In a former commit 37358e42c4 ("alsa: Suppress udev detection of sound
card for some units on IEEE 1394 bus"), PulseAudio has udev rules to
suppress handling some units on IEEE 1394 bus for a below issue:
Bug 199365 - repeating bus resets on Firewire bus with Focusrite Saffaire 26/io
https://bugzilla.kernel.org/show_bug.cgi?id=199365
However, I found that the rules match another model; Focusrite Liquid
Saffire 56. For detail, refer to below patch for Linux sound subsystem:
[alsa-devel] [PATCH] ALSA: bebob: use more identical mod_alias for
Saffire Pro 10 I/O against Liquid Saffire 56
https://mailman.alsa-project.org/pipermail/alsa-devel/2019-February/146003.html
For PulseAudio, the udev rule should be improved, because Liquid Saffire 56
(an application of TCAT TCD2200 ASIC, a.k.a Dice Jr.) can be handled by
pulseaudio without the issue.
This commit changes udev rule with model name instead of model_id from
configuration ROM. Below is data on udevd for Liquid Saffire 56, for
your information:
$ udevadm info -q all -p /sys/bus/firewire/devices/fw1.0/sound/card2/
P: /devices/pci0000:00/0000:00:01.2/0000:03:00.2/0000:04:07.0/0000:0a:00.0/0000:0b:00.0/fw1/fw1.0/sound/card2
E: DEVPATH=/devices/pci0000:00/0000:00:01.2/0000:03:00.2/0000:04:07.0/0000:0a:00.0/0000:0b:00.0/fw1/fw1.0/sound/card2
E: ID_BUS=firewire
E: ID_FOR_SEAT=sound-pci-0000_0b_00_0
E: ID_ID=firewire-0x00130e04018001e9
E: ID_MODEL=LIQUID_SAFFIRE_56
E: ID_MODEL_FROM_DATABASE=XIO2213A/B/XIO2221 IEEE-1394b OHCI Controller [Cheetah Express]
E: ID_MODEL_ID=0x000006
E: ID_PATH=pci-0000:0b:00.0
E: ID_PATH_TAG=pci-0000_0b_00_0
E: ID_PCI_CLASS_FROM_DATABASE=Serial bus controller
E: ID_PCI_INTERFACE_FROM_DATABASE=OHCI
E: ID_PCI_SUBCLASS_FROM_DATABASE=FireWire (IEEE 1394)
E: ID_SERIAL=0x00130e04018001e9
E: ID_SERIAL_SHORT=0x00130e04018001e9
E: ID_VENDOR=Focusrite
E: ID_VENDOR_FROM_DATABASE=Texas Instruments
E: ID_VENDOR_ID=0x00130e
E: SOUND_INITIALIZED=1
E: SUBSYSTEM=sound
E: SYSTEMD_WANTS=sound.target
E: TAGS=💺systemd:
E: USEC_INITIALIZED=9802422583
Fixes: 37358e42c4 ("alsa: Suppress udev detection of sound card for some units on IEEE 1394 bus")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Before this commit ucm_port_contains() was using a strncmp to compare
UCM-device-names without first checking that the part of the port_name
being compared and the device-name have the same length, this was causing
it to return true for both "InternalMic-IN1" and "InternalMic-IN12" when
port_name contained "InternalMic-IN1".
We hit this with the bytcr_rt5651 UCM profile which has "InternalMic-IN1",
"InternalMic-IN2" and "InternalMic-IN12" devices, for devices with their
internal mic connected to IN1, or IN2, or using stereo internal mics
connected to both. This problem resulted in various problems including
the RECMIXL? BST2 switch getting turned on when selecting only
"InternalMic-IN1", as well as confusing the gnome-control-center sound
panel, which could not figure out which device is selected in this case.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
The previous commit introduces logic in module-switch-on-port-available
that may change a card's active profile when its availability changes to
PA_AVAILABLE_NO. To choose the new active profile, it needs a consistent
view of the new availability of all profiles, so this commit changes the
order which the ALSA driver updates all profiles' availability to ensure
the active profile is last.
This is not generic enough to cover cases were we may want to take an
action on availability changes of profiles other than the active one
that also need a consistent view of all profiles' availability. But we
don't have any callbacks implementing such action at the moment.
It is helpful to improve reproducibility build [1] since
PA_SRCDIR/PA_BUILDDIR contains build path,
--disable-running-from-build-tree could drop these macros at
precompilation.
[1] https://reproducible-builds.org/
Signed-off-by: Hongxu Jia <hongxu.jia@windriver.com>
Sample format(e.g. 16 bit, 24 bit) was not considered even if the
avoid-resampling option is set or the passthrough mode is used.
This patch checks both sample format and rate of a stream to
determine whether to avoid resampling in case of the option is set.
In other word, it is possble to use the stream's original sample
format and rate without resampling as long as these are supported
by the device.
pa_sink_input_update_rate() and pa_source_output_update_rate() are
renamed to pa_sink_input_update_resampler() and pa_source_output
_update_resampler() respectively.
functions are added as below.
pa_sink_set_sample_format(), pa_sink_set_sample_rate(),
pa_source_set_sample_format(), pa_source_set_sample_rate()
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>
Headphones should have higher priority than lineout. Many people have
speakers always connected to lineout, and when plugging in headphones,
the audio should move to the headphones, which requires headphones
to have higher priority than lineout.
Previously this was handled by marking lineout unavailable when plugging
in headphones, but we don't do that any more.
This reverts commit 66f97c35bd. The commit
message was:
alsa-mixer: Disable line-out if headphone jack is plugged
Line-out gets muted when headphones are plugged in on HDA cards, encode
this in the line-out path so pulse can match that state.
I don't think the mentioned auto-muting happens any more by default,
and some users want to use lineout while having headphones plugged in.
Fixes: https://gitlab.freedesktop.org/pulseaudio/pulseaudio/issues/583
This is to be consistent. In pa currently, as built by the autotools,
libalsa-util is a shared library. Moreover, all the libraries for the
modules, as defined in `src/meson.build`, are also shared libraries.
So let's stick to shared libraries everywhere for now, for simplicity.
We can rework that later on.
Signed-off-by: Arnaud Rebillout <arnaud.rebillout@collabora.com>
A bug was filed to bugzilla.kernel.org for a quirk of some models which
ALSA BeBoB driver supports.
Bug 199365 - repeating bus resets on Firewire bus with Focusrite Saffaire 26/io
https://bugzilla.kernel.org/show_bug.cgi?id=199365
Some models (two models as long as I know) have a quirk to disappear from
IEEE 1394 bus at disconnections of packet streaming. Corresponding
character devices are removed according to 'remove' callbacks of relevant
drivers from Linux dd core. Then the models re-appear on the bus by
generating bus resets and corresponding character devices are added
according to 'probe' callbacks from Linux dd core.
In a view of ALSA applications, this looks that plug-out/plug-in occur in
a sequential order for the models when they stop playback/capture substream.
For most applications, this doesn't cause large issue. However, this quirk
is not good for combination of below modules in PulseAudio. PulseAudio
enters endless loop to detect the models and start/stop PCM substream.
- module-udev-detect
- module-alsa-card
- module-suspend-on-idle
In detail, please read my comment no.6:
https://bugzilla.kernel.org/show_bug.cgi?id=199365#c6
This commit suppressed udev detection of sound card for the issued models.
For the models, 'PULSE_IGNORE' flag is added to udev rules, then
module-udev-detect don't handle the models and PulseAudio never uses the
models automatically. In a scenario for users to load
module-alsa-card/module-alsa-sink/module-alsa-source by hand, although
these modules can still stop PCM substreams with module-suspend-on-idle,
PulseAudio never enters the endless loop because udev detection doesn't
work for the models. In this case, as long as special files for ALSA
character devices for these models are the same, corresponding sinks and
sources are available even if the voluntary plug-out/plug-in occur.
(Focusrite Saffire Pro 10 i/o with systemd 237)
$ udevadm info -q all -p /devices/pci0000:00/0000:00:07.0/fw1/fw1.0/sound/card1
P: /devices/pci0000:00/0000:00:07.0/fw1/fw1.0/sound/card1
E: DEVPATH=/devices/pci0000:00/0000:00:07.0/fw1/fw1.0/sound/card1
E: ID_BUS=firewire
E: ID_FOR_SEAT=sound-pci-0000_00_07_0
E: ID_ID=firewire-0x00130e01000606e0
E: ID_MODEL=Pro10IO
E: ID_MODEL_FROM_DATABASE=XIO2213A/B/XIO2221 IEEE-1394b OHCI Controller [Cheetah Express]
E: ID_MODEL_ID=0x000006
E: ID_PATH=pci-0000:00:07.0
E: ID_PATH_TAG=pci-0000_00_07_0
E: ID_PCI_CLASS_FROM_DATABASE=Serial bus controller
E: ID_PCI_INTERFACE_FROM_DATABASE=OHCI
E: ID_PCI_SUBCLASS_FROM_DATABASE=FireWire (IEEE 1394)
E: ID_SERIAL=0x00130e01000606e0
E: ID_SERIAL_SHORT=0x00130e01000606e0
E: ID_VENDOR=Focusrite
E: ID_VENDOR_FROM_DATABASE=Texas Instruments
E: ID_VENDOR_ID=0x00130e
E: SOUND_INITIALIZED=1
E: SUBSYSTEM=sound
E: SYSTEMD_WANTS=sound.target
E: TAGS=:systemd:seat:
E: USEC_INITIALIZED=957089064
(Focusrite Saffire Pro 26 i/o with systemd 237)
$ udevadm info -q all -p /devices/pci0000:00/0000:00:07.0/fw1/fw1.0/sound/card1
P: /devices/pci0000:00/0000:00:07.0/fw1/fw1.0/sound/card1
E: DEVPATH=/devices/pci0000:00/0000:00:07.0/fw1/fw1.0/sound/card1
E: ID_BUS=firewire
E: ID_FOR_SEAT=sound-pci-0000_00_07_0
E: ID_ID=firewire-0x00130e0100030cdd
E: ID_MODEL=Pro26IO
E: ID_MODEL_FROM_DATABASE=XIO2213A/B/XIO2221 IEEE-1394b OHCI Controller [Cheetah Express]
E: ID_MODEL_ID=0x000003
E: ID_PATH=pci-0000:00:07.0
E: ID_PATH_TAG=pci-0000_00_07_0
E: ID_PCI_CLASS_FROM_DATABASE=Serial bus controller
E: ID_PCI_INTERFACE_FROM_DATABASE=OHCI
E: ID_PCI_SUBCLASS_FROM_DATABASE=FireWire (IEEE 1394)
E: ID_SERIAL=0x00130e0100030cdd
E: ID_SERIAL_SHORT=0x00130e0100030cdd
E: ID_VENDOR=Focusrite
E: ID_VENDOR_FROM_DATABASE=Texas Instruments
E: ID_VENDOR_ID=0x00130e
E: SOUND_INITIALIZED=1
E: SUBSYSTEM=sound
E: SYSTEMD_WANTS=sound.target
E: TAGS=:systemd:seat:
E: USEC_INITIALIZED=1071026684
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
There has been a function to get supported sample rates from alsa and
an array for it in userdata of each module-alsa-sink/source. Similarly,
this patch adds a function to get supported sample formats(bit depth)
from alsa and an array for it to each userdata of the modules.
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>
pa_sink_get_state() and pa_source_get_state() just return the state
variable. We can as well access the state variable directly.
There are no behaviour changes, except that module-virtual-source
accessed the main thread's sink state variable from its push() callback.
I fixed the module so that it uses the thread_info.state variable
instead. Also, the compiler started to complain about comparing a sink
state variable to a source state enum value in protocol-esound.c. The
underlying bug was that a source pointer was assigned to a variable
whose type was a sink pointer (somehow using the pa_source_get_state()
macro confused the compiler enough so that it didn't complain before).
I fixed the variable type.
A bit hacky approach, but it allows to preserve LFE output position
even in reduced output modes 2.1 and 4.1.
Signed-off-by: Nazar Mokrynskyi <nazar@mokrynskyi.com>
If a sound card doesn't have the "front" device defined for it, we have
to use the "hw" device for stereo. Not so long ago, the analog-stereo
mapping had "hw:%f" in its device-strings and everything worked great,
except that it caused trouble with the Intel HDMI LPE driver that uses
the first "hw" device for HDMI, and we were incorrectly detecting it as
an analog device. That problem was fixed in commit ea3ebd09, which
removed "hw:%f" from analog-stereo and added a new stereo fallback
mapping for "hw".
Now the problem is that if a sound card doesn't have the "front" device
defined for it, and it supports both mono and stereo, only the mono
mapping is used, because the stereo mapping is only a fallback. This
patch makes the mono mapping a fallback too, so the mono mapping is used
only if there's absolutely nothing else that works.
This can cause trouble at least in theory. Maybe someone actually wants
to use mono output on a card that supports both mono and stereo. But
that seems quite unlikely.
Previously, the "avoid-resampling" option of daemon.conf is to make the
daemon try to use the stream sample rate if possible for all sinks or
sources.
This patch applies this option to module-udev-detect and module-alsa-card
as a module argument in order to override the default value of daemon.conf.
As a result, user can use this argument for more fine-grained control.
e.g.) set it false in daemon.conf and set it true for module-udev-detect
or a particular module-alsa-card in default.pa.(or vice versa)
To set it, use "avoid_resampling=true or false" as the module argument.
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>
There are only stereo and 5.1 output modes supported natively on this
sound card, but with this config more modes like 2.1, 4.0, 4.1 and 5.0
are now exposed. Also profiles list is cleaner now with all profiles
explicitly specified.
Last thing is removed support for microphone on Linux kernels older than
4.3-rc1, which shouldn't be an issue with future version of PulseAudio
likely be installed on newer kernels anyway.
Signed-off-by: Nazar Mokrynskyi <nazar@mokrynskyi.com>
This should make it easier for clients to elevate their audio threads to
real time priority without having to dig through much through specific
system internals.
When a new card shows up (during pulseaudio startup or hotplugged),
pulseaudio needs to pick the initial profile for the card. Unavailable
profiles shouldn't be picked, but module-alsa-card sometimes marked
unavailable profiles as available, causing bad initial profile choices.
This patch changes module-alsa-card so that it marks all profiles
unavailable whose all output ports or all input ports are unavailable.
Previously only those profiles were marked as unavailable whose all
ports were unavailable. For example, if a profile contains one sink and
one source, and the sink is unavailable and the source is available,
previously such profile was marked as available, but now it's marked as
unavailable.
BugLink: https://bugs.freedesktop.org/show_bug.cgi?id=102902
This is a working implementation of a build with meson. The server,
utils, and most modules build with this, and it is possible to run from
a build tree and play/capture audio on ALSA devices.
There are a number of FIXMEs, of course, and a number of features that
need to be enabled (modules, dependencies, installation, etc.), but this
should provide everything we need to get there relatively quickly.
To use this, install meson (distro package, or mesonbuild.com) and run:
$ cd <pulseaudio src dir>
$ meson <builddir>
$ ninja -C <builddir>
The iec958 output uses device 2 and the iec958 input uses device 0. The
USB configuration in alsa doesn't set up the device numbers correctly,
which is why we need custom configuration in PulseAudio. Ideally this
would be fixed in alsa, but trying to get help for that wasn't
successful.
jack->melem can be null if the jack disappears between probing the card
and the init_jacks() call. I don't know if jacks actually ever disappear
like that (seems unlikely), but this check is in any case needed as long
as init_jacks() has proper handling for the jack disappearing case
(rather than just an assert).
There was a crash report[1] that indicated that card_suspend_changed()
called report_jack_state() with a null melem. I don't know if that was
because the jack actually disappeared, or is there some other bug too.
[1] https://bugs.freedesktop.org/show_bug.cgi?id=104385
The commit "alsa-util: Set ALSA report_delay flag in pa_alsa_safe_delay()"
broke the build on ALSA versions below 1.1.0 because the time stamp
configuration function was introduced in 1.1.0.
This patch makes the usage of snd_pcm_status_set_audio_htstamp_config()
dependent on ALSA version.
The current code does not call snd_pcm_status_set_audio_htstamp_config()
to configure the way timestamps are updated in ALSA. In kernel 4.14 and
above a bug in ALSA has been fixed which changes timmestamp behavior.
This leads to inconsistencies in the delay reporting because the time
stamp no longer reflects the time when the delay was updated if the
ALSA report_delay flag is not set. Therefore latencies are not calculated
correctly.
This patch uses snd_pcm_status_set_audio_htstamp_config() to set the
ALSA report_delay flag to 1 before the call to snd_pcm_status(). With
this, time stamps are updated as expected.
Reconfiguration callback should also be set in case of avoiding resampling
option. This patch set the callback for every case because the callback
has already conditions to leave if it is not needed.
Also unnecessary codes of setting alternate sample rate to 0 are removed.
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>
The alsa sink calls pa_sink_suspend() from the set_port() callback.
pa_sink_suspend() can only be called from the main thread, but the
set_port() callback was often called from the IO thread. That caused an
assertion to be hit in pa_sink_suspend() when switching ports.
Another issue was that pa_sink_suspend() called the set_port() callback,
and if the callback calls pa_sink_suspend() again recursively, nothing
good can be expected from that, so the thread mismatch was not the only
problem.
This patch moves the mixer syncing logic out of pa_sink/source_suspend()
to be handled internally by the alsa sink/source. This removes the
recursive pa_sink_suspend() call. This also removes the need to have the
mixer_dirty flag in pa_sink/source, so the flag and the
pa_sink/source_set_mixer_dirty() functions can be removed.
This patch also changes the threading rules of set_port(). Previously it
was called sometimes from the main thread and sometimes from the IO
thread. Now it's always called from the main thread. When deferred
volumes are used, the alsa sink and source still have to update the
mixer from the IO thread when switching ports, but the thread
synchronization is now handled internally by the alsa sink and source.
The SET_PORT messages are not needed any more and can be removed.
BugLink: https://bugs.freedesktop.org/show_bug.cgi?id=104761
The suspend cause isn't yet used by any of the callbacks. The alsa sink
and source will use it to sync the mixer when the SESSION suspend cause
is removed. Currently the syncing is done in pa_sink/source_suspend(),
and I want to change that, because pa_sink/source_suspend() shouldn't
have any alsa specific code.
There are no behaviour changes, the code from almost all the SET_STATE
handlers is moved with minimal changes to the newly introduced
set_state_in_io_thread() callback. The only exception is module-tunnel,
which has to call pa_sink_render() after pa_sink.thread_info.state has
been updated. The set_state_in_io_thread() callback is called before
updating that variable, so moving the SET_STATE handler code to the
callback isn't possible.
The purpose of this change is to make it easier to get state change
handling right in modules. Hooking to the SET_STATE messages in modules
required care in calling pa_sink/source_process_msg() at the right time
(or not calling it at all, as was the case on resume failures), and
there were a few bugs (fixed before this patch). Now the core takes care
of ordering things correctly.
Another motivation for this change is that there was some talk about
adding a suspend_cause variable to pa_sink/source.thread_info. The
variable would be updated in the core SET_STATE handler, but that would
not work with the old design, because in case of resume failures modules
didn't call the core message handler.
build_pollfd() isn't likely to fail, but if it does, pa_sink/source_put()
will crash on an assertion failure. I haven't seen such crash happening,
this is just something that I noticed while studying the state change
code.
The suspend cause isn't yet used by any of the callbacks. The alsa sink
and source will use it to sync the mixer when the SESSION suspend cause
is removed. Currently the syncing is done in pa_sink/source_suspend(),
and I want to change that, because pa_sink/source_suspend() shouldn't
have any alsa specific code.
This removes the need to hardcode the ELD device index in the path
configuration. The hardcoded values don't work with the Intel HDMI LPE
driver.
BugLink: https://bugs.freedesktop.org/show_bug.cgi?id=100488
We have so far assumed that HDMI always uses device indexes 3, 7, 8, 9,
10, 11, 12 and 13. These values are hardcoded in the path configuration.
The Intel HDMI LPE driver, however, uses different device numbering
scheme. Since the indexes aren't always the same, we need to query the
hw device index from ALSA.
Later patches will use the queried index for HDMI jack detection and ELD
information reading.
BugLink: https://bugs.freedesktop.org/show_bug.cgi?id=100488
This is based on a patch by Rolo <rolo@wildfish.com> that replaced the
old ID with the new one. I deemed it better to leave the old ID in use
(I can't verify if the old ID was correct or not).
The original commit message:
Every time I reinstall or update Ubuntu I have to make this change
to get it to recognise my Native Instruments Traktor Audio 6
external soundcard.
Each time I remember the change by hunting down this forum post in
German,
https://forum.ubuntuusers.de/topic/traktor-audio-6-erkannt-aber-nicht-anwaehlbar/3/#post-8759808
(I don't speak German).
I'm not sure if the ID is just incorrect or if perhaps the hardware
identifies itself differently on slightly different models, so
perhaps we need to duplicate the line - I'm well outside of my
comfort zone here and I know barely anything about how hardware
works on Linux but figured if it helps me it would help others so I
should put it forward.
Thanks!
The Intel HDMI LPE driver works in a peculiar way when the HDMI cable is
not plugged in: any written audio is immediately discarded and underrun
is reported. That resulted in an infinite loop, because PulseAudio tried
to keep the buffer filled, which was futile since the written audio was
immediately consumed/discarded.
This patch adds special handling for the LPE driver: if the active port
of the sink is unavailable, the sink suspends itself. A new suspend
cause is added: PA_SUSPEND_UNAVAILABLE.
BugLink: https://bugs.freedesktop.org/show_bug.cgi?id=100488
