gstreamer/gst-libs/gst/resample/resample.c

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/* Resampling library
* Copyright (C) <2001> David A. Schleef <ds@schleef.org>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#include <string.h>
#include <math.h>
#include <stdio.h>
#include <stdlib.h>
#include <resample.h>
#include <gst/gstplugin.h>
#include <gst/gstversion.h>
inline double sinc(double x)
{
if(x==0)return 1;
return sin(x) / x;
}
inline double window_func(double x)
{
x = 1 - x*x;
return x*x;
}
signed short double_to_s16(double x)
{
if(x<-32768){
printf("clipped\n");
return -32768;
}
if(x>32767){
printf("clipped\n");
return -32767;
}
return rint(x);
}
signed short double_to_s16_ppcasm(double x)
{
if(x<-32768){
return -32768;
}
if(x>32767){
return -32767;
}
return rint(x);
}
static void resample_sinc_ft(resample_t * r);
void resample_init(resample_t * r)
{
r->i_start = 0;
if(r->filter_length&1){
r->o_start = 0;
}else{
r->o_start = r->o_inc * 0.5;
}
memset(r->acc, 0, sizeof(r->acc));
resample_reinit(r);
}
void resample_reinit(resample_t * r)
{
/* i_inc is the number of samples that the output increments for
* each input sample. o_inc is the opposite. */
r->i_inc = (double) r->o_rate / r->i_rate;
r->o_inc = (double) r->i_rate / r->o_rate;
r->halftaps = (r->filter_length - 1.0) * 0.5;
switch (r->method) {
default:
case RESAMPLE_NEAREST:
r->scale = resample_nearest;
break;
case RESAMPLE_BILINEAR:
r->scale = resample_bilinear;
break;
case RESAMPLE_SINC_SLOW:
r->scale = resample_sinc;
break;
case RESAMPLE_SINC:
r->scale = resample_sinc_ft;
break;
}
}
/*
* Prepare to be confused.
*
* We keep a "timebase" that is based on output samples. The zero
* of the timebase cooresponds to the next output sample that will
* be written.
*
* i_start is the "time" that corresponds to the first input sample
* in an incoming buffer. Since the output depends on input samples
* ahead in time, i_start will tend to be around halftaps.
*
* i_start_buf is the time of the first sample in the temporary
* buffer.
*/
void resample_scale(resample_t * r, void *i_buf, unsigned int i_size)
{
int o_size;
r->i_buf = i_buf;
r->i_samples = i_size / 2 / r->channels;
r->i_start_buf = r->i_start - r->filter_length * r->i_inc;
/* i_start is the offset (in a given output sample) that is the
* beginning of the current input buffer */
r->i_end = r->i_start + r->i_inc * r->i_samples;
r->o_samples = floor(r->i_end - r->halftaps * r->i_inc);
o_size = r->o_samples * r->channels * 2;
r->o_buf = r->get_buffer(r->priv, o_size);
if(r->verbose){
printf("resample_scale: i_buf=%p i_size=%d\n",
i_buf,i_size);
printf("resample_scale: i_samples=%d o_samples=%d i_inc=%g o_buf=%p\n",
r->i_samples, r->o_samples, r->i_inc, r->o_buf);
printf("resample_scale: i_start=%g i_end=%g o_start=%g\n",
r->i_start, r->i_end, r->o_start);
}
if ((r->filter_length + r->i_samples)*2*2 > r->buffer_len) {
int size = (r->filter_length + r->i_samples) * sizeof(double) * 2;
if(r->verbose){
printf("resample temp buffer size=%d\n",size);
}
if(r->buffer)free(r->buffer);
r->buffer_len = size;
r->buffer = malloc(size);
memset(r->buffer, 0, size);
}
if(r->channels==2){
conv_double_short(
r->buffer + r->filter_length * sizeof(double) * 2,
r->i_buf, r->i_samples * 2);
}else{
conv_double_short_dstr(
r->buffer + r->filter_length * sizeof(double) * 2,
r->i_buf, r->i_samples, sizeof(double) * 2);
}
r->scale(r);
memcpy(r->buffer,
r->buffer + r->i_samples * sizeof(double) * 2,
r->filter_length * sizeof(double) * 2);
/* updating times */
r->i_start += r->i_samples * r->i_inc;
r->o_start += r->o_samples * r->o_inc - r->i_samples;
/* adjusting timebase zero */
r->i_start -= r->o_samples;
}
void resample_nearest(resample_t * r)
{
signed short *i_ptr, *o_ptr;
int i_count = 0;
double a;
int i;
i_ptr = (signed short *) r->i_buf;
o_ptr = (signed short *) r->o_buf;
a = r->o_start;
i_count = 0;
#define SCALE_LOOP(COPY,INC) \
for (i = 0; i < r->o_samples; i++) { \
COPY; \
a += r->o_inc; \
while (a >= 1) { \
a -= 1; \
i_ptr+=INC; \
i_count++; \
} \
o_ptr+=INC; \
}
switch (r->channels) {
case 1:
SCALE_LOOP(o_ptr[0] = i_ptr[0], 1);
break;
case 2:
SCALE_LOOP(o_ptr[0] = i_ptr[0];
o_ptr[1] = i_ptr[1], 2);
break;
default:
{
int n, n_chan = r->channels;
SCALE_LOOP(for (n = 0; n < n_chan; n++) o_ptr[n] =
i_ptr[n], n_chan);
}
}
if (i_count != r->i_samples) {
printf("handled %d in samples (expected %d)\n", i_count,
r->i_samples);
}
}
void resample_bilinear(resample_t * r)
{
signed short *i_ptr, *o_ptr;
int o_count = 0;
double b;
int i;
double acc0, acc1;
i_ptr = (signed short *) r->i_buf;
o_ptr = (signed short *) r->o_buf;
acc0 = r->acc[0];
acc1 = r->acc[1];
b = r->i_start;
for (i = 0; i < r->i_samples; i++) {
b += r->i_inc;
/*printf("in %d\n",i_ptr[0]); */
if(b>=2){
printf("not expecting b>=2\n");
}
if (b >= 1) {
acc0 += (1.0 - (b-r->i_inc)) * i_ptr[0];
acc1 += (1.0 - (b-r->i_inc)) * i_ptr[1];
o_ptr[0] = rint(acc0);
/*printf("out %d\n",o_ptr[0]); */
o_ptr[1] = rint(acc1);
o_ptr += 2;
o_count++;
b -= 1.0;
acc0 = b * i_ptr[0];
acc1 = b * i_ptr[1];
} else {
acc0 += i_ptr[0] * r->i_inc;
acc1 += i_ptr[1] * r->i_inc;
}
i_ptr += 2;
}
r->acc[0] = acc0;
r->acc[1] = acc1;
if (o_count != r->o_samples) {
printf("handled %d out samples (expected %d)\n", o_count,
r->o_samples);
}
}
void resample_sinc_slow(resample_t * r)
{
signed short *i_ptr, *o_ptr;
int i, j;
double c0, c1;
double a;
int start;
double center;
double weight;
if (!r->buffer) {
int size = r->filter_length * 2 * r->channels;
printf("resample temp buffer\n");
r->buffer = malloc(size);
memset(r->buffer, 0, size);
}
i_ptr = (signed short *) r->i_buf;
o_ptr = (signed short *) r->o_buf;
a = r->i_start;
#define GETBUF(index,chan) (((index)<0) \
? ((short *)(r->buffer))[((index)+r->filter_length)*2+(chan)] \
: i_ptr[(index)*2+(chan)])
{
double sinx, cosx, sind, cosd;
double x, d;
double t;
for (i = 0; i < r->o_samples; i++) {
start = floor(a) - r->filter_length;
center = a - r->halftaps;
x = M_PI * (start - center) * r->o_inc;
sinx = sin(M_PI * (start - center) * r->o_inc);
cosx = cos(M_PI * (start - center) * r->o_inc);
d = M_PI * r->o_inc;
sind = sin(M_PI * r->o_inc);
cosd = cos(M_PI * r->o_inc);
c0 = 0;
c1 = 0;
for (j = 0; j < r->filter_length; j++) {
weight = (x==0)?1:(sinx/x);
/*printf("j %d sin %g cos %g\n",j,sinx,cosx); */
/*printf("j %d sin %g x %g sinc %g\n",j,sinx,x,weight); */
c0 += weight * GETBUF((start + j), 0);
c1 += weight * GETBUF((start + j), 1);
t = cosx * cosd - sinx * sind;
sinx = cosx * sind + sinx * cosd;
cosx = t;
x += d;
}
o_ptr[0] = rint(c0);
o_ptr[1] = rint(c1);
o_ptr += 2;
a += r->o_inc;
}
}
memcpy(r->buffer,
i_ptr + (r->i_samples - r->filter_length) * r->channels,
r->filter_length * 2 * r->channels);
}
void resample_sinc(resample_t * r)
{
double *ptr;
signed short *o_ptr;
int i, j;
double c0, c1;
double a;
int start;
double center;
double weight;
double x0, x, d;
double scale;
ptr = (double *) r->buffer;
o_ptr = (signed short *) r->o_buf;
/* scale provides a cutoff frequency for the low
* pass filter aspects of sinc(). scale=M_PI
* will cut off at the input frequency, which is
* good for up-sampling, but will cause aliasing
* for downsampling. Downsampling needs to be
* cut off at o_rate, thus scale=M_PI*r->i_inc. */
/* actually, it needs to be M_PI*r->i_inc*r->i_inc.
* Need to research why. */
scale = M_PI*r->i_inc;
for (i = 0; i < r->o_samples; i++) {
a = r->o_start + i * r->o_inc;
start = floor(a - r->halftaps);
/*printf("%d: a=%g start=%d end=%d\n",i,a,start,start+r->filter_length-1); */
center = a;
/*x = M_PI * (start - center) * r->o_inc; */
/*d = M_PI * r->o_inc; */
/*x = (start - center) * r->o_inc; */
x0 = (start - center) * r->o_inc;
d = r->o_inc;
c0 = 0;
c1 = 0;
for (j = 0; j < r->filter_length; j++) {
x = x0 + d * j;
weight = sinc(x*scale*r->i_inc)*scale/M_PI;
weight *= window_func(x/r->halftaps*r->i_inc);
c0 += weight * ptr[(start + j + r->filter_length)*2 + 0];
c1 += weight * ptr[(start + j + r->filter_length)*2 + 1];
}
o_ptr[0] = double_to_s16(c0);
o_ptr[1] = double_to_s16(c1);
o_ptr += 2;
}
}
/*
* Resampling audio is best done using a sinc() filter.
*
*
* out[t] = Sum( in[t'] * sinc((t-t')/delta_t), all t')
*
* The immediate problem with this algorithm is that it involves a
* sum over an infinite number of input samples, both in the past
* and future. Note that even though sinc(x) is bounded by 1/x,
* and thus decays to 0 for large x, since sum(x,{x=0,1..,n}) diverges
* as log(n), we need to be careful about convergence. This is
* typically done by using a windowing function, which also makes
* the sum over a finite number of input samples.
*
* The next problem is computational: sinc(), and especially
* sinc() multiplied by a non-trivial windowing function is expensive
* to calculate, and also difficult to find SIMD optimizations. Since
* the time increment on input and output is different, it is not
* possible to use a FIR filter, because the taps would have to be
* recalculated for every t.
*
* To get around the expense of calculating sinc() for every point,
* we pre-calculate sinc() at a number of points, and then interpolate
* for the values we want in calculations. The interpolation method
* chosen is bi-cubic, which requires both the evalated function and
* its derivative at every pre-sampled point. Also, if the sampled
* points are spaced commensurate with the input delta_t, we notice
* that the interpolating weights are the same for every input point.
* This decreases the number of operations to 4 multiplies and 4 adds
* for each tap, regardless of the complexity of the filtering function.
*
* At this point, it is possible to rearrange the problem as the sum
* of 4 properly weghted FIR filters. Typical SIMD computation units
* are highly optimized for FIR filters, making long filter lengths
* reasonable.
*/
static functable_t *ft;
double out_tmp[10000];
static void resample_sinc_ft(resample_t * r)
{
double *ptr;
signed short *o_ptr;
int i;
/*int j; */
double c0, c1;
/*double a; */
double start_f, start_x;
int start;
double center;
/*double weight; */
double x, d;
double scale;
int n = 4;
scale = r->i_inc; /* cutoff at 22050 */
/*scale = 1.0; // cutoff at 24000 */
/*scale = r->i_inc * 0.5; // cutoff at 11025 */
if(!ft){
ft = malloc(sizeof(*ft));
memset(ft,0,sizeof(*ft));
ft->len = (r->filter_length + 2) * n;
ft->offset = 1.0 / n;
ft->start = - ft->len * 0.5 * ft->offset;
ft->func_x = functable_sinc;
ft->func_dx = functable_dsinc;
ft->scale = M_PI * scale;
ft->func2_x = functable_window_std;
ft->func2_dx = functable_window_dstd;
ft->scale2 = 1.0 / r->halftaps;
functable_init(ft);
/*printf("len=%d offset=%g start=%g\n",ft->len,ft->offset,ft->start); */
}
ptr = r->buffer;
o_ptr = (signed short *) r->o_buf;
center = r->o_start;
start_x = center - r->halftaps;
start_f = floor(start_x);
start_x -= start_f;
start = start_f;
for (i = 0; i < r->o_samples; i++) {
/*start_f = floor(center - r->halftaps); */
/*printf("%d: a=%g start=%d end=%d\n",i,a,start,start+r->filter_length-1); */
x = start_f - center;
d = 1;
c0 = 0;
c1 = 0;
/*#define slow */
#ifdef slow
for (j = 0; j < r->filter_length; j++) {
weight = functable_eval(ft,x)*scale;
/*weight = sinc(M_PI * scale * x)*scale*r->i_inc; */
/*weight *= window_func(x / r->halftaps); */
c0 += weight * ptr[(start + j + r->filter_length)*2 + 0];
c1 += weight * ptr[(start + j + r->filter_length)*2 + 1];
x += d;
}
#else
functable_fir2(ft,
&c0,&c1,
x, n,
ptr+(start + r->filter_length)*2,
r->filter_length);
c0 *= scale;
c1 *= scale;
#endif
out_tmp[2 * i + 0] = c0;
out_tmp[2 * i + 1] = c1;
center += r->o_inc;
start_x += r->o_inc;
while(start_x>=1.0){
start_f++;
start_x -= 1.0;
start++;
}
}
if(r->channels==2){
conv_short_double(r->o_buf,out_tmp,2 * r->o_samples);
}else{
conv_short_double_sstr(r->o_buf,out_tmp,r->o_samples,2 * sizeof(double));
}
}
static gboolean
plugin_init (GModule *module, GstPlugin *plugin)
{
gst_plugin_set_longname (plugin, "Resampling routines for use in audio plugins");
return TRUE;
}
GstPluginDesc plugin_desc = {
GST_VERSION_MAJOR,
GST_VERSION_MINOR,
"gstresample",
plugin_init
};