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libs/gst/base/gstbasesink.c: Improve debugging.

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Original commit message from CVS:
* libs/gst/base/gstbasesink.c: (gst_base_sink_loop):
Improve debugging.
* libs/gst/base/gstbasesrc.c: (gst_base_src_query_latency),
(gst_base_src_default_query), (gst_base_src_wait),
(gst_base_src_do_sync), (gst_base_src_change_state):
Rearrange some code so that we can add support for measuring the
startup latency.
This commit is contained in:
Wim Taymans 2007-08-28 15:02:19 +00:00
parent bb82479c74
commit 2db0fa576b
3 changed files with 90 additions and 15 deletions
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11
ChangeLog
View file

@ -1,3 +1,14 @@
2007-08-28 Wim Taymans <wim.taymans@gmail.com>
* libs/gst/base/gstbasesink.c: (gst_base_sink_loop):
Improve debugging.
* libs/gst/base/gstbasesrc.c: (gst_base_src_query_latency),
(gst_base_src_default_query), (gst_base_src_wait),
(gst_base_src_do_sync), (gst_base_src_change_state):
Rearrange some code so that we can add support for measuring the
startup latency.
2007-08-27 Stefan Kost <ensonic@users.sf.net>
* docs/random/ensonic/dynlink.txt:

3
libs/gst/base/gstbasesink.c
View file

@ -2220,6 +2220,9 @@ gst_base_sink_loop (GstPad * pad)
g_assert (basesink->pad_mode == GST_ACTIVATE_PULL);
GST_DEBUG_OBJECT (basesink, "pulling %" G_GUINT64_FORMAT ", %u",
basesink->offset, (guint) DEFAULT_SIZE);
result = gst_pad_pull_range (pad, basesink->offset, DEFAULT_SIZE, &buf);
if (G_UNLIKELY (result != GST_FLOW_OK))
goto paused;

91
libs/gst/base/gstbasesrc.c
View file

@ -100,7 +100,7 @@
* distributed and running.
* </para>
* <para>
* Live sources that synchronize and block on the clock (and audio source, for
* Live sources that synchronize and block on the clock (an audio source, for
* example) can since 0.10.12 use gst_base_src_wait_playing() when the ::create
* function was interrupted by a state change to PAUSED.
* </para>
@ -112,6 +112,12 @@
* a live source.
* </para>
* <para>
* For live sources, the base class will by default measure the time it takes to
* create the first buffer in the PLAYING state and will report this value as
* the latency. Subclasses should override the query function when this
* behaviour is not acceptable.
* </para>
* <para>
* There is only support in #GstBaseSrc for exactly one source pad, which
* should be named "src". A source implementation (subclass of #GstBaseSrc)
* should install a pad template in its base_init function, like so:
@ -234,6 +240,11 @@ struct _GstBaseSrcPrivate
/* two segments to be sent in the streaming thread with STREAM_LOCK */
GstEvent *close_segment;
GstEvent *start_segment;
/* startup latency is the time it takes between going to PLAYING and producing
* the first BUFFER with running_time 0. This value is included in the latency
* reporting. */
GstClockTime startup_latency;
};
static GstElementClass *parent_class = NULL;
@ -554,8 +565,8 @@ gst_base_src_set_format (GstBaseSrc * src, GstFormat format)
* @max_latency: the max latency of the source
*
* Query the source for the latency parameters. @live will be TRUE when @src is
* configured as a live source. @min_latency will be set as the latency between
* calling the create function and the timestamp on the resulting buffer.
* configured as a live source. @min_latency will be set to the difference
* between the running time and the timestamp of the first buffer.
* @max_latency is always the undefined value of -1.
*
* This function is mostly used by subclasses.
@ -568,13 +579,28 @@ gboolean
gst_base_src_query_latency (GstBaseSrc * src, gboolean * live,
GstClockTime * min_latency, GstClockTime * max_latency)
{
GstClockTime min;
GST_LIVE_LOCK (src);
if (live)
*live = src->is_live;
/* if we have a startup latency, report this one, else report 0. Subclasses
* are supposed to override the query function if they want something
* else. */
if (src->priv->startup_latency != -1)
min = src->priv->startup_latency;
else
min = 0;
if (min_latency)
*min_latency = 0;
*min_latency = min;
if (max_latency)
*max_latency = -1;
GST_LOG_OBJECT (src, "latency: live %d, min %" GST_TIME_FORMAT
", max %" GST_TIME_FORMAT, src->is_live, GST_TIME_ARGS (min),
GST_TIME_ARGS (-1));
GST_LIVE_UNLOCK (src);
return TRUE;
@ -784,6 +810,10 @@ gst_base_src_default_query (GstBaseSrc * src, GstQuery * query)
/* Subclasses should override and implement something usefull */
res = gst_base_src_query_latency (src, &live, &min, &max);
GST_LOG_OBJECT (src, "report latency: live %d, min %" GST_TIME_FORMAT
", max %" GST_TIME_FORMAT, live, GST_TIME_ARGS (min),
GST_TIME_ARGS (max));
gst_query_set_latency (query, live, min, max);
break;
}
@ -1364,15 +1394,10 @@ gst_base_src_get_property (GObject * object, guint prop_id, GValue * value,
/* with STREAM_LOCK and LOCK */
static GstClockReturn
gst_base_src_wait (GstBaseSrc * basesrc, GstClockTime time)
gst_base_src_wait (GstBaseSrc * basesrc, GstClock * clock, GstClockTime time)
{
GstClockReturn ret;
GstClockID id;
GstClock *clock;
/* get clock, if no clock, we don't sync */
if ((clock = GST_ELEMENT_CLOCK (basesrc)) == NULL)
return GST_CLOCK_OK;
id = gst_clock_new_single_shot_id (clock, time);
@ -1399,6 +1424,7 @@ gst_base_src_do_sync (GstBaseSrc * basesrc, GstBuffer * buffer)
GstClockTime start, end;
GstBaseSrcClass *bclass;
GstClockTime base_time;
GstClock *clock;
bclass = GST_BASE_SRC_GET_CLASS (basesrc);
@ -1406,20 +1432,47 @@ gst_base_src_do_sync (GstBaseSrc * basesrc, GstBuffer * buffer)
if (bclass->get_times)
bclass->get_times (basesrc, buffer, &start, &end);
/* if we don't have a timestamp, we don't sync */
/* grab the lock to prepare for clocking and calculate the startup
* latency. */
GST_OBJECT_LOCK (basesrc);
base_time = GST_ELEMENT_CAST (basesrc)->base_time;
/* get clock, if no clock, we can't sync or get the latency */
if ((clock = GST_ELEMENT_CLOCK (basesrc)) == NULL)
goto no_clock;
if (basesrc->priv->startup_latency == -1) {
GstClockTime now = gst_clock_get_time (clock);
/* startup latency is the diff between when we went to PLAYING (base_time)
* and the current clock time */
if (now > base_time)
basesrc->priv->startup_latency = now - base_time;
else
basesrc->priv->startup_latency = 0;
GST_LOG_OBJECT (basesrc, "startup latency: %" GST_TIME_FORMAT,
GST_TIME_ARGS (basesrc->priv->startup_latency));
}
/* if we don't have a buffer timestamp, we don't sync */
if (!GST_CLOCK_TIME_IS_VALID (start))
goto invalid_start;
/* now do clocking */
GST_OBJECT_LOCK (basesrc);
base_time = GST_ELEMENT_CAST (basesrc)->base_time;
/* we have a timestamp, we can subtract it from the startup_latency when it is
* smaller. If the timestamp is bigger, there is no startup latency. */
if (start < basesrc->priv->startup_latency)
basesrc->priv->startup_latency -= start;
else
basesrc->priv->startup_latency = 0;
GST_LOG_OBJECT (basesrc,
"waiting for clock, base time %" GST_TIME_FORMAT
", stream_start %" GST_TIME_FORMAT,
GST_TIME_ARGS (base_time), GST_TIME_ARGS (start));
result = gst_base_src_wait (basesrc, start + base_time);
result = gst_base_src_wait (basesrc, clock, start + base_time);
GST_OBJECT_UNLOCK (basesrc);
GST_LOG_OBJECT (basesrc, "clock entry done: %d", result);
@ -1427,9 +1480,16 @@ gst_base_src_do_sync (GstBaseSrc * basesrc, GstBuffer * buffer)
return result;
/* special cases */
no_clock:
{
GST_DEBUG_OBJECT (basesrc, "we have no clock");
GST_OBJECT_UNLOCK (basesrc);
return GST_CLOCK_OK;
}
invalid_start:
{
GST_DEBUG_OBJECT (basesrc, "get_times returned invalid start");
GST_OBJECT_UNLOCK (basesrc);
return GST_CLOCK_OK;
}
}
@ -2260,6 +2320,7 @@ gst_base_src_change_state (GstElement * element, GstStateChange transition)
}
basesrc->priv->last_sent_eos = FALSE;
basesrc->priv->discont = TRUE;
basesrc->priv->startup_latency = -1;
GST_LIVE_UNLOCK (element);
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING: