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  /external/webrtc/webrtc/modules/audio_coding/codecs/isac/fix/source/
settings.h 27 #define FS3 (uint32_t) 48000
bandwidth_estimator.c 233 bweStr->lastReduction = arrivalTime + FS3;
271 if ((arrivalTime - bweStr->lastUpdate) > FS3) {
314 bweStr->lastReduction = arrivalTime + FS3;
320 bweStr->lastReduction = arrivalTime + FS3;
418 bweStr->lastReduction = arrivalTime + FS3;
488 bweStr->lastReduction = arrivalTime + FS3;
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