configure.ac: add audioresample and cairo plugins. Remove

Original commit message from CVS:
* configure.ac: add audioresample and cairo plugins.  Remove
HAVE_MMX stuff, because it's not used.
* ext/Makefile.am: same
* ext/audioresample/Makefile.am: You are not ready for an
audio resampling element based on audioresample.
* ext/audioresample/gstaudioresample.c:
* ext/audioresample/gstaudioresample.h:
* ext/cairo/Makefile.am: You are not ready for overlay elements
based on cairo.  Don't look too closely, these elements kinda
suck right now.
* ext/cairo/gstcairo.c: new
* ext/cairo/gsttextoverlay.c: new
* ext/cairo/gsttextoverlay.h: new
* ext/cairo/gsttimeoverlay.c: new
* ext/cairo/gsttimeoverlay.h: new
* gst-libs/gst/media-info/media-info-priv.h: fix compile
problem with compilers that don't support variadic macros.

7 files changed, 574 insertions, 15 deletions

diff --git a/ChangeLog b/ChangeLog
index b301bcaf..cf7a5a30 100644
--- a/ChangeLog
+++ b/ChangeLog
@@ -1,3 +1,23 @@
+2004-12-15  David Schleef  <ds@schleef.org>
+
+	* configure.ac: add audioresample and cairo plugins.  Remove
+	HAVE_MMX stuff, because it's not used.
+	* ext/Makefile.am: same
+	* ext/audioresample/Makefile.am: You are not ready for an
+	audio resampling element based on audioresample.
+	* ext/audioresample/gstaudioresample.c:
+	* ext/audioresample/gstaudioresample.h:
+	* ext/cairo/Makefile.am: You are not ready for overlay elements
+	based on cairo.  Don't look too closely, these elements kinda
+	suck right now.
+	* ext/cairo/gstcairo.c: new
+	* ext/cairo/gsttextoverlay.c: new
+	* ext/cairo/gsttextoverlay.h: new
+	* ext/cairo/gsttimeoverlay.c: new
+	* ext/cairo/gsttimeoverlay.h: new
+	* gst-libs/gst/media-info/media-info-priv.h: fix compile
+	problem with compilers that don't support variadic macros.
+
 2004-12-15  Balamurali Viswanathan  <balamurali.viswanathan@wipro.com>
 
 	Reviewed by:  David Schleef  <ds@schleef.org>
diff --git a/configure.ac b/configure.ac
index 72514037..e64bea79 100644
--- a/configure.ac
+++ b/configure.ac
@@ -806,6 +806,22 @@ GST_CHECK_FEATURE(AUDIOFILE, [audiofile], afsink afsrc, [
     AC_CHECK_LIB(audiofile, af_virtual_file_new, , HAVE_AUDIOFILE="no")
   fi])
 
+dnl *** audioresample ***
+translit(dnm, m, l) AM_CONDITIONAL(USE_AUDIORESAMPLE, true)
+GST_CHECK_FEATURE(AUDIORESAMPLE, [audioresample plug-in], audioresample, [
+  PKG_CHECK_MODULES(AUDIORESAMPLE, audioresample-0.1, HAVE_AUDIORESAMPLE=yes, HAVE_AUDIORESAMPLE=no)
+  AC_SUBST(AUDIORESAMPLE_CFLAGS)
+  AC_SUBST(AUDIORESAMPLE_LIBS)
+])
+
+dnl *** cairo ***
+translit(dnm, m, l) AM_CONDITIONAL(USE_CAIRO, true)
+GST_CHECK_FEATURE(CAIRO, [cairo plug-in], cairo, [
+  PKG_CHECK_MODULES(CAIRO, cairo, HAVE_CAIRO=yes, HAVE_CAIRO=no)
+  AC_SUBST(CAIRO_CFLAGS)
+  AC_SUBST(CAIRO_LIBS)
+])
+
 dnl *** cdaudio ***
 translit(dnm, m, l) AM_CONDITIONAL(USE_CDAUDIO, true)
 GST_CHECK_FEATURE(CDAUDIO, [cdaudio], cdaudio, [
@@ -1722,15 +1738,6 @@ dnl ######################################################################
 dnl # Check command line parameters, and set shell variables accordingly #
 dnl ######################################################################
 
-AC_ARG_ENABLE(libmmx,
-  AC_HELP_STRING([--enable-libmmx],[use libmmx, if available]),
-[case "${enableval}" in
-  yes) USE_LIBMMX=$HAVE_LIBMMX ;;
-  no)  USE_LIBMMX=no ;;
-  *) AC_MSG_ERROR(bad value ${enableval} for --enable-libmmx) ;;
-esac], 
-[USE_LIBMMX=$HAVE_LIBMMX]) dnl Default value
-
 AC_ARG_ENABLE(atomic,
   AC_HELP_STRING([--enable-atomic],[use atomic reference counting header]),
 [case "${enableval}" in
@@ -1783,10 +1790,6 @@ dnl # Set defines according to variables set above #
 dnl ################################################
 
 
-if test "x$USE_LIBMMX" = xyes; then
-  AC_DEFINE(HAVE_LIBMMX, 1, [Define if libmmx is available])
-fi
-
 if test "x$USE_ATOMIC_H" = xyes; then
   AC_DEFINE(HAVE_ATOMIC_H, 1, [Define if atomic.h header file is available])
 fi
@@ -1804,8 +1807,6 @@ dnl #############################
 
 dnl These should be "USE_*" instead of "HAVE_*", but some packages expect
 dnl HAVE_ and it is likely to be easier to stick with the old name
-AM_CONDITIONAL(HAVE_LIBMMX,         test "x$USE_LIBMMX" = "xyes")
-
 AM_CONDITIONAL(HAVE_ATOMIC_H,       test "x$USE_ATOMIC_H" = "xyes")
 
 AM_CONDITIONAL(EXPERIMENTAL,        test "$EXPERIMENTAL" = "$xyes")
@@ -1980,6 +1981,8 @@ ext/alsa/Makefile
 ext/arts/Makefile
 ext/artsd/Makefile
 ext/audiofile/Makefile
+ext/audioresample/Makefile
+ext/cairo/Makefile
 ext/cdaudio/Makefile
 ext/cdparanoia/Makefile
 ext/dirac/Makefile
diff --git a/ext/Makefile.am b/ext/Makefile.am
index 42faa62c..934e0a2c 100644
--- a/ext/Makefile.am
+++ b/ext/Makefile.am
@@ -34,6 +34,18 @@ else
 AUDIOFILE_DIR=
 endif
 
+if USE_AUDIORESAMPLE
+AUDIORESAMPLE_DIR=audioresample
+else
+AUDIORESAMPLE_DIR=
+endif
+
+if USE_CAIRO
+CAIRO_DIR=cairo
+else
+CAIRO_DIR=
+endif
+
 if USE_CDAUDIO
 CDAUDIO_DIR=cdaudio
 else
@@ -383,6 +395,8 @@ SUBDIRS=\
 	$(ARTS_DIR) \
 	$(ARTSC_DIR) \
 	$(AUDIOFILE_DIR) \
+	$(AUDIORESAMPLE_DIR) \
+	$(CAIRO_DIR) \
 	$(CDAUDIO_DIR) \
 	$(CDPARANOIA_DIR) \
 	$(DIRAC_DIR) \
@@ -446,6 +460,8 @@ DIST_SUBDIRS=\
 	arts \
 	artsd \
 	audiofile \
+	audioresample \
+	cairo \
 	cdaudio \
 	cdparanoia \
 	dirac \
diff --git a/ext/audioresample/Makefile.am b/ext/audioresample/Makefile.am
new file mode 100644
index 00000000..777ff3cf
--- /dev/null
+++ b/ext/audioresample/Makefile.am
@@ -0,0 +1,10 @@
+
+plugin_LTLIBRARIES = libgstaudioresample.la
+
+libgstaudioresample_la_SOURCES = gstaudioresample.c
+libgstaudioresample_la_CFLAGS = $(GST_CFLAGS) $(AUDIORESAMPLE_CFLAGS)
+libgstaudioresample_la_LIBADD = $(AUDIORESAMPLE_LIBS)
+libgstaudioresample_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
+
+noinst_HEADERS = gstaudioresample.h
+
diff --git a/ext/audioresample/gstaudioresample.c b/ext/audioresample/gstaudioresample.c
new file mode 100644
index 00000000..f9b8f656
--- /dev/null
+++ b/ext/audioresample/gstaudioresample.c
@@ -0,0 +1,435 @@
+/* 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;
+
+  /* it's not null if we got it, but it might not be ours */
+    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/ext/audioresample/gstaudioresample.h b/ext/audioresample/gstaudioresample.h
new file mode 100644
index 00000000..fc5115da
--- /dev/null
+++ b/ext/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-libs/gst/media-info/media-info-priv.h b/gst-libs/gst/media-info/media-info-priv.h
index fa4ed5c4..3c98240c 100644
--- a/gst-libs/gst/media-info/media-info-priv.h
+++ b/gst-libs/gst/media-info/media-info-priv.h
@@ -23,6 +23,7 @@
 #define __GST_MEDIA_INFO_PRIV_H__
 
 #include <gst/gst.h>
+#include <glib/gprintf.h>
 
 /* debug */
 GST_DEBUG_CATEGORY_EXTERN (gst_media_info_debug);