The diff between compared timestamps might be outside the gint range
resulting in wrong sorting results. This patch corrects that by
comparing the timestamps and then returning -1, 0 or 1 depending on the
result.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7726>
The wraparound handling code assumes that the PCR gets updated regularly for
being able to detect wraparounds. With ignore-pcr=true that was not the case and
it stayed initialized at 1h forever.
To avoid this problem, update the fake PCR whenever the PTS advanced by more
than 5s, and also detect wraparounds in these fake PCRs.
Problem can be reproduced with
$ gst-launch-1.0 videotestsrc pattern=black ! video/x-raw,framerate=1/5 ! \
x264enc speed-preset=ultrafast tune=zerolatency ! mpegtsmux ! \
tsdemux ignore-pcr=true ! fakesink
which restarts timestamps at 0 after around 26h30m.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7588>
There was an override to fake an IDR as soon as a SPS/PPS
is encountered, but that's not valid, at least an i-slice is needed.
Amend the visl result, as the output is slightly more correct, not
duplicating frame_num.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7330>
This improves the h264parse element to attach LCEVC enhancement data to buffers
using the new GstLcevcMeta API. This metadata will eventually be used downstream
by LCEVC decoders to enhance the RAW video frame.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7330>
This makes sure that if upstream has different latencies that we're still
outputting buffers with increasining timestamps across the different streams
unless buffers are arriving after the latency deadline.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7500>
"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>
