gst-libs/gst/audio/: Various small cleanups.

Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosink.c:
(gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init),
(audioringbuffer_thread_func), (gst_audioringbuffer_init),
(gst_audioringbuffer_acquire), (gst_audioringbuffer_release),
(gst_audioringbuffer_play), (gst_audioringbuffer_stop),
(gst_audioringbuffer_delay), (gst_audiosink_class_init),
(gst_audiosink_create_ringbuffer):
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_baseaudiosink_class_init), (gst_baseaudiosink_init),
(gst_baseaudiosink_get_clock), (gst_baseaudiosink_get_time),
(gst_baseaudiosink_set_property), (build_linear_format),
(debug_spec_caps), (debug_spec_buffer),
(gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times),
(gst_baseaudiosink_event), (gst_baseaudiosink_preroll),
(gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer),
(gst_baseaudiosink_callback), (gst_baseaudiosink_change_state):
* gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type),
(gst_ringbuffer_set_callback), (gst_ringbuffer_acquire),
(gst_ringbuffer_release), (gst_ringbuffer_is_acquired),
(gst_ringbuffer_play), (gst_ringbuffer_pause),
(gst_ringbuffer_stop), (gst_ringbuffer_delay),
(gst_ringbuffer_played_samples), (gst_ringbuffer_set_sample),
(wait_segment), (gst_ringbuffer_commit),
(gst_ringbuffer_prepare_read), (gst_ringbuffer_advance),
(gst_ringbuffer_clear):
Various small cleanups.

* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_chain),
(gst_audio_convert_change_state):
* gst/subparse/gstsubparse.c: (gst_subparse_chain):
No need to take the locks anymore.

ChangeLog

@@ -1,3 +1,37 @@
+2005-05-25  Wim Taymans  <wim@fluendo.com>
+
+	* gst-libs/gst/audio/gstaudiosink.c:
+	(gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init),
+	(audioringbuffer_thread_func), (gst_audioringbuffer_init),
+	(gst_audioringbuffer_acquire), (gst_audioringbuffer_release),
+	(gst_audioringbuffer_play), (gst_audioringbuffer_stop),
+	(gst_audioringbuffer_delay), (gst_audiosink_class_init),
+	(gst_audiosink_create_ringbuffer):
+	* gst-libs/gst/audio/gstbaseaudiosink.c:
+	(gst_baseaudiosink_class_init), (gst_baseaudiosink_init),
+	(gst_baseaudiosink_get_clock), (gst_baseaudiosink_get_time),
+	(gst_baseaudiosink_set_property), (build_linear_format),
+	(debug_spec_caps), (debug_spec_buffer),
+	(gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times),
+	(gst_baseaudiosink_event), (gst_baseaudiosink_preroll),
+	(gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer),
+	(gst_baseaudiosink_callback), (gst_baseaudiosink_change_state):
+	* gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type),
+	(gst_ringbuffer_set_callback), (gst_ringbuffer_acquire),
+	(gst_ringbuffer_release), (gst_ringbuffer_is_acquired),
+	(gst_ringbuffer_play), (gst_ringbuffer_pause),
+	(gst_ringbuffer_stop), (gst_ringbuffer_delay),
+	(gst_ringbuffer_played_samples), (gst_ringbuffer_set_sample),
+	(wait_segment), (gst_ringbuffer_commit),
+	(gst_ringbuffer_prepare_read), (gst_ringbuffer_advance),
+	(gst_ringbuffer_clear):
+	Various small cleanups.
+
+	* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_chain),
+	(gst_audio_convert_change_state):
+	* gst/subparse/gstsubparse.c: (gst_subparse_chain):
+	No need to take the locks anymore.
+
 2005-05-25  Wim Taymans  <wim@fluendo.com>
 
 	* gst/playback/gstdecodebin.c: (gst_decode_bin_init),

gst-libs/gst/audio/gstaudiosink.c

@@ -294,6 +294,7 @@ gst_audioringbuffer_release (GstRingBuffer * buf)
 
   /* free the buffer */
   gst_buffer_unref (buf->data);
+  buf->data = NULL;
 
   if (csink->close)
     result = csink->close (sink);

gst-libs/gst/audio/gstbaseaudiosink.c

@@ -457,6 +457,7 @@ gst_baseaudiosink_event (GstBaseSink * bsink, GstEvent * event)
       sample = 0;
 
     have_value:
+      GST_DEBUG ("discont now at %lld", sample);
       gst_ringbuffer_set_sample (sink->ringbuffer, sample);
       break;
     }
@@ -483,7 +484,8 @@ gst_baseaudiosink_render (GstBaseSink * bsink, GstBuffer * buf)
 
   offset = GST_BUFFER_OFFSET (buf);
 
-  GST_DEBUG ("in offset %llu, time %lld", offset, GST_BUFFER_TIMESTAMP (buf));
+  GST_DEBUG ("in offset %llu, time %" GST_TIME_FORMAT, offset,
+      GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)));
   if (!gst_ringbuffer_is_acquired (sink->ringbuffer))
     goto wrong_state;
 

gst/audioconvert/gstaudioconvert.c

@@ -244,8 +244,6 @@ gst_audio_convert_chain (GstPad * pad, GstBuffer * buf)
    * - convert rate and channels
    * - convert back to output format
    */
-  GST_STREAM_LOCK (pad);
-
   if (!GST_RPAD_CAPS (this->sink)) {
     goto not_negotiated;
   } else if (!GST_RPAD_CAPS (this->src)) {
@@ -262,8 +260,6 @@ gst_audio_convert_chain (GstPad * pad, GstBuffer * buf)
 
   ret = gst_pad_push (this->src, buf);
 
-  GST_STREAM_UNLOCK (pad);
-
   return ret;
 
 not_negotiated:
@@ -271,7 +267,6 @@ not_negotiated:
     GST_ELEMENT_ERROR (this, CORE, NEGOTIATION, (NULL),
         ("Pad not negotiated before chain function was called"));
     gst_buffer_unref (buf);
-    GST_STREAM_UNLOCK (pad);
     return GST_FLOW_NOT_NEGOTIATED;
   }
 no_format:
@@ -279,7 +274,6 @@ no_format:
     GST_ELEMENT_ERROR (this, CORE, NEGOTIATION, (NULL),
         ("Could not negotiate format"));
     gst_buffer_unref (buf);
-    GST_STREAM_UNLOCK (pad);
     return GST_FLOW_ERROR;
   }
 }
@@ -648,12 +642,10 @@ gst_audio_convert_change_state (GstElement * element)
 
   switch (transition) {
     case GST_STATE_PAUSED_TO_READY:
-      GST_STREAM_LOCK (this->sink);
       this->convert_internal = NULL;
       gst_audio_convert_unset_matrix (this);
       gst_caps_replace (&GST_RPAD_CAPS (this->sink), NULL);
       gst_caps_replace (&GST_RPAD_CAPS (this->src), NULL);
-      GST_STREAM_UNLOCK (this->sink);
       break;
     default:
       break;

gst/subparse/gstsubparse.c

@@ -618,12 +618,8 @@ gst_subparse_chain (GstPad * sinkpad, GstBuffer * buf)
   GST_DEBUG ("gst_subparse_chain");
   self = GST_SUBPARSE (GST_OBJECT_PARENT (sinkpad));
 
-  GST_STREAM_LOCK (sinkpad);
-
   ret = handle_buffer (self, buf);
 
-  GST_STREAM_UNLOCK (sinkpad);
-
   return ret;
 }