| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/test/testAPI/ |
| test_api.h | 108 webrtc::WebRtcRTPHeader rtp_header() const { function in class:webrtc::TestRtpReceiver
|
| /external/chromium_org/media/cast/receiver/ |
| frame_receiver.cc | 83 RtpCastHeader rtp_header; local
88 &rtp_header,
94 ProcessParsedPacket(rtp_header, payload_data, payload_size);
95 stats_.UpdateStatistics(rtp_header);
116 void FrameReceiver::ProcessParsedPacket(const RtpCastHeader& rtp_header,
123 frame_id_to_rtp_timestamp_[rtp_header.frame_id & 0xff] =
124 rtp_header.rtp_timestamp;
126 now, PACKET_RECEIVED, event_media_type_, rtp_header.rtp_timestamp,
127 rtp_header.frame_id, rtp_header.packet_id, rtp_header.max_packet_id
[all...] |
| /external/chromium_org/media/cast/transport/rtp_sender/rtp_packetizer/ |
| rtp_packetizer_unittest.cc | 43 void VerifyRtpHeader(const RtpCastTestHeader& rtp_header) {
44 VerifyCommonRtpHeader(rtp_header);
45 VerifyCastRtpHeader(rtp_header);
48 void VerifyCommonRtpHeader(const RtpCastTestHeader& rtp_header) {
49 EXPECT_EQ(kPayload, rtp_header.payload_type);
50 EXPECT_EQ(sequence_number_, rtp_header.sequence_number);
51 EXPECT_EQ(expectd_rtp_timestamp_, rtp_header.rtp_timestamp);
52 EXPECT_EQ(config_.ssrc, rtp_header.ssrc);
53 EXPECT_EQ(0, rtp_header.num_csrcs);
56 void VerifyCastRtpHeader(const RtpCastTestHeader& rtp_header) {
68 RtpCastTestHeader rtp_header; variable
70 VerifyRtpHeader(rtp_header); variable
[all...] |
| /external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/tools/ |
| neteq_performance_test.cc | 57 WebRtcRTPHeader rtp_header; local
63 rtp_gen.GetRtpHeader(kPayloadType, kInputBlockSizeSamples, &rtp_header);
80 lost = ((rtp_header.header.sequenceNumber - 1) % lossrate) == 0;
85 rtp_header, input_payload, payload_len,
94 &rtp_header);
|
| neteq_rtpplay.cc | 109 WebRtcRTPHeader* rtp_header,
235 WebRtcRTPHeader rtp_header; local
236 rtp->parseHeader(&rtp_header);
245 &rtp_header,
249 int error = neteq->InsertPacket(rtp_header, payload_ptr,
505 WebRtcRTPHeader* rtp_header,
509 if (IsComfortNosie(rtp_header->header.payloadType)) {
520 if (next_rtp->sequenceNumber() == rtp_header->header.sequenceNumber + 1) {
522 next_rtp->timeStamp() - rtp_header->header.timestamp) {
524 next_rtp->timeStamp() - rtp_header->header.timestamp
[all...] |
| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
| rtp_sender_audio.cc | 440 RTPHeader rtp_header; local
441 rtp_parser.Parse(rtp_header);
442 _rtpSender->UpdateAudioLevel(dataBuffer, packetSize, rtp_header,
|
| rtp_sender_unittest.cc | 46 const uint8_t* GetPayloadData(const RTPHeader& rtp_header,
48 return packet + rtp_header.headerLength;
51 uint16_t GetPayloadDataLength(const RTPHeader& rtp_header,
53 uint16_t length = packet_length - rtp_header.headerLength -
54 rtp_header.paddingLength;
109 void VerifyRTPHeaderCommon(const RTPHeader& rtp_header) {
110 EXPECT_EQ(kMarkerBit, rtp_header.markerBit);
111 EXPECT_EQ(payload_, rtp_header.payloadType);
112 EXPECT_EQ(kSeqNum, rtp_header.sequenceNumber);
113 EXPECT_EQ(kTimestamp, rtp_header.timestamp)
208 webrtc::RTPHeader rtp_header; local
239 webrtc::RTPHeader rtp_header; local
280 webrtc::RTPHeader rtp_header; local
310 webrtc::RTPHeader rtp_header; local
348 webrtc::RTPHeader rtp_header; local
398 webrtc::RTPHeader rtp_header; local
476 webrtc::RTPHeader rtp_header; local
538 webrtc::RTPHeader rtp_header; local
581 webrtc::RTPHeader rtp_header; local
749 webrtc::RTPHeader rtp_header; local
1023 webrtc::RTPHeader rtp_header; local
1052 webrtc::RTPHeader rtp_header; local
[all...] |
| rtp_sender.cc | 460 RTPHeader rtp_header; local
461 rtp_parser.Parse(rtp_header);
462 bytes_left -= length - rtp_header.headerLength;
795 RTPHeader rtp_header; local
796 rtp_parser.Parse(rtp_header);
798 "timestamp", rtp_header.timestamp,
799 "seqnum", rtp_header.sequenceNumber);
809 UpdateTransmissionTimeOffset(buffer_to_send_ptr, length, rtp_header,
811 UpdateAbsoluteSendTime(buffer_to_send_ptr, length, rtp_header, now_ms);
813 UpdateRtpStats(buffer_to_send_ptr, length, rtp_header, send_over_rtx
910 RTPHeader rtp_header; local
1596 RTPHeader rtp_header; local
[all...] |
| /external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/ |
| neteq_impl_unittest.cc | 261 WebRtcRTPHeader rtp_header; local
262 rtp_header.header.payloadType = kPayloadType;
263 rtp_header.header.sequenceNumber = kFirstSequenceNumber;
264 rtp_header.header.timestamp = kFirstTimestamp;
265 rtp_header.header.ssrc = kSsrc;
320 .WillOnce(Return(&rtp_header.header));
354 neteq_->InsertPacket(rtp_header, payload, kPayloadLength, kFirstReceiveTime);
357 rtp_header.header.timestamp += 160;
358 rtp_header.header.sequenceNumber += 1;
359 neteq_->InsertPacket(rtp_header, payload, kPayloadLength
372 WebRtcRTPHeader rtp_header; local
414 WebRtcRTPHeader rtp_header; local
[all...] |
| neteq_impl.cc | 115 int NetEqImpl::InsertPacket(const WebRtcRTPHeader& rtp_header,
120 LOG(LS_VERBOSE) << "InsertPacket: ts=" << rtp_header.header.timestamp <<
121 ", sn=" << rtp_header.header.sequenceNumber <<
122 ", pt=" << static_cast<int>(rtp_header.header.payloadType) <<
123 ", ssrc=" << rtp_header.header.ssrc <<
125 int error = InsertPacketInternal(rtp_header, payload, length_bytes,
135 int NetEqImpl::InsertSyncPacket(const WebRtcRTPHeader& rtp_header,
139 << rtp_header.header.timestamp <<
140 ", sn=" << rtp_header.header.sequenceNumber <<
141 ", pt=" << static_cast<int>(rtp_header.header.payloadType) <
636 const RTPHeader* rtp_header = packet_buffer_->NextRtpHeader(); local
[all...] |
| /external/chromium_org/third_party/webrtc/modules/video_coding/main/test/ |
| pcap_file_reader.cc | 143 uint8_t pt = packets_[packet_numbers[0]].rtp_header.payloadType;
197 RTPHeader rtp_header; member in struct:webrtc::rtpplayer::PcapFileReaderImpl::RtpPacketMarker
274 rtp_parser.ParseRtcp(&marker.rtp_header);
277 if (!rtp_parser.Parse(marker.rtp_header, NULL)) {
282 uint32_t ssrc = marker.rtp_header.ssrc;
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