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gst-libs/gst/audio/: Only start playback if we are playing. should fix #330748.

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Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_change_state):
* gst-libs/gst/audio/gstringbuffer.c: (wait_segment),
(gst_ring_buffer_may_start):
* gst-libs/gst/audio/gstringbuffer.h:
Only start playback if we are playing.
should fix #330748.
This commit is contained in:
Wim Taymans 2006-03-17 17:48:33 +00:00
parent 17b06edc28
commit 82fd38fbcf
4 changed files with 49 additions and 1 deletions
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10
ChangeLog
View file

@ -1,3 +1,13 @@
2006-03-17 Wim Taymans <wim@fluendo.com>
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_change_state):
* gst-libs/gst/audio/gstringbuffer.c: (wait_segment),
(gst_ring_buffer_may_start):
* gst-libs/gst/audio/gstringbuffer.h:
Only start playback if we are playing.
should fix #330748.
2006-03-17 Jan Schmidt <thaytan@mad.scientist.com>
* ext/alsa/gstalsasink.c: (gst_alsasink_getcaps):

7
gst-libs/gst/audio/gstbaseaudiosink.c
View file

@ -707,14 +707,17 @@ gst_base_audio_sink_change_state (GstElement * element,
goto open_failed;
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
gst_ring_buffer_set_flushing (sink->ringbuffer, FALSE);
sink->next_sample = -1;
gst_ring_buffer_set_flushing (sink->ringbuffer, FALSE);
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)
@ -746,6 +749,8 @@ gst_base_audio_sink_change_state (GstElement * element,
break;
}
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
/* ringbuffer cannot start anymore */
gst_ring_buffer_may_start (sink->ringbuffer, FALSE);
gst_ring_buffer_pause (sink->ringbuffer);
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:

28
gst-libs/gst/audio/gstringbuffer.c
View file

@ -1077,6 +1077,10 @@ wait_segment (GstRingBuffer * buf)
/* buffer must be started now or we deadlock since nobody is reading */
if (G_UNLIKELY (g_atomic_int_get (&buf->state) !=
GST_RING_BUFFER_STATE_STARTED)) {
/* see if we are allowed to start it */
if (G_UNLIKELY (g_atomic_int_get (&buf->abidata.ABI.may_start) == FALSE))
goto no_start;
GST_DEBUG_OBJECT (buf, "start!");
gst_ring_buffer_start (buf);
}
@ -1120,6 +1124,11 @@ flushing:
GST_OBJECT_UNLOCK (buf);
return FALSE;
}
no_start:
{
GST_DEBUG_OBJECT (buf, "not allowed to start");
return FALSE;
}
}
/**
@ -1455,3 +1464,22 @@ gst_ring_buffer_clear (GstRingBuffer * buf, gint segment)
memcpy (data, buf->empty_seg, buf->spec.segsize);
}
/**
* gst_ring_buffer_may_start:
* @buf: the #GstRingBuffer
* @allowed: the new value
*
* Tell the ringbuffer that it is allowed to start playback when
* the ringbuffer is filled with samples.
*
* Since: 0.10.6
*
* MT safe.
*/
void
gst_ring_buffer_may_start (GstRingBuffer * buf, gboolean allowed)
{
GST_LOG_OBJECT (buf, "may start: %d", allowed);
gst_atomic_int_set (&buf->abidata.ABI.may_start, allowed);
}

5
gst-libs/gst/audio/gstringbuffer.h
View file

@ -243,6 +243,8 @@ struct _GstRingBuffer {
union {
struct {
gboolean flushing;
/* ATOMIC */
gint may_start;
} ABI;
/* adding + 0 to mark ABI change to be undone later */
gpointer _gst_reserved[GST_PADDING + 0];
@ -335,6 +337,9 @@ gboolean gst_ring_buffer_prepare_read (GstRingBuffer *buf, gint *segme
void gst_ring_buffer_clear (GstRingBuffer *buf, gint segment);
void gst_ring_buffer_advance (GstRingBuffer *buf, guint advance);
void gst_ring_buffer_may_start (GstRingBuffer *buf, gboolean allowed);
G_END_DECLS
#endif /* __GST_RING_BUFFER_H__ */