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>
Currently module-loopback detects underruns even if sink_input_pop_cb()
was not yet called twice and initial latency adjustments are active.
This leads to unnecessary rewind requests.
This patch delays detecting underruns until the initial adjustments
are done.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/56>
Taking addresses of fields in a packed struct are not guaranteed to be
aligned, resulting in warnings such as:
../src/pulsecore/shm.c: In function 'sharedmem_create':
../src/pulsecore/shm.c:198:25: error: taking address of packed member of 'struct shm_marker' may result in an unaligned pointer value [-Werror=address-of-packed-member]
198 | pa_atomic_store(&marker->pid, (int) getpid());
| ^~~~~~~~~~~~
The struct already has its fields and types laid out in such a way that
the desired packing (without padding) is guaranteed - enforce this with
a `static_assert` to get rid of the unaligned pointer warning.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/653>
GCC warns on all these `fail_unless` calls:
warning: too many arguments for format [-Wformat-extra-args]
`fail_unless` only takes an expression and optionally a string literal
as message with formatting args. Passing NULL for this message should
not be necessary as indicated by all the other tests not passing it
either.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/653>
This patch adds an alternative time smoother implementation based on the theory
found at https://www.freedesktop.org/software/pulseaudio/misc/rate_estimator.odt.
The functions were written to replace the current smoother functions nearly on
a one-to-one basis, though there are a few differences:
- The smoother_2_put() function takes a byte count instead of a sound card
time as argument. This was changed because in most places a sample count
was converted to a time before passing it to the smoother.
- The smoother needs to know sample rate and frame size to convert byte
counts to time.
- A smoother_2_get_delay() function was added to directly retrieve the stream
delay from the smoother.
- A hack for USB devices was added which works around an issue in the alsa
latency reports for USB devices.
The smoother delivers much better precision than the current implementation.
For results, see the document referenced above.
The new functions are still unused. The following patches will convert all
callers of the smoother functions so that they can use both smoother
implementations, depending on a configure option.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/55>
This patch adds a test program that generates a square wave of a given frequency,
length and sample rate. This is then resampled to another rate, rewound and the
rewound part is run through the resampler again. After that, the results of the
first and second resampler pass are compared.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/120>
This patch is in preparation of allowing virtual sinks to specify their own
max_rewind limit.
Currently pa_sink_set_max_rewind_within_thread() simply sets the value of
max_rewind and informs the sink inputs about the new value. Virtual sinks
may however provide their own limit on max_rewind.
This patch allows to query the active sink inputs for the max_rewind value
they support and sets max_rewind to the minimum supported value. This way,
the max_rewind value from the virtual sinks can be communicated to the master
sink.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/120>
