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Searched refs:rtp_rtcp_ (Results 1 – 21 of 21) sorted by relevance

/external/webrtc/audio/voip/
Daudio_channel.cc54 rtp_rtcp_ = ModuleRtpRtcpImpl2::Create(rtp_config); in AudioChannel()
56 rtp_rtcp_->SetSendingMediaStatus(false); in AudioChannel()
57 rtp_rtcp_->SetRTCPStatus(RtcpMode::kCompound); in AudioChannel()
60 process_thread_->RegisterModule(rtp_rtcp_.get(), RTC_FROM_HERE); in AudioChannel()
62 ingress_ = std::make_unique<AudioIngress>(rtp_rtcp_.get(), clock, in AudioChannel()
66 std::make_unique<AudioEgress>(rtp_rtcp_.get(), clock, task_queue_factory); in AudioChannel()
82 process_thread_->DeRegisterModule(rtp_rtcp_.get()); in ~AudioChannel()
89 if (!rtp_rtcp_->Sending() && rtp_rtcp_->SetSendingStatus(true) != 0) { in StartSend()
100 if (!IsPlaying() && rtp_rtcp_->Sending() && in StopSend()
101 rtp_rtcp_->SetSendingStatus(false) != 0) { in StopSend()
[all …]
Daudio_egress.cc23 : rtp_rtcp_(rtp_rtcp), in AudioEgress()
24 rtp_sender_audio_(clock, rtp_rtcp_->RtpSender()), in AudioEgress()
37 return rtp_rtcp_->SendingMedia(); in IsSending()
50 rtp_rtcp_->RegisterSendPayloadFrequency(payload_type, in SetEncoder()
60 rtp_rtcp_->SetSendingMediaStatus(true); in StartSend()
64 rtp_rtcp_->SetSendingMediaStatus(false); in StopSend()
74 if (!rtp_rtcp_->SendingMedia()) { in SendAudioData()
115 if (!rtp_rtcp_->OnSendingRtpFrame(timestamp, kUndefinedCaptureTime, in SendData()
121 const uint32_t rtp_timestamp = timestamp + rtp_rtcp_->StartTimestamp(); in SendData()
139 rtp_rtcp_->RegisterSendPayloadFrequency(rtp_payload_type, sample_rate_hz); in RegisterTelephoneEventType()
Daudio_ingress.cc46 rtp_rtcp_(rtp_rtcp), in AudioIngress()
171 rtp_rtcp_->IncomingRtcpPacket(rtcp_packet.data(), rtcp_packet.size()); in ReceivedRTCPPacket()
180 if (rtp_rtcp_->RemoteNTP(&ntp_secs, &ntp_frac, nullptr, nullptr, in ReceivedRTCPPacket()
194 rtp_rtcp_->GetLatestReportBlockData(); in GetRoundTripTime()
212 rtp_rtcp_->SetRemoteSSRC(sender_ssrc); in GetRoundTripTime()
Daudio_egress.h113 RtpRtcpInterface* const rtp_rtcp_; variable
Daudio_channel.h90 std::unique_ptr<ModuleRtpRtcpImpl2> rtp_rtcp_; variable
Daudio_ingress.h126 RtpRtcpInterface* const rtp_rtcp_; variable
/external/webrtc/audio/
Dchannel_send.cc196 std::unique_ptr<ModuleRtpRtcpImpl2> rtp_rtcp_; member in webrtc::voe::__anonbb1b0d450111::ChannelSend
356 frameType, payloadType, rtp_timestamp, rtp_rtcp_->StartTimestamp(), in SendData()
358 rtp_rtcp_->SSRC()); in SendData()
395 cricket::MEDIA_TYPE_AUDIO, rtp_rtcp_->SSRC(), in SendRtpAudio()
417 if (!rtp_rtcp_->OnSendingRtpFrame(rtp_timestamp, in SendRtpAudio()
434 frameType, payloadType, rtp_timestamp + rtp_rtcp_->StartTimestamp(), in SendRtpAudio()
498 rtp_rtcp_ = ModuleRtpRtcpImpl2::Create(configuration); in ChannelSend()
499 rtp_rtcp_->SetSendingMediaStatus(false); in ChannelSend()
502 rtp_rtcp_->RtpSender()); in ChannelSend()
504 _moduleProcessThreadPtr->RegisterModule(rtp_rtcp_.get(), RTC_FROM_HERE); in ChannelSend()
[all …]
Dchannel_receive.cc219 std::unique_ptr<ModuleRtpRtcpImpl2> rtp_rtcp_; member in webrtc::voe::__anon5d97615e0111::ChannelReceive
300 rtp_rtcp_->RTT(remote_ssrc_, &round_trip_time, NULL, NULL, NULL); in OnReceivedPayloadData()
510 rtp_rtcp_ = ModuleRtpRtcpImpl2::Create(configuration); in ChannelReceive()
511 rtp_rtcp_->SetSendingMediaStatus(false); in ChannelReceive()
512 rtp_rtcp_->SetRemoteSSRC(remote_ssrc_); in ChannelReceive()
514 _moduleProcessThreadPtr->RegisterModule(rtp_rtcp_.get(), RTC_FROM_HERE); in ChannelReceive()
517 rtp_rtcp_->SetRTCPStatus(RtcpMode::kCompound); in ChannelReceive()
530 _moduleProcessThreadPtr->DeRegisterModule(rtp_rtcp_.get()); in ~ChannelReceive()
662 rtp_rtcp_->IncomingRtcpPacket(data, length); in ReceivedRTCPPacket()
674 rtp_rtcp_->RemoteNTP(&ntp_secs, &ntp_frac, NULL, NULL, &rtp_timestamp)) { in ReceivedRTCPPacket()
[all …]
Daudio_send_stream_unittest.cc206 MockRtpRtcpInterface* rtp_rtcp() { return &rtp_rtcp_; } in rtp_rtcp()
220 return &this->rtp_rtcp_; in SetupDefaultChannelSend()
222 EXPECT_CALL(rtp_rtcp_, SSRC).WillRepeatedly(Return(kSsrc)); in SetupDefaultChannelSend()
227 EXPECT_CALL(rtp_rtcp_, SetExtmapAllowMixed(false)).Times(1); in SetupDefaultChannelSend()
234 EXPECT_CALL(rtp_rtcp_, in SetupDefaultChannelSend()
248 EXPECT_CALL(rtp_rtcp_, SetRid(std::string())).Times(1); in SetupDefaultChannelSend()
335 ::testing::NiceMock<MockRtpRtcpInterface> rtp_rtcp_;
/external/webrtc/audio/voip/test/
Daudio_ingress_unittest.cc49 rtp_rtcp_ = ModuleRtpRtcpImpl2::Create(rtp_config); in AudioIngressTest()
51 rtp_rtcp_->SetSendingMediaStatus(false); in AudioIngressTest()
52 rtp_rtcp_->SetRTCPStatus(RtcpMode::kCompound); in AudioIngressTest()
61 ingress_ = std::make_unique<AudioIngress>(rtp_rtcp_.get(), &fake_clock_, in SetUp()
66 egress_ = std::make_unique<AudioEgress>(rtp_rtcp_.get(), &fake_clock_, in SetUp()
73 rtp_rtcp_->SetSendingStatus(true); in SetUp()
77 rtp_rtcp_->SetSendingStatus(false); in TearDown()
98 std::unique_ptr<ModuleRtpRtcpImpl2> rtp_rtcp_; member in webrtc::__anonc756ecca0111::AudioIngressTest
Daudio_egress_unittest.cc59 rtp_rtcp_ = CreateRtpStack(&fake_clock_, &transport_, kRemoteSsrc); in AudioEgressTest()
67 egress_ = std::make_unique<AudioEgress>(rtp_rtcp_.get(), &fake_clock_, in SetUp()
74 rtp_rtcp_->SetSequenceNumber(kSeqNum); in SetUp()
75 rtp_rtcp_->SetSendingStatus(true); in SetUp()
81 rtp_rtcp_->SetSendingStatus(false); in TearDown()
104 std::unique_ptr<ModuleRtpRtcpImpl2> rtp_rtcp_; member in webrtc::__anon7c5499780111::AudioEgressTest
/external/webrtc/video/end_to_end_tests/
Dbandwidth_tests.cc211 SendTask(RTC_FROM_HERE, task_queue_, [this]() { rtp_rtcp_ = nullptr; }); in TEST_F()
248 rtp_rtcp_ = ModuleRtpRtcpImpl2::Create(config); in TEST_F()
249 rtp_rtcp_->SetRemoteSSRC((*receive_configs)[0].rtp.remote_ssrc); in TEST_F()
250 rtp_rtcp_->SetRTCPStatus(RtcpMode::kReducedSize); in TEST_F()
266 rtp_rtcp_->SetRemb( in TEST_F()
269 rtp_rtcp_->SendRTCP(kRtcpRr); in TEST_F()
277 rtp_rtcp_->SetRemb( in TEST_F()
280 rtp_rtcp_->SendRTCP(kRtcpRr); in TEST_F()
307 std::unique_ptr<ModuleRtpRtcpImpl2> rtp_rtcp_; in TEST_F() member in webrtc::TEST_F::BweObserver
/external/webrtc/call/
Dflexfec_receive_stream_impl.cc151 rtp_rtcp_(CreateRtpRtcpModule(clock, in FlexfecReceiveStreamImpl()
159 rtp_rtcp_->SetRTCPStatus(config_.rtcp_mode); in FlexfecReceiveStreamImpl()
160 process_thread_->RegisterModule(rtp_rtcp_.get(), RTC_FROM_HERE); in FlexfecReceiveStreamImpl()
179 process_thread_->DeRegisterModule(rtp_rtcp_.get()); in ~FlexfecReceiveStreamImpl()
Dflexfec_receive_stream_impl.h59 const std::unique_ptr<ModuleRtpRtcpImpl2> rtp_rtcp_; variable
/external/webrtc/video/
Drtp_video_stream_receiver2.cc231 rtp_rtcp_(CreateRtpRtcpModule(clock, in RtpVideoStreamReceiver2()
254 packet_router_->AddReceiveRtpModule(rtp_rtcp_.get(), remb_candidate); in RtpVideoStreamReceiver2()
263 rtp_rtcp_->SetRTCPStatus(config_.rtp.rtcp_mode); in RtpVideoStreamReceiver2()
264 rtp_rtcp_->SetRemoteSSRC(config_.rtp.remote_ssrc); in RtpVideoStreamReceiver2()
281 rtp_rtcp_->SetRtcpXrRrtrStatus(true); in RtpVideoStreamReceiver2()
287 process_thread_->RegisterModule(rtp_rtcp_.get(), RTC_FROM_HERE); in RtpVideoStreamReceiver2()
319 process_thread_->DeRegisterModule(rtp_rtcp_.get()); in ~RtpVideoStreamReceiver2()
322 packet_router_->RemoveReceiveRtpModule(rtp_rtcp_.get()); in ~RtpVideoStreamReceiver2()
343 if (rtp_rtcp_->RemoteNTP(&info.capture_time_ntp_secs, in GetSyncInfo()
684 rtp_rtcp_->SendPictureLossIndication(); in RequestKeyFrame()
[all …]
Drtp_video_stream_receiver.cc258 rtp_rtcp_(CreateRtpRtcpModule(clock, in RtpVideoStreamReceiver()
276 packet_router_->AddReceiveRtpModule(rtp_rtcp_.get(), remb_candidate); in RtpVideoStreamReceiver()
285 rtp_rtcp_->SetRTCPStatus(config_.rtp.rtcp_mode); in RtpVideoStreamReceiver()
286 rtp_rtcp_->SetRemoteSSRC(config_.rtp.remote_ssrc); in RtpVideoStreamReceiver()
303 rtp_rtcp_->SetRtcpXrRrtrStatus(true); in RtpVideoStreamReceiver()
309 process_thread_->RegisterModule(rtp_rtcp_.get(), RTC_FROM_HERE); in RtpVideoStreamReceiver()
351 process_thread_->DeRegisterModule(rtp_rtcp_.get()); in ~RtpVideoStreamReceiver()
354 packet_router_->RemoveReceiveRtpModule(rtp_rtcp_.get()); in ~RtpVideoStreamReceiver()
373 if (rtp_rtcp_->RemoteNTP(&info.capture_time_ntp_secs, in GetSyncInfo()
716 rtp_rtcp_->SendPictureLossIndication(); in RequestKeyFrame()
[all …]
Drtp_video_stream_receiver2.h293 const std::unique_ptr<ModuleRtpRtcpImpl2> rtp_rtcp_; variable
Drtp_video_stream_receiver.h330 const std::unique_ptr<RtpRtcp> rtp_rtcp_; variable
Dvideo_send_stream_tests.cc1639 SendTask(RTC_FROM_HERE, task_queue_, [this]() { rtp_rtcp_ = nullptr; }); in TEST_F()
1660 rtp_rtcp_->SetRemb(kRembBitrateBps, {rtp_packet.Ssrc()}); in TEST_F()
1661 rtp_rtcp_->Process(); in TEST_F()
1680 rtp_rtcp_ = ModuleRtpRtcpImpl2::Create(config); in TEST_F()
1681 rtp_rtcp_->SetRTCPStatus(RtcpMode::kReducedSize); in TEST_F()
1700 std::unique_ptr<ModuleRtpRtcpImpl2> rtp_rtcp_; in TEST_F() member in webrtc::TEST_F::BitrateObserver
/external/webrtc/modules/rtp_rtcp/source/
Drtcp_receiver.cc161 rtp_rtcp_(owner), in RTCPReceiver()
1049 rtp_rtcp_->SetTmmbn(std::move(bounding)); in NotifyTmmbrUpdated()
1070 rtp_rtcp_->OnRequestSendReport(); in TriggerCallbacksFromRtcpPacket()
1076 rtp_rtcp_->OnReceivedNack(packet_information.nack_sequence_numbers); in TriggerCallbacksFromRtcpPacket()
1128 rtp_rtcp_->OnReceivedRtcpReportBlocks(packet_information.report_blocks); in TriggerCallbacksFromRtcpPacket()
Drtcp_receiver.h226 ModuleRtpRtcp* const rtp_rtcp_; variable