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Searched defs:rtp (Results 1 – 19 of 19) sorted by relevance

/external/chromium_org/third_party/webrtc/modules/video_coding/main/test/
Dmt_test_common.h28 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()
Dtest_callbacks.h96 VCMRTPEncodeCompleteCallback(RtpRtcp* rtp) : in VCMRTPEncodeCompleteCallback()
210 PacketRequester(RtpRtcp& rtp) : in PacketRequester()
247 void RegisterRtpModule(RtpRtcp* rtp) {_rtp = rtp;} in RegisterRtpModule()
Dmt_rx_tx_test.cc158 RtpRtcp* rtp = RtpRtcp::CreateRtpRtcp(configuration); in MTRxTxTest() local
/external/chromium_org/third_party/webrtc/video_engine/
Dvie_remb_unittest.cc45 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
/external/chromium_org/media/cast/logging/
Dreceiver_time_offset_estimator_impl.cc20 uint32 rtp, in SetSent()
31 uint32 rtp, in SetReceived()
/external/chromium_org/third_party/libjingle/source/talk/media/webrtc/
Dwebrtcvie.h95 webrtc::ViERender* render, webrtc::ViERTP_RTCP* rtp, in ViEWrapper()
116 webrtc::ViERTP_RTCP* rtp() { return rtp_.get(); } in rtp() function
Dwebrtcvoe.h112 webrtc::VoERTP_RTCP* rtp, in VoEWrapper()
140 webrtc::VoERTP_RTCP* rtp() const { return rtp_.get(); } in rtp() function
Dwebrtcvoiceengine.cc2704 webrtc::VoERTP_RTCP* rtp = engine()->voe()->rtp(); in SetSendCodecs() local
/external/chromium_org/content/browser/renderer_host/p2p/
Dsocket_host.cc71 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()
/external/chromium_org/third_party/webrtc/modules/video_coding/main/source/
Dgeneric_encoder.cc22 void CopyCodecSpecific(const CodecSpecificInfo* info, RTPVideoHeader** rtp) { in CopyCodecSpecific()
/external/chromium_org/third_party/webrtc/
Dvideo_send_stream.h117 } rtp; member
Dvideo_receive_stream.h139 } rtp; member
/external/srtp/include/
Dsrtp.h216 crypto_policy_t rtp; /**< SRTP crypto policy. */ member
/external/chromium_org/third_party/libsrtp/srtp/include/
Dsrtp.h216 crypto_policy_t rtp; /**< SRTP crypto policy. */ member
/external/dhcpcd/
Dconfigure.c642 struct rt *rtp, *rtl, *rtn; in add_router_host_route() local
/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/
Dneteq_unittest.cc309 void NetEqDecodingTest::Process(NETEQTEST_RTPpacket* rtp, int* out_len) { in Process()
364 NETEQTEST_RTPpacket rtp; in DecodeAndCompare() local
/external/chromium_org/third_party/webrtc/examples/android/media_demo/jni/
Dvoice_engine_jni.cc125 webrtc::VoERTP_RTCP* const rtp; member in __anond97007f10111::VoiceEngineData
Dvideo_engine_jni.cc260 webrtc::ViERTP_RTCP* const rtp; member in __anondfde66720111::VideoEngineData
/external/chromium_org/third_party/libjingle/source/talk/session/media/
Dchannel_unittest.cc1775 TransportChannel* rtp = channel1_->transport_channel(); in TestOnReadyToSend() local
1807 TransportChannel* rtp = channel1_->transport_channel(); in TestOnReadyToSendWithRtcpMux() local