actually recurse into sndfile if we are able big ladspa cleanups, mainly to comply with the buffer-frames caps proper...

Original commit message from CVS:
* actually recurse into sndfile if we are able
* big ladspa cleanups, mainly to comply with the buffer-frames caps property, but also general
cleanups
- the samplerate prop is gone, if you want to set it explicitly (as in for get-based plugins)
you need to use a filtered connection, just like with buffer-frames
* big float2int and int2float changes for buffer-frames compatibility - I think it's quite a bit
simpler
* make the ossclock general, add it to gstaudio, and use it in sndfile as well

i need to update mimetypes, but that's coming soon. there are some other plugins that don't
support buffer-frames, i guess i need to get around to fixing them as well.

ext/Makefile.am

@@ -273,7 +273,7 @@ SUBDIRS=$(A52DEC_DIR) $(AALIB_DIR) $(ALSA_DIR) \
 	$(MAD_DIR) $(MATROSKA_DIR) $(MIKMOD_DIR) \
 	$(MPEG2DEC_DIR) $(PANGO_DIR) $(RAW1394_DIR) \
 	$(SDL_DIR) $(SHOUT_DIR) $(SIDPLAY_DIR) \
-	$(SMOOTHWAVE_DIR) $(SWFDEC_DIR) $(TARKIN_DIR) \
+	$(SMOOTHWAVE_DIR) $(SNDFILE_DIR) $(SWFDEC_DIR) $(TARKIN_DIR) \
 	$(VORBIS_DIR) $(XVID_DIR) $(SNAPSHOT_DIR)
 
 DIST_SUBDIRS=\

gst-libs/ext/ffmpeg/Tag

@@ -1 +1 @@
-2003-06-09 22:00 GMT
+2003-07-05 22:00 GMT

gst-libs/gst/audio/Makefile.am

@@ -2,10 +2,10 @@ librarydir = $(libdir)/gstreamer-@GST_MAJORMINOR@
 
 library_LTLIBRARIES = libgstaudio.la
 
-libgstaudio_la_SOURCES = audio.c
+libgstaudio_la_SOURCES = audio.c audioclock.c
 
 libgstaudioincludedir = $(includedir)/gstreamer-@GST_MAJORMINOR@/gst/audio
-libgstaudioinclude_HEADERS = audio.h
+libgstaudioinclude_HEADERS = audio.h audioclock.h
 
 libgstaudio_la_LIBADD =
 libgstaudio_la_CFLAGS = $(GST_CFLAGS) $(GST_OPT_CFLAGS)

gst-libs/gst/audio/audio.c

@@ -177,7 +177,7 @@ gst_audio_is_buffer_framed (GstPad* pad, GstBuffer* buf)
 static gboolean
 plugin_init (GModule *module, GstPlugin *plugin)
 {
-  gst_plugin_set_longname (plugin, "Convenience routines for audio plugins");
+  gst_plugin_set_longname (plugin, "Support services for audio plugins");
   return TRUE;
 }
 

gst-libs/gst/audio/audio.h

@@ -20,6 +20,8 @@
 
 #include <gst/gst.h>
 
+#include <gst/audio/audioclock.h>
+
 /* For people that are looking at this source: the purpose of these defines is
  * to make GstCaps a bit easier, in that you don't have to know all of the
  * properties that need to be defined. you can just use these macros. currently

gst-libs/gst/audio/audioclock.c

@@ -0,0 +1,194 @@
+/* GStreamer
+ * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
+ *                    2000 Wim Taymans <wtay@chello.be>
+ *
+ * audioclock.c: Clock for use by audio plugins
+ *
+ * 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 "audioclock.h"
+
+static void 		gst_audio_clock_class_init 	(GstAudioClockClass *klass);
+static void 		gst_audio_clock_init 		(GstAudioClock *clock);
+
+static GstClockTime	gst_audio_clock_get_internal_time (GstClock *clock);
+static GstClockReturn	gst_audio_clock_id_wait_async	(GstClock *clock,
+                                                         GstClockEntry *entry);
+static void		gst_audio_clock_id_unschedule	(GstClock *clock,
+                                                         GstClockEntry *entry);
+
+static GstSystemClockClass *parent_class = NULL;
+/* static guint gst_audio_clock_signals[LAST_SIGNAL] = { 0 }; */
+  
+GType
+gst_audio_clock_get_type (void)
+{ 
+  static GType clock_type = 0;
+	    
+  if (!clock_type) {
+    static const GTypeInfo clock_info = {
+      sizeof (GstAudioClockClass),
+      NULL,
+      NULL,
+      (GClassInitFunc) gst_audio_clock_class_init,
+      NULL,
+      NULL,
+      sizeof (GstAudioClock),
+      4,
+      (GInstanceInitFunc) gst_audio_clock_init,
+      NULL
+    };
+    clock_type = g_type_register_static (GST_TYPE_SYSTEM_CLOCK, "GstAudioClock",
+                                         &clock_info, 0);
+  }
+  return clock_type;
+}
+
+
+static void
+gst_audio_clock_class_init (GstAudioClockClass *klass)
+{
+  GObjectClass *gobject_class;
+  GstObjectClass *gstobject_class;
+  GstClockClass *gstclock_class;
+
+  gobject_class = (GObjectClass*) klass;
+  gstobject_class = (GstObjectClass*) klass;
+  gstclock_class = (GstClockClass*) klass;
+
+  parent_class = g_type_class_ref (GST_TYPE_SYSTEM_CLOCK);
+
+  gstclock_class->get_internal_time	= gst_audio_clock_get_internal_time;
+  gstclock_class->wait_async		= gst_audio_clock_id_wait_async;
+  gstclock_class->unschedule		= gst_audio_clock_id_unschedule;
+}
+
+static void
+gst_audio_clock_init (GstAudioClock *clock)
+{
+  gst_object_set_name (GST_OBJECT (clock), "GstAudioClock");
+
+  clock->prev1 = 0;
+  clock->prev2 = 0;
+}
+
+GstClock*
+gst_audio_clock_new (gchar *name, GstAudioClockGetTimeFunc func, gpointer user_data)
+{
+  GstAudioClock *aclock = GST_AUDIO_CLOCK (g_object_new (GST_TYPE_AUDIO_CLOCK, NULL));
+
+  aclock->func = func;
+  aclock->user_data = user_data;
+  aclock->adjust = 0;
+
+  return (GstClock*)aclock;
+}
+
+void
+gst_audio_clock_set_active (GstAudioClock *aclock, gboolean active)
+{
+  GTimeVal timeval;
+  GstClockTime time;
+  GstClockTime audiotime;
+
+  g_get_current_time (&timeval);
+  time = GST_TIMEVAL_TO_TIME (timeval);
+  audiotime = aclock->func ((GstClock*)aclock, aclock->user_data);
+
+  if (active) {
+    aclock->adjust = time - audiotime;
+  }
+  else {
+    aclock->adjust = audiotime - time;
+  }
+
+  aclock->active = active;
+}
+
+static GstClockTime
+gst_audio_clock_get_internal_time (GstClock *clock)
+{
+  GstAudioClock *aclock = GST_AUDIO_CLOCK (clock);
+  
+  if (aclock->active) {
+    GstClockTime audiotime;
+    
+    audiotime = aclock->func (clock, aclock->user_data) + aclock->adjust;
+
+    return audiotime;
+  }
+  else {
+    GstClockTime time;
+    GTimeVal timeval;
+    
+    g_get_current_time (&timeval);
+    time = GST_TIMEVAL_TO_TIME (timeval);
+
+    return time;
+  }
+}
+
+void
+gst_audio_clock_update_time (GstAudioClock *aclock, GstClockTime time)
+{
+  /* I don't know of a purpose in updating these; perhaps they can be removed */
+  aclock->prev2 = aclock->prev1;
+  aclock->prev1 = time;
+
+  /* FIXME: the wait_async subsystem should be made threadsafe, but I don't want
+   * to lock and unlock a mutex on every iteration... */
+  while (aclock->async_entries) {
+    GstClockEntry *entry = (GstClockEntry*)aclock->async_entries->data;
+    
+    if (entry->time > time)
+      break;
+
+    entry->func ((GstClock*)aclock, time, entry, entry->user_data);
+
+    aclock->async_entries = g_slist_delete_link (aclock->async_entries,
+                                                 aclock->async_entries);
+    /* do I need to free the entry? */
+  }
+}
+
+static gint
+compare_clock_entries (GstClockEntry *entry1, GstClockEntry *entry2)
+{
+  return entry1->time - entry2->time;
+}
+
+static GstClockReturn
+gst_audio_clock_id_wait_async (GstClock *clock, GstClockEntry *entry)
+{
+  GstAudioClock *aclock = (GstAudioClock*)clock;
+  
+  aclock->async_entries = g_slist_insert_sorted (aclock->async_entries,
+                                                 entry,
+                                                 (GCompareFunc)compare_clock_entries);
+
+  /* is this the proper return val? */
+  return GST_CLOCK_EARLY;
+}
+
+static void
+gst_audio_clock_id_unschedule (GstClock *clock, GstClockEntry *entry)
+{
+  GstAudioClock *aclock = (GstAudioClock*)clock;
+  
+  aclock->async_entries = g_slist_remove (aclock->async_entries,
+                                          entry);
+}

gst-libs/gst/audio/audioclock.h

@@ -0,0 +1,83 @@
+/* GStreamer
+ * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
+ *                    2000 Wim Taymans <wtay@chello.be>
+ *
+ * audioclock.h: Clock for use by audio plugins
+ *
+ * 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_AUDIO_CLOCK_H__
+#define __GST_AUDIO_CLOCK_H__
+
+#include <gst/gstsystemclock.h>
+
+#ifdef __cplusplus
+extern "C" {
+#endif /* __cplusplus */
+
+
+#define GST_TYPE_AUDIO_CLOCK \
+  (gst_audio_clock_get_type())
+#define GST_AUDIO_CLOCK(obj) \
+  (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_CLOCK,GstAudioClock))
+#define GST_AUDIO_CLOCK_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIO_CLOCK,GstAudioClockClass))
+#define GST_IS_AUDIO_CLOCK(obj) \
+  (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_CLOCK))
+#define GST_IS_AUDIO_CLOCK_CLASS(obj) \
+  (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIO_CLOCK))
+
+typedef struct _GstAudioClock GstAudioClock;
+typedef struct _GstAudioClockClass GstAudioClockClass;
+
+typedef GstClockTime (*GstAudioClockGetTimeFunc) (GstClock *clock, gpointer user_data);
+
+
+struct _GstAudioClock {
+  GstSystemClock clock;
+
+  GstClockTime prev1, prev2;
+
+  /* --- protected --- */
+  GstAudioClockGetTimeFunc func;
+  gpointer user_data;
+
+  GstClockTimeDiff adjust;
+
+  GSList *async_entries;
+
+  gboolean active;
+};
+
+struct _GstAudioClockClass {
+  GstSystemClockClass parent_class;
+};
+
+GType           gst_audio_clock_get_type 	(void);
+GstClock*	gst_audio_clock_new		(gchar *name, GstAudioClockGetTimeFunc func,
+                                                 gpointer user_data);
+void		gst_audio_clock_set_active 	(GstAudioClock *aclock, gboolean active);
+
+void		gst_audio_clock_update_time	(GstAudioClock *aclock, GstClockTime time);
+
+#ifdef __cplusplus
+}
+#endif /* __cplusplus */
+
+
+#endif /* __GST_AUDIO_CLOCK_H__ */