Audio Converter

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.