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>
This patch is in preparation of allowing virtual sinks to specify their own
max_rewind limit.
Currently, virtual sinks cannot specify their max_rewind limit, but just copy
the value from the master sink. This may not be correct, if the DSP code of the
virtual sink has limited (or no) rewinding capability.
Because the DSP code of the virtual sink is rewound in the process_rewind()
callback of the sink input, it must be ensured, that rewinding a sink input
to the master of a virtual sink is limited similar to rewinding a sink.
There are two remaining exceptions:
1) If an underrun is detected. In that case, the filter should be reset anyway.
2) When the sink input of the filter is moved and attached to the destination
sink.
The move case is handled without involvement of the implementer, so the implementer
can only receive a rewind larger than max_rewind when the filter should be reset
anyway.
All existing virtual sinks do not distinguish between reset and rewind of the
filter.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/120>
If the output implements a process_rewind() callback, the resampler delay is
not taken into account. This leads to glitches during volume changes when
source and source output rates differ.
This patch fixes the problem.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/120>
The introduction of the history queue makes it possible to implement moving
of streams without involving the implementer. Instead of dropping all data
from the render memblockq and requesting the implementer to rewrite the
data, the render memblockq is now reconstructed from the history queue.
Additionally, the render queue will be filled with silence matching the
amount of audio that is left playing on the old sink to avoid playing
the same audio twice.
This patch slightly breaks moving for virtual sinks because they do not
yet include the resampler delay in their latency reports. This will be
fixed in a different patch set.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/120>
This patch uses the two previous patches to implemnt pseudo-rewinding for the
resamplers by feeding some old data into the resampler after a reset. This is
necessary because PA is using external resamplers that do not implement
rewinding.
To get exactly the same output data from the resampler after a rewind if possible,
the matching period is calculated. This is the number of input samples that produces
an integral number of output samples. After the matching period, the resampler state
repeats. If the matching period is not too large, feeding history into the resampler
will start at a point that is a multiple of the matching period back in time. Then
the resampler will produce exactly the same samples.
The PA_RESAMPLER_MAX_HISTORY value has been replaced by PA_RESAMPLER_MAX_DELAY_USEC
and the required number of history samples is calculated from the sink input sample
rate. The number of history samples can be as large as about 12500.
This fixes glitches during volume changes when the sink runs on a rate different
from the sink input rate.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/120>
The pa_resampler_get_delay() function allows to retrieve the current resampler
delay in input samples for all supported resamplers. The return value is a double
to maintain precision when using variable rate resamplers. Because in many places
the delay is needed in usec, pa_resampler_get_delay_usec() was also supplied.
The speex resampler now skips leading zero samples to provide meaningful delay values.
In the next patch, the pa_resampler_prepare() function will be used to train the
resampler after a rewind. It takes data from a history memblockq and runs it through
the resampler. The output data is discarded.
To make this logic possible, the soxr resampler had to be converted to use variable
rate. The fixed rate version has a variable delay, therefore the logic above could
not be applied. Additionally, with fixed rate, the delay is larger than 150ms in
some situations, while with variable rate the delay is fixed and comparable to the
other resamplers.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/120>
A new memblockq is added to the sink input code to keep some history of the
input data. The queue is kept in sync with the render memblockq. The old input
data will be used to prepare the resampler after a rewind.
pa_resampler_request() and pa_resampler_result() have been changed to round
as good as possible to avoid loosing or duplicating samples during rewinds.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/120>
When sink is suspended for reconfiguration changing sample spec, upon resume
internal thread_info max_request and max_rewind are out of date and possibly
not aligned to frame size anymore.
Recalculate thread max_request and max_rewind before resuming sink.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/658>
When ICE I/O error occurs ICE connection is closed via IceCloseConnection.
This causes crash while releasing session manager connection later because
this ICE connection was initiated and is managed by session manager, and it will
attempt to close this ICE connection again.
Fix this by closing session manager connection instead.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/650>
Commit 7fd89e491 ("bluetooth: Try to reconnect SCO") introduces a call
to pa_msleep but failed to include the header, resulting in a:
../pulseaudio/src/modules/bluetooth/backend-native.c: In function ‘sco_acquire_cb’:
../pulseaudio/src/modules/bluetooth/backend-native.c:336:17: warning: implicit declaration of function ‘pa_msleep’ [-Wimplicit-function-declaration]
336 | pa_msleep(300);
| ^~~~~~~~~
(Un)fortunately this implicit declaration gets resolved at link-time,
otherwise the issue would have been caught sooner.
Fixes: 7fd89e491 ("bluetooth: Try to reconnect SCO")
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/651>
Just like the manpage these written out log levels should correspond to
the numerical values listed before, intead of being in the opposite
order and provoking thoughts of the relation being the wrong way around
where 0=debug and 4=error.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/647>
Requiring user to invoke send-message with correctly quoted parameters string
is not good for usability. Wrap parameters string into JSON string and try
again. If that works, log a warning use wrapped JSON string with parameters.
As an example these two commands will now invoke the same action:
pactl send-message /card/bluez_card... switch-codec '"CODECNAME"'
pactl send-message /card/bluez_card... switch-codec CODECNAME
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/648>
Check whether a Bluetooth profile is supported both by the remote device
and the local host before creating a card profile for it.
This is useful when some of the media profiles have not been registered
with bluetoothd because ex., oFono is not running and the headset
backend is not available.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/638>
The remote device list of UUIDs reflects which profiles are supported by
the remote device alone. We currently rely solely on this list to decide
if a certain card profile is supported, and thus should be created and
get connected.
This used to be accurate when the Bluetooth modules were first written,
but now BlueZ is more dynamic and local profile support can be added or
removed during runtime. The adapter's list of UUIDs is an accurate
representation of the profiles supported by the local host at a certain
moment.
This commit combines the list of UUIDs supported by remote device and
the list of UUIDs supported by the local host to determined whether a
Bluetooth profile is actually supported or not, and whether it should be
expected to get connected during device connection.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/638>
When the SCO connection is in use, if you disconnect first and then connect,
the SCO connection will occasionally fail, and the Bluetooth error code is 42
(0x2A in hexadecimal). This is usually because an error occurred when the SCO
connection was initiated, we need to try to reconnect to optimize the handling
of this problem. The log returned by the kernel is as follows:
Bluetooth: sco_connect_cfm: hcon 0000000003328902 bdaddr 40:ef:4c:0c:11:f0 status 42
Bluetooth: sco_sock_connect status is -38
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/622>
When module-combine-sink is used to create virtual surround card it is expected
that each slave channel receives data for exactly one combined sink channel.
Currently this is not the case because module-combine-sink would follow
core->disable_remixing setting. Usually this means that multiple channels of
combined sink are getting remixed into slave channel, and there is no option to
disable this behavior just for combined sink.
Improve this by implementing "remix" modarg for module-combine-sink,
default to original behavior (follow core->disable_remixing setting).
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/627>
