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