This ensures that they really get processed in order with
buffers. Just waiting for the queue to be empty is sometimes not
enough as the buffers are dropped from the pad before the result is
pushed to the next element, sometimes resulting in surprising
re-ordering.
In the case an aggregator is created and pads are requested but only
linked later, we end up never updating the upstream latency.
This was because latency queries on pads that are not linked succeed,
so we never did a new query once a live source has been linked, so the
thread was never started.
https://bugzilla.gnome.org/show_bug.cgi?id=757548
The function needs to be unlocked if any data is received, but only
end the first buffer processing on an actual buffer, synchronized events
don't matter on the first buffer processing.
https://bugzilla.gnome.org/show_bug.cgi?id=781673
Allowing us to tell GstPad why we are failing an event, which might
be because we are 'flushing' even if the sinkpad is not in flush state
at that point.
Until now we would start the task when the pad is activated. Part of the
activiation concist of testing if the pipeline is live or not.
Unfortunatly, this is often too soon, as it's likely that the pad get
activated before it is fully linked in dynamic pipeline.
Instead, start the task when the first serialized event arrive. This is
a safe moment as we know that the upstream chain is complete and just
like the pad activation, the pads are locked, hence cannot change.
https://bugzilla.gnome.org/show_bug.cgi?id=757548
This fixes a race where we check if there is a clock, then it get
removed and we endup calling gst_clock_new_single_shot_id() with a NULL
pointer instead of a valid clock and also calling gst_object_unref()
with a NULL pointer later.
https://bugzilla.gnome.org/show_bug.cgi?id=757548
Previously, while allocating the pad number for a new pad, aggregator was
maintaining an interesting relationship between the pad count and the pad
number.
If you requested a sink pad called "sink_6", padcount (which is badly named and
actually means number-of-pads-minus-one) would be set to 6. Which means that if
you then requested a sink pad called "sink_0", it would be assigned the name
"sink_6" again, which fails the non-uniqueness test inside gstelement.c.
This can be fixed by instead setting padcount to be 7 in that case, but this
breaks manual management of pad names by the application since it then becomes
impossible to request a pad called "sink_2". Instead, we fix this by always
directly using the requested name as the sink pad name. Uniqueness of the pad
name is tested separately inside gstreamer core. If no name is requested, we use
the next available pad number.
Note that this is important since the sinkpad numbering in aggregator is not
meaningless. Videoaggregator uses it to decide the Z-order of video frames.
This code will never be called as max>=min in all cases. If the upstream
latency query returned min>max, the function already returned and all
values that are added to those have max>= min.
Not all aggregator subclasses will have a single pad template called sink_%u
and might do something special depending on what the application requests.
https://bugzilla.gnome.org/show_bug.cgi?id=757018
Otherwise they will receive a QOS event that has earliest_time=0 (because we
can't have negative timestamps), and consider their buffer as too late
https://bugzilla.gnome.org/show_bug.cgi?id=754356
In the case where you have a source giving the GstAggregator smaller
buffers than it uses, when it reaches a timeout, it will consume the
first buffer, then try to read another buffer for the pad. If the
previous element is not fast enough, it may get the next buffer even
though it may be queued just before. To prevent that race, the easiest
solution is to move the queue inside the GstAggregatorPad itself. It
also means that there is no need for strange code cause by increasing
the min latency without increasing the max latency proportionally.
This also means queuing the synchronized events and possibly acting
on them on the src task.
https://bugzilla.gnome.org/show_bug.cgi?id=745768
Before aggregator based elements always started at running time 0,
now it's possible to select the first input buffer running time or
explicitly set a start-time value.
https://bugzilla.gnome.org/show_bug.cgi?id=749966
Adding a pad will add a new upstream that might have a bigger minimum latency,
so we might have to wait longer. Or it might be the first live upstream, in
which case we will have to start deadline based aggregation.
Removing a pad will remove a new upstream that might have had the biggest
latency, so we can now stop waiting a bit earlier. Or it might be the last
live upstream, in which case we can stop deadline based aggregation.
And keep on querying upstream until we get a reply.
Also, the _get_latency_unlocked() method required being calld
with a private lock, so removed the _unlocked() variant from the API.
And it now returns GST_CLOCK_TIME_NONE when the element is not live as
we think that 0 upstream latency is possible.
https://bugzilla.gnome.org/show_bug.cgi?id=745768
One has to use the src_lock anyway to protect the min/max/live so they
can be notified atomically to the src thread to wake it up on changes,
such as property changes. So no point in having a second lock.
Also, the object lock was being held across a call to
GST_ELEMENT_WARNING, guaranteeing a deadlock.
While gst_aggregator_iterate_sinkpads() makes sure that every pad is only
visited once, even when the iterator has to resync, this is not all we have
to do for querying the latency. When the iterator resyncs we actually have
to query all pads for the latency again and forget our previous results. It
might have happened that a pad was removed, which influenced the result of
the latency query.
It was between another function and its helper function before, which was
confusing when reading the code as it had nothing to do with the other
functions.
This lock is not what is commonly known as a "stream lock" in GStremer,
it's not recursive and it's taken from the non-serialized FLUSH_START event.
https://bugzilla.gnome.org/show_bug.cgi?id=742684
steal_buffer() + unref seems to be a wide-spread idiom
(which perhaps indicates that something is not quite
right with the way aggregator pad works currently).
Instead of using the GST_OBJECT_LOCK we should have
a dedicated mutex for the pad as it is also associated
with the mutex on the EVENT_MUTEX on which we wait
in the _chain function of the pad.
The GstAggregatorPad.segment is still protected with the
GST_OBJECT_LOCK.
Remove the gst_aggregator_pad_peak_unlocked method as it does not make
sense anymore with a private lock.
https://bugzilla.gnome.org/show_bug.cgi?id=742684
Some members sometimes used atomic access, sometimes where not locked at
all. Instead consistently use a mutex to protect them, also document
that.
https://bugzilla.gnome.org/show_bug.cgi?id=742684
Reduce the number of locks simplify code, what is protects
is exposed, but the lock was not.
Also means adding an _unlocked version of gst_aggregator_pad_steal_buffer().
https://bugzilla.gnome.org/show_bug.cgi?id=742684
Whenever a GCond is used, the safest paradigm is to protect
the variable which change is signalled by the GCond with the same
mutex that the GCond depends on.
https://bugzilla.gnome.org/show_bug.cgi?id=742684
No need to use an iterator for this which creates a temporary
structure every time and also involves taking and releasing the
object lock many times in the course of iterating. Not to mention
all that GList handling in gst_aggregator_iterate_sinkpads().
The minimum latency is the latency we have to wait at least
to guarantee that all upstreams have produced data. The maximum
latency has no meaning like that and shouldn't be used for waiting.
When iterating sink pads to collect some data, we should take the stream lock so
we don't get stale data and possibly deadlock because of that. This fixes
a definitive deadlock in _wait_and_check() that manifests with high max
latencies in a live pipeline, and fixes other possible race conditions.
This simplifies the code and also makes sure that we don't forget to check all
conditions for waiting.
Also fix a potential deadlock caused by not checking if we're actually still
running before starting to wait.
When this is TRUE, we really have to produce output. This happens
in live mixing mode when we have to output something for the current
time, no matter if we have enough input or not.
This removes the uses of GAsyncQueue and replaces it with explicit
GMutex, GCond and wakeup count which is used for the non-live case.
For live pipelines, the aggregator waits on the clock until either
data arrives on all sink pads or the expected output buffer time
arrives plus the timeout/latency at which time, the subclass
produces a buffer.
https://bugzilla.gnome.org/show_bug.cgi?id=741146
Otherwise the caps of the pad might change while the subclass still works with
a buffer of the old caps, assuming the the current pad caps apply to that
buffer. Which then leads to crashes and other nice effects.
https://bugzilla.gnome.org/show_bug.cgi?id=740376
Audiomixer blocksize, cant be 0, hence adjusting the minimum value to 1
timeout value of aggregator is defined with MAX of MAXINT64,
but it cannot cross G_MAXLONG * GST_SECOND - 1
Hence changed the max value of the same
https://bugzilla.gnome.org/show_bug.cgi?id=738845
Determines the amount of time that a pad will wait for a buffer before
being marked unresponsive.
Network sources may fail to produce buffers for an extended period of time,
currently causing the pipeline to stall possibly indefinitely, waiting for
these buffers to appear.
Subclasses should render unresponsive pads with either silence (audio), the
last (video) frame or what makes the most sense in the given context.
The previous implementation kept accumulating GSources,
slowing down the iteration and leaking memory.
Instead of trying to fix the main context flushing, replace
it with a GAsyncQueue which is simple to flush and has
less overhead.
https://bugzilla.gnome.org/show_bug.cgi?id=736782
Without a lock that is taken in FLUSH_START we had a rare race where we
end up aggregating a buffer that was before the whole FLUSH_START/STOP
dance. That could lead to very wrong behaviour in subclasses.
Avoiding to be in an inconsistent state where we do not have
actual negotiate caps set as srccaps and leading to point where we
try to unref ->srccaps when they have already been set to NULL.
https://bugzilla.gnome.org/show_bug.cgi?id=735042
Along with the required mandatory dependent events.
Some elements need to perform an allocation query inside
::negotiated_caps(). Without the caps event being sent prior,
downstream elements will be unable to answer and will return
an error.
https://bugzilla.gnome.org/show_bug.cgi?id=732662
The gst_ptp_clock_new() does not actually require a name. However, for
example the rtpjitterbuffer may create a clock without a name, fail, and
fall back to not using the PTP clock.
https://bugzilla.gnome.org/show_bug.cgi?id=791034
Fixes in user code:
undefined reference to `gst_harness_add_element_sink_pad'
Also reorder harness function list to be strictly in alphabetical order and
double check the list with:
awk '{ if ($1 !~ /#define/) if ($2 ~ /gst_harness_/) { print $2 }; if ($3 ~ /gst_harness_/) { print $3} }' libs/gst/check/gstharness.h | sort
There were a few errors:
* The plugin scanner now accepts executable path as an argument.
In case it is NULL, argc == 2
* We find the executable path in init_pre instead of gst_init,
allowing this to work when gst is initialized through the
option group (eg gst-inspect)
* There was a semi-colon missing in the __APPLE__ #ifdef
When a plugin declares a dependency using this flag, all the
relative paths are considered to be relative to the path of
the main executable.
We try to determine the path of the executable portably,
with implementations provided for Linux, Windows and Mac.
If retrieval of the path fails, we will not detect changes.
In order for the main executable path to be the same when
scanning a plugin in a child process, a new variable is
exposed in gst_private.h, _gst_executable_path
https://bugzilla.gnome.org/show_bug.cgi?id=788152
We would constantly re-post the taglist because
posted_avg_rate only gets set to avg_bitrate if
parse->priv->post_avg_bitrate is true, so if it's
false the posted rate will always differ from the
current average rate and we'd queue an update,
which leads to us spamming downstream and the
application with taglist updates.
Fix this by only queuing an update if the average
rate will actually be posted.
These taglists updates could cause expensive
operations on the application side, e.g. in Totem.
https://bugzilla.gnome.org/show_bug.cgi?id=786561
Need to define CK_DLL_EXP to extern as well in libcompat.h
which gets included before the internal-check.h where the
other fallback definition for CK_DLL_EXP is.
duplicate symbol _check_minor_version in:
libcheckinternal.a(libcheckinternal_la-check.o)
libcheckinternal.a(libcheckinternal_la-check_log.o)
Have to modify libcheck header a bit to avoid warnings
about duplicate 'extern extern'.
Also needs some additions to the libcheck meson.build file
to define CK_EXP_DLL when building the static libcheck.
buffer is not unreferened if preroll failed
:Detailed Notes:
- Problem : video freeze when switching from pause to 1/2-FF repeatedly
- RootCause : buffer leaks in basesink
- Solution : unref the buffer if prerolled failed
:Testing Preformed:
How to Test :
pause -> 1/2 FF -> resume -> pause -> 1/2 FF ...
https://bugzilla.gnome.org/show_bug.cgi?id=784932
Holding this lock on live source prevents the source from changing
the caps in ::create() without risking a deadlock. This has consequences
as the LIVE_LOCK was replacing the STREAM_LOCK in many situation. As a
side effect:
- We no longer need to unlock when doing play/pause as the LIVE_LOCK
isn't held. We then let the create() call finish, but will block if
the state have changed meanwhile. This has the benefit that
wait_preroll() calls in subclass is no longer needed.
- We no longer need to change the state to unlock, simplifying the
set_flushing() interface
- We need different handling for EOS depending if we are in push or pull
mode.
This patch also document the locking of each private class member and
the locking order.
https://bugzilla.gnome.org/show_bug.cgi?id=783301
This is something bindings can't handle and it causes leaks. Instead
move the ref_sink() to the explicit, new() constructors.
This means that abstract classes, and anything that can have subclasses,
will have to do ref_sink() in their new() function now. Specifically
this affects GstClock and GstControlSource.
https://bugzilla.gnome.org/show_bug.cgi?id=743062
gst_harness_new_parse() returns without any error even if it doesn't
find the specified element. Then a succeeding call to
gst_harness_set_sink_caps_str() causes an error like this:
Unexpected critical/warning: gst_pad_push_event: assertion 'GST_IS_PAD (pad)' failed
This is a bit cryptic and doesn't give users any clue what was going
on.
gst_harness_new_parse() calls gst_harness_add_parse() with a newly
created empty harness and the given pipeline description string, but
gst_harness_add_parse() does not have a way to propagate the error
back to the caller. Since the function, gst_harness_add_parse(), is a
public API, it's not a good idea to change its signature. This patch,
instead, makes the function to g_error() when it discovers any error.
With this change the same error prints:
** (myelement-test:25345): ERROR **: Unable to create pipeline 'bin.( myelement )': no element "myelement"
The current implementation of gst_parse_launch_full() doesn't return
partially constructed pipeline when GST_PARSE_FLAG_FATAL_ERRORS is
specified, however, this patch also adds a check for it.
https://bugzilla.gnome.org/show_bug.cgi?id=781958
An untested pointer segfaulted in webkit while playing video
on imx6 sabrelite. It turned out that the imx plugin didn't
implement the meta transform function.
The following GST_DEBUG trace was visible:
gstbasetransform.c:1779:foreach_metadata:<conv2> copy metadata
GstImxVpuBufferMetaAPI
Thread 26 vqueue:src received signal SIGSEGV, Segmentation fault.
(gdb) bt
0x00000000 in ?? ()
0x73f8d7d8 in foreach_metadata (inbuf=0xc9b020, meta=0x474b2490,
user_data=<optimized out>) at gstbasetransform.c:1781
0x73eb3ea8 in gst_buffer_foreach_meta (buffer=buffer@entry=0xc9b020,
func=0x73f8d705 <foreach_metadata>,
user_data=user_data@entry=0x474b24d4)
at gstbuffer.c:2234
https://bugzilla.gnome.org/show_bug.cgi?id=782050
This unbalanced closing parenthesis is leftover from the commit
8b739d91e7. It used to wrap the caps but we don't seem to do that in
the current code.
So, just remove it. No functionality has been changed.
https://bugzilla.gnome.org/show_bug.cgi?id=781484
Use g_object_new() instead which nowadays has a shortcut for the
no-properties check. It still does an extra GType check in the
function guard, but there's a pending patch to remove that
and it's hardly going to be a performance issue in practice,
even less so on a system that's compiled without run-time checks.
Alternative would be to move to the new g_object_new_properties()
with a fallback define for older glib versions, but it makes the
code look more unwieldy and doesn't seem worth it.
Fixes deprecation warnings when building against newer GLib versions.
https://bugzilla.gnome.org/show_bug.cgi?id=780903
This patch reorganize the bash completion scripts in order to install
the binary helper (gst-completion-helper) in libexec path rather then
share folder. Most Linux hierarchy compliance requires that no binary
executable are placed in share. We also cleanup the unused .pc entries
and remove copy pasted parts of the script. Note that other project
including the common helper, should now use $_GST_HELPER to read
the binary executable gst-completion-helper. This helper is not longer
version, as it is placed in a versionned subfolder
(libexec/gstreamer.10) just like the other helpers (scanner and ptp).
If parsing returns a non-OK flow return in the middle
of processing an input buffer, don't overwrite that
if a later return is OK again - the subclass might
return not-linked in the middle, and then discard
subsequent data without pushing while returning OK.
A later success doesn't invalidate the earlier failure,
but we should continue processing after not-linked, so
as to keep parse state consistent.
https://bugzilla.gnome.org/show_bug.cgi?id=779831
We would add the offset a second time in _scan_for_start_code()
when we found a result, but it's already been added to the data
pointer at the beginning of _masked_scan_uint32_peek(), so the
peeked value would be wrong if the initial offset was >0, and
we would potentially read memory out-of-bounds.
Add unit test for all of this.
https://bugzilla.gnome.org/show_bug.cgi?id=778365
Otherwise when seeking/looping to the start when reaching the end,
the sink waits for the duration of the stream. So the user hears
nothing for the duration of the stream before it actually loop again.
See example attached to the bug for that.
Existing test:
gst-plugins-good/tests/icles/test-segment-seeks foo.flac
Without the patch the user hears a crack/cut at each seek.
https://bugzilla.gnome.org/show_bug.cgi?id=777780
New API functions to filter log messages before they are processed by
GstCheck. This can be used to discard specific messages that are
accepted by the test or to add callbacks that test specific messages.
Default bevavior when no callback is given to a filter is to discard the
message, because it does not makes sense to have a filter with no
callback which does not discard; that would be a noop.
Discarded messages will in addition to bypass the GstCheck handling also
return to GLib that the message is not fatal if it occurs.
https://bugzilla.gnome.org/show_bug.cgi?id=773091
When malloc is not available, this will set #define malloc rpl_malloc
which is implemented only inside libcheck, and not everything will link
to libcheck.
We don't really need to care too much about how malloc is implemented
and we don't care about platforms that don't implement malloc.
This brings us up-to-speed with the latest compatibility code from upstream
check git. For completeness, we do all the checks that upstream check does, but
we skip the snprintf/vsnprintf code because it's not straightforward (involves
running code and that is bad for cross-compilation) and not necessary for the
platforms we support anyway.
If someone really wants this, they can uncomment this and copy the relevant
checks from the check git repository.
https://bugzilla.gnome.org/show_bug.cgi?id=775870
Makes it clearer which files are actually used in libcheck and which are used
for cross-platform compatibility. This is going to be especially useful when we
add all the libcompat fallback code that upstream libcheck has which will add
about 6 new files.
https://bugzilla.gnome.org/show_bug.cgi?id=775870
Upstream seems to have stopped doing releases, but we need to update for better
Windows and Visual Studio support.
This patch only updates the libcheck sources and ignores the compatibility
sources for now.
https://bugzilla.gnome.org/show_bug.cgi?id=775870
This was totally non-obvious, the kind of big problem is that subclasses must
be able to unblock their streaming thread and continue exactly where they left off
on unpause!
https://bugzilla.gnome.org/show_bug.cgi?id=773912
Allows proxying the control interface from one property on one GstObject
to another property (of the same type) in another GstObject.
E.g. in a parent-child relationship, one may need to
gst_object_sync_values() on the child and have a binding (set elsewhere)
on the parent update the value.
Note: that this doesn't solve GObject property forwarding and must be
taken care of by the implementation manually or using GBinding.
https://bugzilla.gnome.org/show_bug.cgi?id=774657
It might've failed just because of flushing or other things, and we
should retry again on the next possibility if something ever calls in
here again.
https://bugzilla.gnome.org/show_bug.cgi?id=774623
Check the correct segment format value.
parse->segment.format is the format we're outputting in,
not the upstream format. Use parse->priv->upstream_format instead,
and make sure it's set in pull mode.
If the parser is not parsing a raw elementary stream, restrict
the position, duration and conversion query replies to
things we can sensibly answer about - especially don't do
random conversions to/from bytes.
This is cosmetic as 'late' should never be set during preroll (in pause).
Though code may evolve in the future, so this is good for preventing
potential bugs.
https://bugzilla.gnome.org/show_bug.cgi?id=772468
When the first buffer arrives, we endup calling:
->prepare()
->prepare()
->preroll()
->render()
This will likely confuse any element using this method. With this patch,
we ensure the preroll take place before the first render prepare() is
called. This will result in:
->prepare()
->preroll()
->prepare()
->render()
https://bugzilla.gnome.org/show_bug.cgi?id=772468
Hurd also defines __MACH__, but it does not have mach_absolute_time. Use
the more strict __APPLE__ instead.
Has also been sent upstream: https://github.com/libcheck/check/pull/65
This reverts commit 2e278aeb71.
Some parsers, specifically audio parsers, assume to get all remaining
data on EOS and just pass them onwards. While the idea here is correct,
we will probably need a property for this on baseparse for parsers to
opt-in.
https://bugzilla.gnome.org/show_bug.cgi?id=773666
Implement handling in basesink to not unconditionally discard
out-of-segment buffers and expose it as a new property on fakesink
(not unconditionally in all basesink based sinks).
The property defaults to FALSE.
https://bugzilla.gnome.org/show_bug.cgi?id=765734
baseparse would pass whatever is left in the adapter to the
subclass when draining, even if it's less than the minimum
frame size required. This is bogus, baseparse should just
discard that data then. The original intention of that code
seems to have been that if we have more data available than
the minimum required we should pass all of the data available
and not just the minimum required, which does make sense, so
we'll continue to do that in the case that more data is available.
Fixes assertions in rawvideoparse on EOS after not-negotiated with
fakesrc sizetype=random ! queue ! rawvideoparse format=rgb ! appsink caps=video/x-raw,format=I420
https://bugzilla.gnome.org/show_bug.cgi?id=773666
The durations of the buffers are (usually) assuming that no frames are being
dropped and are just the durations coming from the stream. However if we do
trickmodes, frames are being dropped regularly especially if only key units
are supposed to be played.
Fixes completely bogus QoS proportion values in the above case.
https://github.com/mesonbuild/meson
With contributions from:
Tim-Philipp Müller <tim@centricular.com>
Mathieu Duponchelle <mathieu.duponchelle@opencreed.com>
Jussi Pakkanen <jpakkane@gmail.com> (original port)
Highlights of the features provided are:
* Faster builds on Linux (~40-50% faster)
* The ability to build with MSVC on Windows
* Generate Visual Studio project files
* Generate XCode project files
* Much faster builds on Windows (on-par with Linux)
* Seriously fast configure and building on embedded
... and many more. For more details see:
http://blog.nirbheek.in/2016/05/gstreamer-and-meson-new-hope.htmlhttp://blog.nirbheek.in/2016/07/building-and-developing-gstreamer-using.html
Building with Meson should work on both Linux and Windows, but may
need a few more tweaks on other operating systems.
If segment.stop was given, and the subclass provides a size that might be
smaller than segment.stop and also smaller than the actual size, we would
already stop there.
Instead try reading up to segment.stop, the goal is to ignore the (possibly
inaccurate) size the subclass gives and finish until segment.stop or when the
subclass tells us to stop.
Waiting before posting calculated bitrates seems to be the
intent of the code, so avoid adding them to the tag list
pushed with the first frame.
When the threshold is reached, gst_base_parse_update_bitrates
sets tags_changed, so this posts the calculated ones right
that moment.
This prevents an insane average calculated from just the
first (key) frame from getting posted.
https://bugzilla.gnome.org/show_bug.cgi?id=768439
There must be a SEGMENT event before the GAP event, and SEGMENT events must
come after any CAPS event. We however did not produce any CAPS yet, so we need
to ensure to insert the CAPS event before the SEGMENT event into the pending
events list.
https://bugzilla.gnome.org/show_bug.cgi?id=766970
If we were in PAUSED, the current clock time and base time don't have much to
do with the running time anymore as the clock might have advanced while we
were PAUSED. The system clock does that for example, audio clocks often don't.
Updating the start time in PAUSED will cause a) the wrong position to be
reported, b) step events to step not just the requested amount but the amount
of time we spent in PAUSED. The start time should only ever be updated when
going from PLAYING to PAUSED to remember the current running time (to be able
to compensate later when going to PLAYING for the clock time advancing while
PAUSED), not when we are already in PAUSED.
Based on a patch by Kishore Arepalli <kishore.arepalli@gmail.com>
The updating of the start time when the state is lost was added in commit
ba943a82c0 to fix the position reporting when
the state is lost. This still works correctly after this change.
https://bugzilla.gnome.org/show_bug.cgi?id=739289
We don't do calculations with different units (buffer offsets and bytes)
anymore but have functions for:
1) getting the number of bytes since the last discont
2) getting the offset (and pts/dts) at the last discont
and the previously added function to get the last offset and its distance from
the current adapter position.
https://bugzilla.gnome.org/show_bug.cgi?id=766647
API: gst_buffer_prev_offset
API: gst_buffer_get_offset_from_discont
The gst_buffer_get_offset_from_discont() method allows retrieving the current
offset based on the GST_BUFFER_OFFSET of the buffers that were pushed in.
The offset will be set initially by the GST_BUFFER_OFFSET of
DISCONT buffers, and then incremented by the sizes of the following
buffers.
The gst_buffer_prev_offset() method allows retrievent the previous
GST_BUFFER_OFFSET regardless of flags. It works in the same way as
the other gst_buffer_prev_*() methods.
https://bugzilla.gnome.org/show_bug.cgi?id=766647
The subclass should do that already, but just in case do it ourselves too as a
fallback. Without this, e.g. playbin will just wait forever if this fails
because it is triggered as part of an ASYNC state change.
POSIX standards requires strsignal() to return a pointer to a char,
not a const pointer to a char. [1] On uClibc, and possibly other
libc's, that do not HAVE_DECL_STRSIGNAL, libcompat.h declares
const char *strsignal (int sig) which causes a type error.
[1] man 3 strsignal
https://bugzilla.gnome.org/show_bug.cgi?id=763567
To allow the GstTestClock to be used as a GstSystemClock, it is
useful to implement the clock-type property that GstSystemClock
provides. This allows GstTestClock to be used as the system clock
with code that expects a GstSystemClock.
https://bugzilla.gnome.org/show_bug.cgi?id=762147
Otherwise PTS and DTS will come out of sync if upstream continues to provide
PTS and not DTS, and we have to skip some data from the stream or PTS are not
exactly increasing with the duration of each packet.
https://bugzilla.gnome.org/show_bug.cgi?id=765260
gsttypefindhelper.c:485: Warning: GstBase: invalid "transfer" annotation for gsize: only valid for array, struct, union, boxed, object and interface types
If we don't store the value in prev_dts, we would over and over again
initialize the DTS from the last known upstream PTS. If upstream only provides
PTS every now and then, then this causes DTS to be rather static.
For example in adaptive streaming scenarios this means that all buffers in a
fragment will have exactly the same DTS while the PTS is properly updated. As
our queues are now preferring to do buffer fill level calculations on DTS,
this is causing huge problems there.
See https://bugzilla.gnome.org/show_bug.cgi?id=691481#c27 where this part of
the code was introduced.
https://bugzilla.gnome.org/show_bug.cgi?id=765096
It is calling do_sync(), which requires the STREAM_LOCK and PREROLL_LOCK to be
taken. The STREAM_LOCK is already taken in all callers, the PREROLL_LOCK not.
https://bugzilla.gnome.org/show_bug.cgi?id=764939
This is the best guess we can make if such a buffer reached the collect
pad. This is uncommon, we do expect parsers to have tried and fixed that
if possible (or needed).
https://bugzilla.gnome.org/show_bug.cgi?id=762207
POSIX standards requires strsignal() to return a pointer to a char,
not a const pointer to a char. [1] On uClibc, and possibly other
libc's, that do not HAVE_DECL_STRSIGNAL, libcompat.h declares
const char *strsignal (int sig) which causes a type error.
[1] man 3 strsignal
https://bugzilla.gnome.org/show_bug.cgi?id=763567
Many parsers are storing tags only in pre_push_frame(), if we wouldn't check
afterwards we would push buffers before those tags and a lot of code assumes that
tags are available before preroll.
https://bugzilla.gnome.org/show_bug.cgi?id=763553
Similar to the stress test functions for buffers that has a callback to
create the buffer to be pushed, it's useful to have functions that use a
callback to create the event to be pushed.
API: gst_harness_stress_push_event_with_cb_start()
API: gst_harness_stress_push_event_with_cb_start_full()
API: gst_harness_stress_send_upstream_event_with_cb_start()
API: gst_harness_stress_push_upstream_event_with_cb_start_full()
https://bugzilla.gnome.org/show_bug.cgi?id=761932
Depending on when gst_harness_pull was called - before the buffer reached
gst_harness_chain or after we can get different behaviors of the test
with enabled blocking push mode. The fix makes the behavior always the
same. In pull function we get the buffer first, thus making sure
gst_harness_chain waits for the signal, and emitting the signal after.
https://bugzilla.gnome.org/show_bug.cgi?id=761931
Set the sink_forward_pad to NULL before tearing down sink_harness to
avoid that the harness tries to forward events/queries to it while it's
tearing down.
https://bugzilla.gnome.org/show_bug.cgi?id=761904
We have no guarantees about what flags are set on buffers we take
out of the GstAdapter. If we push out multiple buffers from the
first input buffer (which will have discont set), only the first
buffer we push out should be flagged as discont, not all of the
buffers produced from that first initial input buffer.
Fixes issue where the first few mp3 frames/seconds of data in push
mode were skipped or garbled in some cases, and the discont flags
would also trip up decoders which were getting drained/flushed for
every buffer. This was a regression introduced in 1.6 apparently.
