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>
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>
remixing-produce-lfe controls upmixing, and remixing-consume-lfe
controls downmixing. The motivation is that a user might want to
synthesize LFE while playing stereo audio on his/her 5.1 speakers,
but at the same time follow the industry recommendation to omit
the LFE channel when producting a stereo downmix (e.g. for headphones)
from 5.1 content. Or the other way round.
Fixes: #753.
Almost all distributions patch the configuration to disable
flat-volumes, because users tend to find the concept confusing (and it
also causes nasty surprises when some application pushes the volume to
100%). Let's remove the need for patching and disable the feature by
default.
Fixes: https://gitlab.freedesktop.org/pulseaudio/pulseaudio/issues/691
The suspend-sink and suspend-source documentation for pacmd was quite
terse, so I copied the more complete documentation from pactl. I
couldn't resist doing some other minor edits along the way too.
Bug-link: https://bugs.freedesktop.org/show_bug.cgi?id=105907
This adds an "avoid-resampling" option to daemon.conf that makes the
daemon try to use the stream sample rate if possible (the device needs
to support it, which currently only ALSA does), and there should not be
any other stream connected).
This should enable some of the "audiophile" use-cases where users wish
to play high sample rate audio files without resampling.
We still will do conversion if sample formats don't match, though. This
means that if you want to play 96 kHz/24 bit audio without any
modification the default format will need to be set to be 24-bit as
well. This will force all streams to be upconverted, which, other than
the wasted resources, should be relatively harmless.
