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Code Issues Releases Wiki Activity

af_drc: remove

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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>
This commit is contained in:
Jan Janssen 2017-03-17 16:49:28 +01:00 committed by wm4
parent d663a0e90d
commit 222899fbbe
7 changed files with 2 additions and 460 deletions
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1
DOCS/interface-changes.rst
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@ -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.

23
DOCS/man/af.rst
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@ -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).

2
DOCS/mplayer-changes.rst
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@ -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``

99
TOOLS/lua/drc-control.lua
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@ -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)

2
audio/filter/af.c
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@ -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

334
audio/filter/af_drc.c
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@ -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}
},
};

1
wscript_build.py
View File

@ -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" ),