/external/chromium_org/third_party/webrtc/modules/audio_coding/main/test/ |
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; 70 const uint16_t payloadSize, uint32_t frequency); 73 uint16_t payloadSize, uint32_t* offset); 102 const uint16_t payloadSize, uint32_t frequency); 105 uint16_t payloadSize, uint32_t* offset);
|
RTPFile.cc | 62 const uint8_t* payloadData, uint16_t payloadSize, 67 payloadSize(payloadSize), 69 if (payloadSize > 0) { 70 this->payloadData = new uint8_t[payloadSize]; 71 memcpy(this->payloadData, payloadData, payloadSize); 89 const uint16_t payloadSize, uint32_t frequency) { 91 payloadSize, frequency); 98 uint16_t payloadSize, uint32_t* offset) { 108 if (packet->payloadSize > 0 && payloadSize >= packet->payloadSize) [all...] |
Channel.h | 55 const uint16_t payloadSize, 95 void CalcStatistics(WebRtcRTPHeader& rtpInfo, uint16_t payloadSize);
|
/external/chromium_org/third_party/webrtc/modules/utility/source/ |
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;
|
coder.cc | 112 uint16_t payloadSize, 115 memcpy(_encodedData,payloadData,sizeof(uint8_t) * payloadSize); 116 _encodedLengthInBytes = payloadSize;
|
coder.h | 49 uint16_t payloadSize,
|
video_coder.h | 56 uint32_t payloadSize,
|
/external/smack/src/org/xbill/DNS/ |
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) { 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) { 77 this(payloadSize, xrcode, version, flags, null); 85 OPTRecord(int payloadSize, int xrcode, int version) { 86 this(payloadSize, xrcode, version, 0, null) [all...] |
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);
|
SimpleResolver.java | 141 setEDNS(int level, int payloadSize, int flags, List options) { 145 if (payloadSize == 0) 146 payloadSize = DEFAULT_EDNS_PAYLOADSIZE; 147 queryOPT = new OPTRecord(payloadSize, 0, level, flags, options);
|
/external/chromium_org/chrome/renderer/resources/extensions/ |
gcm_custom_bindings.js | 17 var payloadSize = 0; 30 payloadSize += property.length + value.length; 33 if (payloadSize > gcm.MAX_MESSAGE_SIZE) 35 + payloadSize); 37 if (payloadSize == 0)
|
/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/test/testAPI/ |
test_api.h | 91 const uint16_t payloadSize, 93 EXPECT_LE(payloadSize, sizeof(_payloadData)); 94 memcpy(_payloadData, payloadData, payloadSize); 96 _payloadSize = payloadSize;
|
/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/H264/ |
rtp_sender_h264.cc | 408 if (ptrH264Info->payloadSize[idxNALU] > maxPayloadLengthSTAP_A) 422 payloadBytesToSend -= ptrH264Info->payloadSize[idxNALU] + ptrH264Info->startCodeSize[idxNALU]; 423 data += ptrH264Info->payloadSize[idxNALU] + ptrH264Info->startCodeSize[idxNALU]; 428 if(ptrH264Info->payloadSize[idxNALU] + payloadBytesInPacket <= maxPayloadLengthSTAP_A) 435 dataBuffer[dataOffset] = (uint8_t)(ptrH264Info->payloadSize[idxNALU] >> 8); 437 dataBuffer[dataOffset] = (uint8_t)(ptrH264Info->payloadSize[idxNALU] & 0xff); 440 memcpy(&dataBuffer[dataOffset], &data[ptrH264Info->startCodeSize[idxNALU]], ptrH264Info->payloadSize[idxNALU]); 441 dataOffset += ptrH264Info->payloadSize[idxNALU]; 442 data += ptrH264Info->payloadSize[idxNALU] + ptrH264Info->startCodeSize[idxNALU]; 443 payloadBytesInPacket += (uint16_t)(ptrH264Info->payloadSize[idxNALU] + H264_NALU_LENGTH) [all...] |
h264_information.h | 116 memset(payloadSize, 0, sizeof(payloadSize)); 124 uint32_t payloadSize[KMaxNumberOfNALUs];
|
h264_information.cc | 50 memset(_info.payloadSize, 0, sizeof(_info.payloadSize)); 237 _ptrData += (_info.startCodeSize[_info.numNALUs] + _info.payloadSize[_info.numNALUs]); 238 _remLength -= (_info.startCodeSize[_info.numNALUs] + _info.payloadSize[_info.numNALUs]); 286 * - _info.payloadSize[currentNALU] : Payload length in bytes of NAL unit 316 _info.payloadSize[_info.numNALUs] = size - _info.startCodeSize[_info.numNALUs]; 317 _parsedLength += _info.startCodeSize[_info.numNALUs] + _info.payloadSize[_info.numNALUs]; 323 _info.payloadSize[_info.numNALUs] = _remLength - _info.startCodeSize[_info.numNALUs]; 324 if (_info.payloadSize[_info.numNALUs] > 0) 326 _parsedLength += _info.startCodeSize[_info.numNALUs] + _info.payloadSize[_info.numNALUs] [all...] |
/external/chromium_org/third_party/webrtc/modules/video_coding/main/test/ |
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,
|
/frameworks/av/media/libstagefright/codecs/on2/h264dec/source/ |
h264bsd_sei.c | 97 static u32 DecodeFillerPayload(strmData_t *pStrmData, u32 payloadSize); 102 u32 payloadSize); 107 u32 payloadSize); 163 u32 payloadSize); 187 u32 tmp, payloadType, payloadSize, status; 208 payloadSize = 0; 211 payloadSize += 255; 215 payloadSize += tmp; 257 status = DecodeFillerPayload(pStrmData, payloadSize); 264 payloadSize); [all...] |
/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
rtp_sender_audio.cc | 239 uint16_t payloadSize = static_cast<uint16_t>(dataSize); 334 if (payloadSize == 0 || payloadData == NULL) { 366 if (maxPayloadLength < (rtpHeaderLength + payloadSize)) { 407 payloadSize = static_cast<uint16_t>( 417 payloadSize = static_cast<uint16_t>( 428 payloadSize = static_cast<uint16_t>( 431 memcpy(dataBuffer+rtpHeaderLength, payloadData, payloadSize); 438 uint16_t packetSize = payloadSize + rtpHeaderLength; 451 payloadSize,
|
rtp_sender_video.cc | 283 const uint32_t payloadSize, 288 if( payloadSize == 0) 310 capture_time_ms, payloadData, payloadSize); 318 payloadSize, 414 const uint32_t payloadSize, 420 int32_t payloadBytesToSend = payloadSize;
|
rtp_sender_video.h | 54 const uint32_t payloadSize, 110 const uint32_t payloadSize,
|
/frameworks/av/media/libstagefright/mpeg2ts/ |
ESQueue.cpp | 155 unsigned payloadSize = frameSizeTable[frmsizecod >> 1][fscod]; 157 payloadSize += frmsizecod & 1; 159 payloadSize <<= 1; // convert from 16-bit words to bytes 169 return payloadSize; 491 unsigned payloadSize = 0; 498 payloadSize = parseAC3SyncFrame( 502 if (payloadSize > 0) { 508 if (mBuffer->size() < syncStartPos + payloadSize) { 517 sp<ABuffer> accessUnit = new ABuffer(syncStartPos + payloadSize); 518 memcpy(accessUnit->data(), mBuffer->data(), syncStartPos + payloadSize); [all...] |
/external/chromium_org/third_party/webrtc/modules/interface/ |
module_common_types.h | 306 payloadSize(0), 320 payloadSize = data.payloadSize; 324 if (data.payloadSize > 0) { 325 payloadData = new uint8_t[data.payloadSize]; 326 memcpy(payloadData, data.payloadData, data.payloadSize); 347 payloadSize = data.payloadSize; 351 if (data.payloadSize > 0) { 353 payloadData = new uint8_t[data.payloadSize]; [all...] |
/external/webrtc/src/modules/interface/ |
module_common_types.h | 309 payloadSize(0), 322 payloadSize = data.payloadSize; 326 if (data.payloadSize > 0) 328 payloadData = new WebRtc_UWord8[data.payloadSize]; 329 memcpy(payloadData, data.payloadData, data.payloadSize); 356 payloadSize = data.payloadSize; 360 if (data.payloadSize > 0) 363 payloadData = new WebRtc_UWord8[data.payloadSize]; [all...] |
/external/mp4parser/isoparser/src/main/java/com/googlecode/mp4parser/h264/read/ |
CAVLCReader.java | 106 public byte[] read(int payloadSize) throws IOException { 107 byte[] result = new byte[payloadSize]; 108 for (int i = 0; i < payloadSize; i++) {
|
/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/test/BWEStandAlone/ |
TestSenderReceiver.h | 100 const uint16_t payloadSize, 128 const uint32_t payloadSize,
|