From 416c011f1148c714b0eb40cb5340de6f36a4f2c5 Mon Sep 17 00:00:00 2001
From: Wim Taymans <wim.taymans@gmail.com>
Date: Thu, 2 Feb 2006 12:14:35 +0000
Subject: [PATCH] gst-libs/gst/audio/TODO: Updated.

Original commit message from CVS:
* gst-libs/gst/audio/TODO:
Updated.

* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_drain), (gst_base_audio_sink_event):
On EOS, wait till the last sample is played before posting EOS.
---
 ChangeLog                             |  9 ++++++++
 gst-libs/gst/audio/TODO               |  2 --
 gst-libs/gst/audio/gstbaseaudiosink.c | 33 +++++++++++++++++++++++++++
 3 files changed, 42 insertions(+), 2 deletions(-)

diff --git a/ChangeLog b/ChangeLog
index f498e88c58..c3b85acb9b 100644
--- a/ChangeLog
+++ b/ChangeLog
@@ -1,3 +1,12 @@
+2006-02-02  Wim Taymans  <wim@fluendo.com>
+
+	* gst-libs/gst/audio/TODO:
+	Updated.
+
+	* gst-libs/gst/audio/gstbaseaudiosink.c:
+	(gst_base_audio_sink_drain), (gst_base_audio_sink_event):
+	On EOS, wait till the last sample is played before posting EOS.
+
 2006-02-01  Philippe Kalaf <burger at speedy dot org>
 
 	* gst-libs/gst/rtp/gstbasertpdepayload.c:
diff --git a/gst-libs/gst/audio/TODO b/gst-libs/gst/audio/TODO
index ac86f802aa..89fdeb6969 100644
--- a/gst-libs/gst/audio/TODO
+++ b/gst-libs/gst/audio/TODO
@@ -7,8 +7,6 @@ TODO
         is parsed correctly.
       - implement seek/query/convert
       - implement getrange scheduling
-   - on EOS, only post EOS when the complete ringbuffer has been played.      
-   - more accurate clipping of samples outside of the segment
    - simple resampling
    - more accurate master/slave calibration handling
    - faster audio cutoff when going to PAUSED
diff --git a/gst-libs/gst/audio/gstbaseaudiosink.c b/gst-libs/gst/audio/gstbaseaudiosink.c
index 61e879066b..9db8f5f72f 100644
--- a/gst-libs/gst/audio/gstbaseaudiosink.c
+++ b/gst-libs/gst/audio/gstbaseaudiosink.c
@@ -319,6 +319,38 @@ gst_base_audio_sink_get_times (GstBaseSink * bsink, GstBuffer * buffer,
   *end = GST_CLOCK_TIME_NONE;
 }
 
+static gboolean
+gst_base_audio_sink_drain (GstBaseAudioSink * sink)
+{
+  if (!sink->ringbuffer)
+    return TRUE;
+  if (!sink->ringbuffer->spec.rate)
+    return TRUE;
+
+  if (sink->next_sample != -1) {
+    GstClockTime time;
+    GstClock *clock;
+
+    time =
+        gst_util_uint64_scale_int (sink->next_sample, GST_SECOND,
+        sink->ringbuffer->spec.rate);
+
+    GST_OBJECT_LOCK (sink);
+    if ((clock = GST_ELEMENT_CLOCK (sink)) != NULL) {
+      GstClockID id = gst_clock_new_single_shot_id (clock, time);
+
+      GST_OBJECT_UNLOCK (sink);
+
+      GST_DEBUG_OBJECT (sink, "waiting for last sample to play");
+      gst_clock_id_wait (id, NULL);
+      sink->next_sample = -1;
+    } else {
+      GST_OBJECT_UNLOCK (sink);
+    }
+  }
+  return TRUE;
+}
+
 static gboolean
 gst_base_audio_sink_event (GstBaseSink * bsink, GstEvent * event)
 {
@@ -337,6 +369,7 @@ gst_base_audio_sink_event (GstBaseSink * bsink, GstEvent * event)
       /* need to start playback when we reach EOS */
       gst_ring_buffer_start (sink->ringbuffer);
       /* now wait till we played everything */
+      gst_base_audio_sink_drain (sink);
       break;
     default:
       break;