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A bunch of new audio nodes and other related changes #75

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merged 5 commits into from
Jan 5, 2025
Merged

A bunch of new audio nodes and other related changes #75

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6ba0e44
A lot of audio stuff
Nytra Jan 5, 2025
66881e2
Add AudioSourceInfo
Nytra Jan 5, 2025
182be57
Fix channel counts
Nytra Jan 5, 2025
e03b2a5
Remove using
Nytra Jan 5, 2025
3363a34
Fix build error
Nytra Jan 5, 2025
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40 changes: 6 additions & 34 deletions ProjectObsidian/Components/Audio/BandPassFilter.cs
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Original file line number Diff line number Diff line change
@@ -1,13 +1,11 @@
using System;
using FrooxEngine;
using Elements.Assets;
using System.Net;
using ProtoFlux.Runtimes.Execution;
using System.Collections.Generic;
using Obsidian.Elements;

namespace Obsidian.Components.Audio
{
[Category(new string[] { "Obsidian/Audio" })]
[Category(new string[] { "Obsidian/Audio/Filters" })]
public class BandPassFilter : Component, IAudioSource, IWorldElement
{
[Range(0.1f, 1.41f, "0.00")]
Expand All @@ -21,10 +19,7 @@ public class BandPassFilter : Component, IAudioSource, IWorldElement

public readonly SyncRef<IAudioSource> Source;

private double lastTime;

private Dictionary<Type, object> lowFilters = new();
private Dictionary<Type, object> highFilters = new();
private BandPassFilterController _controller = new();

public bool IsActive
{
Expand All @@ -38,8 +33,7 @@ public void Read<S>(Span<S> buffer) where S : unmanaged, IAudioSample<S>
if (!IsActive)
{
buffer.Fill(default(S));
lowFilters.Clear();
highFilters.Clear();
_controller.Clear();
return;
}

Expand All @@ -48,27 +42,7 @@ public void Read<S>(Span<S> buffer) where S : unmanaged, IAudioSample<S>

Source.Target.Read(tempBuffer);

if (!lowFilters.TryGetValue(typeof(S), out object lowFilter))
{
lowFilter = new ButterworthFilter.FilterButterworth<S>();
lowFilters.Add(typeof(S), lowFilter);
}
if (!highFilters.TryGetValue(typeof(S), out object highFilter))
{
highFilter = new ButterworthFilter.FilterButterworth<S>();
highFilters.Add(typeof(S), highFilter);
}

((ButterworthFilter.FilterButterworth<S>)lowFilter).UpdateCoefficients(HighFrequency, Engine.AudioSystem.SampleRate, ButterworthFilter.FilterButterworth<S>.PassType.Lowpass, Resonance);
((ButterworthFilter.FilterButterworth<S>)highFilter).UpdateCoefficients(LowFrequency, Engine.AudioSystem.SampleRate, ButterworthFilter.FilterButterworth<S>.PassType.Highpass, Resonance);

for (int i = 0; i < tempBuffer.Length; i++)
{
((ButterworthFilter.FilterButterworth<S>)lowFilter).Update(ref tempBuffer[i]);
((ButterworthFilter.FilterButterworth<S>)highFilter).Update(ref tempBuffer[i]);
}

lastTime = Engine.Current.AudioSystem.DSPTime;
_controller.Process(tempBuffer, LowFrequency, HighFrequency, Resonance);
}

protected override void OnAwake()
Expand All @@ -77,16 +51,14 @@ protected override void OnAwake()
Resonance.Value = 1.41f;
LowFrequency.Value = 20f;
HighFrequency.Value = 20000f;
lastTime = Engine.Current.AudioSystem.DSPTime;
}

protected override void OnChanges()
{
base.OnChanges();
if (Source.GetWasChangedAndClear())
{
lowFilters.Clear();
highFilters.Clear();
_controller.Clear();
}
}
}
Expand Down
108 changes: 6 additions & 102 deletions ProjectObsidian/Components/Audio/ButterworthFilter.cs
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Original file line number Diff line number Diff line change
@@ -1,11 +1,11 @@
using System;
using FrooxEngine;
using Elements.Assets;
using System.Collections.Generic;
using Obsidian.Elements;

namespace Obsidian.Components.Audio;

[Category(new string[] { "Obsidian/Audio" })]
[Category(new string[] { "Obsidian/Audio/Filters" })]
public class ButterworthFilter : Component, IAudioSource, IWorldElement
{
[Range(20f, 20000f, "0.00")]
Expand All @@ -18,9 +18,7 @@ public class ButterworthFilter : Component, IAudioSource, IWorldElement

public readonly SyncRef<IAudioSource> Source;

private double lastTime;

private Dictionary<Type, object> filters = new();
private ButterworthFilterController _controller = new();

public bool IsActive
{
Expand All @@ -33,7 +31,6 @@ public bool IsActive
protected override void OnAwake()
{
base.OnAwake();
lastTime = Engine.Current.AudioSystem.DSPTime;
Frequency.Value = 20f;
Resonance.Value = 1.41f;
}
Expand All @@ -43,7 +40,7 @@ protected override void OnChanges()
base.OnChanges();
if (Source.GetWasChangedAndClear())
{
filters.Clear();
_controller.Clear();
}
}

Expand All @@ -54,7 +51,7 @@ public void Read<S>(Span<S> buffer) where S : unmanaged, IAudioSample<S>
if (!IsActive)
{
buffer.Fill(default(S));
filters.Clear();
_controller.Clear();
return;
}

Expand All @@ -64,99 +61,6 @@ public void Read<S>(Span<S> buffer) where S : unmanaged, IAudioSample<S>

Source.Target.Read(span);

if (!filters.TryGetValue(typeof(S), out object filter)) {
filter = new FilterButterworth<S>();
filters.Add(typeof(S), filter);
}

((FilterButterworth<S>)filter).UpdateCoefficients(Frequency, Engine.AudioSystem.SampleRate, LowPass ? FilterButterworth<S>.PassType.Lowpass : FilterButterworth<S>.PassType.Highpass, Resonance);

for (int i = 0; i < span.Length; i++)
{
((FilterButterworth<S>)filter).Update(ref span[i]);
}
}

public class FilterButterworth<S> where S: unmanaged, IAudioSample<S>
{
/// <summary>
/// rez amount, from sqrt(2) to ~ 0.1
/// </summary>
private readonly float resonance;

private readonly float frequency;
private readonly int sampleRate;
private readonly PassType passType;

private float c, a1, a2, a3, b1, b2;

/// <summary>
/// Array of input values, latest are in front
/// </summary>
private S[] inputHistory = new S[2];

/// <summary>
/// Array of output values, latest are in front
/// </summary>
private S[] outputHistory = new S[3];

public void UpdateCoefficients(float frequency, int sampleRate, PassType passType, float resonance)
{
switch (passType)
{
case PassType.Lowpass:
c = 1.0f / (float)Math.Tan(Math.PI * frequency / sampleRate);
a1 = 1.0f / (1.0f + resonance * c + c * c);
a2 = 2f * a1;
a3 = a1;
b1 = 2.0f * (1.0f - c * c) * a1;
b2 = (1.0f - resonance * c + c * c) * a1;
break;
case PassType.Highpass:
c = (float)Math.Tan(Math.PI * frequency / sampleRate);
a1 = 1.0f / (1.0f + resonance * c + c * c);
a2 = -2f * a1;
a3 = a1;
b1 = 2.0f * (c * c - 1.0f) * a1;
b2 = (1.0f - resonance * c + c * c) * a1;
break;
}
}

public enum PassType
{
Highpass,
Lowpass,
}

public void Update(ref S newInput)
{
S first = newInput.Multiply(a1);
S second = this.inputHistory[0].Multiply(a2);
S third = this.inputHistory[1].Multiply(a3);
S fourth = this.outputHistory[0].Multiply(b1);
S fifth = this.outputHistory[1].Multiply(b2);
S final = first.Add(second).Add(third).Subtract(fourth).Subtract(fifth);

for (int i = 0; i < final.ChannelCount; i++)
{
if (final[i] > 1f) final = final.SetChannel(i, 1f);
else if (final[i] < -1f) final = final.SetChannel(i, -1f);
}

this.inputHistory[1] = this.inputHistory[0];
this.inputHistory[0] = newInput;

this.outputHistory[2] = this.outputHistory[1];
this.outputHistory[1] = this.outputHistory[0];
this.outputHistory[0] = final;

newInput = final;
}

public S Value
{
get { return this.outputHistory[0]; }
}
_controller.Process(span, LowPass, Frequency, Resonance);
}
}
25 changes: 3 additions & 22 deletions ProjectObsidian/Components/Audio/EMA_IIR_SmoothSignal.cs
View file
Open in desktop
Original file line number Diff line number Diff line change
@@ -1,10 +1,11 @@
using System;
using FrooxEngine;
using Elements.Assets;
using Obsidian.Elements;

namespace Obsidian.Components.Audio;

[Category(new string[] { "Obsidian/Audio" })]
[Category(new string[] { "Obsidian/Audio/Effects" })]
public class EMA_IIR_SmoothSignal : Component, IAudioSource, IWorldElement
{
[Range(0f, 1f, "0.00")]
Expand Down Expand Up @@ -36,26 +37,6 @@ public void Read<S>(Span<S> buffer) where S : unmanaged, IAudioSample<S>

Source.Target.Read(span);

EMAIIRSmoothSignal(ref span, span.Length, SmoothingFactor);
}

// smoothingFactor is between 0.0 (no smoothing) and 0.9999.. (almost smoothing to DC) - *kind* of the inverse of cutoff frequency
public void EMAIIRSmoothSignal<S>(ref Span<S> input, int N, float smoothingFactor = 0.8f) where S : unmanaged, IAudioSample<S>
{
// forward EMA IIR
S acc = input[0];
for (int i = 0; i < N; ++i)
{
acc = input[i].LerpTo(acc, smoothingFactor);
input[i] = acc;
}

// backward EMA IIR - required only if we need to preserve the phase (aka make the filter symetric) - we usually want this
acc = input[N - 1];
for (int i = N - 1; i >= 0; --i)
{
acc = input[i].LerpTo(acc, smoothingFactor);
input[i] = acc;
}
Algorithms.EMAIIRSmoothSignal(ref span, span.Length, SmoothingFactor);
}
}
64 changes: 5 additions & 59 deletions ProjectObsidian/Components/Audio/PhaseModulator.cs
View file
Open in desktop
Original file line number Diff line number Diff line change
@@ -1,11 +1,12 @@
using System;
using FrooxEngine;
using Elements.Assets;
using System.Threading;
using Obsidian.Elements;
using Elements.Core;

namespace Obsidian.Components.Audio
{
[Category(new string[] { "Obsidian/Audio" })]
[Category(new string[] { "Obsidian/Audio/Effects" })]
public class PhaseModulator : Component, IAudioSource, IWorldElement
{
[Range(0f, 5f, "0.00")]
Expand All @@ -29,7 +30,7 @@ public int ChannelCount
{
get
{
return CarrierSource.Target?.ChannelCount ?? 0;
return MathX.Min(CarrierSource.Target?.ChannelCount ?? 0, ModulatorSource.Target?.ChannelCount ?? 0);
}
}

Expand All @@ -39,43 +40,6 @@ protected override void OnAwake()
ModulationIndex.Value = 1f; // Default modulation index
}

/// <summary>
/// Calculates instantaneous phase of a signal using a simple Hilbert transform approximation
/// </summary>
private double[] CalculateInstantaneousPhase<S>(Span<S> buffer) where S : unmanaged, IAudioSample<S>
{
int length = buffer.Length;
double[] phase = new double[length];
double[] avgAmplitudes = new double[length];

for (int i = 1; i < length - 1; i++)
{
for (int j = 0; j < buffer[i].ChannelCount; j++)
{
avgAmplitudes[i] += buffer[i][j];
}
avgAmplitudes[i] /= buffer[i].ChannelCount;
}

// Simple 3-point derivative for phase approximation
for (int i = 1; i < length - 1; i++)
{
double derivative = (avgAmplitudes[i + 1] - avgAmplitudes[i - 1]) / 2.0;
double hilbertApprox = avgAmplitudes[i] / Math.Sqrt(avgAmplitudes[i] * avgAmplitudes[i] + derivative * derivative);
phase[i] = Math.Acos(hilbertApprox);

// Correct phase quadrant based on derivative sign
if (derivative < 0)
phase[i] = 2 * Math.PI - phase[i];
}

// Handle edge cases
phase[0] = phase[1];
phase[length - 1] = phase[length - 2];

return phase;
}

// TODO: Make this not click when the signal goes silent and then not silent?
public void Read<S>(Span<S> buffer) where S : unmanaged, IAudioSample<S>
{
Expand All @@ -100,25 +64,7 @@ public void Read<S>(Span<S> buffer) where S : unmanaged, IAudioSample<S>

float modulationIndex = ModulationIndex.Value;

double[] carrierPhase = CalculateInstantaneousPhase(carrierBuffer);

// Apply phase modulation
for (int i = 0; i < buffer.Length; i++)
{
for (int j = 0; j < buffer[i].ChannelCount; j++)
{
double modifiedPhase = carrierPhase[i] + (modulationIndex * modulatorBuffer[i][j]);

// Calculate amplitude using original carrier amplitude
float amplitude = carrierBuffer[i][j];

// Generate output sample
buffer[i] = buffer[i].SetChannel(j, amplitude * (float)Math.Sin(modifiedPhase));

if (buffer[i][j] > 1f) buffer[i] = buffer[i].SetChannel(j, 1f);
if (buffer[i][j] < -1f) buffer[i] = buffer[i].SetChannel(j, -1f);
}
}
Algorithms.PhaseModulation(buffer, carrierBuffer, modulatorBuffer, modulationIndex, channelCount);
}
}
}
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