They can fail for various reasons.
For non-fatal cases (such as the dump feature of identiy and fakesink),
we just silently skip it.
For other cases post an error message.
https://bugzilla.gnome.org/show_bug.cgi?id=728326
Change the gst_tracer_record_new() api to take the parameters the make the
spec structure directly. This allows us to own the top-level structure and
also collect the args so that we can take ownership of the sub-structures.
https://bugzilla.gnome.org/show_bug.cgi?id=760821
The use-tags-bitrate property makes queue2 look at
tag events in the stream and extract a bitrate for the
stream to use when calculating a duration for buffers
that don't have one explicitly set.
This lets queue2 sensibly buffer to a time threshold
for any bytestream for which the general bitrate is known.
Only hide GstTracer and GstTracerRecord API behind GST_USE_UNSTABLE_API,
but don't spew any warnings, otherwise everyone has to define this
to avoid compiler warnings.
This reverts parts of commit 89ee5d948d.
We use this class to register tracer log entry metadata and build a log
template. With the log template we can serialize log data very efficiently.
This also simplifies the logging code, since that is now a simple varargs
function that is not exposing the implementation details.
Add docs for the new class and basic tests.
Remove the previous log handler.
Fixes#760267
segment.position is meant for internal usage only, but the various
GST_EVENT_SEGMENT creationg/parsing functions won't clear that field.
Use the appropriate segment boundary as an initial value instead
When synchronizing the output by time, there are some use-cases (like
allowing gapless playback downstream) where we want the unlinked streams
to stay slightly behind the linked streams.
The "unlinked-cache-time" property allows the user to specify by how
much time the unlinked streams should wait before pushing again.
Multiqueue should only be used to cope with:
* decoupling upstream and dowstream threading (i.e. having separate threads
for elementary streams).
* Ensuring individual queues have enough space to cope with upstream interleave
(distance in stream time between co-located samples). This is to guarantee
that we have enough room in each individual queues to provide new data in
each, without being blocked.
* Limit the queue sizes to that interleave distance (and an extra minimal
buffering size). This is to ensure we don't consume too much memory.
Based on that, multiqueue now continuously calculates the input interleave
(per incoming streaming thread). Based on that, it calculates a target
interleave (currently 1.5 x real_interleave + 250ms padding).
If the target interleave is greater than the current max_size.time, it will
update it accordingly (to allow enough margin to not block).
If the target interleave goes down by more than 50%, we re-adjust it once
we know we have gone past a safe distance (2 x current max_size.time).
This mode can only be used for incoming streams that are guaranteed to be
properly timestamped.
Furthermore, we ignore sparse streams when calculating interleave and maximum
size of queues.
For the simplest of use-cases (single stream), multiqueue acts as a single
queue with a time limit of 250ms.
If there are multiple inputs, but each come from a different streaming thread,
the maximum time limit will also end up being 250ms.
On regular files (more than one input stream from the same upstream streaming
thread), it can reduce the total memory used as much as 10x, ending up with
max_size.time around 500ms.
Due to the adaptive nature, it can also cope with changing interleave (which
can happen commonly on some files at startup/pre-roll time)
This will mean a much lower delay before a subtitles track changes take
effect. Also avoids excessive memory usage in many cases.
This will also consider sparse streams as (individually) never full, so
as to avoid blocking all playback due to one sparse stream.
https://bugzilla.gnome.org/show_bug.cgi?id=600648
