/external/chromium_org/third_party/webrtc/modules/audio_coding/main/test/ |
D | Channel.cc | 21 int32_t Channel::SendData(const FrameType frameType, const uint8_t payloadType, in SendData() argument 33 rtpInfo.header.payloadType = payloadType; in SendData() 76 rtpInfo.header.payloadType = fragmentation->fragmentationPlType[0]; in SendData() 126 if ((rtpInfo.header.payloadType != _lastPayloadType) in CalcStatistics() 133 if (_lastPayloadType == _payloadStats[n].payloadType) { in CalcStatistics() 139 _lastPayloadType = rtpInfo.header.payloadType; in CalcStatistics() 144 if (rtpInfo.header.payloadType == _payloadStats[n].payloadType) { in CalcStatistics() 191 currentPayloadStr->payloadType = rtpInfo.header.payloadType; in CalcStatistics() 197 while (_payloadStats[n].payloadType != -1) { in CalcStatistics() 204 _payloadStats[n].payloadType = rtpInfo.header.payloadType; in CalcStatistics() [all …]
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D | RTPFile.cc | 31 rtpInfo->header.payloadType = rtpHeader[1]; in ParseRTPHeader() 42 void RTPStream::MakeRTPheader(uint8_t* rtpHeader, uint8_t payloadType, in MakeRTPheader() argument 46 rtpHeader[1] = (unsigned char) (payloadType & 0xFF); in MakeRTPheader() 62 RTPPacket::RTPPacket(uint8_t payloadType, uint32_t timeStamp, int16_t seqNo, in RTPPacket() argument 65 : payloadType(payloadType), in RTPPacket() 88 void RTPBuffer::Write(const uint8_t payloadType, const uint32_t timeStamp, in Write() argument 91 RTPPacket *packet = new RTPPacket(payloadType, timeStamp, seqNo, payloadData, in Write() 105 rtpInfo->header.payloadType = packet->payloadType; in Read() 181 void RTPFile::Write(const uint8_t payloadType, const uint32_t timeStamp, in Write() argument 186 MakeRTPheader(rtpHeader, payloadType, seqNo, timeStamp, 0); in Write()
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D | RTPFile.h | 29 virtual void Write(const uint8_t payloadType, const uint32_t timeStamp, 40 void MakeRTPheader(uint8_t* rtpHeader, uint8_t payloadType, int16_t seqNo, 48 RTPPacket(uint8_t payloadType, uint32_t timeStamp, int16_t seqNo, 54 uint8_t payloadType; variable 68 virtual void Write(const uint8_t payloadType, const uint32_t timeStamp, 100 virtual void Write(const uint8_t payloadType, const uint32_t timeStamp,
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D | Channel.h | 41 int16_t payloadType; member 54 const FrameType frameType, const uint8_t payloadType, 67 void Stats(uint8_t* payloadLenByte, uint32_t* payloadType);
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/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
D | rtp_sender_audio.cc | 88 const int8_t payloadType, in RegisterAudioPayload() argument 98 _cngNBPayloadType = payloadType; in RegisterAudioPayload() 101 _cngWBPayloadType = payloadType; in RegisterAudioPayload() 104 _cngSWBPayloadType = payloadType; in RegisterAudioPayload() 107 _cngFBPayloadType = payloadType; in RegisterAudioPayload() 116 _dtmfPayloadType = payloadType; in RegisterAudioPayload() 132 const int8_t payloadType) in MarkerBit() argument 138 if(_lastPayloadType != payloadType) in MarkerBit() 143 if(_cngNBPayloadType == payloadType) in MarkerBit() 152 if(_cngWBPayloadType == payloadType) in MarkerBit() [all …]
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D | rtp_sender_audio.h | 30 const int8_t payloadType, 37 const int8_t payloadType, 62 int32_t SetRED(const int8_t payloadType); 65 int32_t RED(int8_t& payloadType) const; 76 const int8_t payloadType);
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D | rtp_sender_video.h | 41 const int8_t payloadType, 47 const int8_t payloadType, 98 const int8_t payloadType,
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D | rtp_receiver_impl.cc | 291 if (rtp_header.payloadType == last_received_payload_type) { in CheckSSRCChanged() 296 rtp_header.payloadType, payload)) { in CheckSSRCChanged() 320 id_, rtp_header.payloadType, payload_name, in CheckSSRCChanged() 324 << rtp_header.payloadType; in CheckSSRCChanged() 345 int8_t payload_type = rtp_header.payloadType; in CheckPayloadChanged() 438 rtp_header.header.payloadType)) { in CheckCSRC()
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D | rtp_sender_video.cc | 78 const int8_t payloadType, in RegisterVideoPayload() argument 270 const int8_t payloadType, in SendVideo() argument 294 payloadType, in SendVideo() 319 const int8_t payloadType, in Send() argument 354 dataBuffer, payloadType, last, captureTimeStamp, capture_time_ms); in Send()
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/external/chromium_org/third_party/webrtc/modules/audio_coding/main/acm2/ |
D | initial_delay_manager.cc | 28 last_packet_rtp_info_.header.payloadType = kInvalidPayloadType; in InitialDelayManager() 45 rtp_info.header.payloadType != audio_payload_type_)); in UpdateLastReceivedPacket() 65 last_packet_rtp_info_.header.payloadType == kInvalidPayloadType) { in UpdateLastReceivedPacket() 68 audio_payload_type_ = rtp_info.header.payloadType; in UpdateLastReceivedPacket() 131 sync_stream->rtp_info.header.payloadType = audio_payload_type_; in UpdateLastReceivedPacket() 145 sync_stream->rtp_info.header.payloadType = audio_payload_type_; in UpdateLastReceivedPacket() 207 sync_stream->rtp_info.header.payloadType = audio_payload_type_; in LatePackets() 215 last_packet_rtp_info_.header.payloadType = audio_payload_type_; in LatePackets()
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D | initial_delay_manager_unittest.cc | 35 rtp_info->header.payloadType = kAudioPayloadType; in InitRtpInfo() 99 rtp_info_.header.payloadType = kCngPayloadType; in TEST_F() 105 rtp_info_.header.payloadType = kAvtPayloadType; in TEST_F() 119 rtp_info_.header.payloadType = kAudioPayloadType; in TEST_F() 212 rtp_info_.header.payloadType = kCngPayloadType; in TEST_F() 219 rtp_info_.header.payloadType = kAudioPayloadType; in TEST_F() 336 rtp_info_.header.payloadType = kCngPayloadType; in TEST_F()
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/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/ |
D | packet_buffer.cc | 86 if (decoder_database.IsComfortNoise(packet->header.payloadType)) { in InsertPacketList() 88 *current_cng_rtp_payload_type != packet->header.payloadType) { in InsertPacketList() 94 *current_cng_rtp_payload_type = packet->header.payloadType; in InsertPacketList() 95 } else if (!decoder_database.IsDtmf(packet->header.payloadType)) { in InsertPacketList() 98 *current_rtp_payload_type != packet->header.payloadType) { in InsertPacketList() 103 *current_rtp_payload_type = packet->header.payloadType; in InsertPacketList() 215 decoder_database->GetDecoder(packet->header.payloadType); in NumSamplesInBuffer()
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D | neteq_impl.cc | 124 ", pt=" << static_cast<int>(rtp_header.header.payloadType) << in InsertPacket() 143 ", pt=" << static_cast<int>(rtp_header.header.payloadType) << in InsertSyncPacket() 407 if (decoder_database_->IsDtmf(rtp_header.header.payloadType) || in InsertPacketInternal() 408 decoder_database_->IsRed(rtp_header.header.payloadType) || in InsertPacketInternal() 409 decoder_database_->IsComfortNoise(rtp_header.header.payloadType)) { in InsertPacketInternal() 411 << rtp_header.header.payloadType; in InsertPacketInternal() 415 rtp_header.header.payloadType != current_rtp_payload_type_ || in InsertPacketInternal() 433 packet->header.payloadType = rtp_header.header.payloadType; in InsertPacketInternal() 472 current_rtp_payload_type_ = main_header.payloadType; in InsertPacketInternal() 489 if (decoder_database_->IsRed(main_header.payloadType)) { in InsertPacketInternal() [all …]
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/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/test/ |
D | NETEQTEST_RTPpacket.cc | 159 if (!_blockList.empty() && _blockList.count(payloadType()) > 0) in readFromFile() 199 if (!_blockList.empty() && _blockList.count(payloadType()) > 0) in readFixedFromFile() 289 rtp_header->header.payloadType = _rtpInfo.header.payloadType; in parseHeader() 353 uint8_t NETEQTEST_RTPpacket::payloadType() const in payloadType() function in NETEQTEST_RTPpacket 366 return tempRTPinfo.header.payloadType; in payloadType() 445 _rtpInfo.header.payloadType = pt; in setPayloadType() 553 RTPinfo->header.payloadType, in setRTPheader() 627 void NETEQTEST_RTPpacket::makeRTPheader(unsigned char* rtp_data, uint8_t payloadType, uint16_t seqN… in makeRTPheader() argument 638 rtp_data[1]=(unsigned char)(payloadType & 0xFF); in makeRTPheader() 693 RTPinfo->header.payloadType = (uint8_t) ((rtp_data[0] >> 8) & 0x7F); in parseBasicHeader() [all …]
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/external/chromium_org/third_party/webrtc/modules/video_coding/main/source/ |
D | packet.cc | 20 payloadType(0), in VCMPacket() 40 payloadType(rtpHeader.header.payloadType), in VCMPacket() 61 payloadType(0), in VCMPacket() 80 payloadType = 0; in Reset()
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/external/chromium_org/third_party/webrtc/modules/video_coding/main/test/ |
D | test_callbacks.cc | 56 const uint8_t payloadType, in SendData() argument 90 rtpInfo.header.payloadType = payloadType; in SendData() 148 const uint8_t payloadType, in SendData() argument 160 payloadType, in SendData() 306 header.payloadType, &payload_specific)) { in SendPacket()
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/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/tools/ |
D | neteq_rtpplay.cc | 499 if (IsComfortNosie(rtp_header->header.payloadType)) { in ReplacePayload() 524 if (CodecTimestampRate(rtp_header->header.payloadType) != in ReplacePayload() 525 CodecSampleRate(rtp_header->header.payloadType) || in ReplacePayload() 526 rtp_header->header.payloadType == FLAGS_red || in ReplacePayload() 527 rtp_header->header.payloadType == FLAGS_avt) { in ReplacePayload() 549 switch (CodecSampleRate(rtp_header->header.payloadType)) { in ReplacePayload() 551 rtp_header->header.payloadType = FLAGS_pcm16b; in ReplacePayload() 554 rtp_header->header.payloadType = FLAGS_pcm16b_wb; in ReplacePayload() 557 rtp_header->header.payloadType = FLAGS_pcm16b_swb32; in ReplacePayload() 560 rtp_header->header.payloadType = FLAGS_pcm16b_swb48; in ReplacePayload() [all …]
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D | packet.cc | 99 header->payloadType = payload_ptr[0] & 0x7F; in ExtractRedHeaders() 112 header->payloadType = payload_ptr[0] & 0x7F; in ExtractRedHeaders() 151 destination->payloadType = header_.payloadType; in CopyToHeader()
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D | rtp_analyze.cc | 108 packet->header().payloadType, in main() 120 if (packet->header().payloadType == FLAGS_red) { in main() 131 red->payloadType); in main()
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D | packet_unittest.cc | 56 EXPECT_EQ(kPayloadType, packet.header().payloadType); in TEST() 87 EXPECT_EQ(kPayloadType, packet.header().payloadType); in TEST() 164 EXPECT_EQ(kRedPayloadType, packet.header().payloadType); in TEST() 186 EXPECT_EQ(block_index, red_block->payloadType); in TEST()
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/external/chromium_org/third_party/webrtc/modules/utility/source/ |
D | video_coder.cc | 112 const uint8_t payloadType, in SendData() argument 124 _videoEncodedData->payloadType = payloadType; in SendData()
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/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/test/testAPI/ |
D | test_api_audio.cc | 32 if (rtpHeader->header.payloadType == 98 || in OnReceivedPayloadData() 33 rtpHeader->header.payloadType == 99) { in OnReceivedPayloadData() 45 if (rtpHeader->header.payloadType == 100 || in OnReceivedPayloadData() 46 rtpHeader->header.payloadType == 101 || in OnReceivedPayloadData() 47 rtpHeader->header.payloadType == 102) { in OnReceivedPayloadData() 66 const int8_t payloadType, in OnInitializeDecoder() argument 71 if (payloadType == 96) { in OnInitializeDecoder()
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/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/interface/ |
D | rtp_rtcp.h | 168 const int8_t payloadType) = 0; 276 int* payloadType) const = 0; 330 const int8_t payloadType, 656 const int8_t payloadType) = 0; 664 int8_t& payloadType) const = 0;
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/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/test/BWEStandAlone/ |
D | TestSenderReceiver.cc | 110 const int8_t payloadType /*= 127*/) in InitReceiver() argument 145 if (_rtp->RegisterReceivePayload("I420", payloadType, 90000) != 0) in InitReceiver() 347 const int8_t payloadType /*= 127*/) in InitSender() argument 351 _payloadType = payloadType; in InitSender()
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D | TestSenderReceiver.h | 65 const int8_t payloadType = 127); 76 const int8_t payloadType, in OnInitializeDecoder() argument 127 const int8_t payloadType = 127);
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