gstreamer/docs/design/part-clocks.txt
Wim Taymans b69033434f docs/design/: Documentation updates.
Original commit message from CVS:
* docs/design/draft-klass.txt:
* docs/design/part-clocks.txt:
* docs/design/part-events.txt:
* docs/design/part-gstbin.txt:
* docs/design/part-gstpipeline.txt:
* docs/design/part-messages.txt:
* docs/design/part-negotiation.txt:
* docs/design/part-overview.txt:
* docs/design/part-preroll.txt:
* docs/design/part-seeking.txt:
* docs/design/part-states.txt:
* docs/design/part-streams.txt:
Documentation updates.
2006-03-13 10:32:26 +00:00

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Text

Clocks
------
To synchronize the different elements, the GstPipeline is responsible for
selecting and distributing a global GstClock for all the elements in it.
This selection happens whenever an element is added or removed from the
pipeline. Whever the clock changes in a pipeline, a NEW_CLOCK message is
posted on the bus signaling the new clock to the application.
The GstClock returns a monotonically increasing time with the method
_get_time(). Its accuracy and base time depends on the specific clock
implementation but time is always expessed in nanoseconds. Since the
baseline of the clock is undefined, the clock time returned is not
meaningfull in itself, what matters are the deltas between two clock
times.
The time reported by the clock is called the absolute time.
Time in GStreamer
-----------------
The absolute time is used to calculate the stream time. The stream time
is defined as follows:
- If the pipeline is NULL/READY, the stream time is undefined.
- In PAUSED, the stream time remains at the time when it was last
PAUSED. When the stream is PAUSED for the first time, the stream time
is 0.
- In PLAYING, the stream time is the delta between the absolute time
and the base time. The base time is defined as the absolute time minus
the stream time at the time when the pipeline is set to PLAYING.
- after a seek, the stream time is set to 0 (see part-seeking.txt)
The stream time is completely managed by the GstPipeline object using the
GstClock absolute time.
Timestamps
----------
The combination of the last NEWSEGMENT event and the buffer timestamps
express the presentation stream time of the buffer. The stream time
of a buffer is calculated as follows:
ST = TS - DT where: TS = buffer timestamp
DT = NEWSEGMENT timestamp
ST = buffer stream time
The reason for not making the buffer times express the stream time directly
is for the following reasons:
- demuxers are easier if they can just copy the timestamps as encoded in
the file. The initial NEWSEGMENT event would contain the lowest timestamp in
the stream which makes the buffer stream time start from 0.
- pipelines requiring retimestamping of buffers can efficiently adjust
the timestamp in the NEWSEGMENT events and have all buffers retimestamped
automatically.
- resync after different kinds of seeks is easier.
If an element wants to synchronize a buffer to the clock it needs to first
calculate the buffer stream time and then bring the stream time to the
absolute clock time.
Converting a timestamp (in stream time) to absolute time is performed using
the following formula:
AT = BT + ST where: AT = absolute time
BT = base time
ST = stream time
The pipeline base time is propagated to all the element during the PAUSED
to PLAYING state change. All elements are therefore able to convert the
stream time to the absolute time. It is possible to specify an aditional
delay to the base time to compensate for the delay it takes to perform
the state change using the GstPipeline "delay" property.
Clock features
--------------
The clock supports periodic and single shot clock notifications both
synchronous and asynchronous.
One first needs to create a GstClockID for the periodic or single shot
notification using _clock_new_single_shot_id() or _clock_new_periodic_id().
To perform a blocking wait for the specific time of the GstClockID use the
gst_clock_id_wait(). To receive a callback when the specific time is reached
in the clock use gst_clock_id_wait_async(). Both these calls can be interrupted
with the gst_clock_id_unschedule() call. If the blocking wait is unscheduled
a return value of GST_CLOCK_UNSCHEDULED is returned.
The async callbacks can happen from any thread, either provided by the
core or from a streaming thread. The application should be prepared for this.
A GstClockID that has been unscheduled cannot be used again for any wait
operation.
It is possible to perform a blocking wait on the same ID from multiple
threads. However, registering the same ID for multiple async notifications is
not possible, the callback will only be called once.
None of the wait operations unref the GstClockID, the owner is
responsible for unreffing the ids itself. This holds for both periodic and
single shot notifications. The reason being that the owner of the ClockID
has to keep a handle to the ID to unblock the wait on FLUSHING events
or state changes and if we unref it automatically, the handle might be
invalid.
These clock operations do not operate on the stream time, so the callbacks
will also occur when not in PLAYING state as if the clock just keeps on
running. Some clocks however do not progress when the element that provided
the clock is not PLAYING.
Clock implementations
---------------------
The GStreamer core provides a GstSystemClock based on the system time.
Asynchronous callbacks are scheduled from an internal thread.
Clock implementors are encouraged to subclass this systemclock as it
implements the async notification.
Subclasses can however override all of the important methods for sync and
async notifications to implement their own callback methods or blocking
wait operations.