comparison libaf/af_equalizer.c @ 24900:9079c9745ff9

A/V sync: take audio filter buffers into account Substract the delay caused by filter buffering when calculating currently playing audio position. This matters for af_scaletempo which buffers significant and varying amounts of data. For other current filters the effect is normally insignificant. Instead of the old time-based filter delay field (which was ignored) this version stores the per-filter delay in units of bytes input read without corresponding output. This allows the current scaletempo behavior where other filters before and after it can see the same nominal samplerate even though the real duration of the data varies; in this case the other filters can not know the delay they're causing in terms of real time.

24899:3245638ab801

    // Generate filter taps
    for(k=0;k<s->K;k++)
      bp2(s->a[k],s->b[k],F[k]/((float)af->data->rate),Q);

    // Calculate how much this plugin adds to the overall time delay
    af->delay += 2000.0/((float)af->data->rate);
    
    // Calculate gain factor to prevent clipping at output
    for(k=0;k<AF_NCH;k++)
    {
        for(i=0;i<KM;i++)

24900:9079c9745ff9

    // Generate filter taps
    for(k=0;k<s->K;k++)
      bp2(s->a[k],s->b[k],F[k]/((float)af->data->rate),Q);

    // Calculate how much this plugin adds to the overall time delay
    af->delay = 2 * af->data->nch * af->data->bps;
    
    // Calculate gain factor to prevent clipping at output
    for(k=0;k<AF_NCH;k++)
    {
        for(i=0;i<KM;i++)