| /external/webrtc/audio/voip/test/ |
| D | audio_egress_unittest.cc | 120 RtpPacketReceived rtp; in TEST_F() local
158 RtpPacketReceived rtp; in TEST_F() local
258 auto is_dtmf = [&](RtpPacketReceived& rtp) { in TEST_F()
269 RtpPacketReceived rtp; in TEST_F() local
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| D | audio_channel_unittest.cc | 124 RtpPacketReceived rtp; in TEST_F() local
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| /external/webrtc/rtc_tools/rtp_generator/configs/ |
| D | vp8.json | 8 "rtp" : { object
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| D | vp9.json | 8 "rtp" : { object
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| /external/webrtc/logging/rtc_event_log/ |
| D | logged_events.cc | 14 LoggedPacketInfo::LoggedPacketInfo(const LoggedRtpPacket& rtp, in LoggedPacketInfo()
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| /external/webrtc/test/fuzzers/configs/replay_packet_fuzzer/ |
| D | vp8_config.json | 10 "rtp" : { object
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| D | vp9_config.json | 10 "rtp" : { object
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| D | h264_single_nal_config.json | 36 "rtp" : { object
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| D | h264_non_interleaved_config.json | 36 "rtp" : { object
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| D | vp8_fec_config.json | 11 "rtp" : { object
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| D | vp9_fec_config.json | 16 "rtp" : { object
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| D | h264_fec_config.json | 76 "rtp" : { object
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| /external/webrtc/rtc_tools/rtp_generator/ |
| D | rtp_generator.h | 52 RtpConfig rtp; member
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| /external/webrtc/media/base/ |
| D | rtp_utils.cc | 79 void UpdateRtpAuthTag(uint8_t* rtp, in UpdateRtpAuthTag()
339 bool ValidateRtpHeader(const uint8_t* rtp, in ValidateRtpHeader()
393 bool UpdateRtpAbsSendTimeExtension(uint8_t* rtp, in UpdateRtpAbsSendTimeExtension()
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| /external/webrtc/modules/pacing/ |
| D | packet_router_unittest.cc | 447 NiceMock<MockRtpRtcpInterface> rtp; in TEST() local
473 NiceMock<MockRtpRtcpInterface> rtp; in TEST() local
498 NiceMock<MockRtpRtcpInterface> rtp; in TEST() local
561 NiceMock<MockRtpRtcpInterface> rtp; in TEST() local
724 NiceMock<MockRtpRtcpInterface> rtp; in TEST() local
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| /external/webrtc/pc/ |
| D | rtc_stats_traversal.cc | 103 const auto& rtp = static_cast<const RTCRTPStreamStats&>(stats); in GetStatsReferencedIds() local
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| /external/skqp/src/gpu/ |
| D | GrDrawingManager.cpp | 561 sk_sp<GrRenderTargetOpList> GrDrawingManager::newRTOpList(GrRenderTargetProxy* rtp, in newRTOpList()
717 sk_sp<GrRenderTargetProxy> rtp(sk_ref_sp(sProxy->asRenderTargetProxy())); in makeRenderTargetContext() local
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| D | GrSurfaceProxy.cpp | 263 const GrRenderTargetProxy* rtp = this->asRenderTargetProxy(); in computeScratchKey() local
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| /external/libsrtp2/test/ |
| D | rtp_decoder.c | 689 int rtp; in rtp_decoder_handle_pkt() local
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| /external/curl/lib/ |
| D | rtsp.c | 598 char *rtp; /* moving pointer to rtp data */ in rtsp_rtp_readwrite() local
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| /external/webrtc/call/ |
| D | audio_send_stream.h | 106 } rtp; member
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| D | rtp_payload_params.cc | 37 RTPVideoHeader* rtp) { in PopulateRtpWithCodecSpecifics()
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| D | video_send_stream.h | 160 RtpConfig rtp; member
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| /external/skia/src/gpu/ |
| D | GrSurfaceProxy.cpp | 204 if (const auto* rtp = this->asRenderTargetProxy()) { in computeScratchKey() local
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| /external/mesa3d/src/gallium/drivers/panfrost/ |
| D | pan_mfbd.c | 200 void *rtp, struct pipe_surface *surf, in panfrost_mfbd_emit_rt()
|