[Mplayer-cvslog] CVS: main/libaf af_sub.c,NONE,1.1 Makefile,1.8,1.9 af.c,1.22,1.23 control.h,1.4,1.5 filter.c,1.1,1.2 filter.h,1.1,1.2
Anders Johansson CVS
anders at mplayerhq.hu
Tue Jan 7 11:33:46 CET 2003
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Update of /cvsroot/mplayer/main/libaf
In directory mail:/var/tmp.root/cvs-serv13532
Modified Files:
Makefile af.c control.h filter.c filter.h
Added Files:
af_sub.c
Log Message:
Adding sub-woofer filter, use this filter to add a sub channel to the audio stream
--- NEW FILE ---
/*=============================================================================
//
// 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 at 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
};
Index: Makefile
===================================================================
RCS file: /cvsroot/mplayer/main/libaf/Makefile,v
retrieving revision 1.8
retrieving revision 1.9
diff -u -r1.8 -r1.9
--- Makefile 31 Dec 2002 15:14:13 -0000 1.8
+++ Makefile 7 Jan 2003 10:33:30 -0000 1.9
@@ -2,7 +2,7 @@
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)
Index: af.c
===================================================================
RCS file: /cvsroot/mplayer/main/libaf/af.c,v
retrieving revision 1.22
retrieving revision 1.23
diff -u -r1.22 -r1.23
--- af.c 2 Jan 2003 03:00:44 -0000 1.22
+++ af.c 7 Jan 2003 10:33:30 -0000 1.23
@@ -20,6 +20,7 @@
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 @@
&af_info_comp,\
&af_info_pan,\
&af_info_surround,\
+ &af_info_sub,\
NULL \
};
Index: control.h
===================================================================
RCS file: /cvsroot/mplayer/main/libaf/control.h,v
retrieving revision 1.4
retrieving revision 1.5
diff -u -r1.4 -r1.5
--- control.h 28 Dec 2002 13:59:53 -0000 1.4
+++ control.h 7 Jan 2003 10:33:30 -0000 1.5
@@ -202,8 +202,17 @@
#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 */
Index: filter.c
===================================================================
RCS file: /cvsroot/mplayer/main/libaf/filter.c,v
retrieving revision 1.1
retrieving revision 1.2
diff -u -r1.1 -r1.2
--- filter.c 1 Oct 2002 06:45:08 -0000 1.1
+++ filter.c 7 Jan 2003 10:33:30 -0000 1.2
@@ -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 @@
return (++xi)&(n-1);
}
+/******************************************************************************
+* FIR filter design
+******************************************************************************/
/* Design FIR filter using the Window method
@@ -255,3 +262,172 @@
}
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;
+}
+
Index: filter.h
===================================================================
RCS file: /cvsroot/mplayer/main/libaf/filter.h,v
retrieving revision 1.1
retrieving revision 1.2
diff -u -r1.1 -r1.2
--- filter.h 1 Oct 2002 06:45:08 -0000 1.1
+++ filter.h 7 Jan 2003 10:33:30 -0000 1.2
@@ -45,13 +45,17 @@
// 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
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