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authorJan Janssen <medhefgo@web.de>2017-03-17 16:49:28 +0100
committerwm4 <wm4@nowhere>2017-03-25 12:57:10 +0100
commit222899fbbe523320e66ae1600fabe45b58d48686 (patch)
treecdd094f983583de621ebb3d69c1bd426c5e142f9 /audio
parentd663a0e90dc0df3d8c50a471b918d8a6c6f78da5 (diff)
downloadmpv-222899fbbe523320e66ae1600fabe45b58d48686.tar.bz2
mpv-222899fbbe523320e66ae1600fabe45b58d48686.tar.xz
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>
Diffstat (limited to 'audio')
-rw-r--r--audio/filter/af.c2
-rw-r--r--audio/filter/af_drc.c334
2 files changed, 0 insertions, 336 deletions
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}
- },
-};