Searched defs:rtp_header (Results 1 - 25 of 27) sorted by relevance

12

/external/webrtc/webrtc/modules/audio_coding/neteq/tools/
H A Drtp_generator.cc20 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 Dneteq_external_decoder_test.cc39 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
H A Dneteq_performance_test.cc59 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);
H A Dneteq_rtpplay.cc302 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 Drtcp.cc33 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;
H A Dneteq_impl_unittest.cc267 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 Dneteq_impl.cc125 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 Dloudest_filter.cc43 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;
H A Dconference_transport.cc150 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);
/external/webrtc/webrtc/modules/rtp_rtcp/test/testAPI/
H A Dtest_api.h60 const webrtc::WebRtcRTPHeader* rtp_header) override;
64 webrtc::WebRtcRTPHeader rtp_header() const { return rtp_header_; } function in class:webrtc::TestRtpReceiver
H A Dtest_api.cc72 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
/external/webrtc/webrtc/modules/rtp_rtcp/source/
H A Drtp_receiver_video.cc52 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 Dproducer_fec.cc50 void RedPacket::CreateHeader(const uint8_t* rtp_header, size_t header_length, argument
53 memcpy(data_, rtp_header, header_length);
H A Drtp_receiver_audio.cc181 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 Drtp_receiver_impl.cc161 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 Dreceive_statistics_impl.cc512 void NullReceiveStatistics::IncomingPacket(const RTPHeader& rtp_header, argument
H A Drtp_sender_audio.cc352 RTPHeader rtp_header; local
353 rtp_parser.Parse(&rtp_header);
354 _rtpSender->UpdateAudioLevel(dataBuffer, packetSize, rtp_header,
H A Drtp_sender_video.cc306 RTPHeader rtp_header; local
307 rtp_parser.Parse(&rtp_header);
308 _rtpSender.UpdateVideoRotation(dataBuffer, packetSize, rtp_header,
H A Drtp_sender_unittest.cc58 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 Dacm_receiver.cc168 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
H A Daudio_coding_module_impl.cc655 const WebRtcRTPHeader& rtp_header) {
657 rtp_header,
653 IncomingPacket(const uint8_t* incoming_payload, const size_t payload_length, const WebRtcRTPHeader& rtp_header) argument
H A Daudio_coding_module_unittest_oldapi.cc67 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 DNETEQTEST_RTPpacket.cc283 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;
/external/webrtc/webrtc/video/
H A Dvie_receiver.cc237 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 Drtp_file_reader.cc317 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;

Completed in 251 milliseconds

12