gst-libs/gst/audio/gstbaseaudiosink.c: Implement new async_play vmethod to start...

Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init), (gst_base_audio_sink_async_play),
(gst_base_audio_sink_do_play), (gst_base_audio_sink_change_state):
Implement new async_play vmethod to start slaving and allow
playback start in case of async PLAY state changes.
* gst-libs/gst/video/gstvideosink.c: (gst_video_sink_init):
Enable QoS with new method in base class.

diff --git a/ChangeLog b/ChangeLog
index 21ebd6f..55cf3ff 100644 (file)
--- a/ChangeLog
+++ b/ChangeLog
@@ -1,5 +1,16 @@
 2006-03-23  Wim Taymans  <wim@fluendo.com>
 
+       * gst-libs/gst/audio/gstbaseaudiosink.c:
+       (gst_base_audio_sink_class_init), (gst_base_audio_sink_async_play),
+       (gst_base_audio_sink_do_play), (gst_base_audio_sink_change_state):
+       Implement new async_play vmethod to start slaving and allow
+       playback start in case of async PLAY state changes. 
+
+       * gst-libs/gst/video/gstvideosink.c: (gst_video_sink_init):
+       Enable QoS with new method in base class.
+
+2006-03-23  Wim Taymans  <wim@fluendo.com>
+
        Patch by: Julien MOUTTE <julien at moutte dot net>
 
        * gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_query),

diff --git a/common b/common
index f1c7bfd..252846b 160000 (submodule)
--- a/common
+++ b/common
@@ -1 +1 @@
-Subproject commit f1c7bfd24d0fcc4e5113ce3b96b1fac83a9ec560
+Subproject commit 252846b570144570a0aee25b5adefbfac3f5d4eb

diff --git a/gst-libs/gst/audio/gstbaseaudiosink.c b/gst-libs/gst/audio/gstbaseaudiosink.c
index bb445e5..a45b16d 100644 (file)
--- a/gst-libs/gst/audio/gstbaseaudiosink.c
+++ b/gst-libs/gst/audio/gstbaseaudiosink.c
@@ -64,6 +64,8 @@ static void gst_base_audio_sink_set_property (GObject * object, guint prop_id,
 static void gst_base_audio_sink_get_property (GObject * object, guint prop_id,
     GValue * value, GParamSpec * pspec);
 
+static GstStateChangeReturn gst_base_audio_sink_async_play (GstBaseSink *
+    basesink);
 static GstStateChangeReturn gst_base_audio_sink_change_state (GstElement *
     element, GstStateChange transition);
 
@@ -141,6 +143,8 @@ gst_base_audio_sink_class_init (GstBaseAudioSinkClass * klass)
   gstbasesink_class->get_times =
       GST_DEBUG_FUNCPTR (gst_base_audio_sink_get_times);
   gstbasesink_class->set_caps = GST_DEBUG_FUNCPTR (gst_base_audio_sink_setcaps);
+  gstbasesink_class->async_play =
+      GST_DEBUG_FUNCPTR (gst_base_audio_sink_async_play);
 }
 
 static void
@@ -690,6 +694,62 @@ gst_base_audio_sink_callback (GstRingBuffer * rbuf, guint8 * data, guint len,
   //GstBaseAudioSink *sink = GST_BASE_AUDIO_SINK (data);
 }
 
+/* should be called with the LOCK */
+static GstStateChangeReturn
+gst_base_audio_sink_async_play (GstBaseSink * basesink)
+{
+  GstClock *clock;
+  GstClockTime time, base;
+  GstBaseAudioSink *sink;
+
+  sink = GST_BASE_AUDIO_SINK (basesink);
+
+  GST_DEBUG_OBJECT (sink, "ringbuffer may start now");
+  gst_ring_buffer_may_start (sink->ringbuffer, TRUE);
+
+  clock = GST_ELEMENT_CLOCK (sink);
+  if (clock == NULL)
+    goto no_clock;
+
+  /* FIXME, only start slaving when we really start the ringbuffer */
+  /* if we are slaved to a clock, we need to set the initial
+   * calibration */
+  if (clock != sink->provided_clock) {
+    GstClockTime rate_num, rate_denom;
+
+    base = GST_ELEMENT_CAST (sink)->base_time;
+    time = gst_clock_get_internal_time (sink->provided_clock);
+
+    GST_DEBUG_OBJECT (sink,
+        "time: %" GST_TIME_FORMAT " base: %" GST_TIME_FORMAT,
+        GST_TIME_ARGS (time), GST_TIME_ARGS (base));
+
+    /* FIXME, this is not yet accurate enough for smooth playback */
+    gst_clock_get_calibration (sink->provided_clock, NULL, NULL, &rate_num,
+        &rate_denom);
+    /* Does not work yet. */
+    gst_clock_set_calibration (sink->provided_clock, time, base,
+        rate_num, rate_denom);
+
+    gst_clock_set_master (sink->provided_clock, clock);
+  }
+
+no_clock:
+  return GST_STATE_CHANGE_SUCCESS;
+}
+
+static GstStateChangeReturn
+gst_base_audio_sink_do_play (GstBaseAudioSink * sink)
+{
+  GstStateChangeReturn ret;
+
+  GST_OBJECT_LOCK (sink);
+  ret = gst_base_audio_sink_async_play (GST_BASE_SINK_CAST (sink));
+  GST_OBJECT_UNLOCK (sink);
+
+  return ret;
+}
+
 static GstStateChangeReturn
 gst_base_audio_sink_change_state (GstElement * element,
     GstStateChange transition)
@@ -713,46 +773,16 @@ gst_base_audio_sink_change_state (GstElement * element,
       gst_ring_buffer_may_start (sink->ringbuffer, FALSE);
       break;
     case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
-    {
-      GstClock *clock;
-      GstClockTime time, base;
-
-      gst_ring_buffer_may_start (sink->ringbuffer, TRUE);
-
-      GST_OBJECT_LOCK (sink);
-      clock = GST_ELEMENT_CLOCK (sink);
-      if (clock == NULL)
-        goto no_clock;
-
-      /* FIXME, only start slaving when we really start the ringbuffer */
-      /* if we are slaved to a clock, we need to set the initial
-       * calibration */
-      if (clock != sink->provided_clock) {
-        GstClockTime rate_num, rate_denom;
-
-        base = element->base_time;
-        time = gst_clock_get_internal_time (sink->provided_clock);
-
-        GST_DEBUG_OBJECT (sink,
-            "time: %" GST_TIME_FORMAT " base: %" GST_TIME_FORMAT,
-            GST_TIME_ARGS (time), GST_TIME_ARGS (base));
-
-        gst_clock_set_master (sink->provided_clock, clock);
-        /* FIXME, this is not yet accurate enough for smooth playback */
-        gst_clock_get_calibration (sink->provided_clock, NULL, NULL, &rate_num,
-            &rate_denom);
-        /* Does not work yet. */
-        gst_clock_set_calibration (sink->provided_clock,
-            time, element->base_time, rate_num, rate_denom);
-      }
-    no_clock:
-      GST_OBJECT_UNLOCK (sink);
+      gst_base_audio_sink_do_play (sink);
       break;
-    }
     case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
+      /* need to take the lock so we don't interfere with an
+       * async play */
+      GST_OBJECT_LOCK (sink);
       /* ringbuffer cannot start anymore */
       gst_ring_buffer_may_start (sink->ringbuffer, FALSE);
       gst_ring_buffer_pause (sink->ringbuffer);
+      GST_OBJECT_UNLOCK (sink);
       break;
     case GST_STATE_CHANGE_PAUSED_TO_READY:
       gst_ring_buffer_set_flushing (sink->ringbuffer, TRUE);

diff --git a/gst-libs/gst/video/gstvideosink.c b/gst-libs/gst/video/gstvideosink.c
index 0441373..ec0c30f 100644 (file)
--- a/gst-libs/gst/video/gstvideosink.c
+++ b/gst-libs/gst/video/gstvideosink.c
@@ -106,6 +106,7 @@ gst_video_sink_init (GstVideoSink * videosink)
 
   /* 20ms is more than enough, 80-130ms is noticable */
   gst_base_sink_set_max_lateness (GST_BASE_SINK (videosink), 20 * GST_MSECOND);
+  gst_base_sink_set_qos_enabled (GST_BASE_SINK (videosink), TRUE);
 }
 
 static void