/external/webrtc/webrtc/modules/audio_coding/test/ |
H A D | RTPFile.cc | 61 const uint8_t* payloadData, size_t payloadSize, 66 payloadSize(payloadSize), 68 if (payloadSize > 0) { 69 this->payloadData = new uint8_t[payloadSize]; 70 memcpy(this->payloadData, payloadData, payloadSize); 88 const size_t payloadSize, uint32_t frequency) { 90 payloadSize, frequency); 97 size_t payloadSize, uint32_t* offset) { 107 if (packet->payloadSize > 60 RTPPacket(uint8_t payloadType, uint32_t timeStamp, int16_t seqNo, const uint8_t* payloadData, size_t payloadSize, uint32_t frequency) argument 86 Write(const uint8_t payloadType, const uint32_t timeStamp, const int16_t seqNo, const uint8_t* payloadData, const size_t payloadSize, uint32_t frequency) argument 96 Read(WebRtcRTPHeader* rtpInfo, uint8_t* payloadData, size_t payloadSize, uint32_t* offset) argument 169 Write(const uint8_t payloadType, const uint32_t timeStamp, const int16_t seqNo, const uint8_t* payloadData, const size_t payloadSize, uint32_t frequency) argument 189 Read(WebRtcRTPHeader* rtpInfo, uint8_t* payloadData, size_t payloadSize, uint32_t* offset) argument [all...] |
H A D | RTPFile.h | 31 const size_t payloadSize, uint32_t frequency) = 0; 36 size_t payloadSize, uint32_t* offset) = 0; 49 const uint8_t* payloadData, size_t payloadSize, 58 size_t payloadSize; member in class:webrtc::RTPPacket 72 const size_t payloadSize, 77 size_t payloadSize, 109 const size_t payloadSize, 114 size_t payloadSize,
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H A D | Channel.h | 57 size_t payloadSize, 97 void CalcStatistics(WebRtcRTPHeader& rtpInfo, size_t payloadSize);
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H A D | Channel.cc | 26 size_t payloadSize, 30 size_t payloadDataSize = payloadSize; 100 //fwrite(payloadData, sizeof(uint8_t), payloadSize, _bitStreamFile); 104 CalcStatistics(rtpInfo, payloadSize); 130 void Channel::CalcStatistics(WebRtcRTPHeader& rtpInfo, size_t payloadSize) { argument 201 currentPayloadStr->lastPayloadLenByte = payloadSize; 204 currentPayloadStr->lastPayloadLenByte = payloadSize; 217 _payloadStats[n].lastPayloadLenByte = payloadSize; 22 SendData(FrameType frameType, uint8_t payloadType, uint32_t timeStamp, const uint8_t* payloadData, size_t payloadSize, const RTPFragmentationHeader* fragmentation) argument
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H A D | EncodeDecodeTest.h | 36 const size_t payloadSize,
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H A D | EncodeDecodeTest.cc | 40 const size_t payloadSize, 42 _rtpStream->Write(payloadType, timeStamp, _seqNo++, payloadData, payloadSize, 37 SendData( const FrameType , const uint8_t payloadType, const uint32_t timeStamp, const uint8_t* payloadData, const size_t payloadSize, const RTPFragmentationHeader* ) argument
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/external/webrtc/webrtc/modules/utility/source/ |
H A D | coder.cc | 105 size_t payloadSize, 108 memcpy(_encodedData,payloadData,sizeof(uint8_t) * payloadSize); 109 _encodedLengthInBytes = payloadSize; 100 SendData( FrameType , uint8_t , uint32_t , const uint8_t* payloadData, size_t payloadSize, const RTPFragmentationHeader* ) argument
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H A D | coder.h | 45 size_t payloadSize,
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/external/webrtc/webrtc/modules/rtp_rtcp/source/ |
H A D | rtp_sender_audio.cc | 159 size_t payloadSize = dataSize; local 251 if (payloadSize == 0 || payloadData == NULL) { 283 if (maxPayloadLength < (rtpHeaderLength + payloadSize)) { 321 payloadSize = fragmentation->fragmentationLength[0] + 330 payloadSize = fragmentation->fragmentationLength[0]; 340 payloadSize = fragmentation->fragmentationLength[0]; 342 memcpy(dataBuffer + rtpHeaderLength, payloadData, payloadSize); 350 size_t packetSize = payloadSize + rtpHeaderLength; 360 return _rtpSender->SendToNetwork(dataBuffer, payloadSize, rtpHeaderLength,
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H A D | rtp_sender_audio.h | 40 size_t payloadSize,
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H A D | rtp_sender_video.cc | 231 const size_t payloadSize, 234 if (payloadSize == 0) { 261 size_t payload_bytes_to_send = payloadSize; 304 size_t packetSize = payloadSize + rtp_header_length; 225 SendVideo(const RtpVideoCodecTypes videoType, const FrameType frameType, const int8_t payloadType, const uint32_t captureTimeStamp, int64_t capture_time_ms, const uint8_t* payloadData, const size_t payloadSize, const RTPFragmentationHeader* fragmentation, const RTPVideoHeader* rtpHdr) argument
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H A D | rtp_sender_video.h | 53 const size_t payloadSize,
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H A D | rtcp_receiver_unittest.cc | 61 const size_t payloadSize,
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/external/mp4parser/isoparser/src/main/java/com/googlecode/mp4parser/h264/read/ |
H A D | CAVLCReader.java | 106 public byte[] read(int payloadSize) throws IOException { argument 107 byte[] result = new byte[payloadSize]; 108 for (int i = 0; i < payloadSize; i++) {
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/external/mp4parser/isoparser/src/main/java/com/googlecode/mp4parser/authoring/tracks/ |
H A D | H264TrackImpl.java | 564 int payloadSize = 0; field in class:H264TrackImpl.SEIMessage 594 payloadSize = 0; 607 payloadSize += last_payload_size_bytes; 611 payloadSize += last_payload_size_bytes; 612 if (datasize - read >= payloadSize) { 615 byte[] data = new byte[payloadSize]; 617 read += payloadSize; 689 for (int i = 0; i < payloadSize; i++) { 695 for (int i = 0; i < payloadSize; i++) { 711 ", payloadSize [all...] |
/external/webrtc/webrtc/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ |
H A D | ReleaseTest-API.cc | 68 int16_t payloadSize = 0; local 261 payloadSize = atoi(argv[i + 1]); 262 printf("Maximum Payload Size: %d\n", payloadSize); 529 if (payloadSize != 0) { 530 err = WebRtcIsac_SetMaxPayloadSize(ISAC_main_inst, payloadSize); 596 if ((payloadSize != 0) && (stream_len_int > payloadSize)) { 602 stream_len_int - payloadSize);
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/external/webrtc/webrtc/modules/rtp_rtcp/test/testAPI/ |
H A D | test_api_audio.cc | 30 const size_t payloadSize, 34 EXPECT_EQ(4u, payloadSize); 36 memcpy(str, payloadData, payloadSize);
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/external/webrtc/webrtc/modules/audio_coding/codecs/isac/fix/test/ |
H A D | kenny.cc | 123 int16_t payloadSize = 0; local 279 payloadSize = atoi(argv[i + 1]); 280 printf("Maximum Payload Size: %d\n", payloadSize); 510 if (payloadSize != 0) { 511 err = WebRtcIsacfix_SetMaxPayloadSize(ISAC_main_inst, payloadSize);
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/external/webrtc/webrtc/modules/rtp_rtcp/include/ |
H A D | rtp_rtcp_defines.h | 194 const size_t payloadSize, 351 const size_t payloadSize,
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H A D | rtp_rtcp.h | 293 * payloadSize - size of payload buffer to send 305 size_t payloadSize,
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/external/webrtc/webrtc/modules/rtp_rtcp/mocks/ |
H A D | mock_rtp_rtcp.h | 31 const size_t payloadSize, 129 const size_t payloadSize,
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/external/protobuf/java/src/main/java/com/google/protobuf/nano/ |
H A D | Extension.java | 672 int payloadSize = 674 return payloadSize + CodedOutputByteBufferNano.computeRawVarint32Size(tag);
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/external/webrtc/webrtc/voice_engine/ |
H A D | channel.h | 365 size_t payloadSize, 375 size_t payloadSize,
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H A D | channel.cc | 252 size_t payloadSize, 257 " payloadSize=%" PRIuS ", fragmentation=0x%x)", 259 payloadSize, fragmentation); 280 payloadSize, 457 size_t payloadSize, 461 "Channel::OnReceivedPayloadData(payloadSize=%" PRIuS "," 463 payloadSize, 481 payloadSize, 248 SendData(FrameType frameType, uint8_t payloadType, uint32_t timeStamp, const uint8_t* payloadData, size_t payloadSize, const RTPFragmentationHeader* fragmentation) argument 456 OnReceivedPayloadData(const uint8_t* payloadData, size_t payloadSize, const WebRtcRTPHeader* rtpHeader) argument
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/external/robolectric/v3/libs/ |
H A D | sqlite4java-0.282.jar | META-INF/ META-INF/MANIFEST.MF com/ com/almworks/ com/almworks/sqlite4java/ javolution/ javolution/util/ javolution/ ... |