Currently, virtual sinks and sources are not suspended when the master sink
or source is suspended. To implement this, the slave must be able to track
the suspend cause of the master.
With this patch, the sink input suspend callback will not only be called
when the sink or source is changing state, but also when the suspend cause
changes. Similar to the set_state_in_*_thread_cb() functions, the suspend
callback receives a state and a suspend cause as additional arguments.
Because the new state and suspend cause of the sink or source have already
been set, the old values are passed to the callback.
pa_split_in_place() and pa_split_spaces_in_place() are modifed
to use size_t type instead of integer type.
alsa-ucm.c is revised according to this change.
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>
pa_card_profile_set_available needs to check if the card is linked
before firing PA_CORE_HOOK_CARD_PROFILE_AVAILABLE_CHANGED, so callbacks
connected to it receive a fully initialized card object.
This fixes a crash introduced by commit 30a551bbc
"switch-on-port-available: Check if we need to change the active
profile".
It is helpful to improve reproducibility build [1] since
PA_SRCDIR/PA_BUILDDIR contains build path,
--disable-running-from-build-tree could drop these macros at
precompilation.
[1] https://reproducible-builds.org/
Signed-off-by: Hongxu Jia <hongxu.jia@windriver.com>
Sample format(e.g. 16 bit, 24 bit) was not considered even if the
avoid-resampling option is set or the passthrough mode is used.
This patch checks both sample format and rate of a stream to
determine whether to avoid resampling in case of the option is set.
In other word, it is possble to use the stream's original sample
format and rate without resampling as long as these are supported
by the device.
pa_sink_input_update_rate() and pa_source_output_update_rate() are
renamed to pa_sink_input_update_resampler() and pa_source_output
_update_resampler() respectively.
functions are added as below.
pa_sink_set_sample_format(), pa_sink_set_sample_rate(),
pa_source_set_sample_format(), pa_source_set_sample_rate()
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>
mpv and vlc play "normal" 5.1 AC3 and DTS files as if they had a
"5.1 (Side)" layout. Which is fine and consistent with ITU
recommendations if the user has a proper 7.1 system. But if the user
actually has a 5.1 system, PulseAudio will try to remap, poorly, between
the "5.1 (Side)" and "5.1" layouts, sending either an average between
front and rear channels, or an attenuated version of that average,
depending on the remixing-use-all-sink-channels setting.
This is not desired, the "Side" channels should be sent to "Rear", it is
only an unfortunate nomenclature confusion.
This patch does not fix 5.1 <-> 7.1 remixing.
Signed-off-by: Alexander E. Patrakov <patrakov@gmail.com>
This is based on the initial protocol_native library that is already
defined, and then by looking at the Makefile.am to work out the
dependencies.
It's not clear whether we really need database_c_args, maybe there's
things that can be simplified.
Signed-off-by: Arnaud Rebillout <arnaud.rebillout@collabora.com>
We move over helper functions to get rate, channels, channel map and
sample format (if PCM) in the public API, so users of the extended API
are more easily able to pull out these values from pa_format_info.
Having a single level macro for stringizing LADSPA_PATH doesn't work,
because the '#' preprocessor operator doesn't expand any macros in its
parameter. As a result, we used the string "LADSPA_PATH" as the search
path, and obviously no plugins were ever found.
This adds a two-level macro in macro.h and uses that to expand and
stringize LADSPA_PATH.
Bug link: https://bugs.freedesktop.org/show_bug.cgi?id=107078
pa_sink_get_state() and pa_source_get_state() just return the state
variable. We can as well access the state variable directly.
There are no behaviour changes, except that module-virtual-source
accessed the main thread's sink state variable from its push() callback.
I fixed the module so that it uses the thread_info.state variable
instead. Also, the compiler started to complain about comparing a sink
state variable to a source state enum value in protocol-esound.c. The
underlying bug was that a source pointer was assigned to a variable
whose type was a sink pointer (somehow using the pa_source_get_state()
macro confused the compiler enough so that it didn't complain before).
I fixed the variable type.
pa_sink_input_get_state() and pa_source_output_get_state() just return
the state variable. We can as well access the state variable directly.
There are no behaviour changes, except that some filter sources accessed
the main thread's state variable from their push() callbacks. I fixed
them so that they use the thread_info.state variable instead.
The only thing that the drained state was being used for was "pacmd
list-sink-inputs". In all other cases the drained and running states
were treated as equivalent. IMHO, this usage doesn't justify the
complexity that the additional state brings.
This patch was inspired by a bug report[1] that pointed out an error in
an if condition in pa_sink_input_set_state_within_thread(). The buggy
code is now removed altogether.
[1] https://bugs.freedesktop.org/show_bug.cgi?id=106982
Current code does not check whether pa_play_file call failed. Hence no error is
reported in the cli interface if playback failed because e.g. file isn't
readable by the daemon.
Previously, the "avoid-resampling" option of daemon.conf is to make the
daemon try to use the stream sample rate if possible for all sinks or
sources.
This patch applies this option to module-udev-detect and module-alsa-card
as a module argument in order to override the default value of daemon.conf.
As a result, user can use this argument for more fine-grained control.
e.g.) set it false in daemon.conf and set it true for module-udev-detect
or a particular module-alsa-card in default.pa.(or vice versa)
To set it, use "avoid_resampling=true or false" as the module argument.
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>
This should make it easier for clients to elevate their audio threads to
real time priority without having to dig through much through specific
system internals.
set_nonblock() will always set the file descriptor to non-blocking,
regardless of the nonblock argument.
This patch fixes the issue by passing the correct argument to the
fcntl() call. The bug had no impact because there is only one caller
of pa_make_fd_block() in poll-win32.c
This is a working implementation of a build with meson. The server,
utils, and most modules build with this, and it is possible to run from
a build tree and play/capture audio on ALSA devices.
There are a number of FIXMEs, of course, and a number of features that
need to be enabled (modules, dependencies, installation, etc.), but this
should provide everything we need to get there relatively quickly.
To use this, install meson (distro package, or mesonbuild.com) and run:
$ cd <pulseaudio src dir>
$ meson <builddir>
$ ninja -C <builddir>
The volume_map variable was initialized only for PCM streams, but the
variable was passed to pa_cvolume_remap() also for non-PCM streams. The
volume remapping is never necessary for passthrough streams (PCM or
not), because no volume will be applied anyway, so let's skip the
pa_cvolume_remap() call for all passthrough streams.
Reconfiguration callback should also be set in case of avoiding resampling
option. This patch set the callback for every case because the callback
has already conditions to leave if it is not needed.
Also unnecessary codes of setting alternate sample rate to 0 are removed.
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>
The alsa sink calls pa_sink_suspend() from the set_port() callback.
pa_sink_suspend() can only be called from the main thread, but the
set_port() callback was often called from the IO thread. That caused an
assertion to be hit in pa_sink_suspend() when switching ports.
Another issue was that pa_sink_suspend() called the set_port() callback,
and if the callback calls pa_sink_suspend() again recursively, nothing
good can be expected from that, so the thread mismatch was not the only
problem.
This patch moves the mixer syncing logic out of pa_sink/source_suspend()
to be handled internally by the alsa sink/source. This removes the
recursive pa_sink_suspend() call. This also removes the need to have the
mixer_dirty flag in pa_sink/source, so the flag and the
pa_sink/source_set_mixer_dirty() functions can be removed.
This patch also changes the threading rules of set_port(). Previously it
was called sometimes from the main thread and sometimes from the IO
thread. Now it's always called from the main thread. When deferred
volumes are used, the alsa sink and source still have to update the
mixer from the IO thread when switching ports, but the thread
synchronization is now handled internally by the alsa sink and source.
The SET_PORT messages are not needed any more and can be removed.
BugLink: https://bugs.freedesktop.org/show_bug.cgi?id=104761
