Disabling dB volumes for max_dB < 0 was added in Pulseaudio in 2021,
based on a device which had -128..-127.07 range. However, negative
max_dB is valid value for USB devices, and there are devices that have
it.
Eg. Microsoft LifeChat LX-3000 has
numid=6,iface=MIXER,name='Speaker Playback Volume'
; type=INTEGER,access=rw---R--,values=2,min=0,max=151,step=0
: values=150,150
| dBminmax-min=-28.37dB,max=-0.06dB
and the dB range seems to be OK. Web search for "The decibel volume
range for element" also gives other hits with seemingly OK looking
ranges.
Don't disable dB volume unless both the max is negative and the range is
suspiciously small. This should still disable it for the device this
check was originally added for.
Link: https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/447
Link: 10ac01a206
Some ALSA devices have minimum HW volume value that is muted. ALSA
indicates it with SND_CTL_TLV_DB_GAIN_MUTE = -9999999 dB/100 volume dB.
When rounding down to HW volume, we may get this muted value.
When determining splitting of volumes to mixers and soft volume, we
don't want HW mixers to set volume to muted, unless the target volume is
actually muted.
Fix by adding element_ask_unmuted_dB_vol() that rounds up if the asked
rounding mode resulted to mute.
This fixes mic getting muted at low volume despite ALSA reporting the dB
values correctly.
Fixes#4890
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.
Perform detection of supported sample format and rates just after device is
opened, before `snd_pcm_hw_params()` is called for the first time. This fixes a
problem where device restricts available sample rates after HW params are set
preventing sample rate detection (seen with UAC2 devices and kernel 6.1.9)
Bug: https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/1414
Bug: https://github.com/alsa-project/alsa-lib/issues/119
Link: aed52c507f
[Wim: Apply to acp add_pro_profile(), enable pa_alsa_supported_*()]
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.
Perform detection of supported sample format and rates just after device is
opened, before `snd_pcm_hw_params()` is called for the first time. This fixes a
problem where device restricts available sample rates after HW params are set
preventing sample rate detection (seen with UAC2 devices and kernel 6.1.9)
These headers are designed for including in the project. So the user doesn't
need to install valgrind-devel and we don't have to worry about whether the
headers are available or not.
libacp is a port and wrapper around the pulseaudio card profile code.
It uses a set of templates for construct a card profile and mixer port
settings. It also has support for UCM when available for the hardware.
