| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
| D | rtp_utility.cc | 132 void AssignUWord32ToBuffer(uint8_t* dataBuffer, uint32_t value) { in AssignUWord32ToBuffer() argument
134 dataBuffer[0] = static_cast<uint8_t>(value >> 24); in AssignUWord32ToBuffer()
135 dataBuffer[1] = static_cast<uint8_t>(value >> 16); in AssignUWord32ToBuffer()
136 dataBuffer[2] = static_cast<uint8_t>(value >> 8); in AssignUWord32ToBuffer()
137 dataBuffer[3] = static_cast<uint8_t>(value); in AssignUWord32ToBuffer()
139 uint32_t* ptr = reinterpret_cast<uint32_t*>(dataBuffer); in AssignUWord32ToBuffer()
144 void AssignUWord24ToBuffer(uint8_t* dataBuffer, uint32_t value) { in AssignUWord24ToBuffer() argument
146 dataBuffer[0] = static_cast<uint8_t>(value >> 16); in AssignUWord24ToBuffer()
147 dataBuffer[1] = static_cast<uint8_t>(value >> 8); in AssignUWord24ToBuffer()
148 dataBuffer[2] = static_cast<uint8_t>(value); in AssignUWord24ToBuffer()
[all …]
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| D | rtp_sender_audio.cc | 344 uint8_t dataBuffer[IP_PACKET_SIZE]; in SendAudio() local
355 rtpHeaderLength = _rtpSender->BuildRTPheader(dataBuffer, _REDPayloadType, in SendAudio()
361 rtpHeaderLength = _rtpSender->BuildRTPheader(dataBuffer, payloadType, in SendAudio()
384 dataBuffer[rtpHeaderLength++] = 0x80 + in SendAudio()
393 RtpUtility::AssignUWord24ToBuffer(dataBuffer + rtpHeaderLength, in SendAudio()
397 dataBuffer[rtpHeaderLength++] = fragmentation->fragmentationPlType[0]; in SendAudio()
399 memcpy(dataBuffer+rtpHeaderLength, in SendAudio()
404 memcpy(dataBuffer+rtpHeaderLength + in SendAudio()
414 dataBuffer[rtpHeaderLength++] = fragmentation->fragmentationPlType[0]; in SendAudio()
415 memcpy(dataBuffer+rtpHeaderLength, in SendAudio()
[all …]
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| D | rtp_utility.h | 69 void AssignUWord32ToBuffer(uint8_t* dataBuffer, uint32_t value);
70 void AssignUWord24ToBuffer(uint8_t* dataBuffer, uint32_t value);
71 void AssignUWord16ToBuffer(uint8_t* dataBuffer, uint16_t value);
78 uint16_t BufferToUWord16(const uint8_t* dataBuffer);
85 uint32_t BufferToUWord24(const uint8_t* dataBuffer);
92 uint32_t BufferToUWord32(const uint8_t* dataBuffer);
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| D | rtp_sender_video.cc | 344 uint8_t dataBuffer[IP_PACKET_SIZE] = {0}; in Send() local
347 &dataBuffer[rtp_header_length], &payload_bytes_in_packet, &last)) { in Send()
354 dataBuffer, payloadType, last, captureTimeStamp, capture_time_ms); in Send()
355 if (SendVideoPacket(dataBuffer, in Send()
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| D | rtp_sender_video.h | 87 virtual int32_t SendVideoPacket(uint8_t* dataBuffer,
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| D | rtcp_sender.h | 181 int32_t SendToNetwork(const uint8_t* dataBuffer, const uint16_t length);
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| D | rtcp_sender.cc | 2011 RTCPSender::SendToNetwork(const uint8_t* dataBuffer, in SendToNetwork() argument
2017 if(_cbTransport->SendRTCPPacket(_id, dataBuffer, length) > 0) in SendToNetwork()
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| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/test/testAPI/ |
| D | test_api_video.cc | 81 int32_t BuildRTPheader(uint8_t* dataBuffer, in BuildRTPheader() argument
84 dataBuffer[0] = static_cast<uint8_t>(0x80); // version 2 in BuildRTPheader()
85 dataBuffer[1] = static_cast<uint8_t>(kPayloadType); in BuildRTPheader()
86 RtpUtility::AssignUWord16ToBuffer(dataBuffer + 2, sequence_number); in BuildRTPheader()
87 RtpUtility::AssignUWord32ToBuffer(dataBuffer + 4, timestamp); in BuildRTPheader()
88 RtpUtility::AssignUWord32ToBuffer(dataBuffer + 8, 0x1234); // SSRC. in BuildRTPheader()
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| /external/chromium_org/third_party/libjingle/source/talk/app/webrtc/objc/ |
| D | RTCDataChannel.mm | 50 RTCDataBuffer* dataBuffer =
52 [_channel.delegate channel:_channel didReceiveMessageWithBuffer:dataBuffer];
170 - (const webrtc::DataBuffer*)dataBuffer { method in Internal
251 return _dataChannel->Send(*data.dataBuffer);
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| D | RTCDataChannel+Internal.h | 35 @property(nonatomic, readonly) const webrtc::DataBuffer* dataBuffer;
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| /external/chromium_org/third_party/WebKit/Source/modules/crypto/ |
| D | SubtleCrypto.cpp | 90 …* key, WebCryptoOperation operationType, const ArrayPiece& signature, const ArrayPiece& dataBuffer) in startCryptoOperation() argument
104 if (!ensureNotNull(dataBuffer, "dataBuffer", result.get())) in startCryptoOperation()
114 const unsigned char* data = dataBuffer.bytes(); in startCryptoOperation()
115 unsigned dataSize = dataBuffer.byteLength(); in startCryptoOperation()
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| /external/aac/libMpegTPEnc/src/ |
| D | tpenc_lib.cpp | 576 FDK_BITSTREAM *dataBuffer, in transportEnc_GetConf() argument
590 tpErr = CreateStreamMuxConfig(hLatmConfig, dataBuffer, 0, &hTpEnc->callbacks); in transportEnc_GetConf()
594 if (transportEnc_writeASC(dataBuffer, cc, &hTpEnc->callbacks) != 0) { in transportEnc_GetConf()
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| /external/aac/libMpegTPEnc/include/ |
| D | tpenc_lib.h | 286 FDK_BITSTREAM *dataBuffer,
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| /external/chromium_org/content/common/gpu/media/ |
| D | vt.sig | 6 OSStatus CMSampleBufferCreate(CFAllocatorRef allocator, CMBlockBufferRef dataBuffer, Boolean dataRe…
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| /external/chromium_org/third_party/webrtc/modules/video_coding/main/source/ |
| D | session_info.cc | 26 uint16_t BufferToUWord16(const uint8_t* dataBuffer) { in BufferToUWord16() argument
27 return (dataBuffer[0] << 8) | dataBuffer[1]; in BufferToUWord16()
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| /external/chromium_org/third_party/webrtc/modules/audio_device/test/ |
| D | func_test_manager.h | 65 uint8_t dataBuffer[4 * 960]; member
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| D | func_test_manager.cc | 158 memcpy(packet->dataBuffer, audioSamples, nSamples * nBytesPerSample); in RecordedDataIsAvailable()
342 (const int16_t*) packet->dataBuffer, in NeedMorePlayData()
349 (const int16_t*) packet->dataBuffer, in NeedMorePlayData()
384 (const int16_t*) packet->dataBuffer, in NeedMorePlayData()
391 (const int16_t*) packet->dataBuffer, in NeedMorePlayData()
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| /external/chromium_org/third_party/webrtc/modules/media_file/source/ |
| D | media_file_impl.h | 160 int32_t PlayoutData(int8_t* dataBuffer, uint32_t& dataLengthInBytes,
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| /external/aac/libAACenc/src/ |
| D | bitenc.cpp | 738 UCHAR *dataBuffer, in FDKaacEnc_writeDataStreamElement() argument
803 FDKwriteBits(hBitStream, dataBuffer[i], 8); in FDKaacEnc_writeDataStreamElement()
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