diff options
| -rw-r--r-- | libaf/af.c | 3 | ||||
| -rw-r--r-- | libaf/af.h | 5 | ||||
| -rw-r--r-- | libaf/af_equalizer.c | 2 | ||||
| -rw-r--r-- | libaf/af_lavcresample.c | 2 | ||||
| -rw-r--r-- | libaf/af_resample.c | 2 | ||||
| -rw-r--r-- | libaf/af_scaletempo.c | 5 | ||||
| -rw-r--r-- | mplayer.c | 9 |
7 files changed, 21 insertions, 7 deletions
diff --git a/libaf/af.c b/libaf/af.c index 32c59dbbcd..c229f9eee6 100644 --- a/libaf/af.c +++ b/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; diff --git a/libaf/af.h b/libaf/af.h index 4c983cb6e4..a4dc6f9535 100644 --- a/libaf/af.h +++ b/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); diff --git a/libaf/af_equalizer.c b/libaf/af_equalizer.c index 9eb09a8188..5a92090d32 100644 --- a/libaf/af_equalizer.c +++ b/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++) diff --git a/libaf/af_lavcresample.c b/libaf/af_lavcresample.c index f44f26cabf..026646499d 100644 --- a/libaf/af_lavcresample.c +++ b/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); diff --git a/libaf/af_resample.c b/libaf/af_resample.c index e1bed4976b..57e2d39321 100644 --- a/libaf/af_resample.c +++ b/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; } diff --git a/libaf/af_scaletempo.c b/libaf/af_scaletempo.c index 4cdde2d5c0..3ed14aec68 100644 --- a/libaf/af_scaletempo.c +++ b/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; @@ -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; } |