/external/chromium_org/third_party/webrtc/modules/audio_coding/main/test/ |
D | RTPFile.cc | 63 const uint8_t* payloadData, uint16_t payloadSize, in RTPPacket() 90 const uint16_t payloadSize, uint32_t frequency) { in Write() 99 uint16_t payloadSize, uint32_t* offset) { in Read() 183 const uint16_t payloadSize, uint32_t frequency) { in Write() 211 uint16_t payloadSize, uint32_t* offset) { in Read()
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D | RTPFile.h | 58 uint16_t payloadSize; variable
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D | Channel.cc | 23 const uint16_t payloadSize, in SendData() 124 void Channel::CalcStatistics(WebRtcRTPHeader& rtpInfo, uint16_t payloadSize) { in CalcStatistics()
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D | EncodeDecodeTest.cc | 40 const uint16_t payloadSize, in SendData()
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/external/smack/src/org/xbill/DNS/ |
D | OPTRecord.java | 51 OPTRecord(int payloadSize, int xrcode, int version, int flags, List options) { in OPTRecord() 76 OPTRecord(int payloadSize, int xrcode, int version, int flags) { in OPTRecord() 85 OPTRecord(int payloadSize, int xrcode, int version) { in OPTRecord()
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D | Resolver.java | 55 void setEDNS(int level, int payloadSize, int flags, List options); in setEDNS()
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D | ExtendedResolver.java | 324 setEDNS(int level, int payloadSize, int flags, List options) { in setEDNS()
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D | SimpleResolver.java | 141 setEDNS(int level, int payloadSize, int flags, List options) { in setEDNS()
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/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/test/BWEStandAlone/ |
D | TestSenderReceiver.cc | 310 const uint16_t payloadSize, in OnReceivedPayloadData() 406 const uint32_t payloadSize, in SendOutgoingData()
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D | TestLoadGenerator.cc | 134 const uint32_t payloadSize, in sendPayload()
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/external/chromium_org/third_party/webrtc/modules/video_coding/main/test/ |
D | test_callbacks.cc | 60 const uint32_t payloadSize, in SendData() 152 const uint32_t payloadSize, in SendData()
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D | normal_test.cc | 79 const uint32_t payloadSize, in SendData()
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D | generic_codec_test.cc | 541 const uint32_t payloadSize, in SendData()
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/external/chromium_org/third_party/webrtc/modules/utility/source/ |
D | coder.cc | 112 uint16_t payloadSize, in SendData()
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D | video_coder.cc | 116 uint32_t payloadSize, in SendData()
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/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/test/testAPI/ |
D | test_api.h | 93 const uint16_t payloadSize, in OnReceivedPayloadData()
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D | test_api_audio.cc | 30 const uint16_t payloadSize, in OnReceivedPayloadData()
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/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
D | rtp_sender_video.cc | 274 const uint32_t payloadSize, in SendVideo() 323 const uint32_t payloadSize, in Send()
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D | rtp_sender_audio.cc | 239 uint16_t payloadSize = static_cast<uint16_t>(dataSize); in SendAudio() local
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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()
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/external/chromium_org/third_party/WebKit/Source/modules/websockets/ |
D | DOMWebSocket.cpp | 352 void DOMWebSocket::updateBufferedAmountAfterClose(unsigned long payloadSize) in updateBufferedAmountAfterClose() 674 size_t DOMWebSocket::getFramingOverhead(size_t payloadSize) in getFramingOverhead()
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/external/chromium_org/third_party/WebKit/Source/platform/heap/ |
D | Heap.h | 245 size_t payloadSize() { return heapObjectHeader()->payloadSize(); } in payloadSize() function 519 static size_t payloadSize() in payloadSize() function 1404 size_t HeapObjectHeader::payloadSize() in payloadSize() function 1432 size_t FinalizedHeapObjectHeader::payloadSize() in payloadSize() function 1466 size_t payloadSize = allocationSize - sizeof(Header); in allocate() local
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/external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ |
D | ReleaseTest-API.cc | 70 int16_t payloadSize = 0; in main() local
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/external/webrtc/src/modules/audio_coding/codecs/isac/fix/test/ |
D | kenny.c | 118 WebRtc_Word16 payloadSize = 0; in main() local
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/external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/isac/fix/test/ |
D | kenny.cc | 119 int16_t payloadSize = 0; in main() local
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