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https://github.com/mpv-player/mpv
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audio/filter: remove delay audio filter
Similar filter is available in libavfilter.
This commit is contained in:
Paul B Mahol
wm4
parent
ec4a8bdb3d
commit
e2a54bb1ca
@ -312,34 +312,6 @@ Available filters are:
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``mpv '--af=format=channels=5.1' '--audio-channels=5.1'`` would always force
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``mpv '--af=format=channels=5.1' '--audio-channels=5.1'`` would always force
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remixing audio to 5.1 and output it like this.
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remixing audio to 5.1 and output it like this.
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``delay[=[ch1,ch2,...]]``
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Delays the sound to the loudspeakers such that the sound from the
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different channels arrives at the listening position simultaneously. It is
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only useful if you have more than 2 loudspeakers.
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``[ch1,ch2,...]``
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The delay in ms that should be imposed on each channel (floating point
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number between 0 and 1000).
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To calculate the required delay for the different channels, do as follows:
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1. Measure the distance to the loudspeakers in meters in relation to your
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listening position, giving you the distances s1 to s5 (for a 5.1
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system). There is no point in compensating for the subwoofer (you will
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not hear the difference anyway).
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2. Subtract the distances s1 to s5 from the maximum distance, i.e.
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``s[i] = max(s) - s[i]; i = 1...5``.
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3. Calculate the required delays in ms as ``d[i] = 1000*s[i]/342; i =
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1...5``.
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.. admonition:: Example
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``mpv --af=delay=[10.5,10.5,0,0,7,0] media.avi``
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Would delay front left and right by 10.5 ms, the two rear channels
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and the subwoofer by 0 ms and the center channel by 7 ms.
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``drc[=method:target]``
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``drc[=method:target]``
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Applies dynamic range compression. This maximizes the volume by compressing
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Applies dynamic range compression. This maximizes the volume by compressing
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the audio signal's dynamic range. (Formerly called ``volnorm``.)
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the audio signal's dynamic range. (Formerly called ``volnorm``.)
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@ -31,7 +31,6 @@
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#include "af.h"
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#include "af.h"
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// Static list of filters
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// Static list of filters
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extern const struct af_info af_info_delay;
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extern const struct af_info af_info_channels;
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extern const struct af_info af_info_channels;
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extern const struct af_info af_info_format;
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extern const struct af_info af_info_format;
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extern const struct af_info af_info_volume;
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extern const struct af_info af_info_volume;
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@ -46,7 +45,6 @@ extern const struct af_info af_info_lavfi;
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extern const struct af_info af_info_rubberband;
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extern const struct af_info af_info_rubberband;
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static const struct af_info *const filter_list[] = {
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static const struct af_info *const filter_list[] = {
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&af_info_delay,
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&af_info_channels,
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&af_info_channels,
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&af_info_format,
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&af_info_format,
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&af_info_volume,
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&af_info_volume,
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@ -1,194 +0,0 @@
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/*
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* This audio filter delays the output signal for the different
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* channels and can be used for simple position panning.
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* An extension for this filter would be a reverb.
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*
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* Original author: Anders
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*
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* This file is part of mpv.
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*
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* mpv is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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* (at your option) any later version.
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*
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* mpv is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License along
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* with mpv. If not, see <http://www.gnu.org/licenses/>.
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*/
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#include <stdio.h>
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#include <stdlib.h>
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#include <string.h>
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#include <inttypes.h>
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#include "common/common.h"
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#include "af.h"
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#define L 65536
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#define UPDATEQI(qi) qi=(qi+1)&(L-1)
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// Data for specific instances of this filter
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typedef struct af_delay_s
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{
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void* q[AF_NCH]; // Circular queues used for delaying audio signal
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int wi[AF_NCH]; // Write index
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int ri; // Read index
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float d[AF_NCH]; // Delay [ms]
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char *delaystr;
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}af_delay_t;
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// Initialization and runtime control
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static int control(struct af_instance* af, int cmd, void* arg)
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{
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af_delay_t* s = af->priv;
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switch(cmd){
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case AF_CONTROL_REINIT:{
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int i;
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struct mp_audio *in = arg;
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if (in->bps != 1 && in->bps != 2 && in->bps != 4) {
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MP_FATAL(af, "Sample format not supported\n");
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return AF_ERROR;
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}
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// Free prevous delay queues
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for(i=0;i<af->data->nch;i++)
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free(s->q[i]);
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mp_audio_force_interleaved_format(in);
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mp_audio_copy_config(af->data, in);
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// Allocate new delay queues
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for(i=0;i<af->data->nch;i++){
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s->q[i] = calloc(L,af->data->bps);
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if(NULL == s->q[i])
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MP_FATAL(af, "Out of memory\n");
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}
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if(AF_OK != af_from_ms(AF_NCH, s->d, s->wi, af->data->rate, 0.0, 1000.0))
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return AF_ERROR;
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s->ri = 0;
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for(i=0;i<AF_NCH;i++){
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MP_DBG(af, "Channel %i delayed by %0.3fms\n",
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i,MPCLAMP(s->d[i],0.0,1000.0));
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MP_TRACE(af, "Channel %i delayed by %i samples\n",
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i,s->wi[i]);
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}
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return AF_OK;
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}
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}
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return AF_UNKNOWN;
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}
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// Deallocate memory
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static void uninit(struct af_instance* af)
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{
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int i;
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for(i=0;i<AF_NCH;i++)
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free(((af_delay_t*)(af->priv))->q[i]);
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}
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static int filter_frame(struct af_instance *af, struct mp_audio *c)
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{
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if (!c)
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return 0;
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af_delay_t* s = af->priv; // Setup for this instance
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int nch = c->nch; // Number of channels
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int len = mp_audio_psize(c)/c->bps; // Number of sample in data chunk
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int ri = 0;
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int ch,i;
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if (af_make_writeable(af, c) < 0) {
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talloc_free(c);
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return -1;
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}
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for(ch=0;ch<nch;ch++){
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switch(c->bps){
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case 1:{
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int8_t* a = c->planes[0];
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int8_t* q = s->q[ch];
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int wi = s->wi[ch];
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ri = s->ri;
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for(i=ch;i<len;i+=nch){
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q[wi] = a[i];
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a[i] = q[ri];
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UPDATEQI(wi);
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UPDATEQI(ri);
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}
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s->wi[ch] = wi;
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break;
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}
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case 2:{
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int16_t* a = c->planes[0];
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int16_t* q = s->q[ch];
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int wi = s->wi[ch];
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ri = s->ri;
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for(i=ch;i<len;i+=nch){
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q[wi] = a[i];
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a[i] = q[ri];
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UPDATEQI(wi);
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UPDATEQI(ri);
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}
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s->wi[ch] = wi;
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break;
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}
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case 4:{
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int32_t* a = c->planes[0];
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int32_t* q = s->q[ch];
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int wi = s->wi[ch];
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ri = s->ri;
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for(i=ch;i<len;i+=nch){
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q[wi] = a[i];
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a[i] = q[ri];
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UPDATEQI(wi);
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UPDATEQI(ri);
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}
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s->wi[ch] = wi;
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break;
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}
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}
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}
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s->ri = ri;
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af_add_output_frame(af, c);
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return 0;
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}
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// Allocate memory and set function pointers
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static int af_open(struct af_instance* af){
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af->control=control;
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af->uninit=uninit;
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af->filter_frame = filter_frame;
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af_delay_t *s = af->priv;
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int n = 1;
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int i = 0;
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char* cl = s->delaystr;
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while(cl && n && i < AF_NCH ){
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sscanf(cl,"%f%n",&s->d[i],&n);
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if(n==0 || cl[n-1] == '\0')
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break;
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cl=&cl[n];
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if (*cl != ',')
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break;
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cl++;
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i++;
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}
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return AF_OK;
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}
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#define OPT_BASE_STRUCT af_delay_t
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const struct af_info af_info_delay = {
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.info = "Delay audio filter",
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.name = "delay",
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.open = af_open,
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.priv_size = sizeof(af_delay_t),
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.options = (const struct m_option[]) {
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OPT_STRING("delays", delaystr, 0),
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{0}
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},
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};
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@ -110,7 +110,6 @@ def build(ctx):
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( "audio/decode/dec_audio.c" ),
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( "audio/decode/dec_audio.c" ),
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( "audio/filter/af.c" ),
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( "audio/filter/af.c" ),
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( "audio/filter/af_channels.c" ),
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( "audio/filter/af_channels.c" ),
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( "audio/filter/af_delay.c" ),
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( "audio/filter/af_drc.c" ),
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( "audio/filter/af_drc.c" ),
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( "audio/filter/af_equalizer.c" ),
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( "audio/filter/af_equalizer.c" ),
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( "audio/filter/af_format.c" ),
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( "audio/filter/af_format.c" ),
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