Volume scaling in dB mode is broken if max dB is negative.
I have a Nobsound USB amplifier (1908:2220) that reports a dB range
of -127.07 dB to -128 dB in Alsa.
While this is likely a driver/device bug, in my naive imagination
userspace wouldn't bother too much with the absolute values and just set
out_dB(percent) = min_dB + (max_dB - min_dB) * percent
However, this is not what PulseAudio is doing, instead max_dB is used
as base_volume with which the desired software volume is multiplied
while min_dB does not seem to be taken into account.
The result is that with this device only a tiny portion of the volume
slider is usable.
Setting it to 97% already reaches min_dB which effectively turns any
(software) audio knob to an on/off switch.
To work around this, simply set the has_dB flag to false if max_dB is
negative.
This falls back to using raw Alsa values (ranging from 0 - 255), now
the settings in pavucontrol perfectly mirror those in alsamixer.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/447>
For GNOME X11 sessions, avoid relying on xdg autostart desktop files
to initialize the X11 plugins. This is now handled via a systemd unit
file.
The xdg autostart is still installed, but has been made to instruct
GNOME to skip it with X-GNOME-HiddenUnderSystemd. This is still the
primary way to initialize X11 plugins for other DEs.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/467>
The scripts in this directory are loaded (in GNOME sessions thus far)
at the time of starting Xwayland for X11 clients (may happen on session
start, or on demand whenever X11 clients are started).
This will ensure the relevant X11 modules are loaded as long as there's
a Xwayland instance, thus X11 clients that might make use of them.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/467>
After applying the commit 0d50e787 ("alsa-card: improve the profile
availability logic"), we met an new issue. when system selects the
initial profile, the profile off is selected instead of a profile with
a valid output device on it. That is the issue we met:
Profiles:
HiFi: Default (sinks: 2, sources: 2, priority: 8000, available: no)
off: Off (sinks: 0, sources: 0, priority: 0, available: yes)
Active Profile: off
Ports:
[Out] Headphones: Headphones (priority: 300, latency offset: 0 usec, not available)
Part of profile(s): HiFi
[Out] Speaker: Speaker (priority: 100, latency offset: 0 usec)
Part of profile(s): HiFi
...
I know the commit 0d50e787 really fixed something, but we still need
to fix the new issue, to do so, this patch introduces a priority bonus
for alsa profiles and separate the alsa profiles to 3 groups:
group a (will be granted priority bonus dynamically):
a profile has only output ports and at least one port is not unavailable
a profile has only input ports and at least one port is not unavailable
a profile has both input and output ports, and at least one output and
one input ports are not unavailable
group b (will be marked unavailable)
a profile has only output ports and all ports are unavailable
a profile has only input ports and all ports are unavailable
a profile has both output and input ports, and all ports are unavailable
group c
the rest profiles, their priority and availability is not changed.
With this change, the profile HiFi will become avaialbe:yes, and will
not be granted priority bonus if no input port is plugged.
The priority bonus provides a higher priority base to profiles, this
guarantees this patch doesn't break the fix of 0d50e787.
https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/927
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/355>
WSAStartup was not being called for pacat and pactl built with meson,
causing them to fail in pa_mainloop_new with "cannot create wakeup
pipe". This issue also affects other applications linking to libpulse
other than the pulseaudio daemon, which calls WSAStartup itself.
When built with autotools, WSAStartup would have been called in
DllMain, which is recommended against by the documentation [1].
To fix these issues, the WSAStartup/WSACleanup calls can be moved
into pa_mainloop_new/pa_mainloop_free.
[1] https://docs.microsoft.com/en-us/windows/win32/api/winsock/nf-winsock-wsastartup
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/456>
State database binary file format may depend on system architecture,
for instance gdbm binary format depends on architecture word size,
making x86 and x64 gdbm files incompatible.
If this is the case, it is handled by adding system architecture name to
database file name using automatically configured CANONICAL_HOST string.
Meson build define CANONICAL_HOST to be system architecture name, while
autotools build extends this with vendor and and operating system components.
Switch autotools build to use host_cpu for CANONICAL_HOST to match Meson
configuration. For backwards compatibility always use existing database file
matching CANONICAL_HOST prefix if it exists.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/425>
When an alsa source with fixed latency is used, the actual latency of the source
will only be one fragment size. This is not taken into account when the required
sink latency is calculated.
This patch fixes the issue.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/451>
Document some things that should be helpful to at least new
contributors. Since we don't have a way to show this when people are
creating MRs, also copy over the next to a merge request template so
that creates a dropdown that folks might look at when creating an MR.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/444>
While module-ladspa-sink is still being loaded and before pa_sink_put() is
called there may be an attempt to reconfigure master sink when avoid-resampling
is true. This breaks attempting to suspend ladspa-sink which is still in INIT
state.
Fix this by skipping pa_sink_suspend if PA_SINK_IS_LINKED is false.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/445>
The array read functions need the state pointer as an additional argument because the
array may be in the middle of a parameter list and the state pointer must be advanced
to the element after the array.
Additionally fixes some compiler warnings.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/450>
This is seen at least on HP EliteDesk 800 DM and HP EliteDesk 800 SFF.
This is used by the analog-output-headphones-2 path, but all other paths
on the same sink need to handle the element too. The existing
configuration is inconsistent between files regarding whether headphone
outputs should be muted or not when not using them. I chose to be
consistent within files, which means that Headphone,1 handling is
inconsistent between files in the same way that the existing Headphone
and Headphone2 handling is. (My opinion is that unused paths should be
always muted, but I didn't want to do that policy change in this patch.)
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/272>
Previously both paths had description "Headphones", which I assume can
cause confusion with users who see two ports with identical names. I
don't have this kind of hardware myself nor have I heard complaints from
users, this is just something I noticed while reading the configuration
files.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/272>
On some Dell AIO machines, there is no internal mic, only a multi
function audio jack, so the only input devices are headphone-mic and
headset-mic, and they share the Jack with headphone.
When there is no headset plugged in that Jack, the headphone-mic
and headset-mic are off. And since there is no available port under
the analog input source, this source is unlinked (if there is
internal mic, the source will not be unlinked). so the only pa-source
left in the PA is analog-stereo-monitor.
After the headset is plugged, we need to let switch_to_port() handle
headset-mic and headphone-mic conditionally, this will guarantee the
source will be created if it is unlinked before plugging, and then the
input profile could be selected correctly.
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/390>
We have at least one USB hardware which supports the 8
channels in one mixer element:
https://github.com/alsa-project/alsa-ucm-conf/pull/25
POSITION_MASK_CHANNELS define was added for the future extensions.
The override_map variable was changed from bool to mask (unsigned int).
The channel map override settings is handled for channels up to eight now.
Also added missing override-map.3 .. override-map.8 to the configuration
parser array.
The driver channel position was added to the override mask arguments
(syntax is driver:pulseaudio like left:all-left). If ommited, the ALSA's
channel positions are guessed by index.
Link: https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/292
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/389>
