How to use this workflow
Check the destination
Find out which formats the receiving editor, device or service accepts. There is no single format that is ideal for every destination.
Keep a source for further edits
Retain the best available original. Use PCM or a suitable lossless format between processing steps when the next tool supports it.
Make a delivery copy
Use the converter to choose a supported format and settings. Listen to the exported file and keep the editable source separately.
What the converter’s choices mean
A codec describes how sound data is encoded. A container holds encoded data and associated information. M4A can hold AAC or ALAC; its extension alone does not tell you whether the audio is lossy. The choices below describe this converter’s outputs.
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| Output | Encoding here | Practical consideration |
|---|---|---|
| MP3 | Lossy MP3 | A compact delivery option when the destination accepts it |
| WAV | Uncompressed PCM | Larger files; straightforward editing interchange |
| FLAC | Lossless compression | Smaller than equivalent PCM in many cases; check compatibility |
| AAC / M4A | Lossy AAC in MP4 container | Confirm the receiver accepts M4A/AAC |
| OGG | Lossy Vorbis in Ogg container | Check the target application’s Vorbis support |
| AIFF | Uncompressed PCM | An alternative PCM container for compatible editors |
| ALAC / M4A | Lossless ALAC in MP4 container | Different encoding from AAC despite the same extension |
Bitrate, sample rate and bit depth are different
Bitrate is data per second, commonly shown in kbps for compressed outputs. Sample rate is samples per second per channel, shown in Hz or kHz. PCM bit depth describes the number of bits used for each sample value. They are not interchangeable quality scores.
For uncompressed stereo PCM, 44,100 samples/second × 16 bits × 2 channels equals 1,411,200 bits/second. A sixty-second example contains 10,584,000 bytes of sample data before container overhead. This is arithmetic for those settings, not a claim about every WAV file.
Input support is not determined by the extension alone
Most waveform tools rely on the browser’s audio decoder, while the converter and extractor use a locally loaded FFmpeg engine. A file that opens in one path may fail in another because codec support differs.
Decoding in Web Audio can resample to the audio context’s sample rate. Do not assume every processing path preserves the original encoded sample rate or metadata. If tags, artwork or exact stream preservation matter, verify them in the exported file.
Common questions
Is every .m4a file lossless?
No. This converter writes AAC and ALAC to .m4a. AAC is lossy; ALAC is lossless compression.
Will changing 44.1 kHz to 96 kHz restore detail?
No. Upsampling does not recreate information missing from the original recording. Choose a rate that the next stage requires.