From 1f7decc1a0a7e0f2fb547ee740ee0d7b659c0406 Mon Sep 17 00:00:00 2001 From: Martin Date: Tue, 12 Feb 2013 09:53:33 +0100 Subject: Rename af_volnorm to af_drc MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The previous name of this filter was misleading, because it doesn’t actually normalize volume levels. What it does is closer to performing low-quality dynamic range compression, hence it is now called af_drc. --- audio/filter/af_volnorm.c | 353 ---------------------------------------------- 1 file changed, 353 deletions(-) delete mode 100644 audio/filter/af_volnorm.c (limited to 'audio/filter/af_volnorm.c') diff --git a/audio/filter/af_volnorm.c b/audio/filter/af_volnorm.c deleted file mode 100644 index f49bbc185a..0000000000 --- a/audio/filter/af_volnorm.c +++ /dev/null @@ -1,353 +0,0 @@ -/* - * Copyright (C) 2004 Alex Beregszaszi & Pierre Lombard - * - * This file is part of MPlayer. - * - * MPlayer 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. - * - * MPlayer 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 MPlayer; if not, write to the Free Software Foundation, Inc., - * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. - */ - -#include -#include -#include - -#include -#include -#include - -#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_volnorm_t; - -// Initialization and runtime control -static int control(struct af_instance* af, int cmd, void* arg) -{ - af_volnorm_t* s = (af_volnorm_t*)af->setup; - - switch(cmd){ - case AF_CONTROL_REINIT: - // Sanity check - if(!arg) return AF_ERROR; - - af->data->rate = ((struct mp_audio*)arg)->rate; - af->data->nch = ((struct mp_audio*)arg)->nch; - - if(((struct mp_audio*)arg)->format == (AF_FORMAT_S16_NE)){ - af->data->format = AF_FORMAT_S16_NE; - af->data->bps = 2; - }else{ - af->data->format = AF_FORMAT_FLOAT_NE; - af->data->bps = 4; - } - return af_test_output(af,(struct mp_audio*)arg); - case AF_CONTROL_COMMAND_LINE:{ - int i = 0; - float target = DEFAULT_TARGET; - sscanf((char*)arg,"%d:%f", &i, &target); - if (i != 1 && i != 2) - return AF_ERROR; - s->method = i-1; - s->mid_s16 = ((float)SHRT_MAX) * target; - s->mid_float = target; - return AF_OK; - } - } - return AF_UNKNOWN; -} - -// Deallocate memory -static void uninit(struct af_instance* af) -{ - free(af->data); - free(af->setup); -} - -static void method1_int16(af_volnorm_t *s, struct mp_audio *c) -{ - register int i = 0; - int16_t *data = (int16_t*)c->audio; // Audio data - int len = c->len/2; // 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 = clamp(s->mul, MUL_MIN, MUL_MAX); - } - - // Scale & clamp the samples - for (i = 0; i < len; i++) - { - tmp = s->mul * data[i]; - tmp = clamp(tmp, SHRT_MIN, SHRT_MAX); - data[i] = tmp; - } - - // Evaulation 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_volnorm_t *s, struct mp_audio *c) -{ - register int i = 0; - float *data = (float*)c->audio; // Audio data - int len = c->len/4; // 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 = clamp(s->mul, MUL_MIN, MUL_MAX); - } - - // Scale & clamp the samples - for (i = 0; i < len; i++) - data[i] *= s->mul; - - // Evaulation 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_volnorm_t *s, struct mp_audio *c) -{ - register int i = 0; - int16_t *data = (int16_t*)c->audio; // Audio data - int len = c->len/2; // 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 = clamp(s->mul, MUL_MIN, MUL_MAX); - } - } - - // Scale & clamp the samples - for (i = 0; i < len; i++) - { - tmp = s->mul * data[i]; - tmp = clamp(tmp, SHRT_MIN, SHRT_MAX); - data[i] = tmp; - } - - // Evaulation 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_volnorm_t *s, struct mp_audio *c) -{ - register int i = 0; - float *data = (float*)c->audio; // Audio data - int len = c->len/4; // 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 = clamp(s->mul, MUL_MIN, MUL_MAX); - } - } - - // Scale & clamp the samples - for (i = 0; i < len; i++) - data[i] *= s->mul; - - // Evaulation 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; -} - -// Filter data through filter -static struct mp_audio* play(struct af_instance* af, struct mp_audio* data) -{ - af_volnorm_t *s = af->setup; - - if(af->data->format == (AF_FORMAT_S16_NE)) - { - if (s->method) - method2_int16(s, data); - else - method1_int16(s, data); - } - else if(af->data->format == (AF_FORMAT_FLOAT_NE)) - { - if (s->method) - method2_float(s, data); - else - method1_float(s, data); - } - return data; -} - -// Allocate memory and set function pointers -static int af_open(struct af_instance* af){ - int i = 0; - af->control=control; - af->uninit=uninit; - af->play=play; - af->mul=1; - af->data=calloc(1,sizeof(struct mp_audio)); - af->setup=calloc(1,sizeof(af_volnorm_t)); - if(af->data == NULL || af->setup == NULL) - return AF_ERROR; - - ((af_volnorm_t*)af->setup)->mul = MUL_INIT; - ((af_volnorm_t*)af->setup)->lastavg = ((float)SHRT_MAX) * DEFAULT_TARGET; - ((af_volnorm_t*)af->setup)->idx = 0; - ((af_volnorm_t*)af->setup)->mid_s16 = ((float)SHRT_MAX) * DEFAULT_TARGET; - ((af_volnorm_t*)af->setup)->mid_float = DEFAULT_TARGET; - for (i = 0; i < NSAMPLES; i++) - { - ((af_volnorm_t*)af->setup)->mem[i].len = 0; - ((af_volnorm_t*)af->setup)->mem[i].avg = 0; - } - return AF_OK; -} - -// Description of this filter -struct af_info af_info_volnorm = { - "Volume normalizer filter", - "volnorm", - "Alex Beregszaszi & Pierre Lombard", - "", - AF_FLAGS_NOT_REENTRANT, - af_open -}; -- cgit v1.2.3