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  /external/webrtc/webrtc/modules/audio_coding/codecs/isac/fix/source/
bandwidth_estimator.c 78 bweStr->prevRtpNumber = 0;
244 bweStr->prevRtpNumber = rtpNumber;
327 if ( rtpNumber == bweStr->prevRtpNumber + 1 ) {
508 bweStr->prevRtpNumber = rtpNumber;
    [all...]
structs.h 195 uint16_t prevRtpNumber; /* Previous RTP timestamp from received packet */

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