The delay is always expressed in samples at the output rate of the
resampler. For input streams we need to convert this to the expected
input rate.
Make the delay reporting in playback streams more accurate.
We now automatically move non-rt clients into non-rt threads so the
client-rt.conf is obsolete.
Move the module-rt in client.conf and add conditions to disable modules.
Transparently load client.conf in case applications still specify
client-rt.conf.
Custon configuration in the client-rt.conf.d/ should be moved to
client.conf.d/
The resampler delay was off by one sample, so remove the corresponding
fudge factor here. This matters for BAP output synchronization.
The resampler has also some fractional delay, so there can still be
sub-sample offset between the original and resampled timelines. This is
not currently taken into account.
Resampler delay for N taps is N/2-1 input samples.
Add test that checks this.
When input rate is varying, the resampler also accumulates additional
sub-sample delay. The resampler does not currently have API to report
the instantaneous sub-sample delay. Add knownfail test for it.
Wireplumber and other system services use local real time timestamps in
logging, so it's more convenient if also PW uses them.
Add env var for selecting the timestamp type, default to "local".
In timestamps, support different clocks and local time as formats.
Local real time timestamps are useful when trying to correlate logs from
different sources.
When setting Route param, check that the route actually is part of the
active profile.
Also, check that the device given corresponds to the given route, before
setting properties. acp_device_set_port() also checks this, but we
shouldn't allow updating properties of Routes in non-active profiles.
Setting ports or applying props on devices not part of the profile can
do unexpected things e.g. alter mixer settings.
Don't mix endpoint and transport paths, they're generally different.
If ASHA transport already existed, free the old one but not the device,
as the ASHA device existence should track how it appears in DBus.
G722 codec id for ASHA is chosen arbitrarily to be different from
A2DP and BAP codec IDs. ASHA spec does not specify a codec ID like
A2DP/BAP.
In places where codec_id comparisons are done, ensure that the check
is done against the codec of the right type viz. A2DP/BAP/ASHA.
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).
