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14th November 2015, 12:41 | #13641 | Link | |
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The clipping which occurs in the "correct" result is probably already contained in the mix or mastering, isn't it? @all: since the most recent version of MakeMKV produces the same faulty results, I tried to contact "mike" which seems to be the developer or at least part of the team. Stupidly, I can't register at that forum since my IP range is blocked for some reason which is beyond me. Just in case, anyone knows if and how to enforce MakeMKV to enforce to still use the dtsdecoder.dll instead of the integrated libDca. |
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14th November 2015, 13:07 | #13642 | Link | |
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From what I understand, madshi wants to offer a two-pass option in eac3to to reduce the volume and re-process the file if such clipping issues are detected. It wouldn't be bitexact to ArcSoft or the source, but probably be better than clipping!
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14th November 2015, 14:00 | #13643 | Link |
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I don't understand this.
This is with that required volume reduction in general. I'm aware of eac3to's two-pass-feature and understand the requirement for volume reduction depending on the source when downmixing a number of channels into fewer channels than the source since all channels could carry 0dBFS and have to fit into the lower range. What I never understood is why this is sometimes required when dealing with lossy codecs without downmixing as I don't see the point why I have to reduce the volume just to reconstruct the lossy source. Now with lossy codecs, you got transformation into the frequency domain and stuff and maybe due to errors introduced by the psychoacoustic model, you might get (intersample) peaks here and there asking for a few additional dB of headroom but why this shall still be an issue with lossless sources is beyond me. By definition, there can be only one correct PCM result and audio level for any given lossless source. If that clipped, so shall the PCM of course. When lowering the decoded audio level for a clipped lossless source, I would expect the decoded result to clip as well at -x dBFS then, gaining nothing but slightly rising the noise level due to the lower SNR. |
14th November 2015, 14:08 | #13644 | Link |
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Its probably just a mistake when encoding. Formats like DTS-HD are a bit more complex than a simple codec like FLAC, they have embedded downmixes for stereo and 5.1 and whatnot (and the decoder performs something called "downmix reversal" to get the 7.1 signal), so when all these features are used, its apparently easy enough to screw something up.
I'm sure there are also lossless encodes where the clipping is part of the original PCM, but in this particular sample it is possible to retrieve the audio without clipping, albeit at a slightly reduced volume to make room for the extra data.
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LAV Filters - open source ffmpeg based media splitter and decoders Last edited by nevcairiel; 14th November 2015 at 14:17. |
14th November 2015, 14:22 | #13645 | Link |
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just to confirm, the US Blu-ray of Ex-Machina has indeed a 7.1 DTS:X (+ a DTS:X headphone) track. so this is perhaps why anything else than just extracting the track (well at least if this works as it should) could potentially result in non lossless output as the reis probably no support yet for these kind of tracks via libDAC or arcsoft (unless there are recent updates for arcsoft)
just asking, does that mean that full DTS:X track processing is possible now with that version? or that just one specific error has been fixed?
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14th November 2015, 14:34 | #13646 | Link |
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There are no decoders for DTS:X outside of the hardware receivers.
Note that such things as "bitexact" don't really apply to concepts like DTS:X or Dolby Atmos, as they are designed to be mixed for the speaker setup of the user, so there is no one correct way to "decode" it.
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14th November 2015, 14:42 | #13647 | Link |
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so if you want to have the full Atmos and DTS:X information from an audio track, the only thing which makes sense is to extract the entire track and bitstream it to you receiver, correct?
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14th November 2015, 14:42 | #13648 | Link |
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@madshi
@tebasuna51 @nevcairiel Any comments on my TrueHD sample and AC3 conversion ? Is it a bug of v3.30 ?
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14th November 2015, 14:44 | #13649 | Link |
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Yes, you should always leave the track intact. Even if some day a software decoder appears, it'll still be the same deal - you need to tell it how your speakers are setup to mix the extra "3D" audio properly, so it should only do this conversion at playback time, and not when re-encoding.
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14th November 2015, 16:21 | #13650 | Link |
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eac3to v3.31 released
http://madshi.net/eac3to.zip Code:
* libDcaDec: updated to latest build * libDcaDec: decoding only aborts on critical issues now * libDcaDec: now reports warnings if something isn't 100% perfect * libDcaDec: proper handling of clipped files (2nd pass etc) * libDcaDec: proper handling of tracks that switch bitdepth 16 <-> 24 * fixed: TrueHD decoding -> AC3 encoding didn't work properly |
14th November 2015, 16:31 | #13651 | Link |
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whats the relation between the bitdepth switch fix and your re-opening of that topic at dcadec github?
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14th November 2015, 16:44 | #13654 | Link |
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Is dcadec now more accurate what comes to errors while decoding? I tested the new version on a stream which had no errors with v3.29, the new version reports "libDcaDec reported the warning "XLL output not lossless"." while decoding. I don't know if it's related to this, but in this case I have a DTS-HD MA track which is reported as 24 bits but contains only 16-bit data.
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14th November 2015, 16:53 | #13656 | Link | |
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Decoding of these parts didn't change, it now simply outputs a warning instead of silently ignoring such cases.
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14th November 2015, 16:57 | #13657 | Link |
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Tested with v3.30 as well, the same thing (it doesn't report anything).
If you (or the dcadec dev) want to see the source track, I can provide it. It's a stereo track so it doesn't require a huge amount of space. EDIT: just read nevcairiel's explanation. Makes perfect sense
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14th November 2015, 17:09 | #13658 | Link | |
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It works now.
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14th November 2015, 17:10 | #13659 | Link |
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can DTS:X (headphone) tracks be delayed? is there any way to implement atmos tracks being able to be delayed?
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