More work on the audiosink, mostly debugging and a race in shutdown.

Original commit message from CVS:
* docs/design-audiosinks.txt:
* gst-libs/gst/audio/TODO:
* 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_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):
More work on the audiosink, mostly debugging and a race in
shutdown.

ChangeLog

@@ -1,3 +1,34 @@
+2005-05-05  Wim Taymans  <wim@fluendo.com>
+
+	* docs/design-audiosinks.txt:
+	* gst-libs/gst/audio/TODO:
+	* 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_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):
+	More work on the audiosink, mostly debugging and a race in
+	shutdown.
+
 2005-04-28  Wim Taymans  <wim@fluendo.com>
 
 	* ext/ogg/gstoggdemux.c: (gst_ogg_pad_typefind),

docs/design-audiosinks.txt

@@ -49,8 +49,8 @@ Design:
   The ringbuffer is a contiguous piece of memory divided into segtotal
   pieces of segments. Each segment has segsize bytes.
 
-         play position                    write position
+         play position 
-           v                                 v
+           v          
    +---+---+---+-------------------------------------+----------+
    + 0 | 1 | 2 | ....                                | segtotal |
    +---+---+---+-------------------------------------+----------+
@@ -58,14 +58,9 @@ Design:
      segsize bytes = N samples * bytes_per_sample.
 
   
-  The ringbuffer has a play and write position, which is expressed in
+  The ringbuffer has a play position, which is expressed in
   segments. The play position is where the device is currently reading
-  samples and the write position is where new samples can be written
+  samples from the buffer.
-  into the buffer.
-
-  The latency of the ringbuffer is the distance between the play and
-  write position. The lowest latency is the size of a segment, thus
-  smaller segment sizes allow for lower latency. 
 
   The ringbuffer can be put to the PLAYING or STOPPED state. 
   
@@ -85,6 +80,12 @@ Design:
   The ringbuffer is implemented with lockfree atomic operations, especially
   on the reading side so that low-latency operations are possible.
 
+  Whenever new samples are to be put into the ringbuffer, the position of the
+  read pointer is taken. The required write position is taken and the diff
+  is made between the required qnd actual position. If the defference is <0,
+  the sample is too late. If the difference is bigger than segtotal, the
+  writing part has to wait for the play pointer to advance. 
+
 
 Scheduling:
 

gst-libs/gst/audio/TODO

@@ -3,11 +3,8 @@ TODO
 
 - audio base classes:
    - GstBaseAudioSink
-      - parse caps into rinbuffer spec, also mase sure surround sound
+      - parse caps into rinbuffer spec, also make sure surround sound
         is parsed correctly.
       - implement seek/query/convert
-      - implement clocks
       - implement getrange scheduling
-   - GstRingBuffer
-      - copy samples to right position in ringbuffer
       

gst-libs/gst/audio/gstaudiosink.c

@@ -183,16 +183,15 @@ audioringbuffer_thread_func (GstRingBuffer * buf)
       gst_ringbuffer_advance (buf, 1);
     } else {
       GST_LOCK (abuf);
+      if (!abuf->running)
+        goto stop_running;
       GST_DEBUG ("signal wait");
       GST_AUDIORINGBUFFER_SIGNAL (buf);
       GST_DEBUG ("wait for action");
       GST_AUDIORINGBUFFER_WAIT (buf);
       GST_DEBUG ("got signal");
-      if (!abuf->running) {
+      if (!abuf->running)
-        GST_UNLOCK (abuf);
+        goto stop_running;
-        GST_DEBUG ("stop running");
-        break;
-      }
       GST_DEBUG ("continue running");
       GST_UNLOCK (abuf);
     }
@@ -207,6 +206,12 @@ no_function:
     GST_DEBUG ("no write function, exit thread");
     return;
   }
+stop_running:
+  {
+    GST_UNLOCK (abuf);
+    GST_DEBUG ("stop running, exit thread");
+    return;
+  }
 }
 
 static void

gst-libs/gst/audio/gstbaseaudiosink.c

@@ -65,7 +65,7 @@ static GstFlowReturn gst_baseaudiosink_preroll (GstBaseSink * bsink,
     GstBuffer * buffer);
 static GstFlowReturn gst_baseaudiosink_render (GstBaseSink * bsink,
     GstBuffer * buffer);
-static void gst_baseaudiosink_event (GstBaseSink * bsink, GstEvent * event);
+static gboolean gst_baseaudiosink_event (GstBaseSink * bsink, GstEvent * event);
 static void gst_baseaudiosink_get_times (GstBaseSink * bsink,
     GstBuffer * buffer, GstClockTime * start, GstClockTime * end);
 static gboolean gst_baseaudiosink_setcaps (GstBaseSink * bsink, GstCaps * caps);
@@ -425,7 +425,7 @@ gst_baseaudiosink_get_times (GstBaseSink * bsink, GstBuffer * buffer,
   *end = GST_CLOCK_TIME_NONE;
 }
 
-static void
+static gboolean
 gst_baseaudiosink_event (GstBaseSink * bsink, GstEvent * event)
 {
   GstBaseAudioSink *sink = GST_BASEAUDIOSINK (bsink);
@@ -458,6 +458,7 @@ gst_baseaudiosink_event (GstBaseSink * bsink, GstEvent * event)
     default:
       break;
   }
+  return TRUE;
 }
 
 static GstFlowReturn

gst-libs/gst/audio/gstringbuffer.c

@@ -222,6 +222,9 @@ gst_ringbuffer_release (GstRingBuffer * buf)
   if (rclass->release)
     res = rclass->release (buf);
 
+  /* signal any waiters */
+  GST_RINGBUFFER_SIGNAL (buf);
+
   if (!res) {
     buf->acquired = TRUE;
   } else {