Reverts alsa-mixer.c changes from 32a3ffc74. The comprehensive ACP
layer fix (bdb82be4e) handles all scenarios including pro-audio
profiles that bypass mixer paths, making this approach redundant.
IEC958 (S/PDIF, HDMI, DisplayPort) switches default to muted in ALSA
drivers, causing no audio output on digital devices.
While UCM configurations and mixer paths can handle IEC958 unmuting,
several scenarios lack coverage:
- Pro-audio profiles (bypass UCM and mixer paths by design)
- Devices without UCM configurations
- Devices with incomplete mixer path definitions
- Cards with multiple HDMI/DP outputs (indexed switches)
This ensures IEC958 switches are enabled during device activation and
port changes. The implementation uses the device mixer when available,
falls back to the card mixer for pro-audio profiles, and enables all
IEC958 switches regardless of index.
Safe for all configurations: the operation is idempotent and provides
defense-in-depth even when UCM or mixer paths handle it correctly.
Tested on AMD Rembrandt GPU with 3 HDMI outputs in pro-audio mode.
When selecting an HDMI/DisplayPort (IEC958) output path, the hardware
mute switch remains in kernel default state (muted), causing no audio
output despite correct software routing.
Root cause: pa_alsa_path_select() only sets mute switches when
mute_during_activation is enabled. No mixer paths enable this setting,
making the switch configuration code unreachable for IEC958 paths.
Solution: Always set mute switches to match device mute status after
path activation, regardless of mute_during_activation setting.
Testing: Added test-alsa-path-select tool to verify the fix.
- Loads mixer path and calls pa_alsa_path_select()
- Verifies switch states match expected values
- Tested on AMD Radeon HDMI and Realtek ALC257 analog
Manual verification:
- Before: IEC958 switch OFF, no audio
- After: IEC958 switch set correctly, audio works
This bug was inherited from PulseAudio's ALSA mixer path code where
HDMI path configurations lack IEC958 unmute sections.
Fixes: https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/3261
See-Also: https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/562
See-Also: 33be660e4b
See-Also: https://bugs.launchpad.net/hundredpapercuts/+bug/681996
Log something less confusing when connection to remote device drops
unexpectedly.
Silence logging transport Release() error in cases where the transport
was simultaneously deleted.
We don't want to override the converter flags with the follower flags,
just enhance them with specific follower flags. Otherwise we lose the
DYNAMIC_DATA and other port flags from the converter.
See #4918
When using LC3-24kHz, remote device drops connection after a few seconds
if there is no sink playback. Avoid this by sending silence, one TX
packet for each RX packet, if sink hasn't been feeding data within a
timeout.
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.
Commit cbbf37c3b8 changed the logic of the
Start command. Before this commit, when there was no converter, the
follower would always get the Start command. After the commit, the
follower would only get Start when previously Paused.
This however breaks when we set a format or buffers on the follower
without a converter because those actions might change the state of the
follower to Paused implicitly.
We should simply remove the started check here and always call Start on
the converter and follower, the implementations themselves will keep track
if anything needs to be done.
Fixes#4911
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.
Align RX of streams in same ISO group:
- Ensure all streams in ISO group have same target latency also for BAP
Client
- Determine rate matching to ISO group clock from RX times of all
streams in the group
- Based on this, compute nominal packet RX times, and feed them to
decode-buffer instead of the real RX time. This is enough for
sub-sample level sync.
- Customise buffer overrun handling for ISO so that it drops data to
arrive exactly at the target, for faster convergence at RX start
The ISO clock matching is done based on kernel-provided packet RX times,
so it has unknown offset from the actual ISO clock, probably a few ms.
Current kernels (6.17) do not provide anything better to use for the
clock matching, and doing it properly appears to be controller
vendor-defined (if possible at all).
Take resampler delay into account when computing the buffer fill level,
including the fractional part.
If decode-buffer is now fed nominal packet reference times in
write_packet(), it converges the total buffer + resampler latency to the
target at sub-sample accuracy.
This is needed for aligning RX of ISO streams in the same group, so that
e.g. stereo pair alignment is achieved even though the streams have
separate resamplers. Resampler phases get aligned via independent rate
matching.
The rate matching calculations are done in the system clock domain. If
the driver ticks at a different rate, the correction factor needs to be
adjusted by the rate_diff.
setup_matching() also needs to be before spa_bt_decode_buffer_process():
as follower we should use rate matching value calculated on the
*previous* cycle, because this is what driver is doing when it adjusts
it tick rate.
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.
FireWire ALSA driver latency is determined by the buffer size and not the
period. Timer-based scheduling is then not really useful on these devices as
the latency is fixed.
In pro-audio profile, enable IRQ scheduling unconditionally for these
devices, so that controlling the latency works properly.
See #4785
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
With the removal of `SPA_DATA_MemPtr` support, this member is no longer used.
Fixes: b948ffdb25 ("spa: libcamera: source: remove `SPA_DATA_MemPtr` support")
On production systems, having a constant high latency is favored over
dynamically adjusting it in order to optimize for low latency,
because every time a dynamic adjustment happens, there's a glitch.
This adds an option to let the user specify the exact amount of latency
they want.
The hardcoded latency of 512/<rate> is quite low on some ALSA devices.
Instead of forcing that latency onto the graph, just don't set it at all
unless it originates from the BAP presentation delay. That means that
the functionality remains the same for BAP but changes for A2DP to favor
the preferred quantum of the ALSA sink (or whatever is the driver).
Also, avoid setting an empty string ("") latency and rate in the cases
where it's not defined. This allows users to override those properties
through the wireplumber monitor rules if they need to.
Currently the v4l2 and libcamera plugins map `SPA_PROP_exposure` in incompatible
ways. So change the v4l2 mapping to `V4L2_CID_EXPOSURE_ABSOLUTE` because at least
that is in units of time (a step closer to addressing #4697), and because that
is more relevant for UVC cameras.
Also change the pipewire-v4l2 translation layer.
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
Without this change the playback with different number of channels
fails with `input port %s[%d]:%s already used as input %d, use mixer`
on the first port.
Signed-off-by: Martin Geier <martin.geier@streamunlimited.com>
Fixes#4866
Force filter graph reconfiguration in setup_convert.
When adding/removing filter-graphs, only perform setup when we were
already setup, otherwise we will do this in setup_convert later.
Don't do channelmix_init when we were not setup.
Deactivate the filter-graphs when we suspend.
Fixes#4866
This makes it a bit more generic and allows us to connect other things
that float arrays.
Add a SPA_FGA_PORT_SEQUENCE as a new port type. The data to connect to
this port is a Sequence type with the size field set to the max/size of
the data.
Improve the spa_ump_to_midi function so that it can consume multiple UMP
messages and produce multiple midi messages.
Some UMP messages (like program changes) need to be translated into up
to 3 midi messages. Do this byt adding a state to the function and by
making it consume the input bytes, just like the spa_ump_from_midi
function.
Adapt code to this new world. This is a little API break..
