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>
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 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>
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>
Recently we found an issue of output volume on speaker and headphone,
they should have their own volume but in practice they share one
output volume.
This issue happens on the laptops which use the ucm2 sof-hda-dsp,
originally the speaker has output volume A while the headphone has the
output volume B, suppose the speaker is the active port at the moment
and the output volume is A, users plug a headphone to the jack and the
headphone becomes the active port, in this process, ucm_set_port()
calls _disdev/_enadev which triggers the io_mixer_callback(), in the
meanwhile, the module_device_restore will restore the headphone's
volume to B, it will call set_volume_cb() to set the volume to B, but
this value is not written to hw immediately, during the time of
waiting for the B to be written to the hw, the io_mixer_callback()
calls get_volume_cb(), it reads hw volume and gets the volume A, then
it overrides the output volume to A, this results in the headphone
gets the volume A instead of B.
If a machine doesn't use the ucm, this issue will not happen since the
set_port_cb() will not trigger the io_mixer_callback(). If the ports
don't belong to the same sink/source, this issue also doesn't happen.
BugLink: http://bugs.launchpad.net/bugs/1930188
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/577>
* Minimal implementation of --system on win32.
* Wrap main with a Windows Service on win32 (with a fallback to
running it directly).
* Update PA_SYSTEM_{RUNTIME,STATE,CONFIG}_PATH and HOME dynamically
on Windows (overrides the build config, similar to the existing
config path replacement logic).
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/549>
New card database entry version 5 for card profile is sticky flag.
New messaging API handlers set-profile-sticky and get-profile-sticky.
When card profile is sticky, always restore it even if it is unavailable,
and prevent switching from it when ports become unavailable.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/568>
The hooks are fired when the preferred device changes. This is useful
for module-stream-restore.
I added new set_preferred_sink/source() functions for firing the hooks.
The functions also log the preferred device changes.
There was already pa_sink_input_set_preferred_sink(), but that had a
side effect of moving the stream, so I needed a new function. Since it
can be confusing when the two similarly named functions should be
called, I added a comment for pa_sink_input_set_preferred_sink() that
explains the different situations.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/535>
When the source-output of a virtual source with volume sharing disabled is moved,
the source output volume is reset to 100%. This patch fixes the problem by
applying the virtual source volume to the source-output after the move.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/509>
WSAStartup was not being called for pacat and pactl built with meson,
causing them to fail in pa_mainloop_new with "cannot create wakeup
pipe". This issue also affects other applications linking to libpulse
other than the pulseaudio daemon, which calls WSAStartup itself.
When built with autotools, WSAStartup would have been called in
DllMain, which is recommended against by the documentation [1].
To fix these issues, the WSAStartup/WSACleanup calls can be moved
into pa_mainloop_new/pa_mainloop_free.
[1] https://docs.microsoft.com/en-us/windows/win32/api/winsock/nf-winsock-wsastartup
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/456>
State database binary file format may depend on system architecture,
for instance gdbm binary format depends on architecture word size,
making x86 and x64 gdbm files incompatible.
If this is the case, it is handled by adding system architecture name to
database file name using automatically configured CANONICAL_HOST string.
Meson build define CANONICAL_HOST to be system architecture name, while
autotools build extends this with vendor and and operating system components.
Switch autotools build to use host_cpu for CANONICAL_HOST to match Meson
configuration. For backwards compatibility always use existing database file
matching CANONICAL_HOST prefix if it exists.
Part-of: <https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/425>
