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https://github.com/mpv-player/mpv
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Adding sub-woofer filter, use this filter to add a sub channel to the audio stream
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@8833 b3059339-0415-0410-9bf9-f77b7e298cf2
This commit is contained in:
anders
parent
850c82cf30
commit
4477f1232a
@ -2,7 +2,7 @@ include ../config.mak
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LIBNAME = libaf.a
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SRCS=af.c af_mp.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 af_tools.c af_comp.c af_gate.c af_pan.c af_surround.c
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SRCS=af.c af_mp.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 af_tools.c af_comp.c af_gate.c af_pan.c af_surround.c af_sub.c
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OBJS=$(SRCS:.c=.o)
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@ -20,6 +20,7 @@ extern af_info_t af_info_gate;
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extern af_info_t af_info_comp;
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extern af_info_t af_info_pan;
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extern af_info_t af_info_surround;
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extern af_info_t af_info_sub;
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static af_info_t* filter_list[]={ \
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&af_info_dummy,\
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@ -33,6 +34,7 @@ static af_info_t* filter_list[]={ \
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&af_info_comp,\
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&af_info_pan,\
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&af_info_surround,\
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&af_info_sub,\
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NULL \
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};
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181
libaf/af_sub.c
Normal file
181
libaf/af_sub.c
Normal file
@ -0,0 +1,181 @@
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/*=============================================================================
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//
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// This software has been released under the terms of the GNU Public
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// license. See http://www.gnu.org/copyleft/gpl.html for details.
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//
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// Copyright 2002 Anders Johansson ajh@watri.uwa.edu.au
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//
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//=============================================================================
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*/
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/* This filter adds a sub-woofer channels to the audio stream by
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averaging the left and right channel and low-pass filter them. The
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low-pass filter is implemented as a 4th order IIR Butterworth
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filter, with a variable cutoff frequency between 10 and 300 Hz. The
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filter gives 24dB/octave attenuation. There are two runtime
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controls one for setting which channel to insert the sub-audio into
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called AF_CONTROL_SUB_CH and one for setting the cutoff frequency
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called AF_CONTROL_SUB_FC.
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*/
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#include <stdio.h>
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#include <stdlib.h>
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#include <string.h>
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#include "af.h"
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#include "dsp.h"
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// Q value for low-pass filter
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#define Q 1.0
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// Analog domain biquad section
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typedef struct{
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float a[3]; // Numerator coefficients
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float b[3]; // Denominator coefficients
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} biquad_t;
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// S-parameters for designing 4th order Butterworth filter
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static biquad_t sp[2] = {{{1.0,0.0,0.0},{1.0,0.765367,1.0}},
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{{1.0,0.0,0.0},{1.0,1.847759,1.0}}};
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// Data for specific instances of this filter
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typedef struct af_sub_s
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{
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float w[2][4]; // Filter taps for low-pass filter
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float q[2][2]; // Circular queues
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float fc; // Cutoff frequency [Hz] for low-pass filter
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float k; // Filter gain;
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int ch; // Channel number which to insert the filtered data
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}af_sub_t;
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// Initialization and runtime control
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static int control(struct af_instance_s* af, int cmd, void* arg)
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{
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af_sub_t* s = af->setup;
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switch(cmd){
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case AF_CONTROL_REINIT:{
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// Sanity check
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if(!arg) return AF_ERROR;
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af->data->rate = ((af_data_t*)arg)->rate;
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af->data->nch = max(s->ch+1,((af_data_t*)arg)->nch);
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af->data->format = AF_FORMAT_F | AF_FORMAT_NE;
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af->data->bps = 4;
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// Design low-pass filter
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s->k = 1.0;
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if((-1 == szxform(sp[0].a, sp[0].b, Q, s->fc,
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(float)af->data->rate, &s->k, s->w[0])) ||
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(-1 == szxform(sp[1].a, sp[1].b, Q, s->fc,
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(float)af->data->rate, &s->k, s->w[1])))
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return AF_ERROR;
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return af_test_output(af,(af_data_t*)arg);
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}
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case AF_CONTROL_COMMAND_LINE:{
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int ch=5;
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float fc=60.0;
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sscanf(arg,"%f:%i", &fc , &ch);
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if(AF_OK != control(af,AF_CONTROL_SUB_CH | AF_CONTROL_SET, &ch))
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return AF_ERROR;
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return control(af,AF_CONTROL_SUB_FC | AF_CONTROL_SET, &fc);
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}
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case AF_CONTROL_SUB_CH | AF_CONTROL_SET: // Requires reinit
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// Sanity check
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if((*(int*)arg >= AF_NCH) || (*(int*)arg < 0)){
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af_msg(AF_MSG_ERROR,"[sub] Subwoofer channel number must be between "
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" 0 and %i current value is %i\n", AF_NCH-1, *(int*)arg);
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return AF_ERROR;
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}
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s->ch = *(int*)arg;
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return AF_OK;
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case AF_CONTROL_SUB_CH | AF_CONTROL_GET:
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*(int*)arg = s->ch;
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return AF_OK;
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case AF_CONTROL_SUB_FC | AF_CONTROL_SET: // Requires reinit
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// Sanity check
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if((*(float*)arg > 300) || (*(float*)arg < 20)){
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af_msg(AF_MSG_ERROR,"[sub] Cutoff frequency must be between 20Hz and"
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" 300Hz current value is %0.2f",*(float*)arg);
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return AF_ERROR;
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}
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// Set cutoff frequency
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s->fc = *(float*)arg;
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return AF_OK;
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case AF_CONTROL_SUB_FC | AF_CONTROL_GET:
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*(float*)arg = s->fc;
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return AF_OK;
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}
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return AF_UNKNOWN;
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}
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// Deallocate memory
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static void uninit(struct af_instance_s* af)
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{
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if(af->data)
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free(af->data);
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if(af->setup)
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free(af->setup);
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}
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#ifndef IIR
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#define IIR(in,w,q,out) { \
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float h0 = (q)[0]; \
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float h1 = (q)[1]; \
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float hn = (in) - h0 * (w)[0] - h1 * (w)[1]; \
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out = hn + h0 * (w)[2] + h1 * (w)[3]; \
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(q)[1] = h0; \
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(q)[0] = hn; \
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}
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#endif
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// Filter data through filter
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static af_data_t* play(struct af_instance_s* af, af_data_t* data)
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{
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af_data_t* c = data; // Current working data
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af_sub_t* s = af->setup; // Setup for this instance
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float* a = c->audio; // Audio data
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int len = c->len/4; // Number of samples in current audio block
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int nch = c->nch; // Number of channels
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int ch = s->ch; // Channel in which to insert the sub audio
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register int i;
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// Run filter
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for(i=0;i<len;i+=nch){
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// Average left and right
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register float x = 0.5 * (a[i] + a[i+1]);
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IIR(x * s->k, s->w[0], s->q[0], x);
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IIR(x , s->w[1], s->q[1], a[i+ch]);
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}
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return c;
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}
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// Allocate memory and set function pointers
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static int open(af_instance_t* af){
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af_sub_t* s;
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af->control=control;
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af->uninit=uninit;
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af->play=play;
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af->mul.n=1;
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af->mul.d=1;
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af->data=calloc(1,sizeof(af_data_t));
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af->setup=s=calloc(1,sizeof(af_sub_t));
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if(af->data == NULL || af->setup == NULL)
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return AF_ERROR;
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// Set default values
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s->ch = 5; // Channel nr 6
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s->fc = 60; // Cutoff frequency 60Hz
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return AF_OK;
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}
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// Description of this filter
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af_info_t af_info_sub = {
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"Audio filter for adding a sub-base channel",
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"sub",
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"Anders",
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"",
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AF_FLAGS_NOT_REENTRANT,
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open
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};
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@ -202,8 +202,17 @@ typedef struct af_control_ext_s{
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#define AF_CONTROL_EQUALIZER_GAIN 0x00001C00 | AF_CONTROL_FILTER_SPECIFIC
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// Set delay length in seconds
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// Delay length in ms, arg is a control_ext with a float*
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#define AF_CONTROL_DELAY_LEN 0x00001D00 | AF_CONTROL_FILTER_SPECIFIC
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// Subwoofer
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// Channel number which to insert the filtered data, arg in int*
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#define AF_CONTROL_SUB_CH 0x00001E00 | AF_CONTROL_FILTER_SPECIFIC
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// Cutoff frequency [Hz] for lowpass filter, arg is float*
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#define AF_CONTROL_SUB_FC 0x00001F00 | AF_CONTROL_FILTER_SPECIFIC
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#endif /*__af_control_h */
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176
libaf/filter.c
176
libaf/filter.c
@ -14,6 +14,10 @@
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#include <math.h>
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#include "dsp.h"
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/******************************************************************************
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* FIR filter implementations
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******************************************************************************/
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/* C implementation of FIR filter y=w*x
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n number of filter taps, where mod(n,4)==0
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@ -73,6 +77,9 @@ inline int updatepq(unsigned int n, unsigned int d, unsigned int xi, _ftype_t**
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return (++xi)&(n-1);
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}
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/******************************************************************************
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* FIR filter design
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******************************************************************************/
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/* Design FIR filter using the Window method
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@ -255,3 +262,172 @@ int design_pfir(unsigned int n, unsigned int k, _ftype_t* w, _ftype_t** pw, _fty
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}
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return -1;
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}
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/******************************************************************************
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* IIR filter design
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******************************************************************************/
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/* Helper functions for the bilinear transform */
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/* Pre-warp the coefficients of a numerator or denominator.
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Note that a0 is assumed to be 1, so there is no wrapping
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of it.
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*/
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void prewarp(_ftype_t* a, _ftype_t fc, _ftype_t fs)
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{
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_ftype_t wp;
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wp = 2.0 * fs * tan(M_PI * fc / fs);
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a[2] = a[2]/(wp * wp);
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a[1] = a[1]/wp;
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}
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/* Transform the numerator and denominator coefficients of s-domain
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biquad section into corresponding z-domain coefficients.
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The transfer function for z-domain is:
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1 + alpha1 * z^(-1) + alpha2 * z^(-2)
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H(z) = -------------------------------------
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1 + beta1 * z^(-1) + beta2 * z^(-2)
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Store the 4 IIR coefficients in array pointed by coef in following
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order:
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beta1, beta2 (denominator)
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alpha1, alpha2 (numerator)
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Arguments:
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a - s-domain numerator coefficients
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b - s-domain denominator coefficients
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k - filter gain factor. Initially set to 1 and modified by each
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biquad section in such a way, as to make it the
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coefficient by which to multiply the overall filter gain
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in order to achieve a desired overall filter gain,
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specified in initial value of k.
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fs - sampling rate (Hz)
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coef - array of z-domain coefficients to be filled in.
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Return: On return, set coef z-domain coefficients and k to the gain
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required to maintain overall gain = 1.0;
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*/
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void bilinear(_ftype_t* a, _ftype_t* b, _ftype_t* k, _ftype_t fs, _ftype_t *coef)
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{
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_ftype_t ad, bd;
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/* alpha (Numerator in s-domain) */
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ad = 4. * a[2] * fs * fs + 2. * a[1] * fs + a[0];
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/* beta (Denominator in s-domain) */
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bd = 4. * b[2] * fs * fs + 2. * b[1] * fs + b[0];
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/* Update gain constant for this section */
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*k *= ad/bd;
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/* Denominator */
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*coef++ = (2. * b[0] - 8. * b[2] * fs * fs)/bd; /* beta1 */
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*coef++ = (4. * b[2] * fs * fs - 2. * b[1] * fs + b[0])/bd; /* beta2 */
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/* Numerator */
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*coef++ = (2. * a[0] - 8. * a[2] * fs * fs)/ad; /* alpha1 */
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*coef = (4. * a[2] * fs * fs - 2. * a[1] * fs + a[0])/ad; /* alpha2 */
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}
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/* IIR filter design using bilinear transform and prewarp. Transforms
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2nd order s domain analog filter into a digital IIR biquad link. To
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create a filter fill in a, b, Q and fs and make space for coef and k.
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Example Butterworth design:
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Below are Butterworth polynomials, arranged as a series of 2nd
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order sections:
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Note: n is filter order.
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n Polynomials
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-------------------------------------------------------------------
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2 s^2 + 1.4142s + 1
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4 (s^2 + 0.765367s + 1) * (s^2 + 1.847759s + 1)
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6 (s^2 + 0.5176387s + 1) * (s^2 + 1.414214 + 1) * (s^2 + 1.931852s + 1)
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For n=4 we have following equation for the filter transfer function:
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1 1
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T(s) = --------------------------- * ----------------------------
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s^2 + (1/Q) * 0.765367s + 1 s^2 + (1/Q) * 1.847759s + 1
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The filter consists of two 2nd order sections since highest s power
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is 2. Now we can take the coefficients, or the numbers by which s
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is multiplied and plug them into a standard formula to be used by
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bilinear transform.
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Our standard form for each 2nd order section is:
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a2 * s^2 + a1 * s + a0
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H(s) = ----------------------
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b2 * s^2 + b1 * s + b0
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Note that Butterworth numerator is 1 for all filter sections, which
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means s^2 = 0 and s^1 = 0
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Lets convert standard Butterworth polynomials into this form:
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0 + 0 + 1 0 + 0 + 1
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--------------------------- * --------------------------
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1 + ((1/Q) * 0.765367) + 1 1 + ((1/Q) * 1.847759) + 1
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Section 1:
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a2 = 0; a1 = 0; a0 = 1;
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b2 = 1; b1 = 0.765367; b0 = 1;
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Section 2:
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a2 = 0; a1 = 0; a0 = 1;
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b2 = 1; b1 = 1.847759; b0 = 1;
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Q is filter quality factor or resonance, in the range of 1 to
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1000. The overall filter Q is a product of all 2nd order stages.
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For example, the 6th order filter (3 stages, or biquads) with
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individual Q of 2 will have filter Q = 2 * 2 * 2 = 8.
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Arguments:
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a - s-domain numerator coefficients, a[1] is always assumed to be 1.0
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b - s-domain denominator coefficients
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Q - Q value for the filter
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k - filter gain factor. Initially set to 1 and modified by each
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biquad section in such a way, as to make it the
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coefficient by which to multiply the overall filter gain
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in order to achieve a desired overall filter gain,
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specified in initial value of k.
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fs - sampling rate (Hz)
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coef - array of z-domain coefficients to be filled in.
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Note: Upon return from each call, the k argument will be set to a
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value, by which to multiply our actual signal in order for the gain
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to be one. On second call to szxform() we provide k that was
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changed by the previous section. During actual audio filtering
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k can be used for gain compensation.
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return -1 if fail 0 if success.
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*/
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int szxform(_ftype_t* a, _ftype_t* b, _ftype_t Q, _ftype_t fc, _ftype_t fs, _ftype_t *k, _ftype_t *coef)
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{
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_ftype_t at[3];
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_ftype_t bt[3];
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if(!a || !b || !k || !coef || (Q>1000.0 || Q< 1.0))
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return -1;
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memcpy(at,a,3*sizeof(_ftype_t));
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memcpy(bt,b,3*sizeof(_ftype_t));
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bt[1]/=Q;
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/* Calculate a and b and overwrite the original values */
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prewarp(at, fc, fs);
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prewarp(bt, fc, fs);
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/* Execute bilinear transform */
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bilinear(at, bt, k, fs, coef);
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return 0;
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}
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@ -45,14 +45,18 @@
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// Exported functions
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extern _ftype_t fir(unsigned int n, _ftype_t* w, _ftype_t* x);
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extern _ftype_t* pfir(unsigned int n, unsigned int k, unsigned int xi, _ftype_t** w, _ftype_t** x, _ftype_t* y, unsigned int s);
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extern int updateq(unsigned int n, unsigned int xi, _ftype_t* xq, _ftype_t* in);
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extern int updatepq(unsigned int n, unsigned int k, unsigned int xi, _ftype_t** xq, _ftype_t* in, unsigned int s);
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extern int design_fir(unsigned int n, _ftype_t* w, _ftype_t* fc, unsigned int flags, _ftype_t opt);
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extern int design_pfir(unsigned int n, unsigned int k, _ftype_t* w, _ftype_t** pw, _ftype_t g, unsigned int flags);
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extern int szxform(_ftype_t* a, _ftype_t* b, _ftype_t Q, _ftype_t fc, _ftype_t fs, _ftype_t *k, _ftype_t *coef);
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||||
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||||
/* Add new data to circular queue designed to be used with a FIR
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||||
filter. xq is the circular queue, in pointing at the new sample, xi
|
||||
current index for xq and n the length of the filter. xq must be n*2
|
||||
|
||||
Loading…
Reference in New Issue
Block a user