Searched defs:payloadSize (Results 1 - 25 of 38) sorted by relevance

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/external/chromium_org/third_party/webrtc/modules/audio_coding/main/test/
H A DRTPFile.h31 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;
H A DChannel.cc23 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
H A DEncodeDecodeTest.cc40 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
H A DRTPFile.cc63 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 Dcoder.cc112 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
H A Dvideo_coder.cc66 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
/external/smack/src/org/xbill/DNS/
H A DOPTRecord.java39 * @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 DResolver.java47 * @param payloadSize The maximum DNS packet size that this host is capable
55 void setEDNS(int level, int payloadSize, int flags, List options); argument
H A DExtendedResolver.java324 setEDNS(int level, int payloadSize, int flags, List options) { argument
326 ((Resolver)resolvers.get(i)).setEDNS(level, payloadSize,
H A DSimpleResolver.java141 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);
/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/test/testAPI/
H A Dtest_api.h93 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
H A Dtest_api_audio.cc30 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
/external/mp4parser/isoparser/src/main/java/com/googlecode/mp4parser/h264/read/
H A DCAVLCReader.java106 public byte[] read(int payloadSize) throws IOException { argument
107 byte[] result = new byte[payloadSize];
108 for (int i = 0; i < payloadSize; i++) {
/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/
H A Drtcp_sender_unittest.cc265 const uint16_t payloadSize,
264 OnReceivedPayloadData(const uint8_t* payloadData, const uint16_t payloadSize, const WebRtcRTPHeader* rtpHeader) argument
H A Drtp_sender_audio.cc239 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,
H A Drtp_sender_video.cc274 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
/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/test/BWEStandAlone/
H A DTestLoadGenerator.cc134 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
H A DTestSenderReceiver.cc310 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
/external/chromium_org/third_party/webrtc/modules/video_coding/main/test/
H A Dgeneric_codec_test.cc541 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
H A Dnormal_test.cc79 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
H A Dtest_callbacks.cc60 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
/external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/
H A DReleaseTest-API.cc70 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);
/external/chromium_org/third_party/webrtc/voice_engine/include/
H A Dvoe_rtp_rtcp.h278 const char* payloadData, unsigned short payloadSize) { return -1; };
276 InsertExtraRTPPacket( int channel, unsigned char payloadType, bool markerBit, const char* payloadData, unsigned short payloadSize) argument
/external/chromium_org/third_party/WebKit/Source/modules/websockets/
H A DDOMWebSocket.cpp352 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)
/external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/isac/fix/test/
H A Dkenny.cc119 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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