[Mplayer-cvslog] CVS: main/libaf af_extrastereo.c, NONE, 1.1 af_volnorm.c, NONE, 1.1 Makefile, 1.14, 1.15 af.c, 1.29, 1.30 af_format.h, 1.2, 1.3 control.h, 1.6, 1.7
Alex Beregszaszi
syncmail at mplayerhq.hu
Mon Oct 4 21:11:08 CEST 2004
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CVS change done by Alex Beregszaszi
Update of /cvsroot/mplayer/main/libaf
In directory mail:/var2/tmp/cvs-serv22640
Modified Files:
Makefile af.c af_format.h control.h
Added Files:
af_extrastereo.c af_volnorm.c
Log Message:
reimplementation of the pl_extrastereo and pl_volnorm plugins
--- 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 2004 Alex Beregszaszi & Pierre Lombard
//
//=============================================================================
*/
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <unistd.h>
#include <inttypes.h>
#include <math.h>
#include <limits.h>
#include "af.h"
// Data for specific instances of this filter
typedef struct af_extrastereo_s
{
float mul;
}af_extrastereo_t;
// Initialization and runtime control
static int control(struct af_instance_s* af, int cmd, void* arg)
{
af_extrastereo_t* s = (af_extrastereo_t*)af->setup;
switch(cmd){
case AF_CONTROL_REINIT:
// Sanity check
if(!arg) return AF_ERROR;
if(((af_data_t*)arg)->format != (AF_FORMAT_SI | AF_FORMAT_NE) ||
(((af_data_t*)arg)->nch != 2))
return AF_ERROR;
af->data->rate = ((af_data_t*)arg)->rate;
af->data->nch = 2;
af->data->format = AF_FORMAT_SI | AF_FORMAT_NE;
af->data->bps = 2;
return af_test_output(af,(af_data_t*)arg);
case AF_CONTROL_COMMAND_LINE:{
float f;
sscanf((char*)arg,"%f", &f);
s->mul = f;
return AF_OK;
}
case AF_CONTROL_ES_MUL | AF_CONTROL_SET:
s->mul = *(float*)arg;
return AF_OK;
case AF_CONTROL_ES_MUL | AF_CONTROL_GET:
*(float*)arg = s->mul;
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_extrastereo_t *s = af->setup;
register int i = 0;
int16_t *a = (int16_t*)data->audio; // Audio data
int len = data->len/2; // Number of samples
int avg, l, r;
for (i = 0; i < len; i+=2)
{
avg = (a[i] + a[i + 1]) / 2;
l = avg + (int)(s->mul * (a[i] - avg));
r = avg + (int)(s->mul * (a[i + 1] - avg));
a[i] = clamp(l, SHRT_MIN, SHRT_MAX);
a[i + 1] = clamp(r, SHRT_MIN, SHRT_MAX);
}
return data;
}
// 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_extrastereo_t));
if(af->data == NULL || af->setup == NULL)
return AF_ERROR;
((af_extrastereo_t*)af->setup)->mul = 2.5;
return AF_OK;
}
// Description of this filter
af_info_t af_info_extrastereo = {
"Extra stereo",
"extrastereo",
"Alex Beregszaszi & Pierre Lombard",
"",
AF_FLAGS_NOT_REENTRANT,
open
};
--- 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 2004 Alex Beregszaszi & Pierre Lombard
//
//=============================================================================
*/
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <unistd.h>
#include <inttypes.h>
#include <math.h>
#include <limits.h>
#include "af.h"
// Methods:
// 1: uses a 1 value memory and coefficients new=a*old+b*cur (with a+b=1)
// 2: uses several samples to smooth the variations (standard weighted mean
// on past samples)
// Size of the memory array
// FIXME: should depend on the frequency of the data (should be a few seconds)
#define NSAMPLES 128
// If summing all the mem[].len is lower than MIN_SAMPLE_SIZE bytes, then we
// choose to ignore the computed value as it's not significant enough
// FIXME: should depend on the frequency of the data (0.5s maybe)
#define MIN_SAMPLE_SIZE 32000
// mul is the value by which the samples are scaled
// and has to be in [MUL_MIN, MUL_MAX]
#define MUL_INIT 1.0
#define MUL_MIN 0.1
#define MUL_MAX 5.0
// "Ideal" level
#define MID_S16 (SHRT_MAX * 0.25)
#define MID_FLOAT (INT_MAX * 0.25)
// Silence level
// FIXME: should be relative to the level of the samples
#define SIL_S16 (SHRT_MAX * 0.01)
#define SIL_FLOAT (INT_MAX * 0.01) // FIXME
// smooth must be in ]0.0, 1.0[
#define SMOOTH_MUL 0.06
#define SMOOTH_LASTAVG 0.06
// Data for specific instances of this filter
typedef struct af_volume_s
{
int method; // method used
float mul;
// method 1
float lastavg; // history value of the filter
// method 2
int idx;
struct {
float avg; // average level of the sample
int len; // sample size (weight)
} mem[NSAMPLES];
}af_volnorm_t;
// Initialization and runtime control
static int control(struct af_instance_s* af, int cmd, void* arg)
{
af_volnorm_t* s = (af_volnorm_t*)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 = ((af_data_t*)arg)->nch;
if(((af_data_t*)arg)->format != (AF_FORMAT_F | AF_FORMAT_NE) &&
((af_data_t*)arg)->format != (AF_FORMAT_SI | AF_FORMAT_NE))
return AF_ERROR;
if(((af_data_t*)arg)->format == (AF_FORMAT_SI | AF_FORMAT_NE)){
af->data->format = AF_FORMAT_SI | AF_FORMAT_NE;
af->data->bps = 2;
}else{
af->data->format = AF_FORMAT_F | AF_FORMAT_NE;
af->data->bps = 4;
}
return af_test_output(af,(af_data_t*)arg);
case AF_CONTROL_COMMAND_LINE:{
int i;
sscanf((char*)arg,"%d", &i);
if (i != 1 && i != 2)
return AF_ERROR;
s->method = i-1;
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);
}
static void method1_int16(af_volnorm_t *s, af_data_t *c)
{
register int i = 0;
int16_t *data = (int16_t*)c->audio; // Audio data
int len = c->len/2; // Number of samples
float curavg = 0.0, newavg, neededmul;
int tmp;
for (i = 0; i < len; i++)
{
tmp = data[i];
curavg += tmp * tmp;
}
curavg = sqrt(curavg / (float) len);
// Evaluate an adequate 'mul' coefficient based on previous state, current
// samples level, etc
if (curavg > SIL_S16)
{
neededmul = MID_S16 / (curavg * s->mul);
s->mul = (1.0 - SMOOTH_MUL) * s->mul + SMOOTH_MUL * neededmul;
// clamp the mul coefficient
s->mul = clamp(s->mul, MUL_MIN, MUL_MAX);
}
// Scale & clamp the samples
for (i = 0; i < len; i++)
{
tmp = s->mul * data[i];
tmp = clamp(tmp, SHRT_MIN, SHRT_MAX);
data[i] = tmp;
}
// Evaulation of newavg (not 100% accurate because of values clamping)
newavg = s->mul * curavg;
// Stores computed values for future smoothing
s->lastavg = (1.0 - SMOOTH_LASTAVG) * s->lastavg + SMOOTH_LASTAVG * newavg;
}
static void method1_float(af_volnorm_t *s, af_data_t *c)
{
register int i = 0;
float *data = (float*)c->audio; // Audio data
int len = c->len/4; // Number of samples
float curavg = 0.0, newavg, neededmul, tmp;
for (i = 0; i < len; i++)
{
tmp = data[i];
curavg += tmp * tmp;
}
curavg = sqrt(curavg / (float) len);
// Evaluate an adequate 'mul' coefficient based on previous state, current
// samples level, etc
if (curavg > SIL_FLOAT) // FIXME
{
neededmul = MID_FLOAT / (curavg * s->mul);
s->mul = (1.0 - SMOOTH_MUL) * s->mul + SMOOTH_MUL * neededmul;
// clamp the mul coefficient
s->mul = clamp(s->mul, MUL_MIN, MUL_MAX);
}
// Scale & clamp the samples
for (i = 0; i < len; i++)
data[i] *= s->mul;
// Evaulation of newavg (not 100% accurate because of values clamping)
newavg = s->mul * curavg;
// Stores computed values for future smoothing
s->lastavg = (1.0 - SMOOTH_LASTAVG) * s->lastavg + SMOOTH_LASTAVG * newavg;
}
static void method2_int16(af_volnorm_t *s, af_data_t *c)
{
register int i = 0;
int16_t *data = (int16_t*)c->audio; // Audio data
int len = c->len/2; // Number of samples
float curavg = 0.0, newavg, avg = 0.0;
int tmp, totallen = 0;
for (i = 0; i < len; i++)
{
tmp = data[i];
curavg += tmp * tmp;
}
curavg = sqrt(curavg / (float) len);
// Evaluate an adequate 'mul' coefficient based on previous state, current
// samples level, etc
for (i = 0; i < NSAMPLES; i++)
{
avg += s->mem[i].avg * (float)s->mem[i].len;
totallen += s->mem[i].len;
}
if (totallen > MIN_SAMPLE_SIZE)
{
avg /= (float)totallen;
if (avg >= SIL_S16)
{
s->mul = MID_S16 / avg;
s->mul = clamp(s->mul, MUL_MIN, MUL_MAX);
}
}
// Scale & clamp the samples
for (i = 0; i < len; i++)
{
tmp = s->mul * data[i];
tmp = clamp(tmp, SHRT_MIN, SHRT_MAX);
data[i] = tmp;
}
// Evaulation of newavg (not 100% accurate because of values clamping)
newavg = s->mul * curavg;
// Stores computed values for future smoothing
s->mem[s->idx].len = len;
s->mem[s->idx].avg = newavg;
s->idx = (s->idx + 1) % NSAMPLES;
}
static void method2_float(af_volnorm_t *s, af_data_t *c)
{
register int i = 0;
float *data = (float*)c->audio; // Audio data
int len = c->len/4; // Number of samples
float curavg = 0.0, newavg, avg = 0.0, tmp;
int totallen = 0;
for (i = 0; i < len; i++)
{
tmp = data[i];
curavg += tmp * tmp;
}
curavg = sqrt(curavg / (float) len);
// Evaluate an adequate 'mul' coefficient based on previous state, current
// samples level, etc
for (i = 0; i < NSAMPLES; i++)
{
avg += s->mem[i].avg * (float)s->mem[i].len;
totallen += s->mem[i].len;
}
if (totallen > MIN_SAMPLE_SIZE)
{
avg /= (float)totallen;
if (avg >= SIL_FLOAT)
{
s->mul = MID_FLOAT / avg;
s->mul = clamp(s->mul, MUL_MIN, MUL_MAX);
}
}
// Scale & clamp the samples
for (i = 0; i < len; i++)
data[i] *= s->mul;
// Evaulation of newavg (not 100% accurate because of values clamping)
newavg = s->mul * curavg;
// Stores computed values for future smoothing
s->mem[s->idx].len = len;
s->mem[s->idx].avg = newavg;
s->idx = (s->idx + 1) % NSAMPLES;
}
// Filter data through filter
static af_data_t* play(struct af_instance_s* af, af_data_t* data)
{
af_volnorm_t *s = af->setup;
if(af->data->format == (AF_FORMAT_SI | AF_FORMAT_NE))
{
if (s->method)
method2_int16(s, data);
else
method1_int16(s, data);
}
else if(af->data->format == (AF_FORMAT_F | AF_FORMAT_NE))
{
if (s->method)
method2_float(s, data);
else
method1_float(s, data);
}
return data;
}
// Allocate memory and set function pointers
static int open(af_instance_t* af){
int i = 0;
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_volnorm_t));
if(af->data == NULL || af->setup == NULL)
return AF_ERROR;
((af_volnorm_t*)af->setup)->mul = MUL_INIT;
((af_volnorm_t*)af->setup)->lastavg = MID_S16;
((af_volnorm_t*)af->setup)->idx = 0;
for (i = 0; i < NSAMPLES; i++)
{
((af_volnorm_t*)af->setup)->mem[i].len = 0;
((af_volnorm_t*)af->setup)->mem[i].avg = 0;
}
return AF_OK;
}
// Description of this filter
af_info_t af_info_volnorm = {
"Volume normalizer filter",
"volnorm",
"Alex Beregszaszi & Pierre Lombard",
"",
AF_FLAGS_NOT_REENTRANT,
open
};
Index: Makefile
===================================================================
RCS file: /cvsroot/mplayer/main/libaf/Makefile,v
retrieving revision 1.14
retrieving revision 1.15
diff -u -r1.14 -r1.15
--- Makefile 27 Jun 2004 17:54:31 -0000 1.14
+++ Makefile 4 Oct 2004 19:11:05 -0000 1.15
@@ -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 af_sub.c af_export.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 af_export.c af_volnorm.c af_extrastereo.c
OBJS=$(SRCS:.c=.o)
Index: af.c
===================================================================
RCS file: /cvsroot/mplayer/main/libaf/af.c,v
retrieving revision 1.29
retrieving revision 1.30
diff -u -r1.29 -r1.30
--- af.c 6 Sep 2004 22:27:08 -0000 1.29
+++ af.c 4 Oct 2004 19:11:05 -0000 1.30
@@ -23,6 +23,8 @@
extern af_info_t af_info_surround;
extern af_info_t af_info_sub;
extern af_info_t af_info_export;
+extern af_info_t af_info_volnorm;
+extern af_info_t af_info_extrastereo;
static af_info_t* filter_list[]={
&af_info_dummy,
@@ -40,6 +42,8 @@
#ifdef HAVE_SYS_MMAN_H
&af_info_export,
#endif
+ &af_info_volnorm,
+ &af_info_extrastereo,
NULL
};
Index: af_format.h
===================================================================
RCS file: /cvsroot/mplayer/main/libaf/af_format.h,v
retrieving revision 1.2
retrieving revision 1.3
diff -u -r1.2 -r1.3
--- af_format.h 18 Jan 2003 17:31:58 -0000 1.2
+++ af_format.h 4 Oct 2004 19:11:05 -0000 1.3
@@ -18,7 +18,7 @@
#define AF_FORMAT_US (1<<1) // Un Signed
#define AF_FORMAT_SIGN_MASK (1<<1)
-// Fixed of floating point
+// Fixed or floating point
#define AF_FORMAT_I (0<<2) // Int
#define AF_FORMAT_F (1<<2) // Foating point
#define AF_FORMAT_POINT_MASK (1<<2)
Index: control.h
===================================================================
RCS file: /cvsroot/mplayer/main/libaf/control.h,v
retrieving revision 1.6
retrieving revision 1.7
diff -u -r1.6 -r1.7
--- control.h 20 Sep 2003 13:42:26 -0000 1.6
+++ control.h 4 Oct 2004 19:11:05 -0000 1.7
@@ -218,4 +218,8 @@
// Export
#define AF_CONTROL_EXPORT_SZ 0x00002000 | AF_CONTROL_FILTER_SPECIFIC
+
+// ExtraStereo Multiplier
+#define AF_CONTROL_ES_MUL 0x00002100 | AF_CONTROL_FILTER_SPECIFIC
+
#endif /*__af_control_h */
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