Previously, using the -f json or --format=json flags did not return JSON for the following commands:
- get-sink-volume
- get-source-volume
- get-sink-mute
- get-source-mute
This change adds proper JSON output for these commands.
The CI for merge requests is failing because the meson script is unable
to parse the version. With some print debugging I determined that the
version string being generated on the CI is empty because it has no
git tags. I've added a command to fetch the tags before the build.
This issue was found by enabling ubsan. For me it consistently triggered
after about 28 seconds running a simple example that plays a sine wave
via the mainloop api.
I added a log and confirmed that before the ubsan is triggered the
index variable j is indeed 32 which is out-of-bounds.
Co-authored-by: Arun Raghavan <arun@asymptotic.io>
pa_module_load API's return value is integer which is
enum pa_error_code_t with minus such as -PA_ERR_IO
if the module loading is failed.
pa_cli_command_load gets a return value of pa_module_load
as pa_error_code_t which is wrong.
Minus integer value could not covert to enum which is defined
equal or larger than 0 so that pa_cli_command_load would
recognize the return value as larger than 0 if pa_module_load
return value (integer) is minus.
To fix this issue, I modified return value check logic
of pa_module_load API.
As same as pa_module_load's return type, integer would be used
to check if module load is failed in pa_cli_command_load
and the return value would be compared with minus.
Fixes: https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/3801
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/814>
My log files get completely clobbered by this; thousands of lines of:
Jan 18 18:14:44 pulseaudio[29946]: [pulseaudio] backend-native.c: Dock Status: undocked
Jan 18 18:15:39 pulseaudio[29946]: [pulseaudio] backend-native.c: Battery Level: 50%
Jan 18 18:15:39 pulseaudio[29946]: [pulseaudio] backend-native.c: Dock Status: undocked
Jan 18 18:16:34 pulseaudio[29946]: [pulseaudio] backend-native.c: Battery Level: 50%
Jan 18 18:16:34 pulseaudio[29946]: [pulseaudio] backend-native.c: Dock Status: undocked
Jan 18 18:17:29 pulseaudio[29946]: [pulseaudio] backend-native.c: Battery Level: 50%
Jan 18 18:17:29 pulseaudio[29946]: [pulseaudio] backend-native.c: Dock Status: undocked
Jan 18 18:18:25 pulseaudio[29946]: [pulseaudio] backend-native.c: Battery Level: 50%
Jan 18 18:18:25 pulseaudio[29946]: [pulseaudio] backend-native.c: Dock Status: undocked
Jan 18 18:19:20 pulseaudio[29946]: [pulseaudio] backend-native.c: Battery Level: 50%
Jan 18 18:19:20 pulseaudio[29946]: [pulseaudio] backend-native.c: Dock Status: undocked
Jan 18 18:20:15 pulseaudio[29946]: [pulseaudio] backend-native.c: Battery Level: 50%
This seems like it should be a debug log, not a notice.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/810>
The pa_alsa_ucm_set_port() function is passed both a mapping context and
a device port, and both of these refer to their respective UCM device.
While switching over to having one port per mapping per UCM device, I
expected both of these to be the same device struct, so added an assert
checking so.
This assertion gets triggered when we have multiple UCM verbs declaring
the same UCM device name. The root cause here is that the ports' UCM
device references are set once while creating the ports for the card, so
they happen to be those of a specific verb and may not match those from
a different UCM verb's profiles' mappings.
Solving the root cause necessitates a larger refactor. What we actually
assume here is that name of the UCM device is same for both the port and
the UCM context, which ends up always true in practice. For now, replace
the assert with a check and error.
Signed-off-by: Alper Nebi Yasak <alpernebiyasak@gmail.com>
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/802>
Some versions of the ALSA libraries run into a segmentation fault when
we query a UCM device/modifier status without first setting a UCM verb.
It's not a reasonable thing to do anyway, so check for this case and
return an error. Also do the check in other helpers.
Signed-off-by: Alper Nebi Yasak <alpernebiyasak@gmail.com>
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/801>
When we connect Lenovo XT99 bt headset in the Ubuntu 22.04, this
headset could only work in A2DP profile, couldn't work in HFP profile
with a high chance.
This headset supports mSBC, after pulseaudio replies "+BCS:2" to
headset, we expect to receive a "AT+BCS=2\r" from the headset, but
with a high chance, it will receive 2 AT commands in a buffer like
this "AT+CHLD=?\rAT+BCS=2\r", and we also observed other 2 AT commands
in a buffer like this "AT+NREC=0\rAT+CGMI?\r".
Here we don't suppose there is only one AT command in a buffer, we
will find each command by the delimiter "\r" and handle each command
by sequence.
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/804>
When `pwd.h` header is not available (i.e. not using glibc) and environment
variables are not set (e.g. running via `env --ignore-environment`) client
library would crash due to uninitialized variable in `pa_get_home_dir()`.
Add missing initialization to fix that.
Fixes: #3792
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/800>
If pa_memblockq_push needs to write into the middle of a chunk, target chunk
is split into head and tail sharing the same memblock. Size of head and
tail chunks is adjusted correctly, head chunk pointer into memblock remains
unchanged from target chunk.
The problem is with tail chunk offset into memblock which should be advanced
past write region of memblock, but currently it is left as 0.
This is causing an issue where seeking a few frames back into the middle of
memblock and writing a frame there ends up with tail chunk referencing frames
from very beginning of memblock causing corrupted output from memblockq.
Fix this by adjusting tail chunk offset into memblock past write region and
add a test case.
Fixes#3789
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/798>
ref: https://lore.kernel.org/lkml/20221207154939.2532830-4-jeffxu@google.com/
The new MFD_NOEXEC_SEAL and MFD_EXEC flags allows application to
set executable bit at creation time (memfd_create).
When MFD_NOEXEC_SEAL is set, memfd is created without executable bit
(mode:0666), and sealed with F_SEAL_EXEC, so it can't be chmod to
be executable (mode: 0777) after creation.
when MFD_EXEC flag is set, memfd is created with executable bit
(mode:0777), this is the same as the old behavior of memfd_create.
Signed-off-by: Rudi Heitbaum <rudi@heitbaum.com>
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/792>
From https://bugs.debian.org/1006631:
> dbus supports policy files in both `/usr/share/dbus-1/system.d` and
> `/etc/dbus-1/systemd`. [The] recently released dbus 1.14.0, officially
> deprecates installing packages' default policies into `/etc/dbus-1/systemd`,
> instead reserving it for the sysadmin. This is the same idea as the
> difference between `/usr/lib/udev/rules.d` and `/etc/udev/rules.d`.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/783>
Currently there is no way to unset the default sink or source once it was
configured manually by the user.
This patch introduces the special name @NONE@, which can be used with the pacmd
or pactl set-default-sink and set-default-source commands to unset the user
configured default. When the default is unset, pulseaudio will return to the
standard default sink or source selection mechanism based on priority.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/785>
The code that removes the mixer path if probing fails can be called in
the path that sets a non-off device profile on hotplug *before*
card->active_profile is updated, which results in spuriously removing
the mixer path. By this point, context->ucm->active_verb would be set
to the same as the profile name, so we can use that instead to avoid
the issue.
On Apple Silicon machines with the UCM profiles in the Asahi Linux repo,
this manifests as the headphones jack having hardware volume controls
*only* if PA is started with headphones connected and until they are
disconnected. Hotplugs end up triggering the bad codepath, and it falls
back to software volume (which is particularly a problem when the
hardware volume happens to be very low or 0 at that point).
Fixes: a9cc1373e2 ("alsa: ucm - update the mixer path also after volume probe")
Signed-off-by: Hector Martin <marcan@marcan.st>
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/752>
The ucm_get_device_property() function adds to each UCM device's
playback_volumes (or capture_volumes) hash map an associated volume
mixer keyed with the UCM verb. These key-value pairs are then iterated
over in various places which assume the key is a profile name. This
assumption is no longer true since we can generate multiple profiles to
use conflicting devices.
A previous commit 4527890416 ("alsa-ucm: Stop conflating profile name
with UCM verb name") fixes some instances of this assumption, but misses
the relation explained above. Fix more instances of misleading
"profile"s where the UCM verb name is actually meant.
Fixes: 4527890416 ("alsa-ucm: Stop conflating profile name with UCM verb name")
Signed-off-by: Alper Nebi Yasak <alpernebiyasak@gmail.com>
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/771>
Although it's a valid UCM configuration to have multiple devices using
the same PlaybackPCM or CapturePCM, it's unclear how PulseAudio should
handle the cases where multiple of these devices are enabled. Some
options I can think of are:
- Merge all devices sharing the same PCM into the same mapping, open
only one PCM substream for this mapping, and add 'combination ports'
that enable combinations of the devices. This has been the case until
recently, although the combination port logic was broken. A problem
with this is that we can't independently control device volumes. We
most likely cannot use hardware volumes either.
- Have one mapping for each device in the same profile, and open one PCM
substream for each mapping. This is the current state, and it fails
when there are fewer substreams than devices. Otherwise it works, but
it's still confusing, as sound directed to a device-specific mapping
might end up playing at multiple devices.
- Make multiple profiles each with combinations of upto-substream-count
devices, and have one mapping/substream per device. This still causes
the confusion mentioned above. And it's likely that the substream
count will almost always be one, where this case degenerates into the
last one.
- Have one mapping for each device in the same profile, but open only
one PCM substream. I assume this is possible with software mixing, but
it is still confusing like the above, and probably less performant.
- Generate multiple profiles each with one of the shared-PCM devices,
again with one mapping/substream for that one device. The trade-off
with this is that we can't use multiple of these devices at the same
time. However, this doesn't have the output device confusion,
combination port's volume problems, or the substream count limitation.
This patch takes a short-cut to achieve the last option, by considering
shared-PCM devices implicitly conflicting with each other.
Signed-off-by: Alper Nebi Yasak <alpernebiyasak@gmail.com>
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/743>
While switching profiles of the same UCM verb, existing code first
disables devices that are only on the first profile to avoid conflicts.
However, it only disables devices, not modifiers. Even worse, modifiers
which have PlaybackPCM/CapturePCM are incorrectly treated as devices and
result in a segmentation fault.
Check what we are disabling, and call the appropriate disable function
for both devices and modifiers. Modifiers are disabled before devices,
because _dismod calls fail when the modifier's supported devices are
disabled.
Signed-off-by: Alper Nebi Yasak <alpernebiyasak@gmail.com>
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/742>
Currently module-switch-on-connect overwrites the default sink or source that
the user has configured. This means that when the overwritten default sink or
source becomes unavailable, the new default will be chosen based on priority
and the default will not return to the originally configured value.
This patch solves the issue by introducing new core variables for the sink
or source chosen by the policy module which have higher priority than the
user configured defaults.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/784>
This helps to export correct APIs for compiler toolchain which
does not support version script file. For example, mingw clang.
The APIs in libpulse.def are similar with map-file except those
are in pulse-simple and pulse-mainloop-glib. Those are exported
in different shared library in Windows platform.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/761>
There is no requirement for chunk index to be aligned, we only need chunk length
to be multiple of sample frame size.
Fixes: 6434853b0 ("memblockq: Do not allow non-frame indices in the memblock queue")
Fixes: 22827a5e1 ("protocol-native: Fail if trying to push unaligned memblock into queue")
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/779>
`pa_pstream_send_memblock()` would split incoming memblock into parts not
exceeding maximum pool block size.
To make sure split parts of memblock are still frame-aligned add new `align` arg
to `pa_pstream_send_memblock`, find out required alignment from stream sample
format and pass it there. Bump default alignment to 256 which is good up to
32bit 64ch frames.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/780>
Looks like sbc_decode() would seldom access more than specified input length
bytes from input buffer if input length is less than expected frame size.
Fix potential access past allocated memory by checking if input contains
complete frame before calling sbc_decode()
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/781>
The assumption that the format enum is ordered by size is not valid for quite
some time, since 24bit formats were appended to format enum later than 32bit
formats. This causes resampler to produce properly aligned memblock of size
larger than maximum mempool block size if input format is 24bit and output
format is 32bit.
Oversized block is getting split by `pa_pstream_send_memblock()` into parts of
size not exceeding maximum mempool block size. This usually works well but for
32ch 32bit 48000Hz stream the frame alignment is 128 bytes and maximum mempool
block size value is multiple of 64 but not 128 bytes, therefore resulting parts
are misaligned.
On receiving side this causes extra allocation of 128 byte chunk while `mcalign`
helper reassembles properly aligned frame out of second block of misaligned
size. While first and second properly aligned frames are retrieved successfully
from `mcalign` helper, third retrieved frame would end up with properly aligned
size but misaligned memblock index (in this example, that would be 64 bytes.)
Attempt to push a chunk with misaligned memblock index causes assertion failure
Assertion 'uchunk->index % bq->base == 0' failed at memblockq.c:289,
function pa_memblockq_push(). Aborting.
Fix oversized block issue by checking proper size of format instead of enum
value.
Fixes: a67c21f09 ("merge 'lennart' branch back into trunk.")
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/778>
When tunnel-sink-new was used in combination with module-combine-sink, PA
would hang because the main thread was blocked waiting for the execution
of the latency snapshot message. The message would never be processed
because the rtpoll associated with the control_inq of module-combine-sink
was never run.
This patch fixes the problem by running the rtpoll in the thread function
to process incoming messages. Though there are no users of the rtpoll for
module-tunnel-source-new, the same change is applied there.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/773>
When bluetooth transport has both both sink and source, pulseaudio would
synchronize writing out frames with reading frames from peer to make fair
schedule of reads and writes. Pulseaudio allows two blocks of data to be sent to
peer before synchronizing writes with reads just in case that peer implements
similar write schedule.
It could happen that first blocks are still missed by peer, which would cause
pulseaudio writes to stall waiting for first frames from peer.
Fix this by allowing more data frames out until data from peer is actually
received.
Closes: https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/1424
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/777>
If the same role is named in trigger_roles and cork_roles, a stream with that
role will crash PA. This patch fixes the crash and re-introduces the old
behavior, so that for example specifying trigger_roles=alarm, phone and
cork_roles=alarm, multimedia means that a phone stream will cork alarm and
multimedia streams while an alarm stream will only cork multimedia streams.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/767>
For module-role-ducking, trigger and ducking groups were introduced some years
ago. This patch extends the functionality to module-role-cork, so that trigger
and cork roles may now contain "/" separated groups.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/767>
Incoming RFCOMM string has extra end-of-command terminating character which
breaks both AT+BIA= and AT+BAC= parsers which only expect a comma.
This leads to error parsing last element of response in both cases and could
prevent detecting mSBC availability if mSBC codec id comes last, e.g. AT+BIA=1,2
Fix this by additionally checking for delimiters in both parsers.
Fixes: 3c63f8e6d ("backend-native: Fix stack corruption reading RFCOMM AT+BIA= response")
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/768>
This fixes the rare case of resume_time being bigger than time_stamp. Which
happens sometimes when a gstreamer client is quickly seeking through a
media file. The resulting integer underflow then causes a huge value in
current_time which will break the playback.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/764>
* Enable macOS specific modules (module-bonjour-publish,
module-coreaudio-detect and module-coreaudio-device)
* Correctly set `PA_SOEXT` (.so, .dylib and .dll)
* Build `poll-posix.c` and `semaphore-osx.c`
* Drop linker flag `-Wl,-z,nodelete` on Darwin
* Drop linker flag `-Wl,--no-undefined` on Darwin
* Prefer to `clock_gettime` over compat impl for old Darwin
* Disable SCM credential on Darwin
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/746>
Rate adjustment timer is set up when combine sink is resumed and relased when
combine sink is suspended. Do not create this timer again while module is loaded
to prevent duplicate effort causing assertion in time_callback.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/756>
Faststream backchannel decoder does not know whether incoming stream is mono or
stereo before first packet is decoded, and some devices return stereo stream.
As it is not easy to change source sample spec after source is created, use
stereo sample spec always and perform conversion if mono stream is found.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/628>
When a stream is started but has not yet called smoother_2_put(), pa_smoother_2_get()
returns the time since the start of the stream even if the stream was started paused.
When the stream is started paused, pa_smoother_2_get() should return 0 instead. This
patch fixes the problem.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/745>
When `indicator` is initialized to `1`:
- it always succeeds the `indicator == CIND_CALL_INDICATOR` check;
- hence always calls `continue`;
- hence never reaches the end of the `while` loop where `indicator++` is
called;
- hence `indicator` never contains any other value than `1` meaning
`cind_enabled_indicators` is ever updated.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/740>
This reverts commit b05e34e092.
Now that backend-native uses a different way to get to its own
`native_backend` instance - without going through
`pa_bluetooth_discovery` - this patch can be reverted again, as nothing
outside bluez5-util is supposed to know the internals of this struct.
That's what the many functions are for which all take pointers to this
(at that point) opaque struct instead.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/740>
This removes the inverse/recursive dependency of backend-native on the
`pa_bluetooth_discovery` struct, which is supposed to be opaque outside
of `bluez5-util` in favour of the many accessor functions defined in
`bluez5-util.h`.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/740>
Fix the following build failure without C++:
../output-1/build/pulseaudio-16.1/meson.build:1:0: ERROR: Unknown compiler(s): [['/home/autobuild/autobuild/instance-1/output-1/per-package/pulseaudio/host/bin/powerpc64-buildroot-linux-gnu-g++']]
The following exception(s) were encountered:
Running "/home/autobuild/autobuild/instance-1/output-1/per-package/pulseaudio/host/bin/powerpc64-buildroot-linux-gnu-g++ --version" gave "[Errno 2] No such file or directory: '/home/autobuild/autobuild/instance-1/output-1/per-package/pulseaudio/host/bin/powerpc64-buildroot-linux-gnu-g++'"
Fixes:
- http://autobuild.buildroot.org/results/6526a21bd4da3b8458188f27c1ec04c381e4b673
Signed-off-by: Fabrice Fontaine <fontaine.fabrice@gmail.com>
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/737>
AT+BIA is used to enable/disable CIND indicators by Bluetooth HFP spec.
By default, all indicators are enabled on connection.
AT+BIA will configure which indicators should be disabled then,
the disabled indicators may be enabled later on again with AT+BIA.
When the connection is lost and recovered, all indicators are enabled
again. The HF will reconfigure the indicators again with an AT+BIA
command.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/631>
Add libatomic_ops dependencies to libraries/modules that showed a
failure on an arch that does not have native atomic operations support.
Not all optional dependencies were tested, so it is possible that
some optional modules are still missing libatomic_ops dependencies.
Signed-off-by: Nicolas Cavallari <nicolas.cavallari@green-communications.fr>
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/732>
Attempting to use atomics operations on an architecture that does not
support them generally results in a link error:
ld: /tmp/ccjYcMPP.o: in function `func':
testfile.c:(.text+0x1c): undefined reference to `__sync_bool_compare_and_swap_4'
The current build system uses cc.compiles() to check if atomic ops are
supported, but cc.compiles() does not attempt to link, so the test fails
to enable libatomics_opts.
Fix this by using cc.links() instead of cc.compiles().
Signed-off-by: Nicolas Cavallari <nicolas.cavallari@green-communications.fr>
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/732>
The full identifier check must be executed for the new melem
creation, otherwise the duplicate control element code check
is reached.
Example (using the snd-aloop driver):
numid=56,iface=PCM,name='PCM Notify',device=1,subdevice=1
numid=62,iface=PCM,name='PCM Notify',device=1,subdevice=2
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/730>
The new helem must be tracked and old helem must be cleared
to make the code work properly. Introduce the pointer to helem
as the private value for melem and add the necessary code.
Also, add a check for the duplicate mixer elements. The duplicate
mixer element invokes the abort check in alsa-lib. Print a warning
instead and handle the exit gracefully.
Fixes: def8eb074 ("alsa-mixer: allow to re-attach the mixer control element")
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/730>
Right now we try to add all UCM devices of a verb to a single profile.
But if some devices using different PCMs are configured as conflicting
with one another, we will only be able to utilize one of them, chosen
seemingly based on the order in the UCM config file.
This is not a problem with conflicting devices sharing a PCM, as they
are assigned to the same mapping and the ports mechanism only enables
one of them to be active at a time.
To utilize all devices in a UCM verb even when there are conflicting
devices using different PCMs, calculate subsets of devices which
can be simultaneously used and create a profile for each such set.
Signed-off-by: Alper Nebi Yasak <alpernebiyasak@gmail.com>
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/596>
While switching profiles, it was enough to switch UCM verbs since that
disables all enabled UCM devices and every profile had a distinct verb.
However, switching to the current verb does not disable any devices.
To support multiple profiles for a verb we need to explicitly disable
the old profile's devices, since they might be conflicting with the new
profile's devices and will prevent them from being enabled. Compare both
profiles' mappings, and disable the devices not in the new mappings.
Signed-off-by: Alper Nebi Yasak <alpernebiyasak@gmail.com>
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/596>
After previous patches, we should be generating no combination ports, so
we don't need to store multiple modifiers per mapping. Simplify the code
based on this.
Signed-off-by: Alper Nebi Yasak <alpernebiyasak@gmail.com>
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/596>
After previous patches, we should be generating no combination ports, so
we don't need to store multiple devices per mapping. Simplify the code
based on this.
Signed-off-by: Alper Nebi Yasak <alpernebiyasak@gmail.com>
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/596>
After previous patches, we should be generating no combination ports, so
we don't need to store multiple devices per port. Simplify the code
based on this.
Signed-off-by: Alper Nebi Yasak <alpernebiyasak@gmail.com>
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/596>
A previous commit makes mapping names depend on the UCM device name.
Since UCM device names are unique, this means a mapping will at most
have one port and thus no combination ports can be generated.
This removes the dead code in the pa_alsa_ucm_add_ports_combination()
function, unrolls the remaining code in its helper functions that it
used, and renames it to pa_alsa_ucm_add_port() to signal that it no
longer generates combinations.
Signed-off-by: Alper Nebi Yasak <alpernebiyasak@gmail.com>
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/596>
PulseAudio combines UCM devices that have the same PlaybackPCM or
CapturePCM value into a single mapping with multiple ports. It also
creates ports in the same mapping for each valid combination of those
UCM devices.
Since mappings are the things we put in profiles, we can put in a
profile either all devices of a joint mapping or none of them. This
causes some complications with device conflicts. For example, a
different UCM device might be marked as conflicting with some (but not
all) of the devices in a joint mapping. In this case we can do one of
three things:
- Include all devices in one profile, and hope the conflicting device
isn't chosen as the mapping's active port. We shouldn't do this as it
puts conflicting devices in the same profile.
- Make one profile with the joint group, and one with the other device.
This is somewhat acceptable as we have no conflicts, but we sacrifice
some compatible combinations of devices.
- Do not group the devices into the same mapping, and make one profile
for each compatible combination of devices. This appears to be the
best option, one where we can always have the maximum number of
working devices.
This patch chooses the third option and makes one input and/or output
mapping per UCM device, by using UCM device names instead of PCM device
strings in the mapping names.
Signed-off-by: Alper Nebi Yasak <alpernebiyasak@gmail.com>
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/596>
Combination port logic calculates some useful properties for device
groups that we could reuse while generating multiple profiles to support
conflicting devices. Split them into their own functions.
Signed-off-by: Alper Nebi Yasak <alpernebiyasak@gmail.com>
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/596>
Right now this check is rejecting devices whose UCM config specifies
neither a conflicting device nor a supported device list, and accepting
devices which specify both. However, a device without neither list is
actually unrestricted, and a device with both lists is a configuration
error. Fix the check to accept the former.
Furthermore, this is missing another case where an already selected
device might have a supported devices list that doesn't have the
candidate device. Make this function also check against that, and also
make it accept devices already in the set.
Signed-off-by: Alper Nebi Yasak <alpernebiyasak@gmail.com>
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/596>
The existing code meant to generate device groups for combination ports
is tightly coupled to port creation. Similar functionality would be
useful to generate nonconflicting device groups for multiple profiles as
well, so this tries to rewrite it into a more reusable state.
Several things (e.g devices, mapping contexts) use idxsets to store a
device selection. This also switches this conformance check and device
group generation to using idxsets to make it easier to work with those,
with the eventual aim to unify device group representations.
Also try to adjust users of these functions to use idxsets these will
need/return, without causing too much interference.
Signed-off-by: Alper Nebi Yasak <alpernebiyasak@gmail.com>
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/596>
To support having multiple profiles per UCM verb, split the profile
creation into two parts based on whether they should run once for each
verb or for each profile (maybe multiple times per verb).
Signed-off-by: Alper Nebi Yasak <alpernebiyasak@gmail.com>
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/596>
The ucm_create_mapping() function is not idempotent. It looks like it
was meant to be called once per device for the devices of a UCM verb
and takes a profile argument simply because a verb has generated a
single profile so far.
Make sure creating mappings per device and adding those mappings to the
profiles happens as separate steps to make it easier to split UCM verbs
and profiles as concepts.
Signed-off-by: Alper Nebi Yasak <alpernebiyasak@gmail.com>
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/596>
So far each profile had the exact name as their associated UCM verb,
which caused the one to be used where the other should have been.
Explicitly get and use the verb name where that was intended, and make
sure things about profiles aren't named after verbs.
Signed-off-by: Alper Nebi Yasak <alpernebiyasak@gmail.com>
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/596>
Currently each UCM verb generates one profile named the same as the
verb, meaning it's trivial to know which verb the profile belongs to.
This will be slightly harder to do when we generate multiple profiles
per UCM verb (e.g. to make use of conflicting devices).
It would still be possible to parse the profile name to get the UCM
verb, but instead let's keep track of the struct instance representing
the profile's associated verb. This also lets us remove a block of code
searching for the verb by its name.
Co-authored-by: Jaroslav Kysela <perex@perex.cz>
[Alper: Reused Jaroslav's UCM profile context changes for UCM verb
instead of combined devices.]
Signed-off-by: Alper Nebi Yasak <alpernebiyasak@gmail.com>
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/596>
While switching profiles, it's possible that we will want to do more
work besides switching UCM verbs. The alsa-card module already has our
profiles as structs, but passes in only the names instead of the entire
struct. Make things work with the struct instead, so we can add other
things (like a UCM context) to it and use those here.
Co-authored-by: Tanu Kaskinen <tanuk@iki.fi>
[Alper: Split into its own commit and integrated Tanu's snippet.]
Signed-off-by: Alper Nebi Yasak <alpernebiyasak@gmail.com>
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/596>
Right now manipulating device status is done inline once while setting a
port. However, we will need to reuse this code to disable conflicting
devices of a device we want to enable. Split it into enable and disable
helper functions.
There is another issue with the device enable logic, where trying to
disabling an already disabled device sometimes fails. To avoid that,
implement a status helper and check if the device we want to enable is
already enabled/disabled before trying to do so.
Signed-off-by: Alper Nebi Yasak <alpernebiyasak@gmail.com>
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/596>
Modifiers currently keep their conflicting and supported devices's
names, and these names are resolved to devices every time we need to use
them. Instead, resolve these device names while creating the modifier
struct and keep track of the resulting device structs in idxsets, same
as how device structs keep track of their support relations.
Signed-off-by: Alper Nebi Yasak <alpernebiyasak@gmail.com>
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/596>
This is intended to make the current and upcoming code a bit clearer, as
we won't need to constantly check for the existence of these idxsets
before using or operating on them.
Signed-off-by: Alper Nebi Yasak <alpernebiyasak@gmail.com>
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/596>
Add complementary functions to the existing idxset iterate(),
steal_first(), first(), next() functions that work in the reverse
direction: reverse_iterate(), steal_last(), last() and previous().
Signed-off-by: Alper Nebi Yasak <alpernebiyasak@gmail.com>
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/596>
This is functionally equivalent to get_by_data(s, p, NULL) == p, but
with a more obvious name and form because some existing code is instead
manually iterating through idxsets to check for existence of an item.
Signed-off-by: Alper Nebi Yasak <alpernebiyasak@gmail.com>
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/596>
It may be possible that the ALSA control element appears
again. Allow this combination by checking, if the pulseaudio
mixer element already exists. Do not create the duplicate
mixer element in this case.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/729>
PulseAudio v5.99 or later hits assertion at alsa-lib mixer API due to
wrong handling of removal event for mixer element.
pulseaudio: mixer.c:149: hctl_elem_event_handler: Assertion `bag_empty(bag)' failed.
The removal event is defined as '~0U', thus it's not distinguished from
the other type of event just by bitwise operator.
At the removal event, class implementator for mixer API should detach
mixer element from hcontrol element in callback handler since alsa-lib
has assertion to check the list of mixer elements for a hcontrol element
is empty or not after calling all of handlers. In detail, please refer to
MR to alsa-lib:
* https://github.com/alsa-project/alsa-lib/pull/244
This commit fixes the above two issues. The issue can be regenerated by
`samples/ctl` Python 3 script of alsa-gobject.
* https://github.com/alsa-project/alsa-gobject/
It adds some user-defined elements into sound card 0. When terminated by
SIGINT signal, it removes the elements. Then PulseAudio dies due to the
assertion.
Fixes: 1fd8848e64 ("alsa-util: Add functions for accessing mixer elements through mixer class")
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/728>
GST_TYPE_BITMASK is 64-bit bit mask while corresponding channel_mask in
pulseaudio is int therefore usually 32-bit. Switch to uint64_t instead
to match internal representation in gstreamer.
Fixes pulseaudio crash on ARM 32-bit when pulseaudio is compiled with
gstreamer and either LDAC or aptX support is available.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/723>
A recent commit added i->origin sink for the sink inputs of the combine sinks.
Therefore pa_sink_process_input_underruns() treated the combine sink like
filter sinks. pa_sink_process_input_underruns() calls itself with the
origin sink, which is only correct for filter sinks because they run in the
thread context of the origin sink. The combine sink however has its own
thread context, so pa_sink_process_input_underruns() was executed in the
wrong context.
This patch fixes the issue by skipping the section for module-combine-sink.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/722>
Default build configuration would fail to run on a system without systemd-logind
(or elogind) and without ConsoleKit daemon responding on dbus interface. Here,
module-console-kit would fail to initialize, preventing daemon from starting.
Make module-console-kit an optional build feature to allow opt-out.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/719>
On FreeBSD (and probably other BSDs as well), the FIONREAD ioctl
on UDP sockets does not return the size of the next datagram (like
it does on Linux), but returns the size of the output buffer: this
count contain multiple datagrams and also contains the headers.
We fixed this by taking the result of the FIONREAD as lower bound
for the size, adding an upper bound and then removing the check
that the sizes should be exactly the same.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/718>
Turned out that pa_sdp_info::enable_opus is never initialized, which seldom
makes module-rtp-recv believe it will be playing OPUS-encoded stream even though
discovered SDP record does not indicate OPUS codec in metadata.
Fix this by adding missing initializer.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/720>
pulseaudio crash occurred when I play a file using pacmd play-file command.
The file is not aligned with its frame size and the last rendering size
is also not aligned. Thus, an assertion was generated at the end of the
file as the following.
memblockq.c: Assertion 'uchunk->length % bq->base == 0' failed at
../src/pulsecore/memblockq.c:288, function pa_memblockq_push(). Aborting.
When I play the file using paplay, it works good. So, I changed to
pa_memblockq_push_align instead of pa_memblockq_push to prevent the
assertion.
Signed-off-by: Jaechul Lee <jcsing.lee@samsung.com>
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/717>
The logic for detecting which type of volume was given incorrectly interpreted
any value with a decimal as a VOL_LINEAR. It also could set multiple flags,
which would put the flags variable into an indeterminate state. Additionally,
the flags stack variable was uninitialized which could also lead to an
indeterminate flag state.
Percentages are now prioritized over all other types, and only one type flag
can be set.
RFC 4566 states that SDP record is terminated with CRLF, and parsers should be
able to accept records terminated with just LF. Pulseaudio only accepts LF here.
Fix this by accepting both CRLF and LF terminators.
The combine sink used the current time and counter when calculating
the latency if smoother_2 was enabled. This lead to wrong latency
reports. This patch fixes the problem by using the snapshot time
and counter instead.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/711>
Bluetooth transport layer already allows for packets larger than mSBC frame, and
there are up to 1 + MTU / (mSBC packet size) complete frames to be decoded from
each incoming SCO packet.
Now decoder fails when there is more than one complete frame available, which
could happen if MTU size is larger than 1.5 * (mSBC packet size) = 90
Fix this by adding a loop over avialable frames, and adjust decoded buffer size
to allow decoding up to 1 + MTU / (mSBC packet size) frames at once.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/706>
When the module is loaded and avahi_client_new() fails because the client cannot
connect, a shutdown of the module is scheduled. In parallel, the client_callback
is called with AVAHI_ERR_DISCONNECTED and another connection attempt is made
which also fails and triggers a second unload of the module. This crashes PA,
because there is already an unload in progress.
This patch fixes the problem by checking if an unload is already scheduled.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/710>
The timestamp used for updating the smoother was taken at the wrong time.
It may take some time until an async message is executed (measured up to
2ms), therefore the timestamp used to update the smoother must be taken
before the message is executed and not inside the message.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/705>
When the tunnel modules had no connection and a re-init was pending, the module
could be unloaded without cancelling the pending re-init. When the timer expired
in that situation, this lead to a crash. This patch fixes the problem by keeping
a reference when the module is scheduled to be re-initialized.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/705>
The AVRCP service is known to not be connected before the A2DP transport
is, resulting in PulseAudio asking BlueZ for an initial 'Volume' value
but not getting it because the property doesn't exist.
To prevent end-users from conjecturing this to be the source of whatever
issue they're observing, demote it to a warning.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/707>
When configured, reinitialize the module instead of exiting. This
allows a restart/reconnect, but the module to appear to always be alive
when the user does: "pactl list modules". (The sink will still not
exist until the tcp connection is established.)
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/688>
The io thread, after connection, sends a message asking for a sink to be
created. After the ctl thread is done with creation, it sends a message
back to the io thread so it can continue. This ensures that the sink
only exists when it's connected to something.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/688>
When the --format json parameter is given on the command line, we
attempt to produce a JSON output for most commands.
Our implementation of the JSON serialization uses vsnprintf to output
numbers. Unfortunately, vsnprintf is affected by the locale and more
specifically the LC_NUMERIC variable.
When LC_NUMERIC is set to, for instance, fr_FR.UTF-8, floating-point
numbers are output with a comma as the decimal separator, which is then
considered invalid JSON.
$ LC_NUMERIC=fr_FR.UTF-8 pactl --format json list sinks | jq .
parse error: Objects must consist of key:value pairs at line 1, column 435
This is the token which failed to parse:
}},"balance":0,00,"base_volume":{
Fixed by overriding the LC_NUMERIC value when we request JSON output.
Signed-off-by: Olivier Gayot <olivier.gayot@sigexec.com>
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/702>
When monitor source becomes idle it may happen that monitored sink has no
uncorked inputs anymore and can now be suspended. To allow this, detect if state
is changed for monitor source and check state of monitored sink instead.
This change allows pulseaudio to suspend devices when pavucontrol volume meters
are disabled and corresponding peaks resampled streams are corked.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/697>
Turned out that SelectConfiguration is only used for outgoing connections, and
incoming connection from bluetooth headset using SBC codec ends up with a
bitpool as large as declared by headset. When resulting bitpool is so large that
SBC frame size plus RTP header size exceeds write MTU size, number of frames per
packet becomes zero causing crash dividing by zero in update_sink_buffer_size()
Fix this by limiting available bitpool value exposed for SBC endpoints.
Fixes: 89082cbfa ("bluetooth: a2dp dual channel SBC XQ codec configurations")
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/695>
Commit c6d6ca541 ("bluetooth/gst: Replace buffer accumulation in adapter
with direct pull") removed the `timestamp` parameter from GStreamer
transcoders due to being unused, but these should instead be propagated
to the GStreamer encoding buffers.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/494>
Bluetooth codecs should always have fixed in/output and are hence able
to have their results directly read from the codec, instead of
accumulating in a buffer asynchronously that is subsequently only read
in the transcode callback. The Bluetooth backends calling encode/decode
also expect these fixed buffer sizes.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/494>
Handling multiple threads does not come without overhead, especially
when the end-goal is to ping-pong them making the whole system run
serially. This patch rips out all that thread handling and instead
"chains" buffers to be encoded/decoded directly into the pipeline,
making them execute their work on the current thread. The resulting
buffer can be pulled out from appsink immediately without require extra
locking and signalling. While the overhead on modern systems is found
to be negligible or unnoticable, code complexity of such locking and
signalling systems is prevalent making it the main drive behind this
refactor.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/494>
Drop rtpldacpay and payload the LDAC encoded output manually in the
RTP header.
The RTP payload seems to be required as it carries the frame count
information. Right now, rtpldacpay does not add this so construct
the RTP header and payload manually.
Strangely some devices like Shanling MP4 and Sony XM3 would still
work without this while some like the Sony XM4 does not.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/689>
If UCM defines the private alsa-lib configuration, the ELD controls
are expected to use this device configuration too.
With this change:
I: [pulseaudio] alsa-util.c: Successfully attached to mixer '_ucm0009.hw:Loopback'
Without:
I: [pulseaudio] alsa-util.c: Successfully attached to mixer '_ucm0009.hw:Loopback'
I: [pulseaudio] alsa-util.c: Successfully attached to mixer 'hw:4'
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/673>
The hw: device can be addressed using the card index (hw:0)
or the card identifier (ASCII string - hw:Loopback). Both
mixers are equal.
The previous code was fine for the mixers without the UCM
private prefixes (_ucmXXXX). Make code more robust, create
two aliased mixer structures in the mixers array.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/673>
This makes it possible to define multiple sinks/sources on detection
of the jack server. This allows one to for example create a separate
sink for conferencing software and route that in jack to another
channel on their audio interface.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/669>
Even though the file name is currently behringer-umc22.conf, the USB ID
actually belongs to Texas Instruments PCM2902, which is a generic chip
used in multiple products. Some products have true mono input unlike
Behringer UMC22, which has two mono inputs combined into one stereo PCM
device.
This patch removes the "mono,mono" mapping from Behringer UMC22, which
hopefully won't be missed too much (there are still "mono,aux1" and
"aux1,mono" mappings available for mono recording).
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/667>
If the preferred ports are not set in this function, the
entrys_equal() always returns false in the card_put_hook_callback().
This will make the entry be written into the metadata and the
preferred ports will be cleaned by a mistake.
And we met a hdmi audio bug which has sth to do with this issue, on
the machines with the legacy HDA audio driver, the hdmi port has lower
priority than speaker, users need to manually select the hdmi to be
active output port, then the preferred output port is hdmi for this
sound card, after reboot, the card_put_hook_callback() in the
module-card-restore.c will be called and the preferred ports are
cleaned by a mistake, then the hdmi output port or hdmi sink couldn't
switch to be active after reboot or resume automatically. That is
because the preferred ports are cleaned and hdmi port has lower
priority than speaker, the profile_good_for_output() in the
module-switch-on-port-available.c always returns false.
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Change d7f95170a1 added a dependency on device
adapter pointer being valid while checking if bluetooth profile is supported by
device.
When adapter object is released, each device holding pointer to adapter being
released is notified to reset that to NULL. Since adapter objects are released
first when discovery object is unreferenced, each device will have adapter
pointer reset before the time device objects are released.
Fix observed crash by examining device adapter pointer. If it is NULL report
that device does not support any bluetooth profile instead of looking at UUIDs
supported by adapter.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/646>
Add a log_interval parameter to control the amount of logging. Default is
no logging. Like for adjust_time, the parameter is a double to allow values
below 1s.
If the log interval is too small, logging will occur on every iteration.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/56>
The adjust_time parameter is changed to double to allow better granularity
and adjust times below 1s. This may be useful for a better latency control,
although with alsa devices and the current smoother code no significant
improvement could be found for values below 500ms.
This patch also changes the default adjust time to 1s, the old value of 10s
does not allow a tight control of the end to end latency and would lead to
unnecessary jitter.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/56>
The previous patch slows down initial convergence. Therefore do not use
the controller weight until we can assume that we reached an equilibrium.
Because it takes some time before the reported latency values are reliable,
assume that a steady state is reached when the target latency has been
crossed twice.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/56>
In many situations, the P-controller is too sensitive and therefore exhibits rate hunting.
To avoid rate hunting, the sensibility of the controller is set by the new parameter
adjust_threshold_usec. The parameter value is the deviation from the target latency in usec
which is needed to produce a 1 Hz deviation from the optimum sample rate.
The default is set to 250 usec, which should be sufficient in most cases. If the accuracy
of the latency reports is bad and rate hunting is observed, the parameter must be increased,
while it can be lowered to achieve less latency jitter if the latency reports are accurate.
More details at
https://www.freedesktop.org/software/pulseaudio/misc/rate_estimator.odt
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/56>
The current loopback controller can produce a rate jump of up to 1% at startup,
which may be audible. To prevent large initial jumps, a second controller is
introduced, which produces a rate, that is not more than 2‰ away from the last
rate. Only during the startup phase, the rates produced by this controller will
be nearer to the base rate than those produced by the original controller.
Therefore choosing the rate which is nearer to the base rate will ensure that
the secondary controller only moderates the startup phase and has no influence
during continued operation.
The maximum step size of the original controller after the initial jump is
limited to 2.01‰ of the base rate, see documentation at
https://www.freedesktop.org/software/pulseaudio/misc/rate_estimator.odt
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/56>
Currently module-loopback detects underruns even if sink_input_pop_cb()
was not yet called twice and initial latency adjustments are active.
This leads to unnecessary rewind requests.
This patch delays detecting underruns until the initial adjustments
are done.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/56>
Taking addresses of fields in a packed struct are not guaranteed to be
aligned, resulting in warnings such as:
../src/pulsecore/shm.c: In function 'sharedmem_create':
../src/pulsecore/shm.c:198:25: error: taking address of packed member of 'struct shm_marker' may result in an unaligned pointer value [-Werror=address-of-packed-member]
198 | pa_atomic_store(&marker->pid, (int) getpid());
| ^~~~~~~~~~~~
The struct already has its fields and types laid out in such a way that
the desired packing (without padding) is guaranteed - enforce this with
a `static_assert` to get rid of the unaligned pointer warning.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/653>
GCC warns on all these `fail_unless` calls:
warning: too many arguments for format [-Wformat-extra-args]
`fail_unless` only takes an expression and optionally a string literal
as message with formatting args. Passing NULL for this message should
not be necessary as indicated by all the other tests not passing it
either.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/653>
e04f14eb/dc9dc70f introduced preprocessor warnings to deny the use of
any `alsa-lib` older than `1.2.1`, and with a future patch disallowing
warnings entirely through `-Werror` we now need a distribution that
serves a new enough `alsa-lib`.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/653>
This patch adds an alternative time smoother implementation based on the theory
found at https://www.freedesktop.org/software/pulseaudio/misc/rate_estimator.odt.
The functions were written to replace the current smoother functions nearly on
a one-to-one basis, though there are a few differences:
- The smoother_2_put() function takes a byte count instead of a sound card
time as argument. This was changed because in most places a sample count
was converted to a time before passing it to the smoother.
- The smoother needs to know sample rate and frame size to convert byte
counts to time.
- A smoother_2_get_delay() function was added to directly retrieve the stream
delay from the smoother.
- A hack for USB devices was added which works around an issue in the alsa
latency reports for USB devices.
The smoother delivers much better precision than the current implementation.
For results, see the document referenced above.
The new functions are still unused. The following patches will convert all
callers of the smoother functions so that they can use both smoother
implementations, depending on a configure option.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/55>
This patch adds a test program that generates a square wave of a given frequency,
length and sample rate. This is then resampled to another rate, rewound and the
rewound part is run through the resampler again. After that, the results of the
first and second resampler pass are compared.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/120>
This patch is in preparation of allowing virtual sinks to specify their own
max_rewind limit.
Currently pa_sink_set_max_rewind_within_thread() simply sets the value of
max_rewind and informs the sink inputs about the new value. Virtual sinks
may however provide their own limit on max_rewind.
This patch allows to query the active sink inputs for the max_rewind value
they support and sets max_rewind to the minimum supported value. This way,
the max_rewind value from the virtual sinks can be communicated to the master
sink.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/120>
This patch is in preparation of allowing virtual sinks to specify their own
max_rewind limit.
Currently, virtual sinks cannot specify their max_rewind limit, but just copy
the value from the master sink. This may not be correct, if the DSP code of the
virtual sink has limited (or no) rewinding capability.
Because the DSP code of the virtual sink is rewound in the process_rewind()
callback of the sink input, it must be ensured, that rewinding a sink input
to the master of a virtual sink is limited similar to rewinding a sink.
There are two remaining exceptions:
1) If an underrun is detected. In that case, the filter should be reset anyway.
2) When the sink input of the filter is moved and attached to the destination
sink.
The move case is handled without involvement of the implementer, so the implementer
can only receive a rewind larger than max_rewind when the filter should be reset
anyway.
All existing virtual sinks do not distinguish between reset and rewind of the
filter.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/120>
If the output implements a process_rewind() callback, the resampler delay is
not taken into account. This leads to glitches during volume changes when
source and source output rates differ.
This patch fixes the problem.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/120>
The introduction of the history queue makes it possible to implement moving
of streams without involving the implementer. Instead of dropping all data
from the render memblockq and requesting the implementer to rewrite the
data, the render memblockq is now reconstructed from the history queue.
Additionally, the render queue will be filled with silence matching the
amount of audio that is left playing on the old sink to avoid playing
the same audio twice.
This patch slightly breaks moving for virtual sinks because they do not
yet include the resampler delay in their latency reports. This will be
fixed in a different patch set.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/120>
This patch uses the two previous patches to implemnt pseudo-rewinding for the
resamplers by feeding some old data into the resampler after a reset. This is
necessary because PA is using external resamplers that do not implement
rewinding.
To get exactly the same output data from the resampler after a rewind if possible,
the matching period is calculated. This is the number of input samples that produces
an integral number of output samples. After the matching period, the resampler state
repeats. If the matching period is not too large, feeding history into the resampler
will start at a point that is a multiple of the matching period back in time. Then
the resampler will produce exactly the same samples.
The PA_RESAMPLER_MAX_HISTORY value has been replaced by PA_RESAMPLER_MAX_DELAY_USEC
and the required number of history samples is calculated from the sink input sample
rate. The number of history samples can be as large as about 12500.
This fixes glitches during volume changes when the sink runs on a rate different
from the sink input rate.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/120>
The pa_resampler_get_delay() function allows to retrieve the current resampler
delay in input samples for all supported resamplers. The return value is a double
to maintain precision when using variable rate resamplers. Because in many places
the delay is needed in usec, pa_resampler_get_delay_usec() was also supplied.
The speex resampler now skips leading zero samples to provide meaningful delay values.
In the next patch, the pa_resampler_prepare() function will be used to train the
resampler after a rewind. It takes data from a history memblockq and runs it through
the resampler. The output data is discarded.
To make this logic possible, the soxr resampler had to be converted to use variable
rate. The fixed rate version has a variable delay, therefore the logic above could
not be applied. Additionally, with fixed rate, the delay is larger than 150ms in
some situations, while with variable rate the delay is fixed and comparable to the
other resamplers.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/120>
A new memblockq is added to the sink input code to keep some history of the
input data. The queue is kept in sync with the render memblockq. The old input
data will be used to prepare the resampler after a rewind.
pa_resampler_request() and pa_resampler_result() have been changed to round
as good as possible to avoid loosing or duplicating samples during rewinds.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/120>
When sink is suspended for reconfiguration changing sample spec, upon resume
internal thread_info max_request and max_rewind are out of date and possibly
not aligned to frame size anymore.
Recalculate thread max_request and max_rewind before resuming sink.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/658>
When ICE I/O error occurs ICE connection is closed via IceCloseConnection.
This causes crash while releasing session manager connection later because
this ICE connection was initiated and is managed by session manager, and it will
attempt to close this ICE connection again.
Fix this by closing session manager connection instead.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/650>
Commit 7fd89e491 ("bluetooth: Try to reconnect SCO") introduces a call
to pa_msleep but failed to include the header, resulting in a:
../pulseaudio/src/modules/bluetooth/backend-native.c: In function ‘sco_acquire_cb’:
../pulseaudio/src/modules/bluetooth/backend-native.c:336:17: warning: implicit declaration of function ‘pa_msleep’ [-Wimplicit-function-declaration]
336 | pa_msleep(300);
| ^~~~~~~~~
(Un)fortunately this implicit declaration gets resolved at link-time,
otherwise the issue would have been caught sooner.
Fixes: 7fd89e491 ("bluetooth: Try to reconnect SCO")
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/651>
Just like the manpage these written out log levels should correspond to
the numerical values listed before, intead of being in the opposite
order and provoking thoughts of the relation being the wrong way around
where 0=debug and 4=error.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/647>
Requiring user to invoke send-message with correctly quoted parameters string
is not good for usability. Wrap parameters string into JSON string and try
again. If that works, log a warning use wrapped JSON string with parameters.
As an example these two commands will now invoke the same action:
pactl send-message /card/bluez_card... switch-codec '"CODECNAME"'
pactl send-message /card/bluez_card... switch-codec CODECNAME
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/648>
Check whether a Bluetooth profile is supported both by the remote device
and the local host before creating a card profile for it.
This is useful when some of the media profiles have not been registered
with bluetoothd because ex., oFono is not running and the headset
backend is not available.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/638>
The remote device list of UUIDs reflects which profiles are supported by
the remote device alone. We currently rely solely on this list to decide
if a certain card profile is supported, and thus should be created and
get connected.
This used to be accurate when the Bluetooth modules were first written,
but now BlueZ is more dynamic and local profile support can be added or
removed during runtime. The adapter's list of UUIDs is an accurate
representation of the profiles supported by the local host at a certain
moment.
This commit combines the list of UUIDs supported by remote device and
the list of UUIDs supported by the local host to determined whether a
Bluetooth profile is actually supported or not, and whether it should be
expected to get connected during device connection.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/638>
When the SCO connection is in use, if you disconnect first and then connect,
the SCO connection will occasionally fail, and the Bluetooth error code is 42
(0x2A in hexadecimal). This is usually because an error occurred when the SCO
connection was initiated, we need to try to reconnect to optimize the handling
of this problem. The log returned by the kernel is as follows:
Bluetooth: sco_connect_cfm: hcon 0000000003328902 bdaddr 40:ef:4c:0c:11:f0 status 42
Bluetooth: sco_sock_connect status is -38
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/622>
When module-combine-sink is used to create virtual surround card it is expected
that each slave channel receives data for exactly one combined sink channel.
Currently this is not the case because module-combine-sink would follow
core->disable_remixing setting. Usually this means that multiple channels of
combined sink are getting remixed into slave channel, and there is no option to
disable this behavior just for combined sink.
Improve this by implementing "remix" modarg for module-combine-sink,
default to original behavior (follow core->disable_remixing setting).
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/627>
Mainloop test uses io callback for PA_IO_EVENT_INPUT on stdin.
With glib enabled PA_IO_EVENT_INPUT translates to glib G_IO_IN event which also
matches descriptor in EOF state. While io callback does not check for EOF after
reading from file descriptor this is causing mainloop-test to repeatedly read 0
bytes once EOF is reached, rearm defer callback and spam test log.
Fix this by disarming io callback when EOF is reached in test run.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/625>
The word variables used in the _pactl and _pacmd functions are
unlocalized. Thus, the variable appears in the user's environment when
tab-completing with pactl or pacmd. This may clobber another variable
of the same name, which is undesirable.
Localize the word variable to fix this issue.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/623>
When packaging a new version for OpenEmbedded, I use the
buildhistory-diff tool to check what changed between the versions. The
version number in the module directory means that I get tons of diff
output due to changes in file paths. There are many removed and added
files and it's hard to see if something else than just the version
number changed.
That motivated me to write this patch. Removing the version number has
the downside that it makes it easier to have version mismatches between
the daemon and the modules, but
https://gitlab.freedesktop.org/pulseaudio/pulseaudio/merge_requests/249
will make the handling of such situations better.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/230>
Currently, module-tunnel uses the default fixed latency of 250ms as fixed
latency.
There is no reason for such a large latency. This patch adds a parameter
latency_msec to the module to set the fixed latency at load time of the
module. The parameter can range from 5 to 500 milliseconds. With this
patch, I was able to run a tunnel sink at 7ms latency without problems.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/53>
Currently module-tunnel uses only a rough estimate of the current stream
latency and reports wrong latencies in certain situations. This leads to
very inexact and unstable latency reports for the virtual sink.
This patch fixes the issue by introducing latency snapshots like they
are used in module-loopback. Because the latency reports are now correct,
the update interval for latency re-calculations can be reduced to 1s.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/53>
Currently the combine-sink uses the trivial resampler by default.
This patch changes the default to the configured resampler.
Also the default update time is changed from 10s to 1s to achieve
faster convergence and higher precision.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/53>
Currently, it takes one adjust time before the smoother is updated after an
unsuspend. Before the first update, the smoother will not be aware of the
slave sink latencies, leading to incorrect latency reports.
This patch moves the first smoother update to one latency time after the
sink was unsuspended, thereby improving initial latency reports. This
only partially resolves the problem because the smoother takes multiple
updates to adapt to the slave sink latencies.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/53>
This patch adds a rate controller similar to the one used in module-loopback
to limit step size and maximum deviation from the base rate. Rate changes
are handled more smoothly by the controller. The patch has not much impact
on the behavior of the module, except that there is less rate hunting.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/53>
Currently module-combine-sink uses only a rough estimate of the current
slave sink latencies to calculate the rate for the various sink inputs.
This leads to very inexact and unstable latency reports for the virtual
sink.
This patch fixes the issue by introducing latency snapshots like they
are used in module-loopback. It also changes the definition of the
target latency to ensure that there is always one sink which uses the
base rate.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/53>
Commit f89d64b98e fixed a crash
when disabling adapters. However, now if any device is removed
ofono card is removed, even if it belongs to different device.
Add a check for the device being unlinked to our callback to fix.
Signed-off-by: Juho Hämäläinen <juho.hamalainen@jolla.com>
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/624>
With USB Alternate Setting 3 size of HCI payload is 72 bytes which is already
larger than mSBC frame size. Largest known size of HCI payload is with USB
Alternate Setting 5 (144 bytes), make it the default SCO socket MTU.
Reserve additional space in bluetooth encoder buffer to cover this case.
Since mSBC encoder and decoder will now work with larger packet sizes, drop
assertions about MTU larger than mSBC frame size.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/617>
While the threading model for combine is different from other filters
(which expect to just piggy-back on the I/O thread of the most
downstream sink), it might still be valuable to set this field to
indicate that this sink input is intended to behave as a filter stream
rather than a conventional stream.
At the very least, routing behaviour and cycle detection should act on
these streams as with any other filter.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/399>
With clang compiler including cpuid.h will produce error if architecture is not
x86-based, and cheching if cpuid.h exists via Meson has_header() is not enough.
Fix this by creating a list of headers checked to be usable via Meson
check_header() function, and move cpuid.h to that list.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/614>
This reverts commit 3ac73598c6.
Systemd v249 has new entries of hwdb for node and unit in IEEE 1394 bus
(hwdb.d/80-ieee1394-unit-function.hwdb). It can obsolete my workaround
added by commit 3ac73598c6 ("udev: use ID_MODEL/ID_VENDOR to give
friendly name for FireWire devices"). The hwdb entry is handy prepared.
When user finds missing entry, it's preferable to file issue or merge
request in systemd project site.
IEEE 1394 bus is enough legacy and it's easy to expect that few developer
can evaluate the change. For reviewers, I describe the original issues and
the integration of hwdb in systemd side.
In systemd, udev rule for sound card (rules.d/78-sound-card.rules) has
below line to assign information in hwdb to instance for sound card.
```
IMPORT{builtin}="hwdb"
```
In the case, the udev hwdb builtin finds information according to
modalias by following nodes in device topology tree toward root. For
sound card associated to unit in node in IEEE 1394 bus, it's inconvenient
since hwdb had no entry for the unit. The instance for node in IEEE 1394
bus doesn't have modalias. As a result, the builtin reaches 1394 OHCI
controller in PCI Express bus which maintains the IEEE 1394 bus, then the
value for ID_VENDOR_FROM_DATABASE and ID_MODEL_FROM_DATABASE properties
from hwdb of pci device (hwdb.d/20-pci-vendor-model.hwdb) for the sound
card.
For example, when two nodes are in IEEE 1394 bus and one of them has
unit instance for audio and music functions, the topology of the bus is
depicted in following diagram:
```
* 1394 OHCI controller (pci*, modalias)
* node A - (pci*/fw0, /dev/fw0)
* node B - (pci*/fw1, /dev/fw1)
* unit B-1 - (pci*/fw1/fw1.0, modalias)
* sound card 0 - (pci*/fw1/fw1.0/sound/card0, card0)
```
In the case, the udev hwdb builtin picks up from hwdb of pci device for
the sound card:
```
$ udevadm test-builtin hwdb /sys/class/sound/card2
Load module index
Parsed configuration file /usr/lib/systemd/network/99-default.link
Parsed configuration file /usr/lib/systemd/network/73-usb-net-by-mac.link
Created link configuration context.
ID_PCI_CLASS_FROM_DATABASE=Serial bus controller
ID_PCI_SUBCLASS_FROM_DATABASE=FireWire (IEEE 1394)
ID_PCI_INTERFACE_FROM_DATABASE=OHCI
ID_VENDOR_FROM_DATABASE=Texas Instruments
ID_MODEL_FROM_DATABASE=XIO2213A/B/XIO2221 IEEE-1394b OHCI Controller [Cheetah Express]
Unload module index
Unloaded link configuration context.
```
The aim of my workaround is to avoid using ID_VENDOR_FROM_DATABASE and
ID_MODEL_FROM_DATABASE for sound card associated to unit in IEEE 1394
bus. Instead, ID_VENDOR and ID_MODEL properties are used.
However, it has another issue. For the properties, the udev rule for
sound card has the other lines for sound card associated to unit in
IEEE 1394 bus, below:
```
SUBSYSTEMS=="firewire", ATTRS{guid}=="?*", \
ENV{ID_BUS}="firewire", ENV{ID_SERIAL}="$attr{guid}", ENV{ID_SERIAL_SHORT}="$attr{guid}", \
ENV{ID_VENDOR_ID}="$attr{vendor}", ENV{ID_MODEL_ID}="$attr{model}", \
ENV{ID_VENDOR}="$attr{vendor_name}", ENV{ID_MODEL}="$attr{model_name}"
SUBSYSTEMS=="firewire", GOTO="skip_pci"
```
The values of ID_VENDOR and ID_MODEL properties come from vendor_name and
model_name attributes in parent instance of the sound card, therefore
they come from audio and music units in IEEE 1394 bus. Unfortunately
these attributes are not available always.
All of nodes in IEEE 1394 bus should have configuration ROM in place
according to IEEE 1212 and Linux FireWire subsystem parses the content of
ROM to detect units in the node. At the same time, the subsystem manages
to detect information about vendor and model according to standard layout
defined by 1394 Trading Association[1].
When the content of ROM is against the standard, the subsystem is
discouraged the name detection. In the case, vendor_name and model_name
attributes are not available, and supplemental information should be from
software implementation.
The new hwdb (hwdb.d/80-ieee1394-unit-function.hwdb) added to systemd v249
can solve the above issues. The prepared names for vendor and model in
hwdb are assigned to both node and unit. The udev hwdb builtin can find
the vendor and model names for the unit according to modalias before
arriving at pci-device. Regardless of standard or non-standard
configuration ROM, the hwdb gives prepared names of vendor and model.
This is an example of Mark of the Unicorn (MOTU) Traveler. The search
finishes at instance for unit in IEEE 1394 bus expectedly:
```
$ udevadm test-builtin hwdb /sys/class/sound/card2
Load module index
Parsed configuration file /usr/lib/systemd/network/99-default.link
Parsed configuration file /usr/lib/systemd/network/73-usb-net-by-mac.link
Created link configuration context.
ID_MODEL_FROM_DATABASE=Traveler
ID_VENDOR_FROM_DATABASE=MOTU
IEEE1394_UNIT_FUNCTION_AUDIO=1
IEEE1394_UNIT_FUNCTION_MIDI=1
Unload module index
Unloaded link configuration context.
```
[1] Configuration ROM for AV/C Devices 1.0 (Dec. 12, 2000, 1394 Trading
Association, TA Document 1999027)
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/610>
Try to register profile support again after RegisterProfile fails, when
BlueZ indicates no one else is implementing the profiles we are
interested in.
Ideally this would rely on a list of UUIDs supported by the profile
manager instead of the adapter, but BlueZ has no such API.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/593>
Create pa_bluetooth_profile_status_t to represent all stages an external
Bluetooth profile can go through:
0. Inactive: Initial state, no D-Bus object has been registered for
this profile yet.
1. Active: an object implementing the org.bluez.Profile1 interface has
been registered on the system bus.
2. Registering: RegisterProfile has been called.
3. Registered: RegisterProfile succeeded.
This will be useful to handle RegisterProfile failures, as well as
dynamically register and un-register a profile based on the current
active seat.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/593>
HF indicator 2 (see [assigned-numbers], Hands-Free Profile) is able to
report battery percentage at 1% intervals (in range [0, 100]), contrary
to the `+XAPL` `+IPHONEACCEV` extension which only supports 10%
increments. This does not guarantee increased granularity however, as
peers may still be limited to imprecise battery measurements internally
or round to coarser percentages.
Supporting both additionally broadens the range of devices for which PA
can report its battery level.
[assigned-numbers]: https://www.bluetooth.com/specifications/assigned-numbers/
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/482>
Whenever a device disconnects the device is not removed from BlueZ, only
the profiles that had an active connection are disconnected. Since we
were providing this battery level based on AT commands received through
HSP/HFP these services should be responsible for deregistering it again.
Deregister the interface to signal BlueZ (And UPower in return) that the
battery level won't be accurate/updated anymore.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/482>
The peer will wait some time and eventually time out the connection if
no reply is sent back. When sending `ERROR` the peer can decide to break
the RFCOMM connection immediately or continue when a command is not
critical.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/482>
Devices for Apple's iOS uses a few extra HFP AT commands to
inform the iPhone about the headphone's battery status.
Apple documented the AT commands in the following document:
https://developer.apple.com/hardwaredrivers/BluetoothDesignGuidelines.pdf
The patch has been tested with a Bose QC35, which results
in the following communication:
D: [pulseaudio] backend-native.c: RFCOMM << AT+VGS=14
D: [pulseaudio] backend-native.c: RFCOMM >> OK
D: [pulseaudio] backend-native.c: RFCOMM << AT+XAPL=009E-400C-0129,3
D: [pulseaudio] backend-native.c: RFCOMM >> +XAPL=iPhone,2
D: [pulseaudio] backend-native.c: RFCOMM >> OK
D: [pulseaudio] backend-native.c: RFCOMM << AT+XEVENT=Bose SoundLink,158
D: [pulseaudio] backend-native.c: RFCOMM >> OK
D: [pulseaudio] backend-native.c: RFCOMM << AT+IPHONEACCEV=2,1,4,2,0
N: [pulseaudio] backend-native.c: Battery Level: 50%
N: [pulseaudio] backend-native.c: Dock Status: undocked
D: [pulseaudio] backend-native.c: RFCOMM >> OK
[Marijn: Adapt for recent HSP/HFP code changes]
Co-authored-by: Marijn Suijten <marijns95@gmail.com>
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/482>
Alsa UCM device string can contain private configuration prefix required to make
correct device open call. Private prefix is dynamically generated by UCM manager
depending on internal state. Since pulseaudio sink/source port names currently
contain device string, these may change between runs breaking volume database
and module arguments referring to sink/source.
Fix this by skipping UCM private prefix available via `_alibpref` key while
creating UCM mapping name. Mapping object will still contain unmodified
device string for device open call.
See also https://github.com/alsa-project/alsa-ucm-conf/issues/104
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/598>
When child `gsettings-helper` terminates prematurely, unconditionally reading
from child pipe fails in a busy loop until child process is reaped.
Fix this by terminating module upon PA_IO_EVENT_HANGUP or PA_IO_EVENT_ERROR.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/600>
These two log messages are most likely intended for the path that was
just tried, but they are mistakenly printing the name of the port's
current path. Fix them.
Signed-off-by: Alper Nebi Yasak <alpernebiyasak@gmail.com>
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/594>
Recently we found an issue of output volume on speaker and headphone,
they should have their own volume but in practice they share one
output volume.
This issue happens on the laptops which use the ucm2 sof-hda-dsp,
originally the speaker has output volume A while the headphone has the
output volume B, suppose the speaker is the active port at the moment
and the output volume is A, users plug a headphone to the jack and the
headphone becomes the active port, in this process, ucm_set_port()
calls _disdev/_enadev which triggers the io_mixer_callback(), in the
meanwhile, the module_device_restore will restore the headphone's
volume to B, it will call set_volume_cb() to set the volume to B, but
this value is not written to hw immediately, during the time of
waiting for the B to be written to the hw, the io_mixer_callback()
calls get_volume_cb(), it reads hw volume and gets the volume A, then
it overrides the output volume to A, this results in the headphone
gets the volume A instead of B.
If a machine doesn't use the ucm, this issue will not happen since the
set_port_cb() will not trigger the io_mixer_callback(). If the ports
don't belong to the same sink/source, this issue also doesn't happen.
BugLink: http://bugs.launchpad.net/bugs/1930188
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/577>
* Minimal implementation of --system on win32.
* Wrap main with a Windows Service on win32 (with a fallback to
running it directly).
* Update PA_SYSTEM_{RUNTIME,STATE,CONFIG}_PATH and HOME dynamically
on Windows (overrides the build config, similar to the existing
config path replacement logic).
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/549>
Having G_MESSAGES_DEBUG=all set in the environment (a normal thing to do
when debugging Gnome troubles) causes gsettings-helper to emit a bunch
of helpful gnome debug logs (which is good), but before this change they
were printed on stdout rather than stderr (which was bad!). Rather than
going somewhere the user could see, these log messages were being sent
to the pulesaudio server and interpreted as the src/modules/stdin-util.c
protocol. pulseadio waits to see a '!' message from gsettings-helper
before continuing startup. With the log messages mixed in messing up
the stdin-util protocol, pulseaudio never saw the '!' message, and so
never completed startup.
This simple fix relies on a recent glib > 2.68 (Mar 2021), so builds
against old versions of glib will still have this problem! We consider
this good enough until some complains otherwise.
Fixes: https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/1222
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/579>
New card database entry version 5 for card profile is sticky flag.
New messaging API handlers set-profile-sticky and get-profile-sticky.
When card profile is sticky, always restore it even if it is unavailable,
and prevent switching from it when ports become unavailable.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/568>
It seems that in sound context environment variable is not available for
match expression.
This commit utilizes walkthrough to refer to attributes in fw node. The
combination of vendor, model, units is enough to match the node since
the attributes of fw unit doesn't have vendor.
Fix: 37358e42c4 ("alsa: Suppress udev detection of sound card for some units on IEEE 1394 bus")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/566>
The Volume property on org.bluez.MediaTransport1 is required to utilize
Absolute Volume, but it will only become availabe if the peer device
supports the feature. This happens asynchronously somewhere after the
transport itself has been acquired, after which the callbacks are
attached and software volume is reset.
To prevent race conditions availability of the property is also checked
on startup through a "Get" call.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/239>
Write the current volume to the `Volume` DBus property to keep the
volume on the remote in sync. Without this the remote device shows the
wrong volume, and any attempts to change it will cause an unexpected
jump when the local volume has also been adjusted.
Thanks to prior investments to improve volume synchronization, setting
up callbacks and sending initial volume to the peer for HFP/HSP
implementing this feature is as easy as unconditionally assigning a
valid function to `set_source_volume`. `source_setup_volume_callback`
is already responsible for attaching a `SOURCE_VOLUME_CHANGED` hook and
sending initial (restored) volume to the peer (signifying support for
Absolute Volume - if not derived from the presence of FEATURE_CATEGORY_2
on the profile yet).
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/239>
Like the previous commit this handles `Volume` property changes but
applies them to an A2DP sink instead of source stream. As mentioned in
the AVRCP spec v1.6.2 §5.8 the rendering device (A2DP sink) is
responsible for performing volume attenuation meaning PulseAudio should
pass through audio as-is without performing any attenuation in SW.
Setting a valid pointer to `set_sink_volume` and returning `true` from
`should_attenuate_volume` attaches a hardware callback to `pa_sink` such
that no volume attenuation is performed anymore.
In addition to receiving volume change notifications it is also possible
to control remote volume by writing a new value to the DBus property.
This is especially useful when playing back to in-ear audio devices
which usually lack physical buttons to adjust the final volume on the
sink.
While software volume (used before this patch) is generally fine it is
annoying to crank it up all the way to 100% when a previous connection
to a different device left saved volume on the peer at a low volume.
Providing this bidirectional synchronization is most natural to users
who wish to use physical controls on their headphones, are used to this
from their smartphone, or aforementioned volume mismatches where both PA
as source and the peer as sink/rendering device are performing
attenutation.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/239>
The A2DP spec mandates that the audio rendering device - the device
receiving audio, in our case a `pa_source` - is responsible for
performing attenuation:
AVRCP v1.6.2, §5.8:
The SetAbsoluteVolume command is used to set an absolute volume to be used by the rendering device.
BlueZ models this call as a change of the `Volume` property on the
`org.bluez.MediaTransport1` interface. Supporting Absolute Volume is
optional but BlueZ unconditionally reports feature category 2 in its
profile, mandating support. Hence remote devices (ie. a phone) playing
back audio to a machine running PulseAudio assume volume is to be
changed through SetAbsoluteVolume, without performing any local
attenuation.
Future changes will implement this feature the other way around: setting
an initial value for the `Volume` property as well as propagating
`pa_source` volume changes back to the peer.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/239>
In the case, where the latency is larger than the maximum block size,
module-null-sink will request multiples of the maximum block size from
the sink input instead of limiting the requested amount of data to the
the configured latency.
This patch fixes the problem.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/554>
This reverts commit 96369919e5.
The commit was originally for the issue of Headphone can't output
sound, that was because the Headphone and Lineout share the 1st alsa
mixer and DAC, but this commit introduced a new issue of the speaker
is not muted after switching to headphone.
A recent merged kernel commit (f48652bbe3ae@linux) could fix the 1st
issue, so we could revert the fix of the 1st issue from PA, then the
2nd issue is fixed automatically.
BugLink: https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/747
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/553>
The variable card_name in sink_input_preferred_sink_changed_cb and
source_output_preferred_source_changed_cb could be used uninitialized,
which leads to invalid database entries.
This patch fixes the problem.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/543>
The loopback message may be called after the sink input is already destroyed which causes
a crash. Also memory is leaked because the message object is not correctly freed.
This patch fixes the problems by adding a "dead" flag to the message structure and freeing
the message object on exit.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/541>
When fallback mapping is selected all subsequent profile sets containing
selected mapping are ignored. When there are only e.g. fallback input mappings
available, admitted profile set will only contain one profile with selected
first input fallback mapping and no outputs, and rest of profiles will only
contain outputs and no inputs. When there are only fallback input and output
mappings, there will be no profiles admitted at all.
Fix this by making sure that selected first fallback input or output mapping
is actually allowed to exist in all probed profile sets.
Note while this change allows selected fallback mappings to be found in duplex
configuraitons, probing fallbacks still can fail if there is more than one input
fallback and first one (selected) does not work in duplex configurations.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/539>
This patch fixes the following error:
$ pacat --file-format=ogg -r test.ogg
Failed to open audio file.
$ parecord sep.flac
Failed to open audio file.
libsndfile errors out if a WAV or OGG file is set to have anything but
SF_ENDIAN_FILE:
f4d1646e5c
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/526>
A few headsets have issues if HFP HF profile connection is attempted before
HSP HS profile connection is closed. Looks like this could happen because
bluez bluetoothd alows to make simultaneous HSP HS and HFP HF peer connections.
One of affected headsets is WH-1000XM2
Until we find out how to prevent simultaneous HSP HS and HFP HF connections,
when native backend has HFP HF profile enabled (this is the default) do disable
HSP HS completely unless user explicitly request it via discovery modarg.
Do this by adding module-bluetooth-discover arg enable_native_hsp_hs,
default to inverse of enable_native_hfp_hf.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/538>
For mSBC to work correctly the following must be set correctly
- codec object
- transport write method
- transport setsockopt method
Use helper method to set all three simultaneously.
Static configuration structure may be cleaner solution.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/507>
HFP Audio Connection SCO configuration is negotiated symmetrically in both
directions, and USB HCI SCO packet framing is also symmetric in both directions.
This means that packet size will be the same for reads and writes over HFP SCO
socket.
HFP profile specification states that valid speech data shall exist on the
Synchronous Connection in both directions after the Audio Connection is
established.
This guarantees that an incoming packet will arrive shortly after SCO connection
is established. Use it's size to fix write MTU in case kernel value is wrong.
Discussion here https://lore.kernel.org/patchwork/patch/1303411/
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/507>
The HFP protocol supports the ability to negotiate codecs if that is
supported by both AG and HF. This patch adds advertising of codec
negotiation support and the ability to negotiate a codec change. The
only currently supported extra codec (as of HF 1.7.1) is mSBC. mSBC
requires that the transmission be done over an eSCO link with
Transparent Data. The linux kernel ensures the former, but we have to
manually set the socket to transparent data.
Signed-off-by: James Bottomley <James.Bottomley@HansenPartnership.com>
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/507>
Adding processing support for the mSBC codec is somewhat problematic,
because, although it is a SBC codec, the a2dp handling can't simply be
reused because the codec is used on an eSCO link with transparent
data, meaning the transmission unit has to be 48 bytes (fragmenting
the codec packets) and reassembly and boundary detection is required
to be done by the implementation. Therefore we have to implement
separate render and push routines for msbc that do this fragmentation.
Fragmentation is done by emulating circular buffers. The receive
(push) buffer is easy, since the mSBC packet size is 60, simply have a
buffer of this size in the sbc_info area where the fragments are
reassembled. Once we have a full 60 bytes, decode and restart from
zero. The send (render) buffer is more problematic, since the
transmit must be done from contiguous memory. This means that the
buffer must be the lowest common multiple of the transmission unit and
the packet size. This value is 240 since 240/48 == 5 and 240/60 == 4.
So the buffer pointers are reset at 240 which is a whole number of
both rendered packets and eSCO transmission units.
Signed-off-by: James Bottomley <James.Bottomley@HansenPartnership.com>
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/507>
When an application sets a device for a newly created stream, we treat
that as a temporary setting, and don't save it as the preferred device
for future streams. The handling for this was broken, however: if the
stream already had a preferred device saved in the stream-restore
database, that was unset.
This was a regression introduced in
bc0e728320 and
70bbbcdc84. These commits tried to detect
in subscribe_callback() when the preferred device is cleared, but as a
side effect the preferred device started to get cleared from the
database also when a stream was created with a device set by the
application.
There's no way for subscribe_callback() to distinguish the different
cases of the preferred device being NULL. This problem is solved by
using the PREFERRED_SINK/SOURCE_CHANGED hooks. The hooks are only called
when the preferred device really changes.
Fixes: https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/1063
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/535>
The hooks are fired when the preferred device changes. This is useful
for module-stream-restore.
I added new set_preferred_sink/source() functions for firing the hooks.
The functions also log the preferred device changes.
There was already pa_sink_input_set_preferred_sink(), but that had a
side effect of moving the stream, so I needed a new function. Since it
can be confusing when the two similarly named functions should be
called, I added a comment for pa_sink_input_set_preferred_sink() that
explains the different situations.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/535>
Bluez prepends newly registered profile to a list of supported profiles,
and new peer profile connections are attempted in reverse order of profile
registration.
Currently native backend would register HFP AG profile before HSP AG profile.
When peer supports both HFP HF and HSP HS profiles, this registration order
causes extra HSP HS connection attempt before native backend would reject it
to make sure peer is reconnected with HFP HF profile.
Reorder HSP AG profile registration before HFP AG to make sure peer supporting
both profiles connects with HFP HF profile first.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/534>
Setting these callbacks adds the HW_{VOLUME,MUTE}_CTRL flag even when
PulseAudio is solely responsible for performing attenuation whilst only
keeping the peer posted on changes. For this case the hardware callback
is not registered at all but instead a hook is attached to catch
PA_CORE_HOOK_{SINK,SOURCE}_VOLUME_CHANGED. Only when the peer performs
attenuation (the peer is in HeadSet/HandsFree role) are the callbacks
used, without touching PA software volume at all. A future change could
potentially use software volume to compensate for the extremely coarse
16 steps of volume control in HSP and HFP, and to allow volume over
100%.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/519>
Since commit cb91d7a1 the watermark is increased when there is nothing to rewind.
This is also done in the case when there was actually no rewind requested at all,
so the watermark is increased needlessly.
This patch fixes the issue by skipping the rewind if none is requested.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/530>
Originally written for A2DP this rework of that patch enables late-bound
hardware volume control on HFP and HSP. As per the specification the
headphones (where gain control for both speaker and microphone could
happen in hardware on the peer) are supposed to send initial values for
these before the SCO connection is created; these `AT+VG[MS]` commands
are also used to determine support for it. PA uses this information in
`add_{sink,source}` to attach hardware volume callbacks, _if_ it is
supported. Otherwise PA performs the attenuation in software.
Unfortunately headphones like the WH-1000XM3's connect to A2DP
initially and only send `AT+VGS` (microphone hardware gain is not
supported) _during_ SCO connection when the user switches to the HFP
profile afterwards; the callbacks set up dynamically in
`rfcomm_io_callback` are written after the sink and source have been
created (`add_{sink,source}`), leaving them without hardware volume
callbacks and with software volume when adjusted on the PA side. (The
headphones can still send volume updates resulting in abrupt changes if
software and peer volume differ. Furthermore the same attenuation is
applied twice - once in PA software, once on the peer).
To solve this problem we simply check whether the callbacks have been
attached whenever the peer sends a volume change, and if not attach the
callbacks to the sink/source and reset software volume.
Fixes: d510ddc7f ("bluetooth: Perform software attenuation until HF/HS reports gain control")
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/528>
HF/HS hardware attenuation is optional on HFP: the peer indicates
support with the AT+BRSF command, when bit 4 is set. That does not
explicitly mandate speaker or microphone gain control; either is
dynamically detected as soon as `AT+VG[MS]=` is received. Otherwise
software attenuation is performed.
It is also optional on HSP but nothing is mentioned about feature
detection, assume it is the same as HFP: perform software attenuation
until the HF/HS peer sends an `AT+VG[MS]=` command.
When PA is a HS/HF (and the peer the AG) we attenuate both channels in
software and unconditionally keep the peer up to date with
`AT+VGM/AT+VGS` commands.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/521>
Generalize the distinction between local and peer-attenuated volumes
into a function, paving the way for future changes where this needs to
be checked in more places and when A2DP Absolute Volume support is
added.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/521>
Sink and source naming is more generic when dealing with audio that is
directional in the sense that it either goes to or comes from the other
device, but not necessarily a microphone or speaker. A concrete example
is the swapped meaning when the current device is in the HeadSet
profile. The incoming audio can come from any source, not necessarily a
microphone. Likewise, audio captured by the microphone of the headset is
not necessarily played back by a speaker on the AG, it is merely acting
as a sink for the data: further handling is irrelevant to the naming.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/521>
For the upcoming A2DP AVRCP Absolute Volume feature the code in BlueZ5
has to be generic to be reusable. Move this conversion so that it
becomes possible to implement A2DP volume - which uses different values
- on top without duplicating existing callback functionality.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/521>
The format of COMMAND line sent from HS to AG is COMMAND<cr>
The format of RESPONSE line sent from AG to HS is <cr><lf>RESPONSE<cr><lf>
Split rfcomm_write into rfcomm_write_command and rfcomm_write_response to handle
line formatting correctly.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/520>
Commit 4868fcf5f3 ("daemon: Rely on
systemd unit file for X11 plugin initialization") added a new systemd
unit file, pulseaudio-x11.service, generated from a respective .in file.
Unfortunately, this was only hooked up to meson, and is not currently
installed by autotools. Among other breakage, "make dist" produces a
tarball that meson is then unable to build (because a file is missing).
Signed-off-by: Faidon Liambotis <paravoid@debian.org>
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/514>
On cpu-volume-test, cpu_info is initialized only on i386/amd64 systems,
and otherwise passed on to pa_cpu_init_orc() uninitialized.
If one was unlucky enough, they could end up with cpu_info.cpu_type ==
PA_CPU_X86 on a non-x86 system, and use and test the Orc codepath
without that being functional, and thus with the test failing.
This has been observed in the wild on the ppc64el Debian buildds. See
Debian bug #982740 for more context.
Define cpu_info here in the same way as in other tests.
Signed-off-by: Faidon Liambotis <paravoid@debian.org>
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/511>
Intel TGL HDMI/DP codec provides 9 pins (Linux kernel, 9a11ba7388f16:
ALSA: hda: hdmi - add Tigerlake support), and with the DP MST enabled,
the linux kernel will build 11 output devices (3, 7, 8, 9, 10, 11, 12,
13, 14, 15, 16), and the alsa-lib will map 11 PCM devices from HDMI:0
to HDMI:10, but current pulseaudio only supports 8 HDMI/DP devices,
if users plug the HDMI/DP monitor to the last 3 ports, the users will
not see the output device from pulseaudio or gnome.
We have experienced this issue on a dell TGL machine with a dock, we
plugged 2 HDMP/DP monitors on the dock, but we could only see 1
HDMI/DP output device from pulseaudio or gnome, through investigation,
we found one monitor is plugged in the 2nd port from last.
Here we add 3 HDMI/DP output devices.
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/495>
When HFP HF support is enabled in native backend, peer HFP HF profile connection
is preferred over same peer HSP HS profile connection if peer supports both
profiles.
Enforce the preference by rejecting HSP HS profile connections from such peer.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/491>
Native backend implements HFP AG but not HFP HF yet, therefore headset=auto
functionality is still needed if HFP HF is required.
To make headset=auto work again, drop both HFP AG and HSP AG roles while
performing handover from native backend when oFono is detected running.
While at it, restore profile description to Headset Head Unit (HSP/HFP)
to note that HFP may be still provided via oFono backend.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/491>
Change default backend from 'auto' to 'native' so that in the usual
install pulseaudio uses the native backend with HFP_HF handling.
set default to false unless the backend is the native one, in which
case the default becomes true.
Additionally set default value of enable_native_hfp_hf to false unless
the backend is the native one, in which case the default becomes
true. so that we only bind the HFP_HF end point in the native case
(leaving it free for ofono in the ofono backend or auto case)
Signed-off-by: James Bottomley <James.Bottomley@HansenPartnership.com>
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/491>
HFP 1.6 requires a stateful negotiation of AT commands. The prior
version got away with initialising HFP simply by replying 'OK' to
every negotiation attempt. This one actually tries to parse the state
and make sure the negotiation occurs correctly
Signed-off-by: James Bottomley <James.Bottomley@HansenPartnership.com>
---
v4:
- Update for PA 11.0
- Finally sort out CIND negotiaton for complex headsets
v3:
- remove internal debugging
- added comment for t->config being not null for hfp
- removed unused returns from hfp_rfcomm_handle()
- remove rfcomm comment
- use pa_startswith
- simplify negotiation
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/491>
When all headsets supported both HSP and HFP, life was good and we
only needed to implement HSP in the native backend. Unfortunately
some headsets have started supporting HFP only. Unfortuantely, we
can't simply switch to HFP only because that might break older HSP
only headsets meaning we need to support both HSP and HFP separately.
This patch separates them from a joint profile to being two separate
ones. The older one retains the headset_head_unit name, meaning any
saved parameters will still select this (keeping us backward
compatible). It also introduces a new headset_handsfree.
For headsets that support both HSP and HFP, the two profiles will
become separately visible and selectable. This will only matter once
we start adding features to HFP that HSP can't support (like wideband
audio).
Signed-off-by: <James.Bottomley@HansenPartnership.com>
---
v6:
- merge profile switching fixes patch from Rodrigo Araujo
v5:
- rename option to enable_native_hfp_hf
- don't call profile_done for HFP_HF unless it was initialised
v3:
- Update for PA 11.0
v2:
- fold in review feedback
- add global disable option for not registering HFP
v3:
- change parameter to enable_profile_hfp
- update device_supports_profile to be aware of hfp/hsp exclusivity
- change parameter to enable_profile_hfp_hf
bluetooth: separate HSP and HFP (to me merged with this patch)
Hi.
First, just to say that your patches are going great. Finally I can use
the microphone of my HFP only headset (a version of a Bluedio T2+).
So far, I've only encontered one problem: the auto_switch option of
module_bluetooth_policy stops working. Dug through the code and I think
you missed a few spots were you have to hangle the new headset_handsfree
profile in module_bluetooth_policy.c
Applying the following after applying your v5 patches fixed the issue
for me, now when I start making a VOIP call the profile switches to
headset_handsfree and the mic works automatically, and when the call
finishes it reverts back to a2dp.
Thanks and best regards.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/491>
The PA_BLUETOOTH_PROFILE names should mirror the PA_BLUETOOTH_UUID
names using profile_function instead of randomly made up names. Fix
this with the transformation:
PA_BLUETOOTH_PROFILE_HEADSET_HEAD_UNIT -> PA_BLUETOOTH_PROFILE_HSP_HS
PA_BLUETOOTH_PROFILE_HEADSET_AUDIO_GATEWAY -> PA_BLUETOOTH_PROFILE_HFP_AG
Signed-off-by: James Bottomley <James.Bottomley@HansenPartnership.com>
---
v4: update for PA 11.0
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/491>
When the source-output of a virtual source with volume sharing disabled is moved,
the source output volume is reset to 100%. This patch fixes the problem by
applying the virtual source volume to the source-output after the move.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/509>
The encoding and decoding pipeline are essentially identical: both push
data in via an appsrc, route it through a codec-specific (opaque)
element, and finally pull data out of an appsink. The code already makes
it impossible to have an encoding and decoding pipeline simultaneously
set up in `gst_info`, and converting `bool for_encoding` to a tri-state
(encode, decode, or both) would be messy; particularly when encoding and
decoding could possibly differ in format.
This change removes a swath of code and removes the possibility of
misusing `enc_` or `dec_` in the wrong place (ie. after copying a bit of
code and forgetting to rename one or two). When bidirectional codecs
come online a second codec instance (`gst_info`) can simply be created
and controlled independently.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/487>
LDAC encoder already supports S16, S24, S32 and F32LE. Using FLOAT32LE
for the sample format would avoid the additional call for conversion to
pa_sconv_s32le_from_float32ne. perf tool shows this as being the function
called frequently after encode. So, just avoid this by using sample format
as F32LE.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/486>
Make the code ever so slightly more generic by not using appsrc and
appsink in codec-specific logic when assigning caps specific to the raw
(PCM) format provided by or returned to PA.
Note that caps have to be set (= event) after starting, can't send
events in flushing state.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/484>
SBC codec decrements bitpool value by fixed amount each time it is asked to
reduce output bitrate. This results in reduced audio quality with SBC codec.
Implement increase_encoder_bitrate for SBC codec by adding 1 to bitpool value
each time encoder bitrate needs to be increased to restore SBC audio quality.
While at it, remove bitpool decrement limit to use connection agreed value
instead as we will be able to restore quality later.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/474>
Bluetooth thread may ask encoder to reduce bitrate if writing is not keeping up
with inputs or writing to bluetooth socket takes too much time.
Assuming conditions leading to reduced bitrate are intermittent, allow periodic
attempts to increase encoder bitrate, by default at most twice per second.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/481>
Similar to the situation/comment in `endpoint_release` BlueZ does not
request any reply to `ClearConfiguration()` either; sending one results
in the same "0 matched rules" warning from dbus-daemon:
dbus-daemon[1309]: [system] Rejected send message, 0 matched rules; type="method_return", sender=":1.71" (uid=1000 pid=87548 comm="../build/src/daemon/pulseaudio -vvvv -n -F ../buil") interface="(unset)" member="(unset)" error name="(unset)" requested_reply="0" destination=":1.3" (uid=0 pid=1308 comm="/usr/lib/bluetooth/bluetoothd -d ")
Solve this by only creating a return message when an (othwise empty)
reply is solicited for, just like in `endpoint_release`.
Unfortunately we also have to make sure to not send any error back if no
reply is requested, but fortunately an argument parsing error here is
extremely unlikely.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/472>
We move the codec specific bits to their own respective files and now
make the codec specific initialisation use a GstBin, which the generic
GStreamer module now uses in the pipeline.
It is job of the codec specific function to add elements in the GstBin
and link the added elements in the bin. It should also set up the ghost
pads as a GstBin has no pads of it's own and without which the bin
cannot be linked to the appsrc/appsink.
Also, we now only initialise either the encoding or the decoding
pipeline and not both. The codec init API already gets passed the
for_encoding flag. We pass and use the same to codec specific init
functions.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/440>
As we now support codecs other than SBC, we might have codec which does
not have an encode or a decode capability. Specifically, in the case of
LDAC there isn't a known decoder implementation available. For such a
case, we should not register the corresponding endpoint.
In case of LDAC, as decoding cannot be supported, we should not register
a sink endpoint or vice versa in the other scenario.
To do this, we check if encode_buffer or decode_buffer entry for a codec
has been set in pa_a2dp_codec and accordingly prevent or allow it's
registration.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/440>
When it comes to codecs provided via GStreamer, we register all codecs
if GStreamer option is enabled for bluez5 via meson. However, the
GStreamer plugin required for the codec might not be present on the
system. This results in the codec being available for registration with
the bluez stack or selection by the user, but, trying to use the said
codec then fails.
To prevent the above, we now use the can_be_supported codec API to check
if the codec is usable and if not, we do not register the said codec and
also prevent users from switching to it.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/440>
This API internally checks if a requested codec can be supported on the
system. This is especially required for codecs supported via GStreamer
where the availability of a plugin decides if the said codec can be
supported.
This will be used to prevent registration of a codec which the remote
endpoint device might be able to support, but, PulseAudio can't as the
codec is not available on the system due to the absence of a plugin.
We can also prevent listing or switching to an unavailable codec.
Note that the codec negotiation happens with the bluez stack even before
a device is connected. Because of this, we need to make sure that gst_init
is called before checking for the availability of a plugin. Since
module-bluez5-device gets loaded only after a connection to the device
has been established, doing the gst_init in that or one of the bluetooth
modules is not feasible.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/440>
For example, using the following on the command line will return the
current codec for a bluetooth device
pacmd send-message /card/bluez_card.4C_BC_98_80_01_9B/bluez get-codec
where 4C_BC_98_80_01_9B is the bluetooth device.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/440>
This exposes the currently active codec on the source or sink via the
proplist and can be seen in output of pacmd list-sinks/list-sources.
Also set it on the card. In case of a bi-directional codec, the codec
for the sink and source could be different. For example, for aptX-LL,
the codec name on card, sink and source would be aptx-ll, aptx and sbc
respectively.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/440>
For example, using the following on the command line will return the
list of possible codecs for a bluetooth device
pacmd send-message /card/bluez_card.4C_BC_98_80_01_9B/bluez list-codecs
where 4C_BC_98_80_01_9B is the bluetooth device.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/440>
This adds a generic gstreamer codec module based on which other
bluetooth codecs viz. aptX, aptX-HD, LDAC and AAC can be supported.
The GStreamer codec plugins used here themselves depend on the native
codec implementation.
aptX/aptX-HD -> libopenaptx
LDAC -> libldac
AAC -> Fraunhofer FDK AAC
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/440>
This uses the messaging API to initiate a codec switch.
While a particular codec might be applicable only for a particular
profile, for eg. aptX can only be applicable for A2DP sink or source
and not for let's say HSP, the codec switching logic has not been
tied to the logic for switching profiles.
Codec can be switched by running the following on the command line.
pacmd send-message /card/bluez_card.XX_XX_XX_XX_XX_XX/bluez switch-codec{"ldac_hq"}
pacmd send-message /card/bluez_card.XX_XX_XX_XX_XX_XX/bluez switch-codec {"ldac_mq"}
pacmd send-message /card/bluez_card.XX_XX_XX_XX_XX_XX/bluez switch-codec {"ldac_sq"}
pacmd send-message /card/bluez_card.XX_XX_XX_XX_XX_XX/bluez switch-codec {"aptx_hd"}
pacmd send-message /card/bluez_card.XX_XX_XX_XX_XX_XX/bluez switch-codec {"aptx"}
pacmd send-message /card/bluez_card.XX_XX_XX_XX_XX_XX/bluez switch-codec {"sbc"}
Codec name passed above is matched against pa_a2dp_codec->name. Note that
the match is case sensitive. XX_XX_XX_XX_XX_XX needs to be substituted with
the actual bluetooth device id.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/440>
Instead of letting a codec with higher index have higher priority,
just use a lower index for high priority. This allows the for loop
iterating over the codecs to be written in a straightforward manner
and not have to iterate from the end. FWIW Pipewire does the same.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/440>
If the current active profile is off, it has no sinks and sources, and
if users plug a headset to the audio port, the profile including this
audio port becomes available and should be selected as active profile.
But with the current design, the profile_good_for_output() will return
false because the sources in off profile and target profile doesn't
match.
For example:
(Before users plug headset)
Profiles:
HiFi (Speaker): Default (sinks: 1, sources: 1, priority: 8100, available: no)
HiFi (Headphones): Default (sinks: 1, sources: 1, priority: 8200, available: no)
off: Off (sinks: 0, sources: 0, priority: 0, available: yes)
Active Profile: off
(After users plug headset)
Profiles:
HiFi (Speaker): Default (sinks: 1, sources: 1, priority: 8100, available: yes)
HiFi (Headphones): Default (sinks: 1, sources: 1, priority: 8200, available: yes)
off: Off (sinks: 0, sources: 0, priority: 0, available: yes)
Active Profile: off
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/354>
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>
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>
Some gaming sound cards have custom profiles with analog-game and
analog-chat mappings that exist simultaneously. The game sink should
have higher priority than the chat sink, but currently there's no way to
affect the sink priorities from the profile-set configuration (the
mapping priority is not propagated to the sink priority).
I first thought about adding the mapping priority to the sink priority,
but that could mess up the prioritization system in
pa_device_init_priority(). I ended up checking for the intended roles
property to reduce the chat sink priority. I also reduced the iec958
priority so that the chat and iec958 sinks don't end up with the same
priority.
Fixes: https://gitlab.freedesktop.org/pulseaudio/pulseaudio/issues/818
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/266>
In OpenEmbedded the PulseAudio recipe currently disables Valgrind
support by passing "ac_cv_header_valgrind_memcheck_h=no" to the
configure script (this was added to make it deterministic whether
Valgrdind support gets enabled or not). I'm converting the PulseAudio
recipe to use Meson, and I needed an option to disable Valgrind.
Since there's no stable API for modules, all modules need to be compiled
together with the server. This version check tries to ensure that if
a version mismatch happens, there will be an informative error message
rather than a random crash.
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.
I believe nobody needs to pass octal numbers to PulseAudio, and if we
encounter integer strings starting with zeros, the intention is to use
them in base 10. Hexadecimal numbers are more common, and they can't be
interpreted in base 10 anyway, so they are still supported.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/51>
If an application calls the function when the server doesn't support the
feature, the result should be just an error from the function. Without
the check the whole connection gets terminated due to protocol error.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/51>
The following new functions have been added:
pa_message_params_read_double_array() - read an array of double from list
pa_message_params_read_int64_array() - read an array of int64 from list
pa_message_params_read_uint64_array() - read an array of uint64 from list
pa_message_params_read_string_array() - read an array of strings from list
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/51>
The following functions have been added:
pa_message_params_write_double() - writes a double to a pa_message_params structure
pa_message_params_write_int64() - writes an integer to a pa_message_params structure
pa_message_params_write_uint64() - writes an unsigned to a pa_message_params structure
pa_message_params_write_bool() - writes a boolean to a pa_message_params structure
pa_message_params_read_double() - read a double from a parameter list
pa_message_params_read_int64() - read an integer from a parameter list
pa_message_params_read_uint64() - read an unsigned from a parameter list
pa_message_params_read_bool() - read a boolean from a parameter list
The patch also improves the doxygen documentation im message-params.h
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/51>
The patch adds the possibility to escape curly braces within parameter strings
and introduces several new functions that can be used for writing parameters.
For writing, the structure pa_message_params, which is a wrapper for pa_strbuf
has been created. Following new write functions are available:
pa_message_params_new() - creates a new pa_message_params structure
pa_message_params_free() - frees a pa_message_params structure
pa_message_param_to_string_free() - converts a pa_message_param to string and
frees the structure
pa_message_params_begin_list() - starts a list
pa_message_params_end_list() - ends a list
pa_message_params_write_string() - writes a string to a pa_message_params structure
pa_message_params_write_raw() - writes a raw string to a pa_message_params structure
For string parameters that contain curly braces or backslashes, those characters
will be escaped when using pa_message_params_write_string(), while write_raw() will
put the string into the buffer without any changes.
For reading, pa_message_params_read_string() reverts the changes that
pa_message_params_write_string() might have introduced.
The patch also adds more restrictions on the object path name. Now only
alphanumeric characters and one of "_", ".", "-" and "/" are allowed.
The path name may not end with a / or contain a double slash. If the user
specifies a trailing / when sending a message, it will be silently removed.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/51>
For better readability, "pactl list message-handlers" is introduced which
prints a formatted output of "pactl send-message /core list-handlers".
The patch also adds the functions pa_message_params_read_raw() and
pa_message_params_read_string() for easy parsing of the message response
string. Because the functions need to modify the parameter string,
the message handler and the pa_context_string_callback function now
receive a char* instead of a const char* as parameter argument.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/51>
This patch adds a small message handler to the core which enables
clients to list available handlers via the list-handlers message.
Command: pacmd send-message /core list-handlers
pactl can be used with the same parameters.
The patch also introduces a convention for the return string.
It consists of a list of elements where curly braces are used
to separate elements. Each element can itself contain further
elements. For example consider a message that returns multiple
elements which each contain an integer and an array of float.
A response string would look like that:
{{Integer} {{1st float} {2nd float} ...}}{...}
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/51>
This patch adds the PA_COMMAND_SEND_OBJECT_MESSAGE command to protocol-native
so that clients can use the messaging feature introduced in the previous patch.
Sending messages can in effect replace the extension system for modules. The
approach is more flexible than the extension interface because a generic string
format is used to exchange information. Furthermore the messaging system can be
used for any object, not only for modules, and is easier to implement than
extensions.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/51>
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
The .include meta command already supports specifying a directory and
when including a directory, all files with the extension '.pa' in that
directory will be parsed in alphabetical order.
This feature can be used to add support for default.pa.d directory, so
that packages for other applications or users can just drop in a file
for configuration without changing the default.pa which is shipped.
We use the PA_DEFAULT_CONFIG_DIR for this, however, since meson quotes
this build variable, introduce an unquoted version for this purpose and
use it with .include.
Fixes: https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/909
Signed-off-by: Sanchayan Maity <sanchayan@asymptotic.io>
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
The webrtc backend of module-echo-cancel uses sscanf() to parse floating
point numbers from module arguments, which didn't work when the locale
used a comma for the decimal point. Setting the LC_NUMERIC locale
variable to C makes the pulseaudio process use a period as the decimal
point regardless of the user's locale configuration.
Fixes: https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/89
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))) {
|
<optdesc><p>Specify the symbolic name of the sink/source to play/record this stream on/from.</p></optdesc>
<optdesc><p>Specify the symbolic name of the sink/source to play/record this stream on/from. The special names @DEFAULT_SINK@, @DEFAULT_SOURCE@ and @DEFAULT_MONITOR@ can be used to specify the default sink, source and monitor respectively.</p></optdesc>
</option>
<option>
@ -202,7 +202,7 @@ License along with PulseAudio; if not, see <http://www.gnu.org/licenses/>.
<optdesc><p>Suspend or resume the specified source (which may be
specified either by its name or index), depending whether true
specified either by its symbolic name or numerical index), depending whether true
(suspend) or false (resume) is passed as last argument. Suspending
a source will pause all capturing. Depending on the module implementing
the source this might have the effect that the underlying device is
@ -161,9 +167,16 @@ License along with PulseAudio; if not, see <http://www.gnu.org/licenses/>.
<optdesc><p>Set the specified card (identified by its symbolic name or numerical index) to the specified profile (identified by its symbolic name).</p></optdesc>
</option>
<option>
<p><opt>get-default-sink</opt></p>
<optdesc><p>Returns the symbolic name of the default sink.</p></optdesc>
</option>
<option>
<p><opt>set-default-sink</opt><arg>SINK</arg></p>
<optdesc><p>Make the specified sink (identified by its symbolic name) the default sink.</p></optdesc>
<optdesc><p>Make the specified sink (identified by its symbolic name or numerical index) the default sink.
Use the special name \@NONE@ to unset the user defined default sink. This will make pulseaudio return to the default
sink selection based on sink priority.</p></optdesc>
</option>
<option>
@ -171,9 +184,16 @@ License along with PulseAudio; if not, see <http://www.gnu.org/licenses/>.
<optdesc><p>Set the specified sink (identified by its symbolic name or numerical index) to the specified port (identified by its symbolic name).</p></optdesc>
</option>
<option>
<p><opt>get-default-source</opt></p>
<optdesc><p>Returns the symbolic name of the default source.</p></optdesc>
<optdesc><p>Make the specified source (identified by its symbolic name) the default source.</p></optdesc>
<optdesc><p>Make the specified source (identified by its symbolic name or numerical index) the default source.
Use the special name \@NONE@ to unset the user defined default source. This will make pulseaudio return to the default
source selection based on source priority.</p></optdesc>
</option>
<option>
@ -187,6 +207,11 @@ License along with PulseAudio; if not, see <http://www.gnu.org/licenses/>.
<arg>OFFSET</arg> is a number which represents the latency offset in microseconds</p></optdesc>
</option>
<option>
<p><opt>get-sink-volume</opt><arg>SINK</arg></p>
<optdesc><p>Get the volume of the specified sink (identified by its symbolic name or numerical index) displayed in the same format as the `info` command.</p></optdesc>
<optdesc><p>Get the volume of the specified source (identified by its symbolic name or numerical index) displayed in the same format as the `info` command.</p></optdesc>
