| /external/webrtc/webrtc/modules/audio_coding/neteq/tools/ |
| packet_unittest.cc | 27 uint8_t* rtp_data) {
28 rtp_data[0] = 0x80;
29 rtp_data[1] = static_cast<uint8_t>(payload_type);
30 rtp_data[2] = (seq_number >> 8) & 0xFF;
31 rtp_data[3] = (seq_number) & 0xFF;
32 rtp_data[4] = timestamp >> 24;
33 rtp_data[5] = (timestamp >> 16) & 0xFF;
34 rtp_data[6] = (timestamp >> 8) & 0xFF;
35 rtp_data[7] = timestamp & 0xFF;
36 rtp_data[8] = ssrc >> 24
[all...] |
| /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...] |
| NETEQTEST_RTPpacket.h | 91 void makeRTPheader(unsigned char* rtp_data, uint8_t payloadType,
|
| /external/curl/tests/libtest/ |
| lib571.c | 49 static const char *RTP_DATA = "$_1234\n\0asdf";
74 if(memcmp(RTP_DATA, data + i, RTP_DATA_SIZE) != 0) {
79 if (memcmp(RTP_DATA, data + i, message_size - i) != 0) {
|
| /external/webrtc/webrtc/test/ |
| rtp_file_reader.cc | 175 uint8_t* rtp_data = packet->data; variable
193 if (fread(rtp_data, 1, len, file_) != len) {
|
| /external/webrtc/webrtc/video/ |
| vie_channel.cc | 818 StreamDataCounters rtp_data; local
820 rtp_rtcp->GetSendStreamDataCounters(&rtp_data, &rtx_data);
821 rtp_counters->Add(rtp_data);
[all...] |
| /external/curl/tests/server/ |
| rtspd.c | 199 static const char *RTP_DATA = "$_1234\n\0asdf";
498 memcpy(rtp_scratch + 4 + i, RTP_DATA, RTP_DATA_SIZE);
[all...] |
| /external/webrtc/talk/session/media/ |
| channel_unittest.cc | 131 const uint8_t* rtp_data,
140 rtp_packet_(reinterpret_cast<const char*>(rtp_data), rtp_len),
[all...] |
