Add some guards against doing processing when there has been an error or
the node is not started. Set error status to IO. Continue driving on IO
errors.
In media-sink, there's no need to set RCVBUF.
In media-source, we don't need to set NONBLOCK, as reads are done with
DONTWAIT. Don't set SNDBUF as it's not needed there. Don't set RCVBUF,
but use the (big) kernel default value: decode-buffer will handle any
overruns. Small values of RCVBUF might cause problems if kernel is
sending packets in a burst faster than we wake up.
Single argument static_assert() is only available since
C++17 and C23. Some compilers accept it even in earlier language
versions, but some do not.
Single argument static assertions can be supported by using
a GCC extensions, namely that `, ## __VA_ARGS__` removes the
comma if the variadic argument list is empty. This enables a
construction which passes a pre-determined string as the second
argument of the underlying static_assert() when only a single
argument is used in the `SPA_STATIC_ASSERT()` macro.
Fixes#3050
Don't just limit the max delay of samples we keep in the ALSA ringbuffer
to the buffer_size but to half of it. Make this into a max_delay
variable.
If we have a buffer size of 8192 samples and a headroom of 8192 samples,
when capturing, we would wait for the ringbuffer to contain at least
8192 samples, which would always xrun. When we limit the size to
half, we can still read the data without xruns.
Fixes#2972
On underflow in sources, pad with explicit silence. This avoids the
audioadapter from getting off sync from the cycle. That causes problems
as driver when we want to produce a buffer only a the start of the
cycle.
In some cases, it's also possible that the io already has buffer at the
start of the cycle when rate matching as driver. Currently, we don't
produce buffer in this case, but we should. Fix that by doing things in
the exact same way as ALSA sources do.
On glibc, `pthread_t` is `unsigned long int` while on musl
it has a pointer type. To avoid format string warnings,
cast it to `void *` and use the `%p` format specifier.
Delay output by one packet, so that we never need to wait for
node_process to supply more data when a packet is due out, and can write
audio packets at exactly equal intervals (up to timer/io accuracy).
In principle, this should not be necessary. However, enable it for now,
in case this improves the various stutter/etc. bug reports.
After flushing a packet, encode the next one immediately if we already
have the data. This makes the flush timing more accurate (std ~4x
smaller) as we don't need to wait for the encode.
* Add support for running the sink as a driver
* Detect which compressed formats are actually supported
* Correctly open/close/start/stop device according to the node commands
* Shift away from tinycompress and use Compress-Offload ioctls directly
to be able to access various caps information (including fragment sizes)
which are unavailable in the tinycompress API
* Implement SPA_PARAM_PropInfo and SPA_PARAM_Props support
The maximum receive buffer target of 6 packets may be too small when
there's huge jitter in reception. Increase it so that we may use all
buffer available if needed (2*quantum_limit = 370 ms @ 44100).
For SCO, explicitly set maximum buffer to 40 ms, so that latency cannot
grow too large there. For A2DP duplex, set it to 80 ms for same reason.
These are close to the old 6*packet limit.
For BAP server audio sink, set buffering target so that we try to match
the target presentation delay. Also adjust requested node latency to be
smaller than the delay.
Also fix BAP transport presentation delay value parsing, and parse also
the other BAP transport properties. Of these, transport latency value
needs to be taken into account in the total sink latency.
Use a 0 mask to handle unknown layouts. When the source or destination
is an unknown layout, pair, distribute or average. When pairing, keep
track how we paired and use that to construct the matrix later.
This fixes [ UNK UNK ] -> [ FL FR ] mapping by pairing.
Many distributions provide outdated libcamera versions. This change should also help making changes to libcamera itself.
System libcamera is kept a default to avoid breaking existing build processes relying to packaged libcamera.
Handle MONO layout as a real layout, not just like FC. This means it
does not share the FC mixing weights.
Only distribute and combine MONO channels when the target is also
MONO, enable normalization in this case.
Otherwise downmix and upmix the mono channels like any other channel,
which will make it respect the upmix and other settings.
Change some tests with this new way of doing things.
Fixes#3010
Codec switching does not currently work properly for source/duplex.
With BAP it's also possible only when we're BAP client.
When we can't codec switch, emit the "codecless" BAP profile.
When we are rate matching, keep some more headroom to make sure we
have enough data for the adaptive resampler.
Fixes crackling when following the dummy node and probably also when
following another capture device.
Add support for using other clocks.
clock.id can be used to set one of the system clocks.
clock.device can be used to open a clock device such as a PTP clock
device.
Use a dll to track the progress of non-monotonic clocks.
We always probe the Pro Audio profile with the maximum number of
channels but this can lead to a more limited amount of sample rates.
Add an option to set the channels used when probing so that the other
samplerates become available.
Fixes#2990
