- Changed plugins for new APIs

Original commit message from CVS:
- Changed plugins for new APIs
- modularized audiofile.
- added seeking, query and convert functions for mad, mpeg2dec,
avidemux, mpegdemux, mpegparse
- sync updates to oss. removed the ossclock for now

ext/audiofile/Makefile.am

@@ -1,15 +1,10 @@
 plugindir = $(libdir)/gst
 
-plugin_LTLIBRARIES = libgstafsink.la libgstafsrc.la
+plugin_LTLIBRARIES = libgstaf.la
 
-libgstafsink_la_SOURCES = gstafsink.c
+libgstaf_la_SOURCES = gstaf.c gstafsink.c gstafsrc.c gstafparse.c
-libgstafsink_la_CFLAGS = $(GST_CFLAGS) $(AUDIOFILE_CFLAGS)
+libgstaf_la_CFLAGS = $(GST_CFLAGS) $(AUDIOFILE_CFLAGS)
-libgstafsink_la_LIBADD = $(AUDIOFILE_LIBS)
+libgstaf_la_LIBADD = $(AUDIOFILE_LIBS)
-libgstafsink_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
+libgstaf_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
-
-libgstafsrc_la_SOURCES = gstafsrc.c
-libgstafsrc_la_CFLAGS = $(GST_CFLAGS) $(AUDIOFILE_CFLAGS)
-libgstafsrc_la_LIBADD = $(AUDIOFILE_LIBS)
-libgstafsrc_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
 
 noinst_HEADERS = gstafsink.h gstafsrc.h 

ext/audiofile/gstaf.c

@@ -0,0 +1,40 @@
+/* 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.
+ */
+
+
+#include "gstafsrc.h"
+#include "gstafsink.h"
+#include "gstafparse.h"
+
+static gboolean
+plugin_init (GModule *module, GstPlugin *plugin)
+{
+  gst_afsink_plugin_init (module, plugin);
+  gst_afsrc_plugin_init (module, plugin);
+  gst_afparse_plugin_init (module, plugin);
+
+  return TRUE;
+}
+
+GstPluginDesc plugin_desc = {
+  GST_VERSION_MAJOR,
+  GST_VERSION_MINOR,
+  "gstaf",
+  plugin_init
+};

ext/audiofile/gstafparse.c

@@ -180,7 +180,7 @@ gst_afparse_loop(GstElement *element)
   GstAFParse *afparse;
   GstBuffer *buf;
   GstBufferPool *bufpool;
-  gint numframes, frames_to_bytes, frames_per_read, bytes_per_read;
+  gint numframes = 0, frames_to_bytes, frames_per_read, bytes_per_read;
   guint8 *data;
   gboolean bypass_afread = TRUE;
   GstByteStream *bs;
@@ -315,8 +315,8 @@ gst_afparse_get_property (GObject *object, guint prop_id, GValue *value, GParamS
   }
 }
 
-static gboolean
+gboolean
-plugin_init (GModule *module, GstPlugin *plugin)
+gst_afparse_plugin_init (GModule *module, GstPlugin *plugin)
 {
   GstElementFactory *factory;
   
@@ -343,14 +343,6 @@ plugin_init (GModule *module, GstPlugin *plugin)
   return TRUE;
 }
 
-
-GstPluginDesc plugin_desc = {
-  GST_VERSION_MAJOR,
-  GST_VERSION_MINOR,
-  "afparse",
-  plugin_init
-};
-
 /* this is where we open the audiofile */
 static gboolean
 gst_afparse_open_file (GstAFParse *afparse)
@@ -483,7 +475,7 @@ static long
 gst_afparse_vf_seek   (AFvirtualfile *vfile, long offset, int is_relative)
 {
   GstByteStream *bs = (GstByteStream*)vfile->closure;
-  GstSeekType type;
+  GstSeekType method;
   guint64 current_offset = gst_bytestream_tell(bs);
 
   if (!is_relative){
@@ -492,16 +484,15 @@ gst_afparse_vf_seek   (AFvirtualfile *vfile, long offset, int is_relative)
       return offset;
     }
 
-    type = GST_SEEK_BYTEOFFSET_SET;
+    method = GST_SEEK_METHOD_SET;
   }
   else {
     if (offset == 0) return current_offset;
-    type = GST_SEEK_BYTEOFFSET_CUR; 
+    method = GST_SEEK_METHOD_CUR; 
   }
   
-  g_print("doing seek to %d, current offset %lld\n", (gint)offset, current_offset);  
+  if (gst_bytestream_seek(bs, (gint64)offset, method)){
-  if (gst_bytestream_seek(bs, type, (gint64)offset)){
+    g_print("doing seek to %d\n", (gint)offset);
-
     return offset;
   }
   return 0;

ext/audiofile/gstafparse.h

@@ -95,6 +95,8 @@ struct _GstAFParseClass {
   void (*handoff) (GstElement *element,GstPad *pad);
 };
 
+gboolean 	gst_afparse_plugin_init 	(GModule *module, GstPlugin *plugin);
+
 #ifdef __cplusplus
 }
 #endif /* __cplusplus */

ext/audiofile/gstafsink.c

@@ -260,8 +260,8 @@ gst_afsink_get_property (GObject *object, guint prop_id, GValue *value, GParamSp
   }
 }
 
-static gboolean
+gboolean
-plugin_init (GModule *module, GstPlugin *plugin)
+gst_afsink_plugin_init (GModule *module, GstPlugin *plugin)
 {
   GstElementFactory *factory;
   
@@ -276,14 +276,6 @@ plugin_init (GModule *module, GstPlugin *plugin)
   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)

ext/audiofile/gstafsink.h

@@ -53,7 +53,7 @@ typedef struct _GstAFSink GstAFSink;
 typedef struct _GstAFSinkClass GstAFSinkClass;
 
 typedef enum {
-  GST_AFSINK_OPEN             = GST_ELEMENT_FLAG_LAST,
+  GST_AFSINK_OPEN       = GST_ELEMENT_FLAG_LAST,
 
   GST_AFSINK_FLAG_LAST 	= GST_ELEMENT_FLAG_LAST + 2,
 } GstAFSinkFlags;
@@ -88,8 +88,9 @@ struct _GstAFSinkClass {
   void (*handoff) (GstElement *element,GstPad *pad);
 };
 
-GType gst_afsink_get_type(void);
+GType 		gst_afsink_get_type	(void);
-/*gboolean gst_afsink_factory_init (GstElementFactory *factory); */
+gboolean 	gst_afsink_plugin_init 	(GModule *module, GstPlugin *plugin);
+
 
 
 #ifdef __cplusplus

ext/audiofile/gstafsrc.c

@@ -271,8 +271,8 @@ gst_afsrc_get_property (GObject *object, guint prop_id, GValue *value, GParamSpe
   }
 }
 
-static gboolean
+gboolean
-plugin_init (GModule *module, GstPlugin *plugin)
+gst_afsrc_plugin_init (GModule *module, GstPlugin *plugin)
 {
   GstElementFactory *factory;
   
@@ -295,13 +295,6 @@ plugin_init (GModule *module, GstPlugin *plugin)
 }
 
 
-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)

ext/audiofile/gstafsrc.h

@@ -94,8 +94,10 @@ struct _GstAFSrcClass {
   void (*handoff) (GstElement *element,GstPad *pad);
 };
 
-GType gst_afsrc_get_type(void);
+GType 		gst_afsrc_get_type	(void);
-/*gboolean gst_afsrc_factory_init (GstElementFactory *factory);*/
+gboolean        gst_afsrc_plugin_init  (GModule *module, GstPlugin *plugin);
+
+
 
 
 #ifdef __cplusplus

ext/sdl/sdlvideosink.c

@@ -190,6 +190,7 @@ gst_sdlvideosink_init (GstSDLVideoSink *sdlvideosink)
   GST_ELEMENT (sdlvideosink)->setclockfunc    = gst_sdlvideosink_set_clock;
 
   GST_FLAG_SET(sdlvideosink, GST_ELEMENT_THREAD_SUGGESTED);
+  GST_FLAG_SET(sdlvideosink, GST_ELEMENT_EVENT_AWARE);
 }
 
 
@@ -399,7 +400,7 @@ static void
 gst_sdlvideosink_chain (GstPad *pad, GstBuffer *buf)
 {
   GstSDLVideoSink *sdlvideosink;
-  SDL_Event event;
+  SDL_Event sdl_event;
 
   g_return_if_fail (pad != NULL);
   g_return_if_fail (GST_IS_PAD (pad));
@@ -407,24 +408,42 @@ gst_sdlvideosink_chain (GstPad *pad, GstBuffer *buf)
 
   sdlvideosink = GST_SDLVIDEOSINK (gst_pad_get_parent (pad));
 
-  GST_DEBUG (0,"videosink: clock wait: %llu", GST_BUFFER_TIMESTAMP(buf));
 
-  while (SDL_PollEvent(&event))
+  while (SDL_PollEvent(&sdl_event))
   {
-    switch(event.type)
+    switch(sdl_event.type)
     {
       case SDL_VIDEORESIZE:
         /* create a SDL window of the size requested by the user */
-        sdlvideosink->window_width = event.resize.w;
+        sdlvideosink->window_width = sdl_event.resize.w;
-        sdlvideosink->window_height = event.resize.h;
+        sdlvideosink->window_height = sdl_event.resize.h;
         gst_sdlvideosink_create(sdlvideosink, FALSE);
         break;
     }
   }
 
+  if (GST_IS_EVENT (buf)) {
+    GstEvent *event = GST_EVENT (buf);
+    gint64 offset;
+
+    switch (GST_EVENT_TYPE (event)) {
+      case GST_EVENT_DISCONTINUOUS:
+	offset = GST_EVENT_DISCONT_OFFSET (event, 0).value;
+	g_print ("sdl discont %lld\n", offset);
+	gst_clock_handle_discont (sdlvideosink->clock, (guint64) GST_EVENT_DISCONT_OFFSET (event, 0).value);
+	break;
+      default:
+	gst_pad_event_default (pad, event);
+	break;
+    }
+    gst_event_free (event);
+    return;
+  }
+
+  GST_DEBUG (0,"videosink: clock wait: %llu", GST_BUFFER_TIMESTAMP(buf));
   if (sdlvideosink->clock) {
     gst_element_clock_wait (GST_ELEMENT (sdlvideosink),
-		  sdlvideosink->clock, GST_BUFFER_TIMESTAMP (buf));
+		  sdlvideosink->clock, GST_BUFFER_TIMESTAMP (buf), NULL);
   }
 
   if (!gst_sdlvideosink_lock(sdlvideosink))