Add a bunch of vararg getter convenience functions to complement
the vararg setter functions, and a basic unit test. Fixes#534208.
API: gst_structure_get()
API: gst_structure_id_get()
API: gst_structure_get_valist()
API: gst_structure_id_get_valist()
Note in the docs that a flushing step in PLAYING brings the pipeline to the lost
state and skips the data before prerolling again.
Implement the flushing step correctly by invalidating the current step
operation, which would activate the new step operation.
Add gst_segment_to_position() that converts a running_time to a position in the
segment. A faulty variant of this function is currently used in inputselector
but we'll need it for frame stepping too.
API: GstSegment::gst_segment_to_position()
Update framestep document, we want to pass the flush flag in the step-done
message.
Add flush flag to the gstmessage.
Update examples to use the new step-done message api.
Implement framestep with playback rates < 0.0 too.
Add new STEP event and methods for creating/parsing the event
Update design docs.
Add new STEP_DONE message and method to create/parse.
API: GstEvent::gst_event_new_step()
API: GstEvent::gst_event_parse_step()
API: GstMessage::gst_message_new_step_done()
API: GstMessage::gst_message_parse_step_done()
Add a start_time field and some methods. The start_time will contain the
running_time of when the element last went to paused. This time can be user to
report the position in PAUSED but also to do more correct clipping and
stepping later.
Add a gst_element_lost_state_full() with an extra argument to control
distribution of a new base_time. We will need this for flushing step
operations.
API: GstElement::gst_element_lost_state_full()
Reuse buffer code for bufferlists. Not sure if this measurably impacts performance
for the simple buffer case, if it does after doing some benchmarks, we can
decouple it later.
Fixes#572285
Remove the static function set on the TaskPool before _prepare() is called and
allow for assigning a function to a Task when we _push().
Update the examples
Pass the thread object in a GValue, which would allow the application to figure
out the type of the object instead of us having to explicitly code it in a
message field.
Add the first version of the STREAM_STATUS message design docs.
This message will be used to give applications more control over the
streaming threads.
Two new log levels to dump FIXMEs into the log and to log data
in form of a hex dump (#578114).
API: GST_CAT_FIXME_OBJECT
API: GST_CAT_MEMDUMP_OBJECT
API: GST_CAT_FIXME
API: GST_CAT_MEMDUMP
API: GST_FIXME_OBJECT
API: GST_MEMDUMP_OBJECT
API: GST_FIXME
API: GST_MEMDUMP
Yes, 'codec' isn't exactly the best word, but let's be consistent with AUDIO_CODEC
and VIDEO_CODEC (which may be 'raw' formats as well after all). Prerequisite for
bug #576552.
Add some of the bits needed for an uninstalled gst-rtsp-server (so gdb works
on the examples etc.). Python bits are still missing, and we might need an
-uninstalled.pc file as well in the future. Break up very long lines to make
them easier to read and maintain. Also remove gst-plugins paths from the
old days.
This will be mostly useful in all elements that have some kind of internal
seek/index table. Currently almost all of them (or even all of them)
are using a linear search although the used array is already sorted,
wasting some CPU time without good reason.
Fixes bug #573623.
We got the constants for G_LITTLE_ENDIAN and G_BIG_ENDIAN the wrong way around in some docs (fixes: #572392). Also mention
G_BYTE_ORDER in the audio types section.
Add a GST_MESSAGE_REQUEST_STATE that can be posted by element when they would
like to have the application change the state of the pipeline. the primary use
case is to pause the pipeline when an audio mixer is mixing a higher priority
stream but it can also be used for other purposes.
Add some docs and a unit test.
Implement the REQUEST_STATE message in gst-launch.
API: gst_message_new_request_state()
API: gst_message_parse_request_state()
API: GST_MESSAGE_REQUEST_STATE
Replace all mentions of CVS with git. Add link to gst-uninstalled script in cgit and to SubmittingPatches page in wiki. Fix some typos. Update indenting rules to what we actually use (#571646).
This tag will list a homepage for the media,
i.e. the artist's or movie's homepage.
This is different to GST_TAG_LOCATION as the latter
lists the original location of the media.
Fixes bug #571227.
Add a property to select the clock type, currently REALTIME and MONOTONIC when
posix timers are available.
Implement the systemclock with GstPoll instead of GCond. This allows us to
schedule timeouts with nanosecond precission on newer kernels and with ppoll
support. It's also resilient to changes to the systemclock because of NTP or
similar.
Add a special timer mode in GstPoll that makes it only use the control socket
with a timeout to schedule timeouts. Also add a pair of methods to wakeup the
timeout thread.
API: GstPoll::gst_poll_new_timer()
API: GstPoll::gst_poll_write_control()
API: GstPoll::gst_poll_read_control()
We don't want to use -Wall -Werror and friends when building the gtk-doc-generated
$docmodule-scan.c, since we can't easily fix stuff if a certain gtk-doc/compiler
combination breaks the build. Fixes build on ubuntu intrepid.
Original commit message from CVS:
* docs/faq/gst-uninstalled:
Add libgstapp-0.10 from -base to search path and remove the old
lib from -bad from the search path.
Original commit message from CVS:
* docs/gst/gstreamer-sections.txt:
* gst/gstquark.c:
* gst/gstquark.h:
* gst/gstquery.c: (gst_query_new_uri), (gst_query_set_uri),
(gst_query_parse_uri):
* gst/gstquery.h:
API: Add URI query type. This is useful to query the URI
of a sink/source element and can be used by demuxers that
need to get data from other files.
This query should go upstream by default.
Fixes bug #562949.
* plugins/elements/gstfdsink.c: (gst_fd_sink_query):
* plugins/elements/gstfdsrc.c: (gst_fd_src_class_init),
(gst_fd_src_query):
* plugins/elements/gstfilesink.c: (gst_file_sink_query):
* plugins/elements/gstfilesrc.c: (gst_file_src_class_init),
(gst_file_src_query):
Implement URI query.
Original commit message from CVS:
* docs/gst/gstreamer-sections.txt:
* gst/gsttagsetter.c:
* gst/gsttagsetter.h:
Rename api added in previous commit and add since tag to docs.
API: gst_tag_setter_reset_tags()
Original commit message from CVS:
* docs/gst/gstreamer-sections.txt:
* gst/gsttagsetter.c:
* gst/gsttagsetter.h:
Add function to reset tagsetter for element reuse.
API: gst_tag_setter_flush()
Original commit message from CVS:
* docs/design/part-TODO.txt:
Remove the seqnum entry that we implemented in 0.10 already.
Add entry about removing the format return value for queries.
Original commit message from CVS:
* docs/gst/gstreamer-sections.txt:
* gst/gstbin.c: (gst_bin_recalculate_latency),
(gst_bin_change_state_func):
* gst/gstbin.h:
Add method to recalculate and redistribute the latency on a bin.
API: gst_bin_recalculate_latency().
Original commit message from CVS:
2008-11-04 Andy Wingo <wingo@pobox.com>
Add sequence numbers to events and messages. See #559250.
* gst/gstutils.c (gst_util_seqnum_next, gst_util_seqnum_compare):
New functions.
* gst/gstevent.h:
* gst/gstevent.c (_gst_event_copy, gst_event_new): Initialize new
events with a new sequence number, and copy it when copying.
(gst_event_get_seqnum, gst_event_set_seqnum): Accessors for an
event's sequence number.
* gst/gstmessage.h:
* gst/gstmessage.c (_gst_message_copy, gst_message_new_custom):
(gst_event_get_seqnum, gst_event_set_seqnum): As with events, so
with messages.
* docs/gst/gstreamer-sections.txt: Add new functions to the docs.
