For whatever reason, building under MXE (for windows) causes FLAC to try
to use dll import references, but setting FLAC__NO_DLL before including
FLAC/export.h fixes the issue.
Most were pretty easy and fairly logical, but gib's regex was a bit of a
pain until I figured out the real problem was the conditional
assignments.
However, libs/gamecode/test/test-conv4 fails when optimizing due to gcc
using vcvttps2dq (which is nice, actually) for vector forms, but not the
single equivalent other times. I haven't decided what to do with the
test (I might abandon it as it does seem to be UD).
This gets ambient sounds (in particular, water and sky) working again
for quakeworld after the recent sound changes, and again for nq after I
don't know how long.
I never liked "cache" as a name because it said where the sound was
stored rather than how it was loaded/played, but "stream" is ok, since
that's pretty much spot on. I'm not sure "block" is the best, but it at
least makes sense because the sounds are loaded as a single block (as
opposed to being streamed). An now, neither use the cache system.
Nuclear powered audio ;)
More seriously, use _Atomic on a few fields that very obviously need it.
That is, channel's buffer pointer (used to signal to the mixer that the
channel is ready for use) and "flow control" flags (stop, done and
pause), and head and tail in the buffer itself. Since QF has been
working without _Atomic (admittedly, thanks to luck and x86's strong
memory model), this should do until proven otherwise. I imagine getting
stream reading out of the RT thread will highlight any issues.
Turned out the channels simply weren't being freed by SND_ScanChannels
when they should have been (probably a good thing, too, as it wasn't
being told to wait for the mixer).
Care needs to be taken when freeing channels as doing so while an
asynchronous mixer is using them is unlikely to end well. However,
whether the mixer is asynchronous depends on the output driver. This
lets the driver inform the rest of the system that the output and mixer
are running asynchronously.
SYS_dev is a holdover from when we had only the one flag and is not
meant to be used for tests (I seem to remember mentioning an audit was
necessary, but obviously forgotten). One step at a time, I guess :)
This improves the locality of reference when mixing and removes the
proxy sfx for streamed sounds.
The buffer for streamed sounds is allocated when the stream is opened
(since streamed sounds can't share buffers), and freed when the stream
is closed.
For block sounds, the buffer is reference counted (with the sfx holding
one reference, so currently block buffers never get freed), with their
reference count getting incremented on open and decremented on close.
That the reference counts get to 1 has been confirmed, so all that
should be needed is proper destruction of the sfx instances.
Still need to sort out just why channels leak across level changes.
They're currently treated as non-fatal, those sounds just won't ever
play. This allows ad_tears to at least load with only 32MB of locked
memory (it needs somewhere between 64 and 96).
Sounds no longer use the cache, which is good for multi-threaded, but a
pain for memory management: the buffers are shared between channels that
play back the sounds, but when the sounds were cached, they were
automagically (thus problematically) freed when the space was needed.
That no longer happens, so they leak. I think the solution is to use
reference counting and retain/release in sfx->open() and sfx->close().
Streams are the easy one as they were never in the cache. As a side
effect, sfxstream_t is much smaller as it no longer has the buffer
embedded in the struct.
SND_AllocChannel is a little too aggressive in freeing channels that
have finished as the channel may be externally owned (eg, by cd_file).
Get bgm looping working again.
More shrinkage. It turned out the mixer uses the phase fields, so they
couldn't be removed, but even at 192kHz, +/- 127 samples produces
sufficient phase separation for a 21cm head (which is, actually, pretty
big: mine is about 15cm across), but that change can come later.
The ambient sound loading has been removed from snd_channels because 1)
it doesn't work for nq, 2) it should never have been there in the first
place (it belongs in the client, but that needs some more API).
This is part of a process to shrink channel_t so it doesn't waste locked
memory when it gets moved there. Eventually, only the fields the mixer
needs will be in channel_t itself: those needed for spacialization will
be moved into a separate array.
In the process, I found that channels leak across level changes, but
this appears to be due to the cached sounds being removed during loading
and the mixer never marking them as done (it sees the null sfx pointer
and assumes the channel was never in use). Having the mixer mark the
channel as done seems to fix the leak, but cause a free channel list
overflow. Rather than fight with that, I'll leave the leak for now and
fix it at its root cause: the management of the sound samples
themselves.
Sys_DoubleTime starts at 4Gs in order to keep its precision fixed for a
nice long time (about 120 years, iirc).
This fixes an instant watchdog trigger when first starting up in
testsound. I'm not sure why it didn't happen with nq, but I guess that
doesn't really matter
The scaling up of the volumes when setting a channel's volume bothered
me. The biggest issue being it hasn't been necessary for over a decade
since the conversion to a float-mixer. Now the volume and attenuation
scaling from protocol bytes is entirely in the client's hands.
sfx_t is now private, and cd_file no longer accesses channel_t's
internals. This is necessary for hiding the code needed to make mixing
and channel management *properly* lock-free (I've been getting away with
murder thanks to x86's strong memory model and just plain luck with
gcc).
Sounds in Arcane Dimensions (at least those used by ad_tears) specify
start and end cue points. The code was using only the final point in the
list and thus breaking looped sounds. Now, the first cue point is used
as the loop start, and the second (if present), the sample length. Both
are bounds-checked against the wav's sample count. Fixes sound locking
up during the first seconds in ad_tears.
The misinterpretations were due to either the cvar not being accessed
directly by the engine, but via only the callback, or the cvars were
accesssed only by progs (in which case, they should be float). The
remainder are a potential enum (hud gravity) and a "too hard basket"
(rcon password: need to figure out how I want to handle secret strings).
This is an extremely extensive patch as it hits every cvar, and every
usage of the cvars. Cvars no longer store the value they control,
instead, they use a cexpr value object to reference the value and
specify the value's type (currently, a null type is used for strings).
Non-string cvars are passed through cexpr, allowing expressions in the
cvars' settings. Also, cvars have returned to an enhanced version of the
original (id quake) registration scheme.
As a minor benefit, relevant code having direct access to the
cvar-controlled variables is probably a slight optimization as it
removed a pointer dereference, and the variables can be located for data
locality.
The static cvar descriptors are made private as an additional safety
layer, though there's nothing stopping external modification via
Cvar_FindVar (which is needed for adding listeners).
While not used yet (partly due to working out the design), cvars can
have a validation function.
Registering a cvar allows a primary listener (and its data) to be
specified: it will always be called first when the cvar is modified. The
combination of proper listeners and direct access to the controlled
variable greatly simplifies the more complex cvar interactions as much
less null checking is required, and there's no need for one cvar's
callback to call another's.
nq-x11 is known to work at least well enough for the demos. More testing
will come.
This is the bulk of the work for recording the resource pointer with
with builtin data. I don't know how much of a difference it makes for
most things, but it's probably pretty big for qwaq-curses due to the
very high number of calls to the curses builtins.
Closes#26
This is part of the work for #26 (Record resource pointer with builtin
function data). Currently, the data pointer gets as far as the
per-instance VM function table (I don't feel like tackling the job of
converting all the builtin functions tonight). All the builtin modules
that register a resources data block pass that block on to
PR_RegisterBuiltins.
This will make it possible for the engine to set up their parameter
pointers when running Ruamoko progs. At this stage, it doesn't matter
*too* much, except for varargs functions, because no builtin yet takes
anything larger than a float quaternion, but it will be critical when
double or long vec3 and vec4 values are passed.
This is needed for cleaning up excess memsets when loading files because
Hunk_RawAllocName has nonnull on its hunk pointer (as the rest of the
hunk functions really should, but not just yet).
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.
on_update is for pull-model outpput targets to do periodic synchronous
checks (eg, checking that the connection to the actual output device is
still alive and reviving it if necessary)
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).
