byuu says:
Changelog:
- improved attenuation of biquad filter by computing butterworth Q
coefficients correctly (instead of using the same constant)
- adding 1e-25 to each input sample into the biquad filters to try and
prevent denormalization
- updated normalization from [0.0 to 1.0] to [-1.0 to +1.0]; volume/reverb
happen in floating-point mode now
- good amount of work to make the base Emulator::Audio support any number
of output channels
- so that we don't have to do separate work on left/right channels;
and can instead share the code for each channel
- Emulator::Interface::audioSample(int16 left, int16 right); changed to:
- Emulator::Interface::audioSample(double* samples, uint channels);
- samples are normalized [-1.0 to +1.0]
- for now at least, channels will be the value given to
Emulator::Audio::reset()
- fixed GUI crash on startup when audio driver is set to None
I'm probably going to be updating ruby to accept normalized doubles as
well; but I'm not sure if I will try and support anything other 2-channel
audio output. It'll depend on how easy it is to do so; perhaps it'll be
a per-driver setting.
The denormalization thing is fierce. If that happens, it drops the
emulator framerate from 220fps to about 20fps for Game Boy emulation. And
that happens basically whenever audio output is silent. I'm probably
also going to make a nall/denormal.hpp file at some point with
platform-specific functionality to set the CPU state to "denormals as
zero" where applicable. I'll still add the 1e-25 offset (inaudible)
as another fallback.
byuu says:
Changelog:
- nall/dsp returns with new iir/biquad.hpp and resampler/cubic.hpp files
- nall/queue.hpp added (simple ring buffer ... nall/vector wouldn't
cause too many moves with FIFO)
- audio streams now only buffer 20ms; so even if multiple audio streams
desync, latency can never exceed 20ms
- replaced blackman windwed sinc FIR hermite audio filter with transposed
direct form II biquadratic sixth-order IIR butterworth filter (better
attenuation of frequencies above 20KHz, faster, no need for decimation,
less code)
- put in experimental eight-tap echo filter (a lot better than what I
had before, but still rather weak)
- substantial cleanups to the SuperFX GSU processor core (slightly
faster, 479KB->100KB object file, 42.7KB->33.4KB source code size,
way less code duplication)
We'll definitely want to test the whole SuperFX library (not many games)
just to make sure there's no regressions caused by this one.
Not sure what I want to do with audio processing effects yet. I've always
really wanted lots of fun controls to customize audio, and now finally
with this new biquad filter, I can finally start implementing real
effects. For instance, an equalizer wouldn't be too complicated anymore.
The new reverb effect is still a poor man's version. I need to find human
readable source for implementing a comb-filter properly. I'm pretty sure
I can already treat nall::queue as an all-pass filter since all that
does is phase shift (fancy audio term for "delay audio"). What's really
going to be hard is figuring out how to expose user-friendly settings for
controlling it. It looks like you need a bunch of coprime coefficients,
and I don't think casual users are going to be able to hand-enter coprime
values to get the echo effect they want. I uh ... don't even know how
to calculate coprime values dynamically right now >_> But we're going
to have to, as they are correlated to the output sampling rate.
We'll definitely want to make some audio profiles so that users can
quickly select pre-configured themes that sound nice, but expose the
underlying coefficients so that they can tweak stuff to their liking. This
isn't just about higan, this is about me trying to learn digital signal
processing, so please don't be too upset about feature creep or anything
on this.
Anyway ... I'm having some difficulties with my audio right now. When
the reverb effect is enabled, there's a bunch of static on system
reset for just a moment. But this should not be possible. nall::queue
is initializing all previous reverb sample elements to 0.0. I don't
understand where static is coming in from. Further, we have the same
issue with both the windowed sinc and the biquad filters ... a bit of
a popping sound when starting a game. Any help tracking this down would
be appreciated.
There's also one really annoying issue ... I can't seem to do reverb
or volume adjustments with normalized samples. If I say "volume *= 0.5"
in higan/audio/audio.cpp line 68, it doesn't just halve the volume, it
adds a whole bunch of distortion. This makes absolutely zero sense to
me. The sample values are between 0.0 (mute) and 1.0 (full volume) here,
so multiplying a double by 0.5 shouldn't cause distortion. So right now,
I'm doing these adjustments with less precision after denormalizing back
to int16. Anyone ever see something like that? :/
byuu says:
Changelog:
- emulation cores now refresh video from host thread instead of
cothreads (fix AMD crash)
- SFC: fixed another bug with leap year months in SharpRTC emulation
- SFC: cleaned up camelCase on function names for
armdsp,epsonrtc,hitachidsp,mcc,nss,sharprtc classes
- GB: added MBC1M emulation (requires manually setting mapper=MBC1M in
manifest.bml for now, sorry)
- audio: implemented Emulator::Audio mixer and effects processor
- audio: implemented Emulator::Stream interface
- it is now possible to have more than two audio streams: eg SNES
+ SGB + MSU1 + Voicer-Kun (eventually)
- audio: added reverb delay + reverb level settings; exposed balance
configuration in UI
- video: reworked palette generation to re-enable saturation, gamma,
luminance adjustments
- higan/emulator.cpp is gone since there was nothing left in it
I know you guys are going to say the color adjust/balance/reverb stuff
is pointless. And indeed it mostly is. But I like the idea of allowing
some fun special effects and configurability that isn't system-wide.
Note: there seems to be some kind of added audio lag in the SGB
emulation now, and I don't really understand why. The code should be
effectively identical to what I had before. The only main thing is that
I'm sampling things to 48000hz instead of 32040hz before mixing. There's
no point where I'm intentionally introducing added latency though. I'm
kind of stumped, so if anyone wouldn't mind taking a look at it, it'd be
much appreciated :/
I don't have an MSU1 test ROM, but the latency issue may affect MSU1 as
well, and that would be very bad.
byuu says:
Changelog:
- WS/WSC: re-added support for screen rotation (code is inside WS core)
- ruby: changed sample(uint16_t left, uint16_t right) to sample(int16_t
left, int16_t right);
- requires casting to uint prior to shifting in each driver, but
I felt it was misleading to use uint16_t just to avoid that
- ruby: WASAPI is now built in by default; has wareya's improvements,
and now supports latency adjust
- tomoko: audio settings panel has new "Exclusive Mode" checkbox for
WASAPI driver only
- note: although the setting *does* take effect in real-time, I'd
suggest restarting the emulator after changing it
- tomoko: audio latency can now be set to 0ms (which in reality means
"the minimum supported by the driver")
- all: increased cothread size from 512KiB to 2MiB to see if it fixes
bullshit AMD driver crashes
- this appears to cause a slight speed penalty due to cache locality
going down between threads, though
byuu says:
Changelog:
- SFC: balanced profile removed
- SFC: performance profile removed
- SFC: code for handling non-threaded CPU, SMP, DSP, PPU removed
- SFC: Coprocessor, Controller (and expansion port) shared Thread code
merged to SFC::Cothread
- Cothread here just means "Thread with CPU affinity" (couldn't think
of a better name, sorry)
- SFC: CPU now has vector<Thread*> coprocessors, peripherals;
- this is the beginning of work to allow expansion port devices to be
dynamically changed at run-time
- ruby: all audio drivers default to 48000hz instead of 22050hz now if
no frequency is assigned
- note: the WASAPI driver can default to whatever the native frequency
is; doesn't have to be 48000hz
- tomoko: removed the ability to change the frequency from the UI (but
it will display the frequency used)
- tomoko: removed the timing settings panel
- the goal is to work toward smooth video via adaptive sync
- the model is broken by not being in control of the audio frequency
anyway
- it's further broken by PAL running at 50hz and WSC running at 75hz
- it was always broken anyway by SNES interlace timing varying from
progressive timing
- higan: audio/ stub created (for now, it's just nall/dsp/ moved here
and included as a header)
- higan: video/ stub created
- higan/GNUmakefile: now includes build rules for essential components
(libco, emulator, audio, video)
The audio changes are in preparation to merge wareya's awesome WASAPI
work without the need for the nall/dsp resampler.
byuu says:
New update. Most of the work today went into eliminating hiro::Image
from all objects in all ports, replacing with nall::image. That took an
eternity.
Changelog:
- fixed crashing bug when loading games [thanks endrift!!]
- toggling "show status bar" option adjusts window geometry (not
supposed to recenter the window, though)
- button sizes improved; icon-only button icons no longer being cut off
byuu says:
Changelog:
- restructured the project and removed a whole bunch of old/dead
directives from higan/GNUmakefile
- huge amounts of work on hiro/cocoa (compiles but ~70% of the
functionality is commented out)
- fixed a masking error in my ARM CPU disassembler [Lioncash]
- SFC: decided to change board cic=(411,413) back to board
region=(ntsc,pal) ... the former was too obtuse
If you rename Boolean (it's a problem with an include from ruby, not
from hiro) and disable all the ruby drivers, you can compile an
OS X binary, but obviously it's not going to do anything.
It's a boring WIP, I just wanted to push out the project structure
change now at the start of this WIP cycle.
Tim Allen