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-rw-r--r--audio/filter/af.c9
-rw-r--r--audio/filter/af.h5
-rw-r--r--audio/filter/af_channels.c255
-rw-r--r--audio/filter/af_equalizer.c215
-rw-r--r--audio/filter/af_pan.c206
-rw-r--r--audio/filter/af_volume.c188
6 files changed, 0 insertions, 878 deletions
diff --git a/audio/filter/af.c b/audio/filter/af.c
index a76945feea..dd78bb0cb5 100644
--- a/audio/filter/af.c
+++ b/audio/filter/af.c
@@ -31,25 +31,16 @@
#include "af.h"
// Static list of filters
-extern const struct af_info af_info_channels;
extern const struct af_info af_info_format;
-extern const struct af_info af_info_volume;
-extern const struct af_info af_info_equalizer;
-extern const struct af_info af_info_pan;
extern const struct af_info af_info_lavcac3enc;
extern const struct af_info af_info_lavrresample;
extern const struct af_info af_info_scaletempo;
-extern const struct af_info af_info_bs2b;
extern const struct af_info af_info_lavfi;
extern const struct af_info af_info_lavfi_bridge;
extern const struct af_info af_info_rubberband;
static const struct af_info *const filter_list[] = {
- &af_info_channels,
&af_info_format,
- &af_info_volume,
- &af_info_equalizer,
- &af_info_pan,
&af_info_lavcac3enc,
&af_info_lavrresample,
#if HAVE_RUBBERBAND
diff --git a/audio/filter/af.h b/audio/filter/af.h
index f66b189f14..58f67727a2 100644
--- a/audio/filter/af.h
+++ b/audio/filter/af.h
@@ -120,11 +120,6 @@ struct af_stream {
enum af_control {
AF_CONTROL_REINIT = 1,
AF_CONTROL_RESET,
- AF_CONTROL_SET_VOLUME,
- AF_CONTROL_SET_PAN_LEVEL,
- AF_CONTROL_SET_PAN_NOUT,
- AF_CONTROL_SET_PAN_BALANCE,
- AF_CONTROL_GET_PAN_BALANCE,
AF_CONTROL_SET_PLAYBACK_SPEED,
AF_CONTROL_SET_PLAYBACK_SPEED_RESAMPLE,
AF_CONTROL_GET_METADATA,
diff --git a/audio/filter/af_channels.c b/audio/filter/af_channels.c
deleted file mode 100644
index 7cd7810d08..0000000000
--- a/audio/filter/af_channels.c
+++ /dev/null
@@ -1,255 +0,0 @@
-/*
- * Audio filter that adds and removes channels, according to the
- * command line parameter channels. It is stupid and can only add
- * silence or copy channels, not mix or filter.
- *
- * Original author: Anders
- *
- * This file is part of mpv.
- *
- * mpv is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * mpv is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License along
- * with mpv. If not, see <http://www.gnu.org/licenses/>.
- */
-
-#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-#include <inttypes.h>
-
-#include "common/common.h"
-#include "af.h"
-
-#define FR 0
-#define TO 1
-
-typedef struct af_channels_s{
- int route[AF_NCH][2];
- int nch, nr;
- int router;
- char *routes;
-}af_channels_t;
-
-// Local function for copying data
-static void copy(struct af_instance *af, void* in, void* out,
- int ins, int inos,int outs, int outos, int len, int bps)
-{
- switch(bps){
- case 1:{
- int8_t* tin = (int8_t*)in;
- int8_t* tout = (int8_t*)out;
- tin += inos;
- tout += outos;
- len = len/ins;
- while(len--){
- *tout=*tin;
- tin +=ins;
- tout+=outs;
- }
- break;
- }
- case 2:{
- int16_t* tin = (int16_t*)in;
- int16_t* tout = (int16_t*)out;
- tin += inos;
- tout += outos;
- len = len/(2*ins);
- while(len--){
- *tout=*tin;
- tin +=ins;
- tout+=outs;
- }
- break;
- }
- case 3:{
- int8_t* tin = (int8_t*)in;
- int8_t* tout = (int8_t*)out;
- tin += 3 * inos;
- tout += 3 * outos;
- len = len / ( 3 * ins);
- while (len--) {
- tout[0] = tin[0];
- tout[1] = tin[1];
- tout[2] = tin[2];
- tin += 3 * ins;
- tout += 3 * outs;
- }
- break;
- }
- case 4:{
- int32_t* tin = (int32_t*)in;
- int32_t* tout = (int32_t*)out;
- tin += inos;
- tout += outos;
- len = len/(4*ins);
- while(len--){
- *tout=*tin;
- tin +=ins;
- tout+=outs;
- }
- break;
- }
- case 8:{
- int64_t* tin = (int64_t*)in;
- int64_t* tout = (int64_t*)out;
- tin += inos;
- tout += outos;
- len = len/(8*ins);
- while(len--){
- *tout=*tin;
- tin +=ins;
- tout+=outs;
- }
- break;
- }
- default:
- MP_ERR(af, "Unsupported number of bytes/sample: %i"
- " please report this error on the MPlayer mailing list. \n",bps);
- }
-}
-
-// Make sure the routes are sane
-static int check_routes(struct af_instance *af, int nin, int nout)
-{
- af_channels_t* s = af->priv;
- int i;
- if((s->nr < 1) || (s->nr > AF_NCH)){
- MP_ERR(af, "The number of routing pairs must be"
- " between 1 and %i. Current value is %i\n",AF_NCH,s->nr);
- return AF_ERROR;
- }
-
- for(i=0;i<s->nr;i++){
- if((s->route[i][FR] >= nin) || (s->route[i][TO] >= nout)){
- MP_ERR(af, "Invalid routing in pair nr. %i.\n", i);
- return AF_ERROR;
- }
- }
- return AF_OK;
-}
-
-// Initialization and runtime control
-static int control(struct af_instance* af, int cmd, void* arg)
-{
- af_channels_t* s = af->priv;
- switch(cmd){
- case AF_CONTROL_REINIT: ;
-
- struct mp_chmap chmap;
- mp_chmap_set_unknown(&chmap, s->nch);
- mp_audio_set_channels(af->data, &chmap);
-
- // Set default channel assignment
- if(!s->router){
- int i;
- // Make sure this filter isn't redundant
- if(af->data->nch == ((struct mp_audio*)arg)->nch)
- return AF_DETACH;
-
- // If mono: fake stereo
- if(((struct mp_audio*)arg)->nch == 1){
- s->nr = MPMIN(af->data->nch,2);
- for(i=0;i<s->nr;i++){
- s->route[i][FR] = 0;
- s->route[i][TO] = i;
- }
- }
- else{
- s->nr = MPMIN(af->data->nch, ((struct mp_audio*)arg)->nch);
- for(i=0;i<s->nr;i++){
- s->route[i][FR] = i;
- s->route[i][TO] = i;
- }
- }
- }
-
- af->data->rate = ((struct mp_audio*)arg)->rate;
- mp_audio_force_interleaved_format((struct mp_audio*)arg);
- mp_audio_set_format(af->data, ((struct mp_audio*)arg)->format);
- return check_routes(af,((struct mp_audio*)arg)->nch,af->data->nch);
- }
- return AF_UNKNOWN;
-}
-
-static int filter_frame(struct af_instance *af, struct mp_audio *c)
-{
- af_channels_t* s = af->priv;
- int i;
-
- if (!c)
- return 0;
-
- struct mp_audio *l = mp_audio_pool_get(af->out_pool, &af->fmt_out, c->samples);
- if (!l) {
- talloc_free(c);
- return -1;
- }
- mp_audio_copy_attributes(l, c);
-
- // Reset unused channels
- memset(l->planes[0],0,mp_audio_psize(c) / c->nch * l->nch);
-
- if(AF_OK == check_routes(af,c->nch,l->nch))
- for(i=0;i<s->nr;i++)
- copy(af, c->planes[0],l->planes[0],c->nch,s->route[i][FR],
- l->nch,s->route[i][TO],mp_audio_psize(c),c->bps);
-
- talloc_free(c);
- af_add_output_frame(af, l);
- return 0;
-}
-
-// Allocate memory and set function pointers
-static int af_open(struct af_instance* af){
- af->control=control;
- af->filter_frame = filter_frame;
- af_channels_t *s = af->priv;
-
- MP_WARN(af, "This filter is deprecated (no replacement).\n");
-
- // If router scan commandline for routing pairs
- if(s->routes && s->routes[0]){
- char* cp = s->routes;
- int ch = 0;
- // Scan for pairs on commandline
- do {
- int n = 0;
- if (ch >= AF_NCH) {
- MP_FATAL(af, "Can't have more than %d routes.\n", AF_NCH);
- return AF_ERROR;
- }
- sscanf(cp, "%i-%i%n" ,&s->route[ch][FR], &s->route[ch][TO], &n);
- MP_VERBOSE(af, "Routing from channel %i to"
- " channel %i\n",s->route[ch][FR],s->route[ch][TO]);
- cp = &cp[n];
- ch++;
- } while(*cp == ',' && *(cp++));
- s->nr = ch;
- if (s->nr > 0)
- s->router = 1;
- }
-
- return AF_OK;
-}
-
-#define OPT_BASE_STRUCT af_channels_t
-const struct af_info af_info_channels = {
- .info = "Insert or remove channels",
- .name = "channels",
- .open = af_open,
- .priv_size = sizeof(af_channels_t),
- .options = (const struct m_option[]) {
- OPT_INTRANGE("nch", nch, 0, 1, AF_NCH, OPTDEF_INT(2)),
- OPT_STRING("routes", routes, 0),
- {0}
- },
-};
diff --git a/audio/filter/af_equalizer.c b/audio/filter/af_equalizer.c
deleted file mode 100644
index 3f132fdc0c..0000000000
--- a/audio/filter/af_equalizer.c
+++ /dev/null
@@ -1,215 +0,0 @@
-/*
- * Equalizer filter, implementation of a 10 band time domain graphic
- * equalizer using IIR filters. The IIR filters are implemented using a
- * Direct Form II approach, but has been modified (b1 == 0 always) to
- * save computation.
- *
- * Copyright (C) 2001 Anders Johansson ajh@atri.curtin.edu.au
- *
- * This file is part of mpv.
- *
- * mpv is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * mpv is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License along
- * with mpv. If not, see <http://www.gnu.org/licenses/>.
- */
-
-#include <stdio.h>
-#include <stdlib.h>
-
-#include <inttypes.h>
-#include <math.h>
-
-#include "common/common.h"
-#include "af.h"
-
-#define L 2 // Storage for filter taps
-#define KM 10 // Max number of bands
-
-#define Q 1.2247449 /* Q value for band-pass filters 1.2247=(3/2)^(1/2)
- gives 4dB suppression @ Fc*2 and Fc/2 */
-
-/* Center frequencies for band-pass filters
- The different frequency bands are:
- nr. center frequency
- 0 31.25 Hz
- 1 62.50 Hz
- 2 125.0 Hz
- 3 250.0 Hz
- 4 500.0 Hz
- 5 1.000 kHz
- 6 2.000 kHz
- 7 4.000 kHz
- 8 8.000 kHz
- 9 16.00 kHz
-*/
-#define CF {31.25,62.5,125,250,500,1000,2000,4000,8000,16000}
-
-// Maximum and minimum gain for the bands
-#define G_MAX +12.0
-#define G_MIN -12.0
-
-// Data for specific instances of this filter
-typedef struct af_equalizer_s
-{
- float a[KM][L]; // A weights
- float b[KM][L]; // B weights
- float wq[AF_NCH][KM][L]; // Circular buffer for W data
- float g[AF_NCH][KM]; // Gain factor for each channel and band
- int K; // Number of used eq bands
- int channels; // Number of channels
- float gain_factor; // applied at output to avoid clipping
- double p[KM];
-} af_equalizer_t;
-
-// 2nd order Band-pass Filter design
-static void bp2(float* a, float* b, float fc, float q){
- double th= 2.0 * M_PI * fc;
- double C = (1.0 - tan(th*q/2.0))/(1.0 + tan(th*q/2.0));
-
- a[0] = (1.0 + C) * cos(th);
- a[1] = -1 * C;
-
- b[0] = (1.0 - C)/2.0;
- b[1] = -1.0050;
-}
-
-// Initialization and runtime control
-static int control(struct af_instance* af, int cmd, void* arg)
-{
- af_equalizer_t* s = (af_equalizer_t*)af->priv;
-
- switch(cmd){
- case AF_CONTROL_REINIT:{
- int k =0, i =0;
- float F[KM] = CF;
-
- s->gain_factor=0.0;
-
- // Sanity check
- if(!arg) return AF_ERROR;
-
- mp_audio_copy_config(af->data, (struct mp_audio*)arg);
- mp_audio_set_format(af->data, AF_FORMAT_FLOAT);
-
- // Calculate number of active filters
- s->K=KM;
- while(F[s->K-1] > (float)af->data->rate/2.2)
- s->K--;
-
- if(s->K != KM)
- MP_INFO(af, "Limiting the number of filters to"
- " %i due to low sample rate.\n",s->K);
-
- // Generate filter taps
- for(k=0;k<s->K;k++)
- 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 = 2.0 / (double)af->data->rate;
-
- // Calculate gain factor to prevent clipping at output
- for(k=0;k<AF_NCH;k++)
- {
- for(i=0;i<KM;i++)
- {
- if(s->gain_factor < s->g[k][i]) s->gain_factor=s->g[k][i];
- }
- }
-
- s->gain_factor=log10(s->gain_factor + 1.0) * 20.0;
-
- if(s->gain_factor > 0.0)
- {
- s->gain_factor=0.1+(s->gain_factor/12.0);
- }else{
- s->gain_factor=1;
- }
-
- return af_test_output(af,arg);
- }
- }
- return AF_UNKNOWN;
-}
-
-static int filter(struct af_instance* af, struct mp_audio* data)
-{
- struct mp_audio* c = data; // Current working data
- if (!c)
- return 0;
- af_equalizer_t* s = (af_equalizer_t*)af->priv; // Setup
- uint32_t ci = af->data->nch; // Index for channels
- uint32_t nch = af->data->nch; // Number of channels
-
- if (af_make_writeable(af, data) < 0) {
- talloc_free(data);
- return -1;
- }
-
- while(ci--){
- float* g = s->g[ci]; // Gain factor
- float* in = ((float*)c->planes[0])+ci;
- float* out = ((float*)c->planes[0])+ci;
- float* end = in + c->samples*c->nch; // Block loop end
-
- while(in < end){
- register int k = 0; // Frequency band index
- register float yt = *in; // Current input sample
- in+=nch;
-
- // Run the filters
- for(;k<s->K;k++){
- // Pointer to circular buffer wq
- register float* wq = s->wq[ci][k];
- // Calculate output from AR part of current filter
- register float w=yt*s->b[k][0] + wq[0]*s->a[k][0] + wq[1]*s->a[k][1];
- // Calculate output form MA part of current filter
- yt+=(w + wq[1]*s->b[k][1])*g[k];
- // Update circular buffer
- wq[1] = wq[0];
- wq[0] = w;
- }
- // Calculate output
- *out=yt*s->gain_factor;
- out+=nch;
- }
- }
- af_add_output_frame(af, data);
- return 0;
-}
-
-// Allocate memory and set function pointers
-static int af_open(struct af_instance* af){
- MP_WARN(af, "This filter is deprecated. Use 'anequalizer' or 'firequalizer' instead.\n");
- af->control=control;
- af->filter_frame = filter;
- af_equalizer_t *priv = af->priv;
- for(int i=0;i<AF_NCH;i++){
- for(int j=0;j<KM;j++){
- priv->g[i][j] = pow(10.0,MPCLAMP(priv->p[j],G_MIN,G_MAX)/20.0)-1.0;
- }
- }
- return AF_OK;
-}
-
-#define OPT_BASE_STRUCT af_equalizer_t
-const struct af_info af_info_equalizer = {
- .info = "Equalizer audio filter",
- .name = "equalizer",
- .open = af_open,
- .priv_size = sizeof(af_equalizer_t),
- .options = (const struct m_option[]) {
-#define BAND(n) OPT_DOUBLE("e" #n, p[n], 0)
- BAND(0), BAND(1), BAND(2), BAND(3), BAND(4),
- BAND(5), BAND(6), BAND(7), BAND(8), BAND(9),
- {0}
- },
-};
diff --git a/audio/filter/af_pan.c b/audio/filter/af_pan.c
deleted file mode 100644
index b2233a7191..0000000000
--- a/audio/filter/af_pan.c
+++ /dev/null
@@ -1,206 +0,0 @@
-/*
- * Copyright (C) 2002 Anders Johansson ajh@atri.curtin.edu.au
- *
- * This file is part of mpv.
- *
- * mpv is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * mpv is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License along
- * with mpv. If not, see <http://www.gnu.org/licenses/>.
- */
-
-#include <stdio.h>
-#include <stdlib.h>
-
-#include <inttypes.h>
-#include <math.h>
-#include <limits.h>
-
-#include "common/common.h"
-#include "af.h"
-
-// Data for specific instances of this filter
-typedef struct af_pan_s {
- int nch; // Number of output channels; zero means same as input
- float level[AF_NCH][AF_NCH]; // Gain level for each channel
- char *matrixstr;
-} af_pan_t;
-
-static void set_channels(struct mp_audio *mpa, int num)
-{
- struct mp_chmap map;
- // "unknown" channel layouts make it easier to pass through audio data,
- // without triggering remixing.
- mp_chmap_set_unknown(&map, num);
- mp_audio_set_channels(mpa, &map);
-}
-
-static void parse_matrix(struct af_instance *af, const char *cp)
-{
- af_pan_t *s = af->priv;
- int j = 0, k = 0, n;
- while (*cp && k < AF_NCH) {
- sscanf(cp, "%f%n" , &s->level[j][k], &n);
- MP_VERBOSE(af, "Pan level from channel %i to"
- " channel %i = %f\n", k, j, s->level[j][k]);
- cp = &cp[n];
- j++;
- if (j >= s->nch) {
- j = 0;
- k++;
- }
- if (*cp != ',')
- break;
- cp++;
- }
-
-}
-
-// Initialization and runtime control
-static int control(struct af_instance *af, int cmd, void *arg)
-{
- af_pan_t* s = af->priv;
-
- switch(cmd){
- case AF_CONTROL_REINIT:
- // Sanity check
- if (!arg)
- return AF_ERROR;
-
- af->data->rate = ((struct mp_audio*)arg)->rate;
- mp_audio_set_format(af->data, AF_FORMAT_FLOAT);
- set_channels(af->data, s->nch ? s->nch : ((struct mp_audio*)arg)->nch);
-
- if ((af->data->format != ((struct mp_audio*)arg)->format) ||
- (af->data->bps != ((struct mp_audio*)arg)->bps)) {
- mp_audio_set_format((struct mp_audio*)arg, af->data->format);
- return AF_FALSE;
- }
- return AF_OK;
- case AF_CONTROL_SET_PAN_LEVEL: {
- int i;
- int ch = ((af_control_ext_t*)arg)->ch;
- float *level = ((af_control_ext_t*)arg)->arg;
- if (ch >= AF_NCH)
- return AF_FALSE;
- for (i = 0; i < AF_NCH; i++)
- s->level[ch][i] = level[i];
- return AF_OK;
- }
- case AF_CONTROL_SET_PAN_NOUT:
- // Reinit must be called after this function has been called
- // Sanity check
- if (((int*)arg)[0] <= 0 || ((int*)arg)[0] > AF_NCH) {
- MP_ERR(af, "The number of output channels must be"
- " between 1 and %i. Current value is %i\n",
- AF_NCH, ((int*)arg)[0]);
- return AF_ERROR;
- }
- s->nch = ((int*)arg)[0];
- return AF_OK;
- case AF_CONTROL_SET_PAN_BALANCE: {
- float val = *(float*)arg;
- if (s->nch)
- return AF_ERROR;
- if (af->data->nch >= 2) {
- s->level[0][0] = MPMIN(1.f, 1.f - val);
- s->level[0][1] = MPMAX(0.f, val);
- s->level[1][0] = MPMAX(0.f, -val);
- s->level[1][1] = MPMIN(1.f, 1.f + val);
- }
- return AF_OK;
- }
- case AF_CONTROL_GET_PAN_BALANCE:
- if (s->nch)
- return AF_ERROR;
- *(float*)arg = s->level[0][1] - s->level[1][0];
- return AF_OK;
- case AF_CONTROL_COMMAND: {
- char **args = arg;
- if (!strcmp(args[0], "set-matrix")) {
- parse_matrix(af, args[1]);
- return CONTROL_OK;
- } else {
- return CONTROL_ERROR;
- }
- }
- }
- return AF_UNKNOWN;
-}
-
-static int filter_frame(struct af_instance *af, struct mp_audio *c)
-{
- if (!c)
- return 0;
- struct mp_audio *l = mp_audio_pool_get(af->out_pool, &af->fmt_out, c->samples);
- if (!l) {
- talloc_free(c);
- return -1;
- }
- mp_audio_copy_attributes(l, c);
-
- af_pan_t* s = af->priv; // Setup for this instance
- float *in = c->planes[0]; // Input audio data
- float *out = NULL; // Output audio data
- float *end = in+c->samples * c->nch; // End of loop
- int nchi = c->nch; // Number of input channels
- int ncho = l->nch; // Number of output channels
- register int j, k;
-
- out = l->planes[0];
- // Execute panning
- // FIXME: Too slow
- while (in < end) {
- for (j = 0; j < ncho; j++) {
- register float x = 0.0;
- register float *tin = in;
- for (k = 0; k < nchi; k++)
- x += tin[k] * s->level[j][k];
- out[j] = x;
- }
- out += ncho;
- in += nchi;
- }
-
- talloc_free(c);
- af_add_output_frame(af, l);
- return 0;
-}
-
-// Allocate memory and set function pointers
-static int af_open(struct af_instance *af)
-{
- af->control = control;
- af->filter_frame = filter_frame;
- MP_WARN(af, "This filter is deprecated. Use lavfi pan instead.\n");
- af_pan_t *s = af->priv;
- int nch = s->nch;
- if (nch && AF_OK != control(af, AF_CONTROL_SET_PAN_NOUT, &nch))
- return AF_ERROR;
-
- // Read pan values
- if (s->matrixstr)
- parse_matrix(af, s->matrixstr);
- return AF_OK;
-}
-
-#define OPT_BASE_STRUCT af_pan_t
-const struct af_info af_info_pan = {
- .info = "Panning audio filter",
- .name = "pan",
- .open = af_open,
- .priv_size = sizeof(af_pan_t),
- .options = (const struct m_option[]) {
- OPT_INTRANGE("channels", nch, 0, 0, AF_NCH),
- OPT_STRING("matrix", matrixstr, 0),
- {0}
- },
-};
diff --git a/audio/filter/af_volume.c b/audio/filter/af_volume.c
deleted file mode 100644
index 8fffc08853..0000000000
--- a/audio/filter/af_volume.c
+++ /dev/null
@@ -1,188 +0,0 @@
-/*
- * Copyright (C)2002 Anders Johansson ajh@atri.curtin.edu.au
- *
- * This file is part of mpv.
- *
- * mpv is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * mpv is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License along
- * with mpv. If not, see <http://www.gnu.org/licenses/>.
- */
-
-#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-
-#include <inttypes.h>
-#include <math.h>
-#include <limits.h>
-
-#include "common/common.h"
-#include "af.h"
-#include "demux/demux.h"
-
-struct priv {
- float level; // User-specified gain level for each channel
- float rgain; // Replaygain level
- int rgain_track; // Enable/disable track based replaygain
- int rgain_album; // Enable/disable album based replaygain
- float rgain_preamp; // Set replaygain pre-amplification
- int rgain_clip; // Enable/disable clipping prevention
- float replaygain_fallback;
- int soft; // Enable/disable soft clipping
- int fast; // Use fix-point volume control
- int detach; // Detach if gain volume is neutral
- float cfg_volume;
- int warn;
-};
-
-// Convert to gain value from dB. input <= -200dB will become 0 gain.
-static float from_dB(float in, float k, float mi, float ma)
-{
- if (in <= -200)
- return 0.0;
- return pow(10.0, MPCLAMP(in, mi, ma) / k);
-}
-
-static int control(struct af_instance *af, int cmd, void *arg)
-{
- struct priv *s = af->priv;
-
- switch (cmd) {
- case AF_CONTROL_REINIT: {
- struct mp_audio *in = arg;
-
- mp_audio_copy_config(af->data, in);
- mp_audio_force_interleaved_format(af->data);
-
- if (s->fast && af_fmt_from_planar(in->format) != AF_FORMAT_FLOAT) {
- mp_audio_set_format(af->data, AF_FORMAT_S16);
- } else {
- mp_audio_set_format(af->data, AF_FORMAT_FLOAT);
- }
- if (af_fmt_is_planar(in->format))
- mp_audio_set_format(af->data, af_fmt_to_planar(af->data->format));
- s->rgain = 1.0;
- struct replaygain_data *rg = af->replaygain_data;
- if ((s->rgain_track || s->rgain_album) && rg) {
- MP_VERBOSE(af, "Replaygain: Track=%f/%f Album=%f/%f\n",
- rg->track_gain, rg->track_peak,
- rg->album_gain, rg->album_peak);
-
- float gain, peak;
- if (s->rgain_track) {
- gain = rg->track_gain;
- peak = rg->track_peak;
- } else {
- gain = rg->album_gain;
- peak = rg->album_peak;
- }
-
- gain += s->rgain_preamp;
- s->rgain = from_dB(gain, 20.0, -200.0, 60.0);
-
- MP_VERBOSE(af, "Applying replay-gain: %f\n", s->rgain);
-
- if (!s->rgain_clip) { // clipping prevention
- s->rgain = MPMIN(s->rgain, 1.0 / peak);
- MP_VERBOSE(af, "...with clipping prevention: %f\n", s->rgain);
- }
- } else if (s->replaygain_fallback) {
- s->rgain = from_dB(s->replaygain_fallback, 20.0, -200.0, 60.0);
- MP_VERBOSE(af, "Applying fallback gain: %f\n", s->rgain);
- }
- if (s->detach && fabs(s->level * s->rgain - 1.0) < 0.00001)
- return AF_DETACH;
- return af_test_output(af, in);
- }
- case AF_CONTROL_SET_VOLUME:
- s->level = *(float *)arg;
- MP_VERBOSE(af, "volume gain: %f\n", s->level);
- return AF_OK;
- }
- return AF_UNKNOWN;
-}
-
-static void filter_plane(struct af_instance *af, struct mp_audio *data, int p)
-{
- struct priv *s = af->priv;
-
- float level = s->level * s->rgain * from_dB(s->cfg_volume, 20.0, -200.0, 60.0);
- int num_samples = data->samples * data->spf;
-
- if (af_fmt_from_planar(af->data->format) == AF_FORMAT_S16) {
- int vol = 256.0 * level;
- if (vol != 256) {
- if (af_make_writeable(af, data) < 0)
- return; // oom
- int16_t *a = data->planes[p];
- for (int i = 0; i < num_samples; i++) {
- int x = (a[i] * vol) >> 8;
- a[i] = MPCLAMP(x, SHRT_MIN, SHRT_MAX);
- }
- }
- } else if (af_fmt_from_planar(af->data->format) == AF_FORMAT_FLOAT) {
- float vol = level;
- if (vol != 1.0) {
- if (af_make_writeable(af, data) < 0)
- return; // oom
- float *a = data->planes[p];
- for (int i = 0; i < num_samples; i++) {
- float x = a[i] * vol;
- a[i] = s->soft ? af_softclip(x) : MPCLAMP(x, -1.0, 1.0);
- }
- }
- }
-}
-
-static int filter(struct af_instance *af, struct mp_audio *data)
-{
- if (data) {
- for (int n = 0; n < data->num_planes; n++)
- filter_plane(af, data, n);
- af_add_output_frame(af, data);
- }
- return 0;
-}
-
-static int af_open(struct af_instance *af)
-{
- struct priv *s = af->priv;
- if (s->warn)
- MP_WARN(af, "This filter is deprecated. Use --volume directly.\n");
- af->control = control;
- af->filter_frame = filter;
- s->level = 1.0;
- return AF_OK;
-}
-
-#define OPT_BASE_STRUCT struct priv
-
-// Description of this filter
-const struct af_info af_info_volume = {
- .info = "Volume control audio filter",
- .name = "volume",
- .open = af_open,
- .priv_size = sizeof(struct priv),
- .options = (const struct m_option[]) {
- OPT_FLOATRANGE("volumedb", cfg_volume, 0, -200, 60),
- OPT_FLAG("replaygain-track", rgain_track, 0),
- OPT_FLAG("replaygain-album", rgain_album, 0),
- OPT_FLOATRANGE("replaygain-preamp", rgain_preamp, 0, -15, 15),
- OPT_FLAG("replaygain-clip", rgain_clip, 0),
- OPT_FLOATRANGE("replaygain-fallback", replaygain_fallback, 0, -200, 60),
- OPT_FLAG("softclip", soft, 0),
- OPT_FLAG("s16", fast, 0),
- OPT_FLAG("detach", detach, 0),
- OPT_FLAG("warn", warn, 0, OPTDEF_INT(1)),
- {0}
- },
-};