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Supports: ASF
An .asf is Microsoft's Advanced Systems Format — the container behind Windows Media, normally holding WMV video next to a WMA audio track. Converting it to .ac3 does two things at once: it discards the picture entirely and re-encodes the WMA soundtrack as a Dolby Digital (AC-3) elementary bitstream, the audio format DVD-Video authoring tools, AV receivers and S/PDIF links expect.
That makes this a delivery conversion, not an archival one. WMA is already lossy, and AC-3 is lossy too, so this is a second generation of perceptual coding on top of a first. It is the right move when something downstream specifically demands an AC-3 stream; it is the wrong move if you just want the audio in a general-purpose format, where ASF to MP3 or the lossless-PCM ASF to WAV will serve you better at a fraction of the bitrate.
.asf onto the page or click "+ Add Files". Several files can be queued and all of them run with the same settings. The video track is dropped automatically; only the audio is converted..ac3. Files are uploaded over an encrypted connection, decoded and re-encoded on our servers, and deleted automatically after a few hours — no sign-up, no watermark, never shared or made public.They are not three routes to the same place. Each one reaches the AC-3 encoder by a different path, and only one of them is clamped.
| Mode | What it offers | Range that actually reaches the encoder | When to use it |
|---|---|---|---|
| Quality Preset | Highest / Very High / High / Medium / Low / Very Low / Lowest | Computed from the preset, the channel count and the sample rate, then held inside 192-640 kbps | Only if you have no target spec in mind — and be aware it opens on Highest |
| Constant Bitrate | A fixed list: 32, 40, 48, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320, 384 kbps | Passed through exactly, including the sub-192 kbps rungs | Stereo or mono delivery where you want a specific low or mid rate |
| Custom Bitrate | A typed number with a bps / Kbps / Mbps unit selector | Clamped into 192-640 kbps — anything lower is raised to 192, anything higher pulled back to 640 | The two rates the dropdown cannot reach: 448 and 640 kbps |
The clamp asymmetry is deliberate and it matters. If you type 96 into Custom Bitrate you will get a 192 kbps file; pick 96 kbps from the Constant Bitrate list instead and you get exactly 96 kbps. We confirmed the pass-through on a 6-second stereo 44.1 kHz source: the 32 kbps selection encoded at 32,000 bps into 24,102 bytes, 192 kbps at 192,000 bps into 144,612 bytes, and 384 kbps at 384,000 bps into 289,226 bytes. Conversely, the 448 kbps DVD ceiling and the 640 kbps AC-3 ceiling are absent from the dropdown and can only be reached by typing them into Custom Bitrate.
| Delivery target | Rate to set | How to set it |
|---|---|---|
| Stereo over S/PDIF to an AV receiver | 192 kbps | Constant Bitrate dropdown |
| Stereo, comfortable headroom | 256 kbps | Constant Bitrate dropdown |
| 5.1 for DVD-Video authoring | 448 kbps — the DVD-Video maximum and the rate commercial discs use | Type 448 into Custom Bitrate |
| 5.1 over an optical or coaxial S/PDIF link | 448 kbps — optical tops out there regardless of what the codec allows | Type 448 into Custom Bitrate |
| 5.1 over HDMI or on Blu-ray | 640 kbps — the AC-3 ceiling in ATSC A/52 | Type 640 into Custom Bitrate |
| Speech-only mono, small file | 96-128 kbps | Constant Bitrate dropdown |
No rung is marked as selected in this page's option set, so the dropdown presents its first entry — Highest. On AC-3, which has no variable-bitrate mode, the preset is converted into a constant bitrate derived from the preset level, the channel count and the sample rate, then held inside 192-640 kbps. At Highest with a stereo 44.1 kHz source, that arithmetic lands on the ceiling: 640 kbps, which is 4.8 MB of audio per minute. Encoding a 6-second stereo source at 640 kbps produces a 482,046-byte file — for six seconds of audio. If you have not chosen a rate on purpose, you are probably shipping a much bigger file than you needed.
AC-3 defines exactly three sample rates — 32, 44.1 and 48 kHz — and our validation layer snaps whatever you choose to the nearest legal one before the encoder ever sees it. The 8000, 12000, 16000 and 24000 HZ entries come from the shared audio panel used across every audio target on the site, and all four are nearer to 32000 than to 44100, so all four land on 32 kHz. That keeps the output a valid AC-3 stream instead of failing the job. If your source is 96 kHz or 192 kHz it is resampled down the same way, normally to 48 kHz.
Only if the ASF already contains six discrete channels. AC-3 can carry up to 5.1 here, and leaving Audio Channel on ORIGINAL preserves whatever the source held, but no encoder invents rear or LFE channels that were never recorded. Most .asf files are stereo WMA, and a stereo source produces a 2.0 AC-3 stream no matter what bitrate you set. Check the source first — VLC shows the channel count under Tools then Codec Information.
Nothing — the job fails rather than producing a silent file. We ran the negative control: feeding a video-only .asf into this converter produces the error "Output file does not contain any stream" and no file is written. That is the correct behaviour for an audio-only target, since an AC-3 file with no audio would be meaningless, but it does mean a failure here is usually a diagnosis rather than a fault. Open the ASF in a player and confirm it has an audio track before retrying.
No. An .ac3 is a raw elementary bitstream, not a tagged container — there is no ID3 or ASF-descriptor equivalent inside it, so titles, artists, album art and any DRM-free descriptors the ASF carried are all dropped. This is normal for the format's intended use, where the AC-3 becomes an audio track inside a disc image or a multiplexed video file and the metadata lives at the container level. If you need tags to survive, ASF to MP3 keeps them.
Because Dolby encoders write a dialnorm loudness value into the bitstream and compliant decoders — AV receivers, televisions, set-top boxes — attenuate playback to a reference level accordingly. The samples themselves are not being turned down; the decoder is normalising. Software players that ignore dialnorm will play the same file louder. It is not a fault in the conversion and there is no setting here that changes it.
No, and it cannot be. The audio inside an .asf is normally WMA, which is already a lossy perceptual codec, and AC-3 is a second lossy perceptual codec with a different psychoacoustic model. The server decodes the WMA to PCM and re-encodes it as AC-3, so this is a genuine second generation of loss — there is no stream copy on this route. At 448 kbps for surround or 256 kbps for stereo the damage is very hard to hear on home-theatre equipment, but keep the ASF if it is your only copy.
It is deliberately hidden on AC-3 output. AC-3 is a fixed-bitrate codec with a discrete rate ladder, so the output size is simply bitrate multiplied by duration and nothing else — a percentage target would have no mechanism to hit. That also makes the size completely predictable: 192 kbps is 1.44 MB per minute, 448 kbps is 3.36 MB per minute, and 640 kbps is 4.8 MB per minute. Use Constant Bitrate or Custom Bitrate and do the arithmetic; if you need a shorter file rather than a smaller one, use Trim.
There is no fixed per-file cap — what limits you in practice is upload size and connection speed, and since the video track is discarded on the server the size that matters is the whole ASF, not just its audio. Files travel over an encrypted connection, are converted on our servers, and both the upload and the result are deleted automatically after a few hours. Nothing is shared or made public and no account is required. If you only need a section of a long recording, set a Start time and a Duration under Trim so the output is cut before you download it.