Trim, resample, mono/stereo, gain, fade — convert any audio to 16-bit WAV right in your browser.
Drop an audio file here or click to browse
Supports WAV, MP3, OGG, M4A, FLAC, WebM Opus — any format your browser can decode.
Audio is decoded and rendered entirely in your browser. Your files never leave the page.
Normalize audio loudness to Spotify/YouTube/Podcast/Broadcast targets via peak or LUFS — entirely in-browser.
Detect the tempo (BPM) of any song or beat by analyzing peak intervals — runs entirely in your browser.
Convert any text to speech using your browser's built-in voices. 50+ languages, runs offline once voices are installed.
The Audio Converter & Trimmer is a fully in-browser tool for the most common audio prep tasks — cutting a clip down to a specific time range, resampling to a target rate (8 kHz for telephony, 16 kHz for speech recognition, 44.1 kHz for CD quality, 48 kHz for video), converting between mono and stereo, applying volume gain in decibels, and adding fade-in/out curves. All processing happens in your browser via the Web Audio API: decoding leverages the codecs your browser already ships (so WAV, MP3, OGG Vorbis, M4A/AAC, FLAC, and WebM Opus all "just work" with no JavaScript decoder added to the bundle), and re-encoding produces standard 16-bit PCM WAV files that every DAW, video editor, and downstream pipeline can read. Because nothing leaves your machine, you can safely process voice memos, confidential interviews, unreleased music, or client recordings without uploading them anywhere. The tool is ideal for shrinking podcast clips for transcription services that have file-size limits, normalizing the format of an audio dataset before machine learning, or just trimming silence from the start of a voice note.
16-bit PCM WAV at a chosen sample rate is the lingua franca of audio: every DAW (Pro Tools, Logic, Reaper, Audacity), every speech-recognition API (Whisper, Google Cloud STT, Azure), and every video editor accepts it directly. It is uncompressed, so no information is lost from the trim or resample step, and there is no encoding-induced quality loss to debug later. For long files where size matters more than quality, you can compress the WAV output with ffmpeg (or use this project's Video Toolkit which has ffmpeg.wasm built in).