Copy the server value to the context so that the locally allocated
buffers match the server quantum-limit and we don't cause xruns because
of too small buffers.
See #4490
Make sure we only make the buffer for the follower larger when we
downsample because then we need to ask for more data from the follower
to fill up a quantum.
Never try to make the follower buffer smaller than the quantum limit.
The reason is that the graph rate could be decreased dynamically and
then we would end up with too small buffers.
See #4490
Load multiple graphs with audioconvert.filter-graph.N where N is the
order where the graph is inserted/replaced. Run the graphs before the
channelmixer.
Graphs can be added and removed at runtime.
Instead of recalculating what to do every cycle, we can prepare a
static schedule and just run that. We only need to reevaluate it when
something changes.
For input streams, first run the resampler and then the channelmix. This
ensures that the channelmix is run with the rate of the graph instead
of the rate of the input. This is nicer because rate and quantum align
with the graph and the sample accurate volume ramps will work as
intended.
For output streams, leave the resampler after the channelmix for the same
reasons.
Remove the chunk and add separate arrays with data and n_samples. This aligns
better with other methods and makes it possible to more easily reuse
arrays of pointers as input and output.
An input port has at most 1 link so we can avoid the loop and the
default port data. We can then also improve the logging and log
something when we set silence.
`(n_items) + 1 * sizeof(*items)` is not the correct size to allocate
for `n_items + 1` count of objects each with size `sizeof(*items)`,
the `+ 1` should be inside the parentheses.
Fixes#4481
Fixes: 4baa94fce2 ("thread: make it possible to set a custom create function")
Both `spa_rectangle` and `spa_fraction` store unsigned numbers,
so print them as unsigned, the same way it is already done in
`spa_debug_strbuf_format_value()`.
For some streams, the buffer size is changed and may exceed
the acquired buffer size which is acquired from the pool of
pipewiresink. Need split buffer and send them in turn for
this case.
We want to track the difference between the PTP timestamp (now) and the
last RTP send, not the synthesized next RTP timestamp (which will always
be smoothly incrementing).
When api.alsa.split-enable=true for ACP device, instruct UCM to not
use alsa-lib plugins for SplitPCM devices.
Grab the information from UCM for the intended channel remapping, and
add the splitting information to the nodes emitted.
Session manager can then look at that, and load nodes to do the channel
splitting.
The current biquad calculations are based on RBJ's cookbook [1],
except for low-/highpass. Since the filter configuration is also
based on using the definition of Q, it makes sense to also align
the remaining calculations to use the same filter cookbook instead
of using resonance which doesn't result in the same coefficients
as when using Q.
[1] = https://www.w3.org/TR/audio-eq-cookbook/
Add support for latencyOffsetNsec Prop, which just controls the nsec
part of the ProcessLatency.
This is needed to support latency offset in Pulseaudio apps when using
loopbacks as front-end nodes to underlying sinks.
The expression `VBAN_PROTOCOL_SERIAL | vban_BPSList[14]` is assigned
to an 8 bit field of the header, but, `vban_BPSList[14]` being
115200, it does not fit. Instead, its index, 14, should be
placed in the header.
In addition to fixing the issue, add `-Werror=constant-conversion`,
and clang diagnostic that catches such issues.
Fixes: 1a5514e5cf ("module-vban: create streams per stream_name")
