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https://github.com/mpv-player/mpv
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![alex](/assets/img/avatar_default.png)
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@14246 b3059339-0415-0410-9bf9-f77b7e298cf2
251 lines
5.3 KiB
C
251 lines
5.3 KiB
C
/* Normalizer plugin
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*
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* Limitations:
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* - only AFMT_S16_LE supported
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* - no parameters yet => tweak the values by editing the #defines
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*
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* License: GPLv2
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* Author: pl <p_l@gmx.fr> (c) 2002 and beyond...
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*
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* Sources: some ideas from volnorm plugin for xmms
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*
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* */
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#define PLUGIN
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/* Values for AVG:
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* 1: uses a 1 value memory and coefficients new=a*old+b*cur (with a+b=1)
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*
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* 2: uses several samples to smooth the variations (standard weighted mean
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* on past samples)
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*
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* */
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#define AVG 1
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#include <stdio.h>
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#include <stdlib.h>
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#include <inttypes.h>
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#include <math.h> // for sqrt()
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#include "audio_out.h"
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#include "audio_plugin.h"
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#include "audio_plugin_internal.h"
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#include "libaf/af_format.h"
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static ao_info_t info = {
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"Volume normalizer",
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"volnorm",
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"pl <p_l@gmx.fr>",
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""
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};
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LIBAO_PLUGIN_EXTERN(volnorm)
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// mul is the value by which the samples are scaled
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// and has to be in [MUL_MIN, MUL_MAX]
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#define MUL_INIT 1.0
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#define MUL_MIN 0.1
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#define MUL_MAX 5.0
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static float mul;
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#if AVG==1
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// "history" value of the filter
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static float lastavg;
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// SMOOTH_* must be in ]0.0, 1.0[
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// The new value accounts for SMOOTH_MUL in the value and history
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#define SMOOTH_MUL 0.06
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#define SMOOTH_LASTAVG 0.06
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#elif AVG==2
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// Size of the memory array
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// FIXME: should depend on the frequency of the data (should be a few seconds)
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#define NSAMPLES 128
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// Indicates where to write (in 0..NSAMPLES-1)
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static int idx;
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// The array
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static struct {
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float avg; // average level of the sample
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int32_t len; // sample size (weight)
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} mem[NSAMPLES];
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// If summing all the mem[].len is lower than MIN_SAMPLE_SIZE bytes, then we
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// choose to ignore the computed value as it's not significant enough
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// FIXME: should depend on the frequency of the data (0.5s maybe)
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#define MIN_SAMPLE_SIZE 32000
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#else
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// Kab00m !
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#error "Unknown AVG"
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#endif
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// Some limits
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#define MIN_S16 -32768
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#define MAX_S16 32767
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// "Ideal" level
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#define MID_S16 (MAX_S16 * 0.25)
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// Silence level
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// FIXME: should be relative to the level of the samples
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#define SIL_S16 (MAX_S16 * 0.01)
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// Local data
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static struct {
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int inuse; // This plugin is in use TRUE, FALSE
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int format; // sample fomat
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} pl_volnorm = {0, 0};
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// minimal interface
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static int control(int cmd,void *arg){
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switch(cmd){
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case AOCONTROL_PLUGIN_SET_LEN:
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return CONTROL_OK;
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}
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return CONTROL_UNKNOWN;
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}
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// minimal interface
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// open & setup audio device
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// return: 1=success 0=fail
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static int init(){
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switch(ao_plugin_data.format){
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case(AF_FORMAT_S16_NE):
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break;
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default:
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fprintf(stderr,"[pl_volnorm] Audio format not yet supported.\n");
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return 0;
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}
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pl_volnorm.format = ao_plugin_data.format;
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pl_volnorm.inuse = 1;
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reset();
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printf("[pl_volnorm] Normalizer plugin in use.\n");
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return 1;
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}
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// close plugin
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static void uninit(){
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pl_volnorm.inuse=0;
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}
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// empty buffers
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static void reset(){
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int i;
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mul = MUL_INIT;
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switch(ao_plugin_data.format) {
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case(AF_FORMAT_S16_NE):
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#if AVG==1
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lastavg = MID_S16;
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#elif AVG==2
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for(i=0; i < NSAMPLES; ++i) {
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mem[i].len = 0;
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mem[i].avg = 0;
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}
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idx = 0;
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#endif
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break;
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default:
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fprintf(stderr,"[pl_volnorm] internal inconsistency - bugreport !\n");
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*(char *) 0 = 0;
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}
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}
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// processes 'ao_plugin_data.len' bytes of 'data'
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// called for every block of data
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static int play(){
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switch(pl_volnorm.format){
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case(AF_FORMAT_S16_NE): {
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#define CLAMP(x,m,M) do { if ((x)<(m)) (x) = (m); else if ((x)>(M)) (x) = (M); } while(0)
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int16_t* data=(int16_t*)ao_plugin_data.data;
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int len=ao_plugin_data.len / 2; // 16 bits samples
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int32_t i, tmp;
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float curavg, newavg;
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#if AVG==1
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float neededmul;
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#elif AVG==2
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float avg;
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int32_t totallen;
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#endif
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// Evaluate current samples average level
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curavg = 0.0;
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for (i = 0; i < len ; ++i) {
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tmp = data[i];
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curavg += tmp * tmp;
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}
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curavg = sqrt(curavg / (float) len);
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// Evaluate an adequate 'mul' coefficient based on previous state, current
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// samples level, etc
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#if AVG==1
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if (curavg > SIL_S16) {
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neededmul = MID_S16 / ( curavg * mul);
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mul = (1.0 - SMOOTH_MUL) * mul + SMOOTH_MUL * neededmul;
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// Clamp the mul coefficient
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CLAMP(mul, MUL_MIN, MUL_MAX);
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}
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#elif AVG==2
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avg = 0.0;
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totallen = 0;
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for (i = 0; i < NSAMPLES; ++i) {
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avg += mem[i].avg * (float) mem[i].len;
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totallen += mem[i].len;
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}
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if (totallen > MIN_SAMPLE_SIZE) {
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avg /= (float) totallen;
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if (avg >= SIL_S16) {
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mul = (float) MID_S16 / avg;
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CLAMP(mul, MUL_MIN, MUL_MAX);
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}
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}
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#endif
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// Scale & clamp the samples
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for (i = 0; i < len ; ++i) {
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tmp = mul * data[i];
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CLAMP(tmp, MIN_S16, MAX_S16);
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data[i] = tmp;
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}
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// Evaluation of newavg (not 100% accurate because of values clamping)
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newavg = mul * curavg;
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// Stores computed values for future smoothing
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#if AVG==1
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lastavg = (1.0 - SMOOTH_LASTAVG) * lastavg + SMOOTH_LASTAVG * newavg;
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//printf("\rmul=%02.1f ", mul);
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#elif AVG==2
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mem[idx].len = len;
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mem[idx].avg = newavg;
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idx = (idx + 1) % NSAMPLES;
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//printf("\rmul=%02.1f (%04dKiB) ", mul, totallen/1024);
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#endif
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//fflush(stdout);
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break;
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}
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default:
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return 0;
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}
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return 1;
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}
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