Call gst_aggregator_selected_samples() after identifying the
caption buffers that will be added as a meta on the next video
buffer.
Implement GstAggregator.peek_next_sample.
Add an example that demonstrates usage of the new API in
combination with the existing buffer-consumed signal.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1390>
A pipeline like this:
closedcaption/x-cea-708,format=cdp,framerate=30000/1001 ! ccconverter ! closedcaption/x-cea-708,format=cc_data
would produce a critical/assert:
GStreamer-CRITICAL **: 14:21:11.509: gst_util_fraction_multiply: assertion 'a_d != 0' failed
because there would be no framerate field on ccconverter's output.
Fixed by always fixating a framerate if the input has a framerate.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1393>
Instead of storing the raw cc_data, store the 2 cea608 fields individually
as well as the ccp data.
Simply copying the input cc_data to the output cc_data violates a number of
requirements in the cea708 specification. The most prominent being, that
cea608 triples must be placed at the beginning of each cdp.
We also need to comply with the framerate-dpendent limits for both the
cea608 and the ccp data which may involve splitting or merging some
cea608 data but not ccp data or vice versa.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1116>
This will stop stripping four bytes start code. This was fixed and broken
again as it was causing the a timestamp shift. We now call
gst_base_parse_set_ts_at_offset() with the offset of the first NAL to ensure
that fixing a moderatly broken input stream won't affect the timestamps. We
also fixes the unit test, removing a comment about the stripping behaviour not
being correct.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1251>
TSN streams are expected to send packets to the network in a well
defined "pace", which is arbitrarily defined for each stream. This pace
is defined by the "measurement interval" property of a stream.
When the AVTP CVF payloader element - avtpcvfpay - fragments a video
frame that is too big to be sent to the network, it currently defines
that all fragments should be transmitted at the same time (via DTS
property of GstBuffers generated, as sink will use those to time the
transmission of the AVTPDU). This doesn't comply with stream definition,
which also has a limit on how many packets can be sent on a given
measurement interval.
This patch solves that by spreading in time the DTS of the GstBuffers
containing the AVTPDUs. Two new properties, "measurement-interval" and
"max-interval-frames", added to avptcvfpay element so that it knows
stream measurement interval and how many AVTPDUs it can send on any of
them. More details on the method used to proper spread DTS/PTS according
to measurement interval can be found in a code commentary inside this patch.
Tests also added for the new property and behaviour.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1004>
This adds tests to validate whether the avtpcrfsync element applies the
adjustment correctly.
Also, the infrastructure to include additional source files while compiling
is added. This change is exactly the same as the one in gst-plugins-good.
We can't be sure about the reference count if the muxer is currently
running, which can happen in the test_reappearing_pad test. An
additional reference might temporarily be owned by the srcpad task of
tsmux while iterating over the pads.
Until now, any streams in tsmux had to be present when the element
started its first buffer. Now they can appear at any point during the
stream, or even disappear and reappear later using the same PID.
Some raw h264 encoded files trigger the assignment of wrong PTS to buffers
when some SEI data is provided. This change prevents it to happen.
Also ensure this behavior is being tested.
