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authorNobuhiro Iwamatsu <nobuhiro.iwamatsu.yj@renesas.com>2015-03-25 10:47:45 +0900
committerNobuhiro Iwamatsu <nobuhiro.iwamatsu.yj@renesas.com>2015-03-25 10:47:45 +0900
commit1c35920d85e424b3f65aa6df1dbde689dd6ec007 (patch)
tree58b2cacb3674111aad5a4ded694db0cef5cf55f3 /common/recipes-multimedia/gstreamer/gst-plugins-base
commit BSP v1.8.0
Signed-off-by: Nobuhiro Iwamatsu <nobuhiro.iwamatsu.yj@renesas.com>
Diffstat (limited to 'common/recipes-multimedia/gstreamer/gst-plugins-base')
-rw-r--r--common/recipes-multimedia/gstreamer/gst-plugins-base/0001-gstaudiosink-wait-for-gst_ring_buffer_advance-when-p.patch79
1 files changed, 79 insertions, 0 deletions
diff --git a/common/recipes-multimedia/gstreamer/gst-plugins-base/0001-gstaudiosink-wait-for-gst_ring_buffer_advance-when-p.patch b/common/recipes-multimedia/gstreamer/gst-plugins-base/0001-gstaudiosink-wait-for-gst_ring_buffer_advance-when-p.patch
new file mode 100644
index 0000000..7d04c8c
--- /dev/null
+++ b/common/recipes-multimedia/gstreamer/gst-plugins-base/0001-gstaudiosink-wait-for-gst_ring_buffer_advance-when-p.patch
@@ -0,0 +1,79 @@
+From d46fff879acb78b6cd42937864f80e22389e3acb Mon Sep 17 00:00:00 2001
+From: Hideki EIRAKU <hdk@igel.co.jp>
+Date: Fri, 18 May 2012 19:30:27 +0900
+Subject: [PATCH] gstaudiosink: wait for gst_ring_buffer_advance when pausing
+
+If gst_ring_buffer_advance is called after paused, the time of
+audio sink is updated and the base_time will be increased when resuming.
+The base_time change makes a cleared buffer play.
+Gstaudiosrc may have the same problem.
+---
+ gst-libs/gst/audio/gstaudiosink.c | 9 +++++++++
+ gst-libs/gst/audio/gstaudiosink.h | 3 +++
+ 2 files changed, 12 insertions(+)
+
+diff --git a/gst-libs/gst/audio/gstaudiosink.c b/gst-libs/gst/audio/gstaudiosink.c
+index c5fec73..72984bf 100644
+--- a/gst-libs/gst/audio/gstaudiosink.c
++++ b/gst-libs/gst/audio/gstaudiosink.c
+@@ -265,6 +265,8 @@ audioringbuffer_thread_func (GstRingBuffer * buf)
+ gst_ring_buffer_advance (buf, 1);
+ } else {
+ GST_OBJECT_LOCK (abuf);
++ sink->buffer_running = FALSE;
++ g_cond_signal (sink->pause_cond);
+ if (!abuf->running)
+ goto stop_running;
+ if (G_UNLIKELY (g_atomic_int_get (&buf->state) ==
+@@ -280,6 +282,7 @@ audioringbuffer_thread_func (GstRingBuffer * buf)
+ if (!abuf->running)
+ goto stop_running;
+ GST_DEBUG_OBJECT (sink, "continue running");
++ sink->buffer_running = TRUE;
+ GST_OBJECT_UNLOCK (abuf);
+ }
+ }
+@@ -429,6 +432,8 @@ gst_audioringbuffer_activate (GstRingBuffer * buf, gboolean active)
+
+ GST_DEBUG_OBJECT (sink, "starting thread");
+
++ sink->pause_cond = g_cond_new ();
++ sink->buffer_running = TRUE;
+ #if !GLIB_CHECK_VERSION (2, 31, 0)
+ sink->thread =
+ g_thread_create ((GThreadFunc) audioringbuffer_thread_func, buf, TRUE,
+@@ -454,6 +459,7 @@ gst_audioringbuffer_activate (GstRingBuffer * buf, gboolean active)
+
+ /* join the thread */
+ g_thread_join (sink->thread);
++ g_cond_free (sink->pause_cond);
+
+ GST_OBJECT_LOCK (buf);
+ }
+@@ -531,6 +537,9 @@ gst_audioringbuffer_pause (GstRingBuffer * buf)
+ GST_DEBUG_OBJECT (sink, "reset done");
+ }
+
++ if (sink->buffer_running)
++ g_cond_wait (sink->pause_cond, GST_OBJECT_GET_LOCK (buf));
++
+ return TRUE;
+ }
+
+diff --git a/gst-libs/gst/audio/gstaudiosink.h b/gst-libs/gst/audio/gstaudiosink.h
+index 83a4e95..8c32d5a 100644
+--- a/gst-libs/gst/audio/gstaudiosink.h
++++ b/gst-libs/gst/audio/gstaudiosink.h
+@@ -49,6 +49,9 @@ struct _GstAudioSink {
+ /*< private >*/ /* with LOCK */
+ GThread *thread;
+
++ GCond *pause_cond;
++ gboolean buffer_running;
++
+ /*< private >*/
+ gpointer _gst_reserved[GST_PADDING];
+ };
+--
+1.7.10.4
+