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Supports: MTS
An .mts file is what a Sony or Panasonic AVCHD camcorder writes to its memory card: MPEG-4 AVC/H.264 video with a Dolby Digital (AC-3) or linear PCM soundtrack alongside it. This page discards the picture and re-encodes only that soundtrack to Opus, the IETF codec defined in RFC 6716 that WebRTC, Discord and most modern voice apps rely on.
This tutorial is for anyone lifting an interview, a recital, a lecture or wildlife ambience off camcorder footage. Two things trip people up on AVCHD specifically — which audio track gets used when the camera recorded more than one, and why asking for 44.1 kHz does not give you 44.1 kHz. Both are covered below.
.mts clip onto the page or click "Add Files" to browse. Several clips can queue together and they share one set of settings..opus file. Files upload over an encrypted connection, are processed on our servers, and are deleted automatically after a few hours — no sign-up, no watermark.Bitrate. Opus reaches transparency far lower than MP3 does, so the MP3 habit of reaching for 320 kbps wastes bytes here. RFC 6716 puts the codec's operating range at 6 to 510 kbit/s and names 64–128 kbit/s as the sweet spot for fullband stereo music. Custom Bitrate accepts anything inside the encoder's 8–512 kbps range, the Constant Bitrate dropdown runs from 8 up to 384 kbps, and the Quality Preset levels are scaled across that same range, adjusted for the channel count and sample rate you picked.
Channels. Audio Channel set to Original keeps whatever the camcorder recorded. That matters more than usual on AVCHD, because 5.1-capable handycams write a six-channel AC-3 track, and Original will carry all six into the Opus file. Choose Mono for a single-mic interview — it roughly halves the data rate at the same perceived quality — or Stereo to force a downmix of a surround track.
Sample rate. This is the AVCHD-specific gotcha. 44100 Hz appears in the dropdown because that list is shared across every audio codec we support, but Opus has no 44.1 kHz mode: RFC 6716 defines exactly five rates — 8, 12, 16, 24 and 48 kHz. Rather than failing the job, our validation layer snaps your request to the nearest legal rate before ffmpeg ever sees it, so 44100 silently becomes 48000. That is worth knowing because running the same request through raw ffmpeg on your own machine would error out.
| What you recorded | Bitrate | Channel setting | Notes |
|---|---|---|---|
| Single-mic interview or lecture | 24–48 kbps | Mono | Speech stays clear well below where MP3 falls apart |
| Voice with room tone or wind | 48–64 kbps | Mono or Stereo | Keeps ambience from turning into swirl |
| General event audio | 64–96 kbps | Stereo | Sensible default for a stereo camcorder track |
| Music, recital, live performance | 96–128 kbps | Stereo | RFC 6716's stated sweet spot for fullband stereo |
| Surround AC-3 kept intact | 160–256 kbps | Original | Six channels need proportionally more bits |
| Source track on the card | Nature | Going to Opus |
|---|---|---|
| Dolby Digital AC-3, stereo, 64–640 kbps | Lossy | Lossy-to-lossy re-encode; one extra generation of loss |
| Dolby Digital AC-3, 5.1 | Lossy | Carried at up to six channels when Audio Channel is Original |
| Linear PCM, 1.5 Mbps stereo | Uncompressed | First lossy step; use 128 kbps or higher to stay clean |
| Any track, 48 kHz | — | Passes through untouched; 48 kHz is Opus's native rate |
| Any track, requested at 44100 Hz | — | Snapped to 48000 Hz before encoding |
.mts only. If your import software already renamed the clip to .m2ts — Sony's does — the byte-identical stream will be refused here. Use the video to Opus extractor, which accepts both spellings.Opus is an audio codec, so there is no path here that keeps the picture — if you wanted both, MTS to MP4 is the conversion you actually want. Opus also has real playback gaps: it is native in Chrome, Firefox and Edge, on Android, in VLC and Discord, and in Safari on recent Apple systems, but older car head units, legacy portable players and pre-Windows 10 systems generally need a decoder installed. When the file has to open anywhere without discussion, MTS to MP3 is the safer target. And a truncated clip copied out of a camcorder's BDMV folder mid-write may decode a few seconds and stop; re-copy the original file from the card before blaming the settings.
The one with the most channels — so the 5.1 track. We do not send a stream-selection flag to ffmpeg, which means ffmpeg's own default applies: for audio it scores streams by channel count and picks the highest, using the lowest stream index to break ties. If you specifically need the stereo mix, extract it in a desktop tool that lets you name the stream index.
Because the format has no such mode. RFC 6716 defines Opus at 8, 12, 16, 24 and 48 kHz only, corresponding to narrowband through fullband. 44100 Hz still appears in the dropdown because that list is shared across every audio codec on the site. Our validation layer catches the mismatch and snaps to the nearest supported rate, which is 48000 Hz — you get a working file rather than an error.
It adds one generation of lossy-to-lossy loss, which is real but usually inaudible if you do not starve the bitrate. AVCHD records Dolby Digital between 64 and 640 kbps; re-encoding a 256 kbps AC-3 stereo track to 96–128 kbps Opus is generally transparent for speech and close to it for music. Where it bites is repeated round-trips — extract once at a generous bitrate and edit from that file rather than re-converting the same audio several times.
Yes, meaningfully so at the low and middle of the range. Opus was designed to stay intelligible where MP3 becomes watery, which is why it is the codec behind WebRTC and most voice chat. Speech that needs 96–128 kbps as MP3 is usually clean at 32–48 kbps as Opus. The advantage narrows as you climb toward transparency, so at 128 kbps and up the size difference matters less than the compatibility difference.
Nothing is pre-selected in that dropdown — we checked the option surface for this page rather than assuming, and unlike the video pages there is no highlighted default preset. If bitrate matters for your recording, pick a preset explicitly or switch the File Compression mode to Constant Bitrate and type a number. Leaving it untouched is fine for a rough listen but is not a decision you have made.
Yes. Leave Audio Channel on Original and a six-channel AC-3 track is carried through as multichannel Opus. Bear in mind that six channels need proportionally more bits than two, so a bitrate that sounds fine for a stereo track will sound thin spread across 5.1. Set Audio Channel to Stereo instead if the destination is a podcast, a phone or anything that will fold the mix down anyway.
Two routes, and the better one depends on where the size problem is. If the clip is long and you only need a slice, use the Trim controls on this page — set a Start time and a Duration and only that range is encoded. If the upload is the bottleneck because the AVCHD video is huge, trim the source video first with the video cutter and run the short clip through here. Do not send an .mts file to the audio cutter; that tool takes audio files, not video.
No. The date, camera model and lens data that AVCHD carries live in the container and in the MPEG-TS stream structure, and none of it survives a re-encode into a bare .opus file — you get a Vorbis-comment header with nothing meaningful pre-filled. If the capture date matters for your archive, read it off the original clip and store it alongside, or keep the original .mts and treat the Opus file as a derivative.
The .mts file is uploaded over an encrypted connection, processed on our servers, and deleted automatically after a few hours. Nothing is shared or made public, there is no sign-up and no watermark. The practical limit you will run into on AVCHD is upload size and time rather than anything about the conversion itself, since a full HD video stream has to travel before the audio can be pulled out of it.