In order for gst_harness_new_full to be MT-safe the increase and
decrease of HARNESS_REF must be MT-safe. This allows for creating
multiple harnesses from different threads wrapping the same element.
https://bugzilla.gnome.org/show_bug.cgi?id=754661
1. Get a list of pad templates from the element class, not the
factory. This allows us to interact with test-elements that does
not have a factory.
2. Use the pad_template_caps in caps-queries when caps is not set
explicitly on the pad. Not doing so is simply wrong, and prohibits
interactions with special templates used for testing.
https://bugzilla.gnome.org/show_bug.cgi?id=754193
There exist cases where a reconfigure event was propagated from
downstream, but caps didn't change. In this case, we would
reconfigure only when the next buffer arrives. The problem is that
due to the allocation query being cached, the return query parameters
endup outdated.
In this patch we refactor the reconfigurating code into a function, and
along with reconfiguring when a new buffer comes in, we also reconfigure
when a query allocation arrives.
https://bugzilla.gnome.org/show_bug.cgi?id=753850
Explicitly keep track again whether upstream tags or parser tags
already contain bitrate information, and only force a tag update
for a bitrate if we are actually going to add the bitrate to the
taglist later. This fixes constant re-sending of the same taglist,
because upstream provided a bitrate already and we didn't add it,
so we didn't save the 'posted' bitrate, which would then in turn
again trigger the 'bitrate has changed too much, update tags'
code path. Fixes tag spam with m4a files for example.
https://bugzilla.gnome.org/show_bug.cgi?id=679768
In 0.10 there were no sticky events, and all tag events
sent would just be merged with the previously-received
tags. In 1.x we have sticky events, and the tags in the
tag event(s) should at all times carry the complete tags,
so we can't just push some tags and then just push tags
with just bitrates to update the bitrates, etc.
Instead we need to keep track of the upstream stream tags
received, of the tags set by the video decoder subclass,
and send an updated tag event with the combined tags
including our own bitrate tags (if applicable) whenever
the upstream tags, the subclass tags or any of our bitrates
change.
https://bugzilla.gnome.org/show_bug.cgi?id=679768
This is needed so that we can do proper tag handling
all around, and combine the upstream tags with the
tags set by the subclass and any extra tags the
base class may want to add.
API: gst_base_parse_merge_tags()
https://bugzilla.gnome.org/show_bug.cgi?id=679768
Use gst_pad_peer_query_duration() and remove a few
unnecessary levels of indentation. Rest of code might
looks a bit questionable, but leave it as is for now.
According to the design docs:
The ACCEPT_CAPS query is not required to work recursively, it can simply
return TRUE if a subsequent CAPS event with those caps would return
success.
So make it a shallow check instead of recursivelly check downstream.
https://bugzilla.gnome.org/show_bug.cgi?id=748635
GstPad has a flag for suggesting if the accept-caps
query should use intersect instead of the default
subset caps operation to verify if the caps would be
acceptable.
basetransform currently always uses the subset check and
this patch makes it honor the flag for using intersect
if it is set.
https://bugzilla.gnome.org/show_bug.cgi?id=748635
To be able to disable the slightly "magic" forwarding of the
necessary events between the harnesses.
Also introduce a new test-suite for GstHarness, that documents the
feature, and should hopefully expand into documenting most of the
features the harness possesses.
https://bugzilla.gnome.org/show_bug.cgi?id=752746
As of now, even for stream completly inside segment, there is no
guarantied that the DTS will be inside the segment. Specifically
for H.264 with B-Frames, the first few frames often have DTS that
are before the segment.
Instead of using the sync timestamp to clip out of segment buffer,
take the duration from the start/stop provided by the sub-class, and
check if the pts and pts_end is out of segment.
https://bugzilla.gnome.org/show_bug.cgi?id=752791
Even though asserts can't be disabled in GstHarness, Coverity still
complains about running code inside them. Moving the code to outside the
g_asserts().
CID #1311326, #1311327, #1311328
gst_query_find_allocation_meta() requires the query to be
writable to work. This patch ensure avoids taking a reference
on the query, so we can now check if a certain allocation meta
is present.
https://bugzilla.gnome.org/show_bug.cgi?id=752661
By introducing gst_harness_add_src_harness and gst_harness_add_sink_harness
we collect all sub-harness setup in one function, making the previous
sub-harness creation functions now calls these directly, and making it
much easier (and less error-prone) to add your own src or sink-harness
using the more generic harness-creation functions.
This line has no purpose, clearly gst_segment_do_seek() is doing
the right job, also, having the start time (a timestamp) be that
same as time (the stream time) is quite odd.
https://bugzilla.gnome.org/show_bug.cgi?id=750783
The element flag does not indicate wether a bin should be tested as a
source or as a sink, eg. a bin with the sink flag may still have a
source pad and a bin with the source flag may have a sink pad. In this
case it is better to determine the element type by looking at the
available pads and pad templates.
Also rename srcpad and sinkpad where it actually represents
element_srcpad_name and element_sinkpad_name.
https://bugzilla.gnome.org/show_bug.cgi?id=752493
For files which are smaller than 1.5 seconds, the duration
estimation does not happen. So the duration will always be
displayed as 0. Updating the duration on EOS when the estimation
has not happened already
https://bugzilla.gnome.org/show_bug.cgi?id=750131
We must make the buffer writable to write its PTS and DTS, and also
reset its duration.
The behaviour is now the same as before commit c3bcbadd, except metas
might still be attached to the buffer extracted from the adapter.
https://bugzilla.gnome.org/show_bug.cgi?id=752092
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
POOL meta just means that this specific instance of the meta is related to a
pool, a copy should be made when reasonable and the flag should just not be
set in the copy.
POOL meta just means that this specific instance of the meta is related to a
pool, a copy should be made when reasonable and the flag should just not be
set in the copy.
This preserves GstMeta properly unless the subclass does special things. It's
enough to make h264parse's stream-format/alignment conversion pass through
metas as needed.
https://bugzilla.gnome.org/show_bug.cgi?id=742385
All functions that return a GstBuffer or a list of them will now copy
all GstMeta from the input buffers except for meta with GST_META_FLAG_POOLED
flag or "memory" tag.
This is similar to the existing behaviour that the caller can't assume
anything about the buffer flags, timestamps or other metadata. And it's
also the same that gst_adapter_take_buffer_fast() did before, and what
gst_adapter_take_buffer() did if part of the first buffer or the complete
first buffer was requested.
https://bugzilla.gnome.org/show_bug.cgi?id=742385
The doc generator get confused with the inline structure. So
workaround by wrapping the inner of the structure with
public/private mark, and document that GST_COLLECT_PADS_DTS macro
shall be used to access this.
* 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
There was few Since: mark missing their column. Also unify the way
we set the Since mark on enum value and structure members. These
sadly don't show up in the index.
These are not usable as they are, and can easily lead to crash
or leaks. This also silence warning from the scanner. If we manage to
make this usable, we can then remove that mark, it will require
to make this type boxed.
gstbasetransform.h:196: Warning: GstBase: "@submit_input_buffer" parameter unexpected at this location:
* @submit_input_buffer: Function which accepts a new input buffer and pre-processes it.
gstnetcontrolmessagemeta.c:103: Warning: GstNet: gst_buffer_add_net_control_message_meta: unknown parameter 'message' in documentation comment, should be 'addr'
Make gst_collect_pads_clip_running_time() function also store the
signed DTS in the CollectData. This signed DTS value can be used by
muxers to properly handle streams where DTS can be negative initially.
https://bugzilla.gnome.org/show_bug.cgi?id=740575
The internal clock is only used for slaving against the remote clock, while
the user-facing GstClock can be additionally slaved to another clock if
desired. By default, if no master clock is set, this has exactly the same
behaviour as before. If a master clock is set (which was not allowed before),
the user-facing clock is reporting the remote clock as internal time and
slaves this to the master clock.
This also removes the weirdness that the internal time of the netclientclock
was always the system clock time, and not the remote clock time.
https://bugzilla.gnome.org/show_bug.cgi?id=750574
Allow for sub-classes which want to collate incoming buffers or
split them into multiple output buffers by separating the input
buffer submission from output buffer generation and allowing
for looping of one of the phases depending on pull or push mode
operation.
https://bugzilla.gnome.org/show_bug.cgi?id=750033
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.
We extend our calculations to work with local send time, remote receive time,
remote send time and local receive time. For the netclientclock protocol,
remote receive and send time are assumed to be the same value.
For the results, this modified calculation makes absolutely no difference
unless the two remote times are different.
This improves accuracy on wifi or similar networks, where the RTT can go very
high up for a single observation every now and then. Without filtering them
away completely, they would still still modify the average RTT, and thus all
clock estimations.
They don't necessarily use the same underlying clocks (e.g. on Windows), or
might be configured to a different clock type (monotonic vs. real time clock).
We need the values a clean system clock returns, as those are the values used
by the internal clocks.
If the delay measurement is too far away from the median of the window of last
delay measurements, we discard it. This increases accuracy on wifi a lot.
https://bugzilla.gnome.org/show_bug.cgi?id=749391
We should do some more measurements with all these and check how much sense
they make for PTP. Also enabling them means not following IEEE1588-2008 by the
letter anymore.
https://bugzilla.gnome.org/show_bug.cgi?id=749391
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
Just create the cancellable fd once and keep it around instead
of creating/closing it for every single packet. Since we spend
most time waiting for packets, an fd is alloced and in use pretty
much all the time anyway.
We were segfaulting because g_sequence_search was returning the iter_end,
and that iterator does not contain anything and thus should not be used
directly
In basesink functions gst_base_sink_chain_unlocked(), below code is used to
checking if buffer is late before doing prepare call to save some effort:
if (syncable && do_sync)
late =
gst_base_sink_is_too_late (basesink, obj, rstart, rstop,
GST_CLOCK_EARLY, 0, FALSE);
if (G_UNLIKELY (late))
goto dropped;
But this code has problem, it should calculate jitter based on current media
clock, rather than just passing 0. I found it will drop all the frames when
rewind in slow speed, such as -2X.
https://bugzilla.gnome.org/show_bug.cgi?id=749258
Since frame->priv->discont was cleared earlier,
GST_BASE_PARSE_FLAG_LOST_SYNC was never being set.
Take the chance to refactor the frame creation a bit to
organize the flags setting and reset.
https://bugzilla.gnome.org/show_bug.cgi?id=738237
Otherwise we're going to set a rather arbitrary DTS of segment.start (usually
0) for live sources, which confuses synchronization if the source started
capturing at a later time. And it's especially wrong for raw media, for which
we should not set any DTS at all.
https://bugzilla.gnome.org/show_bug.cgi?id=747731
It could be triggered by:
gst-launch-1.0 videotestsrc num-buffers=20 ! videcrop bottom=214748364 ! videoconvert ! autovideosink
Spotted while testing:
https://bugzilla.gnome.org/show_bug.cgi?id=743910
The flush-stop event should not restart the task for live sources unless
the element is playing. This was breaking seeks in pause with the rtpsrc.
https://bugzilla.gnome.org/show_bug.cgi?id=635701
Otherwise baseparse will consider empty streams to be an error while
an empty stream is a valid scenario. With this patch, errors would
only be emitted if the parser received data but wasn't able to
produce any output from it.
This change is only for push-mode operation as in pull mode an
empty file can be considered an error for the one driving the
pipeline
Includes a unit test for it
https://bugzilla.gnome.org/show_bug.cgi?id=733171
check_run.c: In function 'sig_handler':
check_run.c:127:13: warning: 'child_sig' may be used uninitialized in this function [-Wmaybe-uninitialized]
killpg(group_pid, child_sig);
^
check_run.c:130:31: warning: 'idx' may be used uninitialized in this function [-Wmaybe-uninitialized]
sigaction(sig_nr, &old_action[idx], NULL);
^
Otherwise e.g. ctrl+c in the test runner exits the test runner, while the test
itself is still running in the background, uses CPU and memory and potentially
never exits (e.g. if the test ran into a deadlock or infinite loop).
The reason why we have to manually kill the actual tests is that after
forking they will be moved to their own process group, and as such are
not receiving any signals sent to the test runner anymore. This is supposed
to be done to make it easier to kill a test, which it only really does if
the test itself is forking off new processes.
This fix is not complete though. SIGKILL can't be caught at all, and error
signals like SIGSEGV, SIGFPE are currently not caught. The latter will only
happen if there is a bug in the test runner itself, and as such seem less
important.
Large scale skip is an optimization, and thus it is safer to
stop skipping than to continue. Clear skip on segments and
discontinuities, as these are points where it is possible that
the original idea of "bytes to skip" changes.
