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.
If the given proplist is NULL, the function creates a new (empty)
proplist. That caused a compiler warning after the constification, which
is why the new proplist is now created using a separate variable.
Existing documentation was unclear about which property list would be the
one changed (merged into), making it seem (along with the non-const
proplist pointer param, which needs changing seperately), that the proplist
object for which a pointer is given will be the one merged into, instead of
the internal cached entry's proplist.
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.
set_nonblock() will always set the file descriptor to non-blocking,
regardless of the nonblock argument.
This patch fixes the issue by passing the correct argument to the
fcntl() call. The bug had no impact because there is only one caller
of pa_make_fd_block() in poll-win32.c
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.
The volume_map variable was initialized only for PCM streams, but the
variable was passed to pa_cvolume_remap() also for non-PCM streams. The
volume remapping is never necessary for passthrough streams (PCM or
not), because no volume will be applied anyway, so let's skip the
pa_cvolume_remap() call for all passthrough streams.
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.
