Audio Converter

Knowledge Base

How to Use the Audio Converter

The Audio Converter allows you to transform audio files between multiple formats, MP3, WAV, FLAC, AAC, M4A, OGG, Opus, and AIF, all within your browser. Upload your file by clicking the drop zone or dragging and dropping, then select your desired output settings before converting.

Why Convert Audio?

Different formats serve different purposes. WAV files are ideal for professional audio work because they store uncompressed, lossless audio. MP3 and AAC are lightweight and compatible with almost all devices, making them perfect for distribution. FLAC offers lossless compression for archiving. The sample rate, bit depth, and channel configuration you choose will affect file size and audio quality.

Choosing the Right Settings

• Sample Rate: 48 kHz is the industry standard for video work. 44.1 kHz is CD quality. Higher rates don't improve perceived quality but do increase file size.
• Bit Depth: 24-bit is professional standard. 16-bit is sufficient for most work and reduces file size.
• Channels: Stereo is standard. Mono is used for voice-only content and podcasts to reduce file size by 50%.

Supported Formats

Upload in: MP3, WAV, FLAC, AAC, M4A, OGG, Opus, AIF, AIFF, AIFC, WebM. Download as: WAV with your chosen sample rate, bit depth, and channel configuration. All conversion happens locally in your browser, your files never leave your device.

FAQ

Q: Will converting to WAV improve my audio quality?
A: No. Converting from a lossy format (like MP3) won't recover lost data. WAV is useful for preserving quality during editing.

Q: What sample rate should I use?
A: Use 48 kHz for video or streaming, 44.1 kHz for audio-only work. Matching your DAW's session rate prevents quality loss.

Q: How much will my file size change?
A: The tool shows an estimate before conversion. Stereo 24-bit 48 kHz WAV is roughly 17 MB per minute of audio.

Choosing sample rate, bit depth and format

Sample rate sets how many times per second the waveform was measured, and it caps the highest frequency the file can hold at half that figure. 44,100 samples per second covers everything human hearing reaches and is the right choice for music and podcasts. 48,000 is the standard for video work, and matching your video project avoids a resampling step later. Higher rates such as 96,000 are useful while recording and processing, and offer nothing on a finished file anyone will listen to.

Bit depth sets how finely each measurement is stored, which determines the dynamic range available. 16 bit covers about 96 decibels, more than enough for delivery and the standard for CD and most distribution. 24 bit covers about 144 decibels and is worth keeping while you edit, because it leaves room to raise quiet material without lifting noise with it. Converting up from 16 bit to 24 adds nothing back, since the detail was never captured.

Convert to WAV when you need a file with no further quality loss, for editing, archiving or handing to someone else's workflow. Every conversion between compressed formats loses a little more, because each one re-encodes what the previous one already approximated. Going from MP3 to AAC to MP3 again audibly degrades the result, so keep an uncompressed master and export compressed copies from it rather than converting the copies.

Mono or stereo is a decision about content rather than quality. A single speaker recorded on one microphone is mono, and saving it as stereo doubles the file size while storing the same thing twice. Music, room ambience and anything with deliberate left and right placement needs stereo. Most spoken word does not.

Vocabulary

Sample rate. How many times per second the audio waveform was measured, in hertz. It caps the highest frequency the file can store at half its value, so 44,100 Hz reaches about 22,000 Hz, comfortably past the limit of human hearing.

Bit depth. How many bits store each individual measurement. More bits mean a finer scale and a wider range between the quietest and loudest sounds a file can hold. 16 bit suits delivery, 24 bit gives editing headroom.

Lossy and lossless. Lossy formats such as MP3 and AAC discard detail to save space, and the discarded part never returns. Lossless formats such as WAV and FLAC reproduce the original exactly, at the cost of a much larger file.

Resampling. Converting a file from one sample rate to another by calculating new measurements between the existing ones. Good resampling is close to inaudible, but each pass costs a little quality, so avoid doing it repeatedly.