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Diffstat (limited to 'libao2/pl_eq.c')
| -rw-r--r-- | libao2/pl_eq.c | 218 |
1 files changed, 0 insertions, 218 deletions
diff --git a/libao2/pl_eq.c b/libao2/pl_eq.c deleted file mode 100644 index 1165b8033d..0000000000 --- a/libao2/pl_eq.c +++ /dev/null @@ -1,218 +0,0 @@ -/*============================================================================= -// -// 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 "libaf/af_format.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,void *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 != AF_FORMAT_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; -} - - - - - - - |