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>
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>
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>
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>
Allow adding module arguments using udev PULSE_MODARGS environment variable and
fail module loading if there is a problem with PULSE_MODARGS
This helps setting e.g. 'tsched=0' for specific devices without a need to create
full load module entry in default.pa.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/436>
With the Auto-Mute enabled, if the headphone jack is plugged, the
alsa hda driver will mute the speaker and set pinctl of the speaker
to Hi-Z state, after this happens, even the pulseaudio unmute the
speaker, the speaker still couldn't output sound because the pinctl
is in Hi-Z state.
We found this issue on a Dell machine which has multi-function audio
jack, after the headphone is plugged in, the speaker's availability is
still unknown, users could select speaker from gnome-sound-setting,
but even the speaker is selected to be the active device, it couldn't
output sound.
The Auto-Mute is not useful if the pulseaudio is running since pa
could mute/unmute devices according to active port change, the ucm
for sof+hda already disabled the Auto-Mute, let us disable it for
hda audio if the machine has the internal speaker.
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/433>
Since there is now support for specifying the index of an Element, add the
same config as is used for the output-mono variant, as they behave the same:
One volume control with no support for adjustments to the left and right
channels.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/274>
Since commit ad447d1468 (in 2009) pa_read and pa_write take care of
handling EINTR error.
So, pa_read, pa_write, pa_iochannel_read and pa_iochannel_write can not
exit with errno set to EINTR, and testing it is useless.
module-jackdbus-detect now accepts sink_name, sink_properties,
sink_client_name, sink_channel_map, source_name, source_properties,
source_client_name, and source_channel_map arguments that will be passed
through to module-jack-source and module-jack-sink (without the sink and
source prefixes, except where needed).
It takes much time when starting to capture because max latency is set
to 2 seconds as a initial value. null-source latency need to be set a
lower value than initial value to improve latency.
playing sound through null sink takes almost 2 seconds at first time
playback when norewinds is set. Because block_usec is set 2 seconds at
initializing time. The value will be changed 50 msec after calling
update_request_latency callback.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/406>
This replaces the original virtual surround sink with a total
rewrite, aiming to implement any number of hrir use cases,
including asymmetrical impulses as two separate left and right
output files. It uses FFTW3 FFT convolution, using the overlap-
save method, with full rewind support. It operates in steps
equal to the resampled length of the hrir, and overlaps input
blocks in increments equal to the size of the FFT block. If
using paired hrirs, it requires matched sample spec and sample
rates and channel maps. For best results, the input files should
have speaker maps, rather than expecting the sample loader to
auto detect the mapping.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/240>
The HP Thunderbolt Dock [1] has two separate USB cards, a headset jack
and an optional module which is a speakerphone.
This patch adds new description for them, and mark the intended-roles as
phone for the speakerphone module.
[1] https://store.hp.com/us/en/pdp/hp-thunderbolt-dock-120w-g2-with-audio
We already supported the CLFE element, which should be semantically
equivalent, so I just copied all the CLFE element definitions.
The Center/LFE element is seen on Creative X-Fi with 20K1 chipset cards.
Fixes: https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/978
Newer GCC warns us that the channel_map and volume in legacy entries are
accessed via pointers, and these might be unaligned as the legacy entry
is a packed structure. For this reason, we read out those values into
local variables before accessing them as pointers.
The warnings are:
[146/433] Compiling C object src/modules/module-device-restore.so.p/module-device-restore.c.o
../src/modules/module-device-restore.c: In function ‘legacy_entry_read’:
../src/modules/module-device-restore.c:554:51: warning: taking address of packed member of ‘struct legacy_entry’ may result in an unaligned pointer value [-Waddress-of-packed-member]
554 | if (le->volume_valid && !pa_channel_map_valid(&le->channel_map)) {
| ^~~~~~~~~~~~~~~~
../src/modules/module-device-restore.c:559:48: warning: taking address of packed member of ‘struct legacy_entry’ may result in an unaligned pointer value [-Waddress-of-packed-member]
559 | if (le->volume_valid && (!pa_cvolume_valid(&le->volume) || !pa_cvolume_compatible_with_channel_map(&le->volume, &le->channel_map))) {
| ^~~~~~~~~~~
../src/modules/module-device-restore.c:559:104: warning: taking address of packed member of ‘struct legacy_entry’ may result in an unaligned pointer value [-Waddress-of-packed-member]
559 | if (le->volume_valid && (!pa_cvolume_valid(&le->volume) || !pa_cvolume_compatible_with_channel_map(&le->volume, &le->channel_map))) {
| ^~~~~~~~~~~
../src/modules/module-device-restore.c:559:117: warning: taking address of packed member of ‘struct legacy_entry’ may result in an unaligned pointer value [-Waddress-of-packed-member]
559 | if (le->volume_valid && (!pa_cvolume_valid(&le->volume) || !pa_cvolume_compatible_with_channel_map(&le->volume, &le->channel_map))) {
| ^~~~~~~~~~~~~~~~
[211/433] Compiling C object src/modules/module-stream-restore.so.p/module-stream-restore.c.o
../src/modules/module-stream-restore.c: In function ‘legacy_entry_read’:
../src/modules/module-stream-restore.c:1076:51: warning: taking address of packed member of ‘struct legacy_entry’ may result in an unaligned pointer value [-Waddress-of-packed-member]
1076 | if (le->volume_valid && !pa_channel_map_valid(&le->channel_map)) {
| ^~~~~~~~~~~~~~~~
../src/modules/module-stream-restore.c:1081:48: warning: taking address of packed member of ‘struct legacy_entry’ may result in an unaligned pointer value [-Waddress-of-packed-member]
1081 | if (le->volume_valid && (!pa_cvolume_valid(&le->volume) || !pa_cvolume_compatible_with_channel_map(&le->volume, &le->channel_map))) {
| ^~~~~~~~~~~
../src/modules/module-stream-restore.c:1081:104: warning: taking address of packed member of ‘struct legacy_entry’ may result in an unaligned pointer value [-Waddress-of-packed-member]
1081 | if (le->volume_valid && (!pa_cvolume_valid(&le->volume) || !pa_cvolume_compatible_with_channel_map(&le->volume, &le->channel_map))) {
| ^~~~~~~~~~~
../src/modules/module-stream-restore.c:1081:117: warning: taking address of packed member of ‘struct legacy_entry’ may result in an unaligned pointer value [-Waddress-of-packed-member]
1081 | if (le->volume_valid && (!pa_cvolume_valid(&le->volume) || !pa_cvolume_compatible_with_channel_map(&le->volume, &le->channel_map))) {
|
modules/alsa/alsa-sink.c: In function ‘pa_alsa_sink_new’:
modules/alsa/alsa-sink.c:2603:15: warning: declaration of ‘state’ shadows a previous local [-Wshadow]
void *state;
^~~~~
modules/alsa/alsa-sink.c:2270:11: note: shadowed declaration is here
void *state = NULL;
^~~~~
CC modules/alsa/module_alsa_sink_la-module-alsa-sink.lo
modules/alsa/alsa-source.c: In function ‘pa_alsa_source_new’:
modules/alsa/alsa-source.c:2289:15: warning: declaration of ‘state’ shadows a previous local [-Wshadow]
void *state;
^~~~~
modules/alsa/alsa-source.c:1975:11: note: shadowed declaration is here
void *state = NULL;
^~~~~
modules/alsa/module-alsa-card.c: In function ‘prune_singleton_availability_groups’:
modules/alsa/module-alsa-card.c:691:71: warning: pointer of type ‘void *’ used in arithmetic [-Wpointer-arith]
pa_hashmap_put(group_counts, p->availability_group, count + 1);
^
Commits 323195e305 ("switch-on-port-available: Switch to headphones on
unknown availability") and d83ad6990e ("module-alsa-card: Drop
availability groups with only one port") broke switching from headphones
to speakers when headphones are unplugged. switch_from_port() selects
speakers, whose availability is unknown and availability group is unset,
and then calls switch_to_port(). The new logic in switch_on_port()
unintentionally blocked that switch.
This patch moves the problematic logic from switch_to_port() to
port_available_hook_callback() where it doesn't interfere with
switch_from_port().
Fixes: https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/1043
