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>
Use safe values for the min_dB and max_dB fields when the position mask
is unset to avoid breakage for the upper levels.
If the range is incorrect, the volume range shown in pavucontrol shows
strange values.
(Thanks to Wim Taymans for the idea.)
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/389>
Some filters take parameters that effectively describe the hardware
they're being applied to (like echo-cancel allowing to specify the
mic array parameters for better noise filtering). This allows system
integrators to set default parameters for such modules per-device,
which will get used when the stream doesn't specify their own.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/400>
The old behaviour was such that if none of the normal mappings worked,
we would probe ALL fallbacks. I don't think that makes sense, and it
caused concrete issues: let's say we have a regular stereo mic device,
but there's no "front" PCM defined for it. In this situation we would
probe the stereo-fallback mapping (which uses "hw" instead of "front"),
and it would work, but then we'd also probe the "multichannel-input"
mapping, which would also work, so we end up with two mappings that
don't have any difference in behaviour.
I think it's better to simply pick the first working fallback and ignore
the rest.
Fixes: https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/901
(issue is marked as confidential due to unreleased hardware)
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/304>
