/external/chromium_org/third_party/webrtc/modules/audio_coding/main/test/ |
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; 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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Channel.cc | 21 int32_t Channel::SendData(const FrameType frameType, const uint8_t payloadType, 33 rtpInfo.header.payloadType = payloadType; 76 rtpInfo.header.payloadType = fragmentation->fragmentationPlType[0]; 126 if ((rtpInfo.header.payloadType != _lastPayloadType) 133 if (_lastPayloadType == _payloadStats[n].payloadType) { 139 _lastPayloadType = rtpInfo.header.payloadType; 144 if (rtpInfo.header.payloadType == _payloadStats[n].payloadType) { 191 currentPayloadStr->payloadType = rtpInfo.header.payloadType [all...] |
RTPFile.cc | 31 rtpInfo->header.payloadType = rtpHeader[1]; 42 void RTPStream::MakeRTPheader(uint8_t* rtpHeader, uint8_t payloadType, 46 rtpHeader[1] = (unsigned char) (payloadType & 0xFF); 62 RTPPacket::RTPPacket(uint8_t payloadType, uint32_t timeStamp, int16_t seqNo, 65 : payloadType(payloadType), 88 void RTPBuffer::Write(const uint8_t payloadType, const uint32_t timeStamp, 91 RTPPacket *packet = new RTPPacket(payloadType, timeStamp, seqNo, payloadData, 105 rtpInfo->header.payloadType = packet->payloadType; [all...] |
Channel.h | 41 int16_t payloadType; 54 const FrameType frameType, const uint8_t payloadType, 67 void Stats(uint8_t* payloadLenByte, uint32_t* payloadType);
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EncodeDecodeTest.h | 33 const FrameType frameType, const uint8_t payloadType, 39 static void MakeRTPheader(uint8_t* rtpHeader, uint8_t payloadType,
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/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
rtp_sender_audio.cc | 88 const int8_t payloadType, 98 _cngNBPayloadType = payloadType; 101 _cngWBPayloadType = payloadType; 104 _cngSWBPayloadType = payloadType; 107 _cngFBPayloadType = payloadType; 115 // we dont want to allow send with a DTMF payloadtype 116 _dtmfPayloadType = payloadType; 132 const int8_t payloadType) 138 if(_lastPayloadType != payloadType) 143 if(_cngNBPayloadType == payloadType) [all...] |
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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rtp_sender_video.h | 41 const int8_t payloadType, 47 const int8_t payloadType, 98 const int8_t payloadType,
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/external/chromium_org/third_party/webrtc/modules/audio_coding/main/acm2/ |
initial_delay_manager.cc | 28 last_packet_rtp_info_.header.payloadType = kInvalidPayloadType; 45 rtp_info.header.payloadType != audio_payload_type_)); 65 last_packet_rtp_info_.header.payloadType == kInvalidPayloadType) { 68 audio_payload_type_ = rtp_info.header.payloadType; 131 sync_stream->rtp_info.header.payloadType = audio_payload_type_; 145 sync_stream->rtp_info.header.payloadType = audio_payload_type_; 207 sync_stream->rtp_info.header.payloadType = audio_payload_type_; 215 last_packet_rtp_info_.header.payloadType = audio_payload_type_;
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/external/chromium_org/third_party/webrtc/modules/video_coding/main/source/ |
packet.cc | 20 payloadType(0), 40 payloadType(rtpHeader.header.payloadType), 61 payloadType(0), 80 payloadType = 0;
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packet.h | 34 uint8_t payloadType;
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generic_encoder.h | 52 void SetPayloadType(uint8_t payloadType) { _payloadType = payloadType; };
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/external/chromium_org/third_party/webrtc/modules/utility/source/ |
video_coder.cc | 112 const uint8_t payloadType, 124 _videoEncodedData->payloadType = payloadType;
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coder.h | 47 uint8_t payloadType,
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coder.cc | 61 const uint8_t payloadType = _receiveCodec.pltype; 65 payloadType, 109 uint8_t /* payloadType */,
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/external/chromium_org/media/cast/net/rtp/ |
mock_rtp_feedback.h | 17 int32(const int8 payloadType,
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/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/mock/ |
mock_rtp_payload_strategy.h | 35 const int8_t payloadType,
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/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/ |
packet_buffer.cc | 86 if (decoder_database.IsComfortNoise(packet->header.payloadType)) { 88 *current_cng_rtp_payload_type != packet->header.payloadType) { 94 *current_cng_rtp_payload_type = packet->header.payloadType; 95 } else if (!decoder_database.IsDtmf(packet->header.payloadType)) { 98 *current_rtp_payload_type != packet->header.payloadType) { 103 *current_rtp_payload_type = packet->header.payloadType; 215 decoder_database->GetDecoder(packet->header.payloadType);
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neteq_impl.cc | 124 ", pt=" << static_cast<int>(rtp_header.header.payloadType) << 143 ", pt=" << static_cast<int>(rtp_header.header.payloadType) << 407 if (decoder_database_->IsDtmf(rtp_header.header.payloadType) || 408 decoder_database_->IsRed(rtp_header.header.payloadType) || 409 decoder_database_->IsComfortNoise(rtp_header.header.payloadType)) { 411 << rtp_header.header.payloadType; 415 rtp_header.header.payloadType != current_rtp_payload_type_ || 433 packet->header.payloadType = rtp_header.header.payloadType; 472 current_rtp_payload_type_ = main_header.payloadType; [all...] |
/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/tools/ |
rtp_analyze.cc | 108 packet->header().payloadType, 120 if (packet->header().payloadType == FLAGS_red) { 131 red->payloadType);
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packet.cc | 99 header->payloadType = payload_ptr[0] & 0x7F; 112 header->payloadType = payload_ptr[0] & 0x7F; 151 destination->payloadType = header_.payloadType;
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/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/test/BWEStandAlone/ |
TestSenderReceiver.h | 65 const int8_t payloadType = 127); 76 const int8_t payloadType, 127 const int8_t payloadType = 127);
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/external/chromium_org/third_party/webrtc/modules/video_coding/main/test/ |
test_callbacks.cc | 56 const uint8_t payloadType, 90 rtpInfo.header.payloadType = payloadType; 148 const uint8_t payloadType, 160 payloadType, 306 header.payloadType, &payload_specific)) {
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/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/test/ |
NETEQTEST_RTPpacket.cc | 159 if (!_blockList.empty() && _blockList.count(payloadType()) > 0) 199 if (!_blockList.empty() && _blockList.count(payloadType()) > 0) 289 rtp_header->header.payloadType = _rtpInfo.header.payloadType; 353 uint8_t NETEQTEST_RTPpacket::payloadType() const 366 return tempRTPinfo.header.payloadType; 445 _rtpInfo.header.payloadType = pt; 553 RTPinfo->header.payloadType, 627 void NETEQTEST_RTPpacket::makeRTPheader(unsigned char* rtp_data, uint8_t payloadType, uint16_t seqNo, uint32_t timestamp, uint32_t ssrc, uint8_t markerBit) const 638 rtp_data[1]=(unsigned char)(payloadType & 0xFF) [all...] |
/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/interface/ |
rtp_rtcp.h | 163 * payloadType - payload type of codec 168 const int8_t payloadType) = 0; 276 int* payloadType) const = 0; 319 * payloadType - payload type of frame to send 330 const int8_t payloadType, 656 const int8_t payloadType) = 0; 664 int8_t& payloadType) const = 0;
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