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;
|
2016-04-23 07:55:59 +00:00
|
|
|
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;
|
2016-04-23 07:55:59 +00:00
|
|
|
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 {
|
2016-06-01 11:23:22 +00:00
|
|
|
auto reset(maybe<uint> channels = nothing, maybe<double> frequency = nothing) -> void;
|
2017-01-13 01:15:45 +00:00
|
|
|
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
|
|
|
|
2016-04-23 07:55:59 +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:
|
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;
|
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;
|
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;
|
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
|
|
|
|
};
|
|
|
|
|
2016-04-23 07:55:59 +00:00
|
|
|
struct Stream {
|
2016-06-01 11:23:22 +00:00
|
|
|
auto reset(uint channels, double inputFrequency, double outputFrequency) -> void;
|
2016-04-23 07:55:59 +00:00
|
|
|
|
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
|
|
|
|
2016-04-23 07:55:59 +00:00
|
|
|
auto pending() const -> bool;
|
2016-06-01 11:23:22 +00:00
|
|
|
auto read(double* samples) -> uint;
|
|
|
|
auto write(const double* samples) -> void;
|
2016-04-23 07:55:59 +00:00
|
|
|
|
|
|
|
template<typename... P> auto sample(P&&... p) -> void {
|
2016-06-01 11:23:22 +00:00
|
|
|
double samples[sizeof...(P)] = {forward<P>(p)...};
|
2016-04-23 07:55:59 +00:00
|
|
|
write(samples);
|
|
|
|
}
|
|
|
|
|
|
|
|
private:
|
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;
|
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;
|
2016-04-23 07:55:59 +00:00
|
|
|
|
|
|
|
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
|
|
|
|
|
|
|
}
|