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.


git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@24928 b3059339-0415-0410-9bf9-f77b7e298cf2

libaf/af.c

@@ -542,7 +542,7 @@ double af_calc_filter_multiplier(af_stream_t* s)
   return mul;
 }
 
-/* Calculate the total delay [ms] caused by the filters */
+/* Calculate the total delay [bytes output] caused by the filters */
 double af_calc_delay(af_stream_t* s)
 {
   af_instance_t* af=s->first; 
@@ -550,6 +550,7 @@ double af_calc_delay(af_stream_t* s)
   // Iterate through all filters 
   while(af){
     delay += af->delay;
+    delay *= af->mul;
     af=af->next;
   }
   return delay;

libaf/af.h

@@ -54,7 +54,8 @@ typedef struct af_instance_s
   af_data_t* data; // configuration for outgoing data stream
   struct af_instance_s* next;
   struct af_instance_s* prev;  
-  double delay; // Delay caused by the filter [ms]
+  double delay; /* Delay caused by the filter, in units of bytes read without
+		 * corresponding output */
   double mul; /* length multiplier: how much does this instance change
 		 the length of the buffer. */
 }af_instance_t;
@@ -196,7 +197,7 @@ double af_calc_filter_multiplier(af_stream_t* s);
 
 /**
  * \brief Calculate the total delay caused by the filters
- * \return delay in seconds
+ * \return delay in bytes of "missing" output
  */
 double af_calc_delay(af_stream_t* s);
 

libaf/af_equalizer.c

@@ -106,7 +106,7 @@ static int control(struct af_instance_s* af, int cmd, void* arg)
       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);
+    af->delay = 2 * af->data->nch * af->data->bps;
     
     // Calculate gain factor to prevent clipping at output
     for(k=0;k<AF_NCH;k++)

libaf/af_lavcresample.c

@@ -48,7 +48,7 @@ static int control(struct af_instance_s* af, int cmd, void* arg)
     af->data->format = AF_FORMAT_S16_NE;
     af->data->bps    = 2;
     af->mul = (double)af->data->rate / data->rate;
-    af->delay = 500*s->filter_length/(double)min(af->data->rate, data->rate);
+    af->delay = af->data->nch * s->filter_length / min(af->mul, 1); // *bps*.5
 
     if(s->avrctx) av_resample_close(s->avrctx);
     s->avrctx= av_resample_init(af->data->rate, /*in_rate*/data->rate, s->filter_length, s->phase_shift, s->linear, s->cutoff);

libaf/af_resample.c

@@ -254,7 +254,7 @@ static int control(struct af_instance_s* af, int cmd, void* arg)
     }
 
     // Set multiplier and delay
-    af->delay = (double)(1000*L/2)/((double)n->rate);
+    af->delay = 0; // not set correctly, but shouldn't be too large anyway
     af->mul = (double)s->up / s->dn;
     return rv;
   }

libaf/af_scaletempo.c

@@ -261,6 +261,11 @@ static af_data_t* play(struct af_instance_s* af, af_data_t* data)
     offset_in += fill_queue(af, data, offset_in);
   }
 
+  // This filter can have a negative delay when scale > 1:
+  // output corresponding to some length of input can be decided and written
+  // after receiving only a part of that input.
+  af->delay = s->bytes_queued - s->bytes_to_slide;
+
   data->audio = af->data->audio;
   data->len   = pout - (int8_t *)af->data->audio;
   return data;

mplayer.c

@@ -1589,9 +1589,16 @@ static double written_audio_pts(sh_audio_t *sh_audio, demux_stream_t *d_audio)
     // Decoded but not filtered
     a_pts -= sh_audio->a_buffer_len / (double)sh_audio->o_bps;
 
+    // Data buffered in audio filters, measured in bytes of "missing" output
+    double buffered_output = af_calc_delay(sh_audio->afilter);
+
     // Data that was ready for ao but was buffered because ao didn't fully
     // accept everything to internal buffers yet
-    a_pts -= sh_audio->a_out_buffer_len * playback_speed / (double)ao_data.bps;
+    buffered_output += sh_audio->a_out_buffer_len;
+
+    // Filters divide audio length by playback_speed, so multiply by it
+    // to get the length in original units without speedup or slowdown
+    a_pts -= buffered_output * playback_speed / ao_data.bps;
 
     return a_pts;
 }