A core meta which helps implement the old concept
of sub-buffering in some situations, by making it
possible for a buffer to keep a ref on a different
parent buffer. The parent buffer is unreffed when
the Meta is freed.
This meta is used to ensure that a buffer whose
memory is being shared to a child buffer isn't freed
and returned to a buffer pool until the memory
is.
https://bugzilla.gnome.org/show_bug.cgi?id=750039
* Fix function name in sections.txt
* Add few missing or fix miss-named
* Workaround gtk-doc being confused with non typedef
types (loose track of public/private
One of the nice feature in GTK doc is that it generate indexes
of added APIs base on the since marker. Include that in our doc
while fixing the issue of duplicate ID (produce xml contains that
id it seems)
Follow up of 7130230ddb
Provide the memory implementation the GstMapInfo that will be used to
map/unmap the memory. This allows the memory implementation to use
some scratch space in GstMapInfo to e.g. track different map/unmap
behaviour or store extra implementation defined data about the map
in use.
https://bugzilla.gnome.org/show_bug.cgi?id=750319
This overrides the default latency handling and configures the specified
latency instead of the minimum latency that was returned from the LATENCY
query.
https://bugzilla.gnome.org/show_bug.cgi?id=750782
This uses all of the netclientclock code, except for the generation and
parsing of packets. Unfortunately some code duplication was necessary
because GstNetTimePacket is public API and couldn't be extended easily
to support NTPv4 packets without breaking API/ABI.
GstPtpClock implements a PTP (IEEE1588:2008) ordinary clock in
slave-only mode, that allows a GStreamer pipeline to synchronize
to a PTP network clock in some specific domain.
The PTP subsystem can be initialized with gst_ptp_init(), which then
starts a helper process to do the actual communication via the PTP
ports. This is required as PTP listens on ports < 1024 and thus
requires special privileges. Once this helper process is started, the
main process will synchronize to all PTP domains that are detected on
the selected interfaces.
gst_ptp_clock_new() then allows to create a GstClock that provides the
PTP time from a master clock inside a specific PTP domain. This clock
will only return valid timestamps once the timestamps in the PTP domain
are known. To check this, the GstPtpClock::internal-clock property and
the related notify::clock signal can be used. Once the internal clock
is not NULL, the PTP domain's time is known. Alternatively you can wait
for this with gst_ptp_clock_wait_ready().
To gather statistics about the PTP clock synchronization,
gst_ptp_statistics_callback_add() can be used. This gives the
application the possibility to collect all kinds of statistics
from the clock synchronization.
https://bugzilla.gnome.org/show_bug.cgi?id=749391
gst_clock_wait_for_sync(), gst_clock_is_synced() and gst_clock_set_synced()
plus a signal to asynchronously wait for the clock to be synced.
This can be used by clocks to signal that they need initial synchronization
before they can report any time, and that this synchronization can also get
completely lost at some point. Network clocks, like the GStreamer
netclientclock, NTP or PTP clocks are examples for clocks where this is useful
to have as they can't report any time at all before they're synced.
https://bugzilla.gnome.org/show_bug.cgi?id=749391
GstFlagSet is a new type designed for negotiating sets
of boolean capabilities flags, consisting of a 32-bit
flags bitfield and 32-bit mask field. The mask field
indicates which of the flags bits an element needs to have
as specific values, and which it doesn't care about.
This allows efficient negotiation of arrays of boolean
capabilities.
The standard serialisation format is FLAGS:MASK, with
flags and mask fields expressed in hexadecimal, however
GstFlagSet has a gst_register_flagset() function, which
associates a new GstFlagSet derived type with an existing
GFlags gtype. When serializing a GstFlagSet with an
associated set of GFlags, it also serializes a human-readable
form of the flags for easier debugging.
It is possible to parse a GFlags style serialisation of a
flagset, without the hex portion on the front. ie,
+flag1/flag2/flag3+flag4, to indicate that
flag1 & flag4 must be set, and flag2/flag3 must be unset,
and any other flags are don't-care.
https://bugzilla.gnome.org/show_bug.cgi?id=746373
The old gst_object_has_ancestor will call the new code. This establishes the
symetry with the new gst_object_has_as_parent.
API: gst_object_has_as_ancestor()
In order to support some types of protected streams (such as those
protected using DASH Common Encryption) some per-buffer information
needs to be passed between elements.
This commit adds a GstMeta type called GstProtectionMeta that allows
protection specific information to be added to a GstBuffer. An example
of its usage is qtdemux providing information to each output sample
that enables a downstream element to decrypt it.
This commit adds a utility function to select a supported protection
system from the installed Decryption elements found in the registry.
The gst_protection_select_system function that takes an array of
identifiers and searches the registry for a element of klass Decryptor that
supports one or more of the supplied identifiers. If multiple elements
are found, the one with the highest rank is selected.
This commit adds a unit test for the gst_protection_select_system
function that adds a fake Decryptor element to the registry and then
checks that it can correctly be selected by the utility function.
This commit adds a unit test for GstProtectionMeta that creates
GstProtectionMeta and adds & removes it from a buffer and performs some
simple reference count checks.
API: gst_buffer_add_protection_meta()
API: gst_buffer_get_protection_meta()
API: gst_protection_select_system()
API: gst_protection_meta_api_get_type()
API: gst_protection_meta_get_info()
https://bugzilla.gnome.org/show_bug.cgi?id=705991
1) segment.accum -> segment.base
2) Refer to GstSegment members as S.foo instead of
NS.foo, the event is now called a segment event
rather than newsegment event.
3) There's no more abs_rate field in GstSegment,
and there never was an abs_applied_rate field.
https://bugzilla.gnome.org/show_bug.cgi?id=690564
Also skip gst_pipeline_get_clock() and gst_pipeline_set_clock() from the
bindings as they are confused with gst_element_*_clock().
API: gst_pipeline_get_pipeline_clock()
https://bugzilla.gnome.org/show_bug.cgi?id=744442
GstNetAddress can be used to store ancillary data which was received with
or is to be sent alongside the buffer data. When used with socket sinks
and sources which understand this meta it allows sending and receiving
ancillary data such as unix credentials (See `GUnixCredentialsMessage`)
and Unix file descriptions (See `GUnixFDMessage`).
This will be useful for implementing protocols which use file-descriptor
passing in payloaders/depayloaders without having to re-implement all the
socket handling code already present in elements such as multisocketsink,
etc. This, in turn, will be useful for implementing zero-copy video IPC.
This meta uses the platform independent `GSocketControlMessage` API
provided by GLib as a part of GIO. As a result this new meta does not
require any new dependencies or any conditional compliation for
portablility, although it is unlikely to do anything useful on non-UNIX
platforms.
gst_bin_sync_children_states() will iterate over all the elements of a bin and
sync their states with the state of the bin. This is useful when adding many
elements to a bin and would otherwise have to call
gst_element_sync_state_with_parent() on each and every one of them.
https://bugzilla.gnome.org/show_bug.cgi?id=745042
These docs missed many details that were not obvious and because of that
handled in a few different, incompatible ways in different elements and base
classes.
https://bugzilla.gnome.org/show_bug.cgi?id=744106
TRUE is 1, but every other non-zero value is also considered true. Comparing
for equality with TRUE would only consider 1 but not the others.
Also normalize booleans in a few places.
The point of this example is to show how to set caps
on the source pad once it has been set on the sink pad.
So, in passthrough mode, the caps is just copied to the
source pad.
https://bugzilla.gnome.org/show_bug.cgi?id=738153
Add a method letting people to ensure that unreffing one object
leads to its destruction, and possibly the destruction of more object
(think destruction of a GstBin etc...).
https://bugzilla.gnome.org/show_bug.cgi?id=736477
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()
* GstGlobalDeviceMonitor was renamed to GstDeviceMonitor
* Expand GST_MESSAGE_DEVICE to the full enum value names
* Correct the incorrect references to the GstDeviceProvider interfaces
* Describe caps arguments for gstcheck interface
* Add missing docs for GstNetAddressMeta and its add function
* Add docs for toc helper macros
* Avoid refering to GstValueList type as done elsewhere
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732786
The start and stop should represent the currently downloading region.
The estimated-total should represent the remaining time to download
the currently downloading region. This makes it a lot more useful
for applications because they can then use those values to update
the fill region and use the estimated time to delay playback.
Update the docs with this clarification.
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
They are very confusing for people, and more often than not
also just not very accurate. Seeing 'last reviewed: 2005' in
your docs is not very confidence-inspiring. Let's just remove
those comments.
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.
Add loop_type and repeat_count fields to GstTocEntry plus setters and getters.
This allows to represent edit-lists in a toc as well as loops in instruemnts (wav, xi).
API: gst_toc_entry_set_loop
API: gst_toc_entry_get_loop
Using info from gst-plugins-base/docs/design .
Encoded streams might make use of the raw properties, so list them all under foo/* .
For foo/raw, only note which of these properties are mandatory.
I didn't take a closer look at the raw formats yet. Those might still be out-of-date.
https://bugzilla.gnome.org/show_bug.cgi?id=724187
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.
* add many missing declarations to sections
* GstController has been removed, update docs
* skip GstIndex when generating documentation
* rephrase so gtkdoc doesn't imagine return value
* add missing argument description for gst_context_new()
* document GstOutputSelectorPadNegotiationMode and move to header-file
https://bugzilla.gnome.org/show_bug.cgi?id=719614
In one of the examples about gst_my_filter_setcaps() there is a variable
declared as "rate", but then the name "samplerate" is used when setting
the caps.
Use the name "rate" everywhere in gst_my_filter_setcaps().
https://bugzilla.gnome.org/show_bug.cgi?id=710876
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.
The check is done using curl (if available). It lists the curl exit code + http
status code (for those > 399) together with the use of the url in the code. The
check is not fatal.
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
and remove all the printf extension/specifier stuff for
the system printf. Next we need to add back the custom
specifiers to our own printf implementation.
https://bugzilla.gnome.org/show_bug.cgi?id=613081
This is equal to any other caps features but results in unfixed caps. It
would be used by elements that only look at the buffer metadata or are
currently working in passthrough mode, and as such don't care about any
specific features.
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 _1_0 suffixed environment variables override the
non-suffixed ones, so if we're in an environment that
sets the _1_0 suffixed ones, such as jhbuild, we need
to set those to make sure ours actually always get
used.
In flush-on-eos=true mode any data remaining in the queue is
discarded when an EOS event is received, and the EOS passed
downstream as soon as possible (instead of waiting for all
buffers in the queue to get processed by downstream first).
May or may not be useful in capture/encoding scenarios.
