Mercurial > mplayer.hg
view libao2/pl_eq.c @ 9009:f2d4324561b2
1) Removed the code to remove {} from sub_read_line_ssa
2) Put this code in a separated function: sub_pp_ssa
3) After recoding added a call to sub_pp_ssa. To be more acqurate: I
added a pointer to function to hold a postprocessing function to be
invoked after using iconv. There are one pointer for each format, if
this pointer isn.t NULL the post processing function is called.
patch by Salvador Eduardo Tropea <salvador@inti.gov.ar>
help by <hephooey@fastmail.fm>
author | arpi |
---|---|
date | Sun, 19 Jan 2003 00:54:55 +0000 |
parents | 46d21c0f36aa |
children | 12b1790038b0 |
line wrap: on
line source
/*============================================================================= // // This software has been released under the terms of the GNU Public // license. See http://www.gnu.org/copyleft/gpl.html for details. // // Copyright 2001 Anders Johansson ajh@atri.curtin.edu.au // //============================================================================= */ /* Equalizer plugin, implementation of a 10 band time domain graphic equalizer using IIR filters. The IIR filters are implemented using a Direct Form II approach. But has been modified (b1 == 0 always) to save computation. */ #define PLUGIN #include <stdio.h> #include <stdlib.h> #include <unistd.h> #include <inttypes.h> #include <math.h> #include "audio_out.h" #include "audio_plugin.h" #include "audio_plugin_internal.h" #include "afmt.h" #include "eq.h" static ao_info_t info = { "Equalizer audio plugin", "eq", "Anders", "" }; LIBAO_PLUGIN_EXTERN(eq) #define CH 6 // Max number of channels #define L 2 // Storage for filter taps #define KM 10 // Max number of octaves #define Q 1.2247 /* Q value for band-pass filters 1.2247=(3/2)^(1/2) gives 4dB suppression @ Fc*2 and Fc/2 */ // Center frequencies for band-pass filters #define CF {31.25,62.5,125,250,500,1000,2000,4000,8000,16000} // local data typedef struct pl_eq_s { int16_t a[KM][L]; // A weights int16_t b[KM][L]; // B weights int16_t wq[CH][KM][L]; // Circular buffer for W data int16_t g[CH][KM]; // Gain factor for each channel and band int16_t K; // Number of used eq bands int channels; // Number of channels } pl_eq_t; static pl_eq_t pl_eq; // to set/get/query special features/parameters static int control(int cmd,int arg){ switch(cmd){ case AOCONTROL_PLUGIN_SET_LEN: return CONTROL_OK; case AOCONTROL_PLUGIN_EQ_SET_GAIN:{ float gain = ((equalizer_t*)arg)->gain; int ch =((equalizer_t*)arg)->channel; int band =((equalizer_t*)arg)->band; if(ch > CH || ch < 0 || band > KM || band < 0) return CONTROL_ERROR; pl_eq.g[ch][band]=(int16_t) 4096 * (pow(10.0,gain/20.0)-1.0); return CONTROL_OK; } case AOCONTROL_PLUGIN_EQ_GET_GAIN:{ int ch =((equalizer_t*)arg)->channel; int band =((equalizer_t*)arg)->band; if(ch > CH || ch < 0 || band > KM || band < 0) return CONTROL_ERROR; ((equalizer_t*)arg)->gain = log10((float)pl_eq.g[ch][band]/4096.0+1) * 20.0; return CONTROL_OK; } } return CONTROL_UNKNOWN; } // return rounded 16bit int static inline int16_t lround16(double n){ return (int16_t)((n)>=0.0?(n)+0.5:(n)-0.5); } // 2nd order Band-pass Filter design void bp2(int16_t* a, int16_t* b, float fc, float q){ double th=2*3.141592654*fc; double C=(1 - tan(th*q/2))/(1 + tan(th*q/2)); a[0] = lround16( 16383.0 * (1 + C) * cos(th)); a[1] = lround16(-16383.0 * C); b[0] = lround16(-16383.0 * (C - 1)/2); b[1] = lround16(-16383.0 * 1.0050); } // empty buffers static void reset(){ int k,l,c; for(c=0;c<pl_eq.channels;c++) for(k=0;k<pl_eq.K;k++) for(l=0;l<L*2;l++) pl_eq.wq[c][k][l]=0; } // open & setup audio device // return: 1=success 0=fail static int init(){ int k = 0; float F[KM] = CF; // Check input format if(ao_plugin_data.format != AFMT_S16_NE){ fprintf(stderr,"[pl_eq] Input audio format not yet supported. \n"); return 0; } // Check number of channels if(ao_plugin_data.channels>CH){ fprintf(stderr,"[pl_eq] Too many channels, max is 6.\n"); return 0; } pl_eq.channels=ao_plugin_data.channels; // Calculate number of active filters pl_eq.K=KM; while(F[pl_eq.K-1] > (float)ao_plugin_data.rate/2) pl_eq.K--; // Generate filter taps for(k=0;k<pl_eq.K;k++) bp2(pl_eq.a[k],pl_eq.b[k],F[k]/((float)ao_plugin_data.rate),Q); // Reset buffers reset(); // Tell ao_plugin how much this plugin adds to the overall time delay ao_plugin_data.delay_fix-=2/((float)pl_eq.channels*(float)ao_plugin_data.rate); // Print some cool remark of what the plugin does printf("[pl_eq] Equalizer in use.\n"); return 1; } // close plugin static void uninit(){ } // processes 'ao_plugin_data.len' bytes of 'data' // called for every block of data static int play(){ uint16_t ci = pl_eq.channels; // Index for channels uint16_t nch = pl_eq.channels; // Number of channels while(ci--){ int16_t* g = pl_eq.g[ci]; // Gain factor int16_t* in = ((int16_t*)ao_plugin_data.data)+ci; int16_t* out = ((int16_t*)ao_plugin_data.data)+ci; int16_t* end = in+ao_plugin_data.len/2; // Block loop end while(in < end){ register int16_t k = 0; // Frequency band index register int32_t yt = 0; // Total output from filters register int16_t x = *in; // Current input sample in+=nch; // Run the filters for(;k<pl_eq.K;k++){ // Pointer to circular buffer wq register int16_t* wq = pl_eq.wq[ci][k]; #if 0 // Calculate output from AR part of current filter register int32_t xt = (x*pl_eq.b[k][0]) >> 4; register int32_t w = xt + wq[0]*pl_eq.a[k][0] + wq[1]*pl_eq.a[k][1]; // Calculate output form MA part of current filter yt+=(((w + wq[1]*pl_eq.b[k][1]) >> 10)*g[k]) >> 12; // Update circular buffer wq[1] = wq[0]; wq[0] = w >> 14; } // Calculate output *out=(int16_t)(yt+x); #else // Calculate output from AR part of current filter register int32_t xt = (x*pl_eq.b[k][0]) / 48; register int32_t w = xt + wq[0]*pl_eq.a[k][0] + wq[1]*pl_eq.a[k][1]; // Calculate output form MA part of current filter yt+=(((w + wq[1]*pl_eq.b[k][1]) >> 10)*g[k]) >> 12; // Update circular buffer wq[1] = wq[0]; wq[0] = w / 24576; } // Calculate output *out=(int16_t)(yt * 0.25 + x * 0.5); #endif out+=nch; } } return 1; }