| /external/chromium_org/third_party/webrtc/modules/audio_coding/main/test/ |
| D | RTPFile.cc | 29 const uint8_t* rtpHeader) { in ParseRTPHeader() argument
30 rtpInfo->header.payloadType = rtpHeader[1]; in ParseRTPHeader()
31 rtpInfo->header.sequenceNumber = (static_cast<uint16_t>(rtpHeader[2]) << 8) | in ParseRTPHeader()
32 rtpHeader[3]; in ParseRTPHeader()
33 rtpInfo->header.timestamp = (static_cast<uint32_t>(rtpHeader[4]) << 24) | in ParseRTPHeader()
34 (static_cast<uint32_t>(rtpHeader[5]) << 16) | in ParseRTPHeader()
35 (static_cast<uint32_t>(rtpHeader[6]) << 8) | rtpHeader[7]; in ParseRTPHeader()
36 rtpInfo->header.ssrc = (static_cast<uint32_t>(rtpHeader[8]) << 24) | in ParseRTPHeader()
37 (static_cast<uint32_t>(rtpHeader[9]) << 16) | in ParseRTPHeader()
38 (static_cast<uint32_t>(rtpHeader[10]) << 8) | rtpHeader[11]; in ParseRTPHeader()
[all …]
|
| D | RTPFile.h | 40 void MakeRTPheader(uint8_t* rtpHeader, uint8_t payloadType, int16_t seqNo,
43 void ParseRTPHeader(WebRtcRTPHeader* rtpInfo, const uint8_t* rtpHeader);
|
| D | EncodeDecodeTest.h | 38 static void MakeRTPheader(uint8_t* rtpHeader, uint8_t payloadType,
|
| /external/chromium_org/third_party/webrtc/modules/video_coding/main/source/ |
| D | packet.cc | 39 const WebRtcRTPHeader& rtpHeader) : in VCMPacket() argument
40 payloadType(rtpHeader.header.payloadType), in VCMPacket()
41 timestamp(rtpHeader.header.timestamp), in VCMPacket()
42 ntp_time_ms_(rtpHeader.ntp_time_ms), in VCMPacket()
43 seqNum(rtpHeader.header.sequenceNumber), in VCMPacket()
46 markerBit(rtpHeader.header.markerBit), in VCMPacket()
48 frameType(rtpHeader.frameType), in VCMPacket()
50 isFirstPacket(rtpHeader.type.Video.isFirstPacket), in VCMPacket()
53 width(rtpHeader.type.Video.width), in VCMPacket()
54 height(rtpHeader.type.Video.height), in VCMPacket()
[all …]
|
| D | packet.h | 25 const WebRtcRTPHeader& rtpHeader);
|
| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/test/testAPI/ |
| D | test_api_audio.cc | 31 const webrtc::WebRtcRTPHeader* rtpHeader) { 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()
|
| D | test_api.h | 92 const webrtc::WebRtcRTPHeader* rtpHeader) { in OnReceivedPayloadData() argument
95 memcpy(&_rtpHeader, rtpHeader, sizeof(_rtpHeader)); in OnReceivedPayloadData()
|
| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/H264/ |
| D | rtp_sender_h264.cc | 219 RTPSenderH264::SendH264FillerData(const WebRtcRTPHeader* rtpHeader, in SendH264FillerData() argument
240 dataBuffer[1] = rtpHeader->header.payloadType; in SendH264FillerData()
242 ModuleRTPUtility::AssignUWord32ToBuffer(dataBuffer+4, rtpHeader->header.timestamp); in SendH264FillerData()
243 ModuleRTPUtility::AssignUWord32ToBuffer(dataBuffer+8, rtpHeader->header.ssrc); in SendH264FillerData()
315 RTPSenderH264::SendH264SVCRelayPacket(const WebRtcRTPHeader* rtpHeader, in SendH264SVCRelayPacket() argument
321 if (rtpHeader->header.sequenceNumber != (uint16_t)(_h264SVCRelaySequenceNumber + 1)) in SendH264SVCRelayPacket()
326 _h264SVCRelaySequenceNumber = rtpHeader->header.sequenceNumber; in SendH264SVCRelayPacket()
329 if (rtpHeader->header.timestamp != _h264SVCRelayTimeStamp) in SendH264SVCRelayPacket()
335 if (rtpHeader->header.timestamp == _h264SVCRelayTimeStamp && in SendH264SVCRelayPacket()
343 _h264SVCRelayTimeStamp = rtpHeader->header.timestamp; in SendH264SVCRelayPacket()
[all …]
|
| D | rtp_sender_h264.h | 81 int32_t SendH264FillerData(const WebRtcRTPHeader* rtpHeader,
89 int32_t SendH264SVCRelayPacket(const WebRtcRTPHeader* rtpHeader,
|
| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/interface/ |
| D | rtp_rtcp_defines.h | 203 const WebRtcRTPHeader* rtpHeader) = 0;
337 const WebRtcRTPHeader* rtpHeader) OVERRIDE { in OnReceivedPayloadData() argument
|
| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/test/BWEStandAlone/ |
| D | TestSenderReceiver.h | 101 const webrtc::WebRtcRTPHeader* rtpHeader);
|
| D | TestSenderReceiver.cc | 311 const webrtc::WebRtcRTPHeader* rtpHeader) in OnReceivedPayloadData() argument
|
| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
| D | fec_receiver_unittest.cc | 33 const WebRtcRTPHeader* rtpHeader));
|
| D | rtcp_sender_unittest.cc | 266 const WebRtcRTPHeader* rtpHeader) { in OnReceivedPayloadData() argument
|
| D | rtcp_receiver_unittest.cc | 245 const WebRtcRTPHeader* rtpHeader) { in OnReceivedPayloadData() argument
|
| /external/chromium_org/third_party/webrtc/voice_engine/ |
| D | channel.cc | 512 const WebRtcRTPHeader* rtpHeader) in OnReceivedPayloadData() argument
518 rtpHeader->header.payloadType, in OnReceivedPayloadData()
519 rtpHeader->type.Audio.channel); in OnReceivedPayloadData()
536 *rtpHeader) != 0) in OnReceivedPayloadData()
545 UpdatePacketDelay(rtpHeader->header.timestamp, in OnReceivedPayloadData()
546 rtpHeader->header.sequenceNumber); in OnReceivedPayloadData()
|
| D | channel.h | 370 const WebRtcRTPHeader* rtpHeader);
|