| /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) OVERRIDE;
73 uint16_t payloadSize, uint32_t* offset) OVERRIDE;
102 const uint16_t payloadSize, uint32_t frequency) OVERRIDE;
105 uint16_t payloadSize, uint32_t* offset) OVERRIDE;
|
| RTPFile.cc | 63 const uint8_t* payloadData, uint16_t payloadSize,
68 payloadSize(payloadSize),
70 if (payloadSize > 0) {
71 this->payloadData = new uint8_t[payloadSize];
72 memcpy(this->payloadData, payloadData, payloadSize);
90 const uint16_t payloadSize, uint32_t frequency) {
92 payloadSize, frequency);
99 uint16_t payloadSize, uint32_t* offset) {
109 if (packet->payloadSize > 0 && payloadSize >= packet->payloadSize)
[all...] |
| Channel.h | 56 const uint16_t payloadSize,
96 void CalcStatistics(WebRtcRTPHeader& rtpInfo, uint16_t payloadSize);
|
| Channel.cc | 23 const uint16_t payloadSize,
27 uint16_t payloadDataSize = payloadSize;
95 //fwrite(payloadData, sizeof(uint8_t), payloadSize, _bitStreamFile);
99 CalcStatistics(rtpInfo, payloadSize);
124 void Channel::CalcStatistics(WebRtcRTPHeader& rtpInfo, uint16_t payloadSize) {
186 currentPayloadStr->lastPayloadLenByte = payloadSize;
189 currentPayloadStr->lastPayloadLenByte = payloadSize;
202 _payloadStats[n].lastPayloadLenByte = payloadSize;
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| EncodeDecodeTest.h | 35 const 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;
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| coder.h | 50 uint16_t payloadSize,
|
| video_coder.h | 56 uint32_t payloadSize,
|
| /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 | 93 const uint16_t payloadSize,
95 EXPECT_LE(payloadSize, sizeof(_payloadData));
96 memcpy(_payloadData, payloadData, payloadSize);
98 _payloadSize = payloadSize;
|
| /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,
|
| normal_test.cc | 79 const uint32_t payloadSize,
87 if (fwrite(payloadData, 1, payloadSize, _encodedFile) != payloadSize) {
117 _encodedBytes += payloadSize;
118 if (payloadSize < 20)
122 _VCMReceiver->IncomingPacket(payloadData, payloadSize, rtpInfo);
|
| test_callbacks.h | 52 const uint32_t payloadSize,
109 const uint32_t payloadSize,
|
| generic_codec_test.h | 99 const uint32_t payloadSize,
|
| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
| rtp_sender_audio.cc | 239 uint16_t payloadSize = static_cast<uint16_t>(dataSize);
336 if (payloadSize == 0 || payloadData == NULL) {
368 if (maxPayloadLength < (rtpHeaderLength + payloadSize)) {
409 payloadSize = static_cast<uint16_t>(
419 payloadSize = static_cast<uint16_t>(
430 payloadSize = static_cast<uint16_t>(
433 memcpy(dataBuffer+rtpHeaderLength, payloadData, payloadSize);
440 uint16_t packetSize = payloadSize + rtpHeaderLength;
453 payloadSize,
|
| rtp_sender_video.h | 51 const uint32_t payloadSize,
102 const uint32_t payloadSize,
|
| rtp_sender_video.cc | 274 const uint32_t payloadSize,
278 if (payloadSize == 0) {
298 payloadSize,
323 const uint32_t payloadSize,
327 int32_t payload_bytes_to_send = payloadSize;
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| rtp_sender_audio.h | 40 const uint32_t payloadSize,
|
| /external/chromium_org/third_party/webrtc/modules/interface/ |
| module_common_types.h | 298 payloadSize(0),
312 payloadSize = data.payloadSize;
316 if (data.payloadSize > 0) {
317 payloadData = new uint8_t[data.payloadSize];
318 memcpy(payloadData, data.payloadData, data.payloadSize);
339 payloadSize = data.payloadSize;
343 if (data.payloadSize > 0) {
345 payloadData = new uint8_t[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 | 103 const uint16_t payloadSize,
131 const uint32_t payloadSize,
|
| /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/chromium_org/third_party/webrtc/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ |
| ReleaseTest-API.cc | 70 int16_t payloadSize = 0;
263 payloadSize = atoi(argv[i + 1]);
264 printf("Maximum Payload Size: %d\n", payloadSize);
582 if(payloadSize != 0)
584 err = WebRtcIsac_SetMaxPayloadSize(ISAC_main_inst, payloadSize);
667 if((payloadSize != 0) && (stream_len > payloadSize))
674 printf("\nError: Streamsize out of range %d\n", stream_len - payloadSize);
[all...] |
| /external/chromium_org/third_party/webrtc/modules/video_coding/main/interface/ |
| video_coding_defines.h | 75 uint32_t payloadSize,
|
