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gst/audioresample/Makefile.am: Leet audioresampling code

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Original commit message from CVS:
* gst/audioresample/Makefile.am: Leet audioresampling code
* gst/audioresample/buffer.c:
* gst/audioresample/buffer.h:
* gst/audioresample/debug.c:
* gst/audioresample/debug.h:
* gst/audioresample/functable.c:
* gst/audioresample/functable.h:
* gst/audioresample/gstaudioresample.c:
* gst/audioresample/gstaudioresample.h:
* gst/audioresample/resample.c:
* gst/audioresample/resample.h:
* gst/audioresample/resample_chunk.c:
* gst/audioresample/resample_functable.c:
* gst/audioresample/resample_ref.c:
This commit is contained in:
David Schleef 2005-08-23 19:29:38 +00:00
parent 84d0eb4f88
commit ae8f41b658
15 changed files with 2271 additions and 0 deletions
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17
ChangeLog
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2005-08-23 David Schleef <ds@schleef.org>
* gst/audioresample/Makefile.am: Leet audioresampling code
* gst/audioresample/buffer.c:
* gst/audioresample/buffer.h:
* gst/audioresample/debug.c:
* gst/audioresample/debug.h:
* gst/audioresample/functable.c:
* gst/audioresample/functable.h:
* gst/audioresample/gstaudioresample.c:
* gst/audioresample/gstaudioresample.h:
* gst/audioresample/resample.c:
* gst/audioresample/resample.h:
* gst/audioresample/resample_chunk.c:
* gst/audioresample/resample_functable.c:
* gst/audioresample/resample_ref.c:
2005-08-23 Wim Taymans <wim@fluendo.com> 2005-08-23 Wim Taymans <wim@fluendo.com>
* examples/seeking/seek.c: (make_vorbis_pipeline), * examples/seeking/seek.c: (make_vorbis_pipeline),

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gst/audioresample/Makefile.am Normal file
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plugin_LTLIBRARIES = libgstaudioresample.la
resample_SOURCES = \
functable.c \
functable.h \
resample.c \
resample_functable.c \
resample_ref.c \
resample_chunk.c \
resample.h \
debug.c \
debug.h \
buffer.c \
buffer.h
libgstaudioresample_la_SOURCES = gstaudioresample.c $(resample_SOURCES)
libgstaudioresample_la_CFLAGS = $(GST_CFLAGS) $(LIBOIL_CFLAGS)
libgstaudioresample_la_LIBADD = $(LIBOIL_LIBS)
libgstaudioresample_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)

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gst/audioresample/buffer.c Normal file
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#ifndef HAVE_CONFIG_H
#include "config.h"
#endif
#include <audioresample/buffer.h>
#include <glib.h>
#include <string.h>
#include <audioresample/debug.h>
static void audioresample_buffer_free_mem (AudioresampleBuffer * buffer,
void *);
static void audioresample_buffer_free_subbuffer (AudioresampleBuffer * buffer,
void *priv);
AudioresampleBuffer *
audioresample_buffer_new (void)
{
AudioresampleBuffer *buffer;
buffer = g_new0 (AudioresampleBuffer, 1);
buffer->ref_count = 1;
return buffer;
}
AudioresampleBuffer *
audioresample_buffer_new_and_alloc (int size)
{
AudioresampleBuffer *buffer = audioresample_buffer_new ();
buffer->data = g_malloc (size);
buffer->length = size;
buffer->free = audioresample_buffer_free_mem;
return buffer;
}
AudioresampleBuffer *
audioresample_buffer_new_with_data (void *data, int size)
{
AudioresampleBuffer *buffer = audioresample_buffer_new ();
buffer->data = data;
buffer->length = size;
buffer->free = audioresample_buffer_free_mem;
return buffer;
}
AudioresampleBuffer *
audioresample_buffer_new_subbuffer (AudioresampleBuffer * buffer, int offset,
int length)
{
AudioresampleBuffer *subbuffer = audioresample_buffer_new ();
if (buffer->parent) {
audioresample_buffer_ref (buffer->parent);
subbuffer->parent = buffer->parent;
} else {
audioresample_buffer_ref (buffer);
subbuffer->parent = buffer;
}
subbuffer->data = buffer->data + offset;
subbuffer->length = length;
subbuffer->free = audioresample_buffer_free_subbuffer;
return subbuffer;
}
void
audioresample_buffer_ref (AudioresampleBuffer * buffer)
{
buffer->ref_count++;
}
void
audioresample_buffer_unref (AudioresampleBuffer * buffer)
{
buffer->ref_count--;
if (buffer->ref_count == 0) {
if (buffer->free)
buffer->free (buffer, buffer->priv);
g_free (buffer);
}
}
static void
audioresample_buffer_free_mem (AudioresampleBuffer * buffer, void *priv)
{
g_free (buffer->data);
}
static void
audioresample_buffer_free_subbuffer (AudioresampleBuffer * buffer, void *priv)
{
audioresample_buffer_unref (buffer->parent);
}
AudioresampleBufferQueue *
audioresample_buffer_queue_new (void)
{
return g_new0 (AudioresampleBufferQueue, 1);
}
int
audioresample_buffer_queue_get_depth (AudioresampleBufferQueue * queue)
{
return queue->depth;
}
int
audioresample_buffer_queue_get_offset (AudioresampleBufferQueue * queue)
{
return queue->offset;
}
void
audioresample_buffer_queue_free (AudioresampleBufferQueue * queue)
{
GList *g;
for (g = g_list_first (queue->buffers); g; g = g_list_next (g)) {
audioresample_buffer_unref ((AudioresampleBuffer *) g->data);
}
g_list_free (queue->buffers);
g_free (queue);
}
void
audioresample_buffer_queue_push (AudioresampleBufferQueue * queue,
AudioresampleBuffer * buffer)
{
queue->buffers = g_list_append (queue->buffers, buffer);
queue->depth += buffer->length;
}
AudioresampleBuffer *
audioresample_buffer_queue_pull (AudioresampleBufferQueue * queue, int length)
{
GList *g;
AudioresampleBuffer *newbuffer;
AudioresampleBuffer *buffer;
AudioresampleBuffer *subbuffer;
g_return_val_if_fail (length > 0, NULL);
if (queue->depth < length) {
return NULL;
}
RESAMPLE_LOG ("pulling %d, %d available", length, queue->depth);
g = g_list_first (queue->buffers);
buffer = g->data;
if (buffer->length > length) {
newbuffer = audioresample_buffer_new_subbuffer (buffer, 0, length);
subbuffer = audioresample_buffer_new_subbuffer (buffer, length,
buffer->length - length);
g->data = subbuffer;
audioresample_buffer_unref (buffer);
} else {
int offset = 0;
newbuffer = audioresample_buffer_new_and_alloc (length);
while (offset < length) {
g = g_list_first (queue->buffers);
buffer = g->data;
if (buffer->length > length - offset) {
int n = length - offset;
memcpy (newbuffer->data + offset, buffer->data, n);
subbuffer =
audioresample_buffer_new_subbuffer (buffer, n, buffer->length - n);
g->data = subbuffer;
audioresample_buffer_unref (buffer);
offset += n;
} else {
memcpy (newbuffer->data + offset, buffer->data, buffer->length);
queue->buffers = g_list_delete_link (queue->buffers, g);
offset += buffer->length;
audioresample_buffer_unref (buffer);
}
}
}
queue->depth -= length;
queue->offset += length;
return newbuffer;
}
AudioresampleBuffer *
audioresample_buffer_queue_peek (AudioresampleBufferQueue * queue, int length)
{
GList *g;
AudioresampleBuffer *newbuffer;
AudioresampleBuffer *buffer;
int offset = 0;
g_return_val_if_fail (length > 0, NULL);
if (queue->depth < length) {
return NULL;
}
RESAMPLE_LOG ("peeking %d, %d available", length, queue->depth);
g = g_list_first (queue->buffers);
buffer = g->data;
if (buffer->length > length) {
newbuffer = audioresample_buffer_new_subbuffer (buffer, 0, length);
} else {
newbuffer = audioresample_buffer_new_and_alloc (length);
while (offset < length) {
buffer = g->data;
if (buffer->length > length - offset) {
int n = length - offset;
memcpy (newbuffer->data + offset, buffer->data, n);
offset += n;
} else {
memcpy (newbuffer->data + offset, buffer->data, buffer->length);
offset += buffer->length;
}
g = g_list_next (g);
}
}
return newbuffer;
}

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#ifndef __AUDIORESAMPLE_BUFFER_H__
#define __AUDIORESAMPLE_BUFFER_H__
#include <glib.h>
typedef struct _AudioresampleBuffer AudioresampleBuffer;
typedef struct _AudioresampleBufferQueue AudioresampleBufferQueue;
struct _AudioresampleBuffer
{
unsigned char *data;
int length;
int ref_count;
AudioresampleBuffer *parent;
void (*free) (AudioresampleBuffer *, void *);
void *priv;
void *priv2;
};
struct _AudioresampleBufferQueue
{
GList *buffers;
int depth;
int offset;
};
AudioresampleBuffer *audioresample_buffer_new (void);
AudioresampleBuffer *audioresample_buffer_new_and_alloc (int size);
AudioresampleBuffer *audioresample_buffer_new_with_data (void *data, int size);
AudioresampleBuffer *audioresample_buffer_new_subbuffer (AudioresampleBuffer * buffer, int offset,
int length);
void audioresample_buffer_ref (AudioresampleBuffer * buffer);
void audioresample_buffer_unref (AudioresampleBuffer * buffer);
AudioresampleBufferQueue *audioresample_buffer_queue_new (void);
void audioresample_buffer_queue_free (AudioresampleBufferQueue * queue);
int audioresample_buffer_queue_get_depth (AudioresampleBufferQueue * queue);
int audioresample_buffer_queue_get_offset (AudioresampleBufferQueue * queue);
void audioresample_buffer_queue_push (AudioresampleBufferQueue * queue,
AudioresampleBuffer * buffer);
AudioresampleBuffer *audioresample_buffer_queue_pull (AudioresampleBufferQueue * queue, int len);
AudioresampleBuffer *audioresample_buffer_queue_peek (AudioresampleBufferQueue * queue, int len);
#endif

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#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <glib.h>
#include <stdio.h>
#include <debug.h>
static const char *resample_debug_level_names[] = {
"NONE",
"ERROR",
"WARNING",
"INFO",
"DEBUG",
"LOG"
};
static int resample_debug_level = RESAMPLE_LEVEL_ERROR;
void
resample_debug_log (int level, const char *file, const char *function,
int line, const char *format, ...)
{
#ifndef GLIB_COMPAT
va_list varargs;
char *s;
if (level > resample_debug_level)
return;
va_start (varargs, format);
s = g_strdup_vprintf (format, varargs);
va_end (varargs);
fprintf (stderr, "RESAMPLE: %s: %s(%d): %s: %s\n",
resample_debug_level_names[level], file, line, function, s);
g_free (s);
#else
va_list varargs;
char s[1000];
if (level > resample_debug_level)
return;
va_start (varargs, format);
vsnprintf (s, 999, format, varargs);
va_end (varargs);
fprintf (stderr, "RESAMPLE: %s: %s(%d): %s: %s\n",
resample_debug_level_names[level], file, line, function, s);
#endif
}
void
resample_debug_set_level (int level)
{
resample_debug_level = level;
}
int
resample_debug_get_level (void)
{
return resample_debug_level;
}

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#ifndef __RESAMPLE_DEBUG_H__
#define __RESAMPLE_DEBUG_H__
enum
{
RESAMPLE_LEVEL_NONE = 0,
RESAMPLE_LEVEL_ERROR,
RESAMPLE_LEVEL_WARNING,
RESAMPLE_LEVEL_INFO,
RESAMPLE_LEVEL_DEBUG,
RESAMPLE_LEVEL_LOG
};
#define RESAMPLE_ERROR(...) \
RESAMPLE_DEBUG_LEVEL(RESAMPLE_LEVEL_ERROR, __VA_ARGS__)
#define RESAMPLE_WARNING(...) \
RESAMPLE_DEBUG_LEVEL(RESAMPLE_LEVEL_WARNING, __VA_ARGS__)
#define RESAMPLE_INFO(...) \
RESAMPLE_DEBUG_LEVEL(RESAMPLE_LEVEL_INFO, __VA_ARGS__)
#define RESAMPLE_DEBUG(...) \
RESAMPLE_DEBUG_LEVEL(RESAMPLE_LEVEL_DEBUG, __VA_ARGS__)
#define RESAMPLE_LOG(...) \
RESAMPLE_DEBUG_LEVEL(RESAMPLE_LEVEL_LOG, __VA_ARGS__)
#define RESAMPLE_DEBUG_LEVEL(level,...) \
resample_debug_log ((level), __FILE__, __FUNCTION__, __LINE__, __VA_ARGS__)
void resample_debug_log (int level, const char *file, const char *function,
int line, const char *format, ...);
void resample_debug_set_level (int level);
int resample_debug_get_level (void);
#endif

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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.
*/
#ifdef HAVE_CONFIG_H
#include <config.h>
#endif
#include <string.h>
#include <math.h>
#include <stdio.h>
#include <stdlib.h>
#include <audioresample/functable.h>
#include <audioresample/debug.h>
void
functable_func_sinc (double *fx, double *dfx, double x, void *closure)
{
if (x == 0) {
*fx = 1;
*dfx = 0;
return;
}
*fx = sin (x) / x;
*dfx = (cos (x) - sin (x) / x) / x;
}
void
functable_func_boxcar (double *fx, double *dfx, double x, void *closure)
{
double width = *(double *) closure;
if (x < width && x > -width) {
*fx = 1;
} else {
*fx = 0;
}
*dfx = 0;
}
void
functable_func_hanning (double *fx, double *dfx, double x, void *closure)
{
double width = *(double *) closure;
if (x < width && x > -width) {
x /= width;
*fx = (1 - x * x) * (1 - x * x);
*dfx = -2 * 2 * x / width * (1 - x * x);
} else {
*fx = 0;
*dfx = 0;
}
}
Functable *
functable_new (void)
{
Functable *ft;
ft = malloc (sizeof (Functable));
memset (ft, 0, sizeof (Functable));
return ft;
}
void
functable_free (Functable * ft)
{
free (ft);
}
void
functable_set_length (Functable * t, int length)
{
t->length = length;
}
void
functable_set_offset (Functable * t, double offset)
{
t->offset = offset;
}
void
functable_set_multiplier (Functable * t, double multiplier)
{
t->multiplier = multiplier;
}
void
functable_calculate (Functable * t, FunctableFunc func, void *closure)
{
int i;
double x;
if (t->fx)
free (t->fx);
if (t->dfx)
free (t->dfx);
t->fx = malloc (sizeof (double) * (t->length + 1));
t->dfx = malloc (sizeof (double) * (t->length + 1));
t->inv_multiplier = 1.0 / t->multiplier;
for (i = 0; i < t->length + 1; i++) {
x = t->offset + t->multiplier * i;
func (&t->fx[i], &t->dfx[i], x, closure);
}
}
void
functable_calculate_multiply (Functable * t, FunctableFunc func, void *closure)
{
int i;
double x;
for (i = 0; i < t->length + 1; i++) {
double afx, adfx, bfx, bdfx;
afx = t->fx[i];
adfx = t->dfx[i];
x = t->offset + t->multiplier * i;
func (&bfx, &bdfx, x, closure);
t->fx[i] = afx * bfx;
t->dfx[i] = afx * bdfx + adfx * bfx;
}
}
double
functable_evaluate (Functable * t, double x)
{
int i;
double f0, f1, w0, w1;
double x2, x3;
double w;
if (x < t->offset || x > (t->offset + t->length * t->multiplier)) {
RESAMPLE_DEBUG ("x out of range %g", x);
}
x -= t->offset;
x *= t->inv_multiplier;
i = floor (x);
x -= i;
x2 = x * x;
x3 = x2 * x;
f1 = 3 * x2 - 2 * x3;
f0 = 1 - f1;
w0 = (x - 2 * x2 + x3) * t->multiplier;
w1 = (-x2 + x3) * t->multiplier;
w = t->fx[i] * f0 + t->fx[i + 1] * f1 + t->dfx[i] * w0 + t->dfx[i + 1] * w1;
/*w = t->fx[i] * (1-x) + t->fx[i+1] * x; */
return w;
}
double
functable_fir (Functable * t, double x, int n, double *data, int len)
{
int i, j;
double f0, f1, w0, w1;
double x2, x3;
double w;
double sum;
x -= t->offset;
x /= t->multiplier;
i = floor (x);
x -= i;
x2 = x * x;
x3 = x2 * x;
f1 = 3 * x2 - 2 * x3;
f0 = 1 - f1;
w0 = (x - 2 * x2 + x3) * t->multiplier;
w1 = (-x2 + x3) * t->multiplier;
sum = 0;
for (j = 0; j < len; j++) {
w = t->fx[i] * f0 + t->fx[i + 1] * f1 + t->dfx[i] * w0 + t->dfx[i + 1] * w1;
sum += data[j * 2] * w;
i += n;
}
return sum;
}
void
functable_fir2 (Functable * t, double *r0, double *r1, double x,
int n, double *data, int len)
{
int i, j;
double f0, f1, w0, w1;
double x2, x3;
double w;
double sum0, sum1;
double floor_x;
x -= t->offset;
x *= t->inv_multiplier;
floor_x = floor (x);
i = floor_x;
x -= floor_x;
x2 = x * x;
x3 = x2 * x;
f1 = 3 * x2 - 2 * x3;
f0 = 1 - f1;
w0 = (x - 2 * x2 + x3) * t->multiplier;
w1 = (-x2 + x3) * t->multiplier;
sum0 = 0;
sum1 = 0;
for (j = 0; j < len; j++) {
w = t->fx[i] * f0 + t->fx[i + 1] * f1 + t->dfx[i] * w0 + t->dfx[i + 1] * w1;
sum0 += data[j * 2] * w;
sum1 += data[j * 2 + 1] * w;
i += n;
}
*r0 = sum0;
*r1 = sum1;
}

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/* Resampling library
* Copyright (C) <2001> David 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.
*/
#ifndef __FUNCTABLE_H__
#define __FUNCTABLE_H__
typedef void FunctableFunc (double *fx, double *dfx, double x, void *closure);
typedef struct _Functable Functable;
struct _Functable {
int length;
double offset;
double multiplier;
double inv_multiplier;
double *fx;
double *dfx;
};
Functable *functable_new (void);
void functable_setup (Functable *t);
void functable_free (Functable *t);
void functable_set_length (Functable *t, int length);
void functable_set_offset (Functable *t, double offset);
void functable_set_multiplier (Functable *t, double multiplier);
void functable_calculate (Functable *t, FunctableFunc func, void *closure);
void functable_calculate_multiply (Functable *t, FunctableFunc func, void *closure);
double functable_evaluate (Functable *t,double x);
double functable_fir(Functable *t,double x0,int n,double *data,int len);
void functable_fir2(Functable *t,double *r0, double *r1, double x0,
int n,double *data,int len);
void functable_func_sinc(double *fx, double *dfx, double x, void *closure);
void functable_func_boxcar(double *fx, double *dfx, double x, void *closure);
void functable_func_hanning(double *fx, double *dfx, double x, void *closure);
#endif /* __PRIVATE_H__ */

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/* GStreamer
* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
* Copyright (C) 2003,2004 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 (at your option) 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.
*/
/* Element-Checklist-Version: 5 */
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <string.h>
#include <math.h>
/*#define DEBUG_ENABLED */
#include "gstaudioresample.h"
#include <gst/audio/audio.h>
GST_DEBUG_CATEGORY_STATIC (audioresample_debug);
#define GST_CAT_DEFAULT audioresample_debug
/* elementfactory information */
static GstElementDetails gst_audioresample_details =
GST_ELEMENT_DETAILS ("Audio scaler",
"Filter/Converter/Audio",
"Resample audio",
"David Schleef <ds@schleef.org>");
/* Audioresample signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
enum
{
ARG_0,
ARG_FILTERLEN
};
#define SUPPORTED_CAPS \
GST_STATIC_CAPS (\
"audio/x-raw-int, " \
"rate = (int) [ 1, MAX ], " \
"channels = (int) [ 1, MAX ], " \
"endianness = (int) BYTE_ORDER, " \
"width = (int) 16, " \
"depth = (int) 16, " \
"signed = (boolean) true")
#if 0
/* disabled because it segfaults */
"audio/x-raw-float, "
"rate = (int) [ 1, MAX ], "
"channels = (int) [ 1, MAX ], "
"endianness = (int) BYTE_ORDER, " "width = (int) 32")
#endif
static GstStaticPadTemplate gst_audioresample_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK, GST_PAD_ALWAYS, SUPPORTED_CAPS);
static GstStaticPadTemplate gst_audioresample_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC, GST_PAD_ALWAYS, SUPPORTED_CAPS);
static void gst_audioresample_base_init (gpointer g_class);
static void gst_audioresample_class_init (AudioresampleClass * klass);
static void gst_audioresample_init (Audioresample * audioresample);
static void gst_audioresample_dispose (GObject * object);
static void gst_audioresample_chain (GstPad * pad, GstData * _data);
static void gst_audioresample_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec);
static void gst_audioresample_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec);
static GstElementClass *parent_class = NULL;
/*static guint gst_audioresample_signals[LAST_SIGNAL] = { 0 }; */
GType audioresample_get_type (void)
{
static GType audioresample_type = 0;
if (!audioresample_type)
{
static const GTypeInfo audioresample_info = {
sizeof (AudioresampleClass),
gst_audioresample_base_init,
NULL,
(GClassInitFunc) gst_audioresample_class_init,
NULL,
NULL,
sizeof (Audioresample), 0,
(GInstanceInitFunc) gst_audioresample_init,};
audioresample_type =
g_type_register_static (GST_TYPE_ELEMENT, "Audioresample",
&audioresample_info, 0);
}
return audioresample_type;
}
static void gst_audioresample_base_init (gpointer g_class)
{
GstElementClass *gstelement_class = GST_ELEMENT_CLASS (g_class);
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&gst_audioresample_src_template));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&gst_audioresample_sink_template));
gst_element_class_set_details (gstelement_class, &gst_audioresample_details);
}
static void gst_audioresample_class_init (AudioresampleClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gobject_class->set_property = gst_audioresample_set_property;
gobject_class->get_property = gst_audioresample_get_property;
gobject_class->dispose = gst_audioresample_dispose;
g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_FILTERLEN,
g_param_spec_int ("filter_length", "filter_length", "filter_length",
0, G_MAXINT, 16, G_PARAM_READWRITE | G_PARAM_CONSTRUCT));
parent_class = g_type_class_ref (GST_TYPE_ELEMENT);
GST_DEBUG_CATEGORY_INIT (audioresample_debug, "audioresample", 0,
"audioresample element");
}
static void gst_audioresample_expand_caps (GstCaps * caps)
{
gint i;
for (i = 0; i < gst_caps_get_size (caps); i++) {
GstStructure *structure = gst_caps_get_structure (caps, i);
const GValue *value;
value = gst_structure_get_value (structure, "rate");
if (value == NULL) {
GST_ERROR ("caps structure doesn't have required rate field");
return;
}
gst_structure_set (structure, "rate", GST_TYPE_INT_RANGE, 1, G_MAXINT, 0);
}
}
static GstCaps *gst_audioresample_getcaps (GstPad * pad)
{
Audioresample *audioresample;
GstCaps *caps;
GstPad *otherpad;
audioresample = GST_AUDIORESAMPLE (gst_pad_get_parent (pad));
otherpad = (pad == audioresample->srcpad) ? audioresample->sinkpad :
audioresample->srcpad;
caps = gst_pad_get_allowed_caps (otherpad);
gst_audioresample_expand_caps (caps);
return caps;
}
static GstCaps *gst_audioresample_fixate (GstPad * pad, const GstCaps * caps)
{
Audioresample *audioresample;
GstPad *otherpad;
int rate;
GstCaps *copy;
GstStructure *structure;
audioresample = GST_AUDIORESAMPLE (gst_pad_get_parent (pad));
if (pad == audioresample->srcpad) {
otherpad = audioresample->sinkpad;
rate = audioresample->i_rate;
} else
{
otherpad = audioresample->srcpad;
rate = audioresample->o_rate;
}
if (!GST_PAD_IS_NEGOTIATING (otherpad))
return NULL;
if (gst_caps_get_size (caps) > 1)
return NULL;
copy = gst_caps_copy (caps);
structure = gst_caps_get_structure (copy, 0);
if (rate) {
if (gst_caps_structure_fixate_field_nearest_int (structure, "rate", rate)) {
return copy;
}
}
gst_caps_free (copy);
return NULL;
}
static GstPadLinkReturn gst_audioresample_link (GstPad * pad,
const GstCaps * caps)
{
Audioresample *audioresample;
GstStructure *structure;
int rate;
int channels;
gboolean ret;
GstPad *otherpad;
audioresample = GST_AUDIORESAMPLE (gst_pad_get_parent (pad));
otherpad = (pad == audioresample->srcpad) ? audioresample->sinkpad :
audioresample->srcpad;
structure = gst_caps_get_structure (caps, 0);
ret = gst_structure_get_int (structure, "rate", &rate);
ret &= gst_structure_get_int (structure, "channels", &channels);
if (!ret)
{
return GST_PAD_LINK_REFUSED;
}
if (gst_pad_is_negotiated (otherpad))
{
GstCaps *othercaps = gst_caps_copy (caps);
int otherrate;
GstPadLinkReturn linkret;
if (pad == audioresample->srcpad) {
otherrate = audioresample->i_rate;
} else {
otherrate = audioresample->o_rate;
}
gst_caps_set_simple (othercaps, "rate", G_TYPE_INT, otherrate, NULL);
linkret = gst_pad_try_set_caps (otherpad, othercaps);
if (GST_PAD_LINK_FAILED (linkret)) {
return GST_PAD_LINK_REFUSED;
}
}
audioresample->channels = channels;
resample_set_n_channels (audioresample->resample, audioresample->channels);
if (pad == audioresample->srcpad) {
audioresample->o_rate = rate;
resample_set_output_rate (audioresample->resample, audioresample->o_rate);
GST_DEBUG ("set o_rate to %d", rate);
} else {
audioresample->i_rate = rate;
resample_set_input_rate (audioresample->resample, audioresample->i_rate);
GST_DEBUG ("set i_rate to %d", rate);
}
return GST_PAD_LINK_OK;
}
static void gst_audioresample_init (Audioresample * audioresample)
{
ResampleState *r;
audioresample->sinkpad =
gst_pad_new_from_template (gst_static_pad_template_get
(&gst_audioresample_sink_template), "sink");
gst_element_add_pad (GST_ELEMENT (audioresample), audioresample->sinkpad);
gst_pad_set_chain_function (audioresample->sinkpad, gst_audioresample_chain);
gst_pad_set_link_function (audioresample->sinkpad, gst_audioresample_link);
gst_pad_set_getcaps_function (audioresample->sinkpad,
gst_audioresample_getcaps);
gst_pad_set_fixate_function (audioresample->sinkpad,
gst_audioresample_fixate);
audioresample->srcpad =
gst_pad_new_from_template (gst_static_pad_template_get
(&gst_audioresample_src_template), "src");
gst_element_add_pad (GST_ELEMENT (audioresample), audioresample->srcpad);
gst_pad_set_link_function (audioresample->srcpad, gst_audioresample_link);
gst_pad_set_getcaps_function (audioresample->srcpad,
gst_audioresample_getcaps);
gst_pad_set_fixate_function (audioresample->srcpad, gst_audioresample_fixate);
r = resample_new ();
audioresample->resample = r;
resample_set_filter_length (r, 64);
resample_set_format (r, RESAMPLE_FORMAT_S16);
}
static void gst_audioresample_dispose (GObject * object)
{
Audioresample *audioresample = GST_AUDIORESAMPLE (object);
if (audioresample->resample) {
resample_free (audioresample->resample);
}
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static void gst_audioresample_chain (GstPad * pad, GstData * _data)
{
GstBuffer *buf = GST_BUFFER (_data);
Audioresample *audioresample;
ResampleState *r;
guchar *data;
gulong size;
int outsize;
GstBuffer *outbuf;
g_return_if_fail (pad != NULL);
g_return_if_fail (GST_IS_PAD (pad));
g_return_if_fail (buf != NULL);
audioresample = GST_AUDIORESAMPLE (gst_pad_get_parent (pad));
if (!GST_IS_BUFFER (_data)) {
gst_pad_push (audioresample->srcpad, _data);
return;
}
if (audioresample->passthru) {
gst_pad_push (audioresample->srcpad, GST_DATA (buf));
return;
}
r = audioresample->resample;
data = GST_BUFFER_DATA (buf);
size = GST_BUFFER_SIZE (buf);
GST_DEBUG ("got buffer of %ld bytes", size);
resample_add_input_data (r, data, size, (ResampleCallback) gst_data_unref,
buf);
outsize = resample_get_output_size (r);
/* FIXME this is audioresample being dumb. dunno why */
if (outsize == 0) {
GST_ERROR ("overriding outbuf size");
outsize = size;
}
outbuf = gst_buffer_new_and_alloc (outsize);
outsize = resample_get_output_data (r, GST_BUFFER_DATA (outbuf), outsize);
GST_BUFFER_SIZE (outbuf) = outsize;
GST_BUFFER_TIMESTAMP (outbuf) =
audioresample->offset * GST_SECOND / audioresample->o_rate;
audioresample->offset += outsize / sizeof (gint16) / audioresample->channels;
gst_pad_push (audioresample->srcpad, GST_DATA (outbuf));
}
static void
gst_audioresample_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
Audioresample *audioresample;
g_return_if_fail (GST_IS_AUDIORESAMPLE (object));
audioresample = GST_AUDIORESAMPLE (object);
switch (prop_id) {
case ARG_FILTERLEN:
audioresample->filter_length = g_value_get_int (value);
GST_DEBUG_OBJECT (GST_ELEMENT (audioresample), "new filter length %d\n",
audioresample->filter_length);
resample_set_filter_length (audioresample->resample,
audioresample->filter_length);
break;
default:G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_audioresample_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
Audioresample *audioresample;
g_return_if_fail (GST_IS_AUDIORESAMPLE (object));
audioresample = GST_AUDIORESAMPLE (object);
switch (prop_id) {
case ARG_FILTERLEN:
g_value_set_int (value, audioresample->filter_length);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static gboolean plugin_init (GstPlugin * plugin)
{
resample_init ();
if (!gst_element_register (plugin, "audioresample", GST_RANK_PRIMARY,
GST_TYPE_AUDIORESAMPLE)) {
return FALSE;
}
return TRUE;
}
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
GST_VERSION_MINOR,
"audioresample",
"Resamples audio", plugin_init, VERSION, "LGPL", GST_PACKAGE, GST_ORIGIN)

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/* GStreamer
* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
*
* 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 (at your option) 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.
*/
#ifndef __AUDIORESAMPLE_H__
#define __AUDIORESAMPLE_H__
#include <gst/gst.h>
#include <audioresample/resample.h>
G_BEGIN_DECLS
#define GST_TYPE_AUDIORESAMPLE \
(audioresample_get_type())
#define GST_AUDIORESAMPLE(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIORESAMPLE,Audioresample))
#define GST_AUDIORESAMPLE_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIORESAMPLE,Audioresample))
#define GST_IS_AUDIORESAMPLE(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIORESAMPLE))
#define GST_IS_AUDIORESAMPLE_CLASS(obj) \
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIORESAMPLE))
typedef struct _Audioresample Audioresample;
typedef struct _AudioresampleClass AudioresampleClass;
struct _Audioresample {
GstElement element;
GstPad *sinkpad,*srcpad;
gboolean passthru;
gint64 offset;
int channels;
int i_rate;
int o_rate;
int filter_length;
ResampleState * resample;
};
struct _AudioresampleClass {
GstElementClass parent_class;
};
GType gst_audioresample_get_type(void);
G_END_DECLS
#endif /* __AUDIORESAMPLE_H__ */

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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.
*/
#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 <audioresample/resample.h>
#include <audioresample/buffer.h>
#include <audioresample/debug.h>
void resample_scale_ref (ResampleState * r);
void resample_scale_functable (ResampleState * r);
void
resample_init (void)
{
static int inited = 0;
if (!inited) {
oil_init ();
inited = 1;
}
}
ResampleState *
resample_new (void)
{
ResampleState *r;
r = malloc (sizeof (ResampleState));
memset (r, 0, sizeof (ResampleState));
r->filter_length = 16;
r->i_start = 0;
if (r->filter_length & 1) {
r->o_start = 0;
} else {
r->o_start = r->o_inc * 0.5;
}
r->queue = audioresample_buffer_queue_new ();
r->out_tmp = malloc (10000 * sizeof (double));
r->need_reinit = 1;
return r;
}
void
resample_free (ResampleState * r)
{
if (r->buffer) {
free (r->buffer);
}
if (r->ft) {
functable_free (r->ft);
}
if (r->queue) {
audioresample_buffer_queue_free (r->queue);
}
if (r->out_tmp) {
free (r->out_tmp);
}
free (r);
}
static void
resample_buffer_free (AudioresampleBuffer * buffer, void *priv)
{
if (buffer->priv2) {
((void (*)(void *)) buffer->priv2) (buffer->priv);
}
}
void
resample_add_input_data (ResampleState * r, void *data, int size,
void (*free_func) (void *), void *closure)
{
AudioresampleBuffer *buffer;
RESAMPLE_DEBUG ("data %p size %d", data, size);
buffer = audioresample_buffer_new_with_data (data, size);
buffer->free = resample_buffer_free;
buffer->priv2 = free_func;
buffer->priv = closure;
audioresample_buffer_queue_push (r->queue, buffer);
}
void
resample_input_eos (ResampleState * r)
{
AudioresampleBuffer *buffer;
int sample_size;
sample_size = r->n_channels * resample_format_size (r->format);
buffer = audioresample_buffer_new_and_alloc (sample_size *
(r->filter_length / 2));
memset (buffer->data, 0, buffer->length);
audioresample_buffer_queue_push (r->queue, buffer);
r->eos = 1;
}
int
resample_get_output_size (ResampleState * r)
{
return floor (audioresample_buffer_queue_get_depth (r->queue) * r->o_rate /
r->i_rate);
}
int
resample_get_output_data (ResampleState * r, void *data, int size)
{
r->o_buf = data;
r->o_size = size;
switch (r->method) {
case 0:
resample_scale_ref (r);
break;
case 1:
resample_scale_functable (r);
break;
default:
break;
}
return size - r->o_size;
}
void
resample_set_filter_length (ResampleState * r, int length)
{
r->filter_length = length;
r->need_reinit = 1;
}
void
resample_set_input_rate (ResampleState * r, double rate)
{
r->i_rate = rate;
r->need_reinit = 1;
}
void
resample_set_output_rate (ResampleState * r, double rate)
{
r->o_rate = rate;
r->need_reinit = 1;
}
void
resample_set_n_channels (ResampleState * r, int n_channels)
{
r->n_channels = n_channels;
r->need_reinit = 1;
}
void
resample_set_format (ResampleState * r, ResampleFormat format)
{
r->format = format;
r->need_reinit = 1;
}
void
resample_set_method (ResampleState * r, int method)
{
r->method = method;
r->need_reinit = 1;
}
int
resample_format_size (ResampleFormat format)
{
switch (format) {
case RESAMPLE_FORMAT_S16:
return 2;
case RESAMPLE_FORMAT_S32:
case RESAMPLE_FORMAT_F32:
return 4;
case RESAMPLE_FORMAT_F64:
return 8;
}
return 0;
}

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/* Resampling library
* Copyright (C) <2001> David 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.
*/
#ifndef __RESAMPLE_H__
#define __RESAMPLE_H__
#include <audioresample/functable.h>
#include <audioresample/buffer.h>
typedef enum {
RESAMPLE_FORMAT_S16 = 0,
RESAMPLE_FORMAT_S32,
RESAMPLE_FORMAT_F32,
RESAMPLE_FORMAT_F64
} ResampleFormat;
typedef void (*ResampleCallback) (void *);
typedef struct _ResampleState ResampleState;
struct _ResampleState {
/* parameters */
int n_channels;
ResampleFormat format;
int filter_length;
double i_rate;
double o_rate;
int method;
/* internal parameters */
int need_reinit;
double halftaps;
/* filter state */
void *o_buf;
int o_size;
AudioresampleBufferQueue *queue;
int eos;
int started;
int sample_size;
void *buffer;
int buffer_len;
double i_start;
double o_start;
double i_inc;
double o_inc;
double sinc_scale;
double i_end;
double o_end;
int i_samples;
int o_samples;
//void *i_buf;
Functable *ft;
double *out_tmp;
};
void resample_init(void);
void resample_cleanup(void);
ResampleState *resample_new (void);
void resample_free (ResampleState *state);
void resample_add_input_data (ResampleState * r, void *data, int size,
ResampleCallback free_func, void *closure);
void resample_input_eos (ResampleState *r);
int resample_get_output_size (ResampleState *r);
int resample_get_output_data (ResampleState *r, void *data, int size);
void resample_set_filter_length (ResampleState *r, int length);
void resample_set_input_rate (ResampleState *r, double rate);
void resample_set_output_rate (ResampleState *r, double rate);
void resample_set_n_channels (ResampleState *r, int n_channels);
void resample_set_format (ResampleState *r, ResampleFormat format);
void resample_set_method (ResampleState *r, int method);
int resample_format_size (ResampleFormat format);
#endif /* __RESAMPLE_H__ */

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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.
*/
#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 <audioresample/resample.h>
#include <audioresample/buffer.h>
#include <audioresample/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;
}
}

272
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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.
*/
#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 <audioresample/resample.h>
#include <audioresample/buffer.h>
#include <audioresample/debug.h>
static void
func_sinc (double *fx, double *dfx, double x, void *closure)
{
//double scale = *(double *)closure;
double scale = M_PI;
if (x == 0) {
*fx = 1;
*dfx = 0;
return;
}
x *= scale;
*fx = sin (x) / x;
*dfx = scale * (cos (x) - sin (x) / x) / x;
}
static void
func_hanning (double *fx, double *dfx, double x, void *closure)
{
double width = *(double *) closure;
if (x < width && x > -width) {
x /= width;
*fx = (1 - x * x) * (1 - x * x);
*dfx = -2 * 2 * x / width * (1 - x * x);
} else {
*fx = 0;
*dfx = 0;
}
}
#if 0
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;
}
#endif
#if 0
static void
functable_test (Functable * ft, double halfwidth)
{
int i;
double x;
for (i = 0; i < 100; i++) {
x = i * 0.1;
printf ("%d %g %g\n", i, resample_sinc_window (x, halfwidth, 1.0),
functable_evaluate (ft, x));
}
exit (0);
}
#endif
void
resample_scale_functable (ResampleState * r)
{
if (r->need_reinit) {
double hanning_width;
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 * r->filter_length;
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;
if (r->ft) {
functable_free (r->ft);
}
r->ft = functable_new ();
functable_set_length (r->ft, r->filter_length * 16);
functable_set_offset (r->ft, -r->filter_length / 2);
functable_set_multiplier (r->ft, 1 / 16.0);
hanning_width = r->filter_length / 2;
functable_calculate (r->ft, func_sinc, NULL);
functable_calculate_multiply (r->ft, func_hanning, &hanning_width);
//functable_test(r->ft, 0.5 * r->filter_length);
#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
r->need_reinit = 0;
}
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 += functable_evaluate (r->ft, offset) * x;
//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 += functable_evaluate (r->ft, offset) * x;
//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 += functable_evaluate (r->ft, offset) * x;
//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 += functable_evaluate (r->ft, offset) * x;
//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;
}
}

210
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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.
*/
#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 <audioresample/resample.h>
#include <audioresample/buffer.h>
#include <audioresample/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_ref (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 * r->filter_length;
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;
}
}