This was being done automatically by autotools, now we need to manually
specify this for each executable/library with a dependency in a
non-standard directory.
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.
The suspend cause isn't yet used by any of the callbacks. The alsa sink
and source will use it to sync the mixer when the SESSION suspend cause
is removed. Currently the syncing is done in pa_sink/source_suspend(),
and I want to change that, because pa_sink/source_suspend() shouldn't
have any alsa specific code.
There are no behaviour changes, the code from almost all the SET_STATE
handlers is moved with minimal changes to the newly introduced
set_state_in_io_thread() callback. The only exception is module-tunnel,
which has to call pa_sink_render() after pa_sink.thread_info.state has
been updated. The set_state_in_io_thread() callback is called before
updating that variable, so moving the SET_STATE handler code to the
callback isn't possible.
The purpose of this change is to make it easier to get state change
handling right in modules. Hooking to the SET_STATE messages in modules
required care in calling pa_sink/source_process_msg() at the right time
(or not calling it at all, as was the case on resume failures), and
there were a few bugs (fixed before this patch). Now the core takes care
of ordering things correctly.
Another motivation for this change is that there was some talk about
adding a suspend_cause variable to pa_sink/source.thread_info. The
variable would be updated in the core SET_STATE handler, but that would
not work with the old design, because in case of resume failures modules
didn't call the core message handler.
The previous code made the SET_STATE message fail if trigger() failed.
However, trigger() was called after pa_sink/source_process_msg(), which
meant that the main thread that sent the SET_STATE thought that resuming
failed, but nothing was undone in the IO thread, so in the IO thread
things seemed as if the sink/source was successfully resumed. (I don't
use OSS myself, so I don't know what kind of practical problems this
could cause).
Unless some complex undo logic is implemented, I believe it's best to
ignore all failures in trigger(). Most error cases were already ignored,
and the only one that wasn't ignored doesn't seem too serious.
I also moved trigger() to happen before pa_sink/source_process_msg(),
which made it necessary to add new state parameters to trigger(). The
reason for this move is that I want to move the SET_STATE handler code
into a separate callback, and if things are done both before and after
pa_sink/source_process_msg(), that makes things more complicated.
The previous code checked the return value of
pa_sink/source_process_msg() before calling trigger(), but that was
unnecessary, since pa_sink/source_process_msg() never fails when
processing the SET_STATE messages.
This removes the symdef header generation m4 magic in favour of a
simpler macro method, allowing us to skip one unnecessary build step
while moving to meson, and removing an 11 year old todo!
The reported latency of source or sink is based on measured initial conditions.
If the conditions contain an error, the estimated latency values may become negative.
This does not indicate that the latency is indeed negative but can be considered
merely an offset error. The current get_latency_in_thread() calls and the
implementations of the PA_{SINK,SOURCE}_MESSAGE_GET_LATENCY messages truncate negative
latencies because they do not make sense from a physical point of view. In fact,
the values are truncated twice, once in the message handler and a second time in
the pa_{source,sink}_get_latency_within_thread() call itself.
This leads to two problems for the latency controller within module-loopback:
- Truncating leads to discontinuities in the latency reports which then trigger
unwanted end to end latency corrections.
- If a large negative port latency offsets is set, the reported latency is always 0,
making it impossible to control the end to end latency at all.
This patch is a pre-condition for solving these problems.
It adds a new flag to pa_{sink,source}_get_latency_within_thread() to allow
negative return values. Truncating is also removed in all implementations of the
PA_{SINK,SOURCE}_MESSAGE_GET_LATENCY message handlers. The allow_negative flag
is set to false for all calls of pa_{sink,source}_get_latency_within_thread()
except when used within PA_{SINK,SOURCE}_MESSAGE_GET_LATENCY. This means that the
original behavior is not altered in most cases. Only if a positive latency offset
is set and the message returns a negative value, the reported latency is smaller
because the values are not truncated twice.
Additionally let PA_SOURCE_MESSAGE_GET_LATENCY return -pa_sink_get_latency_within_thread()
for monitor sources because the source gets the data before it is played.
Bug 96741 shows a case where an assertion is hit, because
pa_asyncq_new() failed due to running out of file descriptors.
pa_asyncq_new() is used in only one place (not counting the call in
asyncq-test): pa_asyncmsgq_new(). Now pa_asyncmsgq_new() can fail too,
which requires error handling in many places. One of those places is
pa_thread_mq_init(), which can now fail too, and that needs additional
error handling in many more places. Luckily there weren't any places
where adding better error handling wouldn't have been easy, so there are
many changes in this patch, but they are not complicated.
BugLink: https://bugs.freedesktop.org/show_bug.cgi?id=96741
FSF addresses used in PA sources are no longer valid and rpmlint
generates numerous warnings during packaging because of this.
This patch changes all FSF addresses to FSF web page according to
the GPL how-to: https://www.gnu.org/licenses/gpl-howto.en.html
Done automatically by sed-ing through sources.
All pa_cvolume_snprint(), pa_volume_snprint(),
pa_sw_cvolume_snprint_dB() and pa_sw_volume_snprint_dB() calls have
been replaced with pa_cvolume_snprint_verbose() and
pa_volume_snprint_verbose() calls, making the log output more
informative and the code sometimes simpler.
This patch removes all occurrences of double and triple
newlines.
Command used for this:
find . -type d \( -name ffmpeg \) -prune -o \
-regex '\(.*\.[hc]\|.*\.cc\)' \
-a -not -name 'adrian-aec.*' -a -not \
-name reserve.c -a -not -name 'rtkit.*' \
-exec sed -i -e '/^$/{N;s/^\n$//}' {} \;
Two passes were needed to remove triple newlines.
The excluded files are mirrored files from external sources.
This patch replaces every occurrence of 'if(' with 'if ('.
The ffmpeg source tree was excluded since it will disappear anyways.
Command used for this:
find . -type d \( -name ffmpeg \) -prune -o \
-regex '\(.*\.[hc]\|.*\.cc\)' \
-exec sed -i -e 's/ if(/ if (/' {} \;
Since some devices can be chatty with regards to how often they return
from poll(), this adds a PA_UNLIKELY() to all the the rewind_requested
checks in our sink modules to make the general case (no rewind was
requested) the fast path.
When a rewind is requested on a sink input, the request parameters are
stored in the pa_sink_input struct. The parameters are reset during
rewind processing, and if the sink decides to ignore the rewind
request due to being suspended, stale parameters are left in
pa_sink_input. It's particularly problematic if the rewrite_bytes
parameter is left at -1, because that will prevent all future rewind
processing on that sink input. So, in order to avoid stale parameters,
every rewind request needs to be processed, even if the sink is
suspended.
Reported-by: Uoti Urpala
Some sink flags are really just a product of what callbacks
are set on the device. We still enforce a degree of sanity
that the flags match the callbacks set, but we also set the
flags automatically in our callback setter functions to
help ensure that a) people use them and b) flags & callbacks
are kept in sync.
This is not currently useful but future commits will make further
changes concerning automatic setting of flags and event delivery
that makes this structure necessary.
This piggy backs onto the previous changes for protocol 22 and
thus does not bump the version. This and the previous commits should be
seen as mostly atomic. Apologies for any bisecting issues this causes
(although I would expect these to be minimal)
Instead <pulsecore/poll.h> should be included. That file includes poll.h on
platform where it is appropriate. Also remove some unnecessary <ioctl.h>
includes.
This is needed to better support out of tree builds (including
distcheck) and to ensure the necessary folders are created in the
build tree on configure and also works around an intl-tools bug
(https://bugs.launchpad.net/intltool/+bug/605826)
The Makefile.am's used are minimal (and in some cases completely
blank). At present they do not include anything interesting
with the majority of the real work still done by the monolitic
src/Makefile.am
It may make sense to start splitting out src/Makefile.am into
smaller chunks but this commit makes the minimum changes to address
the issues that result from using make distcheck and other out of
tree builds.
Note: This 'breaks' the ability to type make in e.g. the src/modules
folder and have all of PA rebuilt accordingly (this is because the
static Makefiles previously present just did a "make -C ..") which
was purportedly for use in emacs. But I'm sure there will be a better
and more robust way to configure emacs to do your builds properly if
this behaviour is still desirable.
This ensures that we always clamp the volume to PA_VOLUME_MAX. While
this currently has no effect, it will be required for making sure we
don't exceed PA_VOLUME_MAX when its value changes in the future.
- We now implement a logic where the sink maintains two distinct
volumes: the 'reference' volume which is shown to the users, and the
'real' volume, which is configured to the hardware. The latter is
configured to the max of all streams. Volume changes on sinks are
propagated back to the streams proportional to the reference volume
change. Volume changes on sink inputs are forwarded to the sink by
'pushing' the volume if necessary.
This renames the old 'virtual_volume' to 'real_volume'. The
'reference_volume' is now the one exposed to users.
By this logic the sink volume visible to the user, will always be the
"upper" boundary for everything that is played. Saved/restored stream
volumes are measured relative to this boundary, the factor here is
always < 1.0.
- introduce accuracy for sink volumes, similar to the accuracy we
already have for source volumes.
- other cleanups.
- drop the 'virtual_' prefix from s->virtual_volume since we don't
distuingish between reference and real volumes for sources
- introduce an accuracy for source volumes: if the hardware can control
the volume "close enough" don't necessarily adjust the rest in
software unless it is beyond a certain threshold. This should save a
little bit of CPU at the expensive of a bit of accuracy in volume
handling.
- other minor cleanups
The reference volume is to be used as reference volume for stored stream
volumes. Previously if a new stream was created the relative volume was
taken relatively to the virtual device volume. Due to the flat volume
logic this could then be fed back to the virtual device volume.
Repeating the whole story over and over would result in a device volume
that would go lower, and lower and lower.
This patch introduces a 'reference' volume for each sink which stays
unmodified by stream volume changes even if flat volumes are used. It is
only modified if the sink volumes are modified directly by the user.
For further explanations see http://pulseaudio.org/wiki/InternalVolumes
