- Welcome to NeXTcommunity.
Recent posts
#1
General Discussion / Re: Steve Jobs’s Wilderness Y...
Last post by Rhetorica - Today at 11:04 AMQuote from: jeffburg on May 17, 2026, 09:40 AM[...] the other forum members have little interest in OS X Macs. In fact they value the 2000's era Macs that can still run Mac OS 9... no idea why lol.I have had not one but two conversations on Reddit that boil down to someone asking, "What's the appropriate OS version for this early New World Mac that I just got?" — only for the correct response (Rhapsody) to get booed by absolute plebeians who expect 8.6 or 9.0 to be the answer!
Quote from: jeffburg on May 17, 2026, 09:40 AMAnd same goes for reverse here, I hope this doesn't offend any members here, but in my opinion if you are a NeXT fan, you have to use a Mac. I am still running NeXTSTEP, it's just much newer and shinier, but the guts are still the same. This was totally my experience when making MathEdit. The amount of changes I needed to make to the app to get it to work on both OPENSTEP 4.2 and on macOS Tahoe were not as large as you think. Not to mention it also runs in GNUStep. So yeah. I love my shiny aluminum ARM64 NeXTSTATIONI think the standard rebuttal to this is something like, "Maybe if Tim Cook drops dead and the walled garden nonsense stops." Let us remember Mavericks Forever!
#2
Virtualization / Re: What Needs to be done for ...
Last post by andreas_g - May 28, 2026, 12:24 PM@Protocol 7
Thank you very much for the link. I had a look at cdda2wav and it seems to confirm my findings. I'll do some testing during the weekend.
Thank you very much for the link. I had a look at cdda2wav and it seems to confirm my findings. I'll do some testing during the weekend.
#3
Virtualization / Re: What Needs to be done for ...
Last post by Protocol 7 - May 28, 2026, 08:31 AMYou can find some info about CDDA de-emphasis here. Maybe it can lead in the right direction. SoX has a de-emphasis filter for example.
#4
Virtualization / Re: What Needs to be done for ...
Last post by andreas_g - May 28, 2026, 05:47 AMThank you, but I generally do not use AI. It produces only garbage anyway. Previous is free from AI generated code.
Back to my question:
Does anyone know how to implement de-emphasis for CD standard emphasised audio at 44,1 kHz?
Back to my question:
Does anyone know how to implement de-emphasis for CD standard emphasised audio at 44,1 kHz?
#5
Virtualization / Re: What Needs to be done for ...
Last post by ZombiePhysicist - May 27, 2026, 04:46 PMI'm not expert.
I just asked some AI and it suggested what follows which may be garbage:
But maybe
[color=rgba(0, 0, 0, 0.96)]b0 = 0.730944081 b1 = -1.432852847 b2 = 0.701908766 a0 = 1.000000000 a1 = -1.421865094 a2 = 0.711709612[/color]
[color=rgba(0, 0, 0, 0.96)]Implementation notes[/color]
Below is a minimal JUCE 7+ example demonstrating how to apply the de‑emphasis biquad in a standalone or plugin processor using JUCE's dsp::IIR::Filter. It loads the provided biquad coefficients into dsp::IIR::Coefficients and processes audio in processBlock.
Paste into your AudioProcessor's header/cpp or into a minimal JUCE Console/Audio plugin project.
Key points:
AudioProcessor.h (excerpt)
#pragma once
#include <JuceHeader.h>
class DeEmphasisProcessor : public juce::AudioProcessor
{
public:
DeEmphasisProcessor();
~DeEmphasisProcessor() override;
void prepareToPlay (double sampleRate, int samplesPerBlock) override;
void releaseResources() override;
void processBlock (juce::AudioBuffer<float>&, juce::MidiBuffer&) override;
// usual AudioProcessor boilerplate...
private:
juce::dsp::IIR::Filter<float> deEmphasisFilter;
JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (DeEmphasisProcessor)
};
AudioProcessor.h (excerpt)
#pragma once
#include <JuceHeader.h>
class DeEmphasisProcessor : public juce::AudioProcessor
{
public:
DeEmphasisProcessor();
~DeEmphasisProcessor() override;
void prepareToPlay (double sampleRate, int samplesPerBlock) override;
void releaseResources() override;
void processBlock (juce::AudioBuffer<float>&, juce::MidiBuffer&) override;
// usual AudioProcessor boilerplate...
private:
juce::dsp::IIR::Filter<float> deEmphasisFilter;
JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (DeEmphasisProcessor)
};
AudioProcessor.cpp (excerpt)
#include "AudioProcessor.h"
DeEmphasisProcessor::DeEmphasisProcessor()
{
// constructor...
}
DeEmphasisProcessor::~DeEmphasisProcessor() = default;
void DeEmphasisProcessor::prepareToPlay (double sampleRate, int samplesPerBlock)
{
juce::dsp::ProcessSpec spec;
spec.sampleRate = sampleRate;
spec.maximumBlockSize = static_cast<uint32> (samplesPerBlock);
spec.numChannels = static_cast<uint32> (getTotalNumOutputChannels());
deEmphasisFilter.reset();
deEmphasisFilter.prepare(spec);
// Ensure sample rate is 44.1 kHz (coeffs provided for 44100)
if (std::abs(sampleRate - 44100.0) > 1.0)
{
// You can recompute coefficients for other rates; here we assert for clarity.
jassertfalse;
}
// Coefficients (Direct Form I, a0 = 1)
const double b0 = 0.730944081;
const double b1 = -1.432852847;
const double b2 = 0.701908766;
const double a0 = 1.0;
const double a1 = -1.421865094;
const double a2 = 0.711709612;
// JUCE expects coefficients as (b0,b1,b2,a0,a1,a2) for IIR::Coefficients::makeDirectFormI
auto coeffs = juce::dsp::IIR::Coefficients<float>::makeDirectFormI(
(float)b0, (float)b1, (float)b2, (float)a0, (float)a1, (float)a2);
*deEmphasisFilter.state = *coeffs; // load coefficients into filter state
}
void DeEmphasisProcessor::releaseResources()
{
deEmphasisFilter.reset();
}
void DeEmphasisProcessor::processBlock (juce::AudioBuffer<float>& buffer, juce::MidiBuffer&)
{
juce::dsp::AudioBlock<float> block (buffer);
juce::dsp::ProcessContextReplacing<float> context (block);
// Process in-place (same coefficients used for all channels)
deEmphasisFilter.process(context);
}
[color=rgba(0, 0, 0, 0.96)][color=var(--ds-text-primary)]Notes and suggestions[/color][/color]
I just asked some AI and it suggested what follows which may be garbage:
But maybe
- [color=rgba(0, 0, 0, 0.96)]JUCE (C++)
[/color]- [color=rgba(0, 0, 0, 0.96)]Cross‑platform, integrates with Core Audio on macOS. Use dsp::IIR::Coefficients anddsp::IIR::Filter (or dsp::ProcessorChain) for SOS processing; ready for plugin or standalone app.[/color]
- [color=rgba(0, 0, 0, 0.96)]Use two first-order sections (cascade) converted via bilinear transform for τ = 50 µs and 15 µs.Implement as two 1st-order IIRs or combine into one biquad (SOS recommended).[/color]
[color=rgba(0, 0, 0, 0.96)]b0 = 0.730944081 b1 = -1.432852847 b2 = 0.701908766 a0 = 1.000000000 a1 = -1.421865094 a2 = 0.711709612[/color]
[color=rgba(0, 0, 0, 0.96)]Implementation notes[/color]
- [color=rgba(0, 0, 0, 0.96)]Use SOS/biquad routines (JUCE dsp::IIR, vDSP_biquadm, AudioUnit DSP code) to ensure numericstability.[/color]
- [color=rgba(0, 0, 0, 0.96)]Normalize or verify 1 kHz reference gain if needed.[/color]
- [color=rgba(0, 0, 0, 0.96)]Test with sweep and reference emphasis test tones.[/color]
Below is a minimal JUCE 7+ example demonstrating how to apply the de‑emphasis biquad in a standalone or plugin processor using JUCE's dsp::IIR::Filter. It loads the provided biquad coefficients into dsp::IIR::Coefficients and processes audio in processBlock.
Paste into your AudioProcessor's header/cpp or into a minimal JUCE Console/Audio plugin project.
Key points:
- Sample rate must be 44100 Hz (coefficients precomputed for 44.1 kHz).
- The coefficients are Direct Form I (a0 = 1) as b0,b1,b2 ; a0,a1,a2.
AudioProcessor.h (excerpt)
#pragma once
#include <JuceHeader.h>
class DeEmphasisProcessor : public juce::AudioProcessor
{
public:
DeEmphasisProcessor();
~DeEmphasisProcessor() override;
void prepareToPlay (double sampleRate, int samplesPerBlock) override;
void releaseResources() override;
void processBlock (juce::AudioBuffer<float>&, juce::MidiBuffer&) override;
// usual AudioProcessor boilerplate...
private:
juce::dsp::IIR::Filter<float> deEmphasisFilter;
JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (DeEmphasisProcessor)
};
AudioProcessor.h (excerpt)
#pragma once
#include <JuceHeader.h>
class DeEmphasisProcessor : public juce::AudioProcessor
{
public:
DeEmphasisProcessor();
~DeEmphasisProcessor() override;
void prepareToPlay (double sampleRate, int samplesPerBlock) override;
void releaseResources() override;
void processBlock (juce::AudioBuffer<float>&, juce::MidiBuffer&) override;
// usual AudioProcessor boilerplate...
private:
juce::dsp::IIR::Filter<float> deEmphasisFilter;
JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (DeEmphasisProcessor)
};
AudioProcessor.cpp (excerpt)
#include "AudioProcessor.h"
DeEmphasisProcessor::DeEmphasisProcessor()
{
// constructor...
}
DeEmphasisProcessor::~DeEmphasisProcessor() = default;
void DeEmphasisProcessor::prepareToPlay (double sampleRate, int samplesPerBlock)
{
juce::dsp::ProcessSpec spec;
spec.sampleRate = sampleRate;
spec.maximumBlockSize = static_cast<uint32> (samplesPerBlock);
spec.numChannels = static_cast<uint32> (getTotalNumOutputChannels());
deEmphasisFilter.reset();
deEmphasisFilter.prepare(spec);
// Ensure sample rate is 44.1 kHz (coeffs provided for 44100)
if (std::abs(sampleRate - 44100.0) > 1.0)
{
// You can recompute coefficients for other rates; here we assert for clarity.
jassertfalse;
}
// Coefficients (Direct Form I, a0 = 1)
const double b0 = 0.730944081;
const double b1 = -1.432852847;
const double b2 = 0.701908766;
const double a0 = 1.0;
const double a1 = -1.421865094;
const double a2 = 0.711709612;
// JUCE expects coefficients as (b0,b1,b2,a0,a1,a2) for IIR::Coefficients::makeDirectFormI
auto coeffs = juce::dsp::IIR::Coefficients<float>::makeDirectFormI(
(float)b0, (float)b1, (float)b2, (float)a0, (float)a1, (float)a2);
*deEmphasisFilter.state = *coeffs; // load coefficients into filter state
}
void DeEmphasisProcessor::releaseResources()
{
deEmphasisFilter.reset();
}
void DeEmphasisProcessor::processBlock (juce::AudioBuffer<float>& buffer, juce::MidiBuffer&)
{
juce::dsp::AudioBlock<float> block (buffer);
juce::dsp::ProcessContextReplacing<float> context (block);
// Process in-place (same coefficients used for all channels)
deEmphasisFilter.process(context);
}
[color=rgba(0, 0, 0, 0.96)][color=var(--ds-text-primary)]Notes and suggestions[/color][/color]
- [color=rgba(0, 0, 0, 0.96)][color=var(--ds-text-primary)]For different sample rates, recompute coefficients via bilinear transform or design an IIR from theanalog poles/zeros for the new sample rate.[/color][/color]
- [color=rgba(0, 0, 0, 0.96)][color=var(--ds-text-primary)]For better numeric stability use an SOS cascade of first‑order sections (two 1st‑order filters) —implement as two dsp::IIR::Filter instances in series.[/color][/color]
- [color=rgba(0, 0, 0, 0.96)][color=var(--ds-text-primary)]In plugins, ensure processing is real‑time safe and avoid heavy allocations inprepareToPlay/processBlock.[/color][/color]
- [color=rgba(0, 0, 0, 0.96)][color=var(--ds-text-primary)]the two 1st‑order section coefficients and an example chaining two dsp::IIR::Filter instances, or[/color][/color]
- [color=rgba(0, 0, 0, 0.96)][color=var(--ds-text-primary)]code that recomputes coefficients at runtime for arbitrary sample rates. Which would you prefer?[/color][/color]
#6
Virtualization / Re: What Needs to be done for ...
Last post by andreas_g - May 27, 2026, 01:59 PMThank you for the response. So I'll change recording to save the data before applying volume and filter. This change will be available in the next version of Previous.
Are there any audio experts in here? While trying to find out how to improve the low-pass filter I realized that in fact I need a de-emphasis filter for CD-standard emphasised sound. In real hardware the filter (analog implementation) is included in the Philips SAA7323 DAC inside the Soundbox. It is enabled via the DEC pin of the chip.
Does anyone have an idea how to implement de-emphasis for CD standard emphasised audio at 44,1 kHz?
Are there any audio experts in here? While trying to find out how to improve the low-pass filter I realized that in fact I need a de-emphasis filter for CD-standard emphasised sound. In real hardware the filter (analog implementation) is included in the Philips SAA7323 DAC inside the Soundbox. It is enabled via the DEC pin of the chip.
Does anyone have an idea how to implement de-emphasis for CD standard emphasised audio at 44,1 kHz?
#7
Virtualization / Re: What Needs to be done for ...
Last post by Rhetorica - May 26, 2026, 05:16 AMYes, just export 44.1 kHz. If someone truly needs to recover the 22050 Hz audio they can always resample it in an editor.
#8
Software / Re: NetSurf for OPENSTEP
Last post by Rhetorica - May 26, 2026, 05:15 AMGetting another browser working on OPENSTEP has been a holy grail for decades. It's fun writing web pages like we're stuck in 1997 but damn, this is amazing work. 
The vintage Mach world will never be the same!
The vintage Mach world will never be the same! #9
Software / NetSurf for OPENSTEP
Last post by turbolent - May 25, 2026, 09:58 PMI took some time to get an OPENSTEP-compatible frontend for NetSurf working.
You can find the fork at https://github.com/turbolent/netsurf-openstep. Building requires several dependencies, which are available via https://github.com/turbolent/openstep-pkg
You can find the fork at https://github.com/turbolent/netsurf-openstep. Building requires several dependencies, which are available via https://github.com/turbolent/openstep-pkg
#10
Virtualization / Re: What Needs to be done for ...
Last post by andreas_g - May 25, 2026, 03:51 PMThis makes sense for me too. Volume can be adjusted easily during playback to the recorded data itself. Filters can also be applied and the current low-pass filter does not match real hardware anyway.
But one issue remains: The Soundbox accepts 44,1 and 22,05 kHz audio streams. Depending on setup it performs upsampling of 22,05 kHz data using sample repetition or zero-filling. Probably this step should be taken before saving the sound?
But one issue remains: The Soundbox accepts 44,1 and 22,05 kHz audio streams. Depending on setup it performs upsampling of 22,05 kHz data using sample repetition or zero-filling. Probably this step should be taken before saving the sound?