Mercurial > mplayer.hg
view libao2/pl_volnorm.c @ 9278:caea8ed36b48
The reason why mplayer crashes (in some cases) when using x11
output and -wid (>0) parameter is this:
Mplayer by default creates a colormap using DirectColor visual. If the
window given to mplayer uses TrueColor visual there will be an error
when mplayer sets the colormap for the window. This patch
modifies mplayer to use TrueColor visual if the window given to mplayer
uses TrueColor. Another solution is to make sure that the window given to
mplayer is created using DirectColor visual if it is supported by the
display.
Jouni Tulkki <jitulkki@cc.hut.fi>
author | arpi |
---|---|
date | Tue, 04 Feb 2003 18:31:44 +0000 |
parents | 46d21c0f36aa |
children | 12b1790038b0 |
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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 plugin for xmms * * */ #define PLUGIN /* Values for AVG: * 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) * * */ #define AVG 1 #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 5.0 static float mul; #if AVG==1 // "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 #elif AVG==2 // Size of the memory array // FIXME: should depend on the frequency of the data (should be a few seconds) #define NSAMPLES 128 // Indicates where to write (in 0..NSAMPLES-1) static int idx; // The array static struct { float avg; // average level of the sample int32_t len; // sample size (weight) } mem[NSAMPLES]; // 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 #else // Kab00m ! #error "Unknown AVG" #endif // Some limits #define MIN_S16 -32768 #define MAX_S16 32767 // "Ideal" level #define MID_S16 (MAX_S16 * 0.25) // Silence level // FIXME: should be relative to the level of the samples #define SIL_S16 (MAX_S16 * 0.01) // 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_NE): 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(){ int i; mul = MUL_INIT; switch(ao_plugin_data.format) { case(AFMT_S16_NE): #if AVG==1 lastavg = MID_S16; #elif AVG==2 for(i=0; i < NSAMPLES; ++i) { mem[i].len = 0; mem[i].avg = 0; } idx = 0; #endif break; default: fprintf(stderr,"[pl_volnorm] internal inconsistency - 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_NE): { #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, tmp; float curavg, newavg; #if AVG==1 float neededmul; #elif AVG==2 float avg; int32_t totallen; #endif // Evaluate current samples average level curavg = 0.0; 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 #if AVG==1 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); } #elif AVG==2 avg = 0.0; totallen = 0; for (i = 0; i < NSAMPLES; ++i) { avg += mem[i].avg * (float) mem[i].len; totallen += mem[i].len; } if (totallen > MIN_SAMPLE_SIZE) { avg /= (float) totallen; if (avg >= SIL_S16) { mul = (float) MID_S16 / avg; CLAMP(mul, MUL_MIN, MUL_MAX); } } #endif // Scale & clamp the samples for (i = 0; i < len ; ++i) { tmp = mul * data[i]; CLAMP(tmp, MIN_S16, MAX_S16); data[i] = tmp; } // Evaluation of newavg (not 100% accurate because of values clamping) newavg = mul * curavg; // Stores computed values for future smoothing #if AVG==1 lastavg = (1.0 - SMOOTH_LASTAVG) * lastavg + SMOOTH_LASTAVG * newavg; //printf("\rmul=%02.1f ", mul); #elif AVG==2 mem[idx].len = len; mem[idx].avg = newavg; idx = (idx + 1) % NSAMPLES; //printf("\rmul=%02.1f (%04dKiB) ", mul, totallen/1024); #endif //fflush(stdout); break; } default: return 0; } return 1; }