I had forgotten to test with shared libs and it turns out jack and alsa
were directly accessing symbols in the renderer (and in jack's case,
linking in a duplicate of the renderer).
Fixes#16.
The JACK Audio Connection Kit support is now just an output target
rather than a full duplicate of the renderer (in pull mode). This is
what I wanted to to back when I first added jack support, but I needed
to get the renderer working asynchronously without affecting any of the
other outputs.
Fixes#16.
Output plugins can use either a push model (synchronous) or a pull
model (asynchronous). The ALSA plugin now uses the pull model. This
paves the way for making jack output a simple output plugin rather than
the combined render/output plugin it currently is (for #16) as now
snd_dma works with both models.
This gets the alsa target working nicely for mmapped outout. I'm not
certain, but I think it will even deal with NPOT buffer sizes (I copied
the code from libasound's sample pcm.c, thus the uncertainty).
Non-mmapped output isn't supported yet, but the alsa target now works
nicely for pull rendering.
However, some work still needs to be done for recovery failure: either
disable the sound system, or restart the driver entirely (preferable).
This brings the alsa driver in line with the jack render (progress
towards #16), but breaks most of the other drivers (for now: one step at
a time). The idea is that once the pull model is working for at least
one other target, the jack renderer can become just another target like
it should have been in the first place (but I needed to get the pull
model working first, then forgot about it).
Correct state checking is not done yet, but testsound does produce what
seems to be fairly good sound when it starts up correctly (part of the
state checking (or lack thereof), I imagine).
and rename the variable since it's not the size of the frame (may be
from the very early days of ALSA development, and I suspect the
terminology changed a bit).
The calculation was including the bits per sample, which makes no sense
as the period size determines the number of samples in a submission
chunk (and thus latency). For now, set it to around 5.5ms (will probably
need a cvar).
There's still some cleanup to do, but everything seems to be working
nicely: `make -j` works, `make distcheck` passes. There is probably
plenty of bitrot in the package directories (RPM, debian), though.
The vc project files have been removed since those versions are way out
of date and quakeforge is pretty much dependent on gcc now anyway.
Most of the old Makefile.am files are now Makemodule.am. This should
allow for new Makefile.am files that allow local building (to be added
on an as-needed bases). The current remaining Makefile.am files are for
standalone sub-projects.a
The installable bins are currently built in the top-level build
directory. This may change if the clutter gets to be too much.
While this does make a noticeable difference in build times, the main
reason for the switch was to take care of the growing dependency issues:
now it's possible to build tools for code generation (eg, using qfcc and
ruamoko programs for code-gen).
I added Sys_RegisterShutdown years ago and never really did anything
with it: now any system that needs to be shutdown can ensure it gets
shutdown on program exit, and in the correct order (ie, reverse to init
order).
o All instances of LIBADD/LDADD have a corresponding DEPENDENCIES
specificatiion.
o libraries now use a lib_ldflags macro to keep things consistent
o duplication of source/lib names has been minimized (particularly in
the libraries; more work needs to be done for the executables)
o automake spec blocks have been organized (again, more work needs to be
done for the executables)
Thanks to "Sander van Dijk" <a.h.vandijk@gmail.com>, we now have much
better SDL sound support.
Here's the promised cleaned up version of the "double buffer" approach
patch for "snd_sdl.c". I've taken some more time to re-read and test
it this time, and it seems to behave well. All memory that is used by
both the main thread and the SDL audio thread is prefixed with "shm_",
and locking is used to ensure that only one thread accesses it at the
same time.
If the default sound device does not support mmap access, retry with
plughw. However, assume the user knows best and do not retry if snd_device
has been set to anything, including "default".
QF alsa support now works out of the box with pulseaudio.
Due to quake's original sound engine using a push model, the actual place
to which the sound data should be written is not necessarily where the
"hardware" dma cursor is, but rather where the last write finished off.
Thus, the correct output location is indicated by snd_paintedtime rather
than snd_shm->framepos.
Unfortuanately, I can't test this properly as I don't have any such
hardware, but as the code is mosly an edited copy of the interleaved code,
any errors should be easy to fix.
