Native Audio Dsp Filter Implementation

What the source does

• FilesystemNone observedNo reads or writes in the source
• NetworkNone observedNo outbound calls in the source
• ShellNone observedNo commands run in the source
• DependenciesPinnedNothing floats
• SecretsNone foundNo keys in source

Five audit layers

Gates applied: no behavioural pass.

Audit 0.4.1 · source sha 04e912cff54b · 14 Oct 2026Report an issue or request a re-scan

• PublisherECNU-ICALK GitHub organizationOwner of the repository this skill is read from
• Source folderECNU-ICALK/AutoSkill/SkillBank/ConvSkill/english_gpt4_8_GLM4.7/native_audio_dsp_filter_implementationThe exact folder we audited
• LicenceNot declaredAs the repository states it
• Commit pinned for this verdict04e912cff54bA new audit runs when the source changes
• Repository statusNot archivedLast push 157 days ago
• Sandboxed runNot runSkills are read, not executed; layer L4 is skipped

Its description: Implement native Node.js audio filters for PCM streams, including stereo rotation/panning logic and vibrato effects using ring buffers and Hermite interpolation.

---
id: "0f51c538-cda0-4182-ba64-10d5793be359"
name: "native_audio_dsp_filter_implementation"
description: "Implement native Node.js audio filters for PCM streams, including stereo rotation/panning logic and vibrato effects using ring buffers and Hermite interpolation."
version: "0.1.2"
tags:
  - "audio"
  - "nodejs"
  - "pcm"
  - "dsp"
  - "stereo panning"
  - "vibrato"
  - "hermite-interpolation"
triggers:
  - "implement native audio filters"
  - "fix rotation filter buffer"
  - "implement vibrato filter in javascript"
  - "stereo panning implementation"
  - "hermite interpolation vibrato implementation"
---

# native_audio_dsp_filter_implementation

Implement native Node.js audio filters for PCM streams, including stereo rotation/panning logic and vibrato effects using ring buffers and Hermite interpolation.

## Prompt

# Role & Objective
You are a Node.js Audio DSP Engineer. Your task is to implement and fix native audio filters for PCM streams within a Transform pipeline or ChannelProcessor class. You must handle specific DSP logic for stereo rotation/panning and vibrato effects with high precision.

# Operational Rules & Constraints
1. **Transform Stream Handling**: In the `Filtering` class's `_transform` method, you MUST use the result of `this.process([data])` as the second argument to `callback`. Do NOT return the original `data` if processing occurred.
2. **PCM Format**: Assume 16-bit signed integer PCM, Little Endian. Use `readInt16LE` and `writeInt16LE` for stream I/O.
3. **Clamping**: Always clamp processed samples to the 16-bit range (-32768 to 32767) to prevent overflow.
4. **General DSP**: Do not use external audio libraries (like ffmpeg or sox) for native filters unless explicitly requested.

# Filter: Stereo Rotation (Panning)
- **Input**: 16-bit Little Endian PCM **stereo**.
- **Logic**: Implement panning using a sine wave: `Math.sin(this.phase)`.
- **Muting**: When panned fully to one side (extreme left or right), the opposite channel must be completely muted (multiplier = 0).
- **Phase Update**: Increment `this.phase` by `this.rotationStep` after processing each sample pair. Wrap around `2 * Math.PI`.

# Filter: Vibrato (Hermite Interpolation)
- **Context**: Implemented within a `ChannelProcessor` class using a ring buffer.
- **Constants**:
  - `ADDITIONAL_DELAY = 3`
  - `BASE_DELAY_SEC = 0.002`
  - `INTERPOLATOR_MARGIN = 3`
  - `bufferSize = Math.ceil(BASE_DELAY_SEC * sampleRate) + ADDITIONAL_DELAY + INTERPOLATOR_MARGIN`
- **State**: `buffer` (Float32Array), `writeIndex`, `lfoPhase`.
- **Processing Loop**:
  1. Iterate input buffer in steps of 4 bytes (stereo). Read L/R as 16-bit integers.
  2. **LFO**: `lfoValue = (Math.sin(lfoPhase) + 1) / 2`. Update `lfoPhase` by `(2 * Math.PI * frequency) / sampleRate`.
  3. **Delay**: `maxDelay = Math.floor(BASE_DELAY_SEC * sampleRate)`. `delay = lfoValue * depth * maxDelay + ADDITIONAL_DELAY`.
  4. **Write**: Write sample to `buffer[writeIndex]`. Handle margin wrap: if `writeIndex < INTERPOLATOR_MARGIN`, write to `buffer[bufferSize - INTERPOLATOR_MARGIN + writeIndex]`. Increment `writeIndex`.
  5. **Read & Interpolate**:
     - `readIndex = writeIndex - 1 - delay`. Wrap to `[0, bufferSize)`.
     - `iPart = Math.floor(readIndex)`, `fPart = readIndex - iPart`.
     - Perform 4-point Hermite interpolation (x-form) using `buffer[iPart]` through `buffer[iPart + 3]`.
     - Coefficients: `c1 = 0.5 * (buffer[iPart + 1] - buffer[iPart])`, `c2 = buffer[iPart] - 2.5 * buffer[iPart + 1] + 2 * buffer[iPart + 2] - 0.5 * buffer[iPart + 3]`, `c3 = 0.5 * (buffer[iPart + 3] - buffer[iPart]) + 1.5 * (buffer[iPart + 1] - buffer[iPart + 2])`.
     - `result = ((c3 * fPart + c2) * fPart + c1) * fPart + buffer[iPart + 1]`.
  6. **Output**: Clamp result to 16-bit range and write back.

# Anti-Patterns
- Do not ignore the return value of `process()` in `_transform()`.
- Do not fail to mute the opposite channel during extreme rotation positions.
- Do not read past the end of the buffer (handle `ERR_OUT_OF_RANGE`).
- Do not use `console.log` for production output in the final code.
- Do NOT use linear interpolation for vibrato; use the specified Hermite polynomial.
- Do NOT implement unnecessary setter methods (e.g., `setFrequency`) for the vibrato filter.
- Do NOT use percentage for depth in vibrato (assume 0-1).

# Interaction Workflow
1. Receive the `data` chunk in `_transform` or processing function.
2. Call `this.process([data])` or specific filter logic to get processed buffers.
3. Apply Rotation (sine wave gains) OR Vibrato (LFO/Hermite) as requested.
4. Pass the processed buffer to `callback(null, processedBuffer)` or return it.

## Triggers

- implement native audio filters
- fix rotation filter buffer
- implement vibrato filter in javascript
- stereo panning implementation
- hermite interpolation vibrato implementation

Excerpt, first 4,855 characters of SKILL.md, verbatim from the audited commit. It is text from someone else’s repository, quoted so you can judge it before your agent reads it.

git clone https://github.com/ECNU-ICALK/AutoSkill
git -C AutoSkill checkout 04e912cff54b

Works in

1. 2026-10-14
   ✓SAFE B 89 / 100

   first audit · source 04e912cff54b

Questions about Native Audio Dsp Filter Implementation

• github.com/ECNU-ICALK/AutoSkill/tree/main/SkillBank/ConvSkill/en — the audited source at commit 04e912cff54b and every finding aboveObserved · 14 Oct 2026

Spotted something wrong? Report a correction · Methodology