bsnes/higan/audio/stream.cpp

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Update to v098r14 release. byuu says: Changelog: - improved attenuation of biquad filter by computing butterworth Q coefficients correctly (instead of using the same constant) - adding 1e-25 to each input sample into the biquad filters to try and prevent denormalization - updated normalization from [0.0 to 1.0] to [-1.0 to +1.0]; volume/reverb happen in floating-point mode now - good amount of work to make the base Emulator::Audio support any number of output channels - so that we don't have to do separate work on left/right channels; and can instead share the code for each channel - Emulator::Interface::audioSample(int16 left, int16 right); changed to: - Emulator::Interface::audioSample(double* samples, uint channels); - samples are normalized [-1.0 to +1.0] - for now at least, channels will be the value given to Emulator::Audio::reset() - fixed GUI crash on startup when audio driver is set to None I'm probably going to be updating ruby to accept normalized doubles as well; but I'm not sure if I will try and support anything other 2-channel audio output. It'll depend on how easy it is to do so; perhaps it'll be a per-driver setting. The denormalization thing is fierce. If that happens, it drops the emulator framerate from 220fps to about 20fps for Game Boy emulation. And that happens basically whenever audio output is silent. I'm probably also going to make a nall/denormal.hpp file at some point with platform-specific functionality to set the CPU state to "denormals as zero" where applicable. I'll still add the 1e-25 offset (inaudible) as another fallback.
2016-06-01 11:23:22 +00:00
auto Stream::reset(uint channels_, double inputFrequency, double outputFrequency) -> void {
channels.reset();
channels.resize(channels_);
for(auto& channel : channels) {
if(outputFrequency / inputFrequency <= 0.5) {
channel.iir.resize(order / 2);
for(auto phase : range(order / 2)) {
double q = DSP::IIR::Biquad::butterworth(order, phase);
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
channel.iir[phase].reset(DSP::IIR::Biquad::Type::LowPass, 20000.0, inputFrequency, q);
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
}
}
Update to v098r14 release. byuu says: Changelog: - improved attenuation of biquad filter by computing butterworth Q coefficients correctly (instead of using the same constant) - adding 1e-25 to each input sample into the biquad filters to try and prevent denormalization - updated normalization from [0.0 to 1.0] to [-1.0 to +1.0]; volume/reverb happen in floating-point mode now - good amount of work to make the base Emulator::Audio support any number of output channels - so that we don't have to do separate work on left/right channels; and can instead share the code for each channel - Emulator::Interface::audioSample(int16 left, int16 right); changed to: - Emulator::Interface::audioSample(double* samples, uint channels); - samples are normalized [-1.0 to +1.0] - for now at least, channels will be the value given to Emulator::Audio::reset() - fixed GUI crash on startup when audio driver is set to None I'm probably going to be updating ruby to accept normalized doubles as well; but I'm not sure if I will try and support anything other 2-channel audio output. It'll depend on how easy it is to do so; perhaps it'll be a per-driver setting. The denormalization thing is fierce. If that happens, it drops the emulator framerate from 220fps to about 20fps for Game Boy emulation. And that happens basically whenever audio output is silent. I'm probably also going to make a nall/denormal.hpp file at some point with platform-specific functionality to set the CPU state to "denormals as zero" where applicable. I'll still add the 1e-25 offset (inaudible) as another fallback.
2016-06-01 11:23:22 +00:00
channel.resampler.reset(inputFrequency, outputFrequency);
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 Stream::pending() const -> bool {
Update to v098r14 release. byuu says: Changelog: - improved attenuation of biquad filter by computing butterworth Q coefficients correctly (instead of using the same constant) - adding 1e-25 to each input sample into the biquad filters to try and prevent denormalization - updated normalization from [0.0 to 1.0] to [-1.0 to +1.0]; volume/reverb happen in floating-point mode now - good amount of work to make the base Emulator::Audio support any number of output channels - so that we don't have to do separate work on left/right channels; and can instead share the code for each channel - Emulator::Interface::audioSample(int16 left, int16 right); changed to: - Emulator::Interface::audioSample(double* samples, uint channels); - samples are normalized [-1.0 to +1.0] - for now at least, channels will be the value given to Emulator::Audio::reset() - fixed GUI crash on startup when audio driver is set to None I'm probably going to be updating ruby to accept normalized doubles as well; but I'm not sure if I will try and support anything other 2-channel audio output. It'll depend on how easy it is to do so; perhaps it'll be a per-driver setting. The denormalization thing is fierce. If that happens, it drops the emulator framerate from 220fps to about 20fps for Game Boy emulation. And that happens basically whenever audio output is silent. I'm probably also going to make a nall/denormal.hpp file at some point with platform-specific functionality to set the CPU state to "denormals as zero" where applicable. I'll still add the 1e-25 offset (inaudible) as another fallback.
2016-06-01 11:23:22 +00:00
return channels && channels[0].resampler.pending();
}
Update to v098r14 release. byuu says: Changelog: - improved attenuation of biquad filter by computing butterworth Q coefficients correctly (instead of using the same constant) - adding 1e-25 to each input sample into the biquad filters to try and prevent denormalization - updated normalization from [0.0 to 1.0] to [-1.0 to +1.0]; volume/reverb happen in floating-point mode now - good amount of work to make the base Emulator::Audio support any number of output channels - so that we don't have to do separate work on left/right channels; and can instead share the code for each channel - Emulator::Interface::audioSample(int16 left, int16 right); changed to: - Emulator::Interface::audioSample(double* samples, uint channels); - samples are normalized [-1.0 to +1.0] - for now at least, channels will be the value given to Emulator::Audio::reset() - fixed GUI crash on startup when audio driver is set to None I'm probably going to be updating ruby to accept normalized doubles as well; but I'm not sure if I will try and support anything other 2-channel audio output. It'll depend on how easy it is to do so; perhaps it'll be a per-driver setting. The denormalization thing is fierce. If that happens, it drops the emulator framerate from 220fps to about 20fps for Game Boy emulation. And that happens basically whenever audio output is silent. I'm probably also going to make a nall/denormal.hpp file at some point with platform-specific functionality to set the CPU state to "denormals as zero" where applicable. I'll still add the 1e-25 offset (inaudible) as another fallback.
2016-06-01 11:23:22 +00:00
auto Stream::read(double* samples) -> uint {
for(auto c : range(channels)) samples[c] = channels[c].resampler.read();
return channels.size();
}
Update to v098r14 release. byuu says: Changelog: - improved attenuation of biquad filter by computing butterworth Q coefficients correctly (instead of using the same constant) - adding 1e-25 to each input sample into the biquad filters to try and prevent denormalization - updated normalization from [0.0 to 1.0] to [-1.0 to +1.0]; volume/reverb happen in floating-point mode now - good amount of work to make the base Emulator::Audio support any number of output channels - so that we don't have to do separate work on left/right channels; and can instead share the code for each channel - Emulator::Interface::audioSample(int16 left, int16 right); changed to: - Emulator::Interface::audioSample(double* samples, uint channels); - samples are normalized [-1.0 to +1.0] - for now at least, channels will be the value given to Emulator::Audio::reset() - fixed GUI crash on startup when audio driver is set to None I'm probably going to be updating ruby to accept normalized doubles as well; but I'm not sure if I will try and support anything other 2-channel audio output. It'll depend on how easy it is to do so; perhaps it'll be a per-driver setting. The denormalization thing is fierce. If that happens, it drops the emulator framerate from 220fps to about 20fps for Game Boy emulation. And that happens basically whenever audio output is silent. I'm probably also going to make a nall/denormal.hpp file at some point with platform-specific functionality to set the CPU state to "denormals as zero" where applicable. I'll still add the 1e-25 offset (inaudible) as another fallback.
2016-06-01 11:23:22 +00:00
auto Stream::write(const double* samples) -> void {
for(auto c : range(channels)) {
Update to v098r14 release. byuu says: Changelog: - improved attenuation of biquad filter by computing butterworth Q coefficients correctly (instead of using the same constant) - adding 1e-25 to each input sample into the biquad filters to try and prevent denormalization - updated normalization from [0.0 to 1.0] to [-1.0 to +1.0]; volume/reverb happen in floating-point mode now - good amount of work to make the base Emulator::Audio support any number of output channels - so that we don't have to do separate work on left/right channels; and can instead share the code for each channel - Emulator::Interface::audioSample(int16 left, int16 right); changed to: - Emulator::Interface::audioSample(double* samples, uint channels); - samples are normalized [-1.0 to +1.0] - for now at least, channels will be the value given to Emulator::Audio::reset() - fixed GUI crash on startup when audio driver is set to None I'm probably going to be updating ruby to accept normalized doubles as well; but I'm not sure if I will try and support anything other 2-channel audio output. It'll depend on how easy it is to do so; perhaps it'll be a per-driver setting. The denormalization thing is fierce. If that happens, it drops the emulator framerate from 220fps to about 20fps for Game Boy emulation. And that happens basically whenever audio output is silent. I'm probably also going to make a nall/denormal.hpp file at some point with platform-specific functionality to set the CPU state to "denormals as zero" where applicable. I'll still add the 1e-25 offset (inaudible) as another fallback.
2016-06-01 11:23:22 +00:00
double sample = samples[c] + 1e-25; //constant offset used to suppress denormals
for(auto& iir : channels[c].iir) sample = iir.process(sample);
channels[c].resampler.write(sample);
}
Update to v098r14 release. byuu says: Changelog: - improved attenuation of biquad filter by computing butterworth Q coefficients correctly (instead of using the same constant) - adding 1e-25 to each input sample into the biquad filters to try and prevent denormalization - updated normalization from [0.0 to 1.0] to [-1.0 to +1.0]; volume/reverb happen in floating-point mode now - good amount of work to make the base Emulator::Audio support any number of output channels - so that we don't have to do separate work on left/right channels; and can instead share the code for each channel - Emulator::Interface::audioSample(int16 left, int16 right); changed to: - Emulator::Interface::audioSample(double* samples, uint channels); - samples are normalized [-1.0 to +1.0] - for now at least, channels will be the value given to Emulator::Audio::reset() - fixed GUI crash on startup when audio driver is set to None I'm probably going to be updating ruby to accept normalized doubles as well; but I'm not sure if I will try and support anything other 2-channel audio output. It'll depend on how easy it is to do so; perhaps it'll be a per-driver setting. The denormalization thing is fierce. If that happens, it drops the emulator framerate from 220fps to about 20fps for Game Boy emulation. And that happens basically whenever audio output is silent. I'm probably also going to make a nall/denormal.hpp file at some point with platform-specific functionality to set the CPU state to "denormals as zero" where applicable. I'll still add the 1e-25 offset (inaudible) as another fallback.
2016-06-01 11:23:22 +00:00
audio.process();
}