audio: clean up headers

diff --git a/gst-libs/gst/audio/gstaudioclock.c b/gst-libs/gst/audio/gstaudioclock.c
index 9d6e485..44879e3 100644 (file)
--- a/gst-libs/gst/audio/gstaudioclock.c
+++ b/gst-libs/gst/audio/gstaudioclock.c
@@ -102,7 +102,7 @@ gst_audio_clock_init (GstAudioClock * clock)
 {
   GST_DEBUG_OBJECT (clock, "init");
   clock->last_time = 0;
-  clock->abidata.ABI.time_offset = 0;
+  clock->time_offset = 0;
   GST_OBJECT_FLAG_SET (clock, GST_CLOCK_FLAG_CAN_SET_MASTER);
 }
 
@@ -111,9 +111,9 @@ gst_audio_clock_dispose (GObject * object)
 {
   GstAudioClock *clock = GST_AUDIO_CLOCK (object);
 
-  if (clock->abidata.ABI.destroy_notify && clock->user_data)
-    clock->abidata.ABI.destroy_notify (clock->user_data);
-  clock->abidata.ABI.destroy_notify = NULL;
+  if (clock->destroy_notify && clock->user_data)
+    clock->destroy_notify (clock->user_data);
+  clock->destroy_notify = NULL;
   clock->user_data = NULL;
 
   G_OBJECT_CLASS (parent_class)->dispose (object);
@@ -168,7 +168,7 @@ gst_audio_clock_new_full (const gchar * name, GstAudioClockGetTimeFunc func,
 
   aclock->func = func;
   aclock->user_data = user_data;
-  aclock->abidata.ABI.destroy_notify = destroy_notify;
+  aclock->destroy_notify = destroy_notify;
 
   return (GstClock *) aclock;
 }
@@ -193,7 +193,7 @@ gst_audio_clock_reset (GstAudioClock * clock, GstClockTime time)
   else
     time_offset = -(time - clock->last_time);
 
-  clock->abidata.ABI.time_offset = time_offset;
+  clock->time_offset = time_offset;
 
   GST_DEBUG_OBJECT (clock,
       "reset clock to %" GST_TIME_FORMAT ", last %" GST_TIME_FORMAT ", offset %"
@@ -219,7 +219,7 @@ gst_audio_clock_get_internal_time (GstClock * clock)
   if (result == GST_CLOCK_TIME_NONE) {
     result = aclock->last_time;
   } else {
-    result += aclock->abidata.ABI.time_offset;
+    result += aclock->time_offset;
     /* clock must be increasing */
     if (aclock->last_time < result)
       aclock->last_time = result;
@@ -256,7 +256,7 @@ gst_audio_clock_get_time (GstClock * clock)
   result = aclock->func (clock, aclock->user_data);
   if (result == GST_CLOCK_TIME_NONE) {
     GST_DEBUG_OBJECT (clock, "no time, reuse last");
-    result = aclock->last_time - aclock->abidata.ABI.time_offset;
+    result = aclock->last_time - aclock->time_offset;
   }
 
   GST_DEBUG_OBJECT (clock,
@@ -285,7 +285,7 @@ gst_audio_clock_adjust (GstClock * clock, GstClockTime time)
 
   aclock = GST_AUDIO_CLOCK_CAST (clock);
 
-  result = time + aclock->abidata.ABI.time_offset;
+  result = time + aclock->time_offset;
 
   return result;
 }

diff --git a/gst-libs/gst/audio/gstaudioclock.h b/gst-libs/gst/audio/gstaudioclock.h
index 906c802..cd96dc0 100644 (file)
--- a/gst-libs/gst/audio/gstaudioclock.h
+++ b/gst-libs/gst/audio/gstaudioclock.h
@@ -68,19 +68,14 @@ struct _GstAudioClock {
 
   /*< protected >*/
   GstAudioClockGetTimeFunc func;
-  gpointer user_data;
-
-  GstClockTime last_time;
+  gpointer                 user_data;
+  GDestroyNotify           destroy_notify;
 
   /*< private >*/
-  union {
-    struct {
-      GstClockTimeDiff   time_offset;
-      GDestroyNotify     destroy_notify;
-    } ABI;
-    /* adding + 0 to mark ABI change to be undone later */
-    gpointer _gst_reserved[GST_PADDING + 0];
-  } abidata;
+  GstClockTime             last_time;
+  GstClockTimeDiff         time_offset;
+
+  gpointer _gst_reserved[GST_PADDING];
 };
 
 struct _GstAudioClockClass {

diff --git a/gst-libs/gst/audio/gstbaseaudiosink.c b/gst-libs/gst/audio/gstbaseaudiosink.c
index 6d09d12..cd42528 100644 (file)
--- a/gst-libs/gst/audio/gstbaseaudiosink.c
+++ b/gst-libs/gst/audio/gstbaseaudiosink.c
@@ -826,7 +826,7 @@ gst_base_audio_sink_drain (GstBaseAudioSink * sink)
 
   /* if PLAYING is interrupted,
    * arrange to have clock running when going to PLAYING again */
-  g_atomic_int_set (&sink->abidata.ABI.eos_rendering, 1);
+  g_atomic_int_set (&sink->eos_rendering, 1);
 
   /* need to start playback before we can drain, but only when
    * we have successfully negotiated a format and thus acquired the
@@ -845,7 +845,7 @@ gst_base_audio_sink_drain (GstBaseAudioSink * sink)
 
     GST_DEBUG_OBJECT (sink, "drained audio");
   }
-  g_atomic_int_set (&sink->abidata.ABI.eos_rendering, 0);
+  g_atomic_int_set (&sink->eos_rendering, 0);
   return TRUE;
 }
 
@@ -1570,8 +1570,7 @@ gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf)
       GST_TIME_ARGS (render_start), GST_TIME_ARGS (render_stop));
 
   /* bring to position in the ringbuffer */
-  time_offset =
-      GST_AUDIO_CLOCK_CAST (sink->provided_clock)->abidata.ABI.time_offset;
+  time_offset = GST_AUDIO_CLOCK_CAST (sink->provided_clock)->time_offset;
   GST_DEBUG_OBJECT (sink,
       "time offset %" GST_TIME_FORMAT, GST_TIME_ARGS (time_offset));
   if (render_start > time_offset)
@@ -1953,7 +1952,7 @@ gst_base_audio_sink_change_state (GstElement * element,
 
       gst_ring_buffer_may_start (sink->ringbuffer, TRUE);
       if (GST_BASE_SINK_CAST (sink)->pad_mode == GST_ACTIVATE_PULL ||
-          g_atomic_int_get (&sink->abidata.ABI.eos_rendering) || eos) {
+          g_atomic_int_get (&sink->eos_rendering) || eos) {
         /* we always start the ringbuffer in pull mode immediatly */
         /* sync rendering on eos needs running clock,
          * and others need running clock when finished rendering eos */

diff --git a/gst-libs/gst/audio/gstbaseaudiosink.h b/gst-libs/gst/audio/gstbaseaudiosink.h
index 95c0774..0511a0f 100644 (file)
--- a/gst-libs/gst/audio/gstbaseaudiosink.h
+++ b/gst-libs/gst/audio/gstbaseaudiosink.h
@@ -124,17 +124,13 @@ struct _GstBaseAudioSink {
   gboolean       provide_clock;
   GstClock      *provided_clock;
 
+  /* with g_atomic_; currently rendering eos */
+  gboolean       eos_rendering;
+
   /*< private >*/
   GstBaseAudioSinkPrivate *priv;
 
-  union {
-    struct {
-      /*< protected >*/
-      /* with g_atomic_; currently rendering eos */
-      gboolean   eos_rendering;
-    } ABI;
-    gpointer _gst_reserved[GST_PADDING - 1];
-  } abidata;
+  gpointer _gst_reserved[GST_PADDING];
 };
 
 /**
@@ -158,9 +154,8 @@ struct _GstBaseAudioSinkClass {
   /* subclass payloader */
   GstBuffer*     (*payload)            (GstBaseAudioSink *sink,
                                         GstBuffer        *buffer);
-
   /*< private >*/
-  gpointer _gst_reserved[GST_PADDING - 1];
+  gpointer _gst_reserved[GST_PADDING];
 };
 
 GType gst_base_audio_sink_get_type(void);