bsnes/nall/queue.hpp

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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
#pragma once
//simple circular ring buffer
namespace nall {
template<typename T>
struct queue {
queue() = default;
queue(const queue& source) {
operator=(source);
}
queue(queue&& source) {
operator=(move(source));
}
auto operator=(const queue& source) -> queue& {
if(this == &source) return *this;
reset();
_size = source._size;
_data = new T[_size];
for(auto n : range(_size)) _data[n] = source._data[n];
_read = source._read;
_write = source._write;
return *this;
}
auto operator=(queue&& source) -> queue& {
if(this == &source) return *this;
_data = source._data;
_size = source._size;
_read = source._read;
_write = source._write;
source._data = nullptr;
source.reset();
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
return *this;
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
}
~queue() {
reset();
}
explicit operator bool() const {
return _size;
}
auto size() const -> uint {
return _size;
}
auto data() -> T* {
return _data;
}
auto data() const -> const T* {
return _data;
}
auto reset() {
delete[] _data;
_data = nullptr;
_size = 0;
_read = 0;
_write = 0;
}
auto resize(uint size, const T& value = {}) -> void {
reset();
_size = size;
_data = new T[_size];
for(auto n : range(_size)) _data[n] = value;
}
auto pending() const -> bool {
return _read != _write;
}
auto read() -> T {
T result = _data[_read];
if(++_read >= _size) _read = 0;
return result;
}
auto last() const -> T {
return _data[_write];
}
auto write(const T& value) -> void {
_data[_write] = value;
if(++_write >= _size) _write = 0;
}
private:
T* _data = nullptr;
uint _size = 0;
uint _read = 0;
uint _write = 0;
};
}