/external/webrtc/webrtc/system_wrappers/source/ |
D | rtp_to_ntp_unittest.cc | 37 RtcpList rtcp; in TEST() local 43 rtcp.push_front(RtcpMeasurement(ntp_sec, ntp_frac, timestamp)); in TEST() 46 rtcp.push_front(RtcpMeasurement(ntp_sec, ntp_frac, timestamp)); in TEST() 52 EXPECT_FALSE(RtpToNtpMs(timestamp, rtcp, ×tamp_in_ms)); in TEST() 56 RtcpList rtcp; in TEST() local 62 rtcp.push_front(RtcpMeasurement(ntp_sec, ntp_frac, timestamp)); in TEST() 65 rtcp.push_front(RtcpMeasurement(ntp_sec, ntp_frac, timestamp)); in TEST() 67 EXPECT_TRUE(RtpToNtpMs(rtcp.back().rtp_timestamp, rtcp, ×tamp_in_ms)); in TEST() 76 RtcpList rtcp; in TEST() local 80 rtcp.push_front(RtcpMeasurement(ntp_sec, ntp_frac, timestamp)); in TEST() [all …]
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D | rtp_to_ntp.cc | 96 const RtcpList& rtcp, in RtpToNtpMs() argument 98 assert(rtcp.size() == 2); in RtpToNtpMs() 99 int64_t rtcp_ntp_ms_new = Clock::NtpToMs(rtcp.front().ntp_secs, in RtpToNtpMs() 100 rtcp.front().ntp_frac); in RtpToNtpMs() 101 int64_t rtcp_ntp_ms_old = Clock::NtpToMs(rtcp.back().ntp_secs, in RtpToNtpMs() 102 rtcp.back().ntp_frac); in RtpToNtpMs() 103 int64_t rtcp_timestamp_new = rtcp.front().rtp_timestamp; in RtpToNtpMs() 104 int64_t rtcp_timestamp_old = rtcp.back().rtp_timestamp; in RtpToNtpMs()
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/external/webrtc/webrtc/modules/rtp_rtcp/source/ |
D | rtcp_receiver_unittest.cc | 156 rtcp::SenderReport sr; in TEST_F() 158 rtc::scoped_ptr<rtcp::RawPacket> packet(sr.Build()); in TEST_F() 170 rtcp::SenderReport sr; in TEST_F() 172 rtc::scoped_ptr<rtcp::RawPacket> packet(sr.Build()); in TEST_F() 180 rtcp::ReceiverReport rr; in TEST_F() 182 rtc::scoped_ptr<rtcp::RawPacket> packet(rr.Build()); in TEST_F() 196 rtcp::ReportBlock rb; in TEST_F() 198 rtcp::ReceiverReport rr; in TEST_F() 201 rtc::scoped_ptr<rtcp::RawPacket> packet(rr.Build()); in TEST_F() 219 rtcp::ReportBlock rb; in TEST_F() [all …]
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D | rtcp_sender.cc | 83 class PacketContainer : public rtcp::CompoundPacket, 84 public rtcp::RtcpPacket::PacketReadyCallback { 99 rtcp::CompoundPacket::Build(this); in SendPackets() 460 rtc::scoped_ptr<rtcp::RtcpPacket> RTCPSender::BuildSR(const RtcpContext& ctx) { in BuildSR() 479 rtcp::SenderReport* report = new rtcp::SenderReport(); in BuildSR() 492 return rtc::scoped_ptr<rtcp::SenderReport>(report); in BuildSR() 495 rtc::scoped_ptr<rtcp::RtcpPacket> RTCPSender::BuildSDES( in BuildSDES() 500 rtcp::Sdes* sdes = new rtcp::Sdes(); in BuildSDES() 506 return rtc::scoped_ptr<rtcp::Sdes>(sdes); in BuildSDES() 509 rtc::scoped_ptr<rtcp::RtcpPacket> RTCPSender::BuildRR(const RtcpContext& ctx) { in BuildRR() [all …]
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D | rtcp_sender.h | 151 bool SendFeedbackPacket(const rtcp::TransportFeedback& packet); 166 rtc::scoped_ptr<rtcp::RtcpPacket> BuildSR(const RtcpContext& context) 168 rtc::scoped_ptr<rtcp::RtcpPacket> BuildRR(const RtcpContext& context) 170 rtc::scoped_ptr<rtcp::RtcpPacket> BuildSDES(const RtcpContext& context) 172 rtc::scoped_ptr<rtcp::RtcpPacket> BuildPLI(const RtcpContext& context) 174 rtc::scoped_ptr<rtcp::RtcpPacket> BuildREMB(const RtcpContext& context) 176 rtc::scoped_ptr<rtcp::RtcpPacket> BuildTMMBR(const RtcpContext& context) 178 rtc::scoped_ptr<rtcp::RtcpPacket> BuildTMMBN(const RtcpContext& context) 180 rtc::scoped_ptr<rtcp::RtcpPacket> BuildAPP(const RtcpContext& context) 182 rtc::scoped_ptr<rtcp::RtcpPacket> BuildVoIPMetric(const RtcpContext& context) [all …]
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D | rtcp_packet_unittest.cc | 24 using webrtc::rtcp::App; 25 using webrtc::rtcp::Bye; 26 using webrtc::rtcp::Dlrr; 27 using webrtc::rtcp::Fir; 28 using webrtc::rtcp::RawPacket; 29 using webrtc::rtcp::ReceiverReport; 30 using webrtc::rtcp::Remb; 31 using webrtc::rtcp::ReportBlock; 32 using webrtc::rtcp::Rpsi; 33 using webrtc::rtcp::Rrtr; [all …]
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/external/webrtc/webrtc/modules/remote_bitrate_estimator/ |
D | remote_estimator_proxy_unittest.cc | 27 MOCK_METHOD1(SendFeedback, bool(rtcp::TransportFeedback* packet)); 58 (rtcp::TransportFeedback::kDeltaScaleFactor * 0xFF) / 1000; 66 .WillOnce(Invoke([this](rtcp::TransportFeedback* packet) { in TEST_F() 71 std::vector<rtcp::TransportFeedback::StatusSymbol> status_vec = in TEST_F() 74 EXPECT_EQ(rtcp::TransportFeedback::StatusSymbol::kReceivedSmallDelta, in TEST_F() 92 .WillOnce(Invoke([this](rtcp::TransportFeedback* packet) { in TEST_F() 97 std::vector<rtcp::TransportFeedback::StatusSymbol> status_vec = in TEST_F() 100 EXPECT_EQ(rtcp::TransportFeedback::StatusSymbol::kReceivedSmallDelta, in TEST_F() 102 EXPECT_EQ(rtcp::TransportFeedback::StatusSymbol::kReceivedSmallDelta, in TEST_F() 104 EXPECT_EQ(rtcp::TransportFeedback::StatusSymbol::kReceivedLargeDelta, in TEST_F() [all …]
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D | transport_feedback_adapter_unittest.cc | 132 rtcp::TransportFeedback feedback; in TEST_F() 168 rtcp::TransportFeedback feedback; in TEST_F() 195 int64_t kHighArrivalTimeMs = rtcp::TransportFeedback::kDeltaScaleFactor * in TEST_F() 207 rtc::scoped_ptr<rtcp::TransportFeedback> feedback( in TEST_F() 208 new rtcp::TransportFeedback()); in TEST_F() 215 rtc::scoped_ptr<rtcp::RawPacket> raw_packet = feedback->Build(); in TEST_F() 216 feedback = rtcp::TransportFeedback::ParseFrom(raw_packet->Buffer(), in TEST_F() 235 rtcp::TransportFeedback::kDeltaScaleFactor * ((1 << 8) - 1); in TEST_F() 237 rtcp::TransportFeedback::kDeltaScaleFactor * in TEST_F() 240 rtcp::TransportFeedback::kDeltaScaleFactor * in TEST_F() [all …]
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D | transport_feedback_adapter.cc | 26 rtcp::TransportFeedback::kDeltaScaleFactor * (1 << 8); 67 const rtcp::TransportFeedback& feedback) { in OnTransportFeedback() 99 if (symbol != rtcp::TransportFeedback::StatusSymbol::kNotReceived) { in OnTransportFeedback()
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/external/webrtc/webrtc/modules/rtp_rtcp/source/rtcp_packet/ |
D | compound_packet_unittest.cc | 19 using webrtc::rtcp::Bye; 20 using webrtc::rtcp::CompoundPacket; 21 using webrtc::rtcp::Fir; 22 using webrtc::rtcp::RawPacket; 23 using webrtc::rtcp::ReceiverReport; 24 using webrtc::rtcp::ReportBlock; 25 using webrtc::rtcp::SenderReport; 97 class Verifier : public rtcp::RtcpPacket::PacketReadyCallback { in TEST() 127 class Verifier : public rtcp::RtcpPacket::PacketReadyCallback { in TEST()
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D | nack_unittest.cc | 21 using webrtc::rtcp::Nack; 22 using webrtc::rtcp::RawPacket; 132 class MockPacketReadyCallback : public rtcp::RtcpPacket::PacketReadyCallback { in TEST() 169 class Verifier : public rtcp::RtcpPacket::PacketReadyCallback { in TEST()
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D | tmmbr_unittest.cc | 18 using webrtc::rtcp::RawPacket; 19 using webrtc::rtcp::Tmmbr;
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D | receiver_report_unittest.cc | 15 using webrtc::rtcp::RawPacket; 16 using webrtc::rtcp::ReceiverReport; 17 using webrtc::rtcp::ReportBlock;
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D | pli_unittest.cc | 16 using webrtc::rtcp::Pli; 17 using webrtc::rtcp::RawPacket;
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/external/webrtc/talk/session/media/ |
D | channel.cc | 117 static const char* PacketType(bool rtcp) { in PacketType() argument 118 return (!rtcp) ? "RTP" : "RTCP"; in PacketType() 121 static bool ValidPacket(bool rtcp, const rtc::Buffer* packet) { in ValidPacket() argument 124 packet->size() >= (!rtcp ? kMinRtpPacketLen : kMinRtcpPacketLen) && in ValidPacket() 158 params->rtcp.reduced_size = desc->rtcp_reduced_size(); in RtpParametersFromMediaDescription() 173 bool rtcp) in BaseChannel() argument 178 rtcp_transport_enabled_(rtcp), in BaseChannel() 505 bool rtcp = PacketIsRtcp(channel, data, len); in OnChannelRead() local 507 HandlePacket(rtcp, &packet, packet_time); in OnChannelRead() 531 void BaseChannel::SetReadyToSend(bool rtcp, bool ready) { in SetReadyToSend() argument [all …]
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D | channelmanager.h | 104 bool rtcp, 114 bool rtcp, 120 bool rtcp, 195 bool rtcp, 202 bool rtcp, 207 bool rtcp,
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D | channelmanager.cc | 254 bool rtcp, in CreateVoiceChannel() argument 258 transport_controller, content_name, rtcp, options)); in CreateVoiceChannel() 265 bool rtcp, in CreateVoiceChannel_w() argument 277 transport_controller, content_name, rtcp); in CreateVoiceChannel_w() 312 bool rtcp, in CreateVideoChannel() argument 316 transport_controller, content_name, rtcp, options)); in CreateVideoChannel() 323 bool rtcp, in CreateVideoChannel_w() argument 335 worker_thread_, media_channel, transport_controller, content_name, rtcp); in CreateVideoChannel_w() 370 bool rtcp, in CreateDataChannel() argument 374 content_name, rtcp, channel_type)); in CreateDataChannel() [all …]
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D | channel.h | 89 bool rtcp); 162 void SignalDtlsSetupFailure_w(bool rtcp); 163 void SignalDtlsSetupFailure_s(bool rtcp); 169 void SetReadyToSend(bool rtcp, bool ready); 228 bool SendPacket(bool rtcp, 231 virtual bool WantsPacket(bool rtcp, rtc::Buffer* packet); 232 void HandlePacket(bool rtcp, rtc::Buffer* packet, 250 bool SetDtlsSrtpCryptoSuites(TransportChannel* tc, bool rtcp); 347 bool rtcp); 449 bool rtcp); [all …]
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/external/webrtc/webrtc/api/objctests/ |
D | RTCSessionDescriptionTest.mm | 71 "a=rtcp:9 IN IP4 0.0.0.0\r\n" 82 "a=rtcp-mux\r\n" 97 "a=rtcp:9 IN IP4 0.0.0.0\r\n" 109 "a=rtcp-mux\r\n" 111 "a=rtcp-fb:100 ccm fir\r\n" 112 "a=rtcp-fb:100 nack\r\n" 113 "a=rtcp-fb:100 nack pli\r\n" 114 "a=rtcp-fb:100 goog-remb\r\n"
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/external/webrtc/webrtc/video/ |
D | stream_synchronization_unittest.cc | 38 RtcpMeasurement rtcp; in GenerateRtcp() local 39 NowNtp(&rtcp.ntp_secs, &rtcp.ntp_frac); in GenerateRtcp() 40 rtcp.rtp_timestamp = NowRtp(frequency, offset); in GenerateRtcp() 41 return rtcp; in GenerateRtcp() 105 audio.rtcp.push_front(send_time_->GenerateRtcp(audio_frequency, in DelayedStreams() 109 video.rtcp.push_front(send_time_->GenerateRtcp(video_frequency, in DelayedStreams() 113 audio.rtcp.push_front(send_time_->GenerateRtcp(audio_frequency, in DelayedStreams() 117 video.rtcp.push_front(send_time_->GenerateRtcp(video_frequency, in DelayedStreams()
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D | stream_synchronization.cc | 63 if (audio_measurement.rtcp.size() < 2 || video_measurement.rtcp.size() < 2) { in ComputeRelativeDelay() 69 audio_measurement.rtcp, in ComputeRelativeDelay() 75 video_measurement.rtcp, in ComputeRelativeDelay()
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D | stream_synchronization.h | 26 Measurements() : rtcp(), latest_receive_time_ms(0), latest_timestamp(0) {} in Measurements() 27 RtcpList rtcp; member
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/external/webrtc/talk/media/base/ |
D | rtpdump.h | 74 RtpDumpPacket(const void* d, size_t s, uint32_t elapsed, bool rtcp) in RtpDumpPacket() 75 : elapsed_time(elapsed), original_data_len((rtcp) ? 0 : s) { in RtpDumpPacket() 219 bool rtcp); 220 size_t FilterPacket(const void* data, size_t data_len, bool rtcp);
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D | rtpdump.cc | 355 bool rtcp) { in WritePacket() argument 369 size_t write_len = FilterPacket(data, data_len, rtcp); in WritePacket() 378 buf.WriteUInt16(static_cast<uint16_t>(rtcp ? 0 : data_len)); in WritePacket() 390 bool rtcp) { in FilterPacket() argument 392 if (!rtcp) { in FilterPacket()
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/external/webrtc/webrtc/modules/pacing/ |
D | packet_router.h | 27 namespace rtcp { 54 virtual bool SendFeedback(rtcp::TransportFeedback* packet);
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