Mercurial > mplayer.hg
view libao2/pl_volnorm.c @ 4964:4cceff46d1d8
many changes but debian version stayed the same.
author | eyck |
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date | Wed, 06 Mar 2002 23:56:28 +0000 |
parents | 511e8d8117e9 |
children | 6d8971d55e40 |
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/* 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 // Some limits #define MIN_S16 -32768 #define MAX_S16 32767 // ideal average level #define MID_S16 (MAX_S16 * 0.25) // silence level #define SIL_S16 (MAX_S16 * 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, MIN_S16, MAX_S16); 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; }