2016-06-01 11:23:22 +00:00
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auto Stream::reset(uint channels_, double inputFrequency, double outputFrequency) -> void {
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channels.reset();
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channels.resize(channels_);
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for(auto& channel : channels) {
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if(outputFrequency / inputFrequency <= 0.5) {
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channel.iir.resize(order / 2);
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for(auto phase : range(order / 2)) {
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double q = DSP::IIR::Biquad::butterworth(order, phase);
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Update to v102r15 release.
byuu says:
Changelog:
- nall: added DSP::IIR::OnePole (which is a first-order IIR filter)
- FC/APU: removed strong highpass, weak hipass filters (and the
dummied out lowpass filter)
- MS,GG,MD/PSG: removed lowpass filter
- MS,GG,MD/PSG: audio was not being centered properly; removed
centering for now
- MD/YM2612: fixed clipping of accumulator from 18 signed bits to 14
signed bits (-0x2000 to +0x1fff) [Cydrak]
- MD/YM2612: removed lowpass filter
- PCE/PSG: audio was not being centered properly; removed centering
for now
First thing is that I've removed all of the ad-hoc audio filtering.
Emulator::Stream intrinsically provides a three-pass, second-order
biquad IIR butterworth lowpass filter that clips frequencies above 20KHz
with very good attenuation (as good as IIR gets, anyway.)
It doesn't really make sense to have the various cores running
additional lowpass filters. If we want to filter frequencies below
20KHz, then I can adapt Emulator::Audio::createStream() to take a cutoff
frequency value, and we can do it all at once, with much better quality.
Right now, I don't know what frequencies are best to cut off the various
other audio cores, so they're just gone for now.
As for the highpass filters for the Famicom core, well ... you don't get
aliasing from resampling low frequencies. And generally speaking, too
low a frequency will be inaudible anyway. All these were doing was
killing possible bass (if they were too strong.) We can add them again,
but only if someone can convert Ryphecha's ad-hoc magic integers into a
frequency cutoff. In which case, I'll use my biquad IIR filter to do it
even better. On this note, it may prove useful to do this for the MD PSG
as well, to try and head off unnecessary clamping when mixing with the
YM2612.
Finally, there was the audio centering issue that affected the
MS,GG,MD,PCE,SG cores. It was flooring the "silent" audio level, which
was resulting in extremely heavy distortion if you tried listening to
higan and, say, audacious at the same time. Without the botched
centering, this distortion is completely gone now.
However, without any centering, we've halved the potential volume range.
This means the audio slider in higan's audio settings panel will start
clamping twice as quickly. So ultimately, we need to figure out how to
fix the centering. This isn't as simple as just subtracting less. We
will probably have to center every individual audio channel before
summing them to do this properly.
Results:
On the Mega Drive, Altered Beast sounds quite a bit better, a lot less
distortion now. But it's still not perfect, especially sound effects.
Further, Bare Knuckle / Streets of Rage still has really bad sound
effects. It looks like I broke something in Cydrak's code when trying to
adapt it to my style =(
2017-03-06 20:23:22 +00:00
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channel.iir[phase].reset(DSP::IIR::Biquad::Type::LowPass, 20000.0, inputFrequency, q);
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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
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}
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2016-04-23 07:55:59 +00:00
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}
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2016-06-01 11:23:22 +00:00
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channel.resampler.reset(inputFrequency, outputFrequency);
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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
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}
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2016-04-23 07:55:59 +00:00
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}
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auto Stream::pending() const -> bool {
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2016-06-01 11:23:22 +00:00
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return channels && channels[0].resampler.pending();
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2016-04-23 07:55:59 +00:00
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}
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2016-06-01 11:23:22 +00:00
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auto Stream::read(double* samples) -> uint {
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for(auto c : range(channels)) samples[c] = channels[c].resampler.read();
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return channels.size();
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2016-04-23 07:55:59 +00:00
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}
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2016-06-01 11:23:22 +00:00
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auto Stream::write(const double* samples) -> void {
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2016-04-23 07:55:59 +00:00
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for(auto c : range(channels)) {
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2016-06-01 11:23:22 +00:00
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double sample = samples[c] + 1e-25; //constant offset used to suppress denormals
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for(auto& iir : channels[c].iir) sample = iir.process(sample);
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channels[c].resampler.write(sample);
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2016-04-23 07:55:59 +00:00
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}
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2016-06-01 11:23:22 +00:00
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audio.process();
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2016-04-23 07:55:59 +00:00
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}
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