| /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,
|
| /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,
|
| /system/core/liblog/ |
| pmsg_writer.c | 105 size_t i, payloadSize;
157 for (payloadSize = 0, i = headerLength; i < nr + headerLength; i++) {
159 payloadSize += newVec[i].iov_len = vec[i - headerLength].iov_len;
161 if (payloadSize > LOGGER_ENTRY_MAX_PAYLOAD) {
162 newVec[i].iov_len -= payloadSize - LOGGER_ENTRY_MAX_PAYLOAD;
166 payloadSize = LOGGER_ENTRY_MAX_PAYLOAD;
170 pmsgHeader.len += payloadSize;
|
| logd_writer.c | 134 size_t i, payloadSize;
226 for (payloadSize = 0, i = headerLength; i < nr + headerLength; i++) {
228 payloadSize += newVec[i].iov_len = vec[i - headerLength].iov_len;
230 if (payloadSize > LOGGER_ENTRY_MAX_PAYLOAD) {
231 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;
|
| rtp_sender_video.h | 53 const size_t payloadSize,
|
| /device/google/cuttlefish_common/host/commands/record_audio/ |
| main.cc | 90 const size_t payloadSize = res - sizeof(gce_audio_message);
105 << payloadSize
131 writer->Append(&buffer[sizeof(gce_audio_message)], payloadSize);
|
| /packages/apps/Car/SystemUpdater/src/com/android/car/systemupdater/ |
| UpdateParser.java | 55 long payloadSize = 0;
74 payloadSize = fileSize;
91 return new ParsedUpdate(file, payloadOffset, payloadSize, props);
|
| /tools/apksig/src/main/java/com/android/apksig/internal/apk/v3/ |
| V3SigningCertificateLineage.java | 198 int payloadSize = 4 + encodedSigningCertificateLineage.length;
199 ByteBuffer encodedWithVersion = ByteBuffer.allocate(payloadSize);
224 int payloadSize = prefixedSignedData.length + 4 + 4 + prefixedSignature.length;
225 ByteBuffer result = ByteBuffer.allocate(payloadSize);
237 int payloadSize = 4 + prefixedCertificate.length;
238 ByteBuffer result = ByteBuffer.allocate(payloadSize);
|
| V3SchemeSigner.java | 238 int payloadSize =
245 ByteBuffer result = ByteBuffer.allocate(payloadSize);
279 int payloadSize =
286 ByteBuffer result = ByteBuffer.allocate(payloadSize);
|
| /system/core/libstats/ |
| statsd_writer.c | 140 size_t i, payloadSize;
199 for (payloadSize = 0, i = headerLength; i < nr + headerLength; i++) {
201 payloadSize += newVec[i].iov_len = vec[i - headerLength].iov_len;
203 if (payloadSize > LOGGER_ENTRY_MAX_PAYLOAD) {
204 newVec[i].iov_len -= payloadSize - LOGGER_ENTRY_MAX_PAYLOAD;
|
| /frameworks/av/media/libstagefright/mpeg2ts/ |
| ESQueue.cpp | 181 unsigned payloadSize = frameSizeTable[frmsizecod >> 1][fscod];
183 payloadSize += frmsizecod & 1;
185 payloadSize <<= 1; // convert from 16-bit words to bytes
195 return payloadSize;
671 unsigned payloadSize = 0;
681 payloadSize = parseAC3SyncFrame(
685 if (payloadSize > 0) {
689 ALOGV("dequeueAccessUnit_AC3[%d]: syncStartPos %u payloadSize %u",
690 mAUIndex, syncStartPos, payloadSize);
695 if (mBuffer->size() < 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);
|
| /tools/apksig/src/main/java/com/android/apksig/internal/apk/v2/ |
| V2SchemeSigner.java | 267 int payloadSize = 4 + 4 + 4;
268 ByteBuffer result = ByteBuffer.allocate(payloadSize);
270 result.putInt(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,
|
