(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.
Make sure ticks start with an accumulator value of 0 by incrementing it
after filling in samples instead of before and by resetting the accumulator
every time a tick begins. This prevents it from being discontinuous at the
beginning of the tick.
https://bugzilla.gnome.org/show_bug.cgi?id=774050
Before it was done the other way around and that can trigger the assert that
already is in place. This also makes more sense; when seeking to time x, we want
then sample that is <= that pos.
Remove the _ in front of the endianness prefix.
Remove the _3 postfix for the 24 bits formats.
Add a _32 postfix after the formats that occupy extra space beyond their
natural size.
The result is that the GST_AUDIO_NE() macro can simply append the endianness
after all formats and that we only specify a different sample width when it is
different from the natural size of the sample. This makes things more consistent
and follows the pulseaudio conventions instead of the alsa ones.
Rework the audio caps similar to the video caps. Remove
width/depth/endianness/signed fields and replace with a simple string
format and media type audio/x-raw.
Create a GstAudioInfo and some helper methods to parse caps.
Remove duplicate code from the ringbuffer and replace with audio info.
Use AudioInfo in the base audio filter class.
Port elements to new API.
