| /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);
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| 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;
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| 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,
|
