From 222899fbbe523320e66ae1600fabe45b58d48686 Mon Sep 17 00:00:00 2001 From: Jan Janssen Date: Fri, 17 Mar 2017 16:49:28 +0100 Subject: 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 --- audio/filter/af.c | 2 - audio/filter/af_drc.c | 334 -------------------------------------------------- 2 files changed, 336 deletions(-) delete mode 100644 audio/filter/af_drc.c (limited to 'audio') 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 . - */ - -#include -#include -#include - -#include -#include -#include - -#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} - }, -}; -- cgit v1.2.3