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docs/random/moving-plugins: Make it clear that the "compiled-in descriptions" really mean the element details.

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Original commit message from CVS:
* docs/random/moving-plugins:
Make it clear that the "compiled-in descriptions" really mean
the element details.
* libs/gst/base/gstbasesink.c: (gst_base_sink_commit_state),
(gst_base_sink_wait_preroll):
Update docs.
* docs/libs/gstreamer-libs-sections.txt:
* libs/gst/base/gstbasesrc.c: (gst_base_src_wait_playing),
(gst_base_src_get_range), (gst_base_src_activate_push):
* libs/gst/base/gstbasesrc.h:
Added function to block while waiting for PLAYING, this function
is used by live sources that block on the clock.
API: gst_base_src_wait_playing()
This commit is contained in:
Wim Taymans 2006-09-27 13:19:55 +00:00
parent 05b05477eb
commit a0e5a4a964
6 changed files with 112 additions and 49 deletions
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18
ChangeLog
View file

@ -1,3 +1,21 @@
2006-09-27 Wim Taymans <wim@fluendo.com>
* docs/random/moving-plugins:
Make it clear that the "compiled-in descriptions" really mean
the element details.
* libs/gst/base/gstbasesink.c: (gst_base_sink_commit_state),
(gst_base_sink_wait_preroll):
Update docs.
* docs/libs/gstreamer-libs-sections.txt:
* libs/gst/base/gstbasesrc.c: (gst_base_src_wait_playing),
(gst_base_src_get_range), (gst_base_src_activate_push):
* libs/gst/base/gstbasesrc.h:
Added function to block while waiting for PLAYING, this function
is used by live sources that block on the clock.
API: gst_base_src_wait_playing()
2006-09-27 Tim-Philipp Müller <tim at centricular dot net>
Patch by: Peter Kjellerstedt <pkj at axis com>

1
docs/libs/gstreamer-libs-sections.txt
View file

@ -149,6 +149,7 @@ GstBaseSrc
GstBaseSrcClass
GstBaseSrcFlags
gst_base_src_wait_playing
gst_base_src_is_live
gst_base_src_set_live
gst_base_src_set_format

2
docs/random/moving-plugins
View file

@ -77,7 +77,7 @@ CHECKLIST
ext/: plug-ins with external dependencies
- The plug-in is documented:
- the compiled-in descriptions should be correct
- the compiled-in descriptions (element details) should be correct
- every element in the plug-in should have gtk-doc documentation:
- longer description of element
- why you would use this element

17
libs/gst/base/gstbasesink.c
View file

@ -35,7 +35,7 @@
* #GstBaseSink infrastructure should be sufficient.
*
* #GstBaseSink provides support for exactly one sink pad, which should be
* named "sink". A sink implementation (subclass of GstBaseSink) should
* named "sink". A sink implementation (subclass of #GstBaseSink) should
* install a pad template in its base_init function, like so:
* <programlisting>
* static void
@ -53,16 +53,16 @@
* </programlisting>
*
* #GstBaseSink will handle the prerolling correctly. This means that it will
* return GST_STATE_CHANGE_ASYNC from a state change to PAUSED until the first
* return #GST_STATE_CHANGE_ASYNC from a state change to PAUSED until the first
* buffer arrives in this element. The base class will call the
* GstBaseSink::preroll vmethod with this preroll buffer and will then commit
* #GstBaseSink::preroll vmethod with this preroll buffer and will then commit
* the state change to the next asynchronously pending state.
*
* When the element is set to PLAYING, #GstBaseSink will synchronise on the
* clock using the times returned from ::get_times. If this function returns
* #GST_CLOCK_TIME_NONE for the start time, no synchronisation will be done.
* Synchronisation can be disabled entirely by setting the object "sync"
* property to FALSE.
* property to %FALSE.
*
* After synchronisation the virtual method #GstBaseSink::render will be called.
* Subclasses should minimally implement this method.
@ -70,7 +70,8 @@
* Since 0.10.3 subclasses that synchronise on the clock in the ::render method
* are supported as well. These classes typically receive a buffer in the render
* method and can then potentially block on the clock while rendering. A typical
* example is an audiosink.
* example is an audiosink. Since 0.10.11 these subclasses can use
* gst_base_sink_wait_preroll() to perform the blocking wait.
*
* Upon receiving the EOS event in the PLAYING state, #GstBaseSink will wait
* for the clock to reach the time indicated by the stop time of the last
@ -78,14 +79,14 @@
* EOS in PAUSED, preroll completes, the event is queued and an EOS message is
* posted when going to PLAYING.
*
* #GstBaseSink will internally use the GST_EVENT_NEW_SEGMENT events to schedule
* #GstBaseSink will internally use the #GST_EVENT_NEWSEGMENT events to schedule
* synchronisation and clipping of buffers. Buffers that fall completely outside
* of the current segment are dropped. Buffers that fall partially in the
* segment are rendered (and prerolled). Subclasses should do any subbuffer
* clipping themselves when needed.
*
* #GstBaseSink will by default report the current playback position in
* GST_FORMAT_TIME based on the current clock time and segment information.
* #GST_FORMAT_TIME based on the current clock time and segment information.
* If no clock has been set on the element, the query will be forwarded
* upstream.
*
@ -127,7 +128,7 @@
* synchronisation. For each dropped buffer it will also send a QoS message
* upstream.
*
* Last reviewed on 2006-03-31 (0.10.5)
* Last reviewed on 2006-09-27 (0.10.11)
*/
#ifdef HAVE_CONFIG_H

115
libs/gst/base/gstbasesrc.c
View file

@ -37,38 +37,38 @@
* <para>
* The source can be configured to operate in any #GstFormat with the
* gst_base_src_set_format() method. The currently set format determines
* the format of the internal #GstSegment and any #GST_EVENT_NEW_SEGMENT
* the format of the internal #GstSegment and any #GST_EVENT_NEWSEGMENT
* events. The default format for #GstBaseSrc is #GST_FORMAT_BYTES.
* </para>
* <para>
* #GstBaseSrc always supports push mode scheduling. If the following
* conditions are met, it also supports pull mode scheduling:
* <itemizedlist>
* <listitem><para>The format is set to GST_FORMAT_BYTES (default).</para>
* <listitem><para>The format is set to #GST_FORMAT_BYTES (default).</para>
* </listitem>
* <listitem><para>#GstBaseSrc::is_seekable returns TRUE.</para>
* <listitem><para>#GstBaseSrc::is_seekable returns %TRUE.</para>
* </listitem>
* </itemizedlist>
* </para>
* <para>
* Since 0.10.9, any #GstBaseSrc can enable pull based scheduling at any
* time by overriding #GstBaseSrc::check_get_range so that it returns TRUE.
* time by overriding #GstBaseSrc::check_get_range so that it returns %TRUE.
* </para>
* <para>
* If all the conditions are met for operating in pull mode, #GstBaseSrc is
* automatically seekable in push mode as well. The following conditions must
* be met to make the element seekable in push mode when the format is not
* GST_FORMAT_BYTES:
* #GST_FORMAT_BYTES:
* <itemizedlist>
* <listitem><para>
* #GstBaseSrc::is_seekable returns TRUE.
* #GstBaseSrc::is_seekable returns %TRUE.
* </para></listitem>
* <listitem><para>
* #GstBaseSrc::query can convert all supported seek formats to the
* internal format as set with gst_base_src_set_format().
* </para></listitem>
* <listitem><para>
* #GstBaseSrc::do_seek is implemented, performs the seek and returns TRUE.
* #GstBaseSrc::do_seek is implemented, performs the seek and returns %TRUE.
* </para></listitem>
* </itemizedlist>
* </para>
@ -91,7 +91,7 @@
* A live source does not produce data in the PAUSED state. This means that the
* #GstBaseSrc::create method will not be called in PAUSED but only in PLAYING.
* To signal the pipeline that the element will not produce data, the return
* value from the READY to PAUSED state will be GST_STATE_CHANGE_NO_PREROLL.
* value from the READY to PAUSED state will be #GST_STATE_CHANGE_NO_PREROLL.
* </para>
* <para>
* A typical live source will timestamp the buffers it creates with the
@ -100,6 +100,11 @@
* distributed and running.
* </para>
* <para>
* Live sources that synchronize and block on the clock (and 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>
* <para>
* The #GstBaseSrc::get_times method can be used to implement pseudo-live
* sources. The base source will wait for the specified stream time returned in
* #GstBaseSrc::get_times before pushing out the buffer.
@ -107,8 +112,8 @@
* a live source.
* </para>
* <para>
* There is only support in GstBaseSrc for exactly one source pad, which
* should be named "src". A source implementation (subclass of GstBaseSrc)
* 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:
* </para>
* <para>
@ -146,8 +151,8 @@
* <programlisting>
* ...
* // stop recording
* gst_element_set_state (audio_source, GST_STATE_NULL);
* gst_element_set_locked_state (audio_source, TRUE);
* gst_element_set_state (audio_source, #GST_STATE_NULL);
* gst_element_set_locked_state (audio_source, %TRUE);
* ...
* </programlisting>
* Now the application should wait for an EOS message
@ -156,8 +161,8 @@
* <programlisting>
* ...
* // everything done - shut down pipeline
* gst_element_set_state (pipeline, GST_STATE_NULL);
* gst_element_set_locked_state (audio_source, FALSE);
* gst_element_set_state (pipeline, #GST_STATE_NULL);
* gst_element_set_locked_state (audio_source, %FALSE);
* ...
* </programlisting>
* </para>
@ -167,7 +172,7 @@
* thread might be blocked in PREROLL.
* </para>
* <para>
* Last reviewed on 2006-07-06 (0.10.9)
* Last reviewed on 2006-09-27 (0.10.11)
* </para>
* </refsect2>
*/
@ -420,6 +425,58 @@ gst_base_src_finalize (GObject * object)
G_OBJECT_CLASS (parent_class)->finalize (object);
}
/**
* gst_base_src_wait_playing:
* @src: the src
*
* If the #GstBaseSrcClass::create method performs its own synchronisation against
* the clock it must unblock when going from PLAYING to the PAUSED state and call
* this method before continuing to produce the remaining data.
*
* This function will block until a state change to PLAYING happens (in which
* case this function returns #GST_FLOW_OK) or the processing must be stopped due
* to a state change to READY or a FLUSH event (in which case this function
* returns #GST_FLOW_WRONG_STATE).
*
* Since: 0.10.12
*
* Returns: #GST_FLOW_OK if @src is PLAYING and processing can
* continue. Any other return value should be returned from the create vmethod.
*/
GstFlowReturn
gst_base_src_wait_playing (GstBaseSrc * src)
{
/* block until the state changes, or we get a flush, or something */
GST_LIVE_LOCK (src);
if (src->is_live) {
while (G_UNLIKELY (!src->live_running)) {
GST_DEBUG ("live source signal waiting");
GST_LIVE_SIGNAL (src);
GST_DEBUG ("live source waiting for running state");
GST_LIVE_WAIT (src);
GST_DEBUG ("live source unlocked");
}
/* FIXME, use another variable to signal stopping so that we don't
* have to grab another lock. */
GST_OBJECT_LOCK (src->srcpad);
if (G_UNLIKELY (GST_PAD_IS_FLUSHING (src->srcpad)))
goto flushing;
GST_OBJECT_UNLOCK (src->srcpad);
}
GST_LIVE_UNLOCK (src);
return GST_FLOW_OK;
/* ERRORS */
flushing:
{
GST_DEBUG_OBJECT (src, "pad is flushing");
GST_OBJECT_UNLOCK (src->srcpad);
GST_LIVE_UNLOCK (src);
return GST_FLOW_WRONG_STATE;
}
}
/**
* gst_base_src_set_live:
* @src: base source instance
@ -1266,23 +1323,9 @@ gst_base_src_get_range (GstBaseSrc * src, guint64 offset, guint length,
bclass = GST_BASE_SRC_GET_CLASS (src);
GST_LIVE_LOCK (src);
if (src->is_live) {
while (G_UNLIKELY (!src->live_running)) {
GST_DEBUG ("live source signal waiting");
GST_LIVE_SIGNAL (src);
GST_DEBUG ("live source waiting for running state");
GST_LIVE_WAIT (src);
GST_DEBUG ("live source unlocked");
}
/* FIXME, use another variable to signal stopping so that we don't
* have to grab another lock. */
GST_OBJECT_LOCK (src->srcpad);
if (G_UNLIKELY (GST_PAD_IS_FLUSHING (src->srcpad)))
goto flushing;
GST_OBJECT_UNLOCK (src->srcpad);
}
GST_LIVE_UNLOCK (src);
ret = gst_base_src_wait_playing (src);
if (ret != GST_FLOW_OK)
goto stopped;
if (G_UNLIKELY (!GST_OBJECT_FLAG_IS_SET (src, GST_BASE_SRC_STARTED)))
goto not_started;
@ -1338,12 +1381,10 @@ done:
return ret;
/* ERROR */
flushing:
stopped:
{
GST_DEBUG_OBJECT (src, "pad is flushing");
GST_OBJECT_UNLOCK (src->srcpad);
GST_LIVE_UNLOCK (src);
return GST_FLOW_WRONG_STATE;
GST_DEBUG_OBJECT (src, "wait_playing returned %d", ret);
return ret;
}
not_started:
{

8
libs/gst/base/gstbasesrc.h
View file

@ -205,10 +205,12 @@ struct _GstBaseSrcClass {
GType gst_base_src_get_type (void);
void gst_base_src_set_live (GstBaseSrc *src, gboolean live);
gboolean gst_base_src_is_live (GstBaseSrc *src);
GstFlowReturn gst_base_src_wait_playing (GstBaseSrc *src);
void gst_base_src_set_format (GstBaseSrc *src, GstFormat format);
void gst_base_src_set_live (GstBaseSrc *src, gboolean live);
gboolean gst_base_src_is_live (GstBaseSrc *src);
void gst_base_src_set_format (GstBaseSrc *src, GstFormat format);
G_END_DECLS