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.
Move the port property logic from the adapter to the port itself.
The port was already doing some of the same work as a fallback but can
just as well do everything. This also makes things more unified when
there is no adapter used.
Add some more common channel name shortcuts supported by pulseaudio.
Make sure we match the full channel-name, not just the prefix.
Generate an invalid channel map when an invalid channel name was
given instead of a partial channel map.
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
When a stream uses the FIX_ flags it should negotiate to the format of
the sink or source it connects to. To do this, look up the sink or
source and look at the format, use this as the allowed format for the
stream when the FIX flags are set.
Make it still possible to override with with properties. Use
audio.position to make it possible to set a channelmap override.
The effect is that the null-sink will report the given format in the
Sample Specification, which is what some applications might expect when
they pass a format parameter.
Use a different key than the usual one to select an audio format when we
are fixating a stream format to avoid confusion.
So pulse.fix.rate, pulse.fix.format, pulse.fix.channels are now used to
force a specific format when the stream has the FIX_ stream flags.
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.
Add support for using custom format, rate or channels when the streams
asks to fix those parameters for us.
Some streams might expect to see S16LE when they connect to a sink in
S16LE when the FIX_FORMAT flag is set but this is not the case in
PipeWire because the audio DSP pipeline works in F32, and so F32 is
choosen. Make it possible to use a pulse.rule with a audio.format
property to control this.
Make pw_getrandom() more usable by handling the EINTR case and returning
< 0 when there was an error or not enough random data was available.
Make a new pw_random() function that uses pw_getrandom() but falls back
to a pseudo random number generator otherwise. This pseudo random number
generator is seeded with either data from the urandom source or from the
current time when pipewire is initialized.
In most cases where crytographic security is not required pw_random()
should be easier to use.
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.
Promote the following warnings to errors:
* implicit-function-declaration
* int-conversion
because the code very likely will not work
correctly if any of these two trigger.
