gstreamer/gst/audioresample/resample_chunk.c
Thomas Vander Stichele c4763dbb33 port audioresample to basetransform
Original commit message from CVS:
port audioresample to basetransform
2005-08-24 14:08:58 +00:00

210 lines
5.7 KiB
C

/* 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.
*/
#ifdef HAVE_CONFIG_H
#include <config.h>
#endif
#include <string.h>
#include <math.h>
#include <stdio.h>
#include <stdlib.h>
#include <limits.h>
#include <liboil/liboil.h>
#include "resample.h"
#include "buffer.h"
#include "debug.h"
static double
resample_sinc_window (double x, double halfwidth, double scale)
{
double y;
if (x == 0)
return 1.0;
if (x < -halfwidth || x > halfwidth)
return 0.0;
y = sin (x * M_PI * scale) / (x * M_PI * scale) * scale;
x /= halfwidth;
y *= (1 - x * x) * (1 - x * x);
return y;
}
void
resample_scale_chunk (ResampleState * r)
{
if (r->need_reinit) {
r->sample_size = r->n_channels * resample_format_size (r->format);
RESAMPLE_DEBUG ("sample size %d", r->sample_size);
if (r->buffer)
free (r->buffer);
r->buffer_len = r->sample_size * 1000;
r->buffer = malloc (r->buffer_len);
memset (r->buffer, 0, r->buffer_len);
r->i_inc = r->o_rate / r->i_rate;
r->o_inc = r->i_rate / r->o_rate;
RESAMPLE_DEBUG ("i_inc %g o_inc %g", r->i_inc, r->o_inc);
r->i_start = -r->i_inc * r->filter_length;
r->need_reinit = 0;
#if 0
if (r->i_inc < 1.0) {
r->sinc_scale = r->i_inc;
if (r->sinc_scale == 0.5) {
/* strange things happen at integer multiples */
r->sinc_scale = 1.0;
}
} else {
r->sinc_scale = 1.0;
}
#else
r->sinc_scale = 1.0;
#endif
}
while (r->o_size > 0) {
double midpoint;
int i;
int j;
RESAMPLE_DEBUG ("i_start %g", r->i_start);
midpoint = r->i_start + (r->filter_length - 1) * 0.5 * r->i_inc;
if (midpoint > 0.5 * r->i_inc) {
RESAMPLE_ERROR ("inconsistent state");
}
while (midpoint < -0.5 * r->i_inc) {
AudioresampleBuffer *buffer;
buffer = audioresample_buffer_queue_pull (r->queue, r->sample_size);
if (buffer == NULL) {
RESAMPLE_ERROR ("buffer_queue_pull returned NULL");
return;
}
r->i_start += r->i_inc;
RESAMPLE_DEBUG ("pulling (i_start = %g)", r->i_start);
midpoint += r->i_inc;
memmove (r->buffer, r->buffer + r->sample_size,
r->buffer_len - r->sample_size);
memcpy (r->buffer + r->buffer_len - r->sample_size, buffer->data,
r->sample_size);
audioresample_buffer_unref (buffer);
}
switch (r->format) {
case RESAMPLE_FORMAT_S16:
for (i = 0; i < r->n_channels; i++) {
double acc = 0;
double offset;
double x;
for (j = 0; j < r->filter_length; j++) {
offset = (r->i_start + j * r->i_inc) * r->o_inc;
x = *(int16_t *) (r->buffer + i * sizeof (int16_t) +
j * r->sample_size);
acc +=
resample_sinc_window (offset, r->filter_length * 0.5,
r->sinc_scale) * x;
}
if (acc < -32768.0)
acc = -32768.0;
if (acc > 32767.0)
acc = 32767.0;
*(int16_t *) (r->o_buf + i * sizeof (int16_t)) = rint (acc);
}
break;
case RESAMPLE_FORMAT_S32:
for (i = 0; i < r->n_channels; i++) {
double acc = 0;
double offset;
double x;
for (j = 0; j < r->filter_length; j++) {
offset = (r->i_start + j * r->i_inc) * r->o_inc;
x = *(int32_t *) (r->buffer + i * sizeof (int32_t) +
j * r->sample_size);
acc +=
resample_sinc_window (offset, r->filter_length * 0.5,
r->sinc_scale) * x;
}
if (acc < -2147483648.0)
acc = -2147483648.0;
if (acc > 2147483647.0)
acc = 2147483647.0;
*(int32_t *) (r->o_buf + i * sizeof (int32_t)) = rint (acc);
}
break;
case RESAMPLE_FORMAT_F32:
for (i = 0; i < r->n_channels; i++) {
double acc = 0;
double offset;
double x;
for (j = 0; j < r->filter_length; j++) {
offset = (r->i_start + j * r->i_inc) * r->o_inc;
x = *(float *) (r->buffer + i * sizeof (float) +
j * r->sample_size);
acc +=
resample_sinc_window (offset, r->filter_length * 0.5,
r->sinc_scale) * x;
}
*(float *) (r->o_buf + i * sizeof (float)) = acc;
}
break;
case RESAMPLE_FORMAT_F64:
for (i = 0; i < r->n_channels; i++) {
double acc = 0;
double offset;
double x;
for (j = 0; j < r->filter_length; j++) {
offset = (r->i_start + j * r->i_inc) * r->o_inc;
x = *(double *) (r->buffer + i * sizeof (double) +
j * r->sample_size);
acc +=
resample_sinc_window (offset, r->filter_length * 0.5,
r->sinc_scale) * x;
}
*(double *) (r->o_buf + i * sizeof (double)) = acc;
}
break;
}
r->i_start -= 1.0;
r->o_buf += r->sample_size;
r->o_size -= r->sample_size;
}
}