af_drc: remove

Remove low quality drc filter. Anyone whishing to have dynamic range
compression should use the much more powerful acompressor ffmpeg filter:

    mpv --af=lavfi=[acompressor] INPUT

Or with parameters:

    mpv --af=lavfi=[acompressor=threshold=-25dB:ratio=3:makeup=8dB] INPUT

Refer to https://ffmpeg.org/ffmpeg-filters.html#acompressor for a full
list of supported parameters.

Signed-off-by: wm4 <wm4@nowhere>

7 files changed, 2 insertions, 460 deletions

diff --git a/DOCS/interface-changes.rst b/DOCS/interface-changes.rst
index c82b0a62d0..911f6f8717 100644
--- a/DOCS/interface-changes.rst
+++ b/DOCS/interface-changes.rst
@@ -25,6 +25,7 @@ Interface changes
     - remove ppm, pgm, pgmyuv, tga choices from the --screenshot-format and
       --vo-image-format options
     - the "jpeg" choice in the option above now leads to a ".jpg" file extension
+    - --af=drc is gone (you can use e.g. lavfi/acompressor instead)
  --- mpv 0.24.0 ---
     - deprecate --hwdec-api and replace it with --opengl-hwdec-interop.
       The new option accepts both --hwdec values, as well as named backends.
diff --git a/DOCS/man/af.rst b/DOCS/man/af.rst
index 4c91b5a9b5..040b44cd64 100644
--- a/DOCS/man/af.rst
+++ b/DOCS/man/af.rst
@@ -319,29 +319,6 @@ Available filters are:
         the mixing matrix at runtime, without reinitializing the entire filter
         chain.
 
-``drc[=method:target]``
-    Applies dynamic range compression. This maximizes the volume by compressing
-    the audio signal's dynamic range. (Formerly called ``volnorm``.)
-
-    ``<method>``
-        Sets the used method.
-
-        1
-            Use a single sample to smooth the variations via the standard
-            weighted mean over past samples (default).
-        2
-            Use several samples to smooth the variations via the standard
-            weighted mean over past samples.
-
-    ``<target>``
-        Sets the target amplitude as a fraction of the maximum for the sample
-        type (default: 0.25).
-
-    .. note::
-
-        This filter can cause distortion with audio signals that have a very
-        large dynamic range.
-
 ``scaletempo[=option1:option2:...]``
     Scales audio tempo without altering pitch, optionally synced to playback
     speed (default).
diff --git a/DOCS/mplayer-changes.rst b/DOCS/mplayer-changes.rst
index df33f66e3f..7c8ec50a90 100644
--- a/DOCS/mplayer-changes.rst
+++ b/DOCS/mplayer-changes.rst
@@ -212,7 +212,7 @@ Command Line Switches
     ``-no<opt>``                ``--no-<opt>`` (add a dash)
     ``-a52drc level``           ``--ad-lavc-ac3drc=level``
     ``-ac spdifac3``            ``--ad=spdif:ac3`` (see ``--ad=help``)
-    ``-af volnorm``             ``--af=drc`` (renamed)
+    ``-af volnorm``             (removed; use acompressor ffmpeg filter instead)
     ``-afm hwac3``              ``--ad=spdif:ac3,spdif:dts``
     ``-ao alsa:device=hw=0.3``  ``--ao=alsa:device=[hw:0,3]``
     ``-aspect``                 ``--video-aspect``
diff --git a/TOOLS/lua/drc-control.lua b/TOOLS/lua/drc-control.lua
deleted file mode 100644
index 83cdb9c935..0000000000
--- a/TOOLS/lua/drc-control.lua
+++ /dev/null
@@ -1,99 +0,0 @@
--- This script enables live control of the dynamic range compression
--- (drc) audio filter while the video is playing back. This can be
--- useful to avoid having to stop and restart mpv to adjust filter
--- parameters. See the entry for "drc" under the "AUDIO FILTERS"
--- section of the man page for a complete description of the filter.
---
--- This script registers the key-binding "\" to toggle the filter between
---
--- * off
--- * method=1 (single-sample smoothing)
--- * method=2 (multi-sample smoothing)
---
--- It registers the keybindings ctrl+9/ctrl+0 to decrease/increase the
--- target ampltiude. These keys will insert the filter at the default
--- target amplitude of 0.25 if it was not previously present.
---
--- OSD feedback of the current filter state is displayed on pressing
--- each bound key.
-
-script_name = mp.get_script_name()
-
-function print_state(params)
-    if params then
-        mp.osd_message(script_name..":\n"
-                           .."method = "..params["method"].."\n"
-                           .."target = "..params["target"])
-    else
-        mp.osd_message(script_name..":\noff")
-    end
-end
-
-function get_index_of_drc(afs)
-    for i,af in pairs(afs) do
-        if af["label"] == script_name then
-            return i
-        end
-    end
-end
-
-function append_drc(afs)
-    afs[#afs+1] = {
-        name   = "drc",
-        label  = script_name,
-        params = {
-            method = "1",
-            target = "0.25"
-        }
-    }
-    print_state(afs[#afs]["params"])
-end
-
-function modify_or_create_af(fun)
-    afs = mp.get_property_native("af")
-    i = get_index_of_drc(afs)
-    if not i then
-        append_drc(afs)
-    else
-        fun(afs, i)
-    end
-    mp.set_property_native("af", afs)
-end
-
-function drc_toggle_method_handler()
-    modify_or_create_af(
-        function (afs, i)
-            new_method=(afs[i]["params"]["method"]+1)%3
-            if new_method == 0 then
-                table.remove(afs, i)
-                print_state(nil)
-            else
-                afs[i]["params"]["method"] = tostring((afs[i]["params"]["method"])%2+1)
-                print_state(afs[i]["params"])
-            end
-        end
-    )
-end
-
-function drc_scale_target(factor)
-    modify_or_create_af(
-        function (afs)
-            afs[i]["params"]["target"] = tostring(afs[i]["params"]["target"]*factor)
-            print_state(afs[i]["params"])
-        end
-    )
-end
-
-function drc_louder_handler()
-    drc_scale_target(2.0)
-end
-
-function drc_quieter_handler()
-    drc_scale_target(0.5)
-end
-
--- toggle between off, method 1 and method 2
-mp.add_key_binding("\\", "drc_toggle_method", drc_toggle_method_handler)
--- increase or decrease target volume
-mp.add_key_binding("ctrl+9", "drc_quieter", drc_quieter_handler)
-mp.add_key_binding("ctrl+0", "drc_louder", drc_louder_handler)
diff --git a/audio/filter/af.c b/audio/filter/af.c
index e6f19b3e37..31f4e45614 100644
--- a/audio/filter/af.c
+++ b/audio/filter/af.c
@@ -36,7 +36,6 @@ 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_drc;
 extern const struct af_info af_info_lavcac3enc;
 extern const struct af_info af_info_lavrresample;
 extern const struct af_info af_info_scaletempo;
@@ -50,7 +49,6 @@ static const struct af_info *const filter_list[] = {
     &af_info_volume,
     &af_info_equalizer,
     &af_info_pan,
-    &af_info_drc,
     &af_info_lavcac3enc,
     &af_info_lavrresample,
 #if HAVE_RUBBERBAND
diff --git a/audio/filter/af_drc.c b/audio/filter/af_drc.c
deleted file mode 100644
index 7b375febf4..0000000000
--- a/audio/filter/af_drc.c
+++ /dev/null
@@ -1,334 +0,0 @@
-/*
- * Copyright (C) 2004 Alex Beregszaszi & Pierre Lombard
- *
- * 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"
-
-// Methods:
-// 1: uses a 1 value memory and coefficients new=a*old+b*cur (with a+b=1)
-// 2: uses several samples to smooth the variations (standard weighted mean
-//    on past samples)
-
-// Size of the memory array
-// FIXME: should depend on the frequency of the data (should be a few seconds)
-#define NSAMPLES 128
-
-// If summing all the mem[].len is lower than MIN_SAMPLE_SIZE bytes, then we
-// choose to ignore the computed value as it's not significant enough
-// FIXME: should depend on the frequency of the data (0.5s maybe)
-#define MIN_SAMPLE_SIZE 32000
-
-// mul is the value by which the samples are scaled
-// and has to be in [MUL_MIN, MUL_MAX]
-#define MUL_INIT 1.0
-#define MUL_MIN 0.1
-#define MUL_MAX 5.0
-
-// Silence level
-// FIXME: should be relative to the level of the samples
-#define SIL_S16 (SHRT_MAX * 0.01)
-#define SIL_FLOAT 0.01
-
-// smooth must be in ]0.0, 1.0[
-#define SMOOTH_MUL 0.06
-#define SMOOTH_LASTAVG 0.06
-
-#define DEFAULT_TARGET 0.25
-
-// Data for specific instances of this filter
-typedef struct af_volume_s
-{
-    int method; // method used
-    float mul;
-    // method 1
-    float lastavg; // history value of the filter
-    // method 2
-    int idx;
-    struct {
-        float avg; // average level of the sample
-        int len; // sample size (weight)
-    } mem[NSAMPLES];
-    // "Ideal" level
-    float mid_s16;
-    float mid_float;
-}af_drc_t;
-
-// Initialization and runtime control
-static int control(struct af_instance* af, int cmd, void* arg)
-{
-  switch(cmd){
-  case AF_CONTROL_REINIT:
-    // Sanity check
-    if(!arg) return AF_ERROR;
-
-    mp_audio_force_interleaved_format((struct mp_audio*)arg);
-    mp_audio_copy_config(af->data, (struct mp_audio*)arg);
-
-    if(((struct mp_audio*)arg)->format != (AF_FORMAT_S16)){
-      mp_audio_set_format(af->data, AF_FORMAT_FLOAT);
-    }
-    return af_test_output(af,(struct mp_audio*)arg);
-  }
-  return AF_UNKNOWN;
-}
-
-static void method1_int16(af_drc_t *s, struct mp_audio *c)
-{
-  register int i = 0;
-  int16_t *data = (int16_t*)c->planes[0];       // Audio data
-  int len = c->samples*c->nch;          // Number of samples
-  float curavg = 0.0, newavg, neededmul;
-  int tmp;
-
-  for (i = 0; i < len; i++)
-  {
-    tmp = data[i];
-    curavg += tmp * tmp;
-  }
-  curavg = sqrt(curavg / (float) len);
-
-  // Evaluate an adequate 'mul' coefficient based on previous state, current
-  // samples level, etc
-
-  if (curavg > SIL_S16)
-  {
-    neededmul = s->mid_s16 / (curavg * s->mul);
-    s->mul = (1.0 - SMOOTH_MUL) * s->mul + SMOOTH_MUL * neededmul;
-
-    // clamp the mul coefficient
-    s->mul = MPCLAMP(s->mul, MUL_MIN, MUL_MAX);
-  }
-
-  // Scale & clamp the samples
-  for (i = 0; i < len; i++)
-  {
-    tmp = s->mul * data[i];
-    tmp = MPCLAMP(tmp, SHRT_MIN, SHRT_MAX);
-    data[i] = tmp;
-  }
-
-  // Evaluation of newavg (not 100% accurate because of values clamping)
-  newavg = s->mul * curavg;
-
-  // Stores computed values for future smoothing
-  s->lastavg = (1.0 - SMOOTH_LASTAVG) * s->lastavg + SMOOTH_LASTAVG * newavg;
-}
-
-static void method1_float(af_drc_t *s, struct mp_audio *c)
-{
-  register int i = 0;
-  float *data = (float*)c->planes[0];   // Audio data
-  int len = c->samples*c->nch;          // Number of samples
-  float curavg = 0.0, newavg, neededmul, tmp;
-
-  for (i = 0; i < len; i++)
-  {
-    tmp = data[i];
-    curavg += tmp * tmp;
-  }
-  curavg = sqrt(curavg / (float) len);
-
-  // Evaluate an adequate 'mul' coefficient based on previous state, current
-  // samples level, etc
-
-  if (curavg > SIL_FLOAT) // FIXME
-  {
-    neededmul = s->mid_float / (curavg * s->mul);
-    s->mul = (1.0 - SMOOTH_MUL) * s->mul + SMOOTH_MUL * neededmul;
-
-    // clamp the mul coefficient
-    s->mul = MPCLAMP(s->mul, MUL_MIN, MUL_MAX);
-  }
-
-  // Scale & clamp the samples
-  for (i = 0; i < len; i++)
-    data[i] *= s->mul;
-
-  // Evaluation of newavg (not 100% accurate because of values clamping)
-  newavg = s->mul * curavg;
-
-  // Stores computed values for future smoothing
-  s->lastavg = (1.0 - SMOOTH_LASTAVG) * s->lastavg + SMOOTH_LASTAVG * newavg;
-}
-
-static void method2_int16(af_drc_t *s, struct mp_audio *c)
-{
-  register int i = 0;
-  int16_t *data = (int16_t*)c->planes[0];       // Audio data
-  int len = c->samples*c->nch;          // Number of samples
-  float curavg = 0.0, newavg, avg = 0.0;
-  int tmp, totallen = 0;
-
-  for (i = 0; i < len; i++)
-  {
-    tmp = data[i];
-    curavg += tmp * tmp;
-  }
-  curavg = sqrt(curavg / (float) len);
-
-  // Evaluate an adequate 'mul' coefficient based on previous state, current
-  // samples level, etc
-  for (i = 0; i < NSAMPLES; i++)
-  {
-    avg += s->mem[i].avg * (float)s->mem[i].len;
-    totallen += s->mem[i].len;
-  }
-
-  if (totallen > MIN_SAMPLE_SIZE)
-  {
-    avg /= (float)totallen;
-    if (avg >= SIL_S16)
-    {
-        s->mul = s->mid_s16 / avg;
-        s->mul = MPCLAMP(s->mul, MUL_MIN, MUL_MAX);
-    }
-  }
-
-  // Scale & clamp the samples
-  for (i = 0; i < len; i++)
-  {
-    tmp = s->mul * data[i];
-    tmp = MPCLAMP(tmp, SHRT_MIN, SHRT_MAX);
-    data[i] = tmp;
-  }
-
-  // Evaluation of newavg (not 100% accurate because of values clamping)
-  newavg = s->mul * curavg;
-
-  // Stores computed values for future smoothing
-  s->mem[s->idx].len = len;
-  s->mem[s->idx].avg = newavg;
-  s->idx = (s->idx + 1) % NSAMPLES;
-}
-
-static void method2_float(af_drc_t *s, struct mp_audio *c)
-{
-  register int i = 0;
-  float *data = (float*)c->planes[0];   // Audio data
-  int len = c->samples*c->nch;          // Number of samples
-  float curavg = 0.0, newavg, avg = 0.0, tmp;
-  int totallen = 0;
-
-  for (i = 0; i < len; i++)
-  {
-    tmp = data[i];
-    curavg += tmp * tmp;
-  }
-  curavg = sqrt(curavg / (float) len);
-
-  // Evaluate an adequate 'mul' coefficient based on previous state, current
-  // samples level, etc
-  for (i = 0; i < NSAMPLES; i++)
-  {
-    avg += s->mem[i].avg * (float)s->mem[i].len;
-    totallen += s->mem[i].len;
-  }
-
-  if (totallen > MIN_SAMPLE_SIZE)
-  {
-    avg /= (float)totallen;
-    if (avg >= SIL_FLOAT)
-    {
-        s->mul = s->mid_float / avg;
-        s->mul = MPCLAMP(s->mul, MUL_MIN, MUL_MAX);
-    }
-  }
-
-  // Scale & clamp the samples
-  for (i = 0; i < len; i++)
-    data[i] *= s->mul;
-
-  // Evaluation of newavg (not 100% accurate because of values clamping)
-  newavg = s->mul * curavg;
-
-  // Stores computed values for future smoothing
-  s->mem[s->idx].len = len;
-  s->mem[s->idx].avg = newavg;
-  s->idx = (s->idx + 1) % NSAMPLES;
-}
-
-static int filter(struct af_instance *af, struct mp_audio *data)
-{
-  af_drc_t *s = af->priv;
-
-  if (!data)
-    return 0;
-
-  if (af_make_writeable(af, data) < 0) {
-    talloc_free(data);
-    return -1;
-  }
-
-  if(af->data->format == (AF_FORMAT_S16))
-  {
-    if (s->method == 2)
-        method2_int16(s, data);
-    else
-        method1_int16(s, data);
-  }
-  else if(af->data->format == (AF_FORMAT_FLOAT))
-  {
-    if (s->method == 2)
-        method2_float(s, data);
-    else
-        method1_float(s, data);
-  }
-  af_add_output_frame(af, data);
-  return 0;
-}
-
-// Allocate memory and set function pointers
-static int af_open(struct af_instance* af){
-  int i = 0;
-  af->control=control;
-  af->filter_frame = filter;
-  af_drc_t *priv = af->priv;
-
-  priv->mul = MUL_INIT;
-  priv->lastavg = ((float)SHRT_MAX) * DEFAULT_TARGET;
-  priv->idx = 0;
-  for (i = 0; i < NSAMPLES; i++)
-  {
-     priv->mem[i].len = 0;
-     priv->mem[i].avg = 0;
-  }
-  priv->mid_s16 = ((float)SHRT_MAX) * priv->mid_float;
-  return AF_OK;
-}
-
-#define OPT_BASE_STRUCT af_drc_t
-const struct af_info af_info_drc = {
-    .info = "Dynamic range compression filter",
-    .name = "drc",
-    .open = af_open,
-    .priv_size = sizeof(af_drc_t),
-    .options = (const struct m_option[]) {
-        OPT_INTRANGE("method", method, 0, 1, 2, OPTDEF_INT(1)),
-        OPT_FLOAT("target", mid_float, 0, OPTDEF_FLOAT(DEFAULT_TARGET)),
-        {0}
-    },
-};
diff --git a/wscript_build.py b/wscript_build.py
index 6014367115..4f1d3440c0 100644
--- a/wscript_build.py
+++ b/wscript_build.py
@@ -124,7 +124,6 @@ def build(ctx):
         ( "audio/decode/dec_audio.c" ),
         ( "audio/filter/af.c" ),
         ( "audio/filter/af_channels.c" ),
-        ( "audio/filter/af_drc.c" ),
         ( "audio/filter/af_equalizer.c" ),
         ( "audio/filter/af_format.c" ),
         ( "audio/filter/af_lavcac3enc.c" ),