MyMediaTools

Convert FLAC to MP3 — Free Online Tool

Convert FLAC lossless audio files to MP3 using the LAME encoder, reducing file size significantly while producing perceptually acceptable quality audio compatible with virtually every device and platform. This tool runs entirely in your browser via FFmpeg.wasm — your audio files never leave your computer.

FFmpeg Command

Copy this command to run the same conversion locally with FFmpeg on your desktop. Download FFmpeg

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Estimated output:

Conversion Complete!

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How It Works

FLAC stores audio as a losslessly compressed PCM signal, meaning the decoded audio is bit-for-bit identical to the original recording. Converting to MP3 is a lossy transcoding process: the browser decodes the FLAC stream back to raw PCM audio, then re-encodes it using the LAME MP3 encoder (libmp3lame) with psychoacoustic compression. LAME discards audio information the human ear is least likely to perceive — such as frequencies masked by louder simultaneous sounds — to achieve the target bitrate. At the default 128k bitrate, file sizes typically shrink by 85–90% compared to the original FLAC. Because this is a decode-then-re-encode operation (not a copy), some audio quality is permanently lost; you cannot recover the original FLAC quality from the resulting MP3.

What Each Flag Does

Flag What it does
ffmpeg Invokes the FFmpeg tool, which handles the full pipeline of demuxing the FLAC container, decoding the lossless FLAC audio stream to raw PCM, re-encoding it with LAME, and writing the output MP3 file.
-i input.flac Specifies the input FLAC file. FFmpeg reads the FLAC container, extracts the compressed lossless audio stream, and decodes it to uncompressed PCM audio in memory as the source material for the MP3 encoder.
-c:a libmp3lame Selects the LAME MP3 encoder for the audio stream. libmp3lame is the gold-standard open-source MP3 encoder and produces the best quality MP3 output available in FFmpeg, using psychoacoustic modeling to discard inaudible audio data from the decoded FLAC source.
-b:a 128k Sets the MP3 output to a constant bitrate of 128 kilobits per second. This is a common default that balances file size and perceived audio quality for general listening; increasing this value (e.g., to 320k) preserves more audio detail at the cost of a larger output file.
output.mp3 Defines the output filename and signals to FFmpeg that the output container should be MP3. The .mp3 extension causes FFmpeg to use the standard MP3 muxer, wrapping the LAME-encoded audio stream with ID3 tag support for metadata like artist and album.

Common Use Cases

  • Uploading music to platforms like SoundCloud, YouTube, or social media that don't require or benefit from lossless quality
  • Transferring a large FLAC music library to a phone or MP3 player with limited storage, where lossless files would fill the device too quickly
  • Preparing audio files for a podcast or web player that streams over the internet, where MP3's wide browser and app compatibility is essential
  • Sending a song or demo recording to a collaborator or client who may not have software capable of playing FLAC files
  • Creating a smaller, portable copy of archival FLAC recordings for everyday listening without carrying the full lossless library
  • Converting FLAC rips of CDs or vinyl records into MP3s for use in car stereos or older hardware players that only support MP3

Frequently Asked Questions

Yes — MP3 is a lossy format, so some audio information is permanently discarded during encoding. At 128k (the default), most listeners find the quality acceptable for casual listening, but audiophiles or people with high-quality playback equipment may notice subtle differences in high-frequency detail and spatial clarity compared to the original FLAC. For critical listening or archival use, always keep the original FLAC and only distribute the MP3 copy. If quality is a priority, increase the bitrate to 192k, 256k, or 320k using the quality selector.
Substantially smaller. A FLAC file typically ranges from 20–35 MB per minute of audio depending on the source material, while a 128k MP3 runs around 1 MB per minute. That means a 40 MB FLAC track might become a 3–4 MB MP3 — roughly a 90% size reduction. Higher MP3 bitrates like 320k will result in larger files (about 2.4 MB per minute) but still far smaller than the original FLAC.
It depends on the converter. FFmpeg does carry over standard metadata during this conversion — artist, album, title, track number, and similar tags stored in the FLAC's Vorbis comment block are mapped to ID3 tags in the output MP3. However, FLAC-specific metadata like ReplayGain values and embedded cue sheets are not transferred, as MP3 does not natively support those features. Embedded cover art is typically preserved.
Yes. This tool produces a new MP3 file as output and never modifies or deletes your original FLAC file. Since the conversion runs entirely in your browser using WebAssembly, your original file stays on your computer untouched. It is good practice to keep the original FLAC as your archival master copy, since the quality reduction from FLAC to MP3 is irreversible.
Replace the value after -b:a in the command. For example, to encode at 320k (the highest standard MP3 quality), use: ffmpeg -i input.flac -c:a libmp3lame -b:a 320k output.mp3. Common options are 64k, 96k, 128k, 160k, 192k, 224k, 256k, and 320k. Higher bitrates produce better audio quality at the cost of larger file sizes. For music you plan to distribute, 192k or 320k is generally recommended; 128k is a reasonable minimum for speech or casual listening.
Yes, using a simple shell loop on your desktop. On Linux or macOS, run: for f in *.flac; do ffmpeg -i "$f" -c:a libmp3lame -b:a 128k "${f%.flac}.mp3"; done. On Windows Command Prompt: for %f in (*.flac) do ffmpeg -i "%f" -c:a libmp3lame -b:a 128k "%~nf.mp3". This browser tool processes one file at a time, so the FFmpeg command approach is especially practical when you have dozens or hundreds of FLAC files to convert, or files larger than 1 GB.

Technical Notes

FLAC uses a linear predictive coding (LPC) algorithm to compress PCM audio losslessly, while MP3 relies on a Modified Discrete Cosine Transform (MDCT) and psychoacoustic modeling via the LAME encoder to achieve lossy compression. Because FLAC is lossless, it is always preferable to convert to MP3 from FLAC rather than from another lossy format like AAC or OGG Vorbis — doing so avoids 'generation loss', where artifacts from multiple rounds of lossy encoding compound. The libmp3lame encoder used here is the highest-quality open-source MP3 encoder available and is the industry standard for FFmpeg-based MP3 output. MP3 does not support features present in FLAC such as cue sheets, ReplayGain tags, or per-sample seeking with the same precision. The output MP3 will use CBR (Constant Bit Rate) encoding at the specified bitrate; if you prefer VBR (Variable Bit Rate) for better quality-per-byte efficiency at the cost of less predictable file sizes, you can modify the command to use -q:a instead of -b:a (e.g., -q:a 2 for high-quality VBR). Sample rates and channel layouts (stereo, mono) are preserved by default unless explicitly changed in the command.

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