bsnes/higan/audio/audio.hpp

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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.
2016-04-09 03:40:12 +00:00
#pragma once
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.
2017-06-26 01:41:58 +00:00
#include <nall/dsp/iir/one-pole.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
#include <nall/dsp/iir/biquad.hpp>
#include <nall/dsp/resampler/cubic.hpp>
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.
2016-04-22 13:35:51 +00:00
namespace Emulator {
struct Interface;
struct Audio;
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.
2017-06-26 01:41:58 +00:00
struct Filter;
struct Stream;
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.
2016-04-22 13:35:51 +00:00
struct Audio {
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.
2016-06-01 11:23:22 +00:00
auto reset(maybe<uint> channels = nothing, maybe<double> frequency = nothing) -> void;
auto setInterface(Interface* interface) -> void;
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.
2016-04-22 13:35:51 +00:00
auto setVolume(double volume) -> void;
auto setBalance(double balance) -> void;
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 setReverb(bool enabled) -> void;
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.
2016-04-22 13:35:51 +00:00
auto createStream(uint channels, double frequency) -> shared_pointer<Stream>;
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.
2016-04-22 13:35:51 +00:00
private:
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.
2016-06-01 11:23:22 +00:00
auto process() -> void;
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.
2016-04-22 13:35:51 +00:00
Interface* interface = nullptr;
vector<shared_pointer<Stream>> streams;
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.
2016-06-01 11:23:22 +00:00
uint channels = 0;
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.
2016-04-22 13:35:51 +00:00
double frequency = 0.0;
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.
2016-06-01 11:23:22 +00:00
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.
2016-04-22 13:35:51 +00:00
double volume = 1.0;
double balance = 0.0;
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
bool reverbEnable = false;
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.
2016-06-01 11:23:22 +00:00
vector<vector<queue<double>>> reverb;
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.
2016-04-22 13:35:51 +00:00
friend class Stream;
};
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.
2017-06-26 01:41:58 +00:00
struct Filter {
enum class Order : uint { First, Second };
enum class Type : uint { LowPass, HighPass };
Order order;
DSP::IIR::OnePole onePole; //first-order
DSP::IIR::Biquad biquad; //second-order
};
struct Stream {
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.
2016-06-01 11:23:22 +00:00
auto reset(uint channels, double inputFrequency, double outputFrequency) -> void;
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.
2017-06-26 01:41:58 +00:00
auto addFilter(Filter::Order order, Filter::Type type, double cutoffFrequency, uint passes = 1) -> void;
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.
2017-03-08 20:20:40 +00:00
auto pending() const -> bool;
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.
2016-06-01 11:23:22 +00:00
auto read(double* samples) -> uint;
auto write(const double* samples) -> void;
template<typename... P> auto sample(P&&... p) -> void {
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.
2016-06-01 11:23:22 +00:00
double samples[sizeof...(P)] = {forward<P>(p)...};
write(samples);
}
private:
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.
2016-06-01 11:23:22 +00:00
struct Channel {
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.
2017-06-26 01:41:58 +00:00
vector<Filter> filters;
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.
2016-06-01 11:23:22 +00:00
DSP::Resampler::Cubic resampler;
};
vector<Channel> channels;
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.
2017-03-08 20:20:40 +00:00
double inputFrequency;
double outputFrequency;
friend class Audio;
};
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.
2016-04-22 13:35:51 +00:00
extern Audio audio;
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.
2016-04-09 03:40:12 +00:00
}