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>
Use snd_ctl_card_info to set some more card properties such as the
alsa.id, alsa.mixer_name and alsa.components.
alsa.id is interesting because it is possible to use udev rules to set a
custom id, which is handy when you have two identical cards in the
system and want to assign unique ids to them.
See #3912
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.
Currently the HDMI output paths have jack mixers named "HDMI/DP" and
with append-pcm-to-name=true. However, most of the SOC audio drivers
are just named "HDMI" and don't add the ",pcm=N". Add these alternate
jack names to the HDMI audio path files so that jack detection will work
on these SOCs.
We would timestamp within an unlikely block, which would introduce
additional jitter to current_time, which would have an impact on
the performance of the timer sensitive code.
Always reevaluate the rate matching even when we did not change the
follower state.
It is possible that we were a follower from some node with the same
clock and now become a follower of a node with a different clock. The
follower state doesn't change but we need to activate the rate matching
logic in that case.
Fixes rate matching in pro audio (playback) when capture and playback
are moved to another driver.
We can increase the MAX_LATENCY again if we increase the amount of
buffers when we are using a small buffer.
Normally we ask for 4 * quantum-limit as the buffer. This should be good
to use 1 buffer and quantum-limit as the quantum with enough headroom
to not run out of buffers.
If we are however using less buffer-frames we need to be careful and
allocate an extra buffer. Imagine using a buffer of 4096 frames, we can
support a quantum of up to 2048 frames if we use 2 buffers.
See #3744
Half of the buffersize is not enough to support as a max-quantum, we
need to divide by (4 * frame_scale) to allow some headroom and account
for the DSD scaling. We do the same calculation to suggest a buffer size
using the quantum-limit.
See #3744
As part of the setup for IRQ based scheduling, a period event
was installed. Not only is a timer based polling unecessary for
IRQ scheduling, depending on the state of the system, the timer
could fire far enough from the IRQ, causing alsa wakeup events
with no data in the ring buffer. Pipewire would identify these
events as an "early wakeup", adding an extra quantum of time
to the next_time estimate, skewing the clock and causing issues
with apps that depend on precise timing.
This reverts commit 49cdb468c2.
We should not do this, the nsec field should be relatable to the clock
monotonic time. If we use the estimated time, without actually using it
as a timer, we might end up with a wakeup time in the future compared to
the MONOTONIC clock time.
Instead, you can use the estimated current time simply by subtracting
the rate corrected duration from the next_nsec. This is really only
useful for some selected use cases (like in the JACK library).
This fixes some issues where in pro-audio mode, a client would try to
compare the current MONOTONIC time to nsec and find that it is in the
past.
This commit was done in an attempt to fix#3657 but it turned out the
real problem was something else.
The alsa sequencer rate matching was not actually working correctly.
It would compare the previous queue time with the current time and
compare that to the quantum. This would include uncorrected errors from
jitter and would result in the timeouts being scaled in the wrong
direction forever.
Instead, calculate an ideal queue time and compare our current queue
time against that. We then use the correction to scale the timeout or
the next queue time prediction.
Also use the predicted time as the base time for the event timestamps.
this results in less jitter.
Fixes#3657
sync_mixer() calls d->set_volume(d, &d->real_volume);
which makes v and &dev->real_volume point to the same memory area
and valgrind complains:
Source and destination overlap in memcpy(0xcc53e2c, 0xcc53e2c, 260)
at 0x488CFA0: __GI_memcpy (vg_replace_strmem.c:1121)
by 0xBB0803F: set_volume (acp.c:1143)
by 0xBB0EDCB: acp_device_set_port (acp.c:1897)
by 0xBA9CD87: impl_set_param (alsa-acp-device.c:757)
because the compiler apparently implicitly converts this into a memcpy()
and memcpy(3) explicitly says "The memory areas must not overlap."
Don't try to multiple the max_buffer_size with the frame scale or else
we might try to set a min_buffer_size larger than the max_buffer_size.
Instead, use the frame_scale only to scale the quantum_limit and then
clamp against the max_buffer size.
See #3000
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).
Don't use the current time as the nsec field in the graph clock because
it can jitter a lot. Instead, use the smoothed next_time, like we do
for timer based scheduling.
Since we track the current time against the rate converted ideal time,
lock on to the first timestamp when we reset the dll.
See #3657
Don't return the value of the last snd_pcm_resume() call because that
might be -ENOSYS when resume is not implemented for the card and then
the non-error (because we used drop/prepare later) propagates and
logs an error.
Backport from Pulseaudio. Reimplement get_data_path. We'll look for the
override files similarly as we do for other config files
(XDG_CONFIG_HOME then /etc then install location), instead of looking at
the Pulseaudio locations ~/.local/share/pulseaudio etc.
Upstream commits:
From: SimonP <simonp.git@gmail.com>
alsa-mixer: Respect XDG base directory spec when loading profile sets
Try $XDG_DATA_HOME, then $XDG_DATA_DIRS, and finally fall back to old behaviour.
From: SimonP <simonp.git@gmail.com>
alsa-mixer: Respect XDG base directory spec when loading path configs
Try $XDG_DATA_HOME, then $XDG_DATA_DIRS, and finally fall back to old
behaviour (prefix-defined directory).
core-util: Ignore non-absolute XDG base dirs
These are invalid per the spec.
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.
Because we now always _drop/_prepare_/_start, the snd_pcm_recover()
before that is no longer useful.
Retry snd_pcm_resume() after suspend when -EAGAIN and fall back to
_drop/_prepare/_start when that fails.
We need to disable the resampler when there is a pitch element. This was
correctly done in setup_matching but not in check_position_config().
See #3628
When the device has not configured a format, remove the properties that
depend on the format so that they don't limit what we can configure the
device with next.
See #3613
We calculate the available frames in read_sync but add another
check in read_frames so that we don't attempt to read more frames
than we have available to avoid xruns.
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.
When checking that a card has all of its PCM devices available, ignore
any specific device with the ACP_IGNORE udev environment variable. This
mirrors how we ignore whole cards, but specifically allows non-PipeWire
software to own specific PCM devices.
Note that this does not actually stop PipeWire from using those
subdevices right now, we assume UCM configs take care of that. This
should probably be implemented later to ensure PipeWire always stays
away from them, but for now this fixes the issue where it refuses to
probe the entire card.
Fixes: #3570
Signed-off-by: Hector Martin <marcan@marcan.st>
All linked PCMs prepare together. If we prepare the secondaries, that
action clobbers the write pointer of every PCM every time, which then
causes playback to fail to start due to lack of data.
Signed-off-by: Hector Martin <marcan@marcan.st>
When the PCM is stopped, don't check for early wakeup because if we
are early, we will never be on time in the next iteration either because the
PCM is stopped and doesn't advance.
Also don't try to align when stopped.
See #3565
