Parse the audio.position spec completely so that we have the right
number of channels but only store the first max_position channels.
Also rename some field to make it clear that this is about the max
number of channel positions.
Add a function that accepts the size of the position array when reading
the audio positions. This makes it possible to decouple the position
array size from SPA_AUDIO_MAX_CHANNELS.
Also use SPA_N_ELEMENTS to pass the number of array elements to
functions instead of a fixed constant. This makes it easier to change
the array size later to a different constant without having to patch up
all the places where the size is used.
Configure the headroom to be equal of the minimum allowed period size for
the configuration.
This is desirable when the ALSA driver's hw_ptr is 'jumpy' due to
underplaying hardware architecture, like SOF.
In case of SOF the DSP firmware will burst read at stream start to fill
it's host facing buffer and later settles to a constant pace. The minimal
period size is constrained by the driver to cover the initial burst and
settling time of the hw_ptr.
Guard this mode of working with a new boolean flag, which is only enabled
for SOF cards, kept it disabled for other cards to avoid any unforeseen
side effects.
Even if the use-period-size-min-as-headroom is set to true, the manual
headroom configuration will take precedence to allow experimentation.
Link: https://github.com/thesofproject/linux/issues/5284
Link: https://github.com/thesofproject/sof/issues/9695#issuecomment-2569033847
Link: https://github.com/thesofproject/sof/issues/10172
Link: https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/4489
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@linux.intel.com>
Replace force_rate with force_quantum. We use force_rate when we need
to play an IEC958 or a DSD format but it does not make sense to just
force the rate without also forcing the duration.
This is also what happens when doing IRQ based scheduling, we then force
both the duration and rate of the graph so we can reuse this logic.
Also when forcing a quantum, take into account the suggested duration
and rate of the graph and scale that with the currently configured rate
for the period size. This gives a quantum that will match the requested
rate better. This is important for the DSD, where rate are very high and
we want the period size to be something reasonable relative to the
selected graph rate.
For batch devices (and when using a timer) we also configure a period
size that is half the duration of the quantum, to make sure we get some
headroom. We however need to force the full duration as the quantum, so
keep track of this scaling and apply when calculating the duration.
Based on testing, ALSA FireWire drivers introduce additional latency
determined by the buffer size.
Report that latency.
Pass device.bus to the node, so it can recognize firewire.
In USB Audio Class 2 (UAC2) setups, pitch control is handled by
feedback endpoints. The host adjusts its data rate accordingly.
When pitch control is active (pitch_elem), applying the default
delay-locked loop (DLL) bandwidth can lead to instability and
oscillations around the target rate.
This patch adds a new parameter, api.alsa.dll-bandwidth-max, to
configure the maximum DLL bandwidth. It introduces a new field
in the ALSA state to store this value.
By default, it uses SPA_DLL_BW_MAX, but when pitch control is in
use, setting it to a lower value (e.g. 0.02) helps ensure better
stability, based on empirical testing.
Including C headers inside of `extern "C"` breaks use from C++. Hoist
the includes of standard C headers above the block so we don't try
to mangle the stdlib.
I initially tried to scope this with a targeted change but it's too
hard to do correctly that way. This way, we avoid whack-a-mole.
Firefox is working around this in their e21461b7b8b39cc31ba53c47d4f6f310c673ff2f
commit.
Bug: https://bugzilla.mozilla.org/1953080
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.
When the driver node is destroyed (like when unplugging the cable) it
will drop/pause all of the follower ALSA nodes. This is something that
happens internally because of how the ALSA nodes work together, on the
PipeWire level, the nodes are still running and they will just be moved to
another driver.
The problem is that nothing will then start the nodes again after moving
it to the new driver. Fix this by keeping track of the desired target
state of the ALSA node and restoring that state when we detect that we are
paused when moved to a new driver.
Fixes#4401
Keep a running average and variance of the error. Use this to
periodically update the DLL bandwidth. When the variance gets smaller,
we update the DLL more slowly to stay closer to the ideal rate.
This seems to improve the rate stability.
Use the helper instead of duplicating the same code.
Also add some helpers to parse a json array of uint32_t
Move some functions to convert between type name and id.
Add spa_json_begin_array/object to replace
spa_json_init+spa_json_begin_array/object
This function is better because it does not waste a useless spa_json
structure as an iterator. The relaxed versions also error out when the
container is mismatched because parsing a mismatched container is not
going to give any results anyway.
This fixes an issue introduced in 771f71f622
where the quantum is forced and may break applications the specify their
own quantum.
Signed-off-by: Lukas Rusak <lorusak@gmail.com>
This does a couple of things: first, we implement revents demangling,
which seems to be required (although hw: devices work fine without it).
The second is to actually read the ctl events so we can tell when
elements we care about have changed, instead of reading everything and
trying to do a diff.
The latter is also required from a correctness perspective, as otherwise
the ctl might keep triggering wakeups while the fd is ready to be read.
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
