It doesn't make sense to hang these on the data loop, so let's have
these on the main loop instead. Also avoids a potential crash while
removing them (since removal happens on the main loop and the data loop
might be polling while we're doing the remove).
This adds an api.alsa.bind-ctls property to alsa-pcm sink and source
nodes, to bind a property to an ALSA PCM ctl. The property is an array
of ctl names that should be bound.
This can be handy, for example, to bind the Playback/Capture Rate
controls on a USB gadget, in order to track the PCM's state via a node
param.
This is currently wired to be read-only, but it should be easy enough to
make it writable.
Make a function to recalculate the headroom and call it whenever the
resample state of the node can change.
When we are IRQ based scheduling but need to resample, we are actually
not driving the graph whit IRQ and need to adjust our period size and
headroom as if we are using timers.
Add an option to automatically use snd_pcm_link when the follower clock
is matching the driver. Only set this to true in pro-audio and when
nodes are scheduled together.
See #3556
Rename some variables to make it clear that we are dealing with the rate
and duration of the driver.
Set the driver rate in the clock instead of the target_rate, which we
might have rejected.
When we are using the same clock (!matching) try to link the two PCM
devices together. This starts and stops the devices at the same time and
gives better latency.
Keep track or the follower state and do the sync from the driver
wakeup.
This improves sync between devices because the sync no longer
depends on when the node was woken up in the graph and scheduled. We
would have been able to handle this difference by using the htimestamp
but that doesn't seen to work reliably on some hardware.
This is also essential for the interrupt based scheduling.
Split the spa_alsa_start() in a prepare and start function. Prepare
will then also reset state and prefill the playback buffer with silence.
This makes it possible to reuse the prepare function in recover().
If we detect Playback/Capture Pitch 1000000, we can adjust those values
to update the feedback endpoint for the host. On the capture side, this
will instruct the host to adjust the rate at which data is being sent.
On the playback side, this will adjust the amount of data the USB gadget
driver sends out in each USB tick.
Use a separate variable to hold the maximum amount of drift we allow
between driver and follower. Ensure this value is smaller than the max_error
and period size so that we have at most 1 period of drift.
Provides configuration to disable timer-based scheduling. This can be
useful at low latencies, for example, where period-based interrupts
might be more reliable than timers.
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
For encoded format, we need to send bigger chunks to make the encoder
happy. Add a new min_delay variable with this info so that we never
leave less than that amount of samples in the ringbuffer.
Fixes#2650
Pass the current time around in various functions.
Make a higher precission htimestamp based get_delay() function. Seems to
work fine for playback but not for capturee.
Parse the quantum_limit parameters and use this to scale the buffers so
that they can contain the maximum allowed samples instead of the
hardcoded 8192 value.
See #1931
The `card` structure uses `uint32_t` for its index member,
on the other hand, the `state` structure uses `int`. No code
depends on it being `int`, therefore change it to `uint32_t`
for consistency.
The id is useless when dealing with props params, we need to use the
name of the property as the key. Also the id can clash with other ids
of other plugins.
Add PropInfo for all the params that we can configure at construct
time and also add them as PROP_params.
This way you can configure the headroom at runtime with this:
pw-cli s <id> Props '{ params = [ "api.alsa.headroom" 1024 ] }'
Expose the card object and always obtain one per pcm.
Keep the configured format in the card object.
Add a api.alsa.multi-rate property. When multi_rate is disabled,
only allow the last configured card rate on all PCMs.
This works around drivers that can't handle multiple samplerates
on their PCMs.
With this patch it should be mostly safe to configure multiple
sample rates in pipewire.conf
See #1547
