The 'new-pref' property sets the preference to use the new (next)
instead of the old (previous) buffer. The default is set to 0.5 to get
a similar behaviour as before the change.
Value 0.0 makes sure that only frames are shown where it's known that
the frame content is visible at that time, always show the old frame
until the new frame timestamp is reached.
Then, if the next buffer replaces the previous buffer the new buffer
is pushed as often as possible until PTS is reached. Before the new
buffer was only pushed once the new next buffer arrived.
Use GstClockTimeDiff because it's known that the current buffer time
is inside the time interval of previous buffer and next buffer the
calculation can be done with building absolute values. Special macros
are not needed here.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8579>
`parsebin` is potentially added by a `typefind` callback.
That `typefind` was activated by a `READY_TO_PAUSED` state change on `urisourcebin`
We want to ensure that it is the "setup_parsebin_for_slot" method that activates
the underlying `parsebin`, and not the external state-change.
Otherwise we would risk a potential deadlock where elements activating in
`parsebin`, and which would cause the upstream `typefind` to switch scheduling
mode, would not be able to acquire the STREAM_LOCK of the `typefind` task.
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/4225
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8511>
The property is defined as:
> The difference between incoming timestamp and next timestamp must exceed
> the given value for audiorate to add or drop samples.
so if the gap duration < tolerance, we should not act.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8498>
AV1 specification [1] define 2 AV1 raw bitstreams storage
formats without containers:
- OBU in chapiter 5.2
- Annexb in chapiter 11.
Implement a detection function for the both cases mostly
by testing OBU forbidden, type, has_size_field and reserved bits.
For annexb case testing if temporal unit size, frame unit size and
obu length are valid. If they are check that the first OBU is
a temporal delimiter.
[1] https://aomediacodec.github.io/av1-spec/av1-spec.pdf
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8502>
The videotestsrc is not build if the adder plugin is disabled. This is a
copy/paste error introduced in Commit 3de86b2b9725 ("docs: port plugins to
explicit sources").
Fix the check to actually test the 'videotestsrc' option.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8399>
Some pixel aspect ratios found in dash streams have very large numerators and
denominators (while being close to 1:1). These values can cause integer overflow
during multiplication, leading to negotiation failures.
Add fallback path using gdouble when integer multiplication would overflow,
trading some precision for reliability instead of failing outright.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8262>
Fix-up after commit e56b78417. Unclear how that could
ever have worked. Code only built because basically
everything was disabled due to the missing config.h
include.
- Include config.h so that HAVE_* defines are available.
- Fix missing variable declarations that somehow disappeared
when moving the stats gathering code block into a separate
function in a new file.
- Fix structure variable modified to match name in new function.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8211>
The reason why the STATE lock was taken was to avoid issues where we would be
adding (and activating) elements at the same time as urisourcebin would be
brought down to READY. That would cause those new elements to potentially return
ERRORS because of not-negotiated/flushing-pads
But that creates a really bad deadlock (state lock is taken to deactivate the
streaming thread which .. is currently grabbing the state lock).
Instead, we can just ignore the warning/error messages that might occur when
shutting down.
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/4075
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/8162>
There is no reason why we should mandate people to "at least" use the static
sink pad. This caused issues, like mandating that it should always have valid
content linked to it (problematic in case of upstream stream changes).
Instead we only use it if it's actually linked to, in which case it gets added
to the list of inputs.
This actually simplifies the code too.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7949>
When checking whether a no-longer used output could be re-used for another slot,
we only want to do that for streams which are not still used.
Otherwise we end up potentially re-assigning a demuxer stream to a completely
different one
Furthermore, if we *are* re-using an output slot, indicate what the replacement
GstStream will be so slot matching can work properly (which can happen in the
case of demuxers which add/remove all pads even if only a single one changed)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7949>
Convert segment to TIME format immediately instead of waiting for
_chain() to be called. This fixes converted segment never being pushed
downstream.
Fix the convert function that was copying some fields in the wrong
direction. Add fast copy if segment is already in TIME format.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7977>
We only want to switch to a selection of an output collection if all streams are
present.
This was previously only done in one place (when triggering by new incoming
streams) but not when triggered by user/application.
Avoid this by moving the check to handle_stream_switch()
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7941>
