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
|
|
|
#include <emulator/emulator.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 {
|
|
|
|
|
2016-04-23 07:55:59 +00:00
|
|
|
#include "stream.cpp"
|
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
|
|
|
Audio audio;
|
|
|
|
|
2016-06-01 11:23:22 +00:00
|
|
|
auto Audio::reset(maybe<uint> channels_, maybe<double> frequency_) -> void {
|
|
|
|
if(channels_) channels = channels_();
|
|
|
|
if(frequency_) frequency = frequency_();
|
|
|
|
|
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
|
|
|
streams.reset();
|
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
|
|
|
reverb.reset();
|
2016-06-01 11:23:22 +00:00
|
|
|
|
|
|
|
reverb.resize(channels);
|
|
|
|
for(auto c : range(channels)) {
|
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
|
|
|
reverb[c].resize(7);
|
|
|
|
reverb[c][0].resize(1229);
|
|
|
|
reverb[c][1].resize(1559);
|
|
|
|
reverb[c][2].resize(1907);
|
|
|
|
reverb[c][3].resize(4057);
|
|
|
|
reverb[c][4].resize(8117);
|
|
|
|
reverb[c][5].resize(8311);
|
|
|
|
reverb[c][6].resize(9931);
|
|
|
|
}
|
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 Audio::setInterface(Interface* interface) -> void {
|
|
|
|
this->interface = interface;
|
|
|
|
}
|
|
|
|
|
|
|
|
auto Audio::setVolume(double volume) -> void {
|
|
|
|
this->volume = volume;
|
|
|
|
}
|
|
|
|
|
|
|
|
auto Audio::setBalance(double balance) -> void {
|
|
|
|
this->balance = balance;
|
|
|
|
}
|
|
|
|
|
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 Audio::setReverb(bool enabled) -> void {
|
|
|
|
this->reverbEnable = enabled;
|
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 Audio::createStream(uint channels, double frequency) -> shared_pointer<Stream> {
|
2016-06-01 11:23:22 +00:00
|
|
|
shared_pointer<Stream> stream = new Stream;
|
|
|
|
stream->reset(channels, frequency, this->frequency);
|
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
|
|
|
streams.append(stream);
|
|
|
|
return stream;
|
|
|
|
}
|
|
|
|
|
2016-06-01 11:23:22 +00:00
|
|
|
auto Audio::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
|
|
|
while(true) {
|
|
|
|
for(auto& stream : streams) {
|
2016-04-23 07:55:59 +00:00
|
|
|
if(!stream->pending()) return;
|
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
|
|
|
}
|
|
|
|
|
2016-06-01 11:23:22 +00:00
|
|
|
double samples[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
|
|
|
for(auto& stream : streams) {
|
2016-06-01 11:23:22 +00:00
|
|
|
double buffer[16];
|
|
|
|
uint length = stream->read(buffer), offset = 0;
|
|
|
|
|
|
|
|
for(auto c : range(channels)) {
|
|
|
|
samples[c] += buffer[offset];
|
|
|
|
if(++offset >= length) offset = 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
|
|
|
}
|
2016-06-01 11:23:22 +00:00
|
|
|
|
|
|
|
for(auto c : range(channels)) {
|
|
|
|
samples[c] /= streams.size();
|
|
|
|
|
|
|
|
if(reverbEnable) {
|
|
|
|
samples[c] *= 0.125;
|
|
|
|
for(auto n : range(7)) samples[c] += 0.125 * reverb[c][n].last();
|
|
|
|
for(auto n : range(7)) reverb[c][n].write(samples[c]);
|
|
|
|
samples[c] *= 8.000;
|
|
|
|
}
|
|
|
|
|
|
|
|
samples[c] *= volume;
|
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
|
|
|
}
|
|
|
|
|
2016-06-01 11:23:22 +00:00
|
|
|
if(channels == 2) {
|
|
|
|
if(balance < 0.0) samples[1] *= 1.0 + balance;
|
|
|
|
if(balance > 0.0) samples[0] *= 1.0 - balance;
|
|
|
|
}
|
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
|
|
|
|
2016-06-01 11:23:22 +00:00
|
|
|
interface->audioSample(samples, channels);
|
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
|
|
|
}
|
|
|
|
}
|
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
|
|
|
|
|
|
|
}
|