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>
