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Searched refs:ssrc_ (Results 1 – 25 of 56) sorted by relevance

123

/external/webrtc/pc/
Dvideo_rtp_receiver.cc79 return ssrc_ ? media_channel_->GetRtpReceiveParameters(*ssrc_) in GetParameters()
88 if (media_channel_ && ssrc_) { in SetFrameDecryptor()
89 media_channel_->SetFrameDecryptor(*ssrc_, frame_decryptor_); in SetFrameDecryptor()
105 ssrc_.value_or(0), frame_transformer_); in SetDepacketizerToDecoderFrameTransformer()
137 if (ssrc == ssrc_) in RestartMediaChannel_w()
148 ssrc_ = std::move(ssrc); in RestartMediaChannel_w()
156 ssrc_.value_or(0), frame_transformer_); in RestartMediaChannel_w()
159 if (media_channel_ && ssrc_) { in RestartMediaChannel_w()
161 media_channel_->SetFrameDecryptor(*ssrc_, frame_decryptor_); in RestartMediaChannel_w()
164 media_channel_->SetBaseMinimumPlayoutDelayMs(*ssrc_, delay_.GetMs()); in RestartMediaChannel_w()
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Daudio_rtp_receiver.cc93 ssrc_ ? media_channel_->SetOutputVolume(*ssrc_, volume) in SetOutputVolume_w()
140 return ssrc_ ? media_channel_->GetRtpReceiveParameters(*ssrc_) in GetParameters()
149 if (media_channel_ && ssrc_) { in SetFrameDecryptor()
150 media_channel_->SetFrameDecryptor(*ssrc_, frame_decryptor_); in SetFrameDecryptor()
191 if (ssrc_ == ssrc) in RestartMediaChannel_w()
193 source_->Stop(media_channel_, ssrc_); in RestartMediaChannel_w()
196 ssrc_ = std::move(ssrc); in RestartMediaChannel_w()
197 source_->Start(media_channel_, ssrc_); in RestartMediaChannel_w()
198 if (ssrc_) { in RestartMediaChannel_w()
199 media_channel_->SetBaseMinimumPlayoutDelayMs(*ssrc_, delay_.GetMs()); in RestartMediaChannel_w()
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Drtp_sender.cc166 if (media_channel_ && ssrc_ && !stopped_) { in SetFrameEncryptor()
168 [&] { media_channel_->SetFrameEncryptor(ssrc_, frame_encryptor_); }); in SetFrameEncryptor()
182 if (media_channel_ && ssrc_ && !stopped_) { in SetEncoderSelectorOnChannel()
184 media_channel_->SetEncoderSelector(ssrc_, encoder_selector_.get()); in SetEncoderSelectorOnChannel()
200 if (!media_channel_ || !ssrc_) { in GetParametersInternal()
204 RtpParameters result = media_channel_->GetRtpSendParameters(ssrc_); in GetParametersInternal()
215 if (!media_channel_ || !ssrc_) { in GetParametersInternalWithAllLayers()
219 RtpParameters result = media_channel_->GetRtpSendParameters(ssrc_); in GetParametersInternalWithAllLayers()
247 if (!media_channel_ || !ssrc_) { in SetParametersInternal()
259 RtpParameters old_parameters = media_channel_->GetRtpSendParameters(ssrc_); in SetParametersInternal()
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/external/webrtc/video/
Drtp_video_stream_receiver_frame_transformer_delegate.cc30 ssrc_(ssrc) {} in TransformableVideoReceiverFrame()
44 uint32_t GetSsrc() const override { return ssrc_; } in GetSsrc()
66 const uint32_t ssrc_; member in webrtc::__anon3d4c0f670111::TransformableVideoReceiverFrame
79 ssrc_(ssrc) {} in RtpVideoStreamReceiverFrameTransformerDelegate()
84 rtc::scoped_refptr<TransformedFrameCallback>(this), ssrc_); in Init()
89 frame_transformer_->UnregisterTransformedFrameSinkCallback(ssrc_); in Reset()
99 ssrc_)); in TransformFrame()
/external/webrtc/modules/rtp_rtcp/source/rtcp_packet/
Dtmmb_item.h26 TmmbItem() : ssrc_(0), bitrate_bps_(0), packet_overhead_(0) {} in TmmbItem()
32 void set_ssrc(uint32_t ssrc) { ssrc_ = ssrc; } in set_ssrc()
36 uint32_t ssrc() const { return ssrc_; } in ssrc()
42 uint32_t ssrc_;
Dtmmb_item.cc20 : ssrc_(ssrc), bitrate_bps_(bitrate_bps), packet_overhead_(overhead) { in TmmbItem()
32 ssrc_ = ByteReader<uint32_t>::ReadBigEndian(&buffer[0]); in Parse()
61 ByteWriter<uint32_t>::WriteBigEndian(&buffer[0], ssrc_); in Create()
/external/webrtc/modules/rtp_rtcp/source/
Drtcp_sender.cc144 ssrc_(config.local_media_ssrc), in RTCPSender()
345 return ssrc_; in SSRC()
350 ssrc_ = ssrc; in SetSsrc()
463 report.SetSenderSsrc(ssrc_); in BuildSR()
477 sdes.AddCName(ssrc_, cname_); in BuildSDES()
483 report.SetSenderSsrc(ssrc_); in BuildRR()
492 pli.SetSenderSsrc(ssrc_); in BuildPLI()
503 fir.SetSenderSsrc(ssrc_); in BuildFIR()
512 remb.SetSenderSsrc(ssrc_); in BuildREMB()
551 candidates.emplace_back(ssrc_, tmmbr_send_bps_, packet_oh_send_); in BuildTMMBR()
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Drtp_sender_video_frame_transformer_delegate.cc44 ssrc_(ssrc) { in TransformableVideoSenderFrame()
61 uint32_t GetSsrc() const override { return ssrc_; } in GetSsrc()
94 const uint32_t ssrc_; member in webrtc::__anon2ea014070111::TransformableVideoSenderFrame
105 ssrc_(ssrc), in RTPSenderVideoFrameTransformerDelegate()
112 rtc::scoped_refptr<TransformedFrameCallback>(this), ssrc_); in Init()
124 expected_retransmission_time_ms, ssrc_)); in TransformFrame()
177 frame_transformer_->UnregisterTransformedFrameSinkCallback(ssrc_); in Reset()
Dreceive_statistics_impl.cc39 : ssrc_(ssrc), in StreamStatisticianImpl()
112 RTC_DCHECK_EQ(ssrc_, packet.Ssrc()); in UpdateCounters()
244 stats.SetMediaSsrc(ssrc_); in MaybeAppendReportBlockAndReset()
268 << ssrc_; in MaybeAppendReportBlockAndReset()
281 cumulative_loss_, ssrc_); in MaybeAppendReportBlockAndReset()
284 ssrc_); in MaybeAppendReportBlockAndReset()
Drtp_sender_egress.cc71 ssrc_(config.local_media_ssrc), in RtpSenderEgress()
122 if (packet->Ssrc() == ssrc_ && in SendPacket()
395 return packet.Ssrc() == ssrc_; in HasCorrectSsrc()
400 return packet.Ssrc() == rtx_ssrc_ || packet.Ssrc() == ssrc_; in HasCorrectSsrc()
403 return packet.Ssrc() == ssrc_ || packet.Ssrc() == flexfec_ssrc_; in HasCorrectSsrc()
423 packet_info.media_ssrc = ssrc_; in AddPacketToTransportFeedback()
429 packet_info.media_ssrc = ssrc_; in AddPacketToTransportFeedback()
601 send_rates[RtpPacketMediaType::kRetransmission].bps(), ssrc_); in UpdateRtpStats()
611 send_rates[RtpPacketMediaType::kRetransmission].bps(), ssrc_); in PeriodicUpdate()
Dflexfec_receiver.cc46 : ssrc_(ssrc), in FlexfecReceiver()
100 if (received_packet->ssrc == ssrc_) { in AddReceivedPacket()
180 << " from FlexFEC stream with SSRC: " << ssrc_; in ProcessReceivedPacket()
Dfec_test_helper.cc42 ssrc_(ssrc), in MediaPacketGenerator()
82 webrtc::ByteWriter<uint32_t>::WriteBigEndian(&data[8], ssrc_); in ConstructMediaPackets()
110 : num_packets_(0), ssrc_(ssrc), seq_num_(0), timestamp_(0) {} in AugmentedPacketGenerator()
136 packet->header.ssrc = ssrc_; in NextPacket()
/external/webrtc/logging/rtc_event_log/events/
Drtc_event_audio_playout.cc22 RtcEventAudioPlayout::RtcEventAudioPlayout(uint32_t ssrc) : ssrc_(ssrc) {} in RtcEventAudioPlayout()
25 : RtcEvent(other.timestamp_us_), ssrc_(other.ssrc_) {} in RtcEventAudioPlayout()
Drtc_event_frame_decoded.cc24 ssrc_(ssrc), in RtcEventFrameDecoded()
33 ssrc_(other.ssrc_), in RtcEventFrameDecoded()
Drtc_event_audio_playout.h53 uint32_t ssrc() const { return ssrc_; } in ssrc()
75 const uint32_t ssrc_; variable
81 {&RtcEventAudioPlayout::ssrc_,
Drtc_event_frame_decoded.h61 uint32_t ssrc() const { return ssrc_; } in ssrc()
84 const uint32_t ssrc_; variable
/external/webrtc/api/
Drtp_packet_info.cc19 : ssrc_(0), rtp_timestamp_(0), receive_time_(Timestamp::MinusInfinity()) {} in RtpPacketInfo()
25 : ssrc_(ssrc), in RtpPacketInfo()
32 : ssrc_(rtp_header.ssrc), in RtpPacketInfo()
Drtp_packet_info.h47 uint32_t ssrc() const { return ssrc_; } in ssrc()
48 void set_ssrc(uint32_t value) { ssrc_ = value; } in set_ssrc()
86 uint32_t ssrc_;
/external/webrtc/video/end_to_end_tests/
Dmulti_stream_tests.cc37 : settings_(settings), ssrc_(ssrc), frame_generator_(frame_generator) {} in TEST()
46 uint32_t Ssrc() { return ssrc_; } in TEST()
52 const uint32_t ssrc_; in TEST() member in webrtc::TEST::VideoOutputObserver
/external/webrtc/modules/audio_coding/neteq/tools/
Drtp_encode.cc105 ssrc_(ssrc), in Packetizer()
137 static_cast<uint8_t>(ssrc_ >> 24), in SendData()
138 static_cast<uint8_t>(ssrc_ >> 16), in SendData()
139 static_cast<uint8_t>(ssrc_ >> 8), in SendData()
140 static_cast<uint8_t>(ssrc_)}; in SendData()
156 const uint32_t ssrc_; member in webrtc::test::__anon6460c84b0111::Packetizer
Drtp_generator.h30 ssrc_(ssrc), in seq_number_()
52 const uint32_t ssrc_; variable
/external/webrtc/audio/
Dchannel_receive_frame_transformer_delegate.cc28 ssrc_(ssrc) {} in TransformableIncomingAudioFrame()
37 uint32_t GetSsrc() const override { return ssrc_; } in GetSsrc()
48 uint32_t ssrc_; member in webrtc::__anon0a35ebb20111::TransformableIncomingAudioFrame
Dchannel_send_frame_transformer_delegate.cc34 ssrc_(ssrc) {} in TransformableOutgoingAudioFrame()
44 uint32_t GetSsrc() const override { return ssrc_; } in GetSsrc()
60 uint32_t ssrc_; member in webrtc::__anona59aa3f90111::TransformableOutgoingAudioFrame
/external/webrtc/modules/rtp_rtcp/source/deprecated/
Ddeprecated_rtp_sender_egress.cc65 : ssrc_(config.local_media_ssrc), in DEPRECATED_RtpSenderEgress()
216 send_rates[RtpPacketMediaType::kRetransmission].bps(), ssrc_); in ProcessBitrateAndNotifyObservers()
294 return packet.Ssrc() == ssrc_; in HasCorrectSsrc()
299 return packet.Ssrc() == rtx_ssrc_ || packet.Ssrc() == ssrc_; in HasCorrectSsrc()
302 return packet.Ssrc() == ssrc_ || packet.Ssrc() == flexfec_ssrc_; in HasCorrectSsrc()
313 packet_info.media_ssrc = ssrc_; in AddPacketToTransportFeedback()
/external/webrtc/modules/rtp_rtcp/include/
Dflexfec_sender.h53 absl::optional<uint32_t> FecSsrc() override { return ssrc_; } in FecSsrc()
85 const uint32_t ssrc_; variable

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