| /external/webrtc/webrtc/modules/audio_coding/test/ |
| 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;
72 const size_t payloadSize,
77 size_t payloadSize,
109 const size_t payloadSize,
114 size_t payloadSize,
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| 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 > 0 && payloadSize >= packet->payloadSize)
[all...] |
| Channel.h | 57 size_t payloadSize,
97 void CalcStatistics(WebRtcRTPHeader& rtpInfo, size_t payloadSize);
|
| 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) {
201 currentPayloadStr->lastPayloadLenByte = payloadSize;
204 currentPayloadStr->lastPayloadLenByte = payloadSize;
217 _payloadStats[n].lastPayloadLenByte = payloadSize;
|
| EncodeDecodeTest.h | 36 const size_t payloadSize,
|
| EncodeDecodeTest.cc | 40 const size_t payloadSize,
42 _rtpStream->Write(payloadType, timeStamp, _seqNo++, payloadData, payloadSize,
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| /external/webrtc/webrtc/modules/utility/source/ |
| coder.cc | 105 size_t payloadSize,
108 memcpy(_encodedData,payloadData,sizeof(uint8_t) * payloadSize);
109 _encodedLengthInBytes = payloadSize;
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| coder.h | 45 size_t 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...] |
| /system/core/liblog/ |
| pmsg_writer.c | 124 size_t i, payloadSize;
176 for (payloadSize = 0, i = headerLength; i < nr + headerLength; i++) {
178 payloadSize += newVec[i].iov_len = vec[i - headerLength].iov_len;
180 if (payloadSize > LOGGER_ENTRY_MAX_PAYLOAD) {
181 newVec[i].iov_len -= payloadSize - LOGGER_ENTRY_MAX_PAYLOAD;
185 payloadSize = LOGGER_ENTRY_MAX_PAYLOAD;
189 pmsgHeader.len += payloadSize;
|
| logd_writer.c | 124 size_t i, payloadSize;
211 for (payloadSize = 0, i = headerLength; i < nr + headerLength; i++) {
213 payloadSize += newVec[i].iov_len = vec[i - headerLength].iov_len;
215 if (payloadSize > LOGGER_ENTRY_MAX_PAYLOAD) {
216 newVec[i].iov_len -= payloadSize - LOGGER_ENTRY_MAX_PAYLOAD;
|
| /external/webrtc/webrtc/modules/rtp_rtcp/source/ |
| rtp_sender_audio.cc | 159 size_t payloadSize = dataSize;
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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| rtp_sender_audio.h | 40 size_t payloadSize,
|
| 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;
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| rtp_sender_video.h | 53 const size_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;
518 unsigned payloadSize = 0;
525 payloadSize = parseAC3SyncFrame(
529 if (payloadSize > 0) {
535 if (mBuffer->size() < syncStartPos + payloadSize) {
544 sp<ABuffer> accessUnit = new ABuffer(syncStartPos + payloadSize);
545 memcpy(accessUnit->data(), mBuffer->data(), syncStartPos + 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/mp4parser/isoparser/src/main/java/com/googlecode/mp4parser/authoring/tracks/ |
| H264TrackImpl.java | 564 int payloadSize = 0;
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=" + payloadSize
[all...] |
| /external/webrtc/webrtc/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ |
| ReleaseTest-API.cc | 68 int16_t payloadSize = 0;
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);
|
| /external/webrtc/webrtc/modules/rtp_rtcp/test/testAPI/ |
| test_api_audio.cc | 30 const size_t payloadSize,
34 EXPECT_EQ(4u, payloadSize);
36 memcpy(str, payloadData, payloadSize);
|
| /external/webrtc/webrtc/modules/rtp_rtcp/include/ |
| rtp_rtcp_defines.h | 194 const size_t payloadSize,
351 const size_t payloadSize,
|
| /external/webrtc/webrtc/modules/audio_coding/codecs/isac/fix/test/ |
| kenny.cc | 123 int16_t payloadSize = 0;
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);
|
| /hardware/intel/common/libmix/mix_vbp/viddec_fw/fw/codecs/h264/parser/ |
| h264parse_sei.c | [all...] |
| /external/webrtc/webrtc/modules/rtp_rtcp/mocks/ |
| mock_rtp_rtcp.h | 31 const size_t payloadSize,
129 const size_t payloadSize,
|
| /build/tools/signapk/src/com/android/signapk/ |
| ApkSignerV2.java | 579 int payloadSize = 0;
581 payloadSize += 4 + element.length;
583 ByteBuffer result = ByteBuffer.allocate(payloadSize);
[all...] |
