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.

1 files changed, 4 insertions, 6 deletions

diff --git a/sys/oss/gstosssrc.c b/sys/oss/gstosssrc.c
index 9eb24aaf..0e41d247 100644
--- a/sys/oss/gstosssrc.c
+++ b/sys/oss/gstosssrc.c
@@ -34,7 +34,7 @@
 
 #include <gstosssrc.h>
 #include <gstosscommon.h>
-#include <gstossclock.h>
+#include <gst/audio/audioclock.h>
 
 /* elementfactory information */
 static GstElementDetails gst_osssrc_details = {
@@ -197,9 +197,7 @@ gst_osssrc_init (GstOssSrc *osssrc)
   osssrc->buffersize = 4096;
   osssrc->curoffset = 0;
 
-  osssrc->provided_clock = GST_CLOCK (gst_oss_clock_new ("ossclock",
-							 gst_osssrc_get_time,
-							 osssrc));
+  osssrc->provided_clock = gst_audio_clock_new ("ossclock", gst_osssrc_get_time, osssrc);
   gst_object_set_parent (GST_OBJECT (osssrc->provided_clock), GST_OBJECT (osssrc));
   
   osssrc->clock = NULL;
@@ -443,10 +441,10 @@ gst_osssrc_change_state (GstElement *element)
       osssrc->curoffset = 0;
       break;
     case GST_STATE_PAUSED_TO_PLAYING:
-      gst_oss_clock_set_active (osssrc->provided_clock, TRUE);
+      gst_audio_clock_set_active (GST_AUDIO_CLOCK (osssrc->provided_clock), TRUE);
       break;
     case GST_STATE_PLAYING_TO_PAUSED:
-      gst_oss_clock_set_active (osssrc->provided_clock, FALSE);
+      gst_audio_clock_set_active (GST_AUDIO_CLOCK (osssrc->provided_clock), FALSE);
       break;
     case GST_STATE_PAUSED_TO_READY:
       if (GST_FLAG_IS_SET (element, GST_OSSSRC_OPEN))