/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/tools/ |
D | rtp_generator.cc | 20 WebRtcRTPHeader* rtp_header) { in GetRtpHeader() 49 WebRtcRTPHeader* rtp_header) { in GetRtpHeader()
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D | neteq_rtpplay.cc | 219 WebRtcRTPHeader rtp_header; in main() local 495 WebRtcRTPHeader* rtp_header, in ReplacePayload()
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D | neteq_performance_test.cc | 57 WebRtcRTPHeader rtp_header; in Run() local
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/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
D | rtp_sender_unittest.cc | 47 const uint8_t* GetPayloadData(const RTPHeader& rtp_header, in GetPayloadData() 52 uint16_t GetPayloadDataLength(const RTPHeader& rtp_header, in GetPayloadDataLength() 110 void VerifyRTPHeaderCommon(const RTPHeader& rtp_header) { in VerifyRTPHeaderCommon() 209 webrtc::RTPHeader rtp_header; in TEST_F() local 240 webrtc::RTPHeader rtp_header; in TEST_F() local 281 webrtc::RTPHeader rtp_header; in TEST_F() local 311 webrtc::RTPHeader rtp_header; in TEST_F() local 349 webrtc::RTPHeader rtp_header; in TEST_F() local 399 webrtc::RTPHeader rtp_header; in TEST_F() local 477 webrtc::RTPHeader rtp_header; in TEST_F() local [all …]
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D | rtp_format_vp8_unittest.cc | 414 WebRtcRTPHeader rtp_header; in TEST_F() local 435 WebRtcRTPHeader rtp_header; in TEST_F() local 468 WebRtcRTPHeader rtp_header; in TEST_F() local 486 WebRtcRTPHeader rtp_header; in TEST_F() local 505 WebRtcRTPHeader rtp_header; in TEST_F() local 526 WebRtcRTPHeader rtp_header; in TEST_F() local 543 WebRtcRTPHeader rtp_header; in TEST_F() local 567 WebRtcRTPHeader rtp_header; in TEST_F() local
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D | rtp_receiver_video.cc | 50 int32_t RTPReceiverVideo::ParseRtpPacket(WebRtcRTPHeader* rtp_header, in ParseRtpPacket() 110 int32_t RTPReceiverVideo::BuildRTPheader(const WebRtcRTPHeader* rtp_header, in BuildRTPheader()
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D | rtp_receiver_impl.cc | 164 const RTPHeader& rtp_header, in IncomingRtpPacket() 265 void RtpReceiverImpl::CheckSSRCChanged(const RTPHeader& rtp_header) { in CheckSSRCChanged() 337 const RTPHeader& rtp_header, in CheckPayloadChanged() 429 void RtpReceiverImpl::CheckCSRC(const WebRtcRTPHeader& rtp_header) { in CheckCSRC()
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D | rtp_format_h264.cc | 40 void ParseSingleNalu(WebRtcRTPHeader* rtp_header, in ParseSingleNalu() 67 void ParseFuaNalu(WebRtcRTPHeader* rtp_header, in ParseFuaNalu() 297 bool RtpDepacketizerH264::Parse(WebRtcRTPHeader* rtp_header, in Parse()
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D | rtp_receiver_audio.cc | 183 int32_t RTPReceiverAudio::ParseRtpPacket(WebRtcRTPHeader* rtp_header, in ParseRtpPacket() 289 WebRtcRTPHeader* rtp_header, in ParseAudioCodecSpecific()
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D | rtp_format_video_generic.cc | 93 bool RtpDepacketizerGeneric::Parse(WebRtcRTPHeader* rtp_header, in Parse()
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D | rtp_sender.cc | 478 RTPHeader rtp_header; in TrySendRedundantPayloads() local 578 RTPHeader rtp_header; in SendPadData() local 820 RTPHeader rtp_header; in PrepareAndSendPacket() local 916 RTPHeader rtp_header; in SendToNetwork() local 1269 const RTPHeader &rtp_header, const int64_t time_diff_ms) const { in UpdateTransmissionTimeOffset() argument 1315 const RTPHeader &rtp_header, in UpdateAudioLevel() 1358 const RTPHeader &rtp_header, const int64_t now_ms) const { in UpdateAbsoluteSendTime() argument 1632 RTPHeader rtp_header; in BuildRtxPacket() local
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D | rtp_format_vp8.cc | 131 int ParseVP8FrameSize(WebRtcRTPHeader* rtp_header, in ParseVP8FrameSize() 173 bool ParseVP8(WebRtcRTPHeader* rtp_header, in ParseVP8() 736 bool RtpDepacketizerVp8::Parse(WebRtcRTPHeader* rtp_header, in Parse()
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D | producer_fec.cc | 49 void RedPacket::CreateHeader(const uint8_t* rtp_header, int header_length, in CreateHeader()
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D | rtp_sender_audio.cc | 442 RTPHeader rtp_header; in SendAudio() local
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/external/chromium_org/media/cast/net/rtp/ |
D | rtp_packetizer_unittest.cc | 42 void VerifyRtpHeader(const RtpCastTestHeader& rtp_header) { in VerifyRtpHeader() 47 void VerifyCommonRtpHeader(const RtpCastTestHeader& rtp_header) { in VerifyCommonRtpHeader() 55 void VerifyCastRtpHeader(const RtpCastTestHeader& rtp_header) { in VerifyCastRtpHeader() 67 RtpCastTestHeader rtp_header; in SendPacket() local
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D | frame_buffer.cc | 27 const RtpCastHeader& rtp_header) { in InsertPacket()
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D | framer.cc | 37 const RtpCastHeader& rtp_header, in InsertPacket()
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/external/chromium_org/third_party/webrtc/modules/video_coding/main/test/ |
D | receiver_tests.h | 33 const webrtc::WebRtcRTPHeader* rtp_header) OVERRIDE { in OnReceivedPayloadData()
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D | vcm_payload_sink_factory.cc | 57 const WebRtcRTPHeader* rtp_header) OVERRIDE { in OnReceivedPayloadData()
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/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/ |
D | rtcp.cc | 33 void Rtcp::Update(const RTPHeader& rtp_header, uint32_t receive_timestamp) { in Update()
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D | neteq_impl_unittest.cc | 261 WebRtcRTPHeader rtp_header; in TEST_F() local 372 WebRtcRTPHeader rtp_header; in TEST_F() local 414 WebRtcRTPHeader rtp_header; in TEST_F() local
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D | neteq_impl.cc | 117 int NetEqImpl::InsertPacket(const WebRtcRTPHeader& rtp_header, in InsertPacket() 137 int NetEqImpl::InsertSyncPacket(const WebRtcRTPHeader& rtp_header, in InsertSyncPacket() 396 int NetEqImpl::InsertPacketInternal(const WebRtcRTPHeader& rtp_header, in InsertPacketInternal() 626 const RTPHeader* rtp_header = packet_buffer_->NextRtpHeader(); in InsertPacketInternal() local
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/external/chromium_org/media/cast/receiver/ |
D | frame_receiver.cc | 79 RtpCastHeader rtp_header; in ProcessPacket() local 112 void FrameReceiver::ProcessParsedPacket(const RtpCastHeader& rtp_header, in ProcessParsedPacket()
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/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/test/testAPI/ |
D | test_api.h | 110 webrtc::WebRtcRTPHeader rtp_header() const { in rtp_header() function
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/external/chromium_org/third_party/webrtc/modules/audio_coding/main/acm2/ |
D | acm_receiver.cc | 256 int AcmReceiver::InsertPacket(const WebRtcRTPHeader& rtp_header, in InsertPacket() 800 const RTPHeader &rtp_header, const uint8_t* payload) const { in RtpHeaderToCodecIndex() argument
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