From d7f6cb23dec483f538239eb42e21c321713c976e Mon Sep 17 00:00:00 2001 From: uau Date: Thu, 1 Nov 2007 06:52:50 +0000 Subject: [PATCH] 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 | 3 ++- libaf/af.h | 5 +++-- libaf/af_equalizer.c | 2 +- libaf/af_lavcresample.c | 2 +- libaf/af_resample.c | 2 +- libaf/af_scaletempo.c | 5 +++++ 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;kdata->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; diff --git a/mplayer.c b/mplayer.c index a50e7f7694..6b7515e64b 100644 --- a/mplayer.c +++ b/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; }