This is a custom mapping. There isn't much needed apart from that to store vp9
in mpeg-ts since the bitstream is self contained.
Since there are no official specification we don't want people to be mistaken in
believing that. Therefore that mapping is only used in the muxer if the (new)
property `enable-custom-mappings` is set to TRUE.
* The MPEG-TS Stream Type is Private Data (0x6) with the registration descriptor
set to `VP09`.
* The Access Unit are VP9 frames stored in PES packets
* As there is no emulation prevention byte in VP9 elementary stream, the can be
misdetection of PES start code. To avoid this, the start of a PES packet must
be signalled using the Payload Unit Start Indicator in the transport packet
header
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7707>
When encoding an image to mpeg2 video, with something like:
gst-launch-1.0 encodebin name=e profile=mpegpsmux:video/mpeg,mpegversion=2,systemstream=false ! \
filesink location=sample.mpg filesrc num-buffers=1 blocksize=$(stat -c%s sample.png) \
location=sample/dts.png ! pngdec ! e.
The only frame's type is set to an invalid value 0
The consequence is that mpegvideoparse sets the delta unit flag on the buffer because
it is not an I frame, then decodebin3 drops this only frame because the delta
unit flag is set and the decoder receives eos before it was able to receive any
encoded data
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7832>
GstTensor contained two fields (data, dims) that were dynamicallay allocated. For
data it's for a GstBuffer and we have pool for efficient memory management. For
dims it's a small array to store the dimension of the tensor. The dims field
can be allocated inplace by moving it at the end of the structure. This will
allow a better memory management when GstTensor is stored in an analytics meta
which will take advantage of the _clear interface for re-use.
- New api to allocate and free GstTensor
To continue to support use-cases where GstTensor is not stored in an
analytics-meta we provide gst_tensor_alloc, gst_tensor_alloc_n and
gst_tensor_free that will facilitate memory management.
- Make GstTensor a boxed type
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6000>
Only in LTC mode we introduce additional latency that is depending on only on a
property and not on the framerate, so waiting for the framerate is not necessary.
In all other modes no latency is introduced at all and the latency query can
simply be proxied.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7831>
The calculated position was off. I'm not sure of the exact cause;
possibly because we're in AU-aligned byte-stream mode, which means
`transform` is true.
Replacing the math that calculates the NALU positions with code more
similar to what is already in use for `idr_pos` seems to have fixed it.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7318>
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>
