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  /external/webrtc/webrtc/modules/audio_coding/codecs/isac/fix/source/
bandwidth_estimator.c 866 if (State->StillBuffered <
873 /* max bps derived from StillBuffered and DelayBuildUp values */
875 if (DelayBuildUp > State->StillBuffered) {
877 State->StillBuffered) * inv_Q12) >> 3)) * BottleNeck;
878 } else if ((den = WEBRTC_SPL_MUL(SAMPLES_PER_MSEC, (State->StillBuffered - DelayBuildUp))) >= FrameSamples) {
938 State->StillBuffered += TransmissionTime;
939 State->StillBuffered -= FrameSamples / SAMPLES_PER_MSEC; /* ms */
940 if (State->StillBuffered < 0) {
941 State->StillBuffered = 0;
944 if (State->StillBuffered > 2000)
    [all...]
structs.h 264 int16_t StillBuffered;
  /external/webrtc/webrtc/modules/audio_coding/codecs/isac/main/source/
bandwidth_estimator.c 839 if (State->StillBuffered < (1.0 - 1.0/BURST_LEN) * DelayBuildUp)
847 // max bps derived from StillBuffered and DelayBuildUp
850 State->StillBuffered) / (double)FrameSamples) * BottleNeck;
907 State->StillBuffered += TransmissionTime;
908 State->StillBuffered -= (FrameSamples * 1000) / FS; /* ms */
909 if (State->StillBuffered < 0.0)
911 State->StillBuffered = 0.0;
934 State->StillBuffered += TransmissionTime;
935 State->StillBuffered -= (FrameSamples * 1000) / FS; /* ms */
936 if (State->StillBuffered < 0.0
    [all...]
structs.h 243 double StillBuffered;

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