RFC 4566 states that SDP record is terminated with CRLF, and parsers should be
able to accept records terminated with just LF. Pulseaudio only accepts LF here.
Fix this by accepting both CRLF and LF terminators.
The combine sink used the current time and counter when calculating
the latency if smoother_2 was enabled. This lead to wrong latency
reports. This patch fixes the problem by using the snapshot time
and counter instead.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/711>
Bluetooth transport layer already allows for packets larger than mSBC frame, and
there are up to 1 + MTU / (mSBC packet size) complete frames to be decoded from
each incoming SCO packet.
Now decoder fails when there is more than one complete frame available, which
could happen if MTU size is larger than 1.5 * (mSBC packet size) = 90
Fix this by adding a loop over avialable frames, and adjust decoded buffer size
to allow decoding up to 1 + MTU / (mSBC packet size) frames at once.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/706>
When the module is loaded and avahi_client_new() fails because the client cannot
connect, a shutdown of the module is scheduled. In parallel, the client_callback
is called with AVAHI_ERR_DISCONNECTED and another connection attempt is made
which also fails and triggers a second unload of the module. This crashes PA,
because there is already an unload in progress.
This patch fixes the problem by checking if an unload is already scheduled.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/710>
The timestamp used for updating the smoother was taken at the wrong time.
It may take some time until an async message is executed (measured up to
2ms), therefore the timestamp used to update the smoother must be taken
before the message is executed and not inside the message.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/705>
When the tunnel modules had no connection and a re-init was pending, the module
could be unloaded without cancelling the pending re-init. When the timer expired
in that situation, this lead to a crash. This patch fixes the problem by keeping
a reference when the module is scheduled to be re-initialized.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/705>
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 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>
The hw: device can be addressed using the card index (hw:0)
or the card identifier (ASCII string - hw:Loopback). Both
mixers are equal.
The previous code was fine for the mixers without the UCM
private prefixes (_ucmXXXX). Make code more robust, create
two aliased mixer structures in the mixers array.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/673>
This makes it possible to define multiple sinks/sources on detection
of the jack server. This allows one to for example create a separate
sink for conferencing software and route that in jack to another
channel on their audio interface.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/669>
Even though the file name is currently behringer-umc22.conf, the USB ID
actually belongs to Texas Instruments PCM2902, which is a generic chip
used in multiple products. Some products have true mono input unlike
Behringer UMC22, which has two mono inputs combined into one stereo PCM
device.
This patch removes the "mono,mono" mapping from Behringer UMC22, which
hopefully won't be missed too much (there are still "mono,aux1" and
"aux1,mono" mappings available for mono recording).
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/667>
If the preferred ports are not set in this function, the
entrys_equal() always returns false in the card_put_hook_callback().
This will make the entry be written into the metadata and the
preferred ports will be cleaned by a mistake.
And we met a hdmi audio bug which has sth to do with this issue, on
the machines with the legacy HDA audio driver, the hdmi port has lower
priority than speaker, users need to manually select the hdmi to be
active output port, then the preferred output port is hdmi for this
sound card, after reboot, the card_put_hook_callback() in the
module-card-restore.c will be called and the preferred ports are
cleaned by a mistake, then the hdmi output port or hdmi sink couldn't
switch to be active after reboot or resume automatically. That is
because the preferred ports are cleaned and hdmi port has lower
priority than speaker, the profile_good_for_output() in the
module-switch-on-port-available.c always returns false.
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Change d7f95170a1 added a dependency on device
adapter pointer being valid while checking if bluetooth profile is supported by
device.
When adapter object is released, each device holding pointer to adapter being
released is notified to reset that to NULL. Since adapter objects are released
first when discovery object is unreferenced, each device will have adapter
pointer reset before the time device objects are released.
Fix observed crash by examining device adapter pointer. If it is NULL report
that device does not support any bluetooth profile instead of looking at UUIDs
supported by adapter.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/646>
Add a log_interval parameter to control the amount of logging. Default is
no logging. Like for adjust_time, the parameter is a double to allow values
below 1s.
If the log interval is too small, logging will occur on every iteration.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/56>
The adjust_time parameter is changed to double to allow better granularity
and adjust times below 1s. This may be useful for a better latency control,
although with alsa devices and the current smoother code no significant
improvement could be found for values below 500ms.
This patch also changes the default adjust time to 1s, the old value of 10s
does not allow a tight control of the end to end latency and would lead to
unnecessary jitter.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/56>
The previous patch slows down initial convergence. Therefore do not use
the controller weight until we can assume that we reached an equilibrium.
Because it takes some time before the reported latency values are reliable,
assume that a steady state is reached when the target latency has been
crossed twice.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/56>
In many situations, the P-controller is too sensitive and therefore exhibits rate hunting.
To avoid rate hunting, the sensibility of the controller is set by the new parameter
adjust_threshold_usec. The parameter value is the deviation from the target latency in usec
which is needed to produce a 1 Hz deviation from the optimum sample rate.
The default is set to 250 usec, which should be sufficient in most cases. If the accuracy
of the latency reports is bad and rate hunting is observed, the parameter must be increased,
while it can be lowered to achieve less latency jitter if the latency reports are accurate.
More details at
https://www.freedesktop.org/software/pulseaudio/misc/rate_estimator.odt
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/56>
The current loopback controller can produce a rate jump of up to 1% at startup,
which may be audible. To prevent large initial jumps, a second controller is
introduced, which produces a rate, that is not more than 2‰ away from the last
rate. Only during the startup phase, the rates produced by this controller will
be nearer to the base rate than those produced by the original controller.
Therefore choosing the rate which is nearer to the base rate will ensure that
the secondary controller only moderates the startup phase and has no influence
during continued operation.
The maximum step size of the original controller after the initial jump is
limited to 2.01‰ of the base rate, see documentation at
https://www.freedesktop.org/software/pulseaudio/misc/rate_estimator.odt
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/56>
