Kempo is a high-performance, pure Kotlin Multiplatform (KMP) audio time-stretching and pitch-shifting library powered by an advanced phase-locked phase vocoder.

Engineered from the ground up for professional digital signal processing (DSP), sub-millisecond streaming latency, and pristine acoustic transparency, Kempo operates natively across JVM, Android, iOS, and WebAssembly (Wasm/JS) without C/C++ native toolchains, dynamic libraries, or JNI runtime overhead.

• Key Highlights
• Architecture & DSP Pipeline
• Module Structure
• Supported Platforms
• Installation
• Getting Started
  • 1. One-Liner Track Processing (AudioTrack)
  • 2. Low-Latency Real-Time Streaming
  • 3. Exact Offline Buffer Rendering
  • 4. Interleaved PCM-16 Serialization
• Deep Dive & Parameter Tuning
  • Pitch & Formant Decoupling
  • Tonality Limits & Custom Frequency Maps
  • Latency & Memory Mechanics (Zero-GC Streaming)
  • Multichannel & 5.1 Surround Pipelines
• Benchmarks
• Credits & Acknowledgements
• License

• ⚡ Zero-Allocation Audio Streaming: The real-time block-by-block streaming loop operates with 0 B heap allocations, making it safe for high-priority real-time audio threads and glitch-free rendering loops.
• 🔒 Vertical Harmonic Phase-Locking: Eliminates the transient smearing, phasing, and "underwater" hollow artifacts typical of classic phase vocoders by locking phase trajectories around prominent spectral peaks.
• 🎧 Multichannel Phase Coherence: Maintains exact Inter-channel Phase Differences (IPD) across stereo, 5.1, and multichannel streams, completely preventing spatial collapse or stereo image drift.
• 🗣️ Independent Formant Scaling: Decouples pitch shifting from vocal tract resonance, allowing natural vocal transposition without the "chipmunk" or "monster" effect, or intentional timbre modification.
• 🚀 Hand-Optimized DSP Foundation:
  • FastFft: In-place mixed-radix FFT engine supporting radices 2, 3, 4, and composite factorization.
  • ModifiedRealFFT: Real-to-complex transform evaluated on half-bin shifted frequency points (k + 0.5) / N to eliminate DC/Nyquist edge singularities.
  • Windows: Parametric Kaiser windows with Weighted Overlap-Add (WOLA) normalization for perfect reconstruction.
• 🔁 Power-of-Two Ring Buffering (MultiChannelBuffer): Circular multichannel ring buffer backed by a single flat array, accelerated with unrolled vector accumulators and split-copy boundary handling.

┌─────────────────────────────────────────────────────────────────────────┐
│                             Audio Ingestion                             │
│       AudioTrack (32-bit Float) / Raw Multi-Channel Float Arrays        │
│                                   │                                     │
│      PcmConverter.fromPcm16 ◄─────┴─────► PcmConverter.toPcm16          │
└───────────────────────────────────┬─────────────────────────────────────┘
                                    │ Feed Blocks
                                    ▼
┌─────────────────────────────────────────────────────────────────────────┐
│                        Dynamic STFT Analysis                            │
│   MultiChannelBuffer (Input Ring) ──► WindowedFFT (WOLA Kaiser Window)  │
│                                                    │                    │
│                                                    ▼                    │
│                                             ModifiedRealFFT             │
│                                      [Half-bin grid: (k + 0.5) / N]     │
└───────────────────────────────────┬─────────────────────────────────────┘
                                    │ Spectral Frames
                                    ▼
┌─────────────────────────────────────────────────────────────────────────┐
│                   Phase-Locked Vocoder Processing Core                  │
│   • Spectral Peak Detection      ──► Slew-Smoothed Energy Contours      │
│   • Vertical Phase-Locking       ──► Harmonic Alignment Across Bins     │
│   • Horizontal Phase Progression ──► Instantaneous Phase Accumulator    │
│   • Inter-Channel Coherence      ──► Locked Inter-channel Phase (IPD)   │
│   • Formant Filtering            ──► Cepstral-like Spectral Envelope    │
└───────────────────────────────────┬─────────────────────────────────────┘
                                    │ Modified Spectrum
                                    ▼
┌─────────────────────────────────────────────────────────────────────────┐
│                       Resynthesis & Overlap-Add                         │
│   Inverse ModifiedRealFFT ──► WOLA Synthesis Window Normalization       │
│                                                    │                    │
│                                                    ▼                    │
│                                      MultiChannelBuffer (Output Ring)   │
└─────────────────────────────────────────────────────────────────────────┘

1. Analysis Stage: Audio frames are pushed into MultiChannelBuffer. Centered circular rotations align windowed frames before computing half-bin shifted real spectra via ModifiedRealFFT.
2. Phase Transformation: Peaks in the spectral envelope are identified via logarithmic smoothing. Phase updates are constrained across harmonic regions (vertical locking) and across channels (stereo coherence).
3. Synthesis & Reconstruction: Inverse transforms are normalized through dual-pass Kaiser WOLA windows and summed into the output accumulation ring.

The codebase is partitioned into targeted modules separating mathematical DSP foundations from interactive UI tooling:

Kempo targets all major Kotlin Multiplatform platforms without external C/C++ runtime requirements:

• JVM: Java 11+, Server, Desktop (macOS, Linux, Windows)
• Android: API 24+ (ARM64, x86_64)
• Apple Native: iOS (iosX64, iosArm64, iosSimulatorArm64)
• WebAssembly: Wasm/JS (wasmJs) for browser and Node.js runtimes

Add the dependency to your module's build.gradle.kts:

kotlin {
    sourceSets {
        commonMain.dependencies {
            implementation("io.github.numq:kempo:1.0.0")
        }
    }
}

Or for single-target JVM / Android projects:

dependencies {
    implementation("io.github.numq:kempo:1.0.0")
}

For batch audio workflows or high-level offline rendering:

import io.github.numq.kempo.AudioTrack

fun processAudio(inputTrack: AudioTrack): AudioTrack {
    // 25% faster (1.25x), transposed up by +3 semitones, preserving natural vocal timbre
    return inputTrack.stretch(
        timeRatio = 1.25f,
        pitchSemitones = 3.0f,
        formantSemitones = 0.0f
    )
}

For real-time audio threads (e.g., audio callbacks, synthesizers, DAW engines):

import io.github.numq.kempo.KempoStretch
import kotlin.math.max

class RealtimeStreamEngine(channels: Int = 2, sampleRate: Float = 44100f) {
    private val stretch = KempoStretch().apply {
        presetDefault(channels, sampleRate)
        setTransposeSemitones(semitones = 2.0f) // Transpose +2 semitones
        setFormantSemitones(semitones = 0.0f)   // Keep natural formants
    }

    private var timeRatio = 1.25f // 1.25x speed
    val outBlockSize: Int = stretch.defaultInterval
    val inBlockSize: Int get() = max(1, (outBlockSize / timeRatio).toInt())

    // Called inside the real-time audio callback: ZERO heap allocations
    fun processBlock(inputChannels: Array<FloatArray>, outputChannels: Array<FloatArray>) {
        stretch.process(
            inputs = inputChannels,
            inputSamples = inBlockSize,
            outputs = outputChannels,
            outputSamples = outBlockSize
        )
    }
}

For deterministic whole-file rendering with automated pre-roll warming, latency compensation, and tail overlap-add flushing:

import io.github.numq.kempo.KempoStretch
import kotlin.math.roundToInt

fun renderTrack(
    rawInput: Array<FloatArray>,
    sampleRate: Float,
    timeRatio: Float = 1.0f,
    pitchSemitones: Float = -3.0f
): Array<FloatArray> {
    val channels = rawInput.size
    val totalInSamples = rawInput[0].size
    val totalOutSamples = (totalInSamples / timeRatio).roundToInt().coerceAtLeast(1)
    val rawOutput = Array(channels) { FloatArray(totalOutSamples) }

    val stretch = KempoStretch()
    stretch.presetDefault(channels, sampleRate)
    stretch.setTransposeSemitones(pitchSemitones)

    val success = stretch.exact(
        inputs = rawInput,
        inputSamples = totalInSamples,
        outputs = rawOutput,
        outputSamples = totalOutSamples
    )

    return rawOutput
}

Convert raw platform audio bytes (e.g., from WAV files or audio recorders) into normalized AudioTrack representations:

import io.github.numq.kempo.PcmConverter

// 1. Decode raw interleaved PCM bytes into [-1.0f, 1.0f] float buffers
val track = PcmConverter.fromPcm16(
    bytes = wavByteArray,
    channels = 2,
    sampleRate = 44100f,
    isLittleEndian = true
)

// 2. Perform transformation
val processedTrack = track.stretch(timeRatio = 1.0f, pitchSemitones = 4.0f)

// 3. Serialize back to 16-bit PCM bytes
val outputBytes = PcmConverter.toPcm16(processedTrack, isLittleEndian = true)

Classical resampling couples pitch and duration ($f_{\text{new}} = f_{\text{old}} \cdot r$). Kempo allows adjusting duration, musical pitch, and spectral timbre completely independently:

• setTransposeSemitones(semitones): Transposes the perceived fundamental frequency and harmonics without modifying playback duration.
• setFormantSemitones(semitones, compensatePitch): Modifies the spectral envelope peak positions. Passing compensatePitch = true automatically shifts the formant envelope inversely to pitch transposition, preserving original vocal identities during large pitch shifts.

For experimental transformations, speech conditioning, or microtonal scales:

• tonalityLimit: Frequencies above this threshold transition from harmonic multiplicative transposition to linear shifting, preserving high-frequency noise and transient clarity.
• setFreqMap(fn: (Float) -> Float): Injects a custom transfer function mapping normalized frequency bins [0.0, 0.5] to arbitrary target frequencies.

• Input/Output Latencies: Retrieve algorithmic delays using stretch.inputLatency() and stretch.outputLatency() to maintain sample-accurate sync across mixer tracks.
• Pre-Allocated Workspaces: All scratch buffers, twiddle factors, and spectral mapping indices are allocated upfront during configure() / presetDefault(). Calling process() inside real-time loops performs zero memory allocation.

Kempo is channel-layout agnostic and accepts arbitrary channel counts (Mono, Stereo, 5.1, 7.1) via flat channel buffers:

• Channel Independence & LFE: Channels are processed in parallel with dynamic inter-channel phase difference (IPD) locking around spectral peaks. Low-frequency effects (LFE) channels can be passed directly without pre-filtering.
• Downmixing Order Contract:
  • Headphone/Stereo Output (Recommended: Downmix First): For rendering 5.1 content over stereo headphones, downmixing prior to stretching (5.1 -> Stereo Downmix -> Kempo) is strongly recommended. It reduces CPU workload by ~3x (processing 2 FFTs instead of 6) and preserves battery on mobile devices while folding dialogue linearly.
  • Discrete Passthrough (Stretch First): If driving discrete multi-speaker surround setups or binaural HRTF spatializers requiring separate 5.1 feeds, stretching all 6 channels simultaneously preserves phase alignment across front, surround, and center channels without comb-filtering.

Measured using JMH 1.37 on OpenJDK 17 (HotSpot 64-Bit Server VM, 17.0.14) with 3 warmup iterations and 5 measurement iterations:

Kempo's core DSP architecture, modified real FFT, and phase-locking vocoder algorithms are directly based on the research and reference implementation of signalsmith-stretch by Geraint Luff (Signalsmith Audio).

Special thanks to Geraint Luff for designing and publishing the mathematical foundation of this phase vocoder.

Apache-2.0 License - see LICENSE file for details.

Support Development: numq.github.io/support