/external/webrtc/modules/audio_coding/neteq/ |
D | neteq_impl_unittest.cc | 187 RTPHeader rtp_header; in TestDtmfPacket() local 188 rtp_header.payloadType = kPayloadType; in TestDtmfPacket() 189 rtp_header.sequenceNumber = 0x1234; in TestDtmfPacket() 190 rtp_header.timestamp = 0x12345678; in TestDtmfPacket() 191 rtp_header.ssrc = 0x87654321; in TestDtmfPacket() 197 EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload)); in TestDtmfPacket() 294 RTPHeader rtp_header; in TEST_F() local 295 rtp_header.payloadType = kPayloadType; in TEST_F() 296 rtp_header.sequenceNumber = kFirstSequenceNumber; in TEST_F() 297 rtp_header.timestamp = kFirstTimestamp; in TEST_F() [all …]
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D | neteq_impl.cc | 192 int NetEqImpl::InsertPacket(const RTPHeader& rtp_header, in InsertPacket() argument 197 if (InsertPacketInternal(rtp_header, payload) != 0) { in InsertPacket() 544 int NetEqImpl::InsertPacketInternal(const RTPHeader& rtp_header, in InsertPacketInternal() argument 556 packet_list.push_back([&rtp_header, &payload, &receive_time_ms] { in InsertPacketInternal() 559 packet.payload_type = rtp_header.payloadType; in InsertPacketInternal() 560 packet.sequence_number = rtp_header.sequenceNumber; in InsertPacketInternal() 561 packet.timestamp = rtp_header.timestamp; in InsertPacketInternal() 563 packet.packet_info = RtpPacketInfo(rtp_header, receive_time_ms); in InsertPacketInternal() 576 if (!decoder_database_->IsRed(rtp_header.payloadType)) { in InsertPacketInternal() 608 nack_->UpdateLastReceivedPacket(rtp_header.sequenceNumber, in InsertPacketInternal() [all …]
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/external/webrtc/modules/audio_coding/neteq/tools/ |
D | rtp_generator.cc | 20 RTPHeader* rtp_header) { in GetRtpHeader() argument 21 assert(rtp_header); in GetRtpHeader() 22 if (!rtp_header) { in GetRtpHeader() 25 rtp_header->sequenceNumber = seq_number_++; in GetRtpHeader() 26 rtp_header->timestamp = timestamp_; in GetRtpHeader() 28 rtp_header->payloadType = payload_type; in GetRtpHeader() 29 rtp_header->markerBit = false; in GetRtpHeader() 30 rtp_header->ssrc = ssrc_; in GetRtpHeader() 31 rtp_header->numCSRCs = 0; in GetRtpHeader() 48 RTPHeader* rtp_header) { in GetRtpHeader() argument [all …]
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D | neteq_performance_test.cc | 63 RTPHeader rtp_header; in Run() local 69 rtp_gen.GetRtpHeader(kPayloadType, kInputBlockSizeSamples, &rtp_header); in Run() 86 lost = ((rtp_header.sequenceNumber - 1) % lossrate) == 0; in Run() 90 int error = neteq->InsertPacket(rtp_header, input_payload); in Run() 97 kPayloadType, kInputBlockSizeSamples, &rtp_header); in Run()
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D | rtp_generator.h | 42 RTPHeader* rtp_header); 71 RTPHeader* rtp_header) override;
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D | rtp_encode.cc | 132 const uint8_t rtp_header[] = {0x80, in SendData() local 144 static_assert(sizeof(rtp_header) == kRtpHeaderLength, ""); in SendData() 146 fwrite(rtp_header, sizeof(uint8_t), kRtpHeaderLength, out_file_), in SendData()
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/external/webrtc/modules/audio_coding/test/ |
D | Channel.cc | 28 RTPHeader rtp_header; in SendData() local 32 rtp_header.markerBit = false; in SendData() 33 rtp_header.ssrc = 0; in SendData() 34 rtp_header.sequenceNumber = in SendData() 38 rtp_header.payloadType = payloadType; in SendData() 39 rtp_header.timestamp = (external_send_timestamp_ < 0) in SendData() 53 _rtp_header = rtp_header; in SendData() 56 rtp_header = _rtp_header; in SendData() 67 CalcStatistics(rtp_header, payloadSize); in SendData() 88 _receiverACM->IncomingPacket(_payloadData, payloadDataSize, rtp_header); in SendData() [all …]
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D | RTPFile.cc | 28 void RTPStream::ParseRTPHeader(RTPHeader* rtp_header, in ParseRTPHeader() argument 30 rtp_header->payloadType = rtpHeader[1]; in ParseRTPHeader() 31 rtp_header->sequenceNumber = in ParseRTPHeader() 33 rtp_header->timestamp = (static_cast<uint32_t>(rtpHeader[4]) << 24) | in ParseRTPHeader() 37 rtp_header->ssrc = (static_cast<uint32_t>(rtpHeader[8]) << 24) | in ParseRTPHeader() 104 size_t RTPBuffer::Read(RTPHeader* rtp_header, in Read() argument 112 rtp_header->markerBit = 1; in Read() 113 rtp_header->payloadType = packet->payloadType; in Read() 114 rtp_header->sequenceNumber = packet->seqNo; in Read() 115 rtp_header->ssrc = 0; in Read() [all …]
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D | TestAllCodecs.cc | 69 RTPHeader rtp_header; in SendData() local 72 rtp_header.markerBit = false; in SendData() 73 rtp_header.ssrc = 0; in SendData() 74 rtp_header.sequenceNumber = sequence_number_++; in SendData() 75 rtp_header.payloadType = payload_type; in SendData() 76 rtp_header.timestamp = timestamp; in SendData() 87 receiver_acm_->IncomingPacket(payload_data_, payload_size, rtp_header); in SendData()
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D | RTPFile.h | 49 void ParseRTPHeader(RTPHeader* rtp_header, const uint8_t* rtpHeader); 84 size_t Read(RTPHeader* rtp_header, 117 size_t Read(RTPHeader* rtp_header,
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D | TestStereo.cc | 49 RTPHeader rtp_header; in SendData() local 52 rtp_header.markerBit = false; in SendData() 53 rtp_header.ssrc = 0; in SendData() 54 rtp_header.sequenceNumber = seq_no_++; in SendData() 55 rtp_header.payloadType = payload_type; in SendData() 56 rtp_header.timestamp = timestamp; in SendData() 64 receiver_acm_->IncomingPacket(payload_data, payload_size, rtp_header); in SendData()
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/external/webrtc/api/ |
D | rtp_packet_info.cc | 35 RtpPacketInfo::RtpPacketInfo(const RTPHeader& rtp_header, in RtpPacketInfo() argument 37 : ssrc_(rtp_header.ssrc), in RtpPacketInfo() 38 rtp_timestamp_(rtp_header.timestamp), in RtpPacketInfo() 40 const auto& extension = rtp_header.extension; in RtpPacketInfo() 41 const auto csrcs_count = std::min<size_t>(rtp_header.numCSRCs, kRtpCsrcSize); in RtpPacketInfo() 43 csrcs_.assign(&rtp_header.arrOfCSRCs[0], &rtp_header.arrOfCSRCs[csrcs_count]); in RtpPacketInfo()
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/external/webrtc/modules/video_coding/ |
D | packet.cc | 34 const RTPHeader& rtp_header, in VCMPacket() argument 38 : payloadType(rtp_header.payloadType), in VCMPacket() 39 timestamp(rtp_header.timestamp), in VCMPacket() 41 seqNum(rtp_header.sequenceNumber), in VCMPacket() 44 markerBit(rtp_header.markerBit), in VCMPacket() 50 packet_info(rtp_header, receive_time_ms) { in VCMPacket()
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D | video_coding_impl.cc | 94 const RTPHeader& rtp_header, in IncomingPacket() argument 96 return receiver_.IncomingPacket(incomingPayload, payloadLength, rtp_header, in IncomingPacket()
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D | video_receiver.cc | 270 const RTPHeader& rtp_header, in IncomingPacket() argument 275 rtp_header.sequenceNumber); in IncomingPacket() 284 const VCMPacket packet(incomingPayload, payloadLength, rtp_header, in IncomingPacket()
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/external/webrtc/test/fuzzers/ |
D | rtp_header_parser_fuzzer.cc | 25 RTPHeader rtp_header; in FuzzOneInput() local 30 rtp_header_parser->Parse(data, size, &rtp_header); in FuzzOneInput() 37 rtp_header_parser->Parse(data, size, &rtp_header); in FuzzOneInput() 45 rtp_header_parser->Parse(data, size, &rtp_header); in FuzzOneInput()
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D | rtp_header_fuzzer.cc | 48 RTPHeader rtp_header; in FuzzOneInput() local 50 rtp_parser.Parse(&rtp_header, &extensions); in FuzzOneInput()
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/external/webrtc/modules/audio_coding/acm2/ |
D | acm_receiver_unittest.cc | 412 RTPHeader rtp_header; in TEST_F() local 418 rtp_header.sequenceNumber = 0xABCD; in TEST_F() 419 rtp_header.timestamp = 0xABCDEF01; in TEST_F() 420 rtp_header.payloadType = kPayloadType; in TEST_F() 421 rtp_header.markerBit = false; in TEST_F() 422 rtp_header.ssrc = 0x1234; in TEST_F() 423 rtp_header.numCSRCs = 0; in TEST_F() 424 rtp_header.payload_type_frequency = kSampleRateHz; in TEST_F() 428 ASSERT_EQ(0, receiver_->InsertPacket(rtp_header, kPayload)); in TEST_F() 429 ++rtp_header.sequenceNumber; in TEST_F() [all …]
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D | acm_receiver.cc | 100 int AcmReceiver::InsertPacket(const RTPHeader& rtp_header, in InsertPacket() argument 103 neteq_->InsertEmptyPacket(rtp_header); in InsertPacket() 107 int payload_type = rtp_header.payloadType; in InsertPacket() 136 if (neteq_->InsertPacket(rtp_header, incoming_payload) < 0) { in InsertPacket() 138 << static_cast<int>(rtp_header.payloadType) in InsertPacket()
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D | audio_coding_module_unittest.cc | 81 void Populate(RTPHeader* rtp_header) { in Populate() argument 82 rtp_header->sequenceNumber = 0xABCD; in Populate() 83 rtp_header->timestamp = 0xABCDEF01; in Populate() 84 rtp_header->payloadType = payload_type_; in Populate() 85 rtp_header->markerBit = false; in Populate() 86 rtp_header->ssrc = 0x1234; in Populate() 87 rtp_header->numCSRCs = 0; in Populate() 89 rtp_header->payload_type_frequency = kSampleRateHz; in Populate() 92 void Forward(RTPHeader* rtp_header) { in Forward() argument 93 ++rtp_header->sequenceNumber; in Forward() [all …]
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/external/adhd/cras/src/server/ |
D | cras_a2dp_info.c | 79 sizeof(struct rtp_header) + sizeof(struct rtp_payload); in init_a2dp() 110 sizeof(struct rtp_header) + sizeof(struct rtp_payload); in a2dp_reset() 119 struct rtp_header *header; in avdtp_write() 122 header = (struct rtp_header *)a2dp->a2dp_buf; in avdtp_write()
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/external/adhd/cras/src/common/ |
D | rtp.h | 26 struct rtp_header { struct 51 struct rtp_header { argument
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/external/webrtc/modules/rtp_rtcp/source/ |
D | rtp_sender_unittest.cc | 1011 webrtc::RTPHeader rtp_header; in TEST_P() local 1012 transport_.last_sent_packet().GetHeader(&rtp_header); in TEST_P() 1015 EXPECT_EQ(kStoredTimeInMs * 90, rtp_header.extension.transmissionTimeOffset); in TEST_P() 1018 EXPECT_EQ(expected_send_time, rtp_header.extension.absoluteSendTime); in TEST_P() 1075 webrtc::RTPHeader rtp_header; in TEST_P() local 1076 transport_.last_sent_packet().GetHeader(&rtp_header); in TEST_P() 1079 EXPECT_EQ(kStoredTimeInMs * 90, rtp_header.extension.transmissionTimeOffset); in TEST_P() 1082 EXPECT_EQ(expected_send_time, rtp_header.extension.absoluteSendTime); in TEST_P() 1108 webrtc::RTPHeader rtp_header; in TEST_P() local 1151 transport_.last_sent_packet().GetHeader(&rtp_header); in TEST_P() [all …]
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/external/webrtc/api/neteq/ |
D | neteq.h | 200 virtual int InsertPacket(const RTPHeader& rtp_header, 207 virtual void InsertEmptyPacket(const RTPHeader& rtp_header) = 0;
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/external/webrtc/test/ |
D | rtp_file_reader.cc | 297 uint8_t pt = packets_[packet_indices[0]].rtp_header.payloadType; in Initialize() 360 RTPHeader rtp_header; member 438 rtp_parser.ParseRtcp(&marker.rtp_header); in ReadPacket() 441 if (!rtp_parser.Parse(&marker.rtp_header, nullptr)) { in ReadPacket() 446 uint32_t ssrc = marker.rtp_header.ssrc; in ReadPacket()
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