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Searched refs:rtp_header (Results 1 – 25 of 55) sorted by relevance

123

/external/webrtc/webrtc/modules/rtp_rtcp/source/
Drtp_sender_unittest.cc58 const uint8_t* GetPayloadData(const RTPHeader& rtp_header, in GetPayloadData() argument
60 return packet + rtp_header.headerLength; in GetPayloadData()
63 size_t GetPayloadDataLength(const RTPHeader& rtp_header, in GetPayloadDataLength() argument
65 return packet_length - rtp_header.headerLength - rtp_header.paddingLength; in GetPayloadDataLength()
150 void VerifyRTPHeaderCommon(const RTPHeader& rtp_header) { in VerifyRTPHeaderCommon() argument
151 VerifyRTPHeaderCommon(rtp_header, kMarkerBit); in VerifyRTPHeaderCommon()
154 void VerifyRTPHeaderCommon(const RTPHeader& rtp_header, bool marker_bit) { in VerifyRTPHeaderCommon() argument
155 EXPECT_EQ(marker_bit, rtp_header.markerBit); in VerifyRTPHeaderCommon()
156 EXPECT_EQ(payload_, rtp_header.payloadType); in VerifyRTPHeaderCommon()
157 EXPECT_EQ(kSeqNum, rtp_header.sequenceNumber); in VerifyRTPHeaderCommon()
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Drtp_receiver_video.cc52 int32_t RTPReceiverVideo::ParseRtpPacket(WebRtcRTPHeader* rtp_header, in ParseRtpPacket() argument
60 "seqnum", rtp_header->header.sequenceNumber, "timestamp", in ParseRtpPacket()
61 rtp_header->header.timestamp); in ParseRtpPacket()
62 rtp_header->type.Video.codec = specific_payload.Video.videoCodecType; in ParseRtpPacket()
64 RTC_DCHECK_GE(payload_length, rtp_header->header.paddingLength); in ParseRtpPacket()
66 payload_length - rtp_header->header.paddingLength; in ParseRtpPacket()
69 return data_callback_->OnReceivedPayloadData(NULL, 0, rtp_header) == 0 ? 0 in ParseRtpPacket()
75 RtpDepacketizer::Create(rtp_header->type.Video.codec)); in ParseRtpPacket()
81 rtp_header->type.Video.isFirstPacket = is_first_packet; in ParseRtpPacket()
86 rtp_header->frameType = parsed_payload.frame_type; in ParseRtpPacket()
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Drtp_receiver_impl.cc161 const RTPHeader& rtp_header, in IncomingRtpPacket() argument
167 CheckSSRCChanged(rtp_header); in IncomingRtpPacket()
172 if (CheckPayloadChanged(rtp_header, first_payload_byte, &is_red, in IncomingRtpPacket()
184 webrtc_rtp_header.header = rtp_header; in IncomingRtpPacket()
187 size_t payload_data_length = payload_length - rtp_header.paddingLength; in IncomingRtpPacket()
194 last_received_sequence_number_ + 1 == rtp_header.sequenceNumber && in IncomingRtpPacket()
195 last_received_timestamp_ != rtp_header.timestamp; in IncomingRtpPacket()
216 if (last_received_timestamp_ != rtp_header.timestamp) { in IncomingRtpPacket()
217 last_received_timestamp_ = rtp_header.timestamp; in IncomingRtpPacket()
220 last_received_sequence_number_ = rtp_header.sequenceNumber; in IncomingRtpPacket()
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Drtp_receiver_audio.cc181 int32_t RTPReceiverAudio::ParseRtpPacket(WebRtcRTPHeader* rtp_header, in ParseRtpPacket() argument
189 "seqnum", rtp_header->header.sequenceNumber, "timestamp", in ParseRtpPacket()
190 rtp_header->header.timestamp); in ParseRtpPacket()
191 rtp_header->type.Audio.numEnergy = rtp_header->header.numCSRCs; in ParseRtpPacket()
192 num_energy_ = rtp_header->type.Audio.numEnergy; in ParseRtpPacket()
193 if (rtp_header->type.Audio.numEnergy > 0 && in ParseRtpPacket()
194 rtp_header->type.Audio.numEnergy <= kRtpCsrcSize) { in ParseRtpPacket()
196 rtp_header->type.Audio.arrOfEnergy, in ParseRtpPacket()
197 rtp_header->type.Audio.numEnergy); in ParseRtpPacket()
200 return ParseAudioCodecSpecific(rtp_header, in ParseRtpPacket()
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Drtp_sender.cc581 RTPHeader rtp_header; in TrySendRedundantPayloads() local
582 rtp_parser.Parse(&rtp_header); in TrySendRedundantPayloads()
583 bytes_left -= static_cast<int>(length - rtp_header.headerLength); in TrySendRedundantPayloads()
672 RTPHeader rtp_header; in SendPadData() local
673 rtp_parser.Parse(&rtp_header); in SendPadData()
677 padding_packet, length, rtp_header, now_ms - capture_time_ms); in SendPadData()
680 UpdateAbsoluteSendTime(padding_packet, length, rtp_header, now_ms); in SendPadData()
685 UpdateTransportSequenceNumber(padding_packet, length, rtp_header); in SendPadData()
696 UpdateRtpStats(padding_packet, length, rtp_header, over_rtx, false); in SendPadData()
914 RTPHeader rtp_header; in PrepareAndSendPacket() local
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Drtp_receiver_impl.h44 bool IncomingRtpPacket(const RTPHeader& rtp_header,
70 void CheckSSRCChanged(const RTPHeader& rtp_header);
71 void CheckCSRC(const WebRtcRTPHeader& rtp_header);
72 int32_t CheckPayloadChanged(const RTPHeader& rtp_header,
Drtp_sender.h82 const RTPHeader& rtp_header,
194 const RTPHeader& rtp_header,
201 const RTPHeader& rtp_header,
207 const RTPHeader& rtp_header,
362 const RTPHeader& rtp_header,
367 const RTPHeader& rtp_header,
371 const RTPHeader& rtp_header,
378 const RTPHeader& rtp_header) const;
/external/webrtc/webrtc/modules/audio_coding/neteq/
Dneteq_impl_unittest.cc267 WebRtcRTPHeader rtp_header; in TEST_F() local
268 rtp_header.header.payloadType = kPayloadType; in TEST_F()
269 rtp_header.header.sequenceNumber = kFirstSequenceNumber; in TEST_F()
270 rtp_header.header.timestamp = kFirstTimestamp; in TEST_F()
271 rtp_header.header.ssrc = kSsrc; in TEST_F()
327 .WillOnce(Return(&rtp_header.header)); in TEST_F()
363 neteq_->InsertPacket(rtp_header, payload, kFirstReceiveTime); in TEST_F()
366 rtp_header.header.timestamp += 160; in TEST_F()
367 rtp_header.header.sequenceNumber += 1; in TEST_F()
368 neteq_->InsertPacket(rtp_header, payload, kFirstReceiveTime + 155); in TEST_F()
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Drtcp.cc33 void Rtcp::Update(const RTPHeader& rtp_header, uint32_t receive_timestamp) { in Update() argument
36 int16_t sn_diff = rtp_header.sequenceNumber - max_seq_no_; in Update()
38 if (rtp_header.sequenceNumber < max_seq_no_) { in Update()
42 max_seq_no_ = rtp_header.sequenceNumber; in Update()
48 int32_t ts_diff = receive_timestamp - (rtp_header.timestamp - transit_); in Update()
54 transit_ = rtp_header.timestamp - receive_timestamp; in Update()
Dneteq_external_decoder_unittest.cc129 void InsertPacket(WebRtcRTPHeader rtp_header, in InsertPacket() argument
134 IncomingPacket(_, payload.size(), rtp_header.header.sequenceNumber, in InsertPacket()
135 rtp_header.header.timestamp, receive_timestamp)); in InsertPacket()
136 NetEqExternalDecoderTest::InsertPacket(rtp_header, payload, in InsertPacket()
212 void InsertPacket(WebRtcRTPHeader rtp_header, in InsertPacket() argument
216 ASSERT_EQ(NetEq::kOK, neteq_internal_->InsertPacket(rtp_header, payload, in InsertPacket()
220 NetEqExternalDecoderUnitTest::InsertPacket(rtp_header, payload, in InsertPacket()
Dneteq_impl.cc125 int NetEqImpl::InsertPacket(const WebRtcRTPHeader& rtp_header, in InsertPacket() argument
131 InsertPacketInternal(rtp_header, payload, receive_timestamp, false); in InsertPacket()
139 int NetEqImpl::InsertSyncPacket(const WebRtcRTPHeader& rtp_header, in InsertSyncPacket() argument
144 InsertPacketInternal(rtp_header, kSyncPayload, receive_timestamp, true); in InsertSyncPacket()
450 int NetEqImpl::InsertPacketInternal(const WebRtcRTPHeader& rtp_header, in InsertPacketInternal() argument
460 if (decoder_database_->IsDtmf(rtp_header.header.payloadType) || in InsertPacketInternal()
461 decoder_database_->IsRed(rtp_header.header.payloadType) || in InsertPacketInternal()
462 decoder_database_->IsComfortNoise(rtp_header.header.payloadType)) { in InsertPacketInternal()
464 << static_cast<int>(rtp_header.header.payloadType); in InsertPacketInternal()
468 rtp_header.header.payloadType != current_rtp_payload_type_ || in InsertPacketInternal()
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/external/webrtc/webrtc/modules/audio_coding/neteq/tools/
Drtp_generator.cc20 WebRtcRTPHeader* rtp_header) { in GetRtpHeader() argument
21 assert(rtp_header); in GetRtpHeader()
22 if (!rtp_header) { in GetRtpHeader()
25 rtp_header->header.sequenceNumber = seq_number_++; in GetRtpHeader()
26 rtp_header->header.timestamp = timestamp_; in GetRtpHeader()
28 rtp_header->header.payloadType = payload_type; in GetRtpHeader()
29 rtp_header->header.markerBit = false; in GetRtpHeader()
30 rtp_header->header.ssrc = ssrc_; in GetRtpHeader()
31 rtp_header->header.numCSRCs = 0; in GetRtpHeader()
32 rtp_header->frameType = kAudioFrameSpeech; in GetRtpHeader()
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Dneteq_rtpplay.cc302 WebRtcRTPHeader* rtp_header, in ReplacePayload() argument
306 if (IsComfortNoise(rtp_header->header.payloadType)) { in ReplacePayload()
318 rtp_header->header.sequenceNumber + 1) { in ReplacePayload()
320 next_packet->header().timestamp - rtp_header->header.timestamp) { in ReplacePayload()
322 next_packet->header().timestamp - rtp_header->header.timestamp; in ReplacePayload()
331 if (CodecTimestampRate(rtp_header->header.payloadType) != in ReplacePayload()
332 CodecSampleRate(rtp_header->header.payloadType) || in ReplacePayload()
333 rtp_header->header.payloadType == FLAGS_red || in ReplacePayload()
334 rtp_header->header.payloadType == FLAGS_avt) { in ReplacePayload()
356 switch (CodecSampleRate(rtp_header->header.payloadType)) { in ReplacePayload()
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Dneteq_performance_test.cc59 WebRtcRTPHeader rtp_header; in Run() local
65 rtp_gen.GetRtpHeader(kPayloadType, kInputBlockSizeSamples, &rtp_header); in Run()
82 lost = ((rtp_header.header.sequenceNumber - 1) % lossrate) == 0; in Run()
87 neteq->InsertPacket(rtp_header, input_payload, in Run()
96 &rtp_header); in Run()
Drtp_generator.h44 WebRtcRTPHeader* rtp_header);
73 WebRtcRTPHeader* rtp_header) override;
Dneteq_external_decoder_test.cc39 WebRtcRTPHeader rtp_header, in InsertPacket() argument
43 neteq_->InsertPacket(rtp_header, payload, receive_timestamp)); in InsertPacket()
/external/webrtc/webrtc/video/
Dvie_receiver.cc237 const WebRtcRTPHeader* rtp_header) { in OnReceivedPayloadData() argument
238 WebRtcRTPHeader rtp_header_with_ntp = *rtp_header; in OnReceivedPayloadData()
240 ntp_estimator_->Estimate(rtp_header->header.timestamp); in OnReceivedPayloadData()
391 WebRtcRTPHeader rtp_header = {}; in NotifyReceiverOfFecPacket() local
392 rtp_header.header = header; in NotifyReceiverOfFecPacket()
393 rtp_header.header.payloadType = last_media_payload_type; in NotifyReceiverOfFecPacket()
394 rtp_header.header.paddingLength = 0; in NotifyReceiverOfFecPacket()
401 rtp_header.type.Video.codec = payload_specific.Video.videoCodecType; in NotifyReceiverOfFecPacket()
402 rtp_header.type.Video.rotation = kVideoRotation_0; in NotifyReceiverOfFecPacket()
404 rtp_header.type.Video.rotation = in NotifyReceiverOfFecPacket()
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/external/webrtc/webrtc/voice_engine/test/auto_test/fakes/
Dloudest_filter.cc43 bool LoudestFilter::ForwardThisPacket(const webrtc::RTPHeader& rtp_header) { in ForwardThisPacket() argument
47 int source_ssrc = rtp_header.ssrc; in ForwardThisPacket()
48 int audio_level = rtp_header.extension.hasAudioLevel ? in ForwardThisPacket()
49 rtp_header.extension.audioLevel : kInvalidAudioLevel; in ForwardThisPacket()
Dconference_transport.cc150 webrtc::RTPHeader rtp_header; in SendPacket() local
151 rtp_header_parser_->Parse(packet.data_, packet.len_, &rtp_header); in SendPacket()
152 if (rtp_header.ssrc == kLocalSsrc) { in SendPacket()
156 if (loudest_filter_.ForwardThisPacket(rtp_header)) { in SendPacket()
157 destination = GetReceiverChannelForSsrc(rtp_header.ssrc); in SendPacket()
/external/adhd/cras/src/server/
Dcras_a2dp_info.c79 sizeof(struct rtp_header) + sizeof(struct rtp_payload); in init_a2dp()
110 sizeof(struct rtp_header) + sizeof(struct rtp_payload); in a2dp_drain()
119 struct rtp_header *header; in avdtp_write()
122 header = (struct rtp_header *)a2dp->a2dp_buf; in avdtp_write()
184 link_mtu - sizeof(struct rtp_header) - sizeof(struct rtp_payload)) in a2dp_write()
/external/webrtc/webrtc/modules/video_coding/test/
Dreceiver_tests.h33 const webrtc::WebRtcRTPHeader* rtp_header) override { in OnReceivedPayloadData() argument
34 return vcm_->IncomingPacket(payload_data, payload_size, *rtp_header); in OnReceivedPayloadData()
/external/webrtc/webrtc/modules/audio_coding/neteq/test/
DNETEQTEST_RTPpacket.cc283 void NETEQTEST_RTPpacket::parseHeader(webrtc::WebRtcRTPHeader* rtp_header) { in parseHeader() argument
287 if (rtp_header) { in parseHeader()
288 rtp_header->header.markerBit = _rtpInfo.header.markerBit; in parseHeader()
289 rtp_header->header.payloadType = _rtpInfo.header.payloadType; in parseHeader()
290 rtp_header->header.sequenceNumber = _rtpInfo.header.sequenceNumber; in parseHeader()
291 rtp_header->header.timestamp = _rtpInfo.header.timestamp; in parseHeader()
292 rtp_header->header.ssrc = _rtpInfo.header.ssrc; in parseHeader()
/external/adhd/cras/src/common/
Drtp.h26 struct rtp_header { struct
51 struct rtp_header { argument
/external/webrtc/webrtc/modules/rtp_rtcp/test/testAPI/
Dtest_api.h60 const webrtc::WebRtcRTPHeader* rtp_header) override;
64 webrtc::WebRtcRTPHeader rtp_header() const { return rtp_header_; } in rtp_header() function
/external/webrtc/webrtc/modules/audio_coding/acm2/
Daudio_coding_module_unittest_oldapi.cc67 void Populate(WebRtcRTPHeader* rtp_header) { in Populate() argument
68 rtp_header->header.sequenceNumber = 0xABCD; in Populate()
69 rtp_header->header.timestamp = 0xABCDEF01; in Populate()
70 rtp_header->header.payloadType = payload_type_; in Populate()
71 rtp_header->header.markerBit = false; in Populate()
72 rtp_header->header.ssrc = 0x1234; in Populate()
73 rtp_header->header.numCSRCs = 0; in Populate()
74 rtp_header->frameType = kAudioFrameSpeech; in Populate()
76 rtp_header->header.payload_type_frequency = kSampleRateHz; in Populate()
77 rtp_header->type.Audio.channel = 1; in Populate()
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