Searched refs:rtpHeader (Results 1 – 11 of 11) sorted by relevance
/external/webrtc/webrtc/modules/audio_coding/test/ |
D | RTPFile.cc | 31 const uint8_t* rtpHeader) { in ParseRTPHeader() argument 32 rtpInfo->header.payloadType = rtpHeader[1]; in ParseRTPHeader() 33 rtpInfo->header.sequenceNumber = (static_cast<uint16_t>(rtpHeader[2]) << 8) | in ParseRTPHeader() 34 rtpHeader[3]; in ParseRTPHeader() 35 rtpInfo->header.timestamp = (static_cast<uint32_t>(rtpHeader[4]) << 24) | in ParseRTPHeader() 36 (static_cast<uint32_t>(rtpHeader[5]) << 16) | in ParseRTPHeader() 37 (static_cast<uint32_t>(rtpHeader[6]) << 8) | rtpHeader[7]; in ParseRTPHeader() 38 rtpInfo->header.ssrc = (static_cast<uint32_t>(rtpHeader[8]) << 24) | in ParseRTPHeader() 39 (static_cast<uint32_t>(rtpHeader[9]) << 16) | in ParseRTPHeader() 40 (static_cast<uint32_t>(rtpHeader[10]) << 8) | rtpHeader[11]; in ParseRTPHeader() [all …]
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D | RTPFile.h | 40 void MakeRTPheader(uint8_t* rtpHeader, uint8_t payloadType, int16_t seqNo, 43 void ParseRTPHeader(WebRtcRTPHeader* rtpInfo, const uint8_t* rtpHeader);
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D | EncodeDecodeTest.h | 40 static void MakeRTPheader(uint8_t* rtpHeader, uint8_t payloadType,
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/external/webrtc/webrtc/modules/video_coding/ |
D | packet.cc | 38 const WebRtcRTPHeader& rtpHeader) in VCMPacket() argument 39 : payloadType(rtpHeader.header.payloadType), in VCMPacket() 40 timestamp(rtpHeader.header.timestamp), in VCMPacket() 41 ntp_time_ms_(rtpHeader.ntp_time_ms), in VCMPacket() 42 seqNum(rtpHeader.header.sequenceNumber), in VCMPacket() 45 markerBit(rtpHeader.header.markerBit), in VCMPacket() 47 frameType(rtpHeader.frameType), in VCMPacket() 49 isFirstPacket(rtpHeader.type.Video.isFirstPacket), in VCMPacket() 52 width(rtpHeader.type.Video.width), in VCMPacket() 53 height(rtpHeader.type.Video.height), in VCMPacket() [all …]
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D | packet.h | 25 const WebRtcRTPHeader& rtpHeader);
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/external/webrtc/webrtc/modules/rtp_rtcp/test/testAPI/ |
D | test_api_audio.cc | 31 const webrtc::WebRtcRTPHeader* rtpHeader) override { in OnReceivedPayloadData() argument 32 if (rtpHeader->header.payloadType == 98 || in OnReceivedPayloadData() 33 rtpHeader->header.payloadType == 99) { in OnReceivedPayloadData() 45 if (rtpHeader->header.payloadType == 100 || in OnReceivedPayloadData() 46 rtpHeader->header.payloadType == 101 || in OnReceivedPayloadData() 47 rtpHeader->header.payloadType == 102) { in OnReceivedPayloadData() 48 if (rtpHeader->type.Audio.channel == 1) { in OnReceivedPayloadData()
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/external/webrtc/webrtc/modules/rtp_rtcp/include/ |
D | rtp_rtcp_defines.h | 195 const WebRtcRTPHeader* rtpHeader) = 0; 352 const WebRtcRTPHeader* rtpHeader) override { in OnReceivedPayloadData() argument
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/external/webrtc/webrtc/modules/rtp_rtcp/mocks/ |
D | mock_rtp_rtcp.h | 32 const WebRtcRTPHeader* rtpHeader));
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/external/webrtc/webrtc/voice_engine/ |
D | channel.cc | 458 const WebRtcRTPHeader* rtpHeader) in OnReceivedPayloadData() argument 464 rtpHeader->header.payloadType, in OnReceivedPayloadData() 465 rtpHeader->type.Audio.channel); in OnReceivedPayloadData() 482 *rtpHeader) != 0) in OnReceivedPayloadData() 491 UpdatePacketDelay(rtpHeader->header.timestamp, in OnReceivedPayloadData() 492 rtpHeader->header.sequenceNumber); in OnReceivedPayloadData()
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D | channel.h | 376 const WebRtcRTPHeader* rtpHeader) override;
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/external/webrtc/webrtc/modules/rtp_rtcp/source/ |
D | rtcp_receiver_unittest.cc | 62 const WebRtcRTPHeader* rtpHeader) override { in OnReceivedPayloadData() argument
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