Kotlin Multiplatform audio I/O library. It captures and plays back linear PCM with a precisely configurable sample rate, channel count and sample format, and reads or writes exactly as many frames, samples or bytes as the caller asks for.

The public API only knows about PCM streams and devices; each platform is implemented on top of its own native audio API behind that API.

Audio is always interleaved, little-endian PCM. Raw streams only: container formats (WAV, MP3, ...) and codecs are out of scope.

commonMain.dependencies {
    implementation("cn.enaium.audio:audio-io-kmp:1.0.1")
}

import cn.enaium.audio.*

fun main() {
    val system = audioSystem()
    try {
        println(system.name)                       // "Core Audio", "WASAPI", ...
        println(system.inputDevices())             // every capture endpoint

        val format = AudioFormat(48000, 1, SampleFormat.PCM_S16)
        val input = system.openInput(format, bufferFrames = 960)   // 20 ms
        try {
            input.start()
            val buffer = AudioBuffer(input.format, input.bufferFrames)
            while (true) {
                val frames = input.read(buffer)    // blocks until audio arrives
                if (frames < 0) break              // stopped or closed
                process(buffer.shorts())           // exactly `buffer.sampleCount` samples
            }
        } finally {
            input.close()
        }
    } finally {
        system.close()
    }
}

Playback works the same way through system.openOutput(...) and output.write(buffer).

AudioBuffer never hides how much data is there. Every accessor takes a count and clamps it to the valid region, so a partially filled capture buffer yields fewer samples instead of zeros or an exception:

val frames = input.read(buffer)

val firstBytes = buffer.bytes(64)      // exactly 64 bytes, or fewer at the end
val samples = buffer.shorts(4096)      // exactly 4096 samples, interleaved
val left = buffer.floats(2048)         // converted to -1.0 .. 1.0

buffer.bytes()                          // the whole valid region
buffer.toShorts(destination)            // conversion without allocating

audioSystem() returns the backend of the current platform. Several systems can coexist, and the caller owns and closes each one.

open -> OPEN, start -> STARTED, stop -> STOPPED, close -> CLOSED

read and write block. stop() and close() from another thread release the blocked call, which then returns -1, so a capture loop ends without needing an extra flag. readNonBlocking/writeNonBlocking never block and report 0 when there is no room or no data.

AudioRingBuffer is the piece to put between the capture thread and a processing or playback thread:

val ring = AudioRingBuffer(format, capacityFrames = 48000)   // one second
// capture thread
ring.write(buffer)
// processing thread
val frames = ring.read(destination, 960)

It is not thread safe by itself: exactly one thread may write and one may read, which is what makes it allocation free.

Every Mixer the JDK reports becomes a device. Capture uses TargetDataLine, playback SourceDataLine, both in blocking mode. JavaSound does not convert between formats: a mixer that does not advertise the requested format fails with an AudioException naming the mixer and the format.

AudioRecord/AudioTrack take the requested format when the device supports it (PCM_U8, PCM_S16, PCM_F32, and PCM_S32 from API 31 on), otherwise the call fails with an AudioException naming the format. The application has to hold RECORD_AUDIO itself; a missing permission surfaces as an AudioException that says so. Device enumeration needs an application Context:

val system = audioSystem(context)   // full AudioManager enumeration
val system = audioSystem()          // default input/output only

AAudio is available from Android 8.0 (API 26). The backend loads libaaudio.so at runtime, so a binary that links this library still starts on older devices and reports the missing API instead. AAudio has no device enumeration: one default endpoint per direction is reported, and a numeric platform device id can be passed through AudioDevice.id to pin a stream to a specific device.

Every shared object that links this library is built for 16 KB memory pages (-Wl,-z,max-page-size=16384 -Wl,-z,common-page-size=16384), which Android 15 and newer require on arm64 devices.

Streams run in shared mode. The requested format is tried first; when the audio engine cannot take it, the client is opened again with AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM so the engine performs the sample rate and format conversion. Devices come from IMMDeviceEnumerator and the AudioDevice.id is the endpoint id string.

libasound.so.2 is loaded at runtime, so nothing has to be installed to link against the library; the runtime package that every Linux installation has is enough. Devices come from ALSA's name hints, so default, sysdefault and hw:CARD,DEV all work as AudioDevice.id. The format is applied exactly as requested: a device that cannot deliver the sample rate, channel count or sample format fails instead of being silently converted. Overruns and suspends are recovered in place, which keeps a long capture running across a machine sleep.

Streams use Audio Queue Services, which runs its own thread and converts between the requested format and the hardware format, so any sample rate and sample format this library can describe works. AudioDevice.id is the Core Audio device UID, which is what pins a queue to a specific device.

audio-io-kmp/                       the library
  src/commonMain                    public API, PCM conversions, ring buffer
  src/jvmMain                       javax.sound.sampled backend
  src/androidMain                   AudioRecord / AudioTrack backend
  src/nativeMain                    shared native code: frame ring, signals, streams
  src/posixNativeMain               mutex/condition signal for macOS and Linux
  src/macosMain                     Core Audio backend
  src/linuxMain                     ALSA backend
  src/mingwMain                     WASAPI backend
  src/androidNativeMain             AAudio backend
  src/nativeInterop/cinterop        cinterop definitions (ALSA, WASAPI, AAudio)
examples/audio-visualizer/          Dear ImGui + ImPlot spectrum visualizer

The native targets compile src/nativeMain plus the directory of their OS family, and each family provides the platform specific pieces such as the backend itself and its wake-up signal.

examples/audio-visualizer captures from the default input and draws a waveform, a live spectrum and a waterfall spectrogram with Dear ImGui and ImPlot, with device/format selection and level meters:

./gradlew :examples:audio-visualizer:jvmRun            # JVM desktop
./gradlew :examples:audio-visualizer:runDebugExecutableMacosArm64

./gradlew :examples:audio-visualizer:android:assembleRelease   # Android APK
adb install -r examples/audio-visualizer/android/build/outputs/apk/release/android-release.apk

Use the release APK on a device: a debug build links an unoptimized Kotlin/Native binary (46 MB against 12 MB for arm64), which renders a frame in the hundreds of milliseconds. The release build type is signed with the debug key, so it installs without a keystore.

The Android application module hosts the SDL activity and packages the libmain.so that the example module links for every ABI (arm64-v8a, armeabi-v7a, x86_64, x86). The activity is locked to landscape, runs immersive, and asks for RECORD_AUDIO before the native side opens the AAudio capture stream; it also forwards the display density so the interface is scaled for the screen instead of using desktop sized text.

The interface fills the whole window: the main ImGui window is re-sized to the ImGui viewport every frame and drawn without decoration, so the layout follows the SDL window on every resize.

The draw data is handed to imgui-kmp's SDL renderer backend, which copies only the vertices each draw command references.

The waterfall is drawn as a 128 x 64 texture rather than as an ImPlot heatmap. A heatmap cell costs four vertices in the render backend, which for a spectrogram of a reasonable size dominates the frame time; the same data as a texture is two triangles, and the values are turned into colors through a lookup table sampled from the selected ImPlot colormap. That is what keeps the example at display refresh rate.

Pass --frames N (JVM) or set AUDIO_VISUALIZER_FRAMES=N (native) to exit after a fixed number of frames, which is what the CI runs use; the run prints the frame rate it reached. Without a capture device the window still opens and shows the error instead of failing.

./gradlew :audio-io-kmp:jvmTest                # core + JavaSound backend tests
./gradlew :audio-io-kmp:macosArm64Test         # the same tests on the native target
./gradlew :audio-io-kmp:compileKotlinLinuxX64  # cross compile a target klib

Every native target except the Apple ones is cross-compiled from any host: the WASAPI bindings use the MinGW headers that Kotlin/Native ships, and AAudio and ALSA are resolved with dlopen, so no Android NDK or ALSA development package is needed to produce the klibs.

The Android target needs an Android SDK (local.properties with sdk.dir, or ANDROID_HOME).

.github/workflows/publish.yml publishes to Maven Central, and .github/workflows/test.yml builds and tests every platform. Both are started manually from the Actions tab.

MIT