(Initially discussed in
https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/issues/305)
The ticks waveform can be useful for audio synchronization diagnostics
and other cases where the time offset between waveforms is important.
However, in its current form, it is too limited, and has problems with
discontinuities, which result in severe artifacts when this waveform
is output by a DAC.
This patch fixes some discontinuities and considerably expand the ticks
waveform's flexibility. They also introduce the notion of a "marker tick";
every Nth tick can have a different amplitude (usually one that is larger
than the others). This is useful for combining frequent oscilloscope
triggering with large time offset detection. For example, without marker
ticks, the tick intervals must not be too small, otherwise the maximum time
offset that can be unambiguously detected is quite small (for example, if
the interval is 50ms, then no time offset larger than 25ms can be
unambiguously recognized). If the tick intervals are too far apart, then
no sudden changes can be clearly observed, since the oscilloscope is not
updated quickly enough. But with marker ticks, this is not an issue: If
there's for example a tick every 100 ms, then the oscilloscope can be
triggered every 100 ms. And, if every 20th tick is a marker tick, then
time offsets of up to 1 second can be discovered, even though the time
between ticks is 100 ms.
The patch also applies some minor cleanup to the audiotestsrc documentation.
Checking the address distance between given begin/end sequence
doesn't make sense. They are output params.
This is to fix weird failure of libs_rtp on Windows
Code in g_return_*() must not have side effects, as it
might be compiled out if -DG_DISABLE_CHECKS is used, in
which case we would read garbage off the stack.
Both versions are basically the same, but version 2.0 also allows
60000/1001 as framerate and allows to specify the field and line number
for each payload.
Put the major version into the caps so that elements can limit via caps
negotiation which versions they can support.
It breaks all the calculations. While it can make sense during
initialization, there's very little API that can be called with such
timecodes without ending up with wrong results.
The old API would only assert or return an invalid timecode, the new API
returns a boolean or NULL. We can't change the existing API
unfortunately but can at least deprecate it.
... instead of hardcoded ':', since G_SEARCHPATH_SEPARATOR_S
varies depending on OS (e.g., ':' for *nix and ';' for Windows).
Note that, when the seperator is not specified explicitly, Meson
will use ';' for Windows and ':' for *nix respectively.
audioconvert's passthrough status can no longer be determined
strictly from input / output caps equality, as a mix-matrix can
now be specified.
We now call gst_base_transform_set_passthrough dynamically, based
on the return from the new gst_audio_converter_is_passthrough()
API, which takes the mix matrix into account.
The presence of `key-mgmt` attribute will set the mikey appropriately.
We therefore don't need to check the return value (which will
be overwritten afterwards).
CEA608_IN_CEA708_RAW is the same format as CEA708_RAW. It's only
difference is that it must contain only CEA608 and a format like this
does not exist in practice. In practice every element that handles raw
cc_data triplets must check each triplet for their actual content and
handle them accordingly.
For CC-only streams a parser could signal the existence of CEA608 and/or
CEA708 inside the caps but for metas this can only potentially be
signalled via the ALLOCATION query for negotiation purposes.
A separate format for this is not very useful and instead it should be a
format qualifier.
CEA608_S334_1A is the format defined by SMPTE S334-1 Annex A and which
is used for transferring CEA608 over SDI instead of CEA708 CDP packets.
