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docs/libs/gstreamer-libs-sections.txt: Added basesink new methods.
Original commit message from CVS: * docs/libs/gstreamer-libs-sections.txt: Added basesink new methods. * gst/gstevent.c: * gst/gstevent.h: Docs updates. Flesh out the QoS docs. * libs/gst/base/gstadapter.c: Small doc clarification about ownership and flushing. * libs/gst/base/gstbasesink.c: (gst_base_sink_set_sync), (gst_base_sink_get_sync), (gst_base_sink_set_max_lateness), (gst_base_sink_get_max_lateness), (gst_base_sink_set_property), (gst_base_sink_get_property), (gst_base_sink_do_sync): * libs/gst/base/gstbasesink.h: Added new methods to allow subclass to control max-lateness and sync. Generate very basic QoS events based on last sync observation. Updated docs, fix typo, added some QoS blurb. * libs/gst/base/gstbasesrc.c: Remove obsolete _get_state() calls from docs.
This commit is contained in:
Wim Taymans
parent
c45bd3fb9f
commit
7a88e2a762
8 changed files with 182 additions and 32 deletions
25
ChangeLog
25
ChangeLog
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@ -1,3 +1,28 @@
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2006-03-07 Wim Taymans <wim@fluendo.com>
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* docs/libs/gstreamer-libs-sections.txt:
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Added basesink new methods.
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* gst/gstevent.c:
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* gst/gstevent.h:
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Docs updates. Flesh out the QoS docs.
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* libs/gst/base/gstadapter.c:
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Small doc clarification about ownership and flushing.
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* libs/gst/base/gstbasesink.c: (gst_base_sink_set_sync),
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(gst_base_sink_get_sync), (gst_base_sink_set_max_lateness),
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(gst_base_sink_get_max_lateness), (gst_base_sink_set_property),
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(gst_base_sink_get_property), (gst_base_sink_do_sync):
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* libs/gst/base/gstbasesink.h:
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Added new methods to allow subclass to control max-lateness
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and sync.
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Generate very basic QoS events based on last sync observation.
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Updated docs, fix typo, added some QoS blurb.
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* libs/gst/base/gstbasesrc.c:
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Remove obsolete _get_state() calls from docs.
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2006-03-07 Wim Taymans <wim@fluendo.com>
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* docs/libs/gstreamer-libs-sections.txt:
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@ -159,6 +159,11 @@ GstBaseSink
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GST_BASE_SINK_PAD
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gst_base_sink_set_sync
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gst_base_sink_get_sync
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gst_base_sink_set_max_lateness
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gst_base_sink_get_max_lateness
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<SUBSECTION Standard>
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GstBaseSinkClass
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GST_BASE_SINK
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@ -615,18 +615,29 @@ gst_event_parse_buffer_size (GstEvent * event, GstFormat * format,
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* Allocate a new qos event with the given values.
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* The QOS event is generated in an element that wants an upstream
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* element to either reduce or increase its rate because of
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* high/low CPU load.
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* high/low CPU load or other resource usage such as network performance.
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*
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* proportion is the requested adjustment in datarate, 1.0 is the normal
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* @proportion is the requested adjustment in datarate, 1.0 is the normal
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* datarate, 0.75 means increase datarate by 75%, 1.5 is 150%. Negative
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* values request a slow down, so -0.75 means a decrease by 75%.
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*
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* diff is the difference against the clock in stream time of the last
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* @diff is the difference against the clock in stream time of the last
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* buffer that caused the element to generate the QOS event.
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*
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* timestamp is the timestamp of the last buffer that cause the element
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* @timestamp is the timestamp of the last buffer that cause the element
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* to generate the QOS event.
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*
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* The upstream element can use the @diff and @timestamp values to decide
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* whether to process more buffers. All buffers with timestamp <=
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* @timestamp + @diff will certainly arrive late in the sink as well.
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*
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* The @proportion value is generally computed based on more long
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* term statistics about the stream quality and can be used in various ways
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* such as lowering or increasing processing quality.
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*
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* The application can use general event probes to intercept the QoS
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* event and implement custom application specific QoS handling.
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*
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* Returns: A new QOS event.
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*/
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GstEvent *
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@ -378,7 +378,6 @@ void gst_event_parse_buffer_size (GstEvent *event, GstFormat *format, gint64 *
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gint64 *maxsize, gboolean *async);
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/* QOS events */
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/* FIXME: QOS events need to be fully specified and implemented */
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GstEvent* gst_event_new_qos (gdouble proportion, GstClockTimeDiff diff,
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GstClockTime timestamp);
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void gst_event_parse_qos (GstEvent *event, gdouble *proportion, GstClockTimeDiff *diff,
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@ -67,6 +67,9 @@
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* gst_adapter_clear(). Data should also be cleared or processed on EOS and
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* when changing state from #GST_STATE_PAUSED to #GST_STATE_READY.
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*
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* Also check the GST_BUFFER_FLAG_DISCONT flag on the buffer. Some elements might
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* need to clear the adapter after a discontinuity.
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*
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* A last thing to note is that while GstAdapter is pretty optimized,
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* merging buffers still might be an operation that requires a memcpy()
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* operation, and this operation is not the fastest. Because of this, some
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@ -76,11 +79,14 @@
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* GstAdapter is not MT safe. All operations on an adapter must be serialized by
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* the caller. This is not normally a problem, however, as the normal use case
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* of GstAdapter is inside one pad's chain function, in which case access is
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* serialized via the pad's stream lock.
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* serialized via the pad's STREAM_LOCK.
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*
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* Note that gst_adapter_push() takes ownership of the buffer passed.
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* Note that gst_adapter_push() takes ownership of the buffer passed. Use
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* gst_buffer_ref() before pushing it into the adapter if you still want to
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* access the buffer later. The adapter will never modify the data in the
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* buffer pushed in it.
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*
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* Last reviewed on 2005-12-18 (0.10.0).
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* Last reviewed on 2006-03-07 (0.10.4).
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*/
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#include <string.h>
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@ -30,7 +30,7 @@
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* you, for example preroll, clock synchronization, state changes, activation in
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* push or pull mode, and queries. In most cases, when writing sink elements,
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* there is no need to implement class methods from #GstElement or to set
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* functions on pads, because the #GstBaseSink infrastructure is sufficient.
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* functions on pads, because the #GstBaseSink infrastructure should be sufficient.
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*
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* There is only support in GstBaseSink for one sink pad, which should be named
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* "sink". A sink implementation (subclass of GstBaseSink) should install a pad
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@ -53,8 +53,8 @@
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* #GstBaseSink will handle the prerolling correctly. This means that it will
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* return GST_STATE_CHANGE_ASYNC from a state change to PAUSED until the first buffer
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* arrives in this element. The base class will call the GstBaseSink::preroll
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* vmethod with this preroll buffer and will then commit the state change to
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* PAUSED.
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* vmethod with this preroll buffer and will then commit the state change to the
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* next asynchronously pending state.
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*
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* When the element is set to PLAYING, #GstBaseSink will synchronize on the clock
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* using the times returned from ::get_times. If this function returns
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@ -88,7 +88,7 @@
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* query will be forwarded upstream.
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*
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* The ::set_caps function will be called when the subclass should configure itself
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* to precess a specific media type.
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* to process a specific media type.
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*
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* The ::start and ::stop virtual methods will be called when resources should be
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* allocated. Any ::preroll, ::render and ::set_caps function will be called
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@ -106,7 +106,16 @@
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* operations they perform in the ::render method. This is mostly usefull when
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* the ::render method performs a blocking write on a file descripter.
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*
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* Last reviewed on 2006-01-30 (0.10.3)
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* The max-lateness property defines how the sink deals with buffers that arrive
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* too late in the sink. A buffer arrives too late in the sink when the presentation
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* time (as a combination of the last segment, buffer timestamp and element
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* base_time) is before the current time of the clock. If the frame is later than
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* max-lateness, the sink will drop the buffer and will generate a QoS event upstream.
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* This feature is disabled if sync is disabled, the ::get-times method does not
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* return a valid start time or max-lateness is set to -1 (the default).
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* Subclasses can use gst_base_sink_set_max_lateness() to configure the max-lateness.
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*
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* Last reviewed on 2006-03-07 (0.10.4)
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*/
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#ifdef HAVE_CONFIG_H
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@ -375,6 +384,95 @@ gst_base_sink_finalize (GObject * object)
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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/**
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* gst_base_sink_set_sync:
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* @sink: the sink
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* @sync: the new sync value.
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*
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* Configures @sink to synchronize on the clock or not. When
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* @sync is FALSE, incomming samples will be played as fast as
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* possible. If @sync is TRUE, the timestamps of the incomming
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* buffers will be used to schedule the exact render time of its
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* contents.
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*
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* Since: 0.10.4
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*/
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void
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gst_base_sink_set_sync (GstBaseSink * sink, gboolean sync)
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{
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GST_OBJECT_LOCK (sink);
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sink->sync = sync;
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GST_OBJECT_UNLOCK (sink);
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}
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/**
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* gst_base_sink_get_sync:
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* @sink: the sink
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*
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* Checks if @sink is currently configured to synchronize against the
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* clock.
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*
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* Returns: TRUE if the sink is configured to synchronize against the clock.
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*
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* Since: 0.10.4
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*/
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gboolean
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gst_base_sink_get_sync (GstBaseSink * sink)
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{
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gboolean res;
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GST_OBJECT_LOCK (sink);
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res = sink->sync;
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GST_OBJECT_UNLOCK (sink);
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return res;
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}
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/**
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* gst_base_sink_set_max_lateness:
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* @sink: the sink
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* @max_lateness: the new max lateness value.
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*
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* Sets the new max lateness value to @max_lateness. This value is
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* used to decide if a buffer should be dropped or not based on the
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* buffer timestamp and the current clock time. A value of -1 means
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* an unlimited time.
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*
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* Since: 0.10.4
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*/
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void
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gst_base_sink_set_max_lateness (GstBaseSink * sink, gint64 max_lateness)
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{
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GST_OBJECT_LOCK (sink);
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sink->abidata.ABI.max_lateness = max_lateness;
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GST_OBJECT_UNLOCK (sink);
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}
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/**
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* gst_base_sink_get_max_lateness:
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* @sink: the sink
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*
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* Gets the max lateness value. See gst_base_sink_set_max_lateness for
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* more details.
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*
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* Returns: The maximum time in nanoseconds that a buffer can be late
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* before it is dropped and not rendered. A value of -1 means an
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* unlimited time.
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*
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* Since: 0.10.4
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*/
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gint64
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gst_base_sink_get_max_lateness (GstBaseSink * sink)
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{
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gint64 res;
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GST_OBJECT_LOCK (sink);
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res = sink->abidata.ABI.max_lateness;
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GST_OBJECT_UNLOCK (sink);
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return res;
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}
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static void
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gst_base_sink_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec)
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GST_PAD_PREROLL_UNLOCK (sink->sinkpad);
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break;
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case PROP_SYNC:
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GST_OBJECT_LOCK (sink);
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sink->sync = g_value_get_boolean (value);
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GST_OBJECT_UNLOCK (sink);
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gst_base_sink_set_sync (sink, g_value_get_boolean (value));
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break;
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case PROP_MAX_LATENESS:
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GST_OBJECT_LOCK (sink);
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sink->abidata.ABI.max_lateness = g_value_get_int64 (value);
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GST_OBJECT_UNLOCK (sink);
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gst_base_sink_set_max_lateness (sink, g_value_get_int64 (value));
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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@ -417,14 +511,10 @@ gst_base_sink_get_property (GObject * object, guint prop_id, GValue * value,
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GST_PAD_PREROLL_UNLOCK (sink->sinkpad);
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break;
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case PROP_SYNC:
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GST_OBJECT_LOCK (sink);
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g_value_set_boolean (value, sink->sync);
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GST_OBJECT_UNLOCK (sink);
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g_value_set_boolean (value, gst_base_sink_get_sync (sink));
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break;
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case PROP_MAX_LATENESS:
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GST_OBJECT_LOCK (sink);
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g_value_set_int64 (value, sink->abidata.ABI.max_lateness);
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GST_OBJECT_UNLOCK (sink);
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g_value_set_int64 (value, gst_base_sink_get_max_lateness (sink));
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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}
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}
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static GstCaps *
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gst_base_sink_get_caps (GstBaseSink * sink)
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{
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@ -838,14 +929,23 @@ again:
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*late = FALSE;
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if (basesink->abidata.ABI.max_lateness != -1) {
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if (status == GST_CLOCK_EARLY
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&& jitter > basesink->abidata.ABI.max_lateness) {
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/* FIXME, update clock stats here and do some QoS */
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if (status == GST_CLOCK_EARLY) {
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if (basesink->abidata.ABI.max_lateness != -1
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&& jitter > basesink->abidata.ABI.max_lateness) {
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GstEvent *event;
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GST_DEBUG_OBJECT (basesink, "late: jitter!! %" G_GINT64_FORMAT "\n",
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jitter);
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*late = TRUE;
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/* generate QoS event, FIXME, calculate decent proportion. */
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event = gst_event_new_qos (-1.0, jitter, start);
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/* send upstream */
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gst_pad_push_event (basesink->sinkpad, event);
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}
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} else {
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}
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return GST_FLOW_OK;
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@ -136,6 +136,12 @@ struct _GstBaseSinkClass {
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GType gst_base_sink_get_type(void);
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void gst_base_sink_set_sync (GstBaseSink *sink, gboolean sync);
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gboolean gst_base_sink_get_sync (GstBaseSink *sink);
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void gst_base_sink_set_max_lateness (GstBaseSink *sink, gint64 max_lateness);
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gint64 gst_base_sink_get_max_lateness (GstBaseSink *sink);
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G_END_DECLS
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#endif /* __GST_BASE_SINK_H__ */
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@ -135,7 +135,6 @@
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* ...
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* // stop recording
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* gst_element_set_state (audio_source, GST_STATE_NULL);
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* gst_element_get_state (audio_source, NULL, NULL, -1);
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* gst_element_set_locked_state (audio_source, TRUE);
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* ...
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* </programlisting>
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@ -146,13 +145,12 @@
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* ...
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* // everything done - shut down pipeline
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* gst_element_set_state (pipeline, GST_STATE_NULL);
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* gst_element_get_state (pipeline, NULL, NULL, -1);
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* gst_element_set_locked_state (audio_source, FALSE);
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* ...
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* </programlisting>
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* </para>
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* <para>
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* Last reviewed on 2005-12-18 (0.10.0)
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* Last reviewed on 2006-03-07 (0.10.4)
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* </para>
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* </refsect2>
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*/
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