| /external/chromium_org/third_party/webrtc/system_wrappers/source/ |
| D | rtp_to_ntp_unittest.cc | 37 RtcpList rtcp; in TEST() local
56 RtcpList rtcp; in TEST() local
76 RtcpList rtcp; in TEST() local
96 RtcpList rtcp; in TEST() local
113 RtcpList rtcp; in TEST() local
133 RtcpList rtcp; in TEST() local
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| D | rtp_to_ntp.cc | 96 const RtcpList& rtcp, in RtpToNtpMs()
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| /external/chromium_org/media/cast/rtcp/ |
| D | rtcp_unittest.cc | 41 void set_rtcp_receiver(Rtcp* rtcp) { rtcp_receiver_ = rtcp; } in set_rtcp_receiver()
80 void set_rtcp_receiver(Rtcp* rtcp) { rtcp_ = rtcp; } in set_rtcp_receiver()
213 Rtcp rtcp(cast_environment_, in TEST_F() local
235 Rtcp rtcp(cast_environment_, in TEST_F() local
251 Rtcp rtcp(cast_environment_, in TEST_F() local
270 Rtcp rtcp(cast_environment_, in TEST_F() local
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| D | rtcp.cc | 26 explicit LocalRtcpRttFeedback(Rtcp* rtcp) : rtcp_(rtcp) {} in LocalRtcpRttFeedback()
41 LocalRtcpReceiverFeedback(Rtcp* rtcp, in LocalRtcpReceiverFeedback()
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| /external/chromium_org/third_party/webrtc/video_engine/ |
| D | stream_synchronization.h | 27 RtcpList rtcp; member
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| D | stream_synchronization_unittest.cc | 37 RtcpMeasurement rtcp; in GenerateRtcp() local
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| /external/chromium_org/third_party/libjingle/source/talk/session/media/ |
| D | bundlefilter.cc | 43 bool BundleFilter::DemuxPacket(const char* data, size_t len, bool rtcp) { in DemuxPacket()
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| D | channelmanager.cc | 321 BaseSession* session, const std::string& content_name, bool rtcp) { in CreateVoiceChannel()
328 BaseSession* session, const std::string& content_name, bool rtcp) { in CreateVoiceChannel_w()
367 BaseSession* session, const std::string& content_name, bool rtcp, in CreateVideoChannel()
375 BaseSession* session, const std::string& content_name, bool rtcp, in CreateVideoChannel_w()
419 bool rtcp, DataChannelType channel_type) { in CreateDataChannel()
427 bool rtcp, DataChannelType data_channel_type) { in CreateDataChannel_w()
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| D | channel.cc | 143 static const char* PacketType(bool rtcp) { in PacketType()
147 static bool ValidPacket(bool rtcp, const talk_base::Buffer* packet) { in ValidPacket()
172 const std::string& content_name, bool rtcp) in BaseChannel()
389 bool rtcp = PacketIsRtcp(channel, data, len); in OnChannelRead() local
424 bool BaseChannel::SendPacket(bool rtcp, talk_base::Buffer* packet, in SendPacket()
554 bool BaseChannel::WantsPacket(bool rtcp, talk_base::Buffer* packet) { in WantsPacket()
567 void BaseChannel::HandlePacket(bool rtcp, talk_base::Buffer* packet, in HandlePacket()
763 bool BaseChannel::SetDtlsSrtpCiphers(TransportChannel *tc, bool rtcp) { in SetDtlsSrtpCiphers()
1251 bool rtcp) in VoiceChannel()
1650 bool rtcp, in VideoChannel()
[all …]
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| D | mediarecorder.cc | 77 void RtpDumpSink::OnPacket(const void* data, size_t size, bool rtcp) { in OnPacket()
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| D | channel.h | 249 bool rtcp() const { return rtcp_; } in rtcp() function
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| D | channel_unittest.cc | 311 bool rtcp) { in CreateChannel()
1818 TransportChannel* rtcp = channel1_->rtcp_transport_channel(); in TestOnReadyToSend() local
1961 bool rtcp) { in CreateChannel()
2753 bool rtcp) { in CreateChannel()
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| D | call.cc | 316 bool rtcp = false; in AddSession() local
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| /external/chromium_org/third_party/libjingle/source/talk/media/base/ |
| D | rtpdump.cc | 353 const void* data, size_t data_len, uint32 elapsed, bool rtcp) { in WritePacket()
388 bool rtcp) { in FilterPacket()
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| D | testutils.cc | 137 size_t count, bool rtcp, uint32 rtp_ssrc, RtpDumpWriter* writer) { in WriteTestPackets()
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| D | mediachannel.h | 676 bool DoSendPacket(talk_base::Buffer* packet, bool rtcp) { in DoSendPacket()
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| /external/chromium_org/third_party/libsrtp/srtp/include/ |
| D | srtp.h | 217 crypto_policy_t rtcp; /**< SRTCP crypto policy. */ member
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| /external/srtp/include/ |
| D | srtp.h | 217 crypto_policy_t rtcp; /**< SRTCP crypto policy. */ member
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| /external/chromium_org/third_party/webrtc/modules/remote_bitrate_estimator/ |
| D | remote_bitrate_estimator_unittest_helper.cc | 84 RtcpPacket* rtcp = new RtcpPacket; in Rtcp() local
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| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
| D | rtcp_packet.cc | 51 namespace rtcp { namespace
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| /external/chromium_org/third_party/webrtc/voice_engine/ |
| D | channel.cc | 50 RtcpStatistics rtcp; member
4034 void Channel::UpdatePlayoutTimestamp(bool rtcp) { in UpdatePlayoutTimestamp()
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| /external/chromium_org/third_party/libjingle/source/talk/media/webrtc/ |
| D | webrtcvoiceengine.cc | 3540 bool rtcp) { in SetSendCodecs()
|