Commit 35e4b75b authored by Wim Taymans's avatar Wim Taymans
Browse files

gst-libs/gst/audio/gstaudioclock.*: Add method to inform the clock that the...

gst-libs/gst/audio/gstaudioclock.*: Add method to inform the clock that the time starts from 0 again. We use this inf...

Original commit message from CVS:
* gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_init),
(gst_audio_clock_reset), (gst_audio_clock_get_internal_time):
* gst-libs/gst/audio/gstaudioclock.h:
Add method to inform the clock that the time starts from 0 again. We use
this info to calculate a clock offset so that the time we report in
internal_time is monotonically increasing, as required by the clock base
class. Fixes #521761.
API: GstAudioClock::gst_audio_clock_reset()
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_skew_slaving),
(gst_base_audio_sink_change_state):
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_create), (gst_base_audio_src_change_state):
Reset reported time when we (re)create the ringbuffer.
parent dc9eb0d6
2008-05-27 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_init),
(gst_audio_clock_reset), (gst_audio_clock_get_internal_time):
* gst-libs/gst/audio/gstaudioclock.h:
Add method to inform the clock that the time starts from 0 again. We use
this info to calculate a clock offset so that the time we report in
internal_time is monotonically increasing, as required by the clock base
class. Fixes #521761.
API: GstAudioClock::gst_audio_clock_reset()
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_skew_slaving),
(gst_base_audio_sink_change_state):
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_create), (gst_base_audio_src_change_state):
Reset reported time when we (re)create the ringbuffer.
2008-05-27 Tim-Philipp Müller <tim.muller at collabora co uk>
 
* ext/alsa/gstalsamixertrack.c:
......@@ -94,6 +94,7 @@ static void
gst_audio_clock_init (GstAudioClock * clock)
{
clock->last_time = 0;
clock->abidata.ABI.time_offset = 0;
GST_OBJECT_FLAG_SET (clock, GST_CLOCK_FLAG_CAN_SET_MASTER);
}
......@@ -122,17 +123,39 @@ gst_audio_clock_new (gchar * name, GstAudioClockGetTimeFunc func,
return (GstClock *) aclock;
}
/**
* gst_audio_clock_reset:
* @clock: a #GstAudioClock
* @time: a #GstClockTime
*
* Inform @clock that future calls to #GstAudioClockGetTimeFunc will return values
* starting from @time. The clock will update an internal offset to make sure that
* future calls to internal_time will return an increasing result as required by
* the #GstClock object.
*/
void
gst_audio_clock_reset (GstAudioClock * clock, GstClockTime time)
{
if (clock->last_time >= time)
clock->abidata.ABI.time_offset = clock->last_time - time;
else
clock->abidata.ABI.time_offset = -(time - clock->last_time);
}
static GstClockTime
gst_audio_clock_get_internal_time (GstClock * clock)
{
GstAudioClock *aclock = GST_AUDIO_CLOCK (clock);
GstAudioClock *aclock;
GstClockTime result;
aclock = GST_AUDIO_CLOCK (clock);
result = aclock->func (clock, aclock->user_data);
if (result == GST_CLOCK_TIME_NONE)
result = aclock->last_time;
else
else {
result += aclock->abidata.ABI.time_offset;
aclock->last_time = result;
}
return result;
}
......@@ -71,7 +71,13 @@ struct _GstAudioClock {
GstClockTime last_time;
/*< private >*/
gpointer _gst_reserved[GST_PADDING];
union {
struct {
GstClockTimeDiff time_offset;
} ABI;
/* adding + 0 to mark ABI change to be undone later */
gpointer _gst_reserved[GST_PADDING + 0];
} abidata;
};
struct _GstAudioClockClass {
......@@ -84,6 +90,7 @@ struct _GstAudioClockClass {
GType gst_audio_clock_get_type (void);
GstClock* gst_audio_clock_new (gchar *name, GstAudioClockGetTimeFunc func,
gpointer user_data);
void gst_audio_clock_reset (GstAudioClock *clock, GstClockTime time);
G_END_DECLS
......
......@@ -1604,6 +1604,7 @@ gst_base_audio_sink_change_state (GstElement * element,
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
if (sink->ringbuffer == NULL) {
gst_audio_clock_reset (GST_AUDIO_CLOCK (sink->provided_clock), 0);
sink->ringbuffer = gst_base_audio_sink_create_ringbuffer (sink);
}
if (!gst_ring_buffer_open_device (sink->ringbuffer))
......
......@@ -870,6 +870,7 @@ gst_base_audio_src_change_state (GstElement * element,
case GST_STATE_CHANGE_NULL_TO_READY:
GST_DEBUG_OBJECT (src, "NULL->READY");
if (src->ringbuffer == NULL) {
gst_audio_clock_reset (GST_AUDIO_CLOCK (src->clock), 0);
src->ringbuffer = gst_base_audio_src_create_ringbuffer (src);
}
if (!gst_ring_buffer_open_device (src->ringbuffer))
......
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