From 4d5388c569cd615aadf3e5beb9a1ca8a53cf2630 Mon Sep 17 00:00:00 2001 From: anders Date: Sun, 3 Nov 2002 09:51:02 +0000 Subject: Adding equalizer filter + some cosmetics git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@8074 b3059339-0415-0410-9bf9-f77b7e298cf2 --- libaf/Makefile | 2 +- libaf/af.c | 2 + libaf/af.h | 6 +- libaf/af_equalizer.c | 217 +++++++++++++++++++++++++++++++++++++++++++++++++++ libaf/control.h | 6 ++ 5 files changed, 231 insertions(+), 2 deletions(-) create mode 100644 libaf/af_equalizer.c (limited to 'libaf') diff --git a/libaf/Makefile b/libaf/Makefile index acac0afada..381d9fcf1b 100644 --- a/libaf/Makefile +++ b/libaf/Makefile @@ -2,7 +2,7 @@ include ../config.mak LIBNAME = libaf.a -SRCS=af.c af_dummy.c af_delay.c af_channels.c af_format.c af_resample.c window.c filter.c af_volume.c +SRCS=af.c af_dummy.c af_delay.c af_channels.c af_format.c af_resample.c window.c filter.c af_volume.c af_equalizer.c OBJS=$(SRCS:.c=.o) diff --git a/libaf/af.c b/libaf/af.c index ea11afb302..63cb692bad 100644 --- a/libaf/af.c +++ b/libaf/af.c @@ -18,6 +18,7 @@ extern af_info_t af_info_channels; extern af_info_t af_info_format; extern af_info_t af_info_resample; extern af_info_t af_info_volume; +extern af_info_t af_info_equalizer; static af_info_t* filter_list[]={ \ &af_info_dummy,\ @@ -26,6 +27,7 @@ static af_info_t* filter_list[]={ \ &af_info_format,\ &af_info_resample,\ &af_info_volume,\ + &af_info_equalizer,\ NULL \ }; diff --git a/libaf/af.h b/libaf/af.h index 7c37f39c75..82a065ac95 100644 --- a/libaf/af.h +++ b/libaf/af.h @@ -184,7 +184,11 @@ int af_lencalc(frac_t mul, af_data_t* data); #endif #ifndef sign -#define sign(a) (((x)>0)?(1):(-1)) +#define sign(a) (((a)>0)?(1):(-1)) +#endif + +#ifndef lround +#define lround(a,b) ((b)((a)>=0.0?(a)+0.5:(a)-0.5)) #endif #endif diff --git a/libaf/af_equalizer.c b/libaf/af_equalizer.c new file mode 100644 index 0000000000..8b6129df54 --- /dev/null +++ b/libaf/af_equalizer.c @@ -0,0 +1,217 @@ +/*============================================================================= +// +// 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 filter, 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. +*/ + +#include +#include + +#include +#include +#include + +#include "../config.h" +#include "../mp_msg.h" +#include "../libao2/afmt.h" + +#include "af.h" +#include "equalizer.h" + +#define NCH 6 // Max number of channels +#define L 2 // Storage for filter taps +#define KM 10 // Max number of bands + +#define Q 1.2247449 /* 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} + +// Maximum and minimum gain for the bands +#define G_MAX +12.0 +#define G_MIN -12.0 + +// Data for specific instances of this filter +typedef struct af_equalizer_s +{ + float a[KM][L]; // A weights + float b[KM][L]; // B weights + float wq[NCH][KM][L]; // Circular buffer for W data + float g[NCH][KM]; // Gain factor for each channel and band + int K; // Number of used eq bands + int channels; // Number of channels +} af_equalizer_t; + +// 2nd order Band-pass Filter design +static void bp2(float* a, float* b, float fc, float q){ + double th= 2.0 * M_PI * fc; + double C = (1.0 - tan(th*q/2.0))/(1.0 + tan(th*q/2.0)); + + a[0] = (1.0 + C) * cos(th); + a[1] = -1 * C; + + b[0] = (1.0 - C)/2.0; + b[1] = -1.0050; +} + +// Initialization and runtime control +static int control(struct af_instance_s* af, int cmd, void* arg) +{ + af_equalizer_t* s = (af_equalizer_t*)af->setup; + + switch(cmd){ + case AF_CONTROL_REINIT:{ + int k =0; + float F[KM] = CF; + + // Sanity check + if(!arg) return AF_ERROR; + + af->data->rate = ((af_data_t*)arg)->rate; + af->data->nch = ((af_data_t*)arg)->nch; + af->data->format = AFMT_S16_LE; + af->data->bps = 2; + + // Calculate number of active filters + s->K=KM; + while(F[s->K-1] > (float)af->data->rate/2.0) + s->K--; + + // Generate filter taps + for(k=0;kK;k++) + bp2(s->a[k],s->b[k],F[k]/((float)af->data->rate),Q); + + // Calculate how much this plugin adds to the overall time delay + af->delay += 2000.0/((float)af->data->rate); + + // Only AFMT_S16_LE is supported + if(af->data->format != ((af_data_t*)arg)->format || + af->data->bps != ((af_data_t*)arg)->bps) + return AF_FALSE; + return AF_OK; + } + case AF_CONTROL_COMMAND_LINE:{ + float g[10]={0.0,0.0,0.0,0.0,0.0,0.0,0.0,0.0,0.0,0.0}; + int i,j; + sscanf((char*)arg,"%f:%f:%f:%f:%f:%f:%f:%f:%f:%f", &g[0], &g[1], + &g[2], &g[3], &g[4], &g[5], &g[6], &g[7], &g[8] ,&g[9]); + for(i=0;isetup)->g[i][j] = + pow(10.0,clamp(g[j],G_MIN,G_MAX)/20.0)-1.0; + } + } + return AF_OK; + } + case AF_CONTROL_EQUALIZER_SET_GAIN:{ + float gain = ((equalizer_t*)arg)->gain; + int ch = ((equalizer_t*)arg)->channel; + int band = ((equalizer_t*)arg)->band; + if(ch > NCH || ch < 0 || band > KM || band < 0) + return AF_ERROR; + + s->g[ch][band] = pow(10.0,clamp(gain,G_MIN,G_MAX)/20.0)-1.0; + return AF_OK; + } + case AF_CONTROL_EQUALIZER_GET_GAIN:{ + int ch =((equalizer_t*)arg)->channel; + int band =((equalizer_t*)arg)->band; + if(ch > NCH || ch < 0 || band > KM || band < 0) + return AF_ERROR; + + ((equalizer_t*)arg)->gain = log10(s->g[ch][band]+1.0) * 20.0; + return AF_OK; + } + } + return AF_UNKNOWN; +} + +// Deallocate memory +static void uninit(struct af_instance_s* af) +{ + if(af->data) + free(af->data); + if(af->setup) + free(af->setup); +} + +// Filter data through filter +static af_data_t* play(struct af_instance_s* af, af_data_t* data) +{ + af_data_t* c = data; // Current working data + af_equalizer_t* s = (af_equalizer_t*)af->setup; // Setup + uint32_t ci = af->data->nch; // Index for channels + uint32_t nch = af->data->nch; // Number of channels + + while(ci--){ + float* g = s->g[ci]; // Gain factor + int16_t* in = ((int16_t*)c->audio)+ci; + int16_t* out = ((int16_t*)c->audio)+ci; + int16_t* end = in + c->len/2; // Block loop end + + while(in < end){ + register uint32_t k = 0; // Frequency band index + register float yt = (float)(*in); // Current input sample + in+=nch; + + // Run the filters + for(;kK;k++){ + // Pointer to circular buffer wq + register float* wq = s->wq[ci][k]; + // Calculate output from AR part of current filter + register float w=yt*s->b[k][0] + wq[0]*s->a[k][0] + wq[1]*s->a[k][1]; + // Calculate output form MA part of current filter + yt+=(w + wq[1]*s->b[k][1])*g[k]; + // Update circular buffer + wq[1] = wq[0]; + wq[0] = w; + } + // Calculate output + *out=(int16_t)(yt/(4.0*10.0)); + out+=nch; + } + } + return c; +} + +// Allocate memory and set function pointers +static int open(af_instance_t* af){ + af->control=control; + af->uninit=uninit; + af->play=play; + af->mul.n=1; + af->mul.d=1; + af->data=calloc(1,sizeof(af_data_t)); + af->setup=calloc(1,sizeof(af_equalizer_t)); + if(af->data == NULL || af->setup == NULL) + return AF_ERROR; + return AF_OK; +} + +// Description of this filter +af_info_t af_info_equalizer = { + "Equalizer audio filter", + "equalizer", + "Anders", + "", + AF_FLAGS_NOT_REENTRANT, + open +}; + + + + + + + diff --git a/libaf/control.h b/libaf/control.h index 7757210647..1e160f460f 100644 --- a/libaf/control.h +++ b/libaf/control.h @@ -79,4 +79,10 @@ // Turn probing on and off, arg is binary #define AF_CONTROL_VOLUME_PROBE_ON_OFF 11 + AF_CONTROL_FILTER_SPECIFIC_BASE +// Set equalizer gain, arg is an equalizer_t* +#define AF_CONTROL_EQUALIZER_SET_GAIN 12 + AF_CONTROL_FILTER_SPECIFIC_BASE + +// Get equalizer gain, arg is an equalizer_t* +#define AF_CONTROL_EQUALIZER_GET_GAIN 13 + AF_CONTROL_FILTER_SPECIFIC_BASE + #endif /*__af_control_h */ -- cgit v1.2.3