"adobe" in app14 marker seem not a null-terminted string. so, when
we use gst_byte_reader_get_string_utf8, more bytes will be read until
null. and "gst_byte_reader_get_uint8 (&reader, &transform)" will almost fail
to read transform
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7356>
librtmp allows for attaching arbitrary AMF objects to the end of the
connect packet, and this is commonly used for authenticating with
servers.
Add a new property, extra-connect-args, that mimics librtmp's behavior.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7054>
A previous fix, a275e1e029, is correct but was too
permissive since it treats all un-matched NAL units the same as AU delimiters
even though some other NAL unit types can be encountered in the processing loop.
The problem this can cause is that some hardware decoders experience bad
performance when handling FD units that precede the SPS.
This change restores the original behavior for FDs so that they're ignored until
the SPS is received and it preserves the codec conformance test gains that the
fix has achieved.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7166>
When configured in constant bitrate mode, the muxer computes timing information
using the configured bitrate and the byte counter (now = bytes sent / byterate).
When an application changes the bitrate in CBR mode during playback, the
relationship between bytes sent and bitrate is no longer valid so new timing
values will be off by the ratio of the old bitrate to the new bitrate.
Furthermore, it will upset the way that padding is generated.
pad_stream() works by trying to fit the byte counter to now * byterate.
The result is that when decreasing bitrate, the muxer stalls, waiting until the
byte counter is in agreement with now * byterate. Also, when increasing
bitrate, the padding will spike in volume until the byte counter fits with
now * byterate.
If the byte counter is scaled by the ratio of new bitrate / old bitrate when
adjusting bitrate, then padding is generated in a way that applications would
more likely expect.
One detail this change doesn't yet address is whether the next PCR will match up
optimally with the previous PCR right after the byte counter is scaled. In that
case, some correction may be necessary. Also, perhaps the user should be
prevented from changing from bitrate=0 to bitrate=nonzero during playback since
it's not straightforward how to scale the byte counter in that case.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7158>
When transforming from unknown alignment to frame or obu, the TU timestamp
was not properly transferred. Fix this by saving the TU DTS as the first
DTS seen within the the TU data, and the PTS as the last PTS seen in that
TU data. Finally, reset the TU timestamp after each TU have completed.
Fixes#1496
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6895>
If the muxer times out because of the latency deadline it can happen
that some pads have no caps yet. In that case skip creation of streams
for these pads and create updated section tables once the first buffer
arrives later.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6823>
This makes sure that for sparse streams (KLV, DVB subtitles, ...) the
muxer does not wait until the next buffer is available for them but
times out on the latency deadline and outputs data.
For non-live pipelines it will still be necessary for upstream to
correctly produce gap events for sparse streams.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6823>
There was an issue with this equality check, which was to figure out what to do
with PCR pids (whether they were part of the streams present or not) and whether
we ignore PCR or not.
Turns out ... we already took care of that further up in the function.
The length check can be simplified by just checking whether the length of
the *original* PMT and the new PMT are identical. Since we don't store "magic"
PCR streams in those, we can just use them as-is.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6713>
The goal of this code was, for programs which were updates (i.e. adding/removing
streams but not completely changing) to allow dynamic addition/removal of
streams without completely removing everything.
But this wasn't 100% tested and there are a bunch of issues which make it fail
in plenty of ways.
For now disable that feature and force the legacy "add all pads again and then
remove old ones" behaviour to make it switch.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6651>
For some cameras `gst_jpeg_parse_app0()` fails on a invalid segment.
While this is likely a driver or firmware bug that should be addressed
accordingly, it's not fatal and likely does not deserve a bus message on
every frame, flooding journals.
Turn down the volume of the warnings by turning them into object
warnings. If we conclude that in some cases we'd still want bus
warnings, they can be done more fine-grained in the
`gst_jpeg_parse_appX()` functions.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6490>
In the situation where playback starts from a keyframe before
the target playback segment, then the first buffers will be
outside the configured segment and gst_segment_to_stream_time()
will return GST_CLOCK_TIME_NONE unconditionally.
If drop-out-of-segment is false, the RTP buffers will not be
dropped, but will be sent witout ONVIF extension timestamps
and given GST_CLOCK_TIME_NONE timestamps on the receiver.
Instead, use gst_segment_to_stream_time_full() to extrapolate
stream time outside the segment so that such buffers still
get assigned their correct timestamps on the receiver.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6248>
Don't accidentally include the stuffing byte (if present)
into the bottom field size. It should only be included in the
total segment length.
Fixes problems with FFmpeg not rendering the subtitles
with a stuffing byte, giving a "Invalid object location!" error.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6250>
- Add the missing field parameter and put the output parameter at the
end.
- Use a switch to verify valid values instead of hard-to-follow range
checks.
- Don't consider bad values a programming error, just a regular failure.
- Set all data fields at the end so we can pass a pointer to an
uninitialized structure without GCC complaining.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5450>
Previously, the path lock was held even while issuing caps queries to
other elements. This can lead to deadlocks in more complex pipelines.
Avoid this by reworking gst_switch_bin_get_allowed_caps() to acquire
references to switchbin paths and then releasing the path lock.
Subsequent operations in that function then act on the acquired
references, thus eliminating the need for holding the path lock for
the entirety of that function.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4632>
The caps query specifies _all_ caps that the element can handle, not just
caps from the current path element. If for example a switchbin has two
paths, with one having an element that handles video/x-h264, and another
path whose element handles video/x-raw, and the second path is the
current path, then the existing code would report only video/x-raw as
supported. Fix this by report all allowed caps, even if there is a
current path defined.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4632>
The rationale is that a passthrough path (= one with no element) behaves
as if the switchbin's sink- and srcpad were one. In particular, internal
caps queries (needed for computing the allowed caps) then go to the peers
instead to path elements. Rework gst_switch_bin_get_allowed_caps () for
a clear handling of NULL path elements and for proper dataflow passthrough
and caps & accept-caps query handling.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4632>
The drop probe was present in early switchbin versions to implement paths
that drop dataflow. However, this feature turned out to be too problematic
and thus was removed. Some bits remained though. This commit removes those
bits and clarifies that in the current switchbin version, a NULL path
element instead means passthrough.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4632>
If the current segment has a configured stop point, detect
when when pad timestamps proceed past that point and mark
them as EOS. Otherwise, tsdemux continues streaming
the whole input downstream (unless something downstream detects
and returns EOS for us)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6023>
Use string parsing instead of pointer arithmetic, which makes the code
easier to understand and less error-prone. This has no functional
changes, and is preparation for the next commit, which extends the
header parsing.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5997>
videoconvertscale advertises `ANY` feature, but it supports it only
in passthrough. Our goal with autoconvert is to ensure that conversion
is possible with the elements that are being plugged so we avoid
plugging `videoconvertscale` if the memory type is not system memory.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/899>
Instead of letting all the elements that were added into the bin,
add them only when strictly needed and removed them when we stop using
them.
With previous refactoring we are keeping them in our own hashmap in
amy case so we can keep reusing the same elements as before.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/899>
We used to conside elements that were subclassses of another
element type as being the same (for example with videoconvertscale,
bother videoconvert and videoscale are subclasses of videoconvertscale
and that code was lost)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/899>
Post a bus message explaining that input buffers must
have timestamps and return GST_FLOW_ERROR, instead of
a confusing NOT-NEGOTIATED
Also remove an errant buffer unref in the error handling
that would lead to crashes after.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5935>
Add a finalize method and release locks and things in there, instead
of in the dispose method. Dispose may be called multiple times,
at any time, and should just safely release references to other
memory that might reference it back.
In this case, timecodestamper would later crash in the element
dispose method trying to take the freed mutex from
gst_timecodestamper_release_pad().
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5935>
- Fix skipsize on _update_backlog failure.
- Add robustness to AU completion detection by using AUD when present. If we've
received a AUD we overwrite the first VCL NAL detection when the result was
negative. VCL following AUD is the first VCL of next AU.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5862>
Instead of only supporting writing SPU data directly to YUV frames,
render the SPU data to an intermediate AYUV overlay buffer. The overlay
data is then blended to the video frame.
For the PGS format, the overlay buffer size is set to the size of the
Composition Window, and its position in the overlay composition is set
to the window position. The objects to render are now cropped when the
cropping flag is set.
For the Vobsub format, the overlay buffer size is set to the size of the
Display Area.
Once rendered, the overlay composition rectangle is now moved and scaled
to fit the video output size, to avoid clipping.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5827>
