2010-08-09 13:28:56 +00:00
|
|
|
//$00 stop
|
2015-06-27 02:38:47 +00:00
|
|
|
auto GSU::op_stop() {
|
2010-08-09 13:28:56 +00:00
|
|
|
if(regs.cfgr.irq == 0) {
|
|
|
|
|
regs.sfr.irq = 1;
|
2012-04-29 06:16:44 +00:00
|
|
|
stop();
|
2010-08-09 13:28:56 +00:00
|
|
|
}
|
|
|
|
|
regs.sfr.g = 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
|
|
|
regs.pipeline = 0x01; //nop
|
2010-08-09 13:28:56 +00:00
|
|
|
regs.reset();
|
|
|
|
|
}
|
|
|
|
|
|
|
|
|
|
//$01 nop
|
2015-06-27 02:38:47 +00:00
|
|
|
auto GSU::op_nop() {
|
2010-08-09 13:28:56 +00:00
|
|
|
regs.reset();
|
|
|
|
|
}
|
|
|
|
|
|
|
|
|
|
//$02 cache
|
2015-06-27 02:38:47 +00:00
|
|
|
auto GSU::op_cache() {
|
2010-08-09 13:28:56 +00:00
|
|
|
if(regs.cbr != (regs.r[15] & 0xfff0)) {
|
|
|
|
|
regs.cbr = regs.r[15] & 0xfff0;
|
2016-06-05 22:10:01 +00:00
|
|
|
flushCache();
|
2010-08-09 13:28:56 +00:00
|
|
|
}
|
|
|
|
|
regs.reset();
|
|
|
|
|
}
|
|
|
|
|
|
|
|
|
|
//$03 lsr
|
2015-06-27 02:38:47 +00:00
|
|
|
auto GSU::op_lsr() {
|
2010-08-09 13:28:56 +00:00
|
|
|
regs.sfr.cy = (regs.sr() & 1);
|
|
|
|
|
regs.dr() = regs.sr() >> 1;
|
|
|
|
|
regs.sfr.s = (regs.dr() & 0x8000);
|
|
|
|
|
regs.sfr.z = (regs.dr() == 0);
|
|
|
|
|
regs.reset();
|
|
|
|
|
}
|
|
|
|
|
|
|
|
|
|
//$04 rol
|
2015-06-27 02:38:47 +00:00
|
|
|
auto GSU::op_rol() {
|
2010-08-09 13:28:56 +00:00
|
|
|
bool carry = (regs.sr() & 0x8000);
|
|
|
|
|
regs.dr() = (regs.sr() << 1) | regs.sfr.cy;
|
|
|
|
|
regs.sfr.s = (regs.dr() & 0x8000);
|
|
|
|
|
regs.sfr.cy = carry;
|
|
|
|
|
regs.sfr.z = (regs.dr() == 0);
|
|
|
|
|
regs.reset();
|
|
|
|
|
}
|
|
|
|
|
|
|
|
|
|
//$05 bra e
|
|
|
|
|
//$06 blt e
|
|
|
|
|
//$07 bge e
|
|
|
|
|
//$08 bne e
|
|
|
|
|
//$09 beq e
|
|
|
|
|
//$0a bpl e
|
|
|
|
|
//$0b bmi e
|
|
|
|
|
//$0c bcc e
|
|
|
|
|
//$0d bcs e
|
|
|
|
|
//$0e bvc e
|
|
|
|
|
//$0f bvs e
|
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 GSU::op_branch(bool c) {
|
|
|
|
|
auto d = (int8)pipe();
|
|
|
|
|
if(c) regs.r[15] += d;
|
2010-08-09 13:28:56 +00:00
|
|
|
}
|
|
|
|
|
|
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
|
|
|
//$10-1f(b0) to rN
|
|
|
|
|
//$10-1f(b1) move rN
|
|
|
|
|
auto GSU::op_to_move(uint n) {
|
|
|
|
|
if(!regs.sfr.b) {
|
2010-08-09 13:28:56 +00:00
|
|
|
regs.dreg = n;
|
|
|
|
|
} else {
|
|
|
|
|
regs.r[n] = regs.sr();
|
|
|
|
|
regs.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
|
|
|
//$20-2f with rN
|
|
|
|
|
auto GSU::op_with(uint n) {
|
2010-08-09 13:28:56 +00:00
|
|
|
regs.sreg = n;
|
|
|
|
|
regs.dreg = n;
|
|
|
|
|
regs.sfr.b = 1;
|
|
|
|
|
}
|
|
|
|
|
|
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
|
|
|
//$30-3b(alt0) stw (rN)
|
|
|
|
|
//$30-3b(alt1) stb (rN)
|
|
|
|
|
auto GSU::op_store(uint n) {
|
2010-08-09 13:28:56 +00:00
|
|
|
regs.ramaddr = regs.r[n];
|
2016-06-05 22:10:01 +00:00
|
|
|
writeRAMBuffer(regs.ramaddr, regs.sr());
|
|
|
|
|
if(!regs.sfr.alt1) writeRAMBuffer(regs.ramaddr ^ 1, regs.sr() >> 8);
|
2010-08-09 13:28:56 +00:00
|
|
|
regs.reset();
|
|
|
|
|
}
|
|
|
|
|
|
|
|
|
|
//$3c loop
|
2015-06-27 02:38:47 +00:00
|
|
|
auto GSU::op_loop() {
|
2010-08-09 13:28:56 +00:00
|
|
|
regs.r[12]--;
|
|
|
|
|
regs.sfr.s = (regs.r[12] & 0x8000);
|
|
|
|
|
regs.sfr.z = (regs.r[12] == 0);
|
|
|
|
|
if(!regs.sfr.z) regs.r[15] = regs.r[13];
|
|
|
|
|
regs.reset();
|
|
|
|
|
}
|
|
|
|
|
|
|
|
|
|
//$3d alt1
|
2015-06-27 02:38:47 +00:00
|
|
|
auto GSU::op_alt1() {
|
2010-08-09 13:28:56 +00:00
|
|
|
regs.sfr.b = 0;
|
|
|
|
|
regs.sfr.alt1 = 1;
|
|
|
|
|
}
|
|
|
|
|
|
|
|
|
|
//$3e alt2
|
2015-06-27 02:38:47 +00:00
|
|
|
auto GSU::op_alt2() {
|
2010-08-09 13:28:56 +00:00
|
|
|
regs.sfr.b = 0;
|
|
|
|
|
regs.sfr.alt2 = 1;
|
|
|
|
|
}
|
|
|
|
|
|
|
|
|
|
//$3f alt3
|
2015-06-27 02:38:47 +00:00
|
|
|
auto GSU::op_alt3() {
|
2010-08-09 13:28:56 +00:00
|
|
|
regs.sfr.b = 0;
|
|
|
|
|
regs.sfr.alt1 = 1;
|
|
|
|
|
regs.sfr.alt2 = 1;
|
|
|
|
|
}
|
|
|
|
|
|
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
|
|
|
//$40-4b(alt0) ldw (rN)
|
|
|
|
|
//$40-4b(alt1) ldb (rN)
|
|
|
|
|
auto GSU::op_load(uint n) {
|
2010-08-09 13:28:56 +00:00
|
|
|
regs.ramaddr = regs.r[n];
|
2016-06-05 22:10:01 +00:00
|
|
|
regs.dr() = readRAMBuffer(regs.ramaddr);
|
|
|
|
|
if(!regs.sfr.alt1) regs.dr() |= readRAMBuffer(regs.ramaddr ^ 1) << 8;
|
2010-08-09 13:28:56 +00:00
|
|
|
regs.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
|
|
|
//$4c(alt0) plot
|
|
|
|
|
//$4c(alt1) rpix
|
|
|
|
|
auto GSU::op_plot_rpix() {
|
|
|
|
|
if(!regs.sfr.alt1) {
|
|
|
|
|
plot(regs.r[1], regs.r[2]);
|
|
|
|
|
regs.r[1]++;
|
|
|
|
|
} else {
|
|
|
|
|
regs.dr() = rpix(regs.r[1], regs.r[2]);
|
|
|
|
|
regs.sfr.s = (regs.dr() & 0x8000);
|
|
|
|
|
regs.sfr.z = (regs.dr() == 0);
|
|
|
|
|
}
|
2010-08-09 13:28:56 +00:00
|
|
|
regs.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
|
|
|
//$4d swap
|
2015-06-27 02:38:47 +00:00
|
|
|
auto GSU::op_swap() {
|
2010-08-09 13:28:56 +00:00
|
|
|
regs.dr() = (regs.sr() >> 8) | (regs.sr() << 8);
|
|
|
|
|
regs.sfr.s = (regs.dr() & 0x8000);
|
|
|
|
|
regs.sfr.z = (regs.dr() == 0);
|
|
|
|
|
regs.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
|
|
|
//$4e(alt0) color
|
|
|
|
|
//$4e(alt1) cmode
|
|
|
|
|
auto GSU::op_color_cmode() {
|
|
|
|
|
if(!regs.sfr.alt1) {
|
|
|
|
|
regs.colr = color(regs.sr());
|
|
|
|
|
} else {
|
|
|
|
|
regs.por = regs.sr();
|
|
|
|
|
}
|
2010-08-09 13:28:56 +00:00
|
|
|
regs.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
|
|
|
//$4f not
|
2015-06-27 02:38:47 +00:00
|
|
|
auto GSU::op_not() {
|
2010-08-09 13:28:56 +00:00
|
|
|
regs.dr() = ~regs.sr();
|
|
|
|
|
regs.sfr.s = (regs.dr() & 0x8000);
|
|
|
|
|
regs.sfr.z = (regs.dr() == 0);
|
|
|
|
|
regs.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
|
|
|
//$50-5f(alt0) add rN
|
|
|
|
|
//$50-5f(alt1) adc rN
|
|
|
|
|
//$50-5f(alt2) add #N
|
|
|
|
|
//$50-5f(alt3) adc #N
|
|
|
|
|
auto GSU::op_add_adc(uint n) {
|
|
|
|
|
if(!regs.sfr.alt2) n = regs.r[n];
|
|
|
|
|
int r = regs.sr() + n + (regs.sfr.alt1 ? regs.sfr.cy : 0);
|
2010-08-09 13:28:56 +00:00
|
|
|
regs.sfr.ov = ~(regs.sr() ^ n) & (n ^ r) & 0x8000;
|
|
|
|
|
regs.sfr.s = (r & 0x8000);
|
|
|
|
|
regs.sfr.cy = (r >= 0x10000);
|
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
|
|
|
regs.sfr.z = ((uint16)r == 0);
|
2010-08-09 13:28:56 +00:00
|
|
|
regs.dr() = r;
|
|
|
|
|
regs.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
|
|
|
//$60-6f(alt0) sub rN
|
|
|
|
|
//$60-6f(alt1) sbc rN
|
|
|
|
|
//$60-6f(alt2) sub #N
|
|
|
|
|
//$60-6f(alt3) cmp rN
|
|
|
|
|
auto GSU::op_sub_sbc_cmp(uint n) {
|
|
|
|
|
if(!regs.sfr.alt2 || regs.sfr.alt1) n = regs.r[n];
|
|
|
|
|
int r = regs.sr() - n - (!regs.sfr.alt2 && regs.sfr.alt1 ? !regs.sfr.cy : 0);
|
2010-08-09 13:28:56 +00:00
|
|
|
regs.sfr.ov = (regs.sr() ^ n) & (regs.sr() ^ r) & 0x8000;
|
|
|
|
|
regs.sfr.s = (r & 0x8000);
|
|
|
|
|
regs.sfr.cy = (r >= 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
|
|
|
regs.sfr.z = ((uint16)r == 0);
|
|
|
|
|
if(!regs.sfr.alt2 || !regs.sfr.alt1) regs.dr() = r;
|
2010-08-09 13:28:56 +00:00
|
|
|
regs.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
|
|
|
//$70 merge
|
2015-06-27 02:38:47 +00:00
|
|
|
auto GSU::op_merge() {
|
2010-08-09 13:28:56 +00:00
|
|
|
regs.dr() = (regs.r[7] & 0xff00) | (regs.r[8] >> 8);
|
|
|
|
|
regs.sfr.ov = (regs.dr() & 0xc0c0);
|
|
|
|
|
regs.sfr.s = (regs.dr() & 0x8080);
|
|
|
|
|
regs.sfr.cy = (regs.dr() & 0xe0e0);
|
|
|
|
|
regs.sfr.z = (regs.dr() & 0xf0f0);
|
|
|
|
|
regs.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
|
|
|
//$71-7f(alt0) and rN
|
|
|
|
|
//$71-7f(alt1) bic rN
|
|
|
|
|
//$71-7f(alt2) and #N
|
|
|
|
|
//$71-7f(alt3) bic #N
|
|
|
|
|
auto GSU::op_and_bic(uint n) {
|
|
|
|
|
if(!regs.sfr.alt2) n = regs.r[n];
|
|
|
|
|
regs.dr() = regs.sr() & (regs.sfr.alt1 ? ~n : n);
|
2010-08-09 13:28:56 +00:00
|
|
|
regs.sfr.s = (regs.dr() & 0x8000);
|
|
|
|
|
regs.sfr.z = (regs.dr() == 0);
|
|
|
|
|
regs.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
|
|
|
//$80-8f(alt0) mult rN
|
|
|
|
|
//$80-8f(alt1) umult rN
|
|
|
|
|
//$80-8f(alt2) mult #N
|
|
|
|
|
//$80-8f(alt3) umult #N
|
|
|
|
|
auto GSU::op_mult_umult(uint n) {
|
|
|
|
|
if(!regs.sfr.alt2) n = regs.r[n];
|
|
|
|
|
regs.dr() = (!regs.sfr.alt1 ? ((int8)regs.sr() * (int8)n) : ((uint8)regs.sr() * (uint8)n));
|
2010-08-09 13:28:56 +00:00
|
|
|
regs.sfr.s = (regs.dr() & 0x8000);
|
|
|
|
|
regs.sfr.z = (regs.dr() == 0);
|
|
|
|
|
regs.reset();
|
2015-06-28 08:44:56 +00:00
|
|
|
if(!regs.cfgr.ms0) step(regs.clsr ? 1 : 2);
|
2010-08-09 13:28:56 +00:00
|
|
|
}
|
|
|
|
|
|
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
|
|
|
//$90 sbk
|
2015-06-27 02:38:47 +00:00
|
|
|
auto GSU::op_sbk() {
|
2016-06-05 22:10:01 +00:00
|
|
|
writeRAMBuffer(regs.ramaddr ^ 0, regs.sr() >> 0);
|
|
|
|
|
writeRAMBuffer(regs.ramaddr ^ 1, regs.sr() >> 8);
|
2010-08-09 13:28:56 +00:00
|
|
|
regs.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
|
|
|
//$91-94 link #N
|
|
|
|
|
auto GSU::op_link(uint n) {
|
2010-08-09 13:28:56 +00:00
|
|
|
regs.r[11] = regs.r[15] + n;
|
|
|
|
|
regs.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
|
|
|
//$95 sex
|
2015-06-27 02:38:47 +00:00
|
|
|
auto GSU::op_sex() {
|
2010-08-09 13:28:56 +00:00
|
|
|
regs.dr() = (int8)regs.sr();
|
|
|
|
|
regs.sfr.s = (regs.dr() & 0x8000);
|
|
|
|
|
regs.sfr.z = (regs.dr() == 0);
|
|
|
|
|
regs.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
|
|
|
//$96(alt0) asr
|
|
|
|
|
//$96(alt1) div2
|
|
|
|
|
auto GSU::op_asr_div2() {
|
2010-08-09 13:28:56 +00:00
|
|
|
regs.sfr.cy = (regs.sr() & 1);
|
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
|
|
|
regs.dr() = ((int16)regs.sr() >> 1) + (regs.sfr.alt1 ? ((regs.sr() + 1) >> 16) : 0);
|
2010-08-09 13:28:56 +00:00
|
|
|
regs.sfr.s = (regs.dr() & 0x8000);
|
|
|
|
|
regs.sfr.z = (regs.dr() == 0);
|
|
|
|
|
regs.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
|
|
|
//$97 ror
|
2015-06-27 02:38:47 +00:00
|
|
|
auto GSU::op_ror() {
|
2010-08-09 13:28:56 +00:00
|
|
|
bool carry = (regs.sr() & 1);
|
|
|
|
|
regs.dr() = (regs.sfr.cy << 15) | (regs.sr() >> 1);
|
|
|
|
|
regs.sfr.s = (regs.dr() & 0x8000);
|
|
|
|
|
regs.sfr.cy = carry;
|
|
|
|
|
regs.sfr.z = (regs.dr() == 0);
|
|
|
|
|
regs.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
|
|
|
//$98-9d(alt0) jmp rN
|
|
|
|
|
//$98-9d(alt1) ljmp rN
|
|
|
|
|
auto GSU::op_jmp_ljmp(uint n) {
|
|
|
|
|
if(!regs.sfr.alt1) {
|
|
|
|
|
regs.r[15] = regs.r[n];
|
|
|
|
|
} else {
|
|
|
|
|
regs.pbr = regs.r[n] & 0x7f;
|
|
|
|
|
regs.r[15] = regs.sr();
|
|
|
|
|
regs.cbr = regs.r[15] & 0xfff0;
|
2016-06-05 22:10:01 +00:00
|
|
|
flushCache();
|
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
|
|
|
}
|
2010-08-09 13:28:56 +00:00
|
|
|
regs.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
|
|
|
//$9e lob
|
2015-06-27 02:38:47 +00:00
|
|
|
auto GSU::op_lob() {
|
2010-08-09 13:28:56 +00:00
|
|
|
regs.dr() = regs.sr() & 0xff;
|
|
|
|
|
regs.sfr.s = (regs.dr() & 0x80);
|
|
|
|
|
regs.sfr.z = (regs.dr() == 0);
|
|
|
|
|
regs.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
|
|
|
//$9f(alt0) fmult
|
|
|
|
|
//$9f(alt1) lmult
|
|
|
|
|
auto GSU::op_fmult_lmult() {
|
|
|
|
|
uint32 result = (int16)regs.sr() * (int16)regs.r[6];
|
|
|
|
|
if(regs.sfr.alt1) regs.r[4] = result;
|
2010-08-09 13:28:56 +00:00
|
|
|
regs.dr() = result >> 16;
|
|
|
|
|
regs.sfr.s = (regs.dr() & 0x8000);
|
|
|
|
|
regs.sfr.cy = (result & 0x8000);
|
|
|
|
|
regs.sfr.z = (regs.dr() == 0);
|
|
|
|
|
regs.reset();
|
2015-06-28 08:44:56 +00:00
|
|
|
step((regs.cfgr.ms0 ? 3 : 7) * (regs.clsr ? 1 : 2));
|
2010-08-09 13:28:56 +00:00
|
|
|
}
|
|
|
|
|
|
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
|
|
|
//$a0-af(alt0) ibt rN,#pp
|
|
|
|
|
//$a0-af(alt1) lms rN,(yy)
|
|
|
|
|
//$a0-af(alt2) sms (yy),rN
|
|
|
|
|
auto GSU::op_ibt_lms_sms(uint n) {
|
|
|
|
|
if(regs.sfr.alt1) {
|
|
|
|
|
regs.ramaddr = pipe() << 1;
|
2016-06-05 22:10:01 +00:00
|
|
|
uint8 lo = readRAMBuffer(regs.ramaddr ^ 0) << 0;
|
|
|
|
|
regs.r[n] = readRAMBuffer(regs.ramaddr ^ 1) << 8 | lo;
|
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
|
|
|
} else if(regs.sfr.alt2) {
|
|
|
|
|
regs.ramaddr = pipe() << 1;
|
2016-06-05 22:10:01 +00:00
|
|
|
writeRAMBuffer(regs.ramaddr ^ 0, regs.r[n] >> 0);
|
|
|
|
|
writeRAMBuffer(regs.ramaddr ^ 1, regs.r[n] >> 8);
|
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
|
|
|
} else {
|
|
|
|
|
regs.r[n] = (int8)pipe();
|
|
|
|
|
}
|
2010-08-09 13:28:56 +00:00
|
|
|
regs.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
|
|
|
//$b0-bf(b0) from rN
|
|
|
|
|
//$b0-bf(b1) moves rN
|
|
|
|
|
auto GSU::op_from_moves(uint n) {
|
|
|
|
|
if(!regs.sfr.b) {
|
2010-08-09 13:28:56 +00:00
|
|
|
regs.sreg = n;
|
|
|
|
|
} else {
|
|
|
|
|
regs.dr() = regs.r[n];
|
|
|
|
|
regs.sfr.ov = (regs.dr() & 0x80);
|
|
|
|
|
regs.sfr.s = (regs.dr() & 0x8000);
|
|
|
|
|
regs.sfr.z = (regs.dr() == 0);
|
|
|
|
|
regs.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
|
|
|
//$c0 hib
|
2015-06-27 02:38:47 +00:00
|
|
|
auto GSU::op_hib() {
|
2010-08-09 13:28:56 +00:00
|
|
|
regs.dr() = regs.sr() >> 8;
|
|
|
|
|
regs.sfr.s = (regs.dr() & 0x80);
|
|
|
|
|
regs.sfr.z = (regs.dr() == 0);
|
|
|
|
|
regs.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
|
|
|
//$c1-cf(alt0) or rN
|
|
|
|
|
//$c1-cf(alt1) xor rN
|
|
|
|
|
//$c1-cf(alt2) or #N
|
|
|
|
|
//$c1-cf(alt3) xor #N
|
|
|
|
|
auto GSU::op_or_xor(uint n) {
|
|
|
|
|
if(!regs.sfr.alt2) n = regs.r[n];
|
|
|
|
|
regs.dr() = (!regs.sfr.alt1 ? (regs.sr() | n) : (regs.sr() ^ n));
|
2010-08-09 13:28:56 +00:00
|
|
|
regs.sfr.s = (regs.dr() & 0x8000);
|
|
|
|
|
regs.sfr.z = (regs.dr() == 0);
|
|
|
|
|
regs.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
|
|
|
//$d0-de inc rN
|
|
|
|
|
auto GSU::op_inc(uint n) {
|
2010-08-09 13:28:56 +00:00
|
|
|
regs.r[n]++;
|
|
|
|
|
regs.sfr.s = (regs.r[n] & 0x8000);
|
|
|
|
|
regs.sfr.z = (regs.r[n] == 0);
|
|
|
|
|
regs.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
|
|
|
//$df(alt0) getc
|
|
|
|
|
//$df(alt2) ramb
|
|
|
|
|
//$df(alt3) romb
|
|
|
|
|
auto GSU::op_getc_ramb_romb() {
|
|
|
|
|
if(!regs.sfr.alt2) {
|
2016-06-05 22:10:01 +00:00
|
|
|
regs.colr = color(readROMBuffer());
|
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
|
|
|
} else if(!regs.sfr.alt1) {
|
2016-06-05 22:10:01 +00:00
|
|
|
syncRAMBuffer();
|
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
|
|
|
regs.rambr = regs.sr() & 0x01;
|
|
|
|
|
} else {
|
2016-06-05 22:10:01 +00:00
|
|
|
syncROMBuffer();
|
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
|
|
|
regs.rombr = regs.sr() & 0x7f;
|
|
|
|
|
}
|
2010-08-09 13:28:56 +00:00
|
|
|
regs.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
|
|
|
//$e0-ee dec rN
|
|
|
|
|
auto GSU::op_dec(uint n) {
|
2010-08-09 13:28:56 +00:00
|
|
|
regs.r[n]--;
|
|
|
|
|
regs.sfr.s = (regs.r[n] & 0x8000);
|
|
|
|
|
regs.sfr.z = (regs.r[n] == 0);
|
|
|
|
|
regs.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
|
|
|
//$ef(alt0) getb
|
|
|
|
|
//$ef(alt1) getbh
|
|
|
|
|
//$ef(alt2) getbl
|
|
|
|
|
//$ef(alt3) getbs
|
2015-06-27 02:38:47 +00:00
|
|
|
auto GSU::op_getb() {
|
Update to v098r13 release.
byuu says:
Changelog:
- nall/dsp returns with new iir/biquad.hpp and resampler/cubic.hpp files
- nall/queue.hpp added (simple ring buffer ... nall/vector wouldn't
cause too many moves with FIFO)
- audio streams now only buffer 20ms; so even if multiple audio streams
desync, latency can never exceed 20ms
- replaced blackman windwed sinc FIR hermite audio filter with transposed
direct form II biquadratic sixth-order IIR butterworth filter (better
attenuation of frequencies above 20KHz, faster, no need for decimation,
less code)
- put in experimental eight-tap echo filter (a lot better than what I
had before, but still rather weak)
- substantial cleanups to the SuperFX GSU processor core (slightly
faster, 479KB->100KB object file, 42.7KB->33.4KB source code size,
way less code duplication)
We'll definitely want to test the whole SuperFX library (not many games)
just to make sure there's no regressions caused by this one.
Not sure what I want to do with audio processing effects yet. I've always
really wanted lots of fun controls to customize audio, and now finally
with this new biquad filter, I can finally start implementing real
effects. For instance, an equalizer wouldn't be too complicated anymore.
The new reverb effect is still a poor man's version. I need to find human
readable source for implementing a comb-filter properly. I'm pretty sure
I can already treat nall::queue as an all-pass filter since all that
does is phase shift (fancy audio term for "delay audio"). What's really
going to be hard is figuring out how to expose user-friendly settings for
controlling it. It looks like you need a bunch of coprime coefficients,
and I don't think casual users are going to be able to hand-enter coprime
values to get the echo effect they want. I uh ... don't even know how
to calculate coprime values dynamically right now >_> But we're going
to have to, as they are correlated to the output sampling rate.
We'll definitely want to make some audio profiles so that users can
quickly select pre-configured themes that sound nice, but expose the
underlying coefficients so that they can tweak stuff to their liking. This
isn't just about higan, this is about me trying to learn digital signal
processing, so please don't be too upset about feature creep or anything
on this.
Anyway ... I'm having some difficulties with my audio right now. When
the reverb effect is enabled, there's a bunch of static on system
reset for just a moment. But this should not be possible. nall::queue
is initializing all previous reverb sample elements to 0.0. I don't
understand where static is coming in from. Further, we have the same
issue with both the windowed sinc and the biquad filters ... a bit of
a popping sound when starting a game. Any help tracking this down would
be appreciated.
There's also one really annoying issue ... I can't seem to do reverb
or volume adjustments with normalized samples. If I say "volume *= 0.5"
in higan/audio/audio.cpp line 68, it doesn't just halve the volume, it
adds a whole bunch of distortion. This makes absolutely zero sense to
me. The sample values are between 0.0 (mute) and 1.0 (full volume) here,
so multiplying a double by 0.5 shouldn't cause distortion. So right now,
I'm doing these adjustments with less precision after denormalizing back
to int16. Anyone ever see something like that? :/
2016-05-31 22:29:36 +00:00
|
|
|
switch(regs.sfr.alt2 << 1 | regs.sfr.alt1 << 0) {
|
2016-06-05 22:10:01 +00:00
|
|
|
case 0: regs.dr() = readROMBuffer(); break;
|
|
|
|
|
case 1: regs.dr() = readROMBuffer() << 8 | (uint8)regs.sr(); break;
|
|
|
|
|
case 2: regs.dr() = (regs.sr() & 0xff00) | readROMBuffer(); break;
|
|
|
|
|
case 3: regs.dr() = (int8)readROMBuffer(); break;
|
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
|
|
|
}
|
2010-08-09 13:28:56 +00:00
|
|
|
regs.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
|
|
|
//$f0-ff(alt0) iwt rN,#xx
|
|
|
|
|
//$f0-ff(alt1) lm rN,(xx)
|
|
|
|
|
//$f0-ff(alt2) sm (xx),rN
|
|
|
|
|
auto GSU::op_iwt_lm_sm(uint n) {
|
|
|
|
|
if(regs.sfr.alt1) {
|
|
|
|
|
regs.ramaddr = pipe() << 0;
|
|
|
|
|
regs.ramaddr |= pipe() << 8;
|
2016-06-05 22:10:01 +00:00
|
|
|
uint8 lo = readRAMBuffer(regs.ramaddr ^ 0) << 0;
|
|
|
|
|
regs.r[n] = readRAMBuffer(regs.ramaddr ^ 1) << 8 | lo;
|
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
|
|
|
} else if(regs.sfr.alt2) {
|
|
|
|
|
regs.ramaddr = pipe() << 0;
|
|
|
|
|
regs.ramaddr |= pipe() << 8;
|
2016-06-05 22:10:01 +00:00
|
|
|
writeRAMBuffer(regs.ramaddr ^ 0, regs.r[n] >> 0);
|
|
|
|
|
writeRAMBuffer(regs.ramaddr ^ 1, regs.r[n] >> 8);
|
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
|
|
|
} else {
|
|
|
|
|
uint8 lo = pipe();
|
|
|
|
|
regs.r[n] = pipe() << 8 | lo;
|
|
|
|
|
}
|
2010-08-09 13:28:56 +00:00
|
|
|
regs.reset();
|
|
|
|
|
}
|