Audio Converter

🎵

Upload Audio File

Drag and drop or click to select an audio file (MP3, WAV, M4A, AAC, FLAC, OGG, OPUS, WMA, AIFF, AMR)

Upload an audio file (drag & drop is supported; MP3, WAV, M4A, FLAC, OGG, WMA, AMR and more).
Preview the audio and select the segment you want to convert.
Choose the output format (MP3, M4A, WAV, FLAC or OGG) and bitrate.
Click "Convert" and wait for the conversion to complete.
Download the converted audio file.

Which output format should you choose?

When converting audio, the tool can output MP3, M4A, WAV, FLAC and OGG. They differ mainly in compatibility, compression and use cases - here is how to pick.

MP3General sharing, car audio, older devices

Lossy compression with the widest compatibility; plays on almost any device.

M4A (AAC)iPhone, iTunes, better quality at smaller sizes

Better quality than MP3 at the same bitrate; used by Apple and YouTube.

WAVAudio editing, DAW projects, further processing

Uncompressed lossless PCM; large files, universally accepted by editors.

FLACMusic archiving, high-fidelity listening

Lossless compression at roughly half the WAV size; no quality loss.

OGGOpen-source projects, games, web audio

Open lossy format with a good size/quality balance, but limited Safari support.

Want a full breakdown of audio codecs? Read the Audio & Video Format Guide. →

Help

FAQ

Is my audio file uploaded to any server?Show

No. All conversion happens locally in your browser using WebAssembly. Your audio file never leaves your device.

Which audio formats can I convert between?Show

Output formats: MP3, M4A (AAC), WAV, FLAC and OGG. Input formats include MP3, WAV, M4A, AAC, FLAC, OGG, OPUS, WMA, AIFF and AMR.

Can I convert only a portion of the audio?Show

Yes! You can set custom start and end times to convert only the segment you need.

Will converting to a higher bitrate improve the quality?Show

No. If the source file is already lossy (e.g. 128 kbps MP3), converting to 320 kbps will not restore lost quality - it only inflates the file size. Choose a bitrate equal to or slightly below the source.

What is the difference between WAV and FLAC?Show

Both are lossless. WAV stores raw PCM audio (large files), while FLAC compresses losslessly to roughly half the size with no quality loss. Use WAV for editing/DAW workflows, FLAC for archiving.

Why can't I preview my audio file in the browser?Show

Some formats (WMA, AMR, AIFF) cannot be decoded by the browser's audio element. The tool still reads their duration with FFmpeg and converts them normally - only the in-page preview is unavailable.

Why is my AAC output a .m4a file?Show

Raw AAC streams (ADTS, .aac files) carry no duration metadata, so many players mis-estimate the length or the sample rate. The tool therefore wraps AAC in the standard M4A container, which fixes this. The audio data itself is identical.

My converted OGG won't play on iPhone - why?Show

Safari has historically limited OGG Vorbis support (iOS added partial support only in recent versions). If your target audience uses Apple devices, choose MP3 or M4A instead - OGG is best for open-source projects, games, and web audio on Chrome/Firefox/Android.

Quick Start

Upload an audio file (drag & drop is supported; MP3, WAV, M4A, FLAC, OGG, WMA, AMR and more).
Preview the audio and select the segment you want to convert.
Choose the output format (MP3, M4A, WAV, FLAC or OGG) and bitrate.
Click "Convert" and wait for the conversion to complete.
Download the converted audio file.

Privacy

Full guide

Audio Converter

Convert audio files directly in your browser. Trim clips, adjust bitrate, and switch between the most common audio formats.

Features

Wide Input Support: MP3, WAV, M4A, AAC, FLAC, OGG, OPUS, WMA, AIFF, AMR
Multiple Outputs: Convert to MP3, M4A (AAC), WAV, FLAC or OGG
Trim Support: Select custom start and end times to convert only what you need
Bitrate Control: Choose from 64 to 320 kbps for lossy formats
100% Local: All processing happens in your browser using WebAssembly

Supported Formats

Input: MP3, WAV, M4A, AAC, FLAC, OGG, OPUS, WMA, AIFF, AMR

Output: MP3, M4A (AAC), WAV, FLAC, OGG

How to Choose a Bitrate?

Bitrate (kbps) is the amount of audio data per second, and it directly determines both quality and file size. The math is simple: file size per minute ≈ bitrate ÷ 8 × 60 seconds.

Bitrate	Size per Minute	Best For
64 kbps	~0.5 MB	Voice recordings, low-quality needs
128 kbps	~1 MB	Podcasts, speech, casual listening
192 kbps	~1.4 MB	Standard choice for music distribution
320 kbps	~2.4 MB	High-quality music, near-lossless listening

A common misconception: converting to a bitrate higher than the source file will not improve quality. If your source is a 128 kbps MP3, outputting 320 kbps just inflates the file for no gain. For lossy sources, choose a bitrate equal to or slightly below the original; use WAV or FLAC only when you need lossless output for editing or archiving. To learn more about formats and codecs, read the Audio & Video Format Guide.

Supported conversion directions

Every pairwise conversion between the five output formats is supported, plus a wide set of legacy input formats. Pick your direction below - each one opens the same tool with your source format in mind:

From MP3: MP3 to WAV, MP3 to FLAC, MP3 to M4A, MP3 to OGG

From WAV: WAV to MP3, WAV to FLAC, WAV to M4A, WAV to OGG

From FLAC: FLAC to MP3, FLAC to WAV, FLAC to M4A, FLAC to OGG

From M4A: M4A to MP3, M4A to WAV, M4A to FLAC, M4A to OGG

From OGG: OGG to MP3, OGG to WAV, OGG to FLAC, OGG to M4A

From legacy formats: OPUS to MP3 / WAV / FLAC, WMA to MP3 / WAV, AMR to MP3 (voice memos), AIFF to WAV / FLAC, AAC to MP3 / FLAC

Which of these conversions make sense? A quick guide by category:

Conversion type	Quality impact	Example
Lossless → lossless (FLAC ↔ WAV)	None - bit-perfect round trips	Archiving, DAW import
Lossless → lossy (FLAC → MP3)	Controlled by your bitrate choice	Everyday listening, car audio
Lossy → lossless (MP3 → WAV/FLAC)	No quality gain - just a bigger lossless copy of already-lossy audio	Only for editing workflows that require PCM
Lossy → lossy (MP3 → OGG)	Compounds quality loss - each re-encode discards more	Compat needs only; avoid re-encoding repeatedly

The key insight from the last two rows: converting a lossy file to WAV or FLAC does not "restore" anything - the information was already discarded when the MP3 was made. The output is lossless in format but identical in fidelity to its lossy source, at 5-10x the file size. Do it when an editor demands WAV; never do it hoping to recover quality.

What lossy compression actually discards

MP3, AAC, and Vorbis all exploit the same fact about human hearing: you cannot perceive sounds that psychoacoustic masking hides. When a loud sound and a quiet sound occur at the same time in the same frequency band, the quiet one is effectively inaudible - so the encoder simply throws it away.

Lossy encoders do this on thousands of frequency bands per frame, discarding:

Frequencies above the hearing threshold (~16-20 kHz, lowered further at low bitrates)
Sounds masked by louder neighbors (in time and in frequency)
Stereo detail your ears cannot localize

At 128 kbps the discards are usually inaudible for casual listening; at 64 kbps they become obvious (cymbals turn hissy, reverb smears). FLAC and WAV keep every bit, which is why a CD-quality WAV is ~10 MB/min while a 128 kbps MP3 of the same minute is ~1 MB - a 10:1 ratio paid for entirely with masked information.

This is also why the loss depends on how many times you re-encode: each pass applies masking again to already-masked audio, and the artifacts compound. For the deeper theory, read Lossless vs Lossy: What Audio Compression Really Does.

Sample rate and bit depth

Two numbers describe every PCM audio stream (WAV/FLAC internals):

Sample rate (Hz) - how many amplitude measurements per second. CD audio is 44.1 kHz; voice memos are often 8-16 kHz; hi-res audio is 96-192 kHz.
Bit depth (bits) - the precision of each measurement. CD is 16-bit; studio work is 24-bit.

A practical rule: 44.1 kHz / 16 bit covers the full range of human hearing. Converting to a higher sample rate cannot add detail that was never recorded, exactly like upscaled video. This tool preserves the source stream's sample rate when re-encoding; only the codec and bitrate change.

A real-world walkthrough: voice memo to MP3

Phone voice memos often arrive as AMR (3-8 kbps, optimized for speech, plays nowhere except phones). To share one via an app that expects standard audio:

Upload the .amr file - the browser can't preview it, but the tool reads duration via FFmpeg and converts it fine.
Pick MP3 at 64-128 kbps. The source is 8 kbps speech; anything higher just wastes space. Mono speech needs nowhere near 192 kbps.
Trim the portion you need (voice memos tend to have long dead air).
Convert and download - the result plays in WeChat, email previews, and every player.

The same flow works for OPUS voice messages (WhatsApp/Telegram exports) and old WMA files.