DFF to AIFF Converter
Convert DFF audio files to AIFF right in your browser with FFmpeg, without uploading them anywhere.
Input
Output
Readme
Why convert audio?
An audio file is a container, named by its extension, that holds sound compressed with a codec. A program, phone, car stereo, or website plays the file only if it supports both. Converting puts the sound into a container and codec that play where you need them.
Lossy formats, such as MP3, AAC, and Opus, leave out detail you're unlikely to hear to make small files. Lossless formats, such as FLAC and ALAC, keep every detail, and uncompressed ones, such as WAV and AIFF, store the sound as it is.
| Format | Good for |
|---|---|
| WAV | Editing and burning CDs; uncompressed and plays almost anywhere |
| MP3 | Sharing and playing almost anywhere |
| M4A | Apple devices and most phones |
| FLAC | Music collections, smaller than WAV without losing anything |
| AAC | Players and streams that take raw AAC |
| OGG | Games and open-source software |
| Opus | Speech and music at low bitrates, such as voice messages |
| WMA | Windows Media Player and older Windows programs |
| AIFF | Apple audio software such as Logic and GarageBand |
| ALAC | Lossless music in Apple Music and on iPhones |
| M4R | iPhone ringtones |
| M4B | Audiobooks |
| WebM | Audio on web pages |
| MKA | Keeping the sound as it is in Matroska, for players such as VLC |
| AC3 | Dolby Digital sound for DVDs, TVs, and AV receivers |
| EAC3 | Dolby Digital Plus sound for streaming and TVs |
| DTS | DTS sound for AV receivers |
| MP2 | Broadcasting, DVDs, and Video CDs |
| CAF | Apple developer tools and iOS apps |
| AU | Older Unix programs and Java |
| WV | Lossless archives with WavPack |
| TTA | Lossless archives with True Audio |
These formats can be converted from, but not to:
| Format | What it is |
|---|---|
| AMR | Voice recordings from phones and voice recorders (AMR-NB and AMR-WB) |
| 3GA | Voice recordings from Android phones |
| APE | Monkey's Audio, a lossless format |
| DSF, DFF | DSD from SACDs and high-resolution music stores |
| MPC | Musepack |
| SPX | Speech compressed with Speex |
| RA | RealAudio |
| VOC | Creative Voice |
| TAK | Tom's lossless Audio Kompressor |
| SHN | Shorten, a lossless format |
Tool description
This tool converts audio between WAV, MP3, M4A, FLAC, AAC, OGG, Opus, WMA, AIFF, ALAC, M4R, M4B, WebM, MKA, AC3, EAC3, DTS, MP2, CAF, AU, WV, and TTA, and from AMR, 3GA, APE, DSF, DFF, MPC, SPX, RA, VOC, TAK, and SHN to any of those. Every audio converter here works the same way, from the general Audio Converter to the ones for a single pair, such as WAV to MP3 or FLAC to ALAC: it starts on the formats in its name, and the Source and Target menus switch to any other. It converts one file or many at once. Browse files lists only files in the source format, but files dropped on it can be in any of them, and each one converts from its own format, so a FLAC file converts even on the WAV to MP3 converter. The Quality, Sample Rate, and Channels options in Settings, under the target format, can make the files smaller or change their sound, and they keep it as it is to begin with. Then press Convert. The progress bar shows while FFmpeg loads and how far the conversion has got, and Cancel stops it. Picking another format or option clears the results until you press Convert again. When it's done, the converted files show up in a grid. The converted files keep the original file names with the new extension.
Audio is converted on your device with FFmpeg and never uploaded. Before the first conversion, the page downloads FFmpeg.
Convert many audio files at once
Pick or drop as many audio files as you like, and add more at any time: this is a batch audio converter, so a whole album or a set of voice recordings converts with one press of Convert, whether it's WAV to MP3, FLAC to ALAC, or M4A to MP3.
- The files convert one after another, in the order they were added, and the progress bar counts them. Cancel stops the file converting and the ones waiting, and the files already converted keep their results.
- Files added later wait for Convert, which then converts only them and any that failed.
- The converted files show up in a grid, each with its own Download button.
- Download All saves every converted file in one ZIP file, or a single converted file as it is.
- When two files would get the same name, a number is added to the later one, as in track (1).mp3.
- If a file can't be converted, an error message names it, and the others still convert. Press Convert to try it again.
Options
Quality sets the bitrate a lossy format is encoded at, or for OGG, the Vorbis quality level, which encodes at about the bitrate shown. Each value is marked High, Medium, or Low: lower ones make smaller files that sound duller or less clear. Every target starts on its bold High value. A value the next target you pick doesn't offer becomes that target's bold value in the same group, so 112 kbps (Low) in MP3 becomes 96 kbps in AAC. Lossless formats (WAV, FLAC, AIFF, ALAC, CAF, AU, WV, and TTA) keep all of the sound, and so does MKA, which copies the sound as it is or stores it as FLAC, so their Quality shows Lossless and can't be changed.
Target High (kbps) Medium (kbps) Low (kbps) MP3 320, 256 224, 192, 160 128, 112, 96, 80, 64, 48, 32 M4A, M4R, M4B, AAC 320, 256 224, 192, 160 128, 96, 64, 48, 32 OGG (Vorbis level) ~500 (10), 320 (9), 256 (8) 224 (7), 192 (6), 160 (5) 128 (4), 112 (3), 96 (2), 80 (1), 64 (0) Opus, WebM 256, 192 160, 128 96, 64, 48, 32 WMA 256, 192 160, 128 96, 64, 48, 32 AC3, EAC3 640, 576, 512, 448, 384 320, 256, 224 192, 160, 128, 96 DTS 1509, 1280 1024, 768 512, 384 MP2 384, 320 256, 224 192, 160, 128, 96, 64 The bitrate is the same for any number of channels. MP2 gets at most 192 kbps for mono and 160 kbps at a sample rate under 32 kHz, and DTS at least 192 kbps for each channel of surround sound.
Sample Rate sets how many times per second the sound is measured. Original keeps the file's own. Lower rates lose the highest pitches, and make smaller lossless files. Audio CDs use 44.1 kHz. Some formats take only certain rates, and a file gets the closest higher one the format takes, or its highest: MP3, WMA, and DTS go up to 48 kHz, AAC-based formats to 96 kHz, AC3 and EAC3 take 32, 44.1, and 48 kHz, MP2 takes 16 to 48 kHz, and Opus takes 8, 12, 16, 24, and 48 kHz.
Channels keeps the file's own (Original), mixes them into two (Stereo), or into one (Mono). MP3, MP2, and WMA files get two channels at most, and AC3, EAC3, and DTS files six (5.1), so surround sound is mixed down to fit.
How it works
FFmpeg, compiled to WebAssembly, runs in your browser. It takes the first audio track of the file and writes it in the target format. Video, cover art, and other tracks are left out, so a WebM video becomes WebM audio. Tags such as the title, artist, and album are kept where the target format holds them.
- When the sound already uses a codec the target format holds, and the options keep its sample rate, channels, and, for a lossy format, the Quality it starts on (its bold High value), it's copied as it is. Copying is much faster than re-encoding and loses no quality: M4A to AAC, M4R, or M4B, Opus to WebM, ALAC to CAF, or almost anything to MKA.
- Otherwise it's re-encoded with the target format's codec, which takes longer. FFmpeg uses one processor core in the browser.
| Target | Codec | Copied as it is |
|---|---|---|
| WAV, CAF | PCM | PCM; CAF also ALAC |
| AIFF, AU | PCM (big-endian) | Big-endian PCM; AU also μ-law and A-law |
| MP3 | MP3 (LAME) | MP3 |
| M4A, M4R, M4B, AAC | AAC | AAC |
| FLAC | FLAC | FLAC |
| ALAC | Apple Lossless | ALAC |
| OGG | Vorbis | Vorbis |
| Opus | Opus | Opus |
| WebM | Opus | Opus and Vorbis |
| WMA | Windows Media Audio 2 | Windows Media Audio 1, 2, Pro, and Lossless |
| MKA | FLAC | AAC, MP3, MP2, AC3, EAC3, DTS, TrueHD, FLAC, ALAC, Opus, Vorbis, WavPack, TTA, PCM |
| AC3 | Dolby Digital | AC3 |
| EAC3 | Dolby Digital Plus | EAC3 |
| DTS | DTS | DTS |
| MP2 | MPEG Audio Layer II | MP2 |
| WV | WavPack | WavPack |
| TTA | True Audio | TTA |
- Lossless formats keep the bits per sample of the original: 24-bit sound stays 24-bit, and 16-bit stays 16-bit. Sound from a lossy format becomes 16-bit. Floating-point sound stays floating point in WAV, CAF, AIFF, AU, and WV, while FLAC, ALAC, and TTA store it as 24-bit.
- DSD (DSF and DFF) becomes PCM at 88.2 kHz, 24-bit in lossless formats. Pick another Sample Rate for a different one. A format that takes lower rates only gets its highest.
- ALAC files are saved with the .m4a extension, as Apple saves them. M4R and M4B files are M4A files under another extension.
Tips
- For a file that plays almost anywhere, convert to MP3.
- Converting a lossy file to a lossless format doesn't bring back the detail it lost; it only makes the file bigger. Each conversion from one lossy format to another loses a little more, so convert from the original when you can.
- To make a file smaller, lower the Quality, pick Mono with a lower Quality for speech, or lower the Sample Rate.
- FFmpeg keeps the whole file and the converted one in memory, so a very long recording may not convert, especially on a device with little memory.