This way we don't have to allocate/free temporary structs
for storing things in the queue array.
API: gst_queue_array_new_for_struct()
API: gst_queue_array_push_tail_struct()
API: gst_queue_array_peek_head_struct()
API: gst_queue_array_pop_head_struct()
API: gst_queue_array_drop_struct()
https://bugzilla.gnome.org/show_bug.cgi?id=750149
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
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
In order for a decrypter element to decrypt media protected using a
specific protection system, it first needs all the protection system
specific information necessary (E.g. information on how to acquire
the decryption keys) for that stream.
The GST_EVENT_PROTECTION defined in this commit enables this information
to be passed from elements that extract it (e.g. qtdemux, dashdemux) to
elements that use it (E.g. a decrypter element).
API: GST_EVENT_PROTECTION
API: gst_event_new_protection()
API: gst_event_parse_protection()
https://bugzilla.gnome.org/show_bug.cgi?id=705991
Do not do any checks for the start/stop in the new
gst_segment_to_running_time_full() method, we can let this be done by
the more capable gst_segment_clip() method. This allows us to remove the
enum of results and only return the sign of the calculated running-time.
We need to put the old clipping checks in the old
gst_segment_to_running_time() still because they work slightly
differently than the _clip methods.
See https://bugzilla.gnome.org/show_bug.cgi?id=740575
Add a helper method to get a running-time with a little more features
such as detecting if the value was before or after the segment and
negative running-time.
API: gst_segment_to_running_time_full()
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=740575
A variant of gst_buffer_copy that forces the underlying memory
to be copied.
This is added to avoid adding an extra reference to a GstMemory
that might belong to a bufferpool that is trying to be drained.
The use case is when the buffer copying is done to release the
old buffer and all its resources.
https://bugzilla.gnome.org/show_bug.cgi?id=745287
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
gst_clock_add_observation_unapplied() adds a new master/slave clock
observation and runs the regression without activating the new
calibration results.
gst_clock_adjust_with_calibration() uses directly passed calibration
parameters, instead of using the clock's current calibration,
allowing for calculations using pending or old calibration params
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
