Mercurial > mplayer.hg
diff libao2/pl_volnorm.c @ 4941:9a468a190c4c
volume normalizer plugin support
author | pl |
---|---|
date | Tue, 05 Mar 2002 09:17:36 +0000 |
parents | |
children | acdff5b36ea9 |
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--- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/libao2/pl_volnorm.c Tue Mar 05 09:17:36 2002 +0000 @@ -0,0 +1,171 @@ +/* Normalizer plugin + * + * Limitations: + * - only AFMT_S16_LE supported + * - no parameters yet => tweak the values by editing the #defines + * + * License: GPLv2 + * Author: pl <p_l@gmx.fr> (c) 2002 and beyond... + * + * Sources: some ideas from volnorm for xmms + * + * */ + +#define PLUGIN + +#include <stdio.h> +#include <stdlib.h> +#include <inttypes.h> +#include <math.h> // for sqrt() + +#include "audio_out.h" +#include "audio_plugin.h" +#include "audio_plugin_internal.h" +#include "afmt.h" + +static ao_info_t info = { + "Volume normalizer", + "volnorm", + "pl <p_l@gmx.fr>", + "" +}; + +LIBAO_PLUGIN_EXTERN(volnorm) + +// 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 15.0 +static float mul; + +// "history" value of the filter +static float lastavg; + +// SMOOTH_* must be in ]0.0, 1.0[ +// The new value accounts for SMOOTH_MUL in the value and history +#define SMOOTH_MUL 0.06 +#define SMOOTH_LASTAVG 0.06 + +// ideal average level +#define MID_S16 (INT16_MAX * 0.25) + +// silence level +#define SIL_S16 (INT16_MAX * 0.02) + +// local data +static struct { + int inuse; // This plugin is in use TRUE, FALSE + int format; // sample fomat +} pl_volnorm = {0, 0}; + + +// minimal interface +static int control(int cmd,int arg){ + switch(cmd){ + case AOCONTROL_PLUGIN_SET_LEN: + return CONTROL_OK; + } + return CONTROL_UNKNOWN; +} + +// minimal interface +// open & setup audio device +// return: 1=success 0=fail +static int init(){ + switch(ao_plugin_data.format){ + case(AFMT_S16_LE): + break; + default: + fprintf(stderr,"[pl_volnorm] Audio format not yet supported.\n"); + return 0; + } + + pl_volnorm.format = ao_plugin_data.format; + pl_volnorm.inuse = 1; + + reset(); + + printf("[pl_volnorm] Normalizer plugin in use.\n"); + return 1; +} + +// close plugin +static void uninit(){ + pl_volnorm.inuse=0; +} + +// empty buffers +static void reset(){ + mul = MUL_INIT; + switch(ao_plugin_data.format) { + case(AFMT_S16_LE): + lastavg = MID_S16; + break; + default: + fprintf(stderr,"[pl_volnorm] internal inconsistency - please bugreport.\n"); + *(char *) 0 = 0; + } +} + +// processes 'ao_plugin_data.len' bytes of 'data' +// called for every block of data +static int play(){ + + switch(pl_volnorm.format){ + case(AFMT_S16_LE): { + +#define CLAMP(x,m,M) do { if ((x)<(m)) (x) = (m); else if ((x)>(M)) (x) = (M); } while(0) + + int16_t* data=(int16_t*)ao_plugin_data.data; + int len=ao_plugin_data.len / 2; // 16 bits samples + + int32_t i; + register int32_t tmp; + register float curavg; + float newavg; + float neededmul; + + // average of the current samples + curavg = 0.0; + for (i = 0; i < len ; ++i) { + tmp = data[i]; + curavg += tmp * tmp; + } + curavg = sqrt(curavg / (float) len); + + if (curavg > SIL_S16) { + neededmul = MID_S16 / ( curavg * mul); + mul = (1.0 - SMOOTH_MUL) * mul + SMOOTH_MUL * neededmul; + + // Clamp the mul coefficient + CLAMP(mul, MUL_MIN, MUL_MAX); + } + + // Scale & clamp the samples + for (i=0; i < len ; ++i) { + tmp = data[i] * mul; + CLAMP(tmp, INT16_MIN, INT16_MAX); + data[i] = tmp; + } + + // Evaluation of newavg (not 100% accurate because of values clamping) + newavg = mul * curavg; + +#if 0 + printf("time = %d len = %d curavg = %6.0f lastavg = %6.0f newavg = %6.0f\n" + " needed_m = %2.2f m = %2.2f\n\n", + time(NULL), len, curavg, lastavg, newavg, neededmul, mul); +#endif + + lastavg = (1.0 - SMOOTH_LASTAVG) * lastavg + SMOOTH_LASTAVG * newavg; + + break; + } + default: + return 0; + } + return 1; + +} +