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Our free online audio converter supports a wide range of audio formats, including MP3, WAV, OGG, AAC, FLAC, WMA, AIFF, ALAC, Opus, and WebM. All conversions are performed directly in your browser, ensuring your files remain private and secure.
Whether you need to convert audio for a presentation, edit a podcast, or prepare music for a video project, our tool provides fast and reliable conversions with high-quality output.
Every digital audio file is a compromise between file size and sound quality. The way a format handles that compromise determines whether it is lossy or lossless, and understanding that distinction is the single most important thing you can learn about audio conversion.
Lossy formats like MP3, AAC, and OGG Vorbis shrink files by permanently removing audio data that psychoacoustic models predict most listeners will not notice. These models are remarkably clever. They know, for example, that a loud cymbal crash masks a quiet guitar note happening at the same instant, so the encoder discards the quiet note. They know that frequencies above roughly 16–18 kHz are inaudible to most adults, so those are trimmed first.
The result is a file that can be 5–12 times smaller than the original with little or no audible difference at higher bitrates. At lower bitrates, though, the encoder has to cut deeper. You may notice cymbals start to sound “watery,” sustained piano notes develop a shimmering artifact, or stereo imaging gets flattened. These artifacts are the price of very small files.
The key word is permanent. Once data is discarded during lossy encoding, it cannot be recovered. Converting a 128 kbps MP3 to FLAC does not restore the lost information — it just wraps what remains in a larger container.
Lossless formats like FLAC, ALAC, and APE compress audio without discarding any data at all. Think of it like a ZIP file for audio: the original waveform is perfectly reconstructed on playback. File sizes are typically 50–70% of the equivalent WAV, which is a meaningful saving but still much larger than lossy formats.
WAV and AIFF are technically lossless too, but they use no compression — they store the raw PCM waveform byte-for-byte. That means a three-minute CD-quality track takes about 30 MB as WAV, roughly 15–20 MB as FLAC, and 3–5 MB as a high-quality MP3.
In controlled double-blind tests, most listeners cannot reliably distinguish a well-encoded 256 kbps AAC or 320 kbps MP3 from the lossless original. Trained audio engineers using studio monitors can sometimes detect differences, especially in complex material like orchestral recordings or dense electronic mixes. For everyday listening through earbuds or car speakers, a good lossy encode at 192 kbps or higher is effectively transparent for the vast majority of people.
That said, there is a real argument for archiving in lossless: you can always create lossy copies from a lossless master later, but you cannot go the other direction. Keeping lossless originals future-proofs your collection against better codecs and hardware that may appear down the road.
| Format | Type | Typical Bitrate | Best For | Device Support |
|---|---|---|---|---|
| MP3 | Lossy | 128 – 320 kbps | General music playback, maximum compatibility | Universal — every device, browser, and operating system |
| AAC | Lossy | 96 – 256 kbps | Streaming, Apple ecosystem, podcasts | Excellent — iOS, Android, all modern browsers, most hardware players |
| OGG Vorbis | Lossy | 96 – 320 kbps | Game audio, Spotify streaming, open-source projects | Good — Android, Linux, most desktop apps; limited on older iOS/hardware |
| Opus | Lossy | 32 – 256 kbps | Voice calls, low-latency streaming, VoIP | Growing — all modern browsers, Android; limited on older hardware |
| WMA | Lossy | 128 – 320 kbps | Legacy Windows applications | Limited — Windows-centric; poor support on Mac, Linux, and mobile |
| FLAC | Lossless | 800 – 1,400 kbps | Music archiving, audiophile playback, mastering | Very good — Android, Windows, Linux, most modern players; Apple added support in 2021 |
| ALAC | Lossless | 800 – 1,400 kbps | Lossless playback on Apple devices | Apple ecosystem — iTunes, iPhone, iPad, Mac; limited elsewhere |
| WAV | Uncompressed | 1,411 kbps (CD quality) | Studio recording, audio editing, professional workflows | Universal — supported everywhere, but files are very large |
Practical guidance on choosing a format:
Bitrate is the number of bits of audio data processed per second of playback, measured in kilobits per second (kbps). A 320 kbps MP3 uses 320,000 bits to represent each second of audio. A higher bitrate gives the encoder more “budget” to describe the sound accurately, which generally means better quality — but also a larger file.
To put it concretely: one minute of audio at 128 kbps is about 0.96 MB, while the same minute at 320 kbps is about 2.4 MB. An uncompressed CD-quality WAV of the same minute is roughly 10.1 MB. So a 320 kbps MP3 is roughly a quarter of the size of the raw audio while retaining virtually all of the perceptible detail.
Human speech has a relatively narrow frequency range and low dynamic complexity. You can get perfectly clear spoken word at surprisingly low bitrates.
Streaming through earbuds, phone speakers, or Bluetooth headphones. Most detail beyond this level is masked by the playback hardware.
Good headphones, quiet room, focused listening. This is where lossy compression starts to matter if you have trained ears.
Long-term storage of music collections, original recordings, or production masters. Use lossless here — storage is cheap, lost audio data is gone forever.
Audio engineers use the term “transparency” to describe the bitrate at which a lossy encode becomes indistinguishable from the lossless source in a proper ABX blind test. This threshold varies by codec:
Below these thresholds, some listeners may notice compression artifacts. Above them, any perceivable differences are extraordinarily difficult to detect, even under ideal conditions. If you are unsure, encoding at or slightly above the transparency threshold gives you the best balance of size and quality.
Not sure which format or bitrate to choose? Here are the most common real-world situations and what works best for each.
Older car head units with USB or aux inputs almost universally support MP3 and very little else. Burn or copy MP3 at 320 kbps — it will play on essentially any hardware manufactured in the last 20 years, and at 320 kbps the quality is indistinguishable from CD over a car sound system.
Rip to FLAC. It preserves every bit of the original CD audio while compressing to about half the size. If you later want MP3s for your phone, you can convert from the FLAC archive without ever touching the CDs again. One rip, endless future flexibility.
For spoken-word content, AAC at 96–128 kbps or Opus at 48–64 kbps gives excellent clarity at very small file sizes. If your hosting platform or audience needs maximum compatibility, use MP3 at 128 kbps mono — it is the podcast industry standard and keeps episode files around 1 MB per minute.
MP3 remains the undisputed king of compatibility. Every phone, computer, car stereo, smart speaker, web browser, and media player supports it. AAC is a close second with slightly better quality at the same bitrate, but a handful of older devices still struggle with it.
Work in WAV (24-bit, 48 kHz or higher) throughout your production and mixing process. WAV is the lingua franca of every major DAW (Ableton, Logic, Pro Tools, FL Studio). When you are ready to distribute, export a lossless FLAC master and generate lossy formats from that. Never re-encode from a lossy source.
Most email services cap attachments at 20–25 MB. Convert to MP3 at 192 kbps for music or Opus at 64 kbps for voice memos. A five-minute MP3 at 192 kbps is about 7 MB — well within attachment limits and perfectly listenable.
It depends on the direction of the conversion. Converting from a lossless format (like WAV or FLAC) to a lossy format (like MP3 or AAC) always discards some audio data — that is what “lossy” means. However, at reasonable bitrates (192 kbps and above for music), the quality loss is inaudible to most listeners. Converting between lossless formats (for example, WAV to FLAC) preserves quality perfectly. The scenario to avoid is converting from one lossy format to another (e.g., MP3 to AAC), because each re-encoding pass discards additional data, compounding quality loss. If you must do a lossy-to-lossy conversion, use the highest bitrate practical to minimize further degradation.
No. This is one of the most common misconceptions about audio. Converting a 128 kbps MP3 to FLAC gives you a FLAC file that is bit-for-bit identical to the MP3 when decoded — it contains exactly the same audio data, just in a different container. The lost frequencies and detail are gone permanently. The file will actually be larger than the MP3 with no quality benefit. Think of it this way: photocopying a blurry photo onto expensive glossy paper does not make the image sharper.
At the same bitrate, modern codecs like Opus and AAC generally outperform MP3 — they can achieve the same perceived quality with a smaller file. At transparency-level bitrates (256–320 kbps for MP3, 192–256 kbps for AAC), the differences between lossy formats become vanishingly small. Among lossless formats, there is literally no quality difference: FLAC, ALAC, and WAV all reproduce the same waveform identically. The “best” format depends on your needs: best compatibility is MP3, best quality-per-bit is Opus, best for Apple devices is AAC or ALAC, and best for archiving is FLAC.
Sample rate (measured in Hz or kHz) describes how many snapshots of the audio waveform are captured per second. CD quality is 44,100 samples per second (44.1 kHz), which can accurately represent frequencies up to about 22 kHz — beyond normal human hearing. Bitrate (measured in kbps) describes how much data is used per second of playback after encoding. A higher sample rate captures a wider frequency range; a higher bitrate allows the encoder to describe the captured audio more accurately. For most listening, CD-standard 44.1 kHz is more than sufficient. Higher sample rates (96 kHz, 192 kHz) are primarily useful in professional recording and editing where additional headroom for processing is valuable.
Our converter runs entirely in your browser using WebAssembly (WASM), a technology that lets compiled C/C++ code execute at near-native speed inside a web page. When you upload a file, it never leaves your computer. The audio decoding and re-encoding happen locally using a WASM build of FFmpeg, the same battle-tested library that powers professional media tools. This means your files stay private, conversion is fast, and there is no server queue to wait in. The trade-off is that very large files or complex conversions may be limited by your device’s available memory and processing power.
This usually happens when you convert from a lossy format to a lossless one (e.g., MP3 to WAV), or when you convert to a lossy format at a higher bitrate than the source. A 128 kbps MP3 converted to WAV balloons in size because WAV stores uncompressed PCM data — it does not know or care that the source was already compressed. Similarly, re-encoding a 128 kbps MP3 as a 320 kbps MP3 creates a larger file without improving quality, because the encoder pads the existing (already degraded) audio data into a bigger container. To keep files small, convert in one direction: from high-quality sources to lower-bitrate lossy targets.