mirror of https://github.com/bsnes-emu/bsnes.git
5 Commits
| Author | SHA1 | Message | Date |
|---|---|---|---|
Tim Allen
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ecc7e899e0 |
Update to v103r01 release.
byuu says:
Changelog:
- nall/dsp: improve one pole coefficient calculations [Fatbag]
- higan/audio: reworked filters to support selection of either one
pole (first-order) or biquad (second-order) filters
- note: the design is not stable yet; so forks should not put too
much effort into synchronizing with this change yet
- fc: added first-order filters as per NESdev wiki (90hz lowpass +
440hz lowpass + 14khz highpass)
- fc: created separate NTSC-J and NTSC-U regions
- NESdev wiki says the Japanese Famicom uses a separate audio
filtering strategy, but details are fuzzy
- there's also cartridge audio output being disabled on NES units;
and differences with controllers
- this stuff will be supported in the future, just adding the
support for it now
- gba: corrected serious bugs in PSG wave channel emulation [Cydrak]
- note that if there are still bugs here, it's my fault
- md/psg,ym2612: added first-order low-pass 2840hz filter to match
VA3-VA6 Mega Drives
- md/psg: lowered volume relative to the YM2612
- using 0x1400; multiple people agreed it was the closest to the
hardware recordings against a VA6
- ms,md/psg: don't serialize the volume levels array
- md/vdp: Hblank bit acts the same during Vblank as outside of it (it
isn't always set during Vblank)
- md/vdp: return isPAL in bit 0 of control port reads
- tomoko: change command-line option separator from : to |
- [Editor's note: This change was present in the public v103,
but it's in this changelog because it was made after the v103 WIP]
- higan/all: change the 20hz high-pass filters from second-order
three-pass to first-order one-pass
- these filters are meant to remove DC bias, but I honestly can't
hear a difference with or without them
- so there's really no sense wasting CPU power with an extremely
powerful filter here
Things I did not do:
- change icarus install rule
- work on 8-bit Mega Drive SRAM
- work on Famicom or Mega Drive region detection heuristics in icarus
My long-term dream plan is to devise a special user-configurable
filtering system where you can set relative volumes and create your own
list of filters (any number of them in any order at any frequency), that
way people can make the systems sound however they want.
Right now, the sanest place to put this information is inside the
$system.sys/manifest.bml files. But that's not very user friendly, and
upgrading to new versions will lose these changes if you don't copy them
over manually. Of course, cluttering the GUI with a fancy filter editor
is probably supreme overkill for 99% of users, so maybe that's fine.
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Tim Allen
|
7e7003fd29 |
Update to v102r15 release.
byuu says:
Changelog:
- nall: added DSP::IIR::OnePole (which is a first-order IIR filter)
- FC/APU: removed strong highpass, weak hipass filters (and the
dummied out lowpass filter)
- MS,GG,MD/PSG: removed lowpass filter
- MS,GG,MD/PSG: audio was not being centered properly; removed
centering for now
- MD/YM2612: fixed clipping of accumulator from 18 signed bits to 14
signed bits (-0x2000 to +0x1fff) [Cydrak]
- MD/YM2612: removed lowpass filter
- PCE/PSG: audio was not being centered properly; removed centering
for now
First thing is that I've removed all of the ad-hoc audio filtering.
Emulator::Stream intrinsically provides a three-pass, second-order
biquad IIR butterworth lowpass filter that clips frequencies above 20KHz
with very good attenuation (as good as IIR gets, anyway.)
It doesn't really make sense to have the various cores running
additional lowpass filters. If we want to filter frequencies below
20KHz, then I can adapt Emulator::Audio::createStream() to take a cutoff
frequency value, and we can do it all at once, with much better quality.
Right now, I don't know what frequencies are best to cut off the various
other audio cores, so they're just gone for now.
As for the highpass filters for the Famicom core, well ... you don't get
aliasing from resampling low frequencies. And generally speaking, too
low a frequency will be inaudible anyway. All these were doing was
killing possible bass (if they were too strong.) We can add them again,
but only if someone can convert Ryphecha's ad-hoc magic integers into a
frequency cutoff. In which case, I'll use my biquad IIR filter to do it
even better. On this note, it may prove useful to do this for the MD PSG
as well, to try and head off unnecessary clamping when mixing with the
YM2612.
Finally, there was the audio centering issue that affected the
MS,GG,MD,PCE,SG cores. It was flooring the "silent" audio level, which
was resulting in extremely heavy distortion if you tried listening to
higan and, say, audacious at the same time. Without the botched
centering, this distortion is completely gone now.
However, without any centering, we've halved the potential volume range.
This means the audio slider in higan's audio settings panel will start
clamping twice as quickly. So ultimately, we need to figure out how to
fix the centering. This isn't as simple as just subtracting less. We
will probably have to center every individual audio channel before
summing them to do this properly.
Results:
On the Mega Drive, Altered Beast sounds quite a bit better, a lot less
distortion now. But it's still not perfect, especially sound effects.
Further, Bare Knuckle / Streets of Rage still has really bad sound
effects. It looks like I broke something in Cydrak's code when trying to
adapt it to my style =(
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Tim Allen
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20ac95ee49 |
Update to v098r15 release.
byuu says: Changelog: - removed template usage from processor/spc700; cleaned up many function names and the switch table - object size: 176.8kb => 127.3kb - source code size: 43.5kb => 37.0kb - fixed processor/r65816 BRK/COP vector regression [hex_usr] - corrected HuC3 unmapped RAM read value; fixes Robopon [endrift] - cosmetic: simplified the butterworth constant calculation [Wolfram|Alpha] The SPC700 core changes took forever, about three hours of work. Only the LR35902 and R6502 still need their template functions removed. The point of this is that it doesn't cause any speed penalty to do so, and it results in smaller binary sizes and faster compilation times. |
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Tim Allen
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fdc41611cf |
Update to v098r14 release.
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.
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Tim Allen
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839813d0f1 |
Update to v098r13 release.
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? :/ |