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.
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.
Some drivers (emu10k1) appear to not necessarily support more than 2
periods.
Don't fail start if snd_pcm_hw_params_set_periods_min() fails, then we
just set nearest possible periods and buffer sizes.
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.
Some devices (FireWire) fail to produce audio if period count is < 3,
and also have small buffer size. When quantum is too large, we might
then get too few periods and broken sound.
Set minimum for the period count in ALSA, to determine the maximum
period size we can use. If smaller than what we were going to use, round
down to power-of-2.
See #4785
The Max latency property only works for timer based scheduling so that
we don't select a quantum larger than we can handle in our buffer.
With IRQ based scheduling this does not make sense because we will
reconfigure the buffer completely when we change quantums and so the
currently selected buffer size does not limit the latency in any way.
Fixes#4877
Some drivers (Firewire) have a latency depending on the ALSA buffer size
instead of the period size.
In IRQ mode, we can safely use 2 (or 3 for batch devices) periods
because we always need to reconfigure the hardware when we want to
change the period and so we don't need to keep some headroom like we do
for timer based scheduling.
See #4785
spa_alsa_read is called from the source process function when we are a
follower and no buffer is ready yet.
Part of the rate correction was performed by the ALSA driver when it
woke up but now, the resampler has updated the requested size and we
need to requery it before we can start reading samples.
Otherwise, we end up with requested samples from before the rate update
and we might not give enough samples to the resampler. In that case, the
adapter will call us again and we will again try to produce a buffer
worth of the requested samples, which will xrun.
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.
In some cases, it is possible that the follower shares a clock with the
driver, but the driver rate is not known when the follower is assigned
to the driver. If this happens, then state->driver_rate is 0, and when
setting the format, we might think that we need to resample (because
follower rate != driver rate). This can cause us to incorrectly halve
the period size for the node.
This was introduced in commit 0b67c10a9c,
which forces reevaluation of matching status on driver change.
To avoid this, let us also probe for the driver rate when updating the
matching status, so we can make the update more accurate.
alsa_write_sync can insert or remove some data from alsa when
resynchronization is needed.
Avail and delay are equal when high precision timestamps are not allowed.
When the high precision timestamps are enabled, the delay is avail
adjusted to current_time.
Signed-off-by: Martin Geier <martin.geier@streamunlimited.com>
When a linked node needs to be resynced we actually never clear the
flag or reset the dll. Move the code around so that it still does
the reset of the flag and dll without actually doing the resync in
the ringbuffer when it is a linked node.
When we do_prepare, always reset the dll. We already set the alsa_sync
field but that is only used by followers to resync in some cases.
When reseting the dll, we also reset the next_time and base_time values,
we however need to do this before calculating the error in update_time
when we are the driver in IRQ mode or else we get some crazy error
that distorts the rate estimation.
Interrupts are disabled in alsa_irq_wakeup_event -> playback_ready method
to not produce another wakeups when waiting for a new data. Interrupts are
enabled again when a new data arrives in a method spa_alsa_write.
In rare cases, when there is multiple streams providing data and one of
them is disconnected, a new data fails to be delivered and the spa_alsa_write
is not called. Not providing data produces underrun and alsa-pcm invokes
recovery process. Recovery process starts a new playback, but without interrupts
enabled is graph not triggered and new data are not delivered (to enable
interrupts). Recovery process keeps running in loop.
Now the interrupts are enabled again after the recovery and the starvation
should not occur.
Signed-off-by: Martin Geier <martin.geier@streamunlimited.com>
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.
If the PCM is loaded directly using only api.alsa.path,
state->card_index may never be initialised, and then we end up opening
the wrong ctl device. Let's try a fallback for this case.
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
Add a new followerClock block in the profiler info. This is only set
when the follower could be a driver and it contains the clock info used
for following the driver, mostly the rate difference and delay.
Dump this info in pw-profiler -J
Make sure we always set the info in the clock, especially also when we
are following.
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.
We have to unlink pcms when they are linked to a driver from
a different pcm.
When a playback and a capture pcm is linked and we start
the playback pcm and the capture pcm later this can leads
to a 'EPIPE' error on the capture device.
...
spa.alsa: hw:3,0c: snd_pcm_start: Broken pipe
...
While the spec allows for 1ppm changes, our rate matching logic applies
these changes quite often, which can be spammy on USB. I haven't seen
hosts mind this, but it seems like it might be a problem at some point.
Additionally, if we also have bind ctls enabled, every pitch update is
also a wakeup for ourselves (whether or not we're listening for the
pitch ctls, since the mixer fd does not distinguish between ctls, those
are filtered after we wake up).
The 10ppm threshold is empirically tested as being not "too noisy" (i.e.
when updates happen, I can see them scroll by with `amixer events`).
If necessary, we can make this configurable in the future.
