| /external/webrtc/webrtc/modules/audio_coding/test/ |
| D | RTPFile.cc | 61 const uint8_t* payloadData, size_t payloadSize, in RTPPacket() argument
66 payloadSize(payloadSize), in RTPPacket()
68 if (payloadSize > 0) { in RTPPacket()
69 this->payloadData = new uint8_t[payloadSize]; in RTPPacket()
70 memcpy(this->payloadData, payloadData, payloadSize); in RTPPacket()
88 const size_t payloadSize, uint32_t frequency) { in Write() argument
90 payloadSize, frequency); in Write()
97 size_t payloadSize, uint32_t* offset) { in Read() argument
107 if (packet->payloadSize > 0 && payloadSize >= packet->payloadSize) { in Read()
108 memcpy(payloadData, packet->payloadData, packet->payloadSize); in Read()
[all …]
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| D | 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; variable
72 const size_t payloadSize,
77 size_t payloadSize,
109 const size_t payloadSize,
114 size_t payloadSize,
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| D | Channel.cc | 26 size_t payloadSize, in SendData() argument
30 size_t payloadDataSize = payloadSize; in SendData()
104 CalcStatistics(rtpInfo, payloadSize); in SendData()
130 void Channel::CalcStatistics(WebRtcRTPHeader& rtpInfo, size_t payloadSize) { in CalcStatistics() argument
201 currentPayloadStr->lastPayloadLenByte = payloadSize; in CalcStatistics()
204 currentPayloadStr->lastPayloadLenByte = payloadSize; in CalcStatistics()
217 _payloadStats[n].lastPayloadLenByte = payloadSize; in CalcStatistics()
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| D | Channel.h | 57 size_t payloadSize,
97 void CalcStatistics(WebRtcRTPHeader& rtpInfo, size_t payloadSize);
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| D | EncodeDecodeTest.cc | 40 const size_t payloadSize, in SendData() argument
42 _rtpStream->Write(payloadType, timeStamp, _seqNo++, payloadData, payloadSize, in SendData()
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| D | EncodeDecodeTest.h | 36 const size_t payloadSize,
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| /external/grpc-grpc/src/csharp/Grpc.Microbenchmarks/ |
| D | PInvokeByteArrayBenchmark.cs | 41 public void Run(int threadCount, int iterations, int payloadSize) in Run() argument
43 …ayBenchmark: threads={0}, iterations={1}, payloadSize={2}", threadCount, iterations, payloadSize)); in Run()
44 … threadedBenchmark = new ThreadedBenchmark(threadCount, () => ThreadBody(iterations, payloadSize)); in Run()
48 private void ThreadBody(int iterations, int payloadSize) in ThreadBody() argument
50 var payload = new byte[payloadSize]; in ThreadBody()
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| D | SendMessageBenchmark.cs | 46 public void Run(int threadCount, int iterations, int payloadSize) in Run() argument
48 …geBenchmark: threads={0}, iterations={1}, payloadSize={2}", threadCount, iterations, payloadSize)); in Run()
49 … threadedBenchmark = new ThreadedBenchmark(threadCount, () => ThreadBody(iterations, payloadSize)); in Run()
53 private void ThreadBody(int iterations, int payloadSize) in ThreadBody() argument
60 var payload = new byte[payloadSize]; in ThreadBody()
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| /external/webrtc/webrtc/modules/rtp_rtcp/source/ |
| D | rtp_sender_audio.cc | 159 size_t payloadSize = dataSize; in SendAudio() local
251 if (payloadSize == 0 || payloadData == NULL) { in SendAudio()
283 if (maxPayloadLength < (rtpHeaderLength + payloadSize)) { in SendAudio()
321 payloadSize = fragmentation->fragmentationLength[0] + in SendAudio()
330 payloadSize = fragmentation->fragmentationLength[0]; in SendAudio()
340 payloadSize = fragmentation->fragmentationLength[0]; in SendAudio()
342 memcpy(dataBuffer + rtpHeaderLength, payloadData, payloadSize); in SendAudio()
350 size_t packetSize = payloadSize + rtpHeaderLength; in SendAudio()
360 return _rtpSender->SendToNetwork(dataBuffer, payloadSize, rtpHeaderLength, in SendAudio()
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| D | rtp_sender_video.cc | 231 const size_t payloadSize, in SendVideo() argument
234 if (payloadSize == 0) { in SendVideo()
261 size_t payload_bytes_to_send = payloadSize; in SendVideo()
304 size_t packetSize = payloadSize + rtp_header_length; in SendVideo()
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| /external/libchrome/mojo/public/js/lib/ |
| D | control_message_proxy.js | 14 var payloadSize =
16 var builder = new internal.MessageV0Builder(messageName, payloadSize);
63 var payloadSize = mojo.interfaceControl.RunMessageParams.encodedSize;
66 payloadSize, internal.kMessageExpectsResponse, 0);
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| D | pipe_control_message_proxy.js | 14 var payloadSize =
17 var builder = new internal.MessageV0Builder(messageName, payloadSize);
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| D | control_message_handler.js | 72 var payloadSize =
76 payloadSize, internal.kMessageIsResponse, requestID);
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| D | codec.js | 573 function MessageV0Builder(messageName, payloadSize) { argument
576 var numberOfBytes = kMessageV0HeaderSize + payloadSize;
610 function MessageV1Builder(messageName, payloadSize, flags, argument
614 var numberOfBytes = kMessageV1HeaderSize + payloadSize;
635 function MessageV2Builder(messageName, payloadSize, flags, requestID) { argument
638 var numberOfBytes = kMessageV2HeaderSize + payloadSize;
693 this.payloadSize = message.buffer.byteLength - messageHeaderSize;
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| /external/mp4parser/isoparser/src/main/java/com/googlecode/mp4parser/authoring/tracks/ |
| D | H264TrackImpl.java | 564 int payloadSize = 0; field in H264TrackImpl.SEIMessage
594 payloadSize = 0; in SEIMessage()
607 payloadSize += last_payload_size_bytes; in SEIMessage()
611 payloadSize += last_payload_size_bytes; in SEIMessage()
612 if (datasize - read >= payloadSize) { in SEIMessage()
615 byte[] data = new byte[payloadSize]; in SEIMessage()
617 read += payloadSize; in SEIMessage()
689 for (int i = 0; i < payloadSize; i++) { in SEIMessage()
695 for (int i = 0; i < payloadSize; i++) { in SEIMessage()
711 ", payloadSize=" + payloadSize; in toString()
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| /external/webrtc/webrtc/modules/utility/source/ |
| D | coder.cc | 105 size_t payloadSize, in SendData() argument
108 memcpy(_encodedData,payloadData,sizeof(uint8_t) * payloadSize); in SendData()
109 _encodedLengthInBytes = payloadSize; in SendData()
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| D | coder.h | 45 size_t payloadSize,
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| /external/webrtc/webrtc/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ |
| D | ReleaseTest-API.cc | 68 int16_t payloadSize = 0; in main() local
261 payloadSize = atoi(argv[i + 1]); in main()
262 printf("Maximum Payload Size: %d\n", payloadSize); in main()
529 if (payloadSize != 0) { in main()
530 err = WebRtcIsac_SetMaxPayloadSize(ISAC_main_inst, payloadSize); in main()
596 if ((payloadSize != 0) && (stream_len_int > payloadSize)) { in main()
602 stream_len_int - payloadSize); in main()
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| /external/mp4parser/isoparser/src/main/java/com/googlecode/mp4parser/h264/read/ |
| D | CAVLCReader.java | 106 public byte[] read(int payloadSize) throws IOException { in read() argument
107 byte[] result = new byte[payloadSize]; in read()
108 for (int i = 0; i < payloadSize; i++) { in read()
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| /external/grpc-grpc-java/alts/src/main/java/io/grpc/alts/internal/ |
| D | AltsTsiFrameProtector.java | 318 int payloadSize = frameSize - HEADER_TYPE_FIELD_BYTES - suffixBytes; in handlePayload() local
327 ciphertextsAndTags.add(lastBuf.readSlice(payloadSize + suffixBytes)); in handlePayload()
329 requiredUnprotectedBytesCompleteFrames += payloadSize; in handlePayload()
330 unprotectedLens.add(payloadSize); in handlePayload()
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| /external/webrtc/webrtc/modules/audio_coding/codecs/isac/fix/test/ |
| D | kenny.cc | 123 int16_t payloadSize = 0; in main() local
279 payloadSize = atoi(argv[i + 1]); in main()
280 printf("Maximum Payload Size: %d\n", payloadSize); in main()
510 if (payloadSize != 0) { in main()
511 err = WebRtcIsacfix_SetMaxPayloadSize(ISAC_main_inst, payloadSize); in main()
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| /external/webrtc/webrtc/modules/rtp_rtcp/test/testAPI/ |
| D | test_api_audio.cc | 30 const size_t payloadSize, in OnReceivedPayloadData() argument
34 EXPECT_EQ(4u, payloadSize); in OnReceivedPayloadData()
36 memcpy(str, payloadData, payloadSize); in OnReceivedPayloadData()
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| /external/grpc-grpc/src/objective-c/tests/ |
| D | InteropTests.m | 44 + (instancetype)messageWithPayloadSize:(NSNumber *)payloadSize
49 + (instancetype)messageWithPayloadSize:(NSNumber *)payloadSize
55 request.payload.body = [NSMutableData dataWithLength:payloadSize.unsignedIntegerValue];
61 + (instancetype)messageWithPayloadSize:(NSNumber *)payloadSize;
65 + (instancetype)messageWithPayloadSize:(NSNumber *)payloadSize {
68 response.payload.body = [NSMutableData dataWithLength:payloadSize.unsignedIntegerValue];
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| /external/webrtc/webrtc/modules/rtp_rtcp/include/ |
| D | rtp_rtcp_defines.h | 194 const size_t payloadSize,
351 const size_t payloadSize, in OnReceivedPayloadData() argument
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| /external/webrtc/webrtc/modules/rtp_rtcp/mocks/ |
| D | mock_rtp_rtcp.h | 31 const size_t payloadSize,
129 const size_t payloadSize,
|