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? :/
2016-05-31 22:29:36 +00:00
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#pragma once
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//simple circular ring buffer
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namespace nall {
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template<typename T>
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struct queue {
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queue() = default;
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queue(const queue& source) {
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operator=(source);
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}
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queue(queue&& source) {
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operator=(move(source));
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}
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auto operator=(const queue& source) -> queue& {
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if(this == &source) return *this;
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reset();
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_size = source._size;
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_data = new T[_size];
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for(auto n : range(_size)) _data[n] = source._data[n];
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_read = source._read;
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_write = source._write;
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return *this;
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}
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auto operator=(queue&& source) -> queue& {
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if(this == &source) return *this;
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_data = source._data;
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_size = source._size;
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_read = source._read;
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_write = source._write;
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source._data = nullptr;
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source.reset();
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Update to v103r23 release.
byuu says:
Changelog:
- gb: added accelerometer X-axis, Y-Axis inputs¹
- gb: added rumble input¹
- gb/mbc5: added rumble support²
- gb/mbc6: added skeleton driver, but it doesn't boot Net de Get
- gb/mbc7: added mostly complete driver (only missing EEPROM), but it
doesn't boot Kirby Tilt 'n' Tumble
- gb/tama: added leap year assignment
- tomoko: fixed macOS compilation [MerryMage]
- hiro/cocoa: fix table cell redrawing on updates and automatic column
resizing [ncbncb]
- hiro/cocoa: fix some weird issue with clicking table view checkboxes
on Retina displays [ncbncb]
- icarus: enhance Game Boy heuristics³
- nall: fix three missing return statements [Jonas Quinn]
- ruby: hopefully fixed all compilation errors reported by Screwtape
et al⁴
¹: because there's no concept of a controller for cartridge inputs,
I'm attaching to the base platform for now. An idea I had was to make
separate ports for each cartridge type ... but this would duplicate the
rumble input between MBC5 and MBC7. And would also be less discoverable.
But it would be more clean in that users wouldn't think the Game Boy
hardware had this functionality. I'll think about it.
²: it probably won't work yet. Rumble isn't documented anywhere, but
I dug through an emulator named GEST and discovered that it seems to use
bit 3 of the RAM bank select to be rumble. I don't know if it sets the
bit for rumbling, then clears when finished, or if it sets it and then
after a few milliseconds it stops rumbling. I couldn't test on my
FreeBSD box because SDL 1.2 doesn't support rumble, udev doesn't exist
on FreeBSD, and nobody has ever posted any working code for how to use
evdev (or whatever it's called) on FreeBSD.
³: I'm still thinking about specifying the MBC7 RAM as EEPROM, since
it's not really static RAM.
⁴: if possible, please test all drivers if you can. I want to ensure
they're all working. Especially let me know if the following work:
macOS: input.carbon Linux: audio.pulseaudiosimple, audio.ao (libao)
If I can confirm these are working, I'm going to then remove them from
being included with stock higan builds.
I'm also considering dropping SDL video on Linux/BSD. XShm is much
faster and supports blurring. I may also drop SDL input on Linux, since
udev works better. That will free a dependency on SDL 1.2 for building
higan. FreeBSD is still going to need it for joypad support, however.
2017-07-30 13:00:31 +00:00
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return *this;
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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? :/
2016-05-31 22:29:36 +00:00
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}
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~queue() {
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reset();
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}
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explicit operator bool() const {
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return _size;
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}
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auto size() const -> uint {
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return _size;
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}
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auto data() -> T* {
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return _data;
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}
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auto data() const -> const T* {
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return _data;
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}
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auto reset() {
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delete[] _data;
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_data = nullptr;
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_size = 0;
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_read = 0;
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_write = 0;
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}
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auto resize(uint size, const T& value = {}) -> void {
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reset();
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_size = size;
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_data = new T[_size];
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for(auto n : range(_size)) _data[n] = value;
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}
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auto pending() const -> bool {
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return _read != _write;
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}
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auto read() -> T {
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T result = _data[_read];
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if(++_read >= _size) _read = 0;
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return result;
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}
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auto last() const -> T {
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return _data[_write];
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}
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auto write(const T& value) -> void {
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_data[_write] = value;
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if(++_write >= _size) _write = 0;
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}
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private:
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T* _data = nullptr;
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uint _size = 0;
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|
uint _read = 0;
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|
uint _write = 0;
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|
};
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}
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