_t ssrc; in WriteAudioRecvConfigs() local
242 uint32_t ssrc; in WriteAudioSendConfigs() local
267 uint32_t ssrc = prng_.Rand<uint32_t>(); in WriteVideoRecvConfigs() local
296 uint32_t ssrc = prng_.Rand<uint32_t>(); in WriteVideoSendConfigs() local
372 uint32_t ssrc = incoming_extensions_[stream].first; in WriteLog() local
456 uint32_t ssrc = incoming_extensions_[stream].first; in WriteLog() local
485 uint32_t ssrc = incoming_extensions_[stream].first; in WriteLog() local
497 uint32_t ssrc = outgoing_extensions_[stream].first; in WriteLog() local
596 uint32_t ssrc = kv.first; in ReadAndVerifyLog() local
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