gst/audioresample/Makefile.am: Leet audioresampling code

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:

14 files changed, 2254 insertions, 0 deletions

diff --git a/gst/audioresample/Makefile.am b/gst/audioresample/Makefile.am
new file mode 100644
index 00000000..bff05034
--- /dev/null
+++ b/gst/audioresample/Makefile.am
@@ -0,0 +1,21 @@
+
+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)
+
diff --git a/gst/audioresample/buffer.c b/gst/audioresample/buffer.c
new file mode 100644
index 00000000..f72e6056
--- /dev/null
+++ b/gst/audioresample/buffer.c
@@ -0,0 +1,238 @@
+
+#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;
+}
diff --git a/gst/audioresample/buffer.h b/gst/audioresample/buffer.h
new file mode 100644
index 00000000..17fb5f90
--- /dev/null
+++ b/gst/audioresample/buffer.h
@@ -0,0 +1,48 @@
+
+#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
diff --git a/gst/audioresample/debug.c b/gst/audioresample/debug.c
new file mode 100644
index 00000000..27877277
--- /dev/null
+++ b/gst/audioresample/debug.c
@@ -0,0 +1,65 @@
+
+#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;
+}
diff --git a/gst/audioresample/debug.h b/gst/audioresample/debug.h
new file mode 100644
index 00000000..2205940c
--- /dev/null
+++ b/gst/audioresample/debug.h
@@ -0,0 +1,34 @@
+
+#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
diff --git a/gst/audioresample/functable.c b/gst/audioresample/functable.c
new file mode 100644
index 00000000..41844015
--- /dev/null
+++ b/gst/audioresample/functable.c
@@ -0,0 +1,254 @@
+/* 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;
+}
diff --git a/gst/audioresample/functable.h b/gst/audioresample/functable.h
new file mode 100644
index 00000000..4349719d
--- /dev/null
+++ b/gst/audioresample/functable.h
@@ -0,0 +1,61 @@
+/* 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__ */
+
diff --git a/gst/audioresample/gstaudioresample.c b/gst/audioresample/gstaudioresample.c
new file mode 100644
index 00000000..363acd9b
--- /dev/null
+++ b/gst/audioresample/gstaudioresample.c
@@ -0,0 +1,434 @@
+/* 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)
diff --git a/gst/audioresample/gstaudioresample.h b/gst/audioresample/gstaudioresample.h
new file mode 100644
index 00000000..fc5115da
--- /dev/null
+++ b/gst/audioresample/gstaudioresample.h
@@ -0,0 +1,74 @@
+/* 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__ */
diff --git a/gst/audioresample/resample.c b/gst/audioresample/resample.c
new file mode 100644
index 00000000..38c6ba84
--- /dev/null
+++ b/gst/audioresample/resample.c
@@ -0,0 +1,219 @@
+/* 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;
+}
diff --git a/gst/audioresample/resample.h b/gst/audioresample/resample.h
new file mode 100644
index 00000000..9be54f46
--- /dev/null
+++ b/gst/audioresample/resample.h
@@ -0,0 +1,114 @@
+/* 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__ */
+
diff --git a/gst/audioresample/resample_chunk.c b/gst/audioresample/resample_chunk.c
new file mode 100644
index 00000000..53755e62
--- /dev/null
+++ b/gst/audioresample/resample_chunk.c
@@ -0,0 +1,210 @@
+/* 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;
+  }
+
+}
diff --git a/gst/audioresample/resample_functable.c b/gst/audioresample/resample_functable.c
new file mode 100644
index 00000000..af5f9253
--- /dev/null
+++ b/gst/audioresample/resample_functable.c
@@ -0,0 +1,272 @@
+/* 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;
+  }
+
+}
diff --git a/gst/audioresample/resample_ref.c b/gst/audioresample/resample_ref.c
new file mode 100644
index 00000000..a4623e71
--- /dev/null
+++ b/gst/audioresample/resample_ref.c
@@ -0,0 +1,210 @@
+/* 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;
+  }
+
+}