/external/webrtc/webrtc/modules/audio_coding/neteq/tools/ |
H A D | rtp_generator.cc | 20 WebRtcRTPHeader* rtp_header) { 21 assert(rtp_header); 22 if (!rtp_header) { 25 rtp_header->header.sequenceNumber = seq_number_++; 26 rtp_header->header.timestamp = timestamp_; 28 rtp_header->header.payloadType = payload_type; 29 rtp_header->header.markerBit = false; 30 rtp_header->header.ssrc = ssrc_; 31 rtp_header->header.numCSRCs = 0; 32 rtp_header 18 GetRtpHeader(uint8_t payload_type, size_t payload_length_samples, WebRtcRTPHeader* rtp_header) argument 47 GetRtpHeader(uint8_t payload_type, size_t payload_length_samples, WebRtcRTPHeader* rtp_header) argument [all...] |
H A D | neteq_external_decoder_test.cc | 39 WebRtcRTPHeader rtp_header, 43 neteq_->InsertPacket(rtp_header, payload, receive_timestamp)); 38 InsertPacket( WebRtcRTPHeader rtp_header, rtc::ArrayView<const uint8_t> payload, uint32_t receive_timestamp) argument
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H A D | 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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H A D | 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 297 ReplacePayload(webrtc::test::InputAudioFile* replacement_audio_file, rtc::scoped_ptr<int16_t[]>* replacement_audio, rtc::scoped_ptr<uint8_t[]>* payload, size_t* payload_mem_size_bytes, size_t* frame_size_samples, WebRtcRTPHeader* rtp_header, const webrtc::test::Packet* next_packet) argument 546 WebRtcRTPHeader rtp_header; local [all...] |
/external/webrtc/webrtc/modules/audio_coding/neteq/ |
H A D | rtcp.cc | 33 void Rtcp::Update(const RTPHeader& rtp_header, uint32_t receive_timestamp) { argument 36 int16_t sn_diff = rtp_header.sequenceNumber - max_seq_no_; 38 if (rtp_header.sequenceNumber < max_seq_no_) { 42 max_seq_no_ = rtp_header.sequenceNumber; 48 int32_t ts_diff = receive_timestamp - (rtp_header.timestamp - transit_); 54 transit_ = rtp_header.timestamp - receive_timestamp;
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H A D | 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, payloa 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...] |
H A D | neteq_impl.cc | 125 int NetEqImpl::InsertPacket(const WebRtcRTPHeader& rtp_header, argument 131 InsertPacketInternal(rtp_header, payload, receive_timestamp, false); 139 int NetEqImpl::InsertSyncPacket(const WebRtcRTPHeader& rtp_header, argument 144 InsertPacketInternal(rtp_header, kSyncPayload, receive_timestamp, true); 450 int NetEqImpl::InsertPacketInternal(const WebRtcRTPHeader& rtp_header, argument 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 686 const RTPHeader* rtp_header = packet_buffer_->NextRtpHeader(); local [all...] |
/external/webrtc/webrtc/voice_engine/test/auto_test/fakes/ |
H A D | loudest_filter.cc | 43 bool LoudestFilter::ForwardThisPacket(const webrtc::RTPHeader& rtp_header) { argument 47 int source_ssrc = rtp_header.ssrc; 48 int audio_level = rtp_header.extension.hasAudioLevel ? 49 rtp_header.extension.audioLevel : kInvalidAudioLevel;
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H A D | 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/test/testAPI/ |
H A D | 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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H A D | test_api.cc | 72 const webrtc::WebRtcRTPHeader* rtp_header) { 75 memcpy(&rtp_header_, rtp_header, sizeof(rtp_header_)); 69 OnReceivedPayloadData( const uint8_t* payload_data, const size_t payload_size, const webrtc::WebRtcRTPHeader* rtp_header) argument
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/external/webrtc/webrtc/modules/rtp_rtcp/source/ |
H A D | rtp_receiver_video.cc | 52 int32_t RTPReceiverVideo::ParseRtpPacket(WebRtcRTPHeader* rtp_header, argument 60 "seqnum", rtp_header->header.sequenceNumber, "timestamp", 61 rtp_header->header.timestamp); 62 rtp_header->type.Video.codec = specific_payload.Video.videoCodecType; 64 RTC_DCHECK_GE(payload_length, rtp_header->header.paddingLength); 66 payload_length - rtp_header->header.paddingLength; 69 return data_callback_->OnReceivedPayloadData(NULL, 0, rtp_header) == 0 ? 0 75 RtpDepacketizer::Create(rtp_header->type.Video.codec)); 81 rtp_header->type.Video.isFirstPacket = is_first_packet; 86 rtp_header [all...] |
H A D | producer_fec.cc | 50 void RedPacket::CreateHeader(const uint8_t* rtp_header, size_t header_length, argument 53 memcpy(data_, rtp_header, header_length);
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H A D | rtp_receiver_audio.cc | 181 int32_t RTPReceiverAudio::ParseRtpPacket(WebRtcRTPHeader* rtp_header, argument 189 "seqnum", rtp_header->header.sequenceNumber, "timestamp", 190 rtp_header->header.timestamp); 191 rtp_header->type.Audio.numEnergy = rtp_header->header.numCSRCs; 192 num_energy_ = rtp_header->type.Audio.numEnergy; 193 if (rtp_header->type.Audio.numEnergy > 0 && 194 rtp_header->type.Audio.numEnergy <= kRtpCsrcSize) { 196 rtp_header->type.Audio.arrOfEnergy, 197 rtp_header 279 ParseAudioCodecSpecific( WebRtcRTPHeader* rtp_header, const uint8_t* payload_data, size_t payload_length, const AudioPayload& audio_specific, bool is_red) argument [all...] |
H A D | rtp_receiver_impl.cc | 161 const RTPHeader& rtp_header, 167 CheckSSRCChanged(rtp_header); 172 if (CheckPayloadChanged(rtp_header, first_payload_byte, &is_red, 184 webrtc_rtp_header.header = rtp_header; 187 size_t payload_data_length = payload_length - rtp_header.paddingLength; 194 last_received_sequence_number_ + 1 == rtp_header.sequenceNumber && 195 last_received_timestamp_ != rtp_header.timestamp; 216 if (last_received_timestamp_ != rtp_header.timestamp) { 217 last_received_timestamp_ = rtp_header.timestamp; 220 last_received_sequence_number_ = rtp_header 160 IncomingRtpPacket( const RTPHeader& rtp_header, const uint8_t* payload, size_t payload_length, PayloadUnion payload_specific, bool in_order) argument 251 CheckSSRCChanged(const RTPHeader& rtp_header) argument 320 CheckPayloadChanged(const RTPHeader& rtp_header, const int8_t first_payload_byte, bool* is_red, PayloadUnion* specific_payload) argument 407 CheckCSRC(const WebRtcRTPHeader& rtp_header) argument [all...] |
H A D | receive_statistics_impl.cc | 512 void NullReceiveStatistics::IncomingPacket(const RTPHeader& rtp_header, argument
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H A D | 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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H A D | 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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H A D | rtp_sender_unittest.cc | 58 const uint8_t* GetPayloadData(const RTPHeader& rtp_header, argument 60 return packet + rtp_header.headerLength; 63 size_t GetPayloadDataLength(const RTPHeader& rtp_header, argument 65 return packet_length - rtp_header.headerLength - rtp_header.paddingLength; 150 void VerifyRTPHeaderCommon(const RTPHeader& rtp_header) { argument 151 VerifyRTPHeaderCommon(rtp_header, kMarkerBit); 154 void VerifyRTPHeaderCommon(const RTPHeader& rtp_header, bool marker_bit) { argument 155 EXPECT_EQ(marker_bit, rtp_header.markerBit); 156 EXPECT_EQ(payload_, rtp_header 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...] |
/external/webrtc/webrtc/modules/audio_coding/acm2/ |
H A D | acm_receiver.cc | 168 int AcmReceiver::InsertPacket(const WebRtcRTPHeader& rtp_header, argument 171 const RTPHeader* header = &rtp_header.header; // Just a shorthand. 203 if (neteq_->InsertPacket(rtp_header, incoming_payload, receive_timestamp) < 507 const RTPHeader& rtp_header, 509 auto it = decoders_.find(rtp_header.payloadType); 506 RtpHeaderToDecoder( const RTPHeader& rtp_header, uint8_t payload_type) const argument
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H A D | audio_coding_module_impl.cc | 655 const WebRtcRTPHeader& rtp_header) { 657 rtp_header, 653 IncomingPacket(const uint8_t* incoming_payload, const size_t payload_length, const WebRtcRTPHeader& rtp_header) argument
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H A D | audio_coding_module_unittest_oldapi.cc | 67 void Populate(WebRtcRTPHeader* rtp_header) { argument 68 rtp_header->header.sequenceNumber = 0xABCD; 69 rtp_header->header.timestamp = 0xABCDEF01; 70 rtp_header->header.payloadType = payload_type_; 71 rtp_header->header.markerBit = false; 72 rtp_header->header.ssrc = 0x1234; 73 rtp_header->header.numCSRCs = 0; 74 rtp_header->frameType = kAudioFrameSpeech; 76 rtp_header->header.payload_type_frequency = kSampleRateHz; 77 rtp_header 81 Forward(WebRtcRTPHeader* rtp_header) argument [all...] |
/external/webrtc/webrtc/modules/audio_coding/neteq/test/ |
H A D | NETEQTEST_RTPpacket.cc | 283 void NETEQTEST_RTPpacket::parseHeader(webrtc::WebRtcRTPHeader* rtp_header) { argument 287 if (rtp_header) { 288 rtp_header->header.markerBit = _rtpInfo.header.markerBit; 289 rtp_header->header.payloadType = _rtpInfo.header.payloadType; 290 rtp_header->header.sequenceNumber = _rtpInfo.header.sequenceNumber; 291 rtp_header->header.timestamp = _rtpInfo.header.timestamp; 292 rtp_header->header.ssrc = _rtpInfo.header.ssrc;
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/external/webrtc/webrtc/video/ |
H A D | 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 235 OnReceivedPayloadData(const uint8_t* payload_data, const size_t payload_size, const WebRtcRTPHeader* rtp_header) argument [all...] |
/external/webrtc/webrtc/test/ |
H A D | 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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