af: remove deprecated audio filters

These couldn't be relicensed, and won't survive the LGPL transition. The
other existing filters are mostly LGPL (except libaf glue code).

This remove the deprecated pan option. I guess it could be restored by
inserting a libavfilter filter (if there's one), but for now let it be
gone.

This temporarily breaks volume control (and things related to it, like
replaygain).

1 files changed, 0 insertions, 215 deletions

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}
-  },
-};