Adds API to get or peek a sub-reader of a certain size from
a given byte reader. This is useful when parsing nested chunks,
one can easily get a byte reader for a sub-chunk and make
sure one never reads beyond the sub-chunk boundary.
API: gst_byte_reader_peek_sub_reader()
API: gst_byte_reader_get_sub_reader()
Adds gst_byte_reader_masked_scan_uint32_peek just like
GstAdapter has a _peek and non _peek version
Upgraded tests to check that the returned value is correct in the
_peek version
API: gst_byte_reader_masked_scan_uint32_peek
https://bugzilla.gnome.org/show_bug.cgi?id=728356
Adds a utility struct that is capable of storing and aggregating flow returns
associated with pads.
This way all demuxers will have a standard function to use and have the
same expected results.
Includes tests.
https://bugzilla.gnome.org/show_bug.cgi?id=709224
Stores the last result of a gst_pad_push or a pull on the GstPad and provides
a getter and a macro to access this field.
Whenever the pad is inactive it is set to FLUSHING
API: gst_pad_get_last_flow_return
https://bugzilla.gnome.org/show_bug.cgi?id=709224
Currently there is no other way to unlock a buffer pool other then
stopping it. This may have the effect of freeing all the buffers,
which is too heavy for a seek. This patch add a method to enter and
leave flushing state. As a convenience, flush_start/flush_stop
virtual are added so pool implementation can also unblock their own
internal poll atomically with the rest of the pool. This is fully
backward compatible with doing stop/start to actually flush the pool
(as being done in GstBaseSrc).
https://bugzilla.gnome.org/show_bug.cgi?id=727611
When we call gst_buffer_pool_set_config() the pool may return FALSE and
slightly change the parameters. This helper is useful to do the minial required
validation before accepting the modified configuration.
https://bugzilla.gnome.org/show_bug.cgi?id=727916
Events passing through #GstPads that have a running time
offset set via gst_pad_set_offset() will get their offset
adjusted according to the pad's offset.
If the event contains any information that related to the
running time, this information will need to be updated
before usage with this offset.
This defaults to TRUE and if it is set to FALSE it is the subclasses
responsibility to return GST_FLOW_EOS from the create() vmethod once
the stream is done.
Adds a variant of the _push function that doesn't check the queue limits
before adding the new item. It is useful when pushing an element to the
queue shouldn't lock the thread.
One particular scenario is when the queue is used to serialize buffers
and events that are going to be pushed from another thread. The
dataqueue should have a limit on the amount of buffers to be stored to
avoid large memory consumption, but events can be considered to have
negligible impact on memory compared to buffers. So it is useful to be
used to push items into the queue that contain events, even though the
queue is already full, it shouldn't matter inserting an item that has
no significative size.
This scenario happens on adaptive elements (dashdemux / mssdemux) as
there is a single download thread fetching buffers and putting into the
dataqueues for the streams. This same download thread can als generate
events in some situations as caps changes, eos or a internal control
events. There can be a deadlock at preroll if the first buffer fetched
is large enough to fill the dataqueue and the download thread and the
next iteration of the download thread decides to push an event to this
same dataqueue before fetching buffers to other streams, if this push
locks, the pipeline will be stuck in preroll as no more buffers will be
downloaded.
There is a somewhat common practice in dash streams to have a single
very large buffer for audio and one for video, so this will always
happen as the download thread will have to push an EOS right after
fetching the first buffer for any stream.
API: gst_data_queue_push_force
https://bugzilla.gnome.org/show_bug.cgi?id=705694
All streams that have the same group id are supposed to be played
together, i.e. all streams inside a container file should have the
same group id but different stream ids. The group id should change
each time the stream is started, resulting in different group ids
each time a file is played for example.
API: gst_value_array_append_and_take_value
API: gst_value_list_append_and_take_value
We were already using this internally, this makes it public for code
which frequently appends values which are expensive to copy (like
structures, arrays, caps, ...).
Avoids copies of the values for users. The passed GValue will also
be 0-memset'ed for re-use.
New users can replace this kind of code:
gst_value_*_append_value(mycontainer, &myvalue);
g_value_unset(&myvalue);
by:
gst_value_*_append_and_take_value(mycontainer, &myvalue);
https://bugzilla.gnome.org/show_bug.cgi?id=701632
This function works just like gst_data_queue_pop, but it doesn't
remove the object from the queue.
Useful when inspecting multiple GstDataQueues to decide from which
to pop the element from.
Add: gst_data_queue_peek
Source elements with limited bandwidth capabilities and supporting
buffering for downstream elements should set this flag when answering
a scheduling query. This is useful for the on-disk buffering scenario
of uridecodebin to avoid checking the URI protocol against a list of
hardcoded protocols.
Bug 693484
These are meant to specify features in caps that are required
for a specific structure, for example a specific memory type
or meta.
Semantically they could be though of as an extension of the media
type name of the structures and are handled exactly like that.
Elements should override GstElement::set_context() and also call
gst_element_set_context() to keep this context up-to-date with
the very latest context they internally use.
The duration should be re-queried via a query using the
normal path, we don't want applications to use the value
from the message itself, since it might no match what a
duration query done from the sink upstream might yield.
Also disables duration caching in GstBin. It should be
added back again at some point.
Not so useful: just adds/reads stuff from an internal GList without
actually doing anything with those paths, so remove for now:
gst_registry_add_path
gst_registry_get_path_list
https://bugzilla.gnome.org/show_bug.cgi?id=608841
Add an alternative version of gst_pad_check_reconfigure that doesn't
clear the reconfigure flag.
Useful for increasing error resilience without duplicating the
reconfigure code in pad task functions.
API: gst_pad_needs_reconfigure
https://bugzilla.gnome.org/show_bug.cgi?id=681198
This is because we need to be able to signal different TOCs
to downstream elements such as muxers and the application,
and because we need to send both types as events (because
the sink should post the TOC messages for the app in the
end, just like tag messages are now posted by the sinks),
and hence need to make TOC events multi-sticky.
https://bugzilla.gnome.org/show_bug.cgi?id=678742
Move the locking methods from GstMemory to GstMiniObject.
Add a miniobject flag to enable LOCKABLE objects. LOCKABLE objects can
use the lock/unlock API to control the access to the object.
Add a minobject flag that allows you to lock an object in readonly mode.
Modify the _is_writable() method to check the shared counter for LOCKABLE
objects. This allows us to control writability separately from the refcount for
LOCKABLE objects.
This is a queue which has the same API as GQueue, except that:
* It uses an array, instead of a doubled-linked-list
* The array can only grow.
This code is not-threadsafe. It is up to the owner to make sure the
proper locking is taken before calling this API.
Make a gst_buffer_append_region() function that allows you to append a memory
region from one buffer to another. This is a more general version of
gst_buffer_append().
Some tag parsers and writers use same datetime format based on ISO 8601.
We can reduce some code by creating some general functions for it.
API: gst_date_time_to_iso8601_string()
API: gst_date_time_new_from_iso8601_string()
https://bugzilla.gnome.org/show_bug.cgi?id=678031
Let's keep it simple for now:
gst_toc_setter_reset_toc() -> gst_toc_setter_reset()
gst_toc_setter_get_toc_copy() -> removed
gst_toc_setter_get_toc() -> returns a ref now
gst_toc_setter_get_toc_entry_copy() -> removed,
use TOC functions instead
gst_toc_setter_get_toc_entry() -> removed,
use TOC functions instead
gst_toc_setter_add_toc_entry() -> removed,
to avoid problems with (refcount-dependent)
writability of TOC; use TOC functions instead
Rename gst_base_sink_wait_eos() to gst_base_sink_wait() to avoid confusion and
introspection problems with the ::wait_eos vmethod. Also this method can be used
to wait for other things than EOS. Update the docs a little.
Add a new message to reset the pipeline running_time. Currently reseting the
pipeline can only be requested in the async_done message which means that the
pipeline needs to be prerolled. It is better to move this to a separate message.
They can be used to select snapping behavior (to previous, next, or
nearest location, where relevant) when seeking.
The seeking implementation (eg, demuxer) may currently ignore some
or all of these flags.
It's only used internally, most other users will likely
want to use gst_registry_find_plugin() directly instead
(and if not, they can easily walk the list and doing the
strcmp themselves).
This is an implementation detail really, and it's not
clear what anyone would do with this. It's unused as
far as I'm aware, so just remove it for now.
