@@ -117,6 +117,7 @@ static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *insamples)

         ff_get_audio_buffer(inlink, AV_PERM_WRITE,

                                   insamples->audio->nb_samples);

     int ret;

+    int len;

     if (!outsamples)

         return AVERROR(ENOMEM);

@@ -126,16 +127,20 @@ static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *insamples)

     out   = (int16_t *)outsamples->data[0];

     in    = (int16_t *)insamples ->data[0];

+    len = FFMIN(NUMTAPS, 2*insamples->audio->nb_samples);

     // copy part of new input and process with saved input

-    memcpy(taps+NUMTAPS, in, NUMTAPS * sizeof(*taps));

-    out   = scalarproduct(taps, taps + NUMTAPS, out);

+    memcpy(taps+NUMTAPS, in, len * sizeof(*taps));

+    out   = scalarproduct(taps, taps + len, out);

     // process current input

-    endin = in + insamples->audio->nb_samples * 2 - NUMTAPS;

-    scalarproduct(in, endin, out);

-

-    // save part of input for next round

-    memcpy(taps, endin, NUMTAPS * sizeof(*taps));

+    if (2*insamples->audio->nb_samples >= NUMTAPS ){

+        endin = in + insamples->audio->nb_samples * 2 - NUMTAPS;

+        scalarproduct(in, endin, out);

+

+        // save part of input for next round

+        memcpy(taps, endin, NUMTAPS * sizeof(*taps));

+    } else

+        memmove(taps, taps + 2*insamples->audio->nb_samples, NUMTAPS * sizeof(*taps));

     ret = ff_filter_frame(outlink, outsamples);

     avfilter_unref_buffer(insamples);