Home
last modified time | relevance | path

Searched refs:rtpHeader (Results 1 – 11 of 11) sorted by relevance

/external/webrtc/webrtc/modules/audio_coding/test/
DRTPFile.cc31 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 …]
DRTPFile.h40 void MakeRTPheader(uint8_t* rtpHeader, uint8_t payloadType, int16_t seqNo,
43 void ParseRTPHeader(WebRtcRTPHeader* rtpInfo, const uint8_t* rtpHeader);
DEncodeDecodeTest.h40 static void MakeRTPheader(uint8_t* rtpHeader, uint8_t payloadType,
/external/webrtc/webrtc/modules/video_coding/
Dpacket.cc38 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 …]
Dpacket.h25 const WebRtcRTPHeader& rtpHeader);
/external/webrtc/webrtc/modules/rtp_rtcp/test/testAPI/
Dtest_api_audio.cc31 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()
/external/webrtc/webrtc/modules/rtp_rtcp/include/
Drtp_rtcp_defines.h195 const WebRtcRTPHeader* rtpHeader) = 0;
352 const WebRtcRTPHeader* rtpHeader) override { in OnReceivedPayloadData() argument
/external/webrtc/webrtc/modules/rtp_rtcp/mocks/
Dmock_rtp_rtcp.h32 const WebRtcRTPHeader* rtpHeader));
/external/webrtc/webrtc/voice_engine/
Dchannel.cc458 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()
Dchannel.h376 const WebRtcRTPHeader* rtpHeader) override;
/external/webrtc/webrtc/modules/rtp_rtcp/source/
Drtcp_receiver_unittest.cc62 const WebRtcRTPHeader* rtpHeader) override { in OnReceivedPayloadData() argument