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  /external/webrtc/src/modules/audio_coding/codecs/isac/main/source/
bandwidth_estimator.c 828 if (State->StillBuffered < (1.0 - 1.0/BURST_LEN) * DelayBuildUp)
836 // max bps derived from StillBuffered and DelayBuildUp
839 State->StillBuffered) / (double)FrameSamples) * BottleNeck;
896 State->StillBuffered += TransmissionTime;
897 State->StillBuffered -= (FrameSamples * 1000) / FS; /* ms */
898 if (State->StillBuffered < 0.0)
900 State->StillBuffered = 0.0;
923 State->StillBuffered += TransmissionTime;
924 State->StillBuffered -= (FrameSamples * 1000) / FS; /* ms */
925 if (State->StillBuffered < 0.0
    [all...]
structs.h 241 double StillBuffered;
  /external/webrtc/src/modules/audio_coding/codecs/isac/fix/source/
bandwidth_estimator.c 856 if (State->StillBuffered < WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL((512 - WEBRTC_SPL_DIV(512, BURST_LEN)), DelayBuildUp), 9)) {
861 /* max bps derived from StillBuffered and DelayBuildUp values */
863 if (DelayBuildUp > State->StillBuffered) {
864 MinRate = WEBRTC_SPL_MUL(512 + WEBRTC_SPL_MUL(SAMPLES_PER_MSEC, WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(DelayBuildUp - State->StillBuffered, inv_Q12), 3)), BottleNeck);
865 } else if ((den = WEBRTC_SPL_MUL(SAMPLES_PER_MSEC, (State->StillBuffered - DelayBuildUp))) >= FrameSamples) {
    [all...]
structs.h 264 WebRtc_Word16 StillBuffered;

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