Searched refs:payloadSize (Results 1 - 25 of 27) sorted by relevance

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/external/webrtc/webrtc/modules/audio_coding/test/
H A DRTPFile.cc61 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 >
60 RTPPacket(uint8_t payloadType, uint32_t timeStamp, int16_t seqNo, const uint8_t* payloadData, size_t payloadSize, uint32_t frequency) argument
86 Write(const uint8_t payloadType, const uint32_t timeStamp, const int16_t seqNo, const uint8_t* payloadData, const size_t payloadSize, uint32_t frequency) argument
96 Read(WebRtcRTPHeader* rtpInfo, uint8_t* payloadData, size_t payloadSize, uint32_t* offset) argument
169 Write(const uint8_t payloadType, const uint32_t timeStamp, const int16_t seqNo, const uint8_t* payloadData, const size_t payloadSize, uint32_t frequency) argument
189 Read(WebRtcRTPHeader* rtpInfo, uint8_t* payloadData, size_t payloadSize, uint32_t* offset) argument
[all...]
H A DRTPFile.h31 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; member in class:webrtc::RTPPacket
72 const size_t payloadSize,
77 size_t payloadSize,
109 const size_t payloadSize,
114 size_t payloadSize,
H A DChannel.h57 size_t payloadSize,
97 void CalcStatistics(WebRtcRTPHeader& rtpInfo, size_t payloadSize);
H A DChannel.cc26 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) { argument
201 currentPayloadStr->lastPayloadLenByte = payloadSize;
204 currentPayloadStr->lastPayloadLenByte = payloadSize;
217 _payloadStats[n].lastPayloadLenByte = payloadSize;
22 SendData(FrameType frameType, uint8_t payloadType, uint32_t timeStamp, const uint8_t* payloadData, size_t payloadSize, const RTPFragmentationHeader* fragmentation) argument
H A DEncodeDecodeTest.h36 const size_t payloadSize,
H A DEncodeDecodeTest.cc40 const size_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 size_t payloadSize, const RTPFragmentationHeader* ) argument
/external/webrtc/webrtc/modules/utility/source/
H A Dcoder.cc105 size_t payloadSize,
108 memcpy(_encodedData,payloadData,sizeof(uint8_t) * payloadSize);
109 _encodedLengthInBytes = payloadSize;
100 SendData( FrameType , uint8_t , uint32_t , const uint8_t* payloadData, size_t payloadSize, const RTPFragmentationHeader* ) argument
H A Dcoder.h45 size_t payloadSize,
/external/webrtc/webrtc/modules/rtp_rtcp/source/
H A Drtp_sender_audio.cc159 size_t payloadSize = dataSize; local
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,
H A Drtp_sender_audio.h40 size_t payloadSize,
H A Drtp_sender_video.cc231 const size_t payloadSize,
234 if (payloadSize == 0) {
261 size_t payload_bytes_to_send = payloadSize;
304 size_t packetSize = payloadSize + rtp_header_length;
225 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 size_t payloadSize, const RTPFragmentationHeader* fragmentation, const RTPVideoHeader* rtpHdr) argument
H A Drtp_sender_video.h53 const size_t payloadSize,
H A Drtcp_receiver_unittest.cc61 const size_t payloadSize,
/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/mp4parser/isoparser/src/main/java/com/googlecode/mp4parser/authoring/tracks/
H A DH264TrackImpl.java564 int payloadSize = 0; field in class:H264TrackImpl.SEIMessage
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
[all...]
/external/webrtc/webrtc/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/
H A DReleaseTest-API.cc68 int16_t payloadSize = 0; local
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/
H A Dtest_api_audio.cc30 const size_t payloadSize,
34 EXPECT_EQ(4u, payloadSize);
36 memcpy(str, payloadData, payloadSize);
/external/webrtc/webrtc/modules/audio_coding/codecs/isac/fix/test/
H A Dkenny.cc123 int16_t payloadSize = 0; local
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);
/external/webrtc/webrtc/modules/rtp_rtcp/include/
H A Drtp_rtcp_defines.h194 const size_t payloadSize,
351 const size_t payloadSize,
H A Drtp_rtcp.h293 * payloadSize - size of payload buffer to send
305 size_t payloadSize,
/external/webrtc/webrtc/modules/rtp_rtcp/mocks/
H A Dmock_rtp_rtcp.h31 const size_t payloadSize,
129 const size_t payloadSize,
/external/protobuf/java/src/main/java/com/google/protobuf/nano/
H A DExtension.java672 int payloadSize =
674 return payloadSize + CodedOutputByteBufferNano.computeRawVarint32Size(tag);
/external/webrtc/webrtc/voice_engine/
H A Dchannel.h365 size_t payloadSize,
375 size_t payloadSize,
H A Dchannel.cc252 size_t payloadSize,
257 " payloadSize=%" PRIuS ", fragmentation=0x%x)",
259 payloadSize, fragmentation);
280 payloadSize,
457 size_t payloadSize,
461 "Channel::OnReceivedPayloadData(payloadSize=%" PRIuS ","
463 payloadSize,
481 payloadSize,
248 SendData(FrameType frameType, uint8_t payloadType, uint32_t timeStamp, const uint8_t* payloadData, size_t payloadSize, const RTPFragmentationHeader* fragmentation) argument
456 OnReceivedPayloadData(const uint8_t* payloadData, size_t payloadSize, const WebRtcRTPHeader* rtpHeader) argument
/external/robolectric/v3/libs/
H A Dsqlite4java-0.282.jarMETA-INF/ META-INF/MANIFEST.MF com/ com/almworks/ com/almworks/sqlite4java/ javolution/ javolution/util/ javolution/ ...

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