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

1 files changed, 425 insertions, 0 deletions

diff --git a/ext/audiofile/gstafsrc.c b/ext/audiofile/gstafsrc.c
new file mode 100644
index 00000000..ceb548f6
--- /dev/null
+++ b/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;
+}