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  /external/chromium_org/third_party/webrtc/modules/audio_coding/main/test/
RTPFile.cc 30 const uint8_t* rtpHeader) {
31 rtpInfo->header.payloadType = rtpHeader[1];
32 rtpInfo->header.sequenceNumber = (static_cast<uint16_t>(rtpHeader[2]) << 8) |
33 rtpHeader[3];
34 rtpInfo->header.timestamp = (static_cast<uint32_t>(rtpHeader[4]) << 24) |
35 (static_cast<uint32_t>(rtpHeader[5]) << 16) |
36 (static_cast<uint32_t>(rtpHeader[6]) << 8) | rtpHeader[7];
37 rtpInfo->header.ssrc = (static_cast<uint32_t>(rtpHeader[8]) << 24) |
38 (static_cast<uint32_t>(rtpHeader[9]) << 16)
    [all...]
RTPFile.h 40 void MakeRTPheader(uint8_t* rtpHeader, uint8_t payloadType, int16_t seqNo,
43 void ParseRTPHeader(WebRtcRTPHeader* rtpInfo, const uint8_t* rtpHeader);
EncodeDecodeTest.h 39 static void MakeRTPheader(uint8_t* rtpHeader, uint8_t payloadType,
  /external/chromium_org/third_party/webrtc/modules/video_coding/main/source/
packet.cc 39 const WebRtcRTPHeader& rtpHeader) :
40 payloadType(rtpHeader.header.payloadType),
41 timestamp(rtpHeader.header.timestamp),
42 ntp_time_ms_(rtpHeader.ntp_time_ms),
43 seqNum(rtpHeader.header.sequenceNumber),
46 markerBit(rtpHeader.header.markerBit),
48 frameType(rtpHeader.frameType),
50 isFirstPacket(rtpHeader.type.Video.isFirstPacket),
53 width(rtpHeader.type.Video.width),
54 height(rtpHeader.type.Video.height)
    [all...]
packet.h 25 const WebRtcRTPHeader& rtpHeader);
  /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/test/testAPI/
test_api.h 53 RTPHeader header;
94 const webrtc::WebRtcRTPHeader* rtpHeader) {
97 memcpy(&_rtpHeader, rtpHeader, sizeof(_rtpHeader));
test_api_audio.cc 31 const webrtc::WebRtcRTPHeader* rtpHeader) {
32 if (rtpHeader->header.payloadType == 98 ||
33 rtpHeader->header.payloadType == 99) {
45 if (rtpHeader->header.payloadType == 100 ||
46 rtpHeader->header.payloadType == 101 ||
47 rtpHeader->header.payloadType == 102) {
48 if (rtpHeader->type.Audio.channel == 1) {
  /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/test/BWEStandAlone/
TestSenderReceiver.h 104 const webrtc::WebRtcRTPHeader* rtpHeader) OVERRIDE;
TestSenderReceiver.cc 311 const webrtc::WebRtcRTPHeader* rtpHeader)
  /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/interface/
rtp_rtcp_defines.h 224 const WebRtcRTPHeader* rtpHeader) = 0;
358 const WebRtcRTPHeader* rtpHeader) OVERRIDE {
  /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/mocks/
mock_rtp_rtcp.h 27 const WebRtcRTPHeader* rtpHeader));
  /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/
rtcp_sender_unittest.cc 266 const WebRtcRTPHeader* rtpHeader) {
373 RTPHeader header;
rtcp_receiver_unittest.cc 55 const WebRtcRTPHeader* rtpHeader) {
  /external/chromium_org/third_party/webrtc/voice_engine/
channel.h 370 const WebRtcRTPHeader* rtpHeader);
489 const RTPHeader& header, bool in_order);
492 const RTPHeader& header);
493 bool IsPacketInOrder(const RTPHeader& header) const;
494 bool IsPacketRetransmitted(const RTPHeader& header, bool in_order) const;
channel.cc 511 const WebRtcRTPHeader* rtpHeader)
517 rtpHeader->header.payloadType,
518 rtpHeader->type.Audio.channel);
535 *rtpHeader) != 0)
544 UpdatePacketDelay(rtpHeader->header.timestamp,
545 rtpHeader->header.sequenceNumber);
563 RTPHeader header;
    [all...]

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