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authorSebastian Dröge <sebastian.droege@collabora.co.uk>2009-01-23 12:46:28 +0100
committerSebastian Dröge <sebastian.droege@collabora.co.uk>2009-01-23 12:47:19 +0100
commite4e3b44e048ddc1d7499c6108175a5f89c6273d9 (patch)
treef88b685f1b6baf849649494ec557d2ef0ef13a88 /gst/audioresample
parent6fec8619b597f5cc9c58d268ddd9f64ea0a94277 (diff)
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Rename audioresample files and types to legacyresample
Finish the move/rename of audioresample to legacyresample to prevent any confusion.
Diffstat (limited to 'gst/audioresample')
-rw-r--r--gst/audioresample/Makefile.am23
-rw-r--r--gst/audioresample/buffer.c253
-rw-r--r--gst/audioresample/buffer.h51
-rw-r--r--gst/audioresample/debug.c65
-rw-r--r--gst/audioresample/debug.h51
-rw-r--r--gst/audioresample/functable.c254
-rw-r--r--gst/audioresample/functable.h61
-rw-r--r--gst/audioresample/gstaudioresample.c860
-rw-r--r--gst/audioresample/gstaudioresample.h79
-rw-r--r--gst/audioresample/resample.c317
-rw-r--r--gst/audioresample/resample.h128
-rw-r--r--gst/audioresample/resample_chunk.c209
-rw-r--r--gst/audioresample/resample_functable.c271
-rw-r--r--gst/audioresample/resample_ref.c223
14 files changed, 0 insertions, 2845 deletions
diff --git a/gst/audioresample/Makefile.am b/gst/audioresample/Makefile.am
deleted file mode 100644
index c08ab262..00000000
--- a/gst/audioresample/Makefile.am
+++ /dev/null
@@ -1,23 +0,0 @@
-plugin_LTLIBRARIES = libgstlegacyresample.la
-
-resample_SOURCES = \
- functable.c \
- resample.c \
- resample_functable.c \
- resample_ref.c \
- resample_chunk.c \
- resample.h \
- buffer.c
-
-noinst_HEADERS = \
- gstaudioresample.h \
- functable.h \
- debug.h \
- buffer.h
-
-libgstlegacyresample_la_SOURCES = gstaudioresample.c $(resample_SOURCES)
-libgstlegacyresample_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(GST_CFLAGS) $(LIBOIL_CFLAGS)
-libgstlegacyresample_la_LIBADD = $(GST_BASE_LIBS) $(GST_LIBS) $(LIBOIL_LIBS)
-libgstlegacyresample_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
-libgstlegacyresample_la_LIBTOOLFLAGS = --tag=disable-static
-
diff --git a/gst/audioresample/buffer.c b/gst/audioresample/buffer.c
deleted file mode 100644
index 442b4f8c..00000000
--- a/gst/audioresample/buffer.c
+++ /dev/null
@@ -1,253 +0,0 @@
-
-#ifndef HAVE_CONFIG_H
-#include "config.h"
-#endif
-
-#include <glib.h>
-#include <string.h>
-
-#include "buffer.h"
-#include "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;
-}
-
-void
-audioresample_buffer_queue_flush (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);
- queue->buffers = NULL;
- queue->depth = 0;
- queue->offset = 0;
-}
diff --git a/gst/audioresample/buffer.h b/gst/audioresample/buffer.h
deleted file mode 100644
index 4cf1fd94..00000000
--- a/gst/audioresample/buffer.h
+++ /dev/null
@@ -1,51 +0,0 @@
-
-#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);
-void audioresample_buffer_queue_flush (AudioresampleBufferQueue * queue);
-
-#endif
diff --git a/gst/audioresample/debug.c b/gst/audioresample/debug.c
deleted file mode 100644
index 27877277..00000000
--- a/gst/audioresample/debug.c
+++ /dev/null
@@ -1,65 +0,0 @@
-
-#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
deleted file mode 100644
index ff7deafb..00000000
--- a/gst/audioresample/debug.h
+++ /dev/null
@@ -1,51 +0,0 @@
-
-#ifndef __RESAMPLE_DEBUG_H__
-#define __RESAMPLE_DEBUG_H__
-
-#if 0
-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);
-#else
-
-#include <gst/gst.h>
-
-GST_DEBUG_CATEGORY_EXTERN (libaudioresample_debug);
-#define GST_CAT_DEFAULT libaudioresample_debug
-
-#define RESAMPLE_ERROR GST_ERROR
-#define RESAMPLE_WARNING GST_WARNING
-#define RESAMPLE_INFO GST_INFO
-#define RESAMPLE_DEBUG GST_DEBUG
-#define RESAMPLE_LOG GST_LOG
-
-#define resample_debug_set_level(x) do { } while (0)
-
-#endif
-
-#endif
diff --git a/gst/audioresample/functable.c b/gst/audioresample/functable.c
deleted file mode 100644
index d627361f..00000000
--- a/gst/audioresample/functable.c
+++ /dev/null
@@ -1,254 +0,0 @@
-/* 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 "functable.h"
-#include "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
deleted file mode 100644
index 5f56e2bd..00000000
--- a/gst/audioresample/functable.h
+++ /dev/null
@@ -1,61 +0,0 @@
-/* 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
deleted file mode 100644
index 4f6f85e0..00000000
--- a/gst/audioresample/gstaudioresample.c
+++ /dev/null
@@ -1,860 +0,0 @@
-/* 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 */
-
-/**
- * SECTION:element-legacyresample
- *
- * legacyresample resamples raw audio buffers to different sample rates using
- * a configurable windowing function to enhance quality.
- *
- * <refsect2>
- * <title>Example launch line</title>
- * |[
- * gst-launch -v filesrc location=sine.ogg ! oggdemux ! vorbisdec ! audioconvert ! legacyresample ! audio/x-raw-int, rate=8000 ! alsasink
- * ]| Decode an Ogg/Vorbis downsample to 8Khz and play sound through alsa.
- * To create the Ogg/Vorbis file refer to the documentation of vorbisenc.
- * </refsect2>
- *
- * Last reviewed on 2006-03-02 (0.10.4)
- */
-
-#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>
-#include <gst/base/gstbasetransform.h>
-
-GST_DEBUG_CATEGORY_STATIC (audioresample_debug);
-#define GST_CAT_DEFAULT audioresample_debug
-
-/* elementfactory information */
-static const GstElementDetails gst_audioresample_details =
-GST_ELEMENT_DETAILS ("Audio scaler",
- "Filter/Converter/Audio",
- "Resample audio",
- "David Schleef <ds@schleef.org>");
-
-#define DEFAULT_FILTERLEN 16
-
-enum
-{
- PROP_0,
- PROP_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;" \
- "audio/x-raw-int, " \
- "rate = (int) [ 1, MAX ], " \
- "channels = (int) [ 1, MAX ], " \
- "endianness = (int) BYTE_ORDER, " \
- "width = (int) 32, " \
- "depth = (int) 32, " \
- "signed = (boolean) true;" \
- "audio/x-raw-float, " \
- "rate = (int) [ 1, MAX ], " \
- "channels = (int) [ 1, MAX ], " \
- "endianness = (int) BYTE_ORDER, " \
- "width = (int) 32; " \
- "audio/x-raw-float, " \
- "rate = (int) [ 1, MAX ], " \
- "channels = (int) [ 1, MAX ], " \
- "endianness = (int) BYTE_ORDER, " \
- "width = (int) 64" \
-)
-
-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_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);
-
-/* vmethods */
-static gboolean audioresample_get_unit_size (GstBaseTransform * base,
- GstCaps * caps, guint * size);
-static GstCaps *audioresample_transform_caps (GstBaseTransform * base,
- GstPadDirection direction, GstCaps * caps);
-static void audioresample_fixate_caps (GstBaseTransform * base,
- GstPadDirection direction, GstCaps * caps, GstCaps * othercaps);
-static gboolean audioresample_transform_size (GstBaseTransform * trans,
- GstPadDirection direction, GstCaps * incaps, guint insize,
- GstCaps * outcaps, guint * outsize);
-static gboolean audioresample_set_caps (GstBaseTransform * base,
- GstCaps * incaps, GstCaps * outcaps);
-static GstFlowReturn audioresample_pushthrough (GstAudioresample *
- audioresample);
-static GstFlowReturn audioresample_transform (GstBaseTransform * base,
- GstBuffer * inbuf, GstBuffer * outbuf);
-static gboolean audioresample_event (GstBaseTransform * base, GstEvent * event);
-static gboolean audioresample_start (GstBaseTransform * base);
-static gboolean audioresample_stop (GstBaseTransform * base);
-
-static gboolean audioresample_query (GstPad * pad, GstQuery * query);
-static const GstQueryType *audioresample_query_type (GstPad * pad);
-
-#define DEBUG_INIT(bla) \
- GST_DEBUG_CATEGORY_INIT (audioresample_debug, "legacyresample", 0, "audio resampling element");
-
-GST_BOILERPLATE_FULL (GstAudioresample, gst_audioresample, GstBaseTransform,
- GST_TYPE_BASE_TRANSFORM, DEBUG_INIT);
-
-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 (GstAudioresampleClass * klass)
-{
- GObjectClass *gobject_class;
-
- gobject_class = (GObjectClass *) klass;
-
- gobject_class->set_property = gst_audioresample_set_property;
- gobject_class->get_property = gst_audioresample_get_property;
-
- g_object_class_install_property (gobject_class, PROP_FILTERLEN,
- g_param_spec_int ("filter-length", "filter length",
- "Length of the resample filter", 0, G_MAXINT, DEFAULT_FILTERLEN,
- G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS));
-
- GST_BASE_TRANSFORM_CLASS (klass)->start =
- GST_DEBUG_FUNCPTR (audioresample_start);
- GST_BASE_TRANSFORM_CLASS (klass)->stop =
- GST_DEBUG_FUNCPTR (audioresample_stop);
- GST_BASE_TRANSFORM_CLASS (klass)->transform_size =
- GST_DEBUG_FUNCPTR (audioresample_transform_size);
- GST_BASE_TRANSFORM_CLASS (klass)->get_unit_size =
- GST_DEBUG_FUNCPTR (audioresample_get_unit_size);
- GST_BASE_TRANSFORM_CLASS (klass)->transform_caps =
- GST_DEBUG_FUNCPTR (audioresample_transform_caps);
- GST_BASE_TRANSFORM_CLASS (klass)->fixate_caps =
- GST_DEBUG_FUNCPTR (audioresample_fixate_caps);
- GST_BASE_TRANSFORM_CLASS (klass)->set_caps =
- GST_DEBUG_FUNCPTR (audioresample_set_caps);
- GST_BASE_TRANSFORM_CLASS (klass)->transform =
- GST_DEBUG_FUNCPTR (audioresample_transform);
- GST_BASE_TRANSFORM_CLASS (klass)->event =
- GST_DEBUG_FUNCPTR (audioresample_event);
-
- GST_BASE_TRANSFORM_CLASS (klass)->passthrough_on_same_caps = TRUE;
-}
-
-static void
-gst_audioresample_init (GstAudioresample * audioresample,
- GstAudioresampleClass * klass)
-{
- GstBaseTransform *trans;
-
- trans = GST_BASE_TRANSFORM (audioresample);
-
- /* buffer alloc passthrough is too impossible. FIXME, it
- * is trivial in the passthrough case. */
- gst_pad_set_bufferalloc_function (trans->sinkpad, NULL);
-
- audioresample->filter_length = DEFAULT_FILTERLEN;
-
- audioresample->need_discont = FALSE;
-
- gst_pad_set_query_function (trans->srcpad, audioresample_query);
- gst_pad_set_query_type_function (trans->srcpad, audioresample_query_type);
-}
-
-/* vmethods */
-static gboolean
-audioresample_start (GstBaseTransform * base)
-{
- GstAudioresample *audioresample = GST_AUDIORESAMPLE (base);
-
- audioresample->resample = resample_new ();
- audioresample->ts_offset = -1;
- audioresample->offset = -1;
- audioresample->next_ts = -1;
-
- resample_set_filter_length (audioresample->resample,
- audioresample->filter_length);
-
- return TRUE;
-}
-
-static gboolean
-audioresample_stop (GstBaseTransform * base)
-{
- GstAudioresample *audioresample = GST_AUDIORESAMPLE (base);
-
- if (audioresample->resample) {
- resample_free (audioresample->resample);
- audioresample->resample = NULL;
- }
-
- gst_caps_replace (&audioresample->sinkcaps, NULL);
- gst_caps_replace (&audioresample->srccaps, NULL);
-
- return TRUE;
-}
-
-static gboolean
-audioresample_get_unit_size (GstBaseTransform * base, GstCaps * caps,
- guint * size)
-{
- gint width, channels;
- GstStructure *structure;
- gboolean ret;
-
- g_assert (size);
-
- /* this works for both float and int */
- structure = gst_caps_get_structure (caps, 0);
- ret = gst_structure_get_int (structure, "width", &width);
- ret &= gst_structure_get_int (structure, "channels", &channels);
- g_return_val_if_fail (ret, FALSE);
-
- *size = width * channels / 8;
-
- return TRUE;
-}
-
-static GstCaps *
-audioresample_transform_caps (GstBaseTransform * base,
- GstPadDirection direction, GstCaps * caps)
-{
- GstCaps *res;
- GstStructure *structure;
-
- /* transform caps gives one single caps so we can just replace
- * the rate property with our range. */
- res = gst_caps_copy (caps);
- structure = gst_caps_get_structure (res, 0);
- gst_structure_set (structure, "rate", GST_TYPE_INT_RANGE, 1, G_MAXINT, NULL);
-
- return res;
-}
-
-/* Fixate rate to the allowed rate that has the smallest difference */
-static void
-audioresample_fixate_caps (GstBaseTransform * base,
- GstPadDirection direction, GstCaps * caps, GstCaps * othercaps)
-{
- GstStructure *s;
- gint rate;
-
- s = gst_caps_get_structure (caps, 0);
- if (!gst_structure_get_int (s, "rate", &rate))
- return;
-
- s = gst_caps_get_structure (othercaps, 0);
- gst_structure_fixate_field_nearest_int (s, "rate", rate);
-}
-
-static gboolean
-resample_set_state_from_caps (ResampleState * state, GstCaps * incaps,
- GstCaps * outcaps, gint * channels, gint * inrate, gint * outrate)
-{
- GstStructure *structure;
- gboolean ret;
- gint myinrate, myoutrate;
- int mychannels;
- gint width, depth;
- ResampleFormat format;
-
- GST_DEBUG ("incaps %" GST_PTR_FORMAT ", outcaps %"
- GST_PTR_FORMAT, incaps, outcaps);
-
- structure = gst_caps_get_structure (incaps, 0);
-
- /* get width */
- ret = gst_structure_get_int (structure, "width", &width);
- if (!ret)
- goto no_width;
-
- /* figure out the format */
- if (g_str_equal (gst_structure_get_name (structure), "audio/x-raw-float")) {
- if (width == 32)
- format = RESAMPLE_FORMAT_F32;
- else if (width == 64)
- format = RESAMPLE_FORMAT_F64;
- else
- goto wrong_depth;
- } else {
- /* for int, depth and width must be the same */
- ret = gst_structure_get_int (structure, "depth", &depth);
- if (!ret || width != depth)
- goto not_equal;
-
- if (width == 16)
- format = RESAMPLE_FORMAT_S16;
- else if (width == 32)
- format = RESAMPLE_FORMAT_S32;
- else
- goto wrong_depth;
- }
- ret = gst_structure_get_int (structure, "rate", &myinrate);
- ret &= gst_structure_get_int (structure, "channels", &mychannels);
- if (!ret)
- goto no_in_rate_channels;
-
- structure = gst_caps_get_structure (outcaps, 0);
- ret = gst_structure_get_int (structure, "rate", &myoutrate);
- if (!ret)
- goto no_out_rate;
-
- if (channels)
- *channels = mychannels;
- if (inrate)
- *inrate = myinrate;
- if (outrate)
- *outrate = myoutrate;
-
- resample_set_format (state, format);
- resample_set_n_channels (state, mychannels);
- resample_set_input_rate (state, myinrate);
- resample_set_output_rate (state, myoutrate);
-
- return TRUE;
-
- /* ERRORS */
-no_width:
- {
- GST_DEBUG ("failed to get width from caps");
- return FALSE;
- }
-not_equal:
- {
- GST_DEBUG ("width %d and depth %d must be the same", width, depth);
- return FALSE;
- }
-wrong_depth:
- {
- GST_DEBUG ("unknown depth %d found", depth);
- return FALSE;
- }
-no_in_rate_channels:
- {
- GST_DEBUG ("could not get input rate and channels");
- return FALSE;
- }
-no_out_rate:
- {
- GST_DEBUG ("could not get output rate");
- return FALSE;
- }
-}
-
-static gboolean
-audioresample_transform_size (GstBaseTransform * base,
- GstPadDirection direction, GstCaps * caps, guint size, GstCaps * othercaps,
- guint * othersize)
-{
- GstAudioresample *audioresample = GST_AUDIORESAMPLE (base);
- ResampleState *state;
- GstCaps *srccaps, *sinkcaps;
- gboolean use_internal = FALSE; /* whether we use the internal state */
- gboolean ret = TRUE;
-
- GST_LOG_OBJECT (base, "asked to transform size %d in direction %s",
- size, direction == GST_PAD_SINK ? "SINK" : "SRC");
- if (direction == GST_PAD_SINK) {
- sinkcaps = caps;
- srccaps = othercaps;
- } else {
- sinkcaps = othercaps;
- srccaps = caps;
- }
-
- /* if the caps are the ones that _set_caps got called with; we can use
- * our own state; otherwise we'll have to create a state */
- if (gst_caps_is_equal (sinkcaps, audioresample->sinkcaps) &&
- gst_caps_is_equal (srccaps, audioresample->srccaps)) {
- use_internal = TRUE;
- state = audioresample->resample;
- } else {
- GST_DEBUG_OBJECT (audioresample,
- "caps are not the set caps, creating state");
- state = resample_new ();
- resample_set_filter_length (state, audioresample->filter_length);
- resample_set_state_from_caps (state, sinkcaps, srccaps, NULL, NULL, NULL);
- }
-
- if (direction == GST_PAD_SINK) {
- /* asked to convert size of an incoming buffer */
- *othersize = resample_get_output_size_for_input (state, size);
- } else {
- /* asked to convert size of an outgoing buffer */
- *othersize = resample_get_input_size_for_output (state, size);
- }
- g_assert (*othersize % state->sample_size == 0);
-
- /* we make room for one extra sample, given that the resampling filter
- * can output an extra one for non-integral i_rate/o_rate */
- GST_LOG_OBJECT (base, "transformed size %d to %d", size, *othersize);
-
- if (!use_internal) {
- resample_free (state);
- }
-
- return ret;
-}
-
-static gboolean
-audioresample_set_caps (GstBaseTransform * base, GstCaps * incaps,
- GstCaps * outcaps)
-{
- gboolean ret;
- gint inrate, outrate;
- int channels;
- GstAudioresample *audioresample = GST_AUDIORESAMPLE (base);
-
- GST_DEBUG_OBJECT (base, "incaps %" GST_PTR_FORMAT ", outcaps %"
- GST_PTR_FORMAT, incaps, outcaps);
-
- ret = resample_set_state_from_caps (audioresample->resample, incaps, outcaps,
- &channels, &inrate, &outrate);
-
- g_return_val_if_fail (ret, FALSE);
-
- audioresample->channels = channels;
- GST_DEBUG_OBJECT (audioresample, "set channels to %d", channels);
- audioresample->i_rate = inrate;
- GST_DEBUG_OBJECT (audioresample, "set i_rate to %d", inrate);
- audioresample->o_rate = outrate;
- GST_DEBUG_OBJECT (audioresample, "set o_rate to %d", outrate);
-
- /* save caps so we can short-circuit in the size_transform if the caps
- * are the same */
- gst_caps_replace (&audioresample->sinkcaps, incaps);
- gst_caps_replace (&audioresample->srccaps, outcaps);
-
- return TRUE;
-}
-
-static gboolean
-audioresample_event (GstBaseTransform * base, GstEvent * event)
-{
- GstAudioresample *audioresample;
-
- audioresample = GST_AUDIORESAMPLE (base);
-
- switch (GST_EVENT_TYPE (event)) {
- case GST_EVENT_FLUSH_START:
- break;
- case GST_EVENT_FLUSH_STOP:
- if (audioresample->resample)
- resample_input_flush (audioresample->resample);
- audioresample->ts_offset = -1;
- audioresample->next_ts = -1;
- audioresample->offset = -1;
- break;
- case GST_EVENT_NEWSEGMENT:
- resample_input_pushthrough (audioresample->resample);
- audioresample_pushthrough (audioresample);
- audioresample->ts_offset = -1;
- audioresample->next_ts = -1;
- audioresample->offset = -1;
- break;
- case GST_EVENT_EOS:
- resample_input_eos (audioresample->resample);
- audioresample_pushthrough (audioresample);
- break;
- default:
- break;
- }
- return parent_class->event (base, event);
-}
-
-static GstFlowReturn
-audioresample_do_output (GstAudioresample * audioresample, GstBuffer * outbuf)
-{
- int outsize;
- int outsamples;
- ResampleState *r;
-
- r = audioresample->resample;
-
- outsize = resample_get_output_size (r);
- GST_LOG_OBJECT (audioresample, "audioresample can give me %d bytes", outsize);
-
- /* protect against mem corruption */
- if (outsize > GST_BUFFER_SIZE (outbuf)) {
- GST_WARNING_OBJECT (audioresample,
- "overriding audioresample's outsize %d with outbuffer's size %d",
- outsize, GST_BUFFER_SIZE (outbuf));
- outsize = GST_BUFFER_SIZE (outbuf);
- }
- /* catch possibly wrong size differences */
- if (GST_BUFFER_SIZE (outbuf) - outsize > r->sample_size) {
- GST_WARNING_OBJECT (audioresample,
- "audioresample's outsize %d too far from outbuffer's size %d",
- outsize, GST_BUFFER_SIZE (outbuf));
- }
-
- outsize = resample_get_output_data (r, GST_BUFFER_DATA (outbuf), outsize);
- outsamples = outsize / r->sample_size;
- GST_LOG_OBJECT (audioresample, "resample gave me %d bytes or %d samples",
- outsize, outsamples);
-
- GST_BUFFER_OFFSET (outbuf) = audioresample->offset;
- GST_BUFFER_TIMESTAMP (outbuf) = audioresample->next_ts;
-
- if (audioresample->ts_offset != -1) {
- audioresample->offset += outsamples;
- audioresample->ts_offset += outsamples;
- audioresample->next_ts =
- gst_util_uint64_scale_int (audioresample->ts_offset, GST_SECOND,
- audioresample->o_rate);
- GST_BUFFER_OFFSET_END (outbuf) = audioresample->offset;
-
- /* we calculate DURATION as the difference between "next" timestamp
- * and current timestamp so we ensure a contiguous stream, instead of
- * having rounding errors. */
- GST_BUFFER_DURATION (outbuf) = audioresample->next_ts -
- GST_BUFFER_TIMESTAMP (outbuf);
- } else {
- /* no valid offset know, we can still sortof calculate the duration though */
- GST_BUFFER_DURATION (outbuf) =
- gst_util_uint64_scale_int (outsamples, GST_SECOND,
- audioresample->o_rate);
- }
-
- /* check for possible mem corruption */
- if (outsize > GST_BUFFER_SIZE (outbuf)) {
- /* this is an error that when it happens, would need fixing in the
- * resample library; we told it we wanted only GST_BUFFER_SIZE (outbuf),
- * and it gave us more ! */
- GST_WARNING_OBJECT (audioresample,
- "audioresample, you memory corrupting bastard. "
- "you gave me outsize %d while my buffer was size %d",
- outsize, GST_BUFFER_SIZE (outbuf));
- return GST_FLOW_ERROR;
- }
- /* catch possibly wrong size differences */
- if (GST_BUFFER_SIZE (outbuf) - outsize > r->sample_size) {
- GST_WARNING_OBJECT (audioresample,
- "audioresample's written outsize %d too far from outbuffer's size %d",
- outsize, GST_BUFFER_SIZE (outbuf));
- }
- GST_BUFFER_SIZE (outbuf) = outsize;
-
- if (G_UNLIKELY (audioresample->need_discont)) {
- GST_DEBUG_OBJECT (audioresample,
- "marking this buffer with the DISCONT flag");
- GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
- audioresample->need_discont = FALSE;
- }
-
- GST_LOG_OBJECT (audioresample, "transformed to buffer of %d bytes, ts %"
- GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT ", offset %"
- G_GINT64_FORMAT ", offset_end %" G_GINT64_FORMAT,
- outsize, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
- GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)),
- GST_BUFFER_OFFSET (outbuf), GST_BUFFER_OFFSET_END (outbuf));
-
-
- return GST_FLOW_OK;
-}
-
-static gboolean
-audioresample_check_discont (GstAudioresample * audioresample,
- GstClockTime timestamp)
-{
- if (timestamp != GST_CLOCK_TIME_NONE &&
- audioresample->prev_ts != GST_CLOCK_TIME_NONE &&
- audioresample->prev_duration != GST_CLOCK_TIME_NONE &&
- timestamp != audioresample->prev_ts + audioresample->prev_duration) {
- /* Potentially a discontinuous buffer. However, it turns out that many
- * elements generate imperfect streams due to rounding errors, so we permit
- * a small error (up to one sample) without triggering a filter
- * flush/restart (if triggered incorrectly, this will be audible) */
- GstClockTimeDiff diff = timestamp -
- (audioresample->prev_ts + audioresample->prev_duration);
-
- if (ABS (diff) > GST_SECOND / audioresample->i_rate) {
- GST_WARNING_OBJECT (audioresample,
- "encountered timestamp discontinuity of %" G_GINT64_FORMAT, diff);
- return TRUE;
- }
- }
-
- return FALSE;
-}
-
-static GstFlowReturn
-audioresample_transform (GstBaseTransform * base, GstBuffer * inbuf,
- GstBuffer * outbuf)
-{
- GstAudioresample *audioresample;
- ResampleState *r;
- guchar *data, *datacopy;
- gulong size;
- GstClockTime timestamp;
-
- audioresample = GST_AUDIORESAMPLE (base);
- r = audioresample->resample;
-
- data = GST_BUFFER_DATA (inbuf);
- size = GST_BUFFER_SIZE (inbuf);
- timestamp = GST_BUFFER_TIMESTAMP (inbuf);
-
- GST_LOG_OBJECT (audioresample, "transforming buffer of %ld bytes, ts %"
- GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT ", offset %"
- G_GINT64_FORMAT ", offset_end %" G_GINT64_FORMAT,
- size, GST_TIME_ARGS (timestamp),
- GST_TIME_ARGS (GST_BUFFER_DURATION (inbuf)),
- GST_BUFFER_OFFSET (inbuf), GST_BUFFER_OFFSET_END (inbuf));
-
- /* check for timestamp discontinuities and flush/reset if needed */
- if (G_UNLIKELY (audioresample_check_discont (audioresample, timestamp))) {
- /* Flush internal samples */
- audioresample_pushthrough (audioresample);
- /* Inform downstream element about discontinuity */
- audioresample->need_discont = TRUE;
- /* We want to recalculate the offset */
- audioresample->ts_offset = -1;
- }
-
- if (audioresample->ts_offset == -1) {
- /* if we don't know the initial offset yet, calculate it based on the
- * input timestamp. */
- if (GST_CLOCK_TIME_IS_VALID (timestamp)) {
- GstClockTime stime;
-
- /* offset used to calculate the timestamps. We use the sample offset for
- * this to make it more accurate. We want the first buffer to have the
- * same timestamp as the incoming timestamp. */
- audioresample->next_ts = timestamp;
- audioresample->ts_offset =
- gst_util_uint64_scale_int (timestamp, r->o_rate, GST_SECOND);
- /* offset used to set as the buffer offset, this offset is always
- * relative to the stream time, note that timestamp is not... */
- stime = (timestamp - base->segment.start) + base->segment.time;
- audioresample->offset =
- gst_util_uint64_scale_int (stime, r->o_rate, GST_SECOND);
- }
- }
- audioresample->prev_ts = timestamp;
- audioresample->prev_duration = GST_BUFFER_DURATION (inbuf);
-
- /* need to memdup, resample takes ownership. */
- datacopy = g_memdup (data, size);
- resample_add_input_data (r, datacopy, size, g_free, datacopy);
-
- return audioresample_do_output (audioresample, outbuf);
-}
-
-/* push remaining data in the buffers out */
-static GstFlowReturn
-audioresample_pushthrough (GstAudioresample * audioresample)
-{
- int outsize;
- ResampleState *r;
- GstBuffer *outbuf;
- GstFlowReturn res = GST_FLOW_OK;
- GstBaseTransform *trans;
-
- r = audioresample->resample;
-
- outsize = resample_get_output_size (r);
- if (outsize == 0) {
- GST_DEBUG_OBJECT (audioresample, "no internal buffers needing flush");
- goto done;
- }
-
- trans = GST_BASE_TRANSFORM (audioresample);
-
- res = gst_pad_alloc_buffer (trans->srcpad, GST_BUFFER_OFFSET_NONE, outsize,
- GST_PAD_CAPS (trans->srcpad), &outbuf);
- if (G_UNLIKELY (res != GST_FLOW_OK)) {
- GST_WARNING_OBJECT (audioresample, "failed allocating buffer of %d bytes",
- outsize);
- goto done;
- }
-
- res = audioresample_do_output (audioresample, outbuf);
- if (G_UNLIKELY (res != GST_FLOW_OK))
- goto done;
-
- res = gst_pad_push (trans->srcpad, outbuf);
-
-done:
- return res;
-}
-
-static gboolean
-audioresample_query (GstPad * pad, GstQuery * query)
-{
- GstAudioresample *audioresample =
- GST_AUDIORESAMPLE (gst_pad_get_parent (pad));
- GstBaseTransform *trans = GST_BASE_TRANSFORM (audioresample);
- gboolean res = TRUE;
-
- switch (GST_QUERY_TYPE (query)) {
- case GST_QUERY_LATENCY:
- {
- GstClockTime min, max;
- gboolean live;
- guint64 latency;
- GstPad *peer;
- gint rate = audioresample->i_rate;
- gint resampler_latency = audioresample->filter_length / 2;
-
- if (gst_base_transform_is_passthrough (trans))
- resampler_latency = 0;
-
- if ((peer = gst_pad_get_peer (trans->sinkpad))) {
- if ((res = gst_pad_query (peer, query))) {
- gst_query_parse_latency (query, &live, &min, &max);
-
- GST_DEBUG ("Peer latency: min %"
- GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
- GST_TIME_ARGS (min), GST_TIME_ARGS (max));
-
- /* add our own latency */
- if (rate != 0 && resampler_latency != 0)
- latency =
- gst_util_uint64_scale (resampler_latency, GST_SECOND, rate);
- else
- latency = 0;
-
- GST_DEBUG ("Our latency: %" GST_TIME_FORMAT, GST_TIME_ARGS (latency));
-
- min += latency;
- if (max != GST_CLOCK_TIME_NONE)
- max += latency;
-
- GST_DEBUG ("Calculated total latency : min %"
- GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
- GST_TIME_ARGS (min), GST_TIME_ARGS (max));
-
- gst_query_set_latency (query, live, min, max);
- }
- gst_object_unref (peer);
- }
- break;
- }
- default:
- res = gst_pad_query_default (pad, query);
- break;
- }
- gst_object_unref (audioresample);
- return res;
-}
-
-static const GstQueryType *
-audioresample_query_type (GstPad * pad)
-{
- static const GstQueryType types[] = {
- GST_QUERY_LATENCY,
- 0
- };
-
- return types;
-}
-
-static void
-gst_audioresample_set_property (GObject * object, guint prop_id,
- const GValue * value, GParamSpec * pspec)
-{
- GstAudioresample *audioresample;
-
- audioresample = GST_AUDIORESAMPLE (object);
-
- switch (prop_id) {
- case PROP_FILTERLEN:
- audioresample->filter_length = g_value_get_int (value);
- GST_DEBUG_OBJECT (GST_ELEMENT (audioresample), "new filter length %d",
- audioresample->filter_length);
- if (audioresample->resample) {
- resample_set_filter_length (audioresample->resample,
- audioresample->filter_length);
- gst_element_post_message (GST_ELEMENT (audioresample),
- gst_message_new_latency (GST_OBJECT (audioresample)));
- }
- 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)
-{
- GstAudioresample *audioresample;
-
- audioresample = GST_AUDIORESAMPLE (object);
-
- switch (prop_id) {
- case PROP_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, "legacyresample", GST_RANK_MARGINAL,
- GST_TYPE_AUDIORESAMPLE)) {
- return FALSE;
- }
-
- return TRUE;
-}
-
-GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
- GST_VERSION_MINOR,
- "legacyresample",
- "Resamples audio", plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME,
- GST_PACKAGE_ORIGIN);
diff --git a/gst/audioresample/gstaudioresample.h b/gst/audioresample/gstaudioresample.h
deleted file mode 100644
index c969ccdb..00000000
--- a/gst/audioresample/gstaudioresample.h
+++ /dev/null
@@ -1,79 +0,0 @@
-/* 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 <gst/base/gstbasetransform.h>
-
-#include "resample.h"
-
-G_BEGIN_DECLS
-
-#define GST_TYPE_AUDIORESAMPLE \
- (gst_audioresample_get_type())
-#define GST_AUDIORESAMPLE(obj) \
- (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIORESAMPLE,GstAudioresample))
-#define GST_AUDIORESAMPLE_CLASS(klass) \
- (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIORESAMPLE,GstAudioresampleClass))
-#define GST_IS_AUDIORESAMPLE(obj) \
- (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIORESAMPLE))
-#define GST_IS_AUDIORESAMPLE_CLASS(klass) \
- (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIORESAMPLE))
-
-typedef struct _GstAudioresample GstAudioresample;
-typedef struct _GstAudioresampleClass GstAudioresampleClass;
-
-/**
- * GstAudioresample:
- *
- * Opaque data structure.
- */
-struct _GstAudioresample {
- GstBaseTransform element;
-
- GstCaps *srccaps, *sinkcaps;
-
- gboolean passthru;
- gboolean need_discont;
-
- guint64 offset;
- guint64 ts_offset;
- GstClockTime next_ts;
- GstClockTime prev_ts, prev_duration;
- int channels;
-
- int i_rate;
- int o_rate;
- int filter_length;
-
- ResampleState * resample;
-};
-
-struct _GstAudioresampleClass {
- GstBaseTransformClass 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
deleted file mode 100644
index c464adf8..00000000
--- a/gst/audioresample/resample.c
+++ /dev/null
@@ -1,317 +0,0 @@
-/* Resampling library
- * Copyright (C) <2001> David A. Schleef <ds@schleef.org>
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- * Boston, MA 02111-1307, USA.
- */
-
-#ifdef HAVE_CONFIG_H
-#include <config.h>
-#endif
-
-
-#include <string.h>
-#include <math.h>
-#include <stdio.h>
-#include <stdlib.h>
-#include <limits.h>
-#include <liboil/liboil.h>
-
-#include "resample.h"
-#include "buffer.h"
-#include "debug.h"
-
-void resample_scale_ref (ResampleState * r);
-void resample_scale_functable (ResampleState * r);
-
-GST_DEBUG_CATEGORY (libaudioresample_debug);
-
-void
-resample_init (void)
-{
- static int inited = 0;
-
- if (!inited) {
- oil_init ();
- inited = 1;
- GST_DEBUG_CATEGORY_INIT (libaudioresample_debug, "libaudioresample", 0,
- "audio resampling library");
-
- }
-}
-
-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);
- }
-}
-
-/*
- * free_func: a function that frees the given closure. If NULL, caller is
- * responsible for freeing.
- */
-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 = (void *) free_func;
- buffer->priv = closure;
-
- audioresample_buffer_queue_push (r->queue, buffer);
-}
-
-void
-resample_input_flush (ResampleState * r)
-{
- RESAMPLE_DEBUG ("flush");
-
- audioresample_buffer_queue_flush (r->queue);
- r->buffer_filled = 0;
- r->need_reinit = 1;
-}
-
-void
-resample_input_pushthrough (ResampleState * r)
-{
- AudioresampleBuffer *buffer;
- int filter_bytes;
- int buffer_filled;
-
- if (r->sample_size == 0)
- return;
-
- filter_bytes = r->filter_length * r->sample_size;
- buffer_filled = r->buffer_filled;
-
- RESAMPLE_DEBUG ("pushthrough filter_bytes %d, filled %d",
- filter_bytes, buffer_filled);
-
- /* if we have no pending samples, we don't need to do anything. */
- if (buffer_filled <= 0)
- return;
-
- /* send filter_length/2 number of samples so we can get to the
- * last queued samples */
- buffer = audioresample_buffer_new_and_alloc (filter_bytes / 2);
- memset (buffer->data, 0, buffer->length);
-
- RESAMPLE_DEBUG ("pushthrough %u", buffer->length);
-
- audioresample_buffer_queue_push (r->queue, buffer);
-}
-
-void
-resample_input_eos (ResampleState * r)
-{
- RESAMPLE_DEBUG ("EOS");
- resample_input_pushthrough (r);
- r->eos = 1;
-}
-
-int
-resample_get_output_size_for_input (ResampleState * r, int size)
-{
- int outsize;
- double outd;
- int avail;
- int filter_bytes;
- int buffer_filled;
-
- if (r->sample_size == 0)
- return 0;
-
- filter_bytes = r->filter_length * r->sample_size;
- buffer_filled = filter_bytes / 2 - r->buffer_filled / 2;
-
- avail =
- audioresample_buffer_queue_get_depth (r->queue) + size - buffer_filled;
-
- RESAMPLE_DEBUG ("avail %d, o_rate %f, i_rate %f, filter_bytes %d, filled %d",
- avail, r->o_rate, r->i_rate, filter_bytes, buffer_filled);
- if (avail <= 0)
- return 0;
-
- outd = (double) avail *r->o_rate / r->i_rate;
-
- outsize = (int) floor (outd);
-
- /* round off for sample size */
- outsize -= outsize % r->sample_size;
-
- return outsize;
-}
-
-int
-resample_get_input_size_for_output (ResampleState * r, int size)
-{
- int outsize;
- double outd;
- int avail;
-
- if (r->sample_size == 0)
- return 0;
-
- avail = size;
-
- RESAMPLE_DEBUG ("size %d, o_rate %f, i_rate %f", avail, r->o_rate, r->i_rate);
- outd = (double) avail *r->i_rate / r->o_rate;
-
- outsize = (int) ceil (outd);
-
- /* round off for sample size */
- outsize -= outsize % r->sample_size;
-
- return outsize;
-}
-
-int
-resample_get_output_size (ResampleState * r)
-{
- return resample_get_output_size_for_input (r, 0);
-}
-
-int
-resample_get_output_data (ResampleState * r, void *data, int size)
-{
- r->o_buf = data;
- r->o_size = size;
-
- if (size == 0)
- return 0;
-
- 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->sample_size = r->n_channels * resample_format_size (r->format);
- r->need_reinit = 1;
-}
-
-void
-resample_set_format (ResampleState * r, ResampleFormat format)
-{
- r->format = format;
- r->sample_size = r->n_channels * resample_format_size (r->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
deleted file mode 100644
index 84bf8f09..00000000
--- a/gst/audioresample/resample.h
+++ /dev/null
@@ -1,128 +0,0 @@
-/* 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 "functable.h"
-#include "buffer.h"
-
-#ifndef M_PI
-#define M_PI 3.14159265358979323846
-#endif
-
-#ifdef WIN32
-#define rint(x) (floor((x)+0.5))
-#endif
-
-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 */
-
- unsigned char *o_buf;
- int o_size;
-
- AudioresampleBufferQueue *queue;
- int eos;
- int started;
-
- int sample_size;
-
- unsigned char *buffer;
- int buffer_len;
- int buffer_filled;
-
- 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);
-void resample_input_flush (ResampleState *r);
-void resample_input_pushthrough (ResampleState *r);
-
-int resample_get_output_size_for_input (ResampleState * r, int size);
-int resample_get_input_size_for_output (ResampleState * r, int size);
-
-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
deleted file mode 100644
index 1cf9f09f..00000000
--- a/gst/audioresample/resample_chunk.c
+++ /dev/null
@@ -1,209 +0,0 @@
-/* Resampling library
- * Copyright (C) <2001> David A. Schleef <ds@schleef.org>
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- * Boston, MA 02111-1307, USA.
- */
-
-#ifdef HAVE_CONFIG_H
-#include <config.h>
-#endif
-
-
-#include <string.h>
-#include <math.h>
-#include <stdio.h>
-#include <stdlib.h>
-#include <limits.h>
-#include <liboil/liboil.h>
-
-#include "resample.h"
-#include "buffer.h"
-#include "debug.h"
-
-
-static double
-resample_sinc_window (double x, double halfwidth, double scale)
-{
- double y;
-
- if (x == 0)
- return 1.0;
- if (x < -halfwidth || x > halfwidth)
- return 0.0;
-
- y = sin (x * M_PI * scale) / (x * M_PI * scale) * scale;
-
- x /= halfwidth;
- y *= (1 - x * x) * (1 - x * x);
-
- return y;
-}
-
-void
-resample_scale_chunk (ResampleState * r)
-{
- if (r->need_reinit) {
- 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
deleted file mode 100644
index af124276..00000000
--- a/gst/audioresample/resample_functable.c
+++ /dev/null
@@ -1,271 +0,0 @@
-/* Resampling library
- * Copyright (C) <2001> David A. Schleef <ds@schleef.org>
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- * Boston, MA 02111-1307, USA.
- */
-
-#ifdef HAVE_CONFIG_H
-#include <config.h>
-#endif
-
-
-#include <string.h>
-#include <math.h>
-#include <stdio.h>
-#include <stdlib.h>
-#include <limits.h>
-#include <liboil/liboil.h>
-
-#include "resample.h"
-#include "buffer.h"
-#include "debug.h"
-
-static 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;
-
- 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
deleted file mode 100644
index bb8d2411..00000000
--- a/gst/audioresample/resample_ref.c
+++ /dev/null
@@ -1,223 +0,0 @@
-/* Resampling library
- * Copyright (C) <2001> David A. Schleef <ds@schleef.org>
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- * Boston, MA 02111-1307, USA.
- */
-
-#ifdef HAVE_CONFIG_H
-#include <config.h>
-#endif
-
-
-#include <string.h>
-#include <math.h>
-#include <stdio.h>
-#include <stdlib.h>
-#include <limits.h>
-#include <liboil/liboil.h>
-
-#include "resample.h"
-#include "buffer.h"
-#include "debug.h"
-
-
-static double
-resample_sinc_window (double x, double halfwidth, double scale)
-{
- double y;
-
- if (x == 0)
- return 1.0;
- if (x < -halfwidth || x > halfwidth)
- return 0.0;
-
- y = sin (x * M_PI * scale) / (x * M_PI * scale) * scale;
-
- x /= halfwidth;
- y *= (1 - x * x) * (1 - x * x);
-
- return y;
-}
-
-void
-resample_scale_ref (ResampleState * r)
-{
- if (r->need_reinit) {
- 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->buffer_filled = 0;
-
- 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
- }
-
- RESAMPLE_DEBUG ("asked to resample %d bytes", r->o_size);
- RESAMPLE_DEBUG ("%d bytes in queue",
- audioresample_buffer_queue_get_depth (r->queue));
-
- while (r->o_size >= r->sample_size) {
- double midpoint;
- int i;
- int j;
-
- midpoint = r->i_start + (r->filter_length - 1) * 0.5 * r->i_inc;
- RESAMPLE_DEBUG
- ("still need to output %d bytes, %d input left, i_start %g, midpoint %f",
- r->o_size, audioresample_buffer_queue_get_depth (r->queue), r->i_start,
- midpoint);
- if (midpoint > 0.5 * r->i_inc) {
- RESAMPLE_ERROR ("inconsistent state");
- }
- while (midpoint < -0.5 * r->i_inc) {
- AudioresampleBuffer *buffer;
-
- RESAMPLE_DEBUG ("midpoint %f < %f, r->i_inc %f", midpoint,
- -0.5 * r->i_inc, r->i_inc);
- buffer = audioresample_buffer_queue_pull (r->queue, r->sample_size);
- if (buffer == NULL) {
- /* FIXME: for the first buffer, this isn't necessarily an error,
- * since because of the filter length we'll output less buffers.
- * deal with that so we don't print to console */
- 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);
- r->buffer_filled = MIN (r->buffer_filled + r->sample_size, r->buffer_len);
-
- 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;
- }
-}