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  /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/
rtp_sender_audio.h 73 const bool markerBit); // set on first packet in talk burst
75 bool MarkerBit(const FrameType frameType,
103 // VAD detection, used for markerbit
rtp_sender_audio.cc 131 RTPSenderAudio::MarkerBit(const FrameType frameType,
137 bool markerBit = false;
196 markerBit = true;
207 markerBit = true;
209 return markerBit;
345 bool markerBit = MarkerBit(frameType, payloadType);
350 if (_REDPayloadType >= 0 && fragmentation && !markerBit &&
356 markerBit, captureTimeStamp,
362 markerBit, captureTimeStamp
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