Convertisseur Audio — Outil en ligne gratuit

Free Audio Converter — WAV, WebM, OGG in Your Browser

Outilolis Audio Converter converts audio files between formats directly in your browser with no server uploads and no software installation. Upload any audio file your browser can decode — MP3, WAV, OGG, WebM, AAC, FLAC, and more — then choose your target format, sample rate, and channel configuration. The tool decodes the source file, resamples as needed, and encodes the output entirely on your device.

This is ideal for quick format conversions, preparing audio for web projects, reducing file sizes for email or messaging, or adjusting sample rates and channel counts for specific applications. Because everything runs locally, there are no upload limits, no wait times, and complete privacy for your audio files.

Supported Output Formats

• WAV (PCM) — uncompressed audio with RIFF headers. Maximum quality, largest file size. Compatible with every audio application and device. The tool manually constructs the WAV file by writing RIFF/WAVE headers and interleaved 16-bit PCM samples from the decoded AudioBuffer.
• WebM/Opus — a modern compressed format offering excellent quality at small file sizes. Opus is the best general-purpose audio codec available, outperforming MP3 and AAC at equivalent bitrates. Supported in Chrome, Firefox, Edge, and recent Safari versions.
• OGG/Vorbis — an open-source lossy format with quality comparable to MP3 at lower bitrates. Supported in Chrome, Firefox, and Edge. Nont natively supported in Safari. Uses the MediaRecorder API for encoding.

Understanding Sample Rates

The sample rate determines how many audio samples are captured per second. Higher sample rates preserve more high-frequency content but produce larger files. Here is a practical guide:

• 8,000 Hz — telephone quality. Only preserves frequencies up to 4 kHz. Sounds muffled but intelligible for speech. Very small files.
• 16,000 Hz — wideband voice quality. Preserves frequencies up to 8 kHz. Good for speech recognition, podcasts, and voice memos.
• 22,050 Hz — AM radio quality. Decent for voice and simple music. Half the data of CD quality.
• 44,100 Hz — CD quality. The standard for music distribution. Preserves the full audible spectrum (up to 22 kHz).
• 48,000 Hz — DVD and professional video standard. Slightly higher than CD quality. Used in film, video production, and professional audio.

Mono vs Stereo

Stereo audio uses two channels (left and right), creating a sense of spatial width. Mono uses a single channel. Mono files are exactly half the size of stereo files at the same sample rate and bit depth. For voice-only content like podcasts, lectures, and phone recordings, mono is perfectly adequate and saves significant storage space. For music, keep stereo to preserve the spatial characteristics of the mix.

When converting stereo to mono, the tool averages the left and right channels. When converting mono to stereo, the single channel is duplicated to both left and right outputs.

Why Non MP3 Output?

MP3 encoding requires the LAME encoder or similar licensed library, which cannot run natively in the browser. Adding a JavaScript-based MP3 encoder (like lamejs) would add 200-300 KB to the page and introduce external dependencies. This tool deliberately uses only native browser APIs for reliability and minimal download size. WAV provides lossless quality, and WebM/Opus provides better compression than MP3 at lower file sizes, making MP3 largely unnecessary.

How the Conversion Works Internally

The conversion process has three steps. First, the source file is decoded into a raw AudioBuffer using AudioContext.decodeAudioData() — this works regardless of the input format. Second, the AudioBuffer is resampled to the target sample rate and channel count using an Hors ligneAudioContext. Third, the resampled buffer is encoded into the target format: for WAV, the tool manually writes RIFF headers and PCM data; for WebM and OGG, it uses the MediaRecorder API with the appropriate MIME type.

Common Conversion Scenarios

• MP3 to WAV — for importing into audio editing software that prefers uncompressed input.
• WAV to WebM — for reducing file size when publishing audio on the web.
• Stereo to Mono — for voice recordings where stereo is unnecessary.
• 48kHz to 44.1kHz — for preparing audio for CD burning or specific applications.
• High sample rate to 16kHz — for speech recognition systems that expect 16kHz input.