diff options
29 files changed, 24 insertions, 56 deletions
diff --git a/audio/decode/dec_audio.c b/audio/decode/dec_audio.c index 3f92c3c4e6..ebb54cb55a 100644 --- a/audio/decode/dec_audio.c +++ b/audio/decode/dec_audio.c @@ -312,10 +312,9 @@ int decode_audio(sh_audio_t *sh_audio, struct mp_audio_buffer *outbuf, max_decode_len -= max_decode_len % unitsize; while (minsamples >= 0 && mp_audio_buffer_samples(outbuf) < minsamples) { - struct af_stream *afs = sh_audio->afilter; - int out_sstride = afs->output.sstride; - int declen = (minsamples - mp_audio_buffer_samples(outbuf)) - * out_sstride / filter_multiplier; + int decsamples = (minsamples - mp_audio_buffer_samples(outbuf)) + / filter_multiplier; + int declen = decsamples * sstride; // + some extra for possible filter buffering declen += unitsize << 5; if (huge_filter_buffer) diff --git a/audio/filter/af.c b/audio/filter/af.c index 182ed27d03..63013e81d9 100644 --- a/audio/filter/af.c +++ b/audio/filter/af.c @@ -703,12 +703,12 @@ struct mp_audio *af_play(struct af_stream *s, struct mp_audio *data) return data; } -// Calculate average ratio of filter output size to input size +// Calculate average ratio of filter output samples to input samples. +// e.g: num_output_samples = mul * num_input_samples double af_calc_filter_multiplier(struct af_stream *s) { struct af_instance *af = s->first; double mul = 1; - // Iterate through all filters and calculate total multiplication factor do { mul *= af->mul; af = af->next; @@ -721,11 +721,9 @@ double af_calc_filter_multiplier(struct af_stream *s) double af_calc_delay(struct af_stream *s) { struct af_instance *af = s->first; - register double delay = 0.0; - // Iterate through all filters + double delay = 0.0; while (af) { delay += af->delay; - delay *= af->mul; af = af->next; } return delay; diff --git a/audio/filter/af.h b/audio/filter/af.h index 36c2cf33fd..7852fa09a6 100644 --- a/audio/filter/af.h +++ b/audio/filter/af.h @@ -64,10 +64,11 @@ struct af_instance { struct mp_audio *data; // configuration and buffer for outgoing data stream struct af_instance *next; struct af_instance *prev; - double delay; /* Delay caused by the filter, in units of bytes read without - * corresponding output */ + double delay; /* Delay caused by the filter, in seconds of audio consumed + * without corresponding output */ double mul; /* length multiplier: how much does this instance change - the length of the buffer. */ + * the number of samples passed though. (Ratio of input + * and output, e.g. mul=4 => 1 sample becomes 4 samples) .*/ bool auto_inserted; // inserted by af.c, such as conversion filters }; diff --git a/audio/filter/af_bs2b.c b/audio/filter/af_bs2b.c index 73388ed5ba..769a2b4577 100644 --- a/audio/filter/af_bs2b.c +++ b/audio/filter/af_bs2b.c @@ -171,7 +171,6 @@ static int af_open(struct af_instance *af) struct af_bs2b *s = af->priv; af->control = control; af->uninit = uninit; - af->mul = 1; // NULL means failed initialization if (!(s->filter = bs2b_open())) { diff --git a/audio/filter/af_center.c b/audio/filter/af_center.c index 018fb756e6..ed482c7a6b 100644 --- a/audio/filter/af_center.c +++ b/audio/filter/af_center.c @@ -107,7 +107,6 @@ static int af_open(struct af_instance* af){ af->control=control; af->uninit=uninit; af->play=play; - af->mul=1; af->setup=s=calloc(1,sizeof(af_center_t)); if(af->setup == NULL) return AF_ERROR; diff --git a/audio/filter/af_channels.c b/audio/filter/af_channels.c index 7e05a9e234..f77807a1af 100644 --- a/audio/filter/af_channels.c +++ b/audio/filter/af_channels.c @@ -168,7 +168,6 @@ static int control(struct af_instance* af, int cmd, void* arg) af->data->rate = ((struct mp_audio*)arg)->rate; mp_audio_set_format(af->data, ((struct mp_audio*)arg)->format); - af->mul = (double)af->data->nch / ((struct mp_audio*)arg)->nch; mp_audio_force_interleaved_format(af->data); int r = af_test_output(af,(struct mp_audio*)arg); if (r != AF_OK) @@ -255,7 +254,6 @@ static int af_open(struct af_instance* af){ af->control=control; af->uninit=uninit; af->play=play; - af->mul=1; af->setup=calloc(1,sizeof(af_channels_t)); if(af->setup == NULL) return AF_ERROR; diff --git a/audio/filter/af_convert24.c b/audio/filter/af_convert24.c index 19904cc8e7..ee8aff5afc 100644 --- a/audio/filter/af_convert24.c +++ b/audio/filter/af_convert24.c @@ -53,8 +53,6 @@ static int control(struct af_instance *af, int cmd, void *arg) assert(test_conversion(in->format, out->format)); - af->mul = (double)out->bps / in->bps; - return mp_audio_config_equals(in, &orig_in) ? AF_OK : AF_FALSE; } case AF_CONTROL_FORMAT_FMT | AF_CONTROL_SET: { diff --git a/audio/filter/af_convertsignendian.c b/audio/filter/af_convertsignendian.c index 18e04be21a..65fffdf487 100644 --- a/audio/filter/af_convertsignendian.c +++ b/audio/filter/af_convertsignendian.c @@ -120,7 +120,6 @@ static int af_open(struct af_instance *af) { af->control = control; af->play = play; - af->mul = 1; return AF_OK; } diff --git a/audio/filter/af_delay.c b/audio/filter/af_delay.c index 27b1ca2a4a..e2ef57221d 100644 --- a/audio/filter/af_delay.c +++ b/audio/filter/af_delay.c @@ -181,7 +181,6 @@ static int af_open(struct af_instance* af){ af->control=control; af->uninit=uninit; af->play=play; - af->mul=1; af->setup=calloc(1,sizeof(af_delay_t)); if(af->setup == NULL) return AF_ERROR; diff --git a/audio/filter/af_drc.c b/audio/filter/af_drc.c index c1e043ac75..17c4a12a95 100644 --- a/audio/filter/af_drc.c +++ b/audio/filter/af_drc.c @@ -319,7 +319,6 @@ static int af_open(struct af_instance* af){ af->control=control; af->uninit=uninit; af->play=play; - af->mul=1; af->setup=calloc(1,sizeof(af_drc_t)); if(af->setup == NULL) return AF_ERROR; diff --git a/audio/filter/af_dummy.c b/audio/filter/af_dummy.c index 87b75fd1b0..d1cb054413 100644 --- a/audio/filter/af_dummy.c +++ b/audio/filter/af_dummy.c @@ -53,7 +53,6 @@ static struct mp_audio* play(struct af_instance* af, struct mp_audio* data) static int af_open(struct af_instance* af){ af->control=control; af->play=play; - af->mul=1; return AF_OK; } diff --git a/audio/filter/af_equalizer.c b/audio/filter/af_equalizer.c index 5fcfa18b35..75a489d867 100644 --- a/audio/filter/af_equalizer.c +++ b/audio/filter/af_equalizer.c @@ -114,7 +114,7 @@ static int control(struct af_instance* 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 = 2 * af->data->nch * af->data->bps; + af->delay = 2.0 / (double)af->data->rate; // Calculate gain factor to prevent clipping at output for(k=0;k<AF_NCH;k++) @@ -203,7 +203,6 @@ static int af_open(struct af_instance* af){ af->control=control; af->uninit=uninit; af->play=play; - af->mul=1; af->setup=calloc(1,sizeof(af_equalizer_t)); if(af->setup == NULL) return AF_ERROR; diff --git a/audio/filter/af_export.c b/audio/filter/af_export.c index 0139b17d4f..5e1096f85a 100644 --- a/audio/filter/af_export.c +++ b/audio/filter/af_export.c @@ -256,7 +256,6 @@ static int af_open( struct af_instance* af ) af->control = control; af->uninit = uninit; af->play = play; - af->mul=1; af->setup = calloc(1, sizeof(af_export_t)); if(af->setup == NULL) return AF_ERROR; diff --git a/audio/filter/af_extrastereo.c b/audio/filter/af_extrastereo.c index 68aeadac5e..4561b60690 100644 --- a/audio/filter/af_extrastereo.c +++ b/audio/filter/af_extrastereo.c @@ -128,7 +128,6 @@ static int af_open(struct af_instance* af){ af->control=control; af->uninit=uninit; af->play=play_s16; - af->mul=1; af->setup=calloc(1,sizeof(af_extrastereo_t)); if(af->setup == NULL) return AF_ERROR; diff --git a/audio/filter/af_format.c b/audio/filter/af_format.c index 58f311735e..5b941951cb 100644 --- a/audio/filter/af_format.c +++ b/audio/filter/af_format.c @@ -106,7 +106,6 @@ static int af_open(struct af_instance *af) { af->control = control; af->play = play; - af->mul = 1; force_in_params(af, af->data); force_out_params(af, af->data); diff --git a/audio/filter/af_hrtf.c b/audio/filter/af_hrtf.c index a0dbb12675..ed51351750 100644 --- a/audio/filter/af_hrtf.c +++ b/audio/filter/af_hrtf.c @@ -313,7 +313,6 @@ static int control(struct af_instance *af, int cmd, void* arg) mp_audio_set_channels_old(af->data, 5); mp_audio_set_format(af->data, AF_FORMAT_S16_NE); test_output_res = af_test_output(af, (struct mp_audio*)arg); - af->mul = 2.0 / af->data->nch; // after testing input set the real output format mp_audio_set_num_channels(af->data, 2); s->print_flag = 1; @@ -598,7 +597,6 @@ static int af_open(struct af_instance* af) af->control = control; af->uninit = uninit; af->play = play; - af->mul = 1; af->setup = calloc(1, sizeof(af_hrtf_t)); if(af->setup == NULL) return AF_ERROR; diff --git a/audio/filter/af_karaoke.c b/audio/filter/af_karaoke.c index 0d0ab7b2eb..8c633b136c 100644 --- a/audio/filter/af_karaoke.c +++ b/audio/filter/af_karaoke.c @@ -69,7 +69,6 @@ static struct mp_audio* play(struct af_instance* af, struct mp_audio* data) static int af_open(struct af_instance* af){ af->control = control; af->play = play; - af->mul = 1; return AF_OK; } diff --git a/audio/filter/af_ladspa.c b/audio/filter/af_ladspa.c index 3a6c258a89..df88c06ab2 100644 --- a/audio/filter/af_ladspa.c +++ b/audio/filter/af_ladspa.c @@ -875,7 +875,6 @@ static int af_open(struct af_instance *af) { af->control=control; af->uninit=uninit; af->play=play; - af->mul=1; af->setup = calloc(1, sizeof(af_ladspa_t)); if (af->setup == NULL) { diff --git a/audio/filter/af_lavcac3enc.c b/audio/filter/af_lavcac3enc.c index 6ecefa0d8f..2820e71f41 100644 --- a/audio/filter/af_lavcac3enc.c +++ b/audio/filter/af_lavcac3enc.c @@ -88,12 +88,13 @@ static int control(struct af_instance *af, int cmd, void *arg) test_output_res = af_test_output(af, data); s->pending_len = 0; - s->expect_len = AC3_FRAME_SIZE * data->nch * af->data->bps; + int expect_samples = AC3_FRAME_SIZE; + s->expect_len = expect_samples * data->nch * af->data->bps; assert(s->expect_len <= s->pending_data_size); if (s->add_iec61937_header) - af->mul = (double)AC3_FRAME_SIZE * 2 * 2 / s->expect_len; + af->mul = 1; else - af->mul = (double)AC3_MAX_CODED_FRAME_SIZE / s->expect_len; + af->mul = (double)(AC3_MAX_CODED_FRAME_SIZE / (2 * 2)) / expect_samples; mp_msg(MSGT_AFILTER, MSGL_DBG2, "af_lavcac3enc reinit: %d, %d, %f, %d.\n", data->nch, data->rate, af->mul, s->expect_len); @@ -306,7 +307,6 @@ static int af_open(struct af_instance* af){ af->control=control; af->uninit=uninit; af->play=play; - af->mul=1; af->setup=s; s->lavc_acodec = avcodec_find_encoder_by_name("ac3"); diff --git a/audio/filter/af_lavfi.c b/audio/filter/af_lavfi.c index e77110c067..24ff8c5985 100644 --- a/audio/filter/af_lavfi.c +++ b/audio/filter/af_lavfi.c @@ -197,7 +197,8 @@ static int control(struct af_instance *af, int cmd, void *arg) p->timebase_out = l_out->time_base; - af->mul = (double) (out->rate * out->nch) / (in->rate * in->nch); + // Blatantly incorrect; we don't know what the filters do. + af->mul = out->rate / (double)in->rate; return mp_audio_config_equals(in, &orig_in) ? AF_OK : AF_FALSE; } @@ -270,7 +271,7 @@ static struct mp_audio *play(struct af_instance *af, struct mp_audio *data) // Need pts past the last output sample. out_time += r->samples / (double)r->rate; - af->delay = (in_time - out_time) * r->rate * r->sstride; + af->delay = in_time - out_time; } *data = *r; @@ -286,7 +287,6 @@ static int af_open(struct af_instance *af) af->control = control; af->uninit = uninit; af->play = play; - af->mul = 1; struct priv *priv = af->priv; af->data = talloc_zero(priv, struct mp_audio), // Removing this requires fixing AVFrame.data vs. AVFrame.extended_data diff --git a/audio/filter/af_lavrresample.c b/audio/filter/af_lavrresample.c index 02d58122fb..2e298a3f39 100644 --- a/audio/filter/af_lavrresample.c +++ b/audio/filter/af_lavrresample.c @@ -249,8 +249,7 @@ static int control(struct af_instance *af, int cmd, void *arg) if (af_to_avformat(out->format) == AV_SAMPLE_FMT_NONE) mp_audio_set_format(out, in->format); - af->mul = (double) (out->rate * out->nch) / (in->rate * in->nch); - af->delay = out->nch * s->opts.filter_size / FFMIN(af->mul, 1); + af->mul = out->rate / (double)in->rate; int r = ((in->format == orig_in.format) && mp_chmap_equals(&in->channels, &orig_in.channels)) @@ -333,9 +332,7 @@ static struct mp_audio *play(struct af_instance *af, struct mp_audio *data) mp_audio_realloc_min(out, out->samples); - af->delay = out->bps * av_rescale_rnd(get_delay(s), - s->ctx.out_rate, s->ctx.in_rate, - AV_ROUND_UP); + af->delay = get_delay(s) / (double)s->ctx.in_rate; #if !USE_SET_CHANNEL_MAPPING do_reorder(in, s->reorder_in); @@ -380,7 +377,6 @@ static int af_open(struct af_instance *af) af->control = control; af->uninit = uninit; af->play = play; - af->mul = 1; if (s->opts.cutoff <= 0.0) s->opts.cutoff = af_resample_default_cutoff(s->opts.filter_size); diff --git a/audio/filter/af_pan.c b/audio/filter/af_pan.c index cc6f129542..3d8c6045d0 100644 --- a/audio/filter/af_pan.c +++ b/audio/filter/af_pan.c @@ -57,7 +57,6 @@ static int control(struct af_instance* af, int cmd, void* arg) af->data->rate = ((struct mp_audio*)arg)->rate; mp_audio_set_format(af->data, AF_FORMAT_FLOAT_NE); set_channels(af->data, s->nch ? s->nch: ((struct mp_audio*)arg)->nch); - af->mul = (double)af->data->nch / ((struct mp_audio*)arg)->nch; if((af->data->format != ((struct mp_audio*)arg)->format) || (af->data->bps != ((struct mp_audio*)arg)->bps)){ @@ -191,7 +190,6 @@ static int af_open(struct af_instance* af){ af->control=control; af->uninit=uninit; af->play=play; - af->mul=1; af->setup=calloc(1,sizeof(af_pan_t)); if(af->setup == NULL) return AF_ERROR; diff --git a/audio/filter/af_scaletempo.c b/audio/filter/af_scaletempo.c index 456de89812..c8560af502 100644 --- a/audio/filter/af_scaletempo.c +++ b/audio/filter/af_scaletempo.c @@ -256,7 +256,8 @@ static struct mp_audio *play(struct af_instance *af, struct mp_audio *data) // 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; + af->delay = (s->bytes_queued - s->bytes_to_slide) / s->scale + / af->data->sstride / af->data->rate; data->planes[0] = af->data->planes[0]; data->samples = (pout - (int8_t *)af->data->planes[0]) / af->data->sstride; @@ -299,9 +300,8 @@ static int control(struct af_instance *af, int cmd, void *arg) s->frames_stride = srate * s->ms_stride; s->bytes_stride = s->frames_stride * bps * nch; s->frames_stride_scaled = s->scale * s->frames_stride; - int bytes_stride_scaled = s->scale * s->bytes_stride; s->frames_stride_error = 0; - af->mul = (double)s->bytes_stride / bytes_stride_scaled; + af->mul = 1.0 / s->scale; af->delay = 0; int frames_overlap = s->frames_stride * s->percent_overlap; @@ -462,7 +462,6 @@ static int af_open(struct af_instance *af) af->control = control; af->uninit = uninit; af->play = play; - af->mul = 1; s->speed_tempo = !!(s->speed_opt & SCALE_TEMPO); s->speed_pitch = !!(s->speed_opt & SCALE_PITCH); diff --git a/audio/filter/af_sinesuppress.c b/audio/filter/af_sinesuppress.c index 33486390b9..ef6fd7d37b 100644 --- a/audio/filter/af_sinesuppress.c +++ b/audio/filter/af_sinesuppress.c @@ -148,7 +148,6 @@ static int af_open(struct af_instance* af){ af->control=control; af->uninit=uninit; af->play=play_s16; - af->mul=1; af->setup=calloc(1,sizeof(af_sinesuppress_t)); if(af->setup == NULL) return AF_ERROR; diff --git a/audio/filter/af_sub.c b/audio/filter/af_sub.c index 561ae94be6..4fd16904c9 100644 --- a/audio/filter/af_sub.c +++ b/audio/filter/af_sub.c @@ -165,7 +165,6 @@ static int af_open(struct af_instance* af){ af->control=control; af->uninit=uninit; af->play=play; - af->mul=1; af->setup=s=calloc(1,sizeof(af_sub_t)); if(af->setup == NULL) return AF_ERROR; diff --git a/audio/filter/af_surround.c b/audio/filter/af_surround.c index 9f9d46b788..efeecdc1e3 100644 --- a/audio/filter/af_surround.c +++ b/audio/filter/af_surround.c @@ -243,7 +243,6 @@ static int af_open(struct af_instance* af){ af->control=control; af->uninit=uninit; af->play=play; - af->mul=2; af->setup=calloc(1,sizeof(af_surround_t)); if(af->setup == NULL) return AF_ERROR; diff --git a/audio/filter/af_sweep.c b/audio/filter/af_sweep.c index 5b1e9d39c1..c153d4261a 100644 --- a/audio/filter/af_sweep.c +++ b/audio/filter/af_sweep.c @@ -84,7 +84,6 @@ static int af_open(struct af_instance* af){ af->control=control; af->uninit=uninit; af->play=play; - af->mul=1; af->setup=calloc(1,sizeof(af_sweept)); return AF_OK; } diff --git a/audio/filter/af_volume.c b/audio/filter/af_volume.c index 982b3816c7..b0eef1865c 100644 --- a/audio/filter/af_volume.c +++ b/audio/filter/af_volume.c @@ -104,7 +104,6 @@ static int af_open(struct af_instance *af) struct priv *s = af->priv; af->control = control; af->play = play; - af->mul = 1; af_from_dB(1, &s->cfg_volume, &s->level, 20.0, -200.0, 60.0); return AF_OK; } diff --git a/mpvcore/player/audio.c b/mpvcore/player/audio.c index fa29778fe9..9bad5140ef 100644 --- a/mpvcore/player/audio.c +++ b/mpvcore/player/audio.c @@ -203,7 +203,7 @@ double written_audio_pts(struct MPContext *mpctx) a_pts -= sh_audio->a_buffer_len / bps; // Data buffered in audio filters, measured in seconds of "missing" output - double buffered_output = af_calc_delay(sh_audio->afilter) / mpctx->ao->bps; + 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 |