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  /external/webrtc/webrtc/modules/rtp_rtcp/source/
rtp_sender_audio.cc 35 _dtmfTimestamp(0),
183 _dtmfTimestamp = captureTimeStamp;
211 uint32_t dtmfDurationSamples = captureTimeStamp - _dtmfTimestamp;
228 SendTelephoneEventPacket(ended, dtmf_payload_type, _dtmfTimestamp,
232 _dtmfTimestamp = captureTimeStamp;
237 ended, dtmf_payload_type, _dtmfTimestamp,
240 if (SendTelephoneEventPacket(ended, dtmf_payload_type, _dtmfTimestamp,
rtp_sender_audio.h 86 uint32_t _dtmfTimestamp;

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