| /external/webrtc/webrtc/modules/audio_coding/neteq/test/ |
| NETEQTEST_RTPpacket.cc | 617 void NETEQTEST_RTPpacket::makeRTPheader(unsigned char* rtp_data,
624 rtp_data[0] = markerBit ? 0x81 : 0x80;
625 rtp_data[1] = payloadType;
626 rtp_data[2] = seqNo >> 8;
627 rtp_data[3] = seqNo & 0xFF;
628 rtp_data[4] = timestamp >> 24;
629 rtp_data[5] = (timestamp >> 16) & 0xFF;
630 rtp_data[6] = (timestamp >> 8) & 0xFF;
631 rtp_data[7] = timestamp & 0xFF;
632 rtp_data[8] = ssrc >> 24
641 uint16_t* rtp_data = reinterpret_cast<uint16_t*>(_datagram); local
664 uint16_t* rtp_data = reinterpret_cast<uint16_t*>(_datagram); local
689 uint16_t* rtp_data = reinterpret_cast<uint16_t*>(_datagram); local
709 uint16_t* rtp_data = reinterpret_cast<uint16_t*>(_datagram); local
[all...] |
| RTPencode.cc | 108 void makeRTPheader(unsigned char* rtp_data,
113 int makeRedundantHeader(unsigned char* rtp_data,
290 unsigned char rtp_data[kRtpDataSize]; local
630 makeRTPheader(rtp_data, NETEQ_CODEC_AVT_PT, seqNo, DTMFtimestamp, ssrc);
632 &rtp_data[12], NTone % 12, 0, 4,
636 makeRTPheader(rtp_data, NETEQ_CODEC_AVT_PT, seqNo, DTMFtimestamp, ssrc);
637 enc_len = makeDTMFpayload(&rtp_data[12], NTone % 12, 1, 4,
657 if (fwrite(rtp_data, 12 + enc_len, 1, out_file) != 1) {
674 NetEQTest_encode(usedCodec, org_data, packet_size, &rtp_data[12], fs,
679 stereoInterleave(&rtp_data[12], enc_len, stereoMode)
[all...] |
| /external/webrtc/webrtc/test/ |
| rtp_file_reader.cc | 175 uint8_t* rtp_data = packet->data; variable
193 if (fread(rtp_data, 1, len, file_) != len) {
|
| /external/webrtc/webrtc/video/ |
| vie_channel.cc | 818 StreamDataCounters rtp_data; local
820 rtp_rtcp->GetSendStreamDataCounters(&rtp_data, &rtx_data);
821 rtp_counters->Add(rtp_data);
[all...] |
