mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-07-05 06:05:53 +00:00
Code Issues 1 Projects Releases Wiki Activity

gst-libs/gst/audio/gstbaseaudiosrc.c: Provide readable actual-buffer-time and actual-latency-time properties that ref...

Browse source 
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_class_init), (gst_base_audio_src_dispose),
(gst_base_audio_src_get_property), (gst_base_audio_src_setcaps),
(gst_base_audio_src_change_state):
Provide readable actual-buffer-time and actual-latency-time properties
that reflect the configured ringbuffer values. Fixes #524724.
This commit is contained in:
Mark Nauwelaerts 2008-05-31 18:10:47 +00:00
parent 11309247f3
commit c660bbd6dd
2 changed files with 51 additions and 0 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

10
ChangeLog
View file

@ -1,3 +1,13 @@
2008-05-31 Mark Nauwelaerts <mnauw@users.sf.net>
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_class_init), (gst_base_audio_src_dispose),
(gst_base_audio_src_get_property), (gst_base_audio_src_setcaps),
(gst_base_audio_src_change_state):
Provide readable actual-buffer-time and actual-latency-time properties
that reflect the configured ringbuffer values. Fixes #524724.
2008-05-30 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push),

41
gst-libs/gst/audio/gstbaseaudiosrc.c
View file

@ -86,6 +86,8 @@ enum
#define DEFAULT_BUFFER_TIME ((200 * GST_MSECOND) / GST_USECOND)
#define DEFAULT_LATENCY_TIME ((10 * GST_MSECOND) / GST_USECOND)
#define DEFAULT_ACTUAL_BUFFER_TIME -1
#define DEFAULT_ACTUAL_LATENCY_TIME -1
#define DEFAULT_PROVIDE_CLOCK TRUE
#define DEFAULT_SLAVE_METHOD GST_BASE_AUDIO_SRC_SLAVE_RETIMESTAMP
@ -94,6 +96,8 @@ enum
PROP_0,
PROP_BUFFER_TIME,
PROP_LATENCY_TIME,
PROP_ACTUAL_BUFFER_TIME,
PROP_ACTUAL_LATENCY_TIME,
PROP_PROVIDE_CLOCK,
PROP_SLAVE_METHOD,
PROP_LAST
@ -179,6 +183,18 @@ gst_base_audio_src_class_init (GstBaseAudioSrcClass * klass)
G_MAXINT64, DEFAULT_LATENCY_TIME,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_ACTUAL_BUFFER_TIME,
g_param_spec_int64 ("actual-buffer-time", "Actual Buffer Time",
"Actual configured size of audio buffer in microseconds",
DEFAULT_ACTUAL_BUFFER_TIME, G_MAXINT64, DEFAULT_ACTUAL_BUFFER_TIME,
G_PARAM_READABLE));
g_object_class_install_property (gobject_class, PROP_ACTUAL_LATENCY_TIME,
g_param_spec_int64 ("actual-latency-time", "Actual Latency Time",
"Actual configured audio latency in microseconds",
DEFAULT_ACTUAL_LATENCY_TIME, G_MAXINT64, DEFAULT_ACTUAL_LATENCY_TIME,
G_PARAM_READABLE));
g_object_class_install_property (gobject_class, PROP_PROVIDE_CLOCK,
g_param_spec_boolean ("provide-clock", "Provide Clock",
"Provide a clock to be used as the global pipeline clock",
@ -241,6 +257,7 @@ gst_base_audio_src_dispose (GObject * object)
src = GST_BASE_AUDIO_SRC (object);
GST_OBJECT_LOCK (src);
if (src->clock)
gst_object_unref (src->clock);
src->clock = NULL;
@ -249,6 +266,7 @@ gst_base_audio_src_dispose (GObject * object)
gst_object_unparent (GST_OBJECT_CAST (src->ringbuffer));
src->ringbuffer = NULL;
}
GST_OBJECT_UNLOCK (src);
G_OBJECT_CLASS (parent_class)->dispose (object);
}
@ -461,6 +479,22 @@ gst_base_audio_src_get_property (GObject * object, guint prop_id,
case PROP_LATENCY_TIME:
g_value_set_int64 (value, src->latency_time);
break;
case PROP_ACTUAL_BUFFER_TIME:
GST_OBJECT_LOCK (src);
if (src->ringbuffer && src->ringbuffer->acquired)
g_value_set_int64 (value, src->ringbuffer->spec.buffer_time);
else
g_value_set_int64 (value, DEFAULT_ACTUAL_BUFFER_TIME);
GST_OBJECT_UNLOCK (src);
break;
case PROP_ACTUAL_LATENCY_TIME:
GST_OBJECT_LOCK (src);
if (src->ringbuffer && src->ringbuffer->acquired)
g_value_set_int64 (value, src->ringbuffer->spec.latency_time);
else
g_value_set_int64 (value, DEFAULT_ACTUAL_LATENCY_TIME);
GST_OBJECT_UNLOCK (src);
break;
case PROP_PROVIDE_CLOCK:
g_value_set_boolean (value, gst_base_audio_src_get_provide_clock (src));
break;
@ -538,6 +572,9 @@ gst_base_audio_src_setcaps (GstBaseSrc * bsrc, GstCaps * caps)
gst_ring_buffer_debug_spec_buff (spec);
g_object_notify (G_OBJECT (src), "actual-buffer-time");
g_object_notify (G_OBJECT (src), "actual-latency-time");
return TRUE;
/* ERRORS */
@ -869,10 +906,12 @@ gst_base_audio_src_change_state (GstElement * element,
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
GST_DEBUG_OBJECT (src, "NULL->READY");
GST_OBJECT_LOCK (src);
if (src->ringbuffer == NULL) {
gst_audio_clock_reset (GST_AUDIO_CLOCK (src->clock), 0);
src->ringbuffer = gst_base_audio_src_create_ringbuffer (src);
}
GST_OBJECT_UNLOCK (src);
if (!gst_ring_buffer_open_device (src->ringbuffer))
goto open_failed;
break;
@ -909,8 +948,10 @@ gst_base_audio_src_change_state (GstElement * element,
case GST_STATE_CHANGE_READY_TO_NULL:
GST_DEBUG_OBJECT (src, "READY->NULL");
gst_ring_buffer_close_device (src->ringbuffer);
GST_OBJECT_LOCK (src);
gst_object_unparent (GST_OBJECT_CAST (src->ringbuffer));
src->ringbuffer = NULL;
GST_OBJECT_UNLOCK (src);
break;
default:
break;