/external/chromium_org/third_party/webrtc/modules/audio_coding/main/test/ |
H A D | RTPFile.h | 31 const uint16_t payloadSize, uint32_t frequency) = 0; 36 uint16_t payloadSize, uint32_t* offset) = 0; 49 const uint8_t* payloadData, uint16_t payloadSize, 58 uint16_t payloadSize; member in class:webrtc::RTPPacket 70 const uint16_t payloadSize, uint32_t frequency) OVERRIDE; 73 uint16_t payloadSize, uint32_t* offset) OVERRIDE; 102 const uint16_t payloadSize, uint32_t frequency) OVERRIDE; 105 uint16_t payloadSize, uint32_t* offset) OVERRIDE;
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H A D | Channel.cc | 23 const uint16_t payloadSize, 27 uint16_t payloadDataSize = payloadSize; 95 //fwrite(payloadData, sizeof(uint8_t), payloadSize, _bitStreamFile); 99 CalcStatistics(rtpInfo, payloadSize); 124 void Channel::CalcStatistics(WebRtcRTPHeader& rtpInfo, uint16_t payloadSize) { argument 186 currentPayloadStr->lastPayloadLenByte = payloadSize; 189 currentPayloadStr->lastPayloadLenByte = payloadSize; 202 _payloadStats[n].lastPayloadLenByte = payloadSize; 21 SendData(const FrameType frameType, const uint8_t payloadType, const uint32_t timeStamp, const uint8_t* payloadData, const uint16_t payloadSize, const RTPFragmentationHeader* fragmentation) argument
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H A D | EncodeDecodeTest.cc | 40 const uint16_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 uint16_t payloadSize, const RTPFragmentationHeader* ) argument
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H A D | RTPFile.cc | 63 const uint8_t* payloadData, uint16_t payloadSize, 68 payloadSize(payloadSize), 70 if (payloadSize > 0) { 71 this->payloadData = new uint8_t[payloadSize]; 72 memcpy(this->payloadData, payloadData, payloadSize); 90 const uint16_t payloadSize, uint32_t frequency) { 92 payloadSize, frequency); 99 uint16_t payloadSize, uint32_t* offset) { 109 if (packet->payloadSize > 62 RTPPacket(uint8_t payloadType, uint32_t timeStamp, int16_t seqNo, const uint8_t* payloadData, uint16_t payloadSize, uint32_t frequency) argument 88 Write(const uint8_t payloadType, const uint32_t timeStamp, const int16_t seqNo, const uint8_t* payloadData, const uint16_t payloadSize, uint32_t frequency) argument 98 Read(WebRtcRTPHeader* rtpInfo, uint8_t* payloadData, uint16_t payloadSize, uint32_t* offset) argument 181 Write(const uint8_t payloadType, const uint32_t timeStamp, const int16_t seqNo, const uint8_t* payloadData, const uint16_t payloadSize, uint32_t frequency) argument 210 Read(WebRtcRTPHeader* rtpInfo, uint8_t* payloadData, uint16_t payloadSize, uint32_t* offset) argument [all...] |
/external/chromium_org/third_party/webrtc/modules/utility/source/ |
H A D | coder.cc | 112 uint16_t payloadSize, 115 memcpy(_encodedData,payloadData,sizeof(uint8_t) * payloadSize); 116 _encodedLengthInBytes = payloadSize; 107 SendData( FrameType , uint8_t , uint32_t , const uint8_t* payloadData, uint16_t payloadSize, const RTPFragmentationHeader* ) argument
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H A D | video_coder.cc | 66 if(encodedData.payloadSize <= 0) 82 videoEncodedData.payloadSize = 0; 116 uint32_t payloadSize, 122 _videoEncodedData->VerifyAndAllocate(payloadSize); 128 sizeof(uint8_t) * payloadSize); 129 _videoEncodedData->payloadSize = payloadSize; 110 SendData( const FrameType frameType, const uint8_t payloadType, const uint32_t timeStamp, int64_t capture_time_ms, const uint8_t* payloadData, uint32_t payloadSize, const RTPFragmentationHeader& fragmentationHeader, const RTPVideoHeader* ) argument
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/external/smack/src/org/xbill/DNS/ |
H A D | OPTRecord.java | 39 * @param payloadSize The size of a packet that can be reassembled on the 51 OPTRecord(int payloadSize, int xrcode, int version, int flags, List options) { argument 52 super(Name.root, Type.OPT, payloadSize, 0); 53 checkU16("payloadSize", payloadSize); 66 * @param payloadSize The size of a packet that can be reassembled on the 76 OPTRecord(int payloadSize, int xrcode, int version, int flags) { argument 77 this(payloadSize, xrcode, version, flags, null); 85 OPTRecord(int payloadSize, int xrcode, int version) { argument 86 this(payloadSize, xrcod [all...] |
H A D | Resolver.java | 47 * @param payloadSize The maximum DNS packet size that this host is capable 55 void setEDNS(int level, int payloadSize, int flags, List options); argument
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H A D | ExtendedResolver.java | 324 setEDNS(int level, int payloadSize, int flags, List options) { argument 326 ((Resolver)resolvers.get(i)).setEDNS(level, payloadSize,
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H A D | SimpleResolver.java | 141 setEDNS(int level, int payloadSize, int flags, List options) { argument 145 if (payloadSize == 0) 146 payloadSize = DEFAULT_EDNS_PAYLOADSIZE; 147 queryOPT = new OPTRecord(payloadSize, 0, level, flags, options);
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/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/test/testAPI/ |
H A D | test_api.h | 93 const uint16_t payloadSize, 95 EXPECT_LE(payloadSize, sizeof(_payloadData)); 96 memcpy(_payloadData, payloadData, payloadSize); 98 _payloadSize = payloadSize; 91 OnReceivedPayloadData( const uint8_t* payloadData, const uint16_t payloadSize, const webrtc::WebRtcRTPHeader* rtpHeader) argument
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H A D | test_api_audio.cc | 30 const uint16_t payloadSize, 34 EXPECT_EQ(4, payloadSize); 36 memcpy(str, payloadData, payloadSize); 28 OnReceivedPayloadData( const uint8_t* payloadData, const uint16_t payloadSize, const webrtc::WebRtcRTPHeader* rtpHeader) argument
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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/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
H A D | rtcp_sender_unittest.cc | 265 const uint16_t payloadSize, 264 OnReceivedPayloadData(const uint8_t* payloadData, const uint16_t payloadSize, const WebRtcRTPHeader* rtpHeader) argument
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H A D | rtp_sender_audio.cc | 239 uint16_t payloadSize = static_cast<uint16_t>(dataSize); local 336 if (payloadSize == 0 || payloadData == NULL) { 368 if (maxPayloadLength < (rtpHeaderLength + payloadSize)) { 409 payloadSize = static_cast<uint16_t>( 419 payloadSize = static_cast<uint16_t>( 430 payloadSize = static_cast<uint16_t>( 433 memcpy(dataBuffer+rtpHeaderLength, payloadData, payloadSize); 440 uint16_t packetSize = payloadSize + rtpHeaderLength; 453 payloadSize,
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H A D | rtp_sender_video.cc | 274 const uint32_t payloadSize, 278 if (payloadSize == 0) { 298 payloadSize, 323 const uint32_t payloadSize, 327 int32_t payload_bytes_to_send = payloadSize; 268 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 uint32_t payloadSize, const RTPFragmentationHeader* fragmentation, VideoCodecInformation* codecInfo, const RTPVideoTypeHeader* rtpTypeHdr) argument 317 Send(const RtpVideoCodecTypes videoType, const FrameType frameType, const int8_t payloadType, const uint32_t captureTimeStamp, int64_t capture_time_ms, const uint8_t* payloadData, const uint32_t payloadSize, const RTPFragmentationHeader* fragmentation, const RTPVideoTypeHeader* rtpTypeHdr) argument
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/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/test/BWEStandAlone/ |
H A D | TestLoadGenerator.cc | 134 const uint32_t payloadSize, 138 return (_sender->SendOutgoingData(timeStamp, payloadData, payloadSize, frameType)); 132 sendPayload(const uint32_t timeStamp, const uint8_t* payloadData, const uint32_t payloadSize, const webrtc::FrameType frameType ) argument
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H A D | TestSenderReceiver.cc | 310 const uint16_t payloadSize, 406 const uint32_t payloadSize, 409 return (_rtp->SendOutgoingData(frameType, _payloadType, timeStamp, payloadData, payloadSize)); 309 OnReceivedPayloadData(const uint8_t* payloadData, const uint16_t payloadSize, const webrtc::WebRtcRTPHeader* rtpHeader) argument 404 SendOutgoingData(const uint32_t timeStamp, const uint8_t* payloadData, const uint32_t payloadSize, const webrtc::FrameType frameType ) argument
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/external/chromium_org/third_party/webrtc/modules/video_coding/main/test/ |
H A D | generic_codec_test.cc | 541 const uint32_t payloadSize, 557 return _vcm.IncomingPacket(payloadData, payloadSize, rtpInfo); 535 SendData( const FrameType frameType, const uint8_t payloadType, const uint32_t timeStamp, int64_t capture_time_ms, const uint8_t* payloadData, const uint32_t payloadSize, const RTPFragmentationHeader& , const webrtc::RTPVideoHeader* ) argument
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H A D | normal_test.cc | 79 const uint32_t payloadSize, 87 if (fwrite(payloadData, 1, payloadSize, _encodedFile) != payloadSize) { 117 _encodedBytes += payloadSize; 118 if (payloadSize < 20) 122 _VCMReceiver->IncomingPacket(payloadData, payloadSize, rtpInfo); 73 SendData( const FrameType frameType, const uint8_t payloadType, const uint32_t timeStamp, int64_t capture_time_ms, const uint8_t* payloadData, const uint32_t payloadSize, const RTPFragmentationHeader& , const webrtc::RTPVideoHeader* videoHdr) argument
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H A D | test_callbacks.cc | 60 const uint32_t payloadSize, 67 if (fwrite(payloadData, 1, payloadSize, _encodedFile) != payloadSize) { 98 _encodedBytes += payloadSize; 100 int ret = _VCMReceiver->IncomingPacket(payloadData, payloadSize, rtpInfo); 152 const uint32_t payloadSize, 157 _encodedBytes+= payloadSize; 164 payloadSize, 54 SendData( const FrameType frameType, const uint8_t payloadType, const uint32_t timeStamp, int64_t capture_time_ms, const uint8_t* payloadData, const uint32_t payloadSize, const RTPFragmentationHeader& fragmentationHeader, const RTPVideoHeader* videoHdr) argument 146 SendData( const FrameType frameType, const uint8_t payloadType, const uint32_t timeStamp, int64_t capture_time_ms, const uint8_t* payloadData, const uint32_t payloadSize, const RTPFragmentationHeader& fragmentationHeader, const RTPVideoHeader* videoHdr) argument
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/external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ |
H A D | ReleaseTest-API.cc | 70 int16_t payloadSize = 0; local 263 payloadSize = atoi(argv[i + 1]); 264 printf("Maximum Payload Size: %d\n", payloadSize); 582 if(payloadSize != 0) 584 err = WebRtcIsac_SetMaxPayloadSize(ISAC_main_inst, payloadSize); 667 if((payloadSize != 0) && (stream_len > payloadSize)) 674 printf("\nError: Streamsize out of range %d\n", stream_len - payloadSize);
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/external/chromium_org/third_party/webrtc/voice_engine/include/ |
H A D | voe_rtp_rtcp.h | 278 const char* payloadData, unsigned short payloadSize) { return -1; }; 276 InsertExtraRTPPacket( int channel, unsigned char payloadType, bool markerBit, const char* payloadData, unsigned short payloadSize) argument
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/external/chromium_org/third_party/WebKit/Source/modules/websockets/ |
H A D | DOMWebSocket.cpp | 352 void DOMWebSocket::updateBufferedAmountAfterClose(unsigned long payloadSize) argument 354 m_bufferedAmountAfterClose = saturateAdd(m_bufferedAmountAfterClose, payloadSize); 355 m_bufferedAmountAfterClose = saturateAdd(m_bufferedAmountAfterClose, getFramingOverhead(payloadSize)); 674 size_t DOMWebSocket::getFramingOverhead(size_t payloadSize) argument 681 if (payloadSize >= minimumPayloadSizeWithEightByteExtendedPayloadLength) 683 else if (payloadSize >= minimumPayloadSizeWithTwoByteExtendedPayloadLength)
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/external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/isac/fix/test/ |
H A D | kenny.cc | 119 int16_t payloadSize = 0; local 275 payloadSize = atoi(argv[i + 1]); 276 printf("Maximum Payload Size: %d\n", payloadSize); 506 if (payloadSize != 0) { 507 err = WebRtcIsacfix_SetMaxPayloadSize(ISAC_main_inst, payloadSize);
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