Theoretically the version number is incremented every time there's a new
section, but in a world of streaming we can't easily make that
assumption.
An example of a broken use case is when we're cat-ing two mpeg-ts files
together, which is equivalent of capturing a DVB stream while switching
channels. A set-top box would know that we switched the channels and
reset the demuxer, but in practice this might not happen.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2468>
Some closedcaption elements like sccenc except input buffers
to have timecode metas. The original use case is to serialize
closed captions extracted from a video stream, in that case
ccextractor copies the video time code metas to the closed
caption buffers, but no such mechanism exists when creating
a CC stream ex nihilo.
Remedy that by having timecodestamper accept closedcaption
input caps, as long as they have a framerate.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2490>
When reaching the end of non-frame wrapping track in pull mode, we want to force
the switch to the next non-eos pad. This is similar to when we exceed the
maximum drift.
Fixes issues on EOS where not everything would be drained out and stray errors
would pop out.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2434>
... for user to be able to set the number of required samples.
For instance, our default value is 240 samples
(about 5ms latency in case that sample rate is 48000), which might
be larger than actual buffer size of audio capture device.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2307>
Some cut VP9 streams begin with a non key frame. The current code
just bail out the parse_process_frame() if not a key frame. Because
of this, we do not set the valid caps before we push the data of the
first frame(even this first frame will be discarded by the downstream
decoder because it is not a key frame).
The pipeline such as:
gst-launch-1.0 filesrc location=some.ivf ! ivfparse ! vp9parse !
vavp9dec ! fakesink
will get a negotiation error and the pipeline can not continue. The
correct behaviour should be: the decoder discard the first frame and
continue to decode later frames successfully.
So, when the parse does not have valid stream info(should be the first
frame case), we should continue and report caps.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2427>
* If we have an index table for non-framed essence, we can handle it
* The demuxer has a state which indicates whether it will next fetch a KLV or
data contained *within* a KLV.
* The position on Essence Tracks always correspond to the next entry to fetch,
demuxer offset will be skipped accordingly whenever we switch between
partitions (in case of resyncs). A copy of the main clip/custom KLV for that
partition is kept to track the position within the essence of that partition.
* For clip/custom-wrapped raw audio, if the edit rate is too small (and would
cause plenty of tiny buffers to be outputted), specify a minimum number of edit
units per buffer.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2371>
* For pull-based, this avoids pulling content if it's not needed (ex: skipping filler
packet, not downloading the content if we only need to know if/where an essence
packet is, etc...). Allows reducing i/o usage to the minimum.
* This also allows doing sub-klv position tracking, and opens the way for
non-frame-wrapping handling
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2371>
In order to figure out the exact start position (backed by a keyframe) accross
all tracks, we first figure out the backing keyframe position, and *then* seek
to that position.
Avoids ending up in situations where we would properly seek to the backing
keyframe on video ... but not on the audio streams (they would have been set to
the original non-keyframe position). Fixes key-unit seeking.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2371>
The picture essence coding matching was wrong. Use the proper "base" MXFUL for
video mpeg compression for matching.
Also handle the case where some old files would put the essence container label
in the essence coding field
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2371>
* Streamline offset <=> entry handling. Historically the demuxer didn't support
information from index tables and stored the discovered information in an array
per track. When index table support was added, a parallel system was setup for
that relationship. This commit unifies this into one system with the
`find_edit_entry()` and `find_entry_for_offset()` functions.
* By extension, per-track offset entry tables are only created/used if no index
table is present for those tracks.
* Use index table information as-is. The index table system from MXF is quite
complex and there are various ways to use the information contained
within. Instead of converting that information we store the data from the tables
as-is and extract the needed information when needed.
* Handle index tables without entries (i.e. all content package units are of the
same size).
* Allow collecting index table segments as we go instead of only once if a
random-index-pack is present. This also improves support of some files in
push-mode.
* When searching for keyframe entries, use the keyframe_offset if
present (speeds up searching).
* For interleaved content (i.e. several tracks in the sample essence container),
we use a system to be able to identify the position of each track in the delta
entries of index tables.
* Handle temporal offset only on tracks which *do* need it (as specified in the
delta entries of the index tables). If present, those offsets are stored in a
pre-processed table which allows computing PTS from DTS with a simple offset.
* Add a quirk for files which are known to be have wrongly stored temporal
offsets.
* Overall opens the way to handle more types of MXF files, especially those with
non-frame-wrapping.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2371>
This is similar to how the same issue was handled in qtdemux.
In order for the "DTS <= PTS" constraint to be respected, we calculate the
maximum temporal reordering that can happen (via index tables).
If there is a non-0 temporal reordering, we:
* Shift all outgoing PTS by that amount
* Shift segment for that stream by that amount
* Don't modify DTS (i.e. they might end up having negative running-time, before
the start of the segment)
Also ensure all entries have a valid PTS set, previously this wouldn't be set
for entries with a temporal offset of 0.
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/584
(and maybe a lot of other issues)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2371>
When the muxer is operating in VBR mode, it's kind of expected
for now that we might not put the PCR in exactly the right place,
because the muxer doesn't schedule packets that way. In that case
don't warn constantly about the PCR ending up a few ms off target.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2295>
This is a work-around to identify some main profile streams having
wrong profile_idc. There are some wrongly encoded main profile streams
which doesn't have any of the profile_idc values mentioned in Annex-A,
instead, general_profile_idc has been set as zero and the
general_profile_compatibility_flag[general_profile_idc] is TRUE.
Assuming them as MAIN profile for now.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2322>
Some elements will make use of the automatically generated names to
create new pads in future muxer instances, for example splitmuxsink.
Previously we would've created a pad with a random pid that would become
"sink_0", and then on a new muxer instance a pad "sink_0" and tsmux
would've then failed because 0 is not a valid PID.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2318>
Previously pads might have been requested already (e.g. in NULL state),
then reset was called (e.g. because changing state) and then a new pad
was requested. Resetting is re-creating the internal muxer object and as
such resetting the pid counter, so the next requested pad would get the
same pid as the first requested pad which then leads to collisions.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2317>
Some decoder implementations might drain out internal buffers and
reset its status on segment-done event. So, in case that
upstream stream-format is packetized but downstream supports only
byte-format, required codec-data might not be forwarded toward
downstream if such parameter set NAL units don't exist in inband
bitstream. Therefore, parse elements should re-send parameter set NAL
units like the case of flush event.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2334>
alphacombine element is a simple element that assumes buffers are always
paired, or at least that missing buffers are signalled with a GAP. The QoS
implementation in the GstVideoDecoder base class allow decoders dropping
frames independently and that could lead to stall in alphacombine.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2326>
This is a video specific sink used to test video CODEC conformance. This is similar
to a combination of filesink and testsink, but will skip over any type of
padding that GStreamer Video library introduces. This is needed in order to obtain the
correct checksum or raw yuv data.
This element currently support writing back non-padded raw I420 through the
location property and will calculate an MD5 and post it as an element message
of type conformance/checksum. More output format or checksum type could be
added in the future as needed.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2287>
Alpha combine works by appending the GstMemory for the alpha channel
to the GstBuffer containing I420, thereby pushing A420 on its src pad.
Add support for the same workflow for NV12, thereby producing the
recently introduced AV12 format (NV12 + Alpha).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2277>
