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Supports: MTS
.mts is AVCHD camcorder footage — the Sony and Panasonic format from 2006 that pairs H.264 video with a Dolby Digital (AC-3) soundtrack, or with uncompressed LPCM on professional bodies. AIFF is the other end of the spectrum: Apple's Audio Interchange File Format from 1988, built on Electronic Arts' IFF chunk structure, storing plain uncompressed PCM samples in big-endian byte order. Converting one to the other is what you do when camcorder audio has to go into Logic, Pro Tools, Final Cut or a sampler that wants an uncompressed file rather than a compressed one. This page covers what the conversion produces, why there are no bitrate controls on it, and the file-size ceiling that catches people with long recordings.
.mts clips onto the page or click "Add Files". AVCHD cameras often split a long take across several numbered files on the card — convert each in turn..aiff. Files upload over an encrypted connection, are processed on our servers, and are deleted automatically after a few hours. No sign-up, no watermark.| Property | Value |
|---|---|
| Full name | Audio Interchange File Format ("Audio IFF") |
| Created by | Apple, 1988, based on Electronic Arts' EA IFF 85 interchange standard |
| Structure | A FORM container chunk with formType AIFF, holding a Common chunk (COMM) and a Sound Data chunk (SSND) |
| Byte order | Big-endian throughout — the spec stores all data in Motorola 68000 format |
| Payload | Uncompressed PCM by default; no compression, no bitrate |
| Bit depth here | 16-bit. This page exposes no 24-bit or 32-bit option |
| Compressed variant | AIFF-C (AIFC), added July 1991, which allows a compression type in the COMM chunk |
| Size ceiling | Chunk sizes are signed 32-bit values, so roughly 2 GiB per file |
| Best for | Handing uncompressed audio to a DAW, sampler or editor on any platform |
The "Audio Codec" group is the only real decision on this page, and it changes what kind of file you get more than it changes how it sounds.
| Option | What it writes | File type | Use when |
|---|---|---|---|
| PCM 16-bit Big Endian (preselected) | Standard big-endian PCM samples | A true FORM…AIFF file |
Always, unless you have a specific reason not to |
| PCM 16-bit Little Endian | Byte-swapped PCM, stored under the sowt compression type |
FORM…AIFC |
A tool specifically wants little-endian samples in an AIFF wrapper |
| PCM A-law | ITU-T G.711 A-law, 8-bit companded | FORM…AIFC |
Telephony-style interchange; not for music |
| PCM mu-law | ITU-T G.711 mu-law, 8-bit companded | FORM…AIFC |
Same as above |
The distinction matters more than the names suggest. AIFF-C is a superset — the header declares AIFC rather than AIFF and names a compression type — and while modern audio software reads it happily, older samplers, hardware and pickier import paths sometimes accept only a plain AIFF. If a file is going somewhere unfamiliar, the preselected big-endian option is the safe one. And despite their names, A-law and mu-law are not uncompressed: they are 8-bit companded telephony codecs, so choosing one throws away most of the fidelity that made you reach for AIFF in the first place.
If you have converted to MP3 or AAC here you will notice something missing: the entire "File Compression" group is absent on this page. That is correct rather than a gap. AIFF stores raw PCM samples, so there is no bitrate to choose and no quality/size trade-off to make — the file size falls straight out of arithmetic. At the camcorder's native 48 kHz, 16-bit stereo, that is 192,000 bytes per second of audio, or about 691 MB per hour. Mono halves it.
The three controls that do change the output size are therefore "Audio Sample Rate" (halving the rate halves the size, at the cost of bandwidth), "Audio Channel" (Mono halves it, at the cost of the stereo image), and "Trim" (converting only the span you need). None of them are quality settings in the compressed-audio sense; they change what is recorded, not how hard it is squeezed.
There is also an honest caveat about fidelity. AVCHD normally records Dolby Digital AC-3, which is lossy. Decoding that to uncompressed PCM gives you a file that is much larger than the source and no better than it — AIFF preserves whatever it is handed, it does not restore anything. That is still the right thing to do when a DAW or sampler needs uncompressed input, but do not expect the conversion itself to improve the recording.
The AIFF specification declares chunk sizes with a signed 32-bit long, whose maximum value is 2,147,483,647 — so an AIFF file tops out near 2 GiB. Uncompressed audio reaches that faster than people expect. At 48 kHz, 16-bit stereo the limit arrives after roughly three hours and six minutes of audio; in mono, roughly six hours and thirteen minutes.
For most camcorder work that is comfortable headroom, since AVCHD cameras split recordings into separate files on the card anyway. It becomes a real constraint on long-form material — a full conference session, a concert, a day of ambient recording assembled into one file. If you are near that boundary, use the "Trim" control to convert in sections, or choose a format without the ceiling: MTS to FLAC gives you lossless audio at roughly half the size, and MTS to WAV is the little-endian equivalent that most Windows-side tools prefer.
No. AIFF is an uncompressed container, and uncompressed does not mean improved — it means nothing further is discarded. AVCHD footage almost always carries Dolby Digital AC-3, which already dropped information when the camcorder recorded it, and decoding that to PCM reproduces exactly what the AC-3 decoder outputs, at roughly ten times the file size. The reason to convert is compatibility: DAWs, samplers and editors often want uncompressed input, and some will not import AVCHD audio at all. It is not a restoration step.
16-bit, and no — this page has no bit-depth control at all. The "Audio Codec" group offers PCM 16-bit Big Endian, PCM 16-bit Little Endian, PCM A-law and PCM mu-law, and that is the complete list. Sixteen bits at 48 kHz is more than enough headroom for AC-3-sourced camcorder audio, since the source has nothing like 96 dB of usable dynamic range left in it. If your project genuinely needs 24-bit files, resave from your DAW after importing.
AIFF stores plain uncompressed PCM; AIFF-C, added in July 1991, extends the format so the Common chunk can name a compression type, and its header reads AIFC instead of AIFF. On this page the preselected PCM 16-bit Big Endian option gives you a true FORM…AIFF. The other three — little-endian PCM (stored as the sowt type), A-law and mu-law — all produce FORM…AIFC. Nearly all current software reads both, but if a file has to satisfy an old sampler or a strict import filter, stay on the preselected option.
Because you removed the compression, not because anything was added. AVCHD encodes its soundtrack with AC-3 at somewhere between 64 and 640 kbit/s; uncompressed 48 kHz 16-bit stereo PCM runs at 1,536 kbit/s regardless of content. That is a jump of several times over, and it is the whole point of the format — every sample is stored literally, with no encoder deciding what you will not miss. If size is a problem and you still want lossless, FLAC stores the identical samples in roughly half the space.
Yes, around 2 GiB. The specification defines each chunk's size field as a signed 32-bit value, which caps out at 2,147,483,647 bytes, and that limit applies to the file as a whole because the audio lives inside a FORM chunk. In practical terms that is about three hours and six minutes at 48 kHz 16-bit stereo, or a little over six hours in mono. Long single-file recordings are the case to watch; use "Trim" to convert in sections, or pick a format without the constraint.
Yes, if you leave "Audio Sample Rate" on Original — and that is what we would recommend, since AVCHD records at 48 kHz and resampling is a lossy arithmetic step you gain nothing from. The dropdown also offers 8000, 12000, 16000, 24000, 44100 and 48000 Hz. The one common reason to change it is matching a session that runs at 44.1 kHz, and even then it is usually better to let your DAW handle the conversion with its own high-quality resampler.
Yes. Switch the "Trim" control away from "Unchanged" and give it a start time and a duration, in seconds or in HH:MM:SS.sss form, and only that span is decoded and written. With uncompressed audio this is worth doing as a matter of habit rather than as an exception, because every minute you convert costs about 11.5 MB in stereo. Note that a generic audio cutter will not accept an MTS file directly, so trimming here — or on the video before conversion — is the practical route.
Your MTS is uploaded to our servers over an encrypted connection, the audio track is decoded and written out as PCM there, and both the upload and the result are deleted automatically after a few hours. There is no sign-up, no watermark, and your recordings are never shared or made public. Expect the upload to be the slow part: AVCHD records at up to 24 Mbit/s, so the video you are discarding dominates the transfer even though only the soundtrack survives the conversion.