fixed some GST_LIBS stuff added audiofile added gst-libs/audio building

Original commit message from CVS:
fixed some GST_LIBS stuff
added audiofile
added gst-libs/audio building

Makefile.am

@@ -1,3 +1,3 @@
-SUBDIRS=sys ext
+SUBDIRS=gst sys ext gst-libs
 
-DIST_SUBDIRS=sys ext
+DIST_SUBDIRS=gst sys ext gst-libs

TODO

@@ -5,3 +5,5 @@
   it better ;)
 
 * check SDL optimisation flags
+
+* check GST_* in configure.ac, there is too much in it

configure.ac

@@ -348,6 +348,16 @@ AC_SUBST(X_PRE_LIBS)
 AC_SUBST(X_EXTRA_LIBS)
 AC_SUBST(X_LIBS)
 
+dnl ==========================================================================
+dnl ============================= gst plugins ================================
+dnl ==========================================================================
+
+dnl *** sine ***
+translit(dnm, m, l) AM_CONDITIONAL(USE_SINE, true)
+GST_CHECK_FEATURE(SINE, [sine plugin], sinesrc, [
+  HAVE_SINE="yes"
+])
+
 dnl ==========================================================================
 dnl ============================= sys plugins ================================
 dnl ==========================================================================
@@ -428,11 +438,11 @@ GST_CHECK_FEATURE(ARTSC, [artsd plugins], artsdsink, [
 
 dnl *** audiofile ***
 dnl this check uses the GST_CHECK_CONFIGPROG macro
-translit(dnm, m, l) AM_CONDITIONAL(USE_LIBAUDIOFILE, true)
+translit(dnm, m, l) AM_CONDITIONAL(USE_AUDIOFILE, true)
-GST_CHECK_FEATURE(LIBAUDIOFILE, [audiofile], afsink afsrc, [
+GST_CHECK_FEATURE(AUDIOFILE, [audiofile], afsink afsrc, [
-  translit(dnm, m, l) AC_SUBST(LIBAUDIOFILE_LIBS)
+  translit(dnm, m, l) AC_SUBST(AUDIOFILE_LIBS)
-  translit(dnm, m, l) AC_SUBST(LIBAUDIOFILE_CFLAGS)
+  translit(dnm, m, l) AC_SUBST(AUDIOFILE_CFLAGS)
-  GST_CHECK_CONFIGPROG(LIBAUDIOFILE, audiofile-config)
+  GST_CHECK_CONFIGPROG(AUDIOFILE, audiofile-config)
 ])
 
 dnl *** avifile ***
@@ -1071,10 +1081,11 @@ AC_SUBST(LIBGST_LIBS)
 AC_SUBST(LIBGST_CFLAGS)
 
 dnl Vars for everyone else
-GST_LIBS="\$(top_builddir)/gst/libgst.la $LIBGST_LIBS"
+dnl FIXME: is there a reason to add this top_builddir stuff ? don't think so
-GST_CFLAGS="-I\$(top_srcdir) -I\$(top_srcdir)/include $LIBGST_CFLAGS"
+dnl GST_LIBS="\$(top_builddir)/gst/libgst.la $LIBGST_LIBS"
-AC_SUBST(GST_LIBS)
+dnl GST_CFLAGS="-I\$(top_srcdir) -I\$(top_srcdir)/include $LIBGST_CFLAGS"
-AC_SUBST(GST_CFLAGS)
+dnl AC_SUBST(GST_LIBS)
+dnl AC_SUBST(GST_CFLAGS)
 
 dnl #############################
 dnl # Configure the subpackages #
@@ -1127,6 +1138,8 @@ dnl stamp.h
 dnl echo "$infomessages", infomessages="$infomessages"
 AC_OUTPUT(
 Makefile
+gst/Makefile
+gst/sine/Makefile
 sys/Makefile
 sys/oss/Makefile
 sys/qcam/Makefile
@@ -1135,11 +1148,14 @@ sys/vcd/Makefile
 sys/vga/Makefile
 sys/xvideo/Makefile
 ext/Makefile
+ext/audiofile/Makefile
 ext/esd/Makefile
 ext/lame/Makefile
 ext/mad/Makefile
 ext/sdl/Makefile
 ext/vorbis/Makefile
+gst-libs/Makefile
+gst-libs/audio/Makefile
 )
 
 echo -e "configure: *** Plugins that will be built : $GST_PLUGINS_YES"

ext/Makefile.am

@@ -1,3 +1,9 @@
+if USE_AUDIOFILE
+AUDIOFILE_DIR=audiofile
+else
+AUDIOFILE_DIR=
+endif
+
 if USE_ESD
 ESD_DIR=esd
 else
@@ -29,6 +35,7 @@ VORBIS_DIR=
 endif
 
 
-SUBDIRS=$(ESD_DIR) $(LAME_DIR) $(MAD_DIR) $(SDL_DIR) $(VORBIS_DIR)
+SUBDIRS=$(AUDIOFILE_DIR) $(ESD_DIR) $(LAME_DIR) $(MAD_DIR) \
+	$(SDL_DIR) $(VORBIS_DIR)
 
-DIST_SUBDIRS=esd lame mad sdl vorbis
+DIST_SUBDIRS=audiofile esd lame mad sdl vorbis

ext/audiofile/Makefile.am

@@ -0,0 +1,14 @@
+plugindir = $(libdir)/gst
+
+plugin_LTLIBRARIES = libafsink.la libafsrc.la
+
+libafsink_la_SOURCES = gstafsink.c
+libafsrc_la_SOURCES = gstafsrc.c
+noinst_HEADERS = gstafsink.h gstafsrc.h
+
+libafsink_la_LIBADD = $(AUDIOFILE_LIBS)
+libafsrc_la_LIBADD = $(AUDIOFILE_LIBS)
+
+libafsink_la_CFLAGS = $(GST_CFLAGS)
+libafsrc_la_CFLAGS = $(GST_CFLAGS)
+

ext/audiofile/README

@@ -0,0 +1,39 @@
+This plugin wraps the SGI Audiofile 
+(http://oss.sgi.com/projects/audiofile/) library into a src and sink
+element.
+
+You can read from and write to the supported formats (WAVE, AIFF, AIFFC,
+NEXTSND).
+
+What is supported :
+* all the file formats
+* integer sample data, both 2's complement and unsigned
+* 8 or 16 bit width & depth (haven't tested others)
+* sample rate
+* some sort of endianness control
+
+What isn't supported yet :
+* float data
+
+What you can do :
+* src element only accepts location argument
+* sink element accepts location, endianness and type
+	- location : file on the system to output
+	- endianness : at this time endianness is still a bit shady
+	  	you can either set 1234 or 4321;
+		setting it to 4321 will byteswap the buffer data
+		you might want to keep it at 1234 for now
+	- type : one of the file types
+
+Use gstreamer-inspect on afsink and afsrc to see all of the supported
+options.
+
+Examples :
+
+* tools/gstreamer-launch afsrc location=/opt/media/wav/dark-480-16-m.wav ! afsink type=2 location=/opt/media/wav/dark-480-16-m.aiff
+
+Future plans :
+
+* add float support
+* wrap up afsink and afsrc with pipe and fork to act like data convertors,
+  allowing arbitrary choice of sink and src element

ext/audiofile/gstafsink.c

@@ -0,0 +1,508 @@
+/* GStreamer
+ * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
+ *                    2000 Wim Taymans <wtay@chello.be>
+ *
+ * gstafsink.c: 
+ *
+ * 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.
+ */
+
+
+#include <gst/gst.h>
+#include "gstafsink.h"
+
+
+static GstElementDetails afsink_details = {
+  "Audiofile Sink",
+  "Sink",
+  "Audiofile sink for audio/raw",
+  VERSION,
+  "Thomas <thomas@apestaart.org>",
+  "(C) 2001"
+};
+
+
+/* AFSink signals and args */
+enum {
+  /* FILL ME */
+  SIGNAL_HANDOFF,
+  LAST_SIGNAL
+};
+
+enum {
+  ARG_0,
+  ARG_TYPE,
+  ARG_OUTPUT_ENDIANNESS,
+  ARG_LOCATION
+};
+
+/* added a sink factory function to force audio/raw MIME type */
+/* I think the caps can be broader, we need to change that somehow */
+GST_PADTEMPLATE_FACTORY (afsink_sink_factory,
+  "sink",
+  GST_PAD_SINK,
+  GST_PAD_ALWAYS,
+  GST_CAPS_NEW (  
+    "audiofile_sink",
+    "audio/raw",
+      "format",             GST_PROPS_STRING ("int"),
+        "law",              GST_PROPS_INT (0),
+        "endianness",       GST_PROPS_INT (G_BYTE_ORDER),
+        "signed",           GST_PROPS_LIST (
+				  GST_PROPS_BOOLEAN (TRUE),
+       				  GST_PROPS_BOOLEAN (FALSE)
+       			    ),
+        "width",            GST_PROPS_INT_RANGE (8, 16),
+        "depth",            GST_PROPS_INT_RANGE (8, 16),
+        "rate",             GST_PROPS_INT_RANGE (4000, 48000), //FIXME
+        "channels",         GST_PROPS_INT_RANGE (1, 2)
+  )
+);
+
+/* we use an enum for the output type arg */
+
+#define GST_TYPE_AFSINK_TYPES (gst_afsink_types_get_type())
+/* FIXME: fix the string ints to be string-converted from the audiofile.h types */
+static GType
+gst_afsink_types_get_type (void) 
+{
+  static GType afsink_types_type = 0;
+  static GEnumValue afsink_types[] = {
+    {AF_FILE_RAWDATA, "0", "raw PCM"},
+    {AF_FILE_AIFFC,   "1", "AIFFC"},
+    {AF_FILE_AIFF,    "2", "AIFF"},
+    {AF_FILE_NEXTSND, "3", "Next/SND"},
+    {AF_FILE_WAVE,    "4", "Wave"},
+    {0, NULL, NULL},
+  };
+  
+  if (!afsink_types_type) 
+  {
+    afsink_types_type = g_enum_register_static ("GstAudiosinkTypes", afsink_types);
+  }
+  return afsink_types_type;
+}
+
+static void		gst_afsink_class_init	(GstAFSinkClass *klass);
+static void		gst_afsink_init		(GstAFSink *afsink);
+
+static gboolean 	gst_afsink_open_file 	(GstAFSink *sink);
+static void 		gst_afsink_close_file 	(GstAFSink *sink);
+
+static void		gst_afsink_chain	(GstPad *pad,GstBuffer *buf);
+
+static void		gst_afsink_set_property	(GObject *object, guint prop_id, const GValue *value, 
+						 GParamSpec *pspec);
+static void		gst_afsink_get_property	(GObject *object, guint prop_id, GValue *value, 
+						 GParamSpec *pspec);
+
+static gboolean		gst_afsink_handle_event (GstPad *pad, GstEvent *event);
+
+static GstElementStateReturn gst_afsink_change_state (GstElement *element);
+
+static GstElementClass *parent_class = NULL;
+static guint gst_afsink_signals[LAST_SIGNAL] = { 0 };
+
+GType
+gst_afsink_get_type (void) 
+{
+  static GType afsink_type = 0;
+
+  if (!afsink_type) {
+    static const GTypeInfo afsink_info = {
+      sizeof (GstAFSinkClass),      NULL,
+      NULL,
+      (GClassInitFunc) gst_afsink_class_init,
+      NULL,
+      NULL,
+      sizeof (GstAFSink),
+      0,
+      (GInstanceInitFunc) gst_afsink_init,
+    };
+    afsink_type = g_type_register_static (GST_TYPE_ELEMENT, "GstAFSink", &afsink_info, 0);
+  }
+  return afsink_type;
+}
+
+static void
+gst_afsink_class_init (GstAFSinkClass *klass) 
+{
+  GObjectClass *gobject_class;
+  GstElementClass *gstelement_class;
+
+  gobject_class = (GObjectClass*)klass;
+  gstelement_class = (GstElementClass*)klass;
+
+  parent_class = g_type_class_ref (GST_TYPE_ELEMENT);
+
+  gst_element_install_std_props (
+         GST_ELEMENT_CLASS (klass),
+         "location",     ARG_LOCATION,     G_PARAM_READWRITE,
+         NULL);
+
+  g_object_class_install_property(G_OBJECT_CLASS(klass), ARG_TYPE,
+    g_param_spec_enum("type","type","type",
+                      GST_TYPE_AFSINK_TYPES,0,G_PARAM_READWRITE)); // CHECKME!
+  g_object_class_install_property(G_OBJECT_CLASS(klass), ARG_OUTPUT_ENDIANNESS,
+    g_param_spec_int("endianness","endianness","endianness",
+                     G_MININT,G_MAXINT,0,G_PARAM_READWRITE)); // CHECKME
+ 
+  gst_afsink_signals[SIGNAL_HANDOFF] =
+    g_signal_new ("handoff", G_TYPE_FROM_CLASS(klass), G_SIGNAL_RUN_LAST,
+                   G_STRUCT_OFFSET (GstAFSinkClass, handoff), NULL, NULL,
+                   g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0);
+
+
+  gobject_class->set_property = gst_afsink_set_property;
+  gobject_class->get_property = gst_afsink_get_property;
+
+  gstelement_class->change_state = gst_afsink_change_state;
+}
+
+static void 
+gst_afsink_init (GstAFSink *afsink) 
+{
+  // GstPad *pad;   this is now done in the struct
+
+  afsink->sinkpad = gst_pad_new_from_template (
+		  GST_PADTEMPLATE_GET (afsink_sink_factory), "sink");
+  gst_element_add_pad (GST_ELEMENT (afsink), afsink->sinkpad);
+
+  gst_pad_set_chain_function (afsink->sinkpad, gst_afsink_chain);
+  gst_pad_set_event_function (afsink->sinkpad, gst_afsink_handle_event);
+
+  afsink->filename = NULL;
+  afsink->file = NULL;
+  /* default values, should never be needed */
+  afsink->channels = 2;
+  afsink->width = 16;
+  afsink->rate = 44100;
+  afsink->type = AF_FILE_WAVE;
+  afsink->endianness_data = 1234;
+  afsink->endianness_wanted = 1234;
+}
+
+static void
+gst_afsink_set_property (GObject *object, guint prop_id, const GValue *value, GParamSpec *pspec)
+{
+  GstAFSink *sink;
+
+  /* it's not null if we got it, but it might not be ours */
+  sink = GST_AFSINK (object);
+
+  switch (prop_id) {
+    case ARG_LOCATION:
+      /* the element must be stopped or paused in order to do this */
+      g_return_if_fail ((GST_STATE (sink) < GST_STATE_PLAYING)
+                      || (GST_STATE (sink) == GST_STATE_PAUSED));
+      if (sink->filename)
+	g_free (sink->filename);
+      sink->filename = g_strdup (g_value_get_string (value));
+      if ( (GST_STATE (sink) == GST_STATE_PAUSED)
+        && (sink->filename != NULL))
+      {
+              gst_afsink_close_file (sink);
+              gst_afsink_open_file (sink);
+      }
+
+      break;
+    case ARG_TYPE:
+      sink->type = g_value_get_enum (value);
+      break;
+    case ARG_OUTPUT_ENDIANNESS:
+      {
+        int end = g_value_get_int (value);
+        if (end == 1234 || end == 4321)
+          sink->endianness_output = end;
+      }
+      break;
+    default:
+      break;
+  }
+}
+
+static void   
+gst_afsink_get_property (GObject *object, guint prop_id, GValue *value, GParamSpec *pspec)
+{
+  GstAFSink *sink;
+ 
+  /* it's not null if we got it, but it might not be ours */
+  g_return_if_fail (GST_IS_AFSINK (object));
+ 
+  sink = GST_AFSINK (object);
+  
+  switch (prop_id) {
+    case ARG_LOCATION:
+      g_value_set_string (value, sink->filename);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static gboolean
+plugin_init (GModule *module, GstPlugin *plugin)
+{
+  GstElementFactory *factory;
+  
+  factory = gst_elementfactory_new ("afsink", GST_TYPE_AFSINK,
+                                    &afsink_details);
+  g_return_val_if_fail (factory != NULL, FALSE);
+  
+  gst_elementfactory_add_padtemplate (factory, GST_PADTEMPLATE_GET (afsink_sink_factory));
+
+  gst_plugin_add_feature (plugin, GST_PLUGIN_FEATURE (factory));
+  
+  return TRUE;
+}
+
+
+GstPluginDesc plugin_desc = {
+  GST_VERSION_MAJOR,
+  GST_VERSION_MINOR,
+  "afsink",
+  plugin_init
+};
+
+/* this is where we open the audiofile */
+static gboolean
+gst_afsink_open_file (GstAFSink *sink)
+{
+  AFfilesetup outfilesetup;
+  GstCaps *caps;
+  int sample_format;			/* audiofile's sample format, look in audiofile.h */
+  int byte_order = 0;			/* audiofile's byte order defines */
+  
+  g_return_val_if_fail (!GST_FLAG_IS_SET (sink, GST_AFSINK_OPEN), FALSE);
+
+  /* open the file */
+/* we use audiofile now
+  sink->file = fopen (sink->filename, "w");
+  if (sink->file == NULL) {
+    perror ("open");
+    gst_element_error (GST_ELEMENT (sink), g_strconcat("opening file \"", sink->filename, "\"", NULL));
+    return FALSE;
+  } 
+*/
+
+  /* get the audio parameters */
+  caps = NULL;
+  g_return_val_if_fail (GST_IS_PAD (sink->sinkpad), FALSE);
+  caps = GST_PAD_CAPS (sink->sinkpad);
+  
+  if (caps == NULL)
+  {
+    // FIXME : Please change this to a better warning method !
+    printf ("WARNING: gstafsink chain : Could not get caps of pad !\n");
+  }
+  else
+  {
+    sink->channels        = gst_caps_get_int (caps, "channels");
+    sink->width           = gst_caps_get_int (caps, "width");
+    sink->rate            = gst_caps_get_int (caps, "rate");
+    sink->is_signed       = gst_caps_get_int (caps, "signed");
+    sink->endianness_data = gst_caps_get_int (caps, "endianness");
+  }
+  GST_DEBUG (GST_CAT_PLUGIN_INFO, "channels %d, width %d, rate %d, signed %s\n",
+  	   		sink->channels, sink->width, sink->rate,
+  	   		sink->is_signed ? "yes" : "no");
+  GST_DEBUG (GST_CAT_PLUGIN_INFO, "endianness: data %d, output %d\n", 
+	   sink->endianness_data, sink->endianness_output);
+  /* setup the output file */
+  if (sink->is_signed)
+    sample_format = AF_SAMPFMT_TWOSCOMP;
+  else
+    sample_format = AF_SAMPFMT_UNSIGNED;
+  // FIXME : this check didn't seem to work, so let the output endianness be set */
+  /*
+  if (sink->endianness_data == sink->endianness_wanted)
+    byte_order = AF_BYTEORDER_LITTLEENDIAN;
+  else
+    byte_order = AF_BYTEORDER_BIGENDIAN;
+  */
+  if (sink->endianness_output == 1234)
+    byte_order = AF_BYTEORDER_LITTLEENDIAN;
+  else
+    byte_order = AF_BYTEORDER_BIGENDIAN;
+
+  outfilesetup = afNewFileSetup ();
+  afInitFileFormat (outfilesetup, sink->type);
+  afInitChannels (outfilesetup, AF_DEFAULT_TRACK, sink->channels);
+  afInitRate (outfilesetup, AF_DEFAULT_TRACK, sink->rate);
+  afInitSampleFormat (outfilesetup, AF_DEFAULT_TRACK, 
+  					  sample_format, sink->width);
+
+  /* open it */
+  sink->file = afOpenFile (sink->filename, "w", outfilesetup);
+  if (sink->file == AF_NULL_FILEHANDLE)
+  {
+    perror ("open");
+    gst_element_error (GST_ELEMENT (sink), g_strconcat("opening file \"", sink->filename, "\"", NULL));
+    return FALSE;
+  } 
+
+  afFreeFileSetup (outfilesetup);
+//  afSetVirtualByteOrder (sink->file, AF_DEFAULT_TRACK, byte_order);
+  
+  GST_FLAG_SET (sink, GST_AFSINK_OPEN);
+
+  return TRUE;
+}
+
+static void
+gst_afsink_close_file (GstAFSink *sink)
+{
+//  g_print ("DEBUG: closing sinkfile...\n");
+  g_return_if_fail (GST_FLAG_IS_SET (sink, GST_AFSINK_OPEN));
+//  g_print ("DEBUG: past flag test\n");
+//  if (fclose (sink->file) != 0)
+  if (afCloseFile (sink->file) != 0)
+  {
+    g_print ("WARNING: afsink: oops, error closing !\n");
+    perror ("close");
+    gst_element_error (GST_ELEMENT (sink), g_strconcat("closing file \"", sink->filename, "\"", NULL));
+  }
+  else {
+    GST_FLAG_UNSET (sink, GST_AFSINK_OPEN);
+  }
+}
+
+/**
+ * gst_afsink_chain:
+ * @pad: the pad this afsink is connected to
+ * @buf: the buffer that has to be absorbed
+ *
+ * take the buffer from the pad and write to file if it's open
+ */
+static void 
+gst_afsink_chain (GstPad *pad, GstBuffer *buf) 
+{
+  GstAFSink *afsink;
+  int ret = 0;
+
+  g_return_if_fail (pad != NULL);
+  g_return_if_fail (GST_IS_PAD (pad));
+  g_return_if_fail (buf != NULL);
+
+  afsink = GST_AFSINK (gst_pad_get_parent (pad));
+/* we use audiofile now
+  if (GST_FLAG_IS_SET (afsink, GST_AFSINK_OPEN))
+  {
+    bytes_written = fwrite (GST_BUFFER_DATA (buf), 1, GST_BUFFER_SIZE (buf), afsink->file);
+    if (bytes_written < GST_BUFFER_SIZE (buf))
+    {
+      printf ("afsink : Warning : %d bytes should be written, only %d bytes written\n",
+      		  GST_BUFFER_SIZE (buf), bytes_written);
+    }
+  }
+*/
+
+  if (!GST_FLAG_IS_SET (afsink, GST_AFSINK_OPEN))
+  {
+    /* it's not open yet, open it */
+    if (!gst_afsink_open_file (afsink))
+          g_print ("WARNING: gstafsink: can't open file !\n");
+//        return FALSE;    Can't return value
+  }
+
+  if (GST_FLAG_IS_SET (afsink, GST_AFSINK_OPEN))
+  {
+    int frameCount = 0;
+
+    frameCount = GST_BUFFER_SIZE (buf) / ((afsink->width / 8) * afsink->channels);
+ //   g_print ("DEBUG: writing %d frames ", frameCount);
+    ret = afWriteFrames (afsink->file, AF_DEFAULT_TRACK, 
+	                 GST_BUFFER_DATA (buf), frameCount);
+    if (ret == AF_BAD_WRITE || ret == AF_BAD_LSEEK)
+    {
+      printf ("afsink : Warning : afWriteFrames returned an error (%d)\n", ret);
+    }
+  }
+
+  gst_buffer_unref (buf);
+
+  g_signal_emit (G_OBJECT (afsink), gst_afsink_signals[SIGNAL_HANDOFF], 0);
+}
+
+static GstElementStateReturn
+gst_afsink_change_state (GstElement *element)
+{
+  g_return_val_if_fail (GST_IS_AFSINK (element), GST_STATE_FAILURE);
+
+  /* if going to NULL? then close the file */
+  if (GST_STATE_PENDING (element) == GST_STATE_NULL) 
+  {
+//    printf ("DEBUG: afsink state change: null pending\n");
+    if (GST_FLAG_IS_SET (element, GST_AFSINK_OPEN))
+    {
+//      g_print ("DEBUG: trying to close the sink file\n");
+      gst_afsink_close_file (GST_AFSINK (element));
+    }
+  } 
+/*
+
+  else
+// this has been moved to the chain function, since it's only then that
+// the caps are set and can be known
+ {
+//    g_print ("DEBUG: it's not going to null\n");
+    if (!GST_FLAG_IS_SET (element, GST_AFSINK_OPEN)) 
+    {
+//      g_print ("DEBUG: GST_AFSINK_OPEN not set\n");
+      if (!gst_afsink_open_file (GST_AFSINK (element)))
+      {
+//        g_print ("DEBUG: element tries to open file\n");
+        return GST_STATE_FAILURE;
+      }
+    }
+  }
+*/
+
+  if (GST_ELEMENT_CLASS (parent_class)->change_state)
+    return GST_ELEMENT_CLASS (parent_class)->change_state (element);
+
+  return GST_STATE_SUCCESS;
+}
+
+/* this function was copied from sinesrc */
+
+static gboolean
+gst_afsink_handle_event (GstPad *pad, GstEvent *event)
+{
+  GstAFSink *afsink;
+
+  afsink = GST_AFSINK (gst_pad_get_parent (pad));
+  GST_DEBUG (0, "DEBUG: afsink: got event\n");
+  gst_afsink_close_file (afsink);
+  GST_FLAG_SET (pad, GST_PAD_EOS);
+
+  return TRUE;
+}
+
+/*
+gboolean
+gst_afsink_factory_init (GstElementFactory *factory)
+{
+  GstPadTemplate *sink_pt;
+  sink_pt = afsink_sink_factory();
+  gst_elementfactory_add_padtemplate (factory, sink_pt);
+    
+  return TRUE;  
+
+}
+*/
+

ext/audiofile/gstafsink.h

@@ -0,0 +1,100 @@
+/* GStreamer
+ * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
+ *                    2000 Wim Taymans <wtay@chello.be>
+ *
+ * gstafsink.h: 
+ *
+ * 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 __GST_AFSINK_H__
+#define __GST_AFSINK_H__
+
+
+#include <config.h>
+#include <gst/gst.h>
+#include <audiofile.h>			/* what else are we to do */
+
+
+#ifdef __cplusplus
+extern "C" {
+#endif /* __cplusplus */
+
+
+//GstElementDetails gst_afsink_details;
+
+
+#define GST_TYPE_AFSINK \
+  (gst_afsink_get_type())
+#define GST_AFSINK(obj) \
+  (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AFSINK,GstAFSink))
+#define GST_AFSINK_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AFSINK,GstAFSinkClass))
+#define GST_IS_AFSINK(obj) \
+  (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AFSINK))
+#define GST_IS_AFSINK_CLASS(obj) \
+  (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AFSINK))
+
+typedef struct _GstAFSink GstAFSink;
+typedef struct _GstAFSinkClass GstAFSinkClass;
+
+typedef enum {
+  GST_AFSINK_OPEN             = GST_ELEMENT_FLAG_LAST,
+
+  GST_AFSINK_FLAG_LAST 	= GST_ELEMENT_FLAG_LAST + 2,
+} GstAFSinkFlags;
+
+struct _GstAFSink {
+  GstElement element;
+  GstPad *sinkpad;
+
+  gchar *filename;
+//  FILE *file;
+
+//  AFfilesetup outfilesetup;
+  AFfilehandle file;
+  int format;
+  int channels;
+  int width;
+  unsigned int rate;
+  gboolean is_signed;
+  int type;				/* type of output, compare to audiofile.h 
+  						 * RAW, AIFF, AIFFC, NEXTSND, WAVE
+  						 */ 
+  // FIXME : endianness is a little cryptic at this point
+  int endianness_data;		/* 4321 or 1234 */
+  int endianness_wanted; /* same thing, but what the output format wants */
+  int endianness_output; /* what the output endianness will be */
+};
+
+struct _GstAFSinkClass {
+  GstElementClass parent_class;
+
+  /* signals */
+  void (*handoff) (GstElement *element,GstPad *pad);
+};
+
+GType gst_afsink_get_type(void);
+//gboolean gst_afsink_factory_init (GstElementFactory *factory);
+
+
+#ifdef __cplusplus
+}
+#endif /* __cplusplus */
+
+
+#endif /* __GST_AFSINK_H__ */

ext/audiofile/gstafsrc.c

@@ -0,0 +1,425 @@
+/* GStreamer
+ * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
+ *                    2000 Wim Taymans <wtay@chello.be>
+ *
+ * gstafsrc.c: 
+ *
+ * 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.
+ */
+
+
+#include <gst/gst.h>
+#include <gst-libs/audio/gstaudio.h>
+#include "gstafsrc.h"
+
+
+static GstElementDetails afsrc_details = {
+  "Audiofile Src",
+  "Src",
+  "Audiofile src for audio/raw",
+  VERSION,
+  "Thomas <thomas@apestaart.org>",
+  "(C) 2001"
+};
+
+
+/* AFSrc signals and args */
+enum {
+  /* FILL ME */
+  SIGNAL_HANDOFF,
+  LAST_SIGNAL
+};
+
+enum {
+  ARG_0,
+  ARG_LOCATION
+};
+
+/* added a src factory function to force audio/raw MIME type */
+/* I think the caps can be broader, we need to change that somehow */
+GST_PADTEMPLATE_FACTORY (afsrc_src_factory,
+  "src",
+  GST_PAD_SRC,
+  GST_PAD_ALWAYS,
+  GST_CAPS_NEW (  
+    "audiofile_src",
+    "audio/raw",
+      "format",             GST_PROPS_STRING ("int"),
+        "law",              GST_PROPS_INT (0),
+        "endianness",       GST_PROPS_INT (G_BYTE_ORDER),
+        "signed",           GST_PROPS_LIST (
+				  GST_PROPS_BOOLEAN (TRUE),
+       				  GST_PROPS_BOOLEAN (FALSE)
+       			    ),
+        "width",            GST_PROPS_INT_RANGE (8, 16),
+        "depth",            GST_PROPS_INT_RANGE (8, 16),
+        "rate",             GST_PROPS_INT_RANGE (4000, 48000), //FIXME
+        "channels",         GST_PROPS_INT_RANGE (1, 2)
+  )
+);
+
+/* we use an enum for the output type arg */
+
+#define GST_TYPE_AFSRC_TYPES (gst_afsrc_types_get_type())
+
+/* FIXME: fix the string ints to be string-converted from the audiofile.h types */
+ 
+static GType
+gst_afsrc_types_get_type (void) 
+{
+  static GType afsrc_types_type = 0;
+  static GEnumValue afsrc_types[] = {
+    {AF_FILE_RAWDATA, "0", "raw PCM"},
+    {AF_FILE_AIFFC,   "1", "AIFFC"},
+    {AF_FILE_AIFF,    "2", "AIFF"},
+    {AF_FILE_NEXTSND, "3", "Next/SND"},
+    {AF_FILE_WAVE,    "4", "Wave"},
+    {0, NULL, NULL},
+  };
+  
+  if (!afsrc_types_type) 
+  {
+    afsrc_types_type = g_enum_register_static ("GstAudiosrcTypes", afsrc_types);
+  }
+  return afsrc_types_type;
+}
+
+static void		gst_afsrc_class_init		(GstAFSrcClass *klass);
+static void		gst_afsrc_init			(GstAFSrc *afsrc);
+
+static gboolean 	gst_afsrc_open_file 		(GstAFSrc *src);
+static void 		gst_afsrc_close_file 		(GstAFSrc *src);
+
+static GstBuffer*	gst_afsrc_get			(GstPad *pad);
+
+static void		gst_afsrc_set_property		(GObject *object, guint prop_id, 
+							 const GValue *value, GParamSpec *pspec);
+static void		gst_afsrc_get_property		(GObject *object, guint prop_id, 
+							 GValue *value, GParamSpec *pspec);
+
+static GstElementStateReturn gst_afsrc_change_state 	(GstElement *element);
+
+static GstElementClass *parent_class = NULL;
+static guint gst_afsrc_signals[LAST_SIGNAL] = { 0 };
+
+GType
+gst_afsrc_get_type (void) 
+{
+  static GType afsrc_type = 0;
+
+  if (!afsrc_type) {
+    static const GTypeInfo afsrc_info = {
+      sizeof (GstAFSrcClass),      NULL,
+      NULL,
+      (GClassInitFunc) gst_afsrc_class_init,
+      NULL,
+      NULL,
+      sizeof (GstAFSrc),
+      0,
+      (GInstanceInitFunc) gst_afsrc_init,
+    };
+    afsrc_type = g_type_register_static (GST_TYPE_ELEMENT, "GstAFSrc", &afsrc_info, 0);
+  }
+  return afsrc_type;
+}
+
+static void
+gst_afsrc_class_init (GstAFSrcClass *klass) 
+{
+  GObjectClass *gobject_class;
+  GstElementClass *gstelement_class;
+
+  gobject_class = (GObjectClass*)klass;
+  gstelement_class = (GstElementClass*)klass;
+
+  parent_class = g_type_class_ref (GST_TYPE_ELEMENT);
+
+  gst_element_install_std_props (
+         GST_ELEMENT_CLASS (klass),
+         "location",     ARG_LOCATION,     G_PARAM_READWRITE,
+         NULL);
+ 
+  gst_afsrc_signals[SIGNAL_HANDOFF] =
+    g_signal_new ("handoff", G_TYPE_FROM_CLASS(klass), G_SIGNAL_RUN_LAST,
+                   G_STRUCT_OFFSET (GstAFSrcClass, handoff), NULL, NULL,
+                   g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0);
+
+
+  gobject_class->set_property = gst_afsrc_set_property;
+  gobject_class->get_property = gst_afsrc_get_property;
+
+  gstelement_class->change_state = gst_afsrc_change_state;
+}
+
+static void 
+gst_afsrc_init (GstAFSrc *afsrc) 
+{
+  // GstPad *pad;   this is now done in the struct
+
+  /* no need for a template, caps are set based on file, right ? */
+  afsrc->srcpad = gst_pad_new ("src", GST_PAD_SRC);
+  gst_element_add_pad (GST_ELEMENT (afsrc), afsrc->srcpad);
+  gst_pad_set_get_function (afsrc->srcpad, gst_afsrc_get);
+
+  afsrc->bytes_per_read = 4096;
+  afsrc->curoffset = 0;
+  afsrc->seq = 0;
+
+  afsrc->filename = NULL;
+  afsrc->file = NULL;
+  /* default values, should never be needed */
+  afsrc->channels = 2;
+  afsrc->width = 16;
+  afsrc->rate = 44100;
+  afsrc->type = AF_FILE_WAVE;
+  afsrc->endianness_data = 1234;
+  afsrc->endianness_wanted = 1234;
+  afsrc->framestamp = 0;
+}
+
+static GstBuffer *
+gst_afsrc_get (GstPad *pad)
+{
+  GstAFSrc *src;
+  GstBuffer *buf;
+
+  glong readbytes, readframes;
+  glong frameCount;
+
+  g_return_val_if_fail (pad != NULL, NULL);
+  src = GST_AFSRC (gst_pad_get_parent (pad));
+
+  buf = gst_buffer_new ();
+  g_return_val_if_fail (buf, NULL);
+  
+  GST_BUFFER_DATA (buf) = (gpointer) g_malloc (src->bytes_per_read);
+ 
+  /* calculate frameCount to read based on file info */
+
+  frameCount = src->bytes_per_read / (src->channels * src->width / 8);
+//  g_print ("DEBUG: gstafsrc: going to read %ld frames\n", frameCount);
+  readframes = afReadFrames (src->file, AF_DEFAULT_TRACK, GST_BUFFER_DATA (buf),
+ 	                    frameCount);
+  readbytes = readframes * (src->channels * src->width / 8);
+  if (readbytes == 0) {
+    gst_element_signal_eos (GST_ELEMENT (src));
+      return NULL;  
+  }
+  
+  GST_BUFFER_SIZE (buf) = readbytes;
+  GST_BUFFER_OFFSET (buf) = src->curoffset;
+
+  src->curoffset += readbytes;
+
+  src->framestamp += gst_audio_frame_length (src->srcpad, buf);
+  GST_BUFFER_TIMESTAMP (buf) = src->framestamp * 1E9
+                             / gst_audio_frame_rate (src->srcpad);
+  printf ("DEBUG: afsrc: timestamp set on output buffer: %f sec\n",
+        GST_BUFFER_TIMESTAMP (buf) / 1E9);
+
+//  g_print("DEBUG: gstafsrc: pushed buffer of %ld bytes\n", readbytes);
+  return buf;
+}
+
+static void
+gst_afsrc_set_property (GObject *object, guint prop_id, const GValue *value, GParamSpec *pspec)
+{
+  GstAFSrc *src;
+
+  /* it's not null if we got it, but it might not be ours */
+  src = GST_AFSRC (object);
+
+  switch (prop_id) {
+    case ARG_LOCATION:
+      if (src->filename)
+	g_free (src->filename);
+      src->filename = g_strdup (g_value_get_string (value));
+      break;
+    default:
+      break;
+  }
+}
+
+static void   
+gst_afsrc_get_property (GObject *object, guint prop_id, GValue *value, GParamSpec *pspec)
+{
+  GstAFSrc *src;
+ 
+  /* it's not null if we got it, but it might not be ours */
+  g_return_if_fail (GST_IS_AFSRC (object));
+ 
+  src = GST_AFSRC (object);
+  
+  switch (prop_id) {
+    case ARG_LOCATION:
+      g_value_set_string (value, src->filename);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static gboolean
+plugin_init (GModule *module, GstPlugin *plugin)
+{
+  GstElementFactory *factory;
+  
+  factory = gst_elementfactory_new ("afsrc", GST_TYPE_AFSRC,
+                                    &afsrc_details);
+  g_return_val_if_fail (factory != NULL, FALSE);
+  
+  gst_elementfactory_add_padtemplate (factory, GST_PADTEMPLATE_GET (afsrc_src_factory));
+
+  gst_plugin_add_feature (plugin, GST_PLUGIN_FEATURE (factory));
+
+  /* load audio support library */
+  if (!gst_library_load ("gstaudio"))
+  {
+    gst_info ("mad: could not load support library: 'gstaudio'\n");
+    return FALSE;
+  }
+  
+  return TRUE;
+}
+
+
+GstPluginDesc plugin_desc = {
+  GST_VERSION_MAJOR,
+  GST_VERSION_MINOR,
+  "afsrc",
+  plugin_init
+};
+
+/* this is where we open the audiofile */
+static gboolean
+gst_afsrc_open_file (GstAFSrc *src)
+{
+  g_return_val_if_fail (!GST_FLAG_IS_SET (src, GST_AFSRC_OPEN), FALSE);
+
+  /* open the file */
+  src->file = afOpenFile (src->filename, "r", AF_NULL_FILESETUP);
+  if (src->file == AF_NULL_FILEHANDLE)
+  {
+    g_print ("ERROR: gstafsrc: Could not open file %s for reading\n",
+		src->filename);
+    gst_element_error (GST_ELEMENT (src), g_strconcat ("opening file \"",
+			src->filename, "\"", NULL));
+    return FALSE;
+  }
+
+  /* get the audiofile audio parameters */
+  {
+    int sampleFormat, sampleWidth;
+    src->channels = afGetChannels (src->file, AF_DEFAULT_TRACK);
+    afGetSampleFormat (src->file, AF_DEFAULT_TRACK, 
+			&sampleFormat, &sampleWidth);
+	switch (sampleFormat)
+	{
+	  case AF_SAMPFMT_TWOSCOMP:
+	    src->is_signed = TRUE;
+	    break;
+	  case AF_SAMPFMT_UNSIGNED:
+	    src->is_signed = FALSE;
+	    break;
+	  case AF_SAMPFMT_FLOAT:
+	  case AF_SAMPFMT_DOUBLE:
+	    GST_DEBUG (GST_CAT_PLUGIN_INFO,
+	    		   "ERROR: float data not supported yet !\n");
+	}
+	src->rate = (guint) afGetRate (src->file, AF_DEFAULT_TRACK);		
+    src->width = sampleWidth;
+    GST_DEBUG (GST_CAT_PLUGIN_INFO, 
+    		   "input file: %d channels, %d width, %d rate, signed %s\n",
+	  			src->channels, src->width, src->rate,
+	  			src->is_signed ? "yes" : "no");
+  }
+  
+  /* set caps on src */
+  //FIXME: add all the possible formats, especially float ! */
+  gst_pad_set_caps (src->srcpad, gst_caps_new (
+    "af_src",
+    "audio/raw",
+    gst_props_new (
+        "format",       GST_PROPS_STRING ("int"),
+          "law",        GST_PROPS_INT (0),              //FIXME
+          "endianness", GST_PROPS_INT (G_BYTE_ORDER),   //FIXME
+          "signed",     GST_PROPS_BOOLEAN (src->is_signed),
+          "width",      GST_PROPS_INT (src->width),
+          "depth",      GST_PROPS_INT (src->width),
+          "rate",       GST_PROPS_INT (src->rate),
+          "channels",   GST_PROPS_INT (src->channels),
+          NULL
+        )
+      ));
+
+  GST_FLAG_SET (src, GST_AFSRC_OPEN);
+
+  return TRUE;
+}
+
+static void
+gst_afsrc_close_file (GstAFSrc *src)
+{
+//  g_print ("DEBUG: closing srcfile...\n");
+  g_return_if_fail (GST_FLAG_IS_SET (src, GST_AFSRC_OPEN));
+//  g_print ("DEBUG: past flag test\n");
+//  if (fclose (src->file) != 0)
+  if (afCloseFile (src->file) != 0)
+  {
+    g_print ("WARNING: afsrc: oops, error closing !\n");
+    perror ("close");
+    gst_element_error (GST_ELEMENT (src), g_strconcat("closing file \"", src->filename, "\"", NULL));
+  }
+  else {
+    GST_FLAG_UNSET (src, GST_AFSRC_OPEN);
+  }
+}
+
+static GstElementStateReturn
+gst_afsrc_change_state (GstElement *element)
+{
+  g_return_val_if_fail (GST_IS_AFSRC (element), GST_STATE_FAILURE);
+
+  /* if going to NULL then close the file */
+  if (GST_STATE_PENDING (element) == GST_STATE_NULL) 
+  {
+//    printf ("DEBUG: afsrc state change: null pending\n");
+    if (GST_FLAG_IS_SET (element, GST_AFSRC_OPEN))
+    {
+//      g_print ("DEBUG: trying to close the src file\n");
+      gst_afsrc_close_file (GST_AFSRC (element));
+    }
+  } 
+  else if (GST_STATE_PENDING (element) == GST_STATE_READY) 
+  {
+//    g_print ("DEBUG: afsrc: ready state pending.  This shouldn't happen at the *end* of a stream\n");
+    if (!GST_FLAG_IS_SET (element, GST_AFSRC_OPEN)) 
+    {
+//      g_print ("DEBUG: GST_AFSRC_OPEN not set\n");
+      if (!gst_afsrc_open_file (GST_AFSRC (element)))
+      {
+//        g_print ("DEBUG: element tries to open file\n");
+        return GST_STATE_FAILURE;
+      }
+    }
+  }
+
+  if (GST_ELEMENT_CLASS (parent_class)->change_state)
+    return GST_ELEMENT_CLASS (parent_class)->change_state (element);
+
+  return GST_STATE_SUCCESS;
+}

ext/audiofile/gstafsrc.h

@@ -0,0 +1,106 @@
+/* GStreamer
+ * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
+ *                    2000 Wim Taymans <wtay@chello.be>
+ *
+ * gstafsrc.h: 
+ *
+ * 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 __GST_AFSRC_H__
+#define __GST_AFSRC_H__
+
+
+#include <config.h>
+#include <gst/gst.h>
+#include <audiofile.h>			/* what else are we to do */
+
+
+#ifdef __cplusplus
+extern "C" {
+#endif /* __cplusplus */
+
+
+//GstElementDetails gst_afsrc_details;
+
+
+#define GST_TYPE_AFSRC \
+  (gst_afsrc_get_type())
+#define GST_AFSRC(obj) \
+  (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AFSRC,GstAFSrc))
+#define GST_AFSRC_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AFSRC,GstAFSrcClass))
+#define GST_IS_AFSRC(obj) \
+  (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AFSRC))
+#define GST_IS_AFSRC_CLASS(obj) \
+  (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AFSRC))
+
+typedef struct _GstAFSrc GstAFSrc;
+typedef struct _GstAFSrcClass GstAFSrcClass;
+
+typedef enum {
+  GST_AFSRC_OPEN             = GST_ELEMENT_FLAG_LAST,
+
+  GST_AFSRC_FLAG_LAST 	= GST_ELEMENT_FLAG_LAST + 2,
+} GstAFSrcFlags;
+
+struct _GstAFSrc {
+  GstElement element;
+  GstPad *srcpad;
+
+  gchar *filename;
+//  FILE *file;
+
+//  AFfilesetup outfilesetup;
+  AFfilehandle file;
+  int format;
+  int channels;
+  int width;
+  unsigned int rate;
+  gboolean is_signed;
+  int type;				/* type of output, compare to audiofile.h 
+  						 * RAW, AIFF, AIFFC, NEXTSND, WAVE
+  						 */ 
+  /* blocking */
+  gulong curoffset;
+  gulong bytes_per_read;
+
+  gulong seq;
+  guint64 framestamp;
+  // FIXME : endianness is a little cryptic at this point
+  int endianness_data;		/* 4321 or 1234 */
+  int endianness_wanted; /* same thing, but what the output format wants */
+  int endianness_output; /* what the output endianness will be */
+};
+
+struct _GstAFSrcClass {
+  GstElementClass parent_class;
+
+  /* signals */
+  void (*handoff) (GstElement *element,GstPad *pad);
+};
+
+GType gst_afsrc_get_type(void);
+//gboolean gst_afsrc_factory_init (GstElementFactory *factory);
+
+
+#ifdef __cplusplus
+}
+#endif /* __cplusplus */
+
+
+#endif /* __GST_AFSRC_H__ */

gst-libs/audio/Makefile.am

@@ -1,4 +1,4 @@
-libdir = $(libdir)/gst
+## libdir = $(libdir)/gst
 
 lib_LTLIBRARIES = libgstaudio.la