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Supports: DV
This page takes a .dv file — a MiniDV, DVCAM, Digital8, or DVCPRO transfer — decodes its soundtrack, and writes it out as a WAV. The picture is discarded; only the audio comes back. DV is an unusually good source for this, because DV never compressed its audio in the first place: the tape holds 16-bit linear PCM at 48 kHz, or four nonlinear 12-bit channels at 32 kHz in the alternate mode. WAV holds linear PCM too. So with the default settings there is no lossy encoder anywhere in the chain — which is exactly what you want when the recording is an interview, a recital, a lecture, or the only surviving audio of someone's voice.
.dv stream and a DV-in-AVI or DV-in-QuickTime capture all decode here, and several files can be queued to run with the same settings.HH:MM:SS.sss to lift one passage out of a long reel.Every lossy audio target on this site has a File Compression group with presets and bitrates. This page does not, and its absence is not an oversight. WAV stores raw samples, so the file size is fixed by arithmetic — sample rate × bit depth × channels — and there is no encoder to tune. The only thing that changes the size is the Audio Codec you pick, and every step away from 16-bit is a trade rather than an improvement.
| Audio Codec | Bits per sample | Size at 48 kHz stereo | Relationship to the DV source |
|---|---|---|---|
| PCM 16-bit Little Endian — the default | 16 | 11.52 MB per minute | Exact match for DV's samples |
| PCM 24-bit Little Endian | 24 | 17.28 MB per minute | Pads each sample with empty bits; no added detail |
| PCM 32-bit Little Endian | 32 | 23.04 MB per minute | Same, more so |
| PCM A-law | 8, logarithmic | 5.76 MB per minute | Discards resolution; a telephony encoding |
| PCM mu-law | 8, logarithmic | 5.76 MB per minute | Same, the North American variant |
Those figures are measured, not estimated — a 48 kHz stereo stream re-encoded through each format lands on exactly those rates, because uncompressed audio has no content-dependent variation. Pick 16-bit unless a mastering or restoration tool downstream has explicitly asked you for a wider container; A-law and mu-law only make sense when you are feeding a legacy telephony system that expects them.
DV recorded audio in one of two modes, and which one your tape used decides what "correct" means. The standard high-quality mode is 16-bit linear PCM, stereo, at 48 kHz — this is what the overwhelming majority of consumer and prosumer tapes used. The alternate mode is four nonlinear 12-bit channels at 32 kHz, which existed so a second stereo pair could be dubbed on after shooting.
.dv in a player first; if it is silent there, the capture is the problem and re-capturing is the only fix..m2t, but they hold MPEG-2 video with MPEG-1 Layer II audio, not DV with PCM. This decoder does not apply to them.WAV is the right answer for a preservation master and the wrong answer for almost everything else. Since DV's audio has never been through a lossy encoder, any of these targets gives you a clean first-generation result — the question is only what you are going to do with the file.
| Target | Lossless? | One hour of 48 kHz stereo | Best for |
|---|---|---|---|
| WAV (PCM 16-bit) | Yes | about 691 MB | Ingest into editing and restoration software, preservation masters |
| FLAC | Yes | typically well under half of WAV, content-dependent | Long-term archives where storage cost matters, tagged libraries |
| MP3 | No | tens of MB at a typical bitrate | Playback on absolutely any device, including 2000s-era hardware |
| Opus | No | smaller again at equal quality | Web delivery, phones, computers, storage-conscious listening copies |
| M4A / AAC | No | comparable to MP3 at better quality | Apple ecosystems and modern mobile playback |
Not with the default settings. DV's standard mode stores 16-bit linear PCM and the default PCM 16-bit Little Endian output stores 16-bit linear PCM, so the samples pass straight through with no lossy encoder in between. We verified this end to end: a known PCM signal muxed into a DV file and extracted back out came through with zero differing samples across the entire stream. The one caveat is the 12-bit four-channel mode, whose nonlinear samples are expanded to linear 16-bit values on decode — that preserves everything the tape held, but it is a conversion rather than a literal copy.
PCM 16-bit Little Endian, unless something downstream has told you otherwise. It is the exact width of DV's own samples, so it neither truncates nor pads. The 24-bit and 32-bit options do not improve anything on a 16-bit source; they just append zero bits to every sample and inflate the file by 50% and 100% respectively. Some mastering and restoration suites prefer to work in a wider format, which is a legitimate reason to pick them — "more bits must be better" is not.
Use Original, which is the default. It reads the rate directly from the DV stream, so a 32 kHz four-channel tape comes out at 32 kHz in the WAV without you selecting anything. The dropdown's fixed entries — 8000, 12000, 16000, 24000, 44100, 48000 Hz — are there for forcing a rate when a header is wrong or when you deliberately want to downsample speech, and none of them is the right tool for a correctly labelled 32 kHz tape.
Because WAV is not a compressed format, so there is nothing to compress. Its size is pure arithmetic: sample rate × bit depth × channels. The lossy pages on this site show a File Compression group with presets and bitrates because their codecs make real quality-versus-size trades; here the only lever is the sample format, and every alternative to 16-bit is either padding or a downgrade. If what you actually want is a small file, you want a different target — see the last question below.
Not as four separate tracks. The 12-bit mode carries two stereo pairs — typically the camera microphone plus a pair dubbed on afterwards. This converter outputs the primary stereo pair, following your Audio Channel setting. Recovering both pairs as independent tracks requires a dedicated multi-track DV demuxer, after which each pair can be converted separately.
Yes, exactly normal. Uncompressed 16-bit/48 kHz stereo runs 192,000 bytes every second — 11.52 MB per minute, 691 MB per hour — regardless of whether the hour is a symphony or a silent room, because every sample is stored at full width whether or not anything is happening. That constancy is the feature: it is why WAV is a preservation format and why it is a poor delivery format.
Both are lossless, so the audio is identical either way; the difference is size and tooling. FLAC compresses PCM losslessly and typically lands well under half the size of the equivalent WAV, and it carries proper metadata tags. WAV is the more universally accepted format for ingest into editing and restoration software, and it has no decode step at all. If the file is going into an archive you will keep for decades, DV to FLAC saves real storage with nothing given up; if it is going straight into an editor tonight, WAV is the frictionless choice.
Then WAV is the wrong target and you should skip the intermediate step. DV to MP3 plays on essentially any device ever made, and DV to Opus is the more efficient modern codec for anything that will be played on a computer, phone, or website. Both re-encode, but since DV's audio has never been through a lossy encoder before, that first encode is as clean as a first encode can be.
It is uploaded over an encrypted connection, processed on our servers, and deleted automatically a few hours later — no sign-up, no watermark, never shared or made public. Be aware of the upload size before you start: DV runs at a fixed 3.6 MB per second of tape, so an hour-long capture is roughly 13 GB. There is no per-file cap, but the whole file has to arrive before the audio can be pulled out of it — the Trim control shortens the output, not the upload — so cutting the capture down beforehand is the only thing that makes the transfer quicker. To split a reel you have already sent, the video cutter handles that as a separate job.