Audio Converter for MP3, WAV, FLAC, and More

Selected Audio Files0 selected

Convert Audio for the App That Needs It

Convert audio between MP3, WAV, FLAC, M4A, OGG and the other formats in the selector. Choose the target format first, then adjust the compatible codec, sample rate, channels and bitrate where available. Multiple outputs can be downloaded separately or in a ZIP. Converting MP3 to WAV does not restore detail already lost in MP3 compression; it changes the format for the receiving app.

Cost: Free to use. File size and usage limits apply.

Example: prepare an MP3 for a WAV-only app

Input: MP3 recording
Setting: Output WAV · choose the required sample rate
Output: WAV audio file

The conversion changes the audio format. It cannot restore detail removed when the MP3 was originally compressed.

How to convert audio files

Add your audio
Click Choose audio or drop files onto the preview. MP3, WAV, FLAC, M4A, AAC, OGG, Opus and audio-only WebM and MP4 files are accepted, up to 200 MB each, and you can add several at once.
Check the selection
Selected Audio Files lists everything that will be converted. Compatible audio already in Files, from earlier uploads or other tools, is selected too when the page opens. Open Files and click a file to add or remove it.
Pick the output format first
Click a tile under Output format. Switching tiles resets Audio codec, Bitrate and Sample rate to that format's defaults, so choose the format first.
Adjust quality if needed
The defaults suit most files. Change codec, bitrate, sample rate or Channels only when a device or app asks for specific values.
Convert and download
Click Convert audio. Each file gets its own Download button; with two or more files and Package converted files as ZIP turned on, you get one Download ZIP button instead.

Which output format to choose

Bitrates in brackets are the defaults each tile starts with.

MP3 or M4A for everyday listening

MP3 starts at 192 kbps. M4A starts with AAC at 192 kbps and also offers lossless ALAC. AAC produces an AAC audio stream without the M4A container. MP4A uses the same container as M4A with a .mp4a extension; check whether your destination app accepts that extension.

Opus, OGG and WebM for web and voice

Opus (128 kbps) is the most efficient lossy codec here and keeps speech clear even at 64k. OGG uses Vorbis (160 kbps) by default and can carry Opus instead; WebM holds Opus (128 kbps) or Vorbis.

Lossless formats for editing and archiving

FLAC compresses audio without discarding samples and keeps 24-bit sources at 24-bit. WAV, W64, AIFF, AU and CAF can store uncompressed PCM, and ALAC in M4A or CAF is Apple's lossless codec. TTA and WV (WavPack) are lossless but less widely supported. MKA wraps FLAC (default), AAC, Opus, Vorbis or PCM in Matroska. Keep bit depth, sample rate and channels unchanged to preserve the decoded audio.

AC3, E-AC3 and MP2 for video and broadcast

Dolby Digital (AC3) and Dolby Digital Plus (E-AC3), both 192 kbps by default, keep 5.1 surround for DVD, Blu-ray and TV production. MP2 (192 kbps) is still used in broadcasting. All three run only at 32,000, 44,100 or 48,000 Hz.

AMR, SPX and WMA for older devices

AMR is a phone speech codec: AMR-NB encodes 8,000 Hz mono at 12.2 kbps and AMR-WB 16,000 Hz mono at 23.85 kbps. SPX (Speex) targets voice at 32 kbps and cannot go much above 44 kbps. WMA (192 kbps) suits older Windows software and holds at most two channels at up to 48,000 Hz.

Settings and when to change them

All settings apply to every selected file.

Audio codec: Most formats have a single codec. The real choices are PCM 16-bit, 24-bit or 32-bit for WAV, W64, AIFF and AU, ALAC or PCM for CAF, AAC or ALAC for M4A and MP4A, Vorbis or Opus for OGG and WebM, AMR-NB or AMR-WB, five codecs for MKA, and MP2 Fixed, a fixed-point variant of the same MP2 encoder. WAV, W64, AIFF and AU start at PCM 16-bit, which reduces a 24-bit recording to 16-bit; choose PCM 24-bit to keep the full depth.
Bitrate: Seven chips from 64k to 320k (kbps) for lossy formats; lossless formats show Lossless instead. As a rule of thumb, 64k to 96k is enough for speech, 128k to 192k for most music, and 256k or 320k keeps music closest to the source. Lossless codecs, including ALAC in M4A and FLAC or PCM in MKA, need no bitrate setting. MP3 caps the bitrate at 160 kbps for 16,000 to 24,000 Hz and at 64 kbps for 8,000 to 12,000 Hz.
Sample rate: Keep source rate (default) leaves the rate unchanged where the codec supports it. The list only offers rates the chosen format and codec can store: up to 96,000 Hz for lossless formats and 48,000 Hz for lossy ones. Opus runs at 8,000, 12,000, 16,000, 24,000 or 48,000 Hz and always plays back at 48,000 Hz. 44,100 Hz is the CD standard, 48,000 Hz the norm for video; a higher rate than the source adds no detail.
Channels: Keep source channels (default) keeps mono, stereo and 5.1 in formats that support surround, such as WAV, FLAC, M4A, Opus, AC3 and E-AC3; MP3, MP2 and SPX mix surround down to stereo. Mono merges everything into one channel, which halves the size of a stereo WAV; for lossy formats, lower the bitrate as well, since it sets the size. Stereo always writes two channels.
Package converted files as ZIP: Appears when two or more files are selected and bundles all converted files into one ZIP archive.
Strip metadata: Off by default, so tags such as title, artist and album are copied into the new file. Turn it on to remove them. Embedded cover art is never carried over.

Lossy, lossless and file size

Every conversion decodes the audio and encodes it again, even when source and target use the same codec. Going from one lossy format to another, such as MP3 to M4A, compresses a second time and loses a little more detail, so start from the best original you have.

Uncompressed size is simple arithmetic: sample rate × bytes per sample × channels × duration. A 4-minute stereo recording at 48,000 Hz and 24-bit is 48,000 × 3 × 2 × 240 = 69,120,000 bytes, about 66 MiB as WAV. At a constant MP3 bitrate, the same duration needs about 9.2 MiB at 320k, 5.5 MiB at 192k and 3.7 MiB at 128k, plus file overhead. FLAC compression depends on the recording's content; it has no fixed reduction ratio.

Formats, limits and file handling

Input formats: MP3, WAV, FLAC, M4A (AAC or ALAC), AAC, OGG, Opus and audio-only WebM and MP4. For the sound of an MP4 video, use MP4 to MP3.
Output formats: MP3, WAV, FLAC, M4A, MP4A, AAC, OGG, Opus, AC3, E-AC3, AIFF, CAF, W64, AU, MP2, AMR, SPX, TTA, WV (WavPack), WMA, MKA and WebM; optional ZIP for batches
Maximum file size: 200 MB per file
Files per job: One or more, converted one after another in a single job. If one file fails, the whole job stops and no files are delivered.
File names: Source name with the new extension; spaces and special characters become hyphens (My Song (live).wav becomes My-Song-live.mp3).
Processing: On our server with FFmpeg. A job stops after 15 minutes.
Watermark: None
Storage: Uploads and results are deleted automatically after 48 hours for guests and after 30 days with a free account. Paid plans can keep files longer. Until then, results stay in Files for other audio tools such as Split Audio.

Frequently asked questions

Does converting MP3 to WAV or FLAC improve the quality?

No. Detail removed when the MP3 was made is gone. A lossless codec can preserve the decoded audio if you keep its bit depth, sample rate and channels. Uncompressed PCM usually needs more space: a 16-bit, 44,100 Hz stereo WAV takes about 10 MB per minute, a 192 kbps MP3 about 1.4 MB. Convert to WAV or FLAC when an editor or upload form requires it.

Why is the converted file larger than the original?

Size depends on duration, audio content and output settings. Uncompressed PCM often needs more space; lossless compression also depends on the recording's content, bit depth, sample rate and channels. A 128 kbps MP3 converted at 192k grows without recovering detail lost earlier. Compare formats and settings for your intended use rather than assuming every lossless result must be larger.

How does the converter handle each codec's limits?

Every codec has a fixed range, and the options adapt to it. WMA holds at most two channels at up to 48,000 Hz, so surround or 96,000 Hz sources are mixed down to stereo and written at 48,000 Hz. Vorbis (OGG, WebM, MKA) works at up to 48,000 Hz with a bitrate that suits the sample rate, and Opus tops out at 256k for a mono track. If a source file is damaged or cannot be decoded, the job stops and names the problem.

Which format plays on iPhone, Android and Windows?

Check the formats supported by the player or app where you will use the file. MP3 and M4A with AAC are common choices for compressed audio; M4A with ALAC and FLAC provide lossless alternatives when the destination supports them. An editor may instead require WAV with a particular sample rate and bit depth.

Who can access my files?

Files are uploaded to our server and converted there with FFmpeg; no AI or third-party service receives them. They are deleted automatically as described above. To send a result to someone, use Share Files.