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audio: update for clock provider API change

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This commit is contained in:
Wim Taymans 2011-11-28 17:51:41 +01:00
parent b4cdf008dd
commit 468d1dde89
3 changed files with 22 additions and 12 deletions
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14
gst-libs/gst/audio/gstaudiobasesink.c
View file

@ -305,7 +305,6 @@ gst_audio_base_sink_init (GstAudioBaseSink * audiobasesink)
audiobasesink->buffer_time = DEFAULT_BUFFER_TIME; audiobasesink->buffer_time = DEFAULT_BUFFER_TIME;
audiobasesink->latency_time = DEFAULT_LATENCY_TIME; audiobasesink->latency_time = DEFAULT_LATENCY_TIME;
audiobasesink->provide_clock = DEFAULT_PROVIDE_CLOCK;
audiobasesink->priv->slave_method = DEFAULT_SLAVE_METHOD; audiobasesink->priv->slave_method = DEFAULT_SLAVE_METHOD;
audiobasesink->priv->drift_tolerance = DEFAULT_DRIFT_TOLERANCE; audiobasesink->priv->drift_tolerance = DEFAULT_DRIFT_TOLERANCE;
audiobasesink->priv->alignment_threshold = DEFAULT_ALIGNMENT_THRESHOLD; audiobasesink->priv->alignment_threshold = DEFAULT_ALIGNMENT_THRESHOLD;
@ -320,6 +319,10 @@ gst_audio_base_sink_init (GstAudioBaseSink * audiobasesink)
basesink->can_activate_pull = DEFAULT_CAN_ACTIVATE_PULL; basesink->can_activate_pull = DEFAULT_CAN_ACTIVATE_PULL;
gst_base_sink_set_last_buffer_enabled (basesink, FALSE); gst_base_sink_set_last_buffer_enabled (basesink, FALSE);
if (DEFAULT_PROVIDE_CLOCK)
GST_OBJECT_FLAG_SET (basesink, GST_ELEMENT_FLAG_PROVIDE_CLOCK);
else
GST_OBJECT_FLAG_UNSET (basesink, GST_ELEMENT_FLAG_PROVIDE_CLOCK);
} }
static void static void
@ -360,7 +363,7 @@ gst_audio_base_sink_provide_clock (GstElement * elem)
goto wrong_state; goto wrong_state;
GST_OBJECT_LOCK (sink); GST_OBJECT_LOCK (sink);
if (!sink->provide_clock) if (!GST_OBJECT_FLAG_IS_SET (sink, GST_ELEMENT_FLAG_PROVIDE_CLOCK))
goto clock_disabled; goto clock_disabled;
clock = GST_CLOCK_CAST (gst_object_ref (sink->provided_clock)); clock = GST_CLOCK_CAST (gst_object_ref (sink->provided_clock));
@ -563,7 +566,10 @@ gst_audio_base_sink_set_provide_clock (GstAudioBaseSink * sink,
g_return_if_fail (GST_IS_AUDIO_BASE_SINK (sink)); g_return_if_fail (GST_IS_AUDIO_BASE_SINK (sink));
GST_OBJECT_LOCK (sink); GST_OBJECT_LOCK (sink);
sink->provide_clock = provide; if (provide)
GST_OBJECT_FLAG_SET (sink, GST_ELEMENT_FLAG_PROVIDE_CLOCK);
else
GST_OBJECT_FLAG_UNSET (sink, GST_ELEMENT_FLAG_PROVIDE_CLOCK);
GST_OBJECT_UNLOCK (sink); GST_OBJECT_UNLOCK (sink);
} }
@ -586,7 +592,7 @@ gst_audio_base_sink_get_provide_clock (GstAudioBaseSink * sink)
g_return_val_if_fail (GST_IS_AUDIO_BASE_SINK (sink), FALSE); g_return_val_if_fail (GST_IS_AUDIO_BASE_SINK (sink), FALSE);
GST_OBJECT_LOCK (sink); GST_OBJECT_LOCK (sink);
result = sink->provide_clock; result = GST_OBJECT_FLAG_IS_SET (sink, GST_ELEMENT_FLAG_PROVIDE_CLOCK);
GST_OBJECT_UNLOCK (sink); GST_OBJECT_UNLOCK (sink);
return result; return result;

1
gst-libs/gst/audio/gstaudiobasesink.h
View file

@ -121,7 +121,6 @@ struct _GstAudioBaseSink {
guint64 next_sample; guint64 next_sample;
/* clock */ /* clock */
gboolean provide_clock;
GstClock *provided_clock; GstClock *provided_clock;
/* with g_atomic_; currently rendering eos */ /* with g_atomic_; currently rendering eos */

17
gst-libs/gst/audio/gstaudiobasesrc.c
View file

@ -73,8 +73,6 @@ gst_audio_base_src_slave_method_get_type (void)
struct _GstAudioBaseSrcPrivate struct _GstAudioBaseSrcPrivate
{ {
gboolean provide_clock;
/* the clock slaving algorithm in use */ /* the clock slaving algorithm in use */
GstAudioBaseSrcSlaveMethod slave_method; GstAudioBaseSrcSlaveMethod slave_method;
}; };
@ -240,7 +238,10 @@ gst_audio_base_src_init (GstAudioBaseSrc * audiobasesrc)
audiobasesrc->buffer_time = DEFAULT_BUFFER_TIME; audiobasesrc->buffer_time = DEFAULT_BUFFER_TIME;
audiobasesrc->latency_time = DEFAULT_LATENCY_TIME; audiobasesrc->latency_time = DEFAULT_LATENCY_TIME;
audiobasesrc->priv->provide_clock = DEFAULT_PROVIDE_CLOCK; if (DEFAULT_PROVIDE_CLOCK)
GST_OBJECT_FLAG_SET (audiobasesrc, GST_ELEMENT_FLAG_PROVIDE_CLOCK);
else
GST_OBJECT_FLAG_UNSET (audiobasesrc, GST_ELEMENT_FLAG_PROVIDE_CLOCK);
audiobasesrc->priv->slave_method = DEFAULT_SLAVE_METHOD; audiobasesrc->priv->slave_method = DEFAULT_SLAVE_METHOD;
/* reset blocksize we use latency time to calculate a more useful /* reset blocksize we use latency time to calculate a more useful
* value based on negotiated format. */ * value based on negotiated format. */
@ -250,6 +251,7 @@ gst_audio_base_src_init (GstAudioBaseSrc * audiobasesrc)
(GstAudioClockGetTimeFunc) gst_audio_base_src_get_time, audiobasesrc, (GstAudioClockGetTimeFunc) gst_audio_base_src_get_time, audiobasesrc,
NULL); NULL);
/* we are always a live source */ /* we are always a live source */
gst_base_src_set_live (GST_BASE_SRC (audiobasesrc), TRUE); gst_base_src_set_live (GST_BASE_SRC (audiobasesrc), TRUE);
/* we operate in time */ /* we operate in time */
@ -295,7 +297,7 @@ gst_audio_base_src_provide_clock (GstElement * elem)
goto wrong_state; goto wrong_state;
GST_OBJECT_LOCK (src); GST_OBJECT_LOCK (src);
if (!src->priv->provide_clock) if (!GST_OBJECT_FLAG_IS_SET (src, GST_ELEMENT_FLAG_PROVIDE_CLOCK))
goto clock_disabled; goto clock_disabled;
clock = GST_CLOCK_CAST (gst_object_ref (src->clock)); clock = GST_CLOCK_CAST (gst_object_ref (src->clock));
@ -364,7 +366,10 @@ gst_audio_base_src_set_provide_clock (GstAudioBaseSrc * src, gboolean provide)
g_return_if_fail (GST_IS_AUDIO_BASE_SRC (src)); g_return_if_fail (GST_IS_AUDIO_BASE_SRC (src));
GST_OBJECT_LOCK (src); GST_OBJECT_LOCK (src);
src->priv->provide_clock = provide; if (provide)
GST_OBJECT_FLAG_SET (src, GST_ELEMENT_FLAG_PROVIDE_CLOCK);
else
GST_OBJECT_FLAG_UNSET (src, GST_ELEMENT_FLAG_PROVIDE_CLOCK);
GST_OBJECT_UNLOCK (src); GST_OBJECT_UNLOCK (src);
} }
@ -387,7 +392,7 @@ gst_audio_base_src_get_provide_clock (GstAudioBaseSrc * src)
g_return_val_if_fail (GST_IS_AUDIO_BASE_SRC (src), FALSE); g_return_val_if_fail (GST_IS_AUDIO_BASE_SRC (src), FALSE);
GST_OBJECT_LOCK (src); GST_OBJECT_LOCK (src);
result = src->priv->provide_clock; result = GST_OBJECT_FLAG_IS_SET (src, GST_ELEMENT_FLAG_PROVIDE_CLOCK);
GST_OBJECT_UNLOCK (src); GST_OBJECT_UNLOCK (src);
return result; return result;