| /external/chromium_org/third_party/webrtc/modules/video_coding/main/test/ |
| D | mt_test_common.h | 28 SendSharedState(webrtc::VideoCodingModule& vcm, webrtc::RtpRtcp& rtp, in SendSharedState()
63 SharedRTPState(webrtc::VideoCodingModule& vcm, webrtc::RtpRtcp& rtp) : in SharedRTPState()
74 SharedTransportState(webrtc::RtpRtcp& rtp, TransportCallback& transport): in SharedTransportState()
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| D | test_callbacks.h | 96 VCMRTPEncodeCompleteCallback(RtpRtcp* rtp) : in VCMRTPEncodeCompleteCallback()
210 PacketRequester(RtpRtcp& rtp) : in PacketRequester()
247 void RegisterRtpModule(RtpRtcp* rtp) {_rtp = rtp;} in RegisterRtpModule()
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| D | mt_rx_tx_test.cc | 158 RtpRtcp* rtp = RtpRtcp::CreateRtpRtcp(configuration); in MTRxTxTest() local
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| /external/chromium_org/third_party/webrtc/video_engine/ |
| D | vie_remb_unittest.cc | 45 MockRtpRtcp rtp; in TEST_F() local
70 MockRtpRtcp rtp; in TEST_F() local
198 MockRtpRtcp rtp; in TEST_F() local
229 MockRtpRtcp rtp; in TEST_F() local
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| /external/chromium_org/media/cast/logging/ |
| D | receiver_time_offset_estimator_impl.cc | 20 uint32 rtp, in SetSent()
31 uint32 rtp, in SetReceived()
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| /external/chromium_org/third_party/libjingle/source/talk/media/webrtc/ |
| D | webrtcvie.h | 95 webrtc::ViERender* render, webrtc::ViERTP_RTCP* rtp, in ViEWrapper()
116 webrtc::ViERTP_RTCP* rtp() { return rtp_.get(); } in rtp() function
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| D | webrtcvoe.h | 112 webrtc::VoERTP_RTCP* rtp, in VoEWrapper()
140 webrtc::VoERTP_RTCP* rtp() const { return rtp_.get(); } in rtp() function
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| D | webrtcvoiceengine.cc | 2704 webrtc::VoERTP_RTCP* rtp = engine()->voe()->rtp(); in SetSendCodecs() local
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| /external/chromium_org/content/browser/renderer_host/p2p/ |
| D | socket_host.cc | 71 bool ValidateRtpHeader(const char* rtp, size_t length, size_t* header_length) { in ValidateRtpHeader()
162 void UpdateRtpAuthTag(char* rtp, in UpdateRtpAuthTag()
373 bool UpdateRtpAbsSendTimeExtension(char* rtp, in UpdateRtpAbsSendTimeExtension()
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| /external/chromium_org/third_party/webrtc/modules/video_coding/main/source/ |
| D | generic_encoder.cc | 22 void CopyCodecSpecific(const CodecSpecificInfo* info, RTPVideoHeader** rtp) { in CopyCodecSpecific()
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| /external/chromium_org/third_party/webrtc/ |
| D | video_send_stream.h | 117 } rtp; member
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| D | video_receive_stream.h | 139 } rtp; member
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| /external/srtp/include/ |
| D | srtp.h | 216 crypto_policy_t rtp; /**< SRTP crypto policy. */ member
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| /external/chromium_org/third_party/libsrtp/srtp/include/ |
| D | srtp.h | 216 crypto_policy_t rtp; /**< SRTP crypto policy. */ member
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| /external/dhcpcd/ |
| D | configure.c | 642 struct rt *rtp, *rtl, *rtn; in add_router_host_route() local
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| /external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/ |
| D | neteq_unittest.cc | 309 void NetEqDecodingTest::Process(NETEQTEST_RTPpacket* rtp, int* out_len) { in Process()
364 NETEQTEST_RTPpacket rtp; in DecodeAndCompare() local
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| /external/chromium_org/third_party/webrtc/examples/android/media_demo/jni/ |
| D | voice_engine_jni.cc | 125 webrtc::VoERTP_RTCP* const rtp; member in __anond97007f10111::VoiceEngineData
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| D | video_engine_jni.cc | 260 webrtc::ViERTP_RTCP* const rtp; member in __anondfde66720111::VideoEngineData
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| /external/chromium_org/third_party/libjingle/source/talk/session/media/ |
| D | channel_unittest.cc | 1775 TransportChannel* rtp = channel1_->transport_channel(); in TestOnReadyToSend() local
1807 TransportChannel* rtp = channel1_->transport_channel(); in TestOnReadyToSendWithRtcpMux() local
|