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>
We can provide a better overall user experience with Bluetooth cards by
always choosing the higher audio quality profile (A2DP) by default and
updating the profile selection dynamically according to which streams
are active at a certain moment. The default initial selection has been
addressed by "85daab272 bluetooth: set better priorities for profiles"
and the dynamic profile selection is covered by module-bluetooth-policy.
In addition, module-card-restore's database entries for Bluetooth devices
are retained after a device is removed from the system, leading to the
previously selected profile being restored after a new pairing with the
same device, with no way for the user to erase this memory and reset the
default profile except manually fiddling with module-card-restore's
database.
This commit adds a module argument to have module-card-restore ignore
Bluetooth profiles and this behavior is set as default.
We recently changed the umask of the daemon from 022 to 077, which broke
module-pipe-sink in the system mode, because nobody was allowed to read
from the pipe.
module-pipe-source in the system mode was probably always broken,
because the old umask of 022 should prevent anyone from writing to the
pipe.
This patch uses chmod() after the file creation to set the permissions
to 0666, which is what the fkfifo() call tried to set.
Bug link: https://bugs.freedesktop.org/show_bug.cgi?id=107070
Having a single level macro for stringizing LADSPA_PATH doesn't work,
because the '#' preprocessor operator doesn't expand any macros in its
parameter. As a result, we used the string "LADSPA_PATH" as the search
path, and obviously no plugins were ever found.
This adds a two-level macro in macro.h and uses that to expand and
stringize LADSPA_PATH.
Bug link: https://bugs.freedesktop.org/show_bug.cgi?id=107078
Add configuration option 'stream_name' for stream/session name so user
will see it on receiver side as RTP Strean ($stream_name)
ex: load-module module-rtp-send source=rtp.monitor stream_name=MyServerMedia
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.
pa_sink_input_get_state() and pa_source_output_get_state() just return
the state variable. We can as well access the state variable directly.
There are no behaviour changes, except that some filter sources accessed
the main thread's state variable from their push() callbacks. I fixed
them so that they use the thread_info.state variable instead.
The only thing that the drained state was being used for was "pacmd
list-sink-inputs". In all other cases the drained and running states
were treated as equivalent. IMHO, this usage doesn't justify the
complexity that the additional state brings.
This patch was inspired by a bug report[1] that pointed out an error in
an if condition in pa_sink_input_set_state_within_thread(). The buggy
code is now removed altogether.
[1] https://bugs.freedesktop.org/show_bug.cgi?id=106982
When the user manually switches the profile of a bluetooth headset from
"off" to "a2dp_sink", the port availability changes from "unknown" to
"yes", which triggered a recursive profile change in
module-switch-on-port-available. Such recursivity isn't (and possibly
can't) be handled well (that is, PulseAudio crashed), so let's avoid
doing bluetooth profile changes from module-switch-on-port-available
(they're useless anyway).
Bug link: https://bugs.freedesktop.org/show_bug.cgi?id=107044
Now that both backend-native and backend-ofono can coexist and
backend-ofono is always loaded, even on systems without oFono, failing
to register with org.ofono is not necessarily an error.
This lowers the failure message log level from error to info.
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.
BlueZ 4 is no longer supported by BlueZ community for a long long time,
also by moving to BlueZ 5 it should make it even more clearer that
BlueZ 4 is no longer an option.
Attempt to use Acquire method if available since it directly returns
the fd in the reply or an error if that the connection could not be
created while Connect offer neither of these and depend on
NewConnection to deliver the fd.
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
Currently the loopback module uses sample spec and channel map of the
sink by default. It leads to double resample if source and sink sample
specs are different and no rate/format specified in arguments. This
patch causes the source sample spec and channel map to be used by
default.
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.
sco_process_render does not unref the memblock when it encounters an error.
This patch fixes the issue. It also changes the return value to 1 in the case
of EAGAIN. Because the data was already rendered and cannot be re-sent, we
have to discard the block.
Because the modified EAGAIN handling prevents the log message about EAGAIN
after POLLOUT from being printed, the log message was moved to
a2dp/sco_process_render().
The rewrite of the thread function does not change functionality much,
most of it is only cleanup, minor bug fixing and documentation work.
This patch also changes the send buffer size for a2dp sink to avoid lags
after temporary connection drops, following the proof-of-concept patch
posted by Dmitry Kalyanov.
Bug-Link: https://bugs.freedesktop.org/show_bug.cgi?id=58746
Additionally the patch changes the fixed latency for HSP playback from 125
to 25 ms. Tests showed that this produces better audio sync, which is
expected as HSP should have smaller latency than A2DP.
