bsnes/higan/processor/gsu/gsu.hpp

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#pragma once
namespace Processor {
struct GSU {
#include "registers.hpp"
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
virtual auto step(uint clocks) -> void = 0;
virtual auto stop() -> void = 0;
virtual auto color(uint8 source) -> uint8 = 0;
virtual auto plot(uint8 x, uint8 y) -> void = 0;
virtual auto rpix(uint8 x, uint8 y) -> uint8 = 0;
virtual auto pipe() -> uint8 = 0;
virtual auto rombuffer_sync() -> void = 0;
virtual auto rombuffer_read() -> uint8 = 0;
virtual auto rambuffer_sync() -> void = 0;
virtual auto rambuffer_read(uint16 addr) -> uint8 = 0;
virtual auto rambuffer_write(uint16 addr, uint8 data) -> void = 0;
virtual auto cache_flush() -> void = 0;
virtual auto bus_read(uint24 addr, uint8 data = 0x00) -> uint8 = 0;
virtual auto bus_write(uint24 addr, uint8 data) -> void = 0;
//gsu.cpp
auto power() -> void;
auto reset() -> void;
//instructions.cpp
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
auto op_add_adc(uint n);
auto op_alt1();
auto op_alt2();
auto op_alt3();
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
auto op_and_bic(uint n);
auto op_asr_div2();
auto op_branch(bool c);
auto op_cache();
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
auto op_color_cmode();
auto op_dec(uint n);
auto op_fmult_lmult();
auto op_from_moves(uint n);
auto op_getb();
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
auto op_getc_ramb_romb();
auto op_hib();
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
auto op_ibt_lms_sms(uint n);
auto op_inc(uint n);
auto op_iwt_lm_sm(uint n);
auto op_jmp_ljmp(uint n);
auto op_link(uint n);
auto op_load(uint n);
auto op_lob();
auto op_loop();
auto op_lsr();
auto op_merge();
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
auto op_mult_umult(uint n);
auto op_nop();
auto op_not();
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
auto op_or_xor(uint n);
auto op_plot_rpix();
auto op_rol();
auto op_ror();
auto op_sbk();
auto op_sex();
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
auto op_store(uint n);
auto op_stop();
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
auto op_sub_sbc_cmp(uint n);
auto op_swap();
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
auto op_to_move(uint n);
auto op_with(uint n);
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
//switch.cpp
auto instruction(uint8 opcode) -> void;
//serialization.cpp
auto serialize(serializer&) -> void;
//disassembler.cpp
auto disassemble_opcode(char* output) -> void;
auto disassemble_alt0(char* output) -> void;
auto disassemble_alt1(char* output) -> void;
auto disassemble_alt2(char* output) -> void;
auto disassemble_alt3(char* output) -> void;
};
}