/external/webrtc/webrtc/modules/rtp_rtcp/test/testAPI/ |
test_api.h | 60 const webrtc::WebRtcRTPHeader* rtp_header) override; 64 webrtc::WebRtcRTPHeader rtp_header() const { return rtp_header_; } function in class:webrtc::TestRtpReceiver
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/external/webrtc/webrtc/modules/audio_coding/neteq/tools/ |
neteq_performance_test.cc | 59 WebRtcRTPHeader rtp_header; local 65 rtp_gen.GetRtpHeader(kPayloadType, kInputBlockSizeSamples, &rtp_header); 82 lost = ((rtp_header.header.sequenceNumber - 1) % lossrate) == 0; 87 neteq->InsertPacket(rtp_header, input_payload, 96 &rtp_header);
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neteq_rtpplay.cc | 302 WebRtcRTPHeader* rtp_header, 306 if (IsComfortNoise(rtp_header->header.payloadType)) { 318 rtp_header->header.sequenceNumber + 1) { 320 next_packet->header().timestamp - rtp_header->header.timestamp) { 322 next_packet->header().timestamp - rtp_header->header.timestamp; 331 if (CodecTimestampRate(rtp_header->header.payloadType) != 332 CodecSampleRate(rtp_header->header.payloadType) || 333 rtp_header->header.payloadType == FLAGS_red || 334 rtp_header->header.payloadType == FLAGS_avt) { 356 switch (CodecSampleRate(rtp_header->header.payloadType)) 546 WebRtcRTPHeader rtp_header; local [all...] |
/external/webrtc/webrtc/voice_engine/test/auto_test/fakes/ |
conference_transport.cc | 150 webrtc::RTPHeader rtp_header; local 151 rtp_header_parser_->Parse(packet.data_, packet.len_, &rtp_header); 152 if (rtp_header.ssrc == kLocalSsrc) { 156 if (loudest_filter_.ForwardThisPacket(rtp_header)) { 157 destination = GetReceiverChannelForSsrc(rtp_header.ssrc);
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/external/webrtc/webrtc/modules/rtp_rtcp/source/ |
rtp_sender_audio.cc | 352 RTPHeader rtp_header; local 353 rtp_parser.Parse(&rtp_header); 354 _rtpSender->UpdateAudioLevel(dataBuffer, packetSize, rtp_header,
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rtp_sender_video.cc | 306 RTPHeader rtp_header; local 307 rtp_parser.Parse(&rtp_header); 308 _rtpSender.UpdateVideoRotation(dataBuffer, packetSize, rtp_header,
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rtp_sender_unittest.cc | 58 const uint8_t* GetPayloadData(const RTPHeader& rtp_header, 60 return packet + rtp_header.headerLength; 63 size_t GetPayloadDataLength(const RTPHeader& rtp_header, 65 return packet_length - rtp_header.headerLength - rtp_header.paddingLength; 150 void VerifyRTPHeaderCommon(const RTPHeader& rtp_header) { 151 VerifyRTPHeaderCommon(rtp_header, kMarkerBit); 154 void VerifyRTPHeaderCommon(const RTPHeader& rtp_header, bool marker_bit) { 155 EXPECT_EQ(marker_bit, rtp_header.markerBit); 156 EXPECT_EQ(payload_, rtp_header.payloadType) 202 webrtc::RTPHeader rtp_header; local 336 webrtc::RTPHeader rtp_header; local 368 webrtc::RTPHeader rtp_header; local 408 webrtc::RTPHeader rtp_header; local 436 webrtc::RTPHeader rtp_header; local 477 webrtc::RTPHeader rtp_header; local 505 webrtc::RTPHeader rtp_header; local 525 webrtc::RTPHeader rtp_header; local 579 webrtc::RTPHeader rtp_header; local 665 webrtc::RTPHeader rtp_header; local 725 webrtc::RTPHeader rtp_header; local 770 webrtc::RTPHeader rtp_header; local 936 webrtc::RTPHeader rtp_header; local 1219 webrtc::RTPHeader rtp_header; local 1248 webrtc::RTPHeader rtp_header; local 1305 webrtc::RTPHeader rtp_header; local [all...] |
rtp_sender.cc | 581 RTPHeader rtp_header; local 582 rtp_parser.Parse(&rtp_header); 583 bytes_left -= static_cast<int>(length - rtp_header.headerLength); 672 RTPHeader rtp_header; local 673 rtp_parser.Parse(&rtp_header); 677 padding_packet, length, rtp_header, now_ms - capture_time_ms); 680 UpdateAbsoluteSendTime(padding_packet, length, rtp_header, now_ms); 685 UpdateTransportSequenceNumber(padding_packet, length, rtp_header); 696 UpdateRtpStats(padding_packet, length, rtp_header, over_rtx, false); 914 RTPHeader rtp_header; local 1033 RTPHeader rtp_header; local 1828 RTPHeader rtp_header; local [all...] |
/external/webrtc/webrtc/video/ |
vie_receiver.cc | 237 const WebRtcRTPHeader* rtp_header) { 238 WebRtcRTPHeader rtp_header_with_ntp = *rtp_header; 240 ntp_estimator_->Estimate(rtp_header->header.timestamp); 391 WebRtcRTPHeader rtp_header = {}; local 392 rtp_header.header = header; 393 rtp_header.header.payloadType = last_media_payload_type; 394 rtp_header.header.paddingLength = 0; 401 rtp_header.type.Video.codec = payload_specific.Video.videoCodecType; 402 rtp_header.type.Video.rotation = kVideoRotation_0; 404 rtp_header.type.Video.rotation [all...] |
/external/webrtc/webrtc/modules/audio_coding/neteq/ |
neteq_impl_unittest.cc | 267 WebRtcRTPHeader rtp_header; local 268 rtp_header.header.payloadType = kPayloadType; 269 rtp_header.header.sequenceNumber = kFirstSequenceNumber; 270 rtp_header.header.timestamp = kFirstTimestamp; 271 rtp_header.header.ssrc = kSsrc; 327 .WillOnce(Return(&rtp_header.header)); 363 neteq_->InsertPacket(rtp_header, payload, kFirstReceiveTime); 366 rtp_header.header.timestamp += 160; 367 rtp_header.header.sequenceNumber += 1; 368 neteq_->InsertPacket(rtp_header, payload, kFirstReceiveTime + 155) 380 WebRtcRTPHeader rtp_header; local 421 WebRtcRTPHeader rtp_header; local 515 WebRtcRTPHeader rtp_header; local 609 WebRtcRTPHeader rtp_header; local 676 WebRtcRTPHeader rtp_header; local 813 WebRtcRTPHeader rtp_header; local 908 WebRtcRTPHeader rtp_header; local 944 WebRtcRTPHeader rtp_header; local 1013 WebRtcRTPHeader rtp_header; local 1138 WebRtcRTPHeader rtp_header; local [all...] |
neteq_impl.cc | 125 int NetEqImpl::InsertPacket(const WebRtcRTPHeader& rtp_header, 131 InsertPacketInternal(rtp_header, payload, receive_timestamp, false); 139 int NetEqImpl::InsertSyncPacket(const WebRtcRTPHeader& rtp_header, 144 InsertPacketInternal(rtp_header, kSyncPayload, receive_timestamp, true); 450 int NetEqImpl::InsertPacketInternal(const WebRtcRTPHeader& rtp_header, 460 if (decoder_database_->IsDtmf(rtp_header.header.payloadType) || 461 decoder_database_->IsRed(rtp_header.header.payloadType) || 462 decoder_database_->IsComfortNoise(rtp_header.header.payloadType)) { 464 << static_cast<int>(rtp_header.header.payloadType); 468 rtp_header.header.payloadType != current_rtp_payload_type_ | 686 const RTPHeader* rtp_header = packet_buffer_->NextRtpHeader(); local [all...] |
neteq_unittest.cc | 413 WebRtcRTPHeader rtp_header; local 414 packet_->ConvertHeader(&rtp_header); 416 rtp_header, [all...] |
/external/webrtc/webrtc/test/ |
rtp_file_reader.cc | 317 uint8_t pt = packets_[packet_indices[0]].rtp_header.payloadType; 380 RTPHeader rtp_header; member in struct:webrtc::test::PcapReader::RtpPacketMarker 458 rtp_parser.ParseRtcp(&marker.rtp_header); 461 if (!rtp_parser.Parse(&marker.rtp_header, nullptr)) { 466 uint32_t ssrc = marker.rtp_header.ssrc;
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