mirror of https://github.com/bsnes-emu/bsnes.git
10 Commits
| Author | SHA1 | Message | Date |
|---|---|---|---|
Tim Allen
|
f87c6b7ecb |
Update to v103r16 release.
byuu says:
Changelog:
- emulator/audio: added the ability to change the output frequency at
run-time without emulator reset
- tomoko: display video synchronize option again¹
- tomoko: Settings→Configuration expanded to Settings→{Video,
Audio, Input, Hotkey, Advanced} Settings²
- tomoko: fix default population of audio settings tab
- ruby: Audio::frequency is a double now (to match both
Emulator::Audio and ASIO)³
- tomoko: changing the audio device will repopulate the frequency and
latency lists
- tomoko: changing the audio frequency can now be done in real-time
- ruby/audio/asio: added missing device() information, so devices can
be changed now
- ruby/audio/openal: ported to new API; added device selection support
- ruby/audio/wasapi: ported to new API, but did not test yet (it's
assuredly still broken)⁴
¹: I'm uneasy about this ... but, I guess if people want to disable
audio and just have smooth scrolling video ... so be it. With
Screwtape's documentation, hopefully that'll help people understand that
video synchronization always breaks audio synchronization. I may change
this to a child menu that lets you pick between {no synchronization,
video synchronization, audio synchronization} as a radio selection.
²: given how much more useful the video and audio tabs are now, I
felt that four extra menu items were worth saving a click and going
right to the tab you want. This also matches the behavior of the Tools
menu displaying all tool options and taking you directly to each tab.
This is kind of a hard change to get used to ... but I think it's for
the better.
³: kind of stupid because I've never seen a hardware sound card where
floor(frequency) != frequency, but whatever. Yay consistency.
⁴: I'm going to move it to be event-driven, and try to support 24-bit
sample formats if possible. Who knows which cards that'll fix and which
cards that'll break. I may end up making multiple WASAPI drivers so
people can find one that actually works for them. We'll see.
|
|
|
Tim Allen
|
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.
|
|
|
Tim Allen
|
04072b278b |
Update to v102r16 release.
byuu says:
Changelog:
- Emulator::Stream now allows adding low-pass and high-pass filters
dynamically
- also accepts a pass# count; each pass is a second-order biquad
butterworth IIR filter
- Emulator::Stream no longer automatically filters out >20KHz
frequencies for all streams
- FC: added 20Hz high-pass filter; 20KHz low-pass filter
- GB: removed simple 'magic constant' high-pass filter of unknown
cutoff frequency (missed this one in the last WIP)
- GB,SGB,GBC: added 20Hz high-pass filter; 20KHz low-pass filter
- MS,GG,MD/PSG: added 20Hz high-pass filter; 20KHz low-pass filter
- MD: added save state support (but it's completely broken for now;
sorry)
- MD/YM2612: fixed Voice#3 per-operator pitch support (fixes sound
effects in Streets of Rage, etc)
- PCE: added 20Hz high-pass filter; 20KHz low-pass filter
- WS,WSC: added 20Hz high-pass filter; 20KHz low-pass filter
So, the point of the low-pass filters is to remove frequencies above
human hearing. If we don't do this, then resampling will introduce
aliasing that results in sounds that are audible to the human ear. Which
basically an annoying buzzing sound. You'll definitely hear the
improvement from these in games like Mega Man 2 on the NES. Of course,
these already existed before, so this WIP won't sound better than
previous WIPs.
The high-pass filters are a little more complicated. Their main role is
to remove DC bias and help to center the audio stream. I don't
understand how they do this at all, but ... that's what everyone who
knows what they're talking about says, thus ... so be it.
I have set all of the high-pass filters to 20Hz, which is below the
limit of human hearing. Now this is where it gets really interesting ...
technically, some of these systems actually cut off a lot of range. For
instance, the GBA should technically use an 800Hz high-pass filter when
output is done through the system's speakers. But of course, if you plug
in headphones, you can hear the lower frequencies.
Now 800Hz ... you definitely can hear. At that level, nearly all of the
bass is stripped out and the audio is very tinny. Just like the real
system. But for now, I don't want to emulate the audio being crushed
that badly.
I'm sticking with 20Hz everywhere since it won't negatively affect audio
quality. In fact, you should not be able to hear any difference between
this WIP and the previous WIP. But theoretically, DC bias should mostly
be removed as a result of these new filters. It may be that we need to
raise the values on some cores in the future, but I don't want to do
that until we know for certain that we have to.
What I can say is that compared to even older WIPs than r15 ... the
removal of the simple one-pole low-pass and high-pass filters with the
newer three-pass, second-order filters should result in much better
attenuation (less distortion of audible frequencies.) Probably not
enough to be noticeable in a blind test, though.
|
|
|
Tim Allen
|
bf90bdfcc8 |
Update to v101r31 release.
byuu says:
Changelog:
- converted Emulator::Interface::Bind to Emulator::Platform
- temporarily disabled SGB hooks
- SMS: emulated Game Gear palette (latching word-write behavior not
implemented yet)
- SMS: emulated Master System 'Reset' button, Game Gear 'Start' button
- SMS: removed reset() functionality, driven by the mappable input now
instead
- SMS: split interface class in two: one for Master System, one for
Game Gear
- SMS: emulated Game Gear video cropping to 160x144
- PCE: started on HuC6280 CPU core—so far only registers, NOP
instruction has been implemented
Errata:
- Super Game Boy support is broken and thus disabled
- if you switch between Master System and Game Gear without
restarting, bad things happen:
- SMS→GG, no video output on the GG
- GG→SMS, no input on the SMS
I'm not sure what's causing the SMS\<-\>GG switch bug, having a hard
time debugging it. Help would be very much appreciated, if anyone's up
for it. Otherwise I'll keep trying to track it down on my end.
|
|
|
Tim Allen
|
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.
|
|
|
Tim Allen
|
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? :/ |
|
|
Tim Allen
|
0955295475 |
Update to v098r08 release.
byuu says: Changelog: - nall/vector rewritten from scratch - higan/audio uses nall/vector instead of raw pointers - higan/sfc/coprocessor/sdd1 updated with new research information - ruby/video/glx and ruby/video/glx2: fuck salt glXSwapIntervalEXT! The big change here is definitely nall/vector. The Windows, OS X and Qt ports won't compile until you change some first/last strings to left/right, but GTK will compile. I'd be really grateful if anyone could stress-test nall/vector. Pretty much everything I do relies on this class. If we introduce a bug, the worst case scenario is my entire SFC game dump database gets corrupted, or the byuu.org server gets compromised. So it's really critical that we test the hell out of this right now. The S-DD1 changes mean you need to update your installation of icarus again. Also, even though the Lunar FMV never really worked on the accuracy core anyway (it didn't initialize the PPU properly), it really won't work now that we emulate the hard-limit of 16MiB for S-DD1 games. |
|
|
Tim Allen
|
7cdae5195a |
Update to v098r07 release.
byuu says: Changelog: - GB: support modeSelect and RAM for MBC1M (Momotarou Collection) - audio: implemented native resampling support into Emulator::Stream - audio: removed nall::DSP completely Unfortunately, the new resampler didn't turn out quite as fast as I had hoped. The final hermite resampling added some overhead; and I had to bump up the kernel count to 500 from 400 to get the buzzing to go away on my main PC. I think that's due to it running at 48000hz output instead of 44100hz output, maybe? Compared to Ryphecha's: (NES) Mega Man 2: 167fps -> 166fps (GB) Mega Man II: 224fps -> 200fps (WSC) Riviera: 143fps -> 151fps Odd that the WS/WSC ends up faster while the DMG/CGB ends up slower. But this knocks 922 lines down to 146 lines. The only files left in all of higan not written (or rewritten) by me are ruby/xaudio2.h and libco/ppc.c |
|
|
Tim Allen
|
e2ee6689a0 |
Update to v098r06 release.
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.
|
|
|
Tim Allen
|
19e1d89f00 |
Update to v098r01 release.
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.
|