[gst-cvs] CVS: gst-plugins/ext/audiofile Makefile.am,NONE,1.1 README,NONE,1.1 gstafsink.c,NONE,1.1 gstafsink.h,NONE,1.1 gstafsrc.c,NONE,1.1 gstafsrc.h,NONE,1.1

[Thomas V. S. <tho...@us...>]

Update of /cvsroot/gstreamer/gst-plugins/ext/audiofile
In directory usw-pr-cvs1:/tmp/cvs-serv18654/ext/audiofile

Added Files:
	Makefile.am README gstafsink.c gstafsink.h gstafsrc.c 
	gstafsrc.h 
Log Message:
fixed some GST_LIBS stuff
added audiofile
added gst-libs/audio building


--- NEW FILE: Makefile.am ---
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)


--- NEW FILE: README ---
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

--- NEW FILE: gstafsink.c ---
/* GStreamer
 * Copyright (C) 1999,2000 Erik Walthinsen <om...@cs...>
 *                    2000 Wim Taymans <wt...@ch...>
 *
 * 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 <th...@ap...>",
  "(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;  

}
*/


--- NEW FILE: gstafsink.h ---
/* GStreamer
 * Copyright (C) 1999,2000 Erik Walthinsen <om...@cs...>
 *                    2000 Wim Taymans <wt...@ch...>
 *
 * 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__ */

--- NEW FILE: gstafsrc.c ---
/* GStreamer
 * Copyright (C) 1999,2000 Erik Walthinsen <om...@cs...>
 *                    2000 Wim Taymans <wt...@ch...>
 *
 * 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 <th...@ap...>",
  "(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;
}

--- NEW FILE: gstafsrc.h ---
/* GStreamer
 * Copyright (C) 1999,2000 Erik Walthinsen <om...@cs...>
 *                    2000 Wim Taymans <wt...@ch...>
 *
 * 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__ */