/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;
|