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junh1024

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Join date
2-Jun-2015
Last activity
16-Jun-2018
Posts
55

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Post
#1008357
Topic
some audio advice please
Time

There’s a lot being said, i hope I’m interpreting it right.

OTT84 said:

i used eac3to to get the wavs to work in vegas

THD & DTS-HD MA are lossless.

Using eac3to to decode THD to WAV is fine. The newest eac3to versions can decode THD & DTS-HD MA losslessly.

and then used surcode to complie to dts

however ive recently been made aware this is a lossy audio and indeed media info says so

DTS (core) itself is lossy.

i read i can use dts master suite to take these wavs from the truehd source and end up with a “better sounding” lossless file and also extract the core, though tbh i dont havea clue what that means and if that’s what i need to do?

Core is always lossy.

i looked at my original audio info and when i re extracted the wavs from truehd today as originals were delted after for space, i notced one major difference the new wavs were 768kbps and only 16bit, my finished film had a higher bitrate of 1510kbps and 24bit

full-rate lossy DTS (core) is 1510kps. Since it’s a lossy codec, exact bitness is somewhat meaningless.

i remembered i didnt do a straight conversion on my prjoect as i htink i noticed this before, i went truehd > eac3to dts @24 > eac3to wavs > vegas > surcode

This is bad, because if you encoded DTS before making it WAV for vegas, DTS (core) is lossy, and you’re editing lossy audio in vegas. Best to go direct from THD to WAV (lossless.)

& As said before, using surcode to make DTS (core), DTS (core) itself is lossy.

i’m concidering using the the new waves to update the audio but i find it odd that the original high bitrate of them is reduced signifiactly, but when renderd through dts-ma suite they do show now as lossless / lossy and unknown / 768 bitrate in media info,

DTSHDMA necessarily includes a DTS core, which can usually range from 752-1510 for surroud

but in the first info page of properties in mpc it shows a bitrate of 4k which is over the original 3k of the truehd files?

When you went from THD --> DTS, this is a lossy process, and the decoder has no knowledge of the original lossless 16bit source, and decoded it to 24bits, and now is bigger. Additionally, you may’ve exported 24bits in vegas.

i want to know if using waves like this is the right way to go about it as they have to go into vegas for the editiding of course, and is any detail lost when i save them individually again after the cuts?

GO straight from THD to WAV using eac3to for vegas for best quality. Export 16bit using vegas.

and from there which settings in dts ma suite should i be using with the edited wavs, i matched all the setting

Using DTSHDMA, and 768/752kps for core is fine.

THD is more inefficient than DTSHDMA for the same input, so DTSHDMA will be smaller.

Post
#955291
Topic
Question About Music Removal
Time

Prepare an intermeriatary stereo file with channels:
L: has soundtrack
R: has music synced to soundtrack

THEN all you nneed is an FFT imager on that intermediatary.

Try Audition’s Center Chnnel Extractor, or try adding QuikQuak Mashtactic to Audition for a moore visual approach.

EDIT: Also maybe http://blog.wavosaur.com/extract-vocals-from-song-with-kn0ck0ut-vst/

Post
#951089
Topic
On MAC: Convert DTS-HD to wav/flac
Time

A few things.

Depending on how old your FFMPEG is , you may actually be decoding the LOSSY DTS core part of DTSHD. DTSHD decoding via dcadec was only very recently added to FFMPEG (mid 2015).

MakeMKV is updated regularly, so as long as you have a recent version of MakeMKV AND the makers of MakeMKV have decided to use a recent FFMPEG, you MIGHT be ok.

I would assume that MakeMKV/FFMPEG decodes to the bitdepth that the lossless part of that particular DTSHD stream specifies. DTSHD MA can be 16, 24, or any number of bits in between.

I would also argue that 16 bit is OK for listening, and even for editing, and dither is bad. It’s:

Potentially harmful
Mostly useless
Self-defeating

For converting to 16bits, I highly reccomend truncating the bits (no dither).

At sensible volumes, you won’t need dither until you get to approx. 12bits.

Post
#792027
Topic
Audio Isolation Using Per-Sample (or near per sample) Mode Averaging
Time

It seems that remixavier uses stretching and FFT (which i can do with a combination of tools, see below)

thanks obi-juan for suggesting Remixavier. There's also utagoerip which is FFT-only. I'll mention other FFT-only tools below.

You can Forget about all this "Audio Isolation Using Per-Sample (or near per sample) Mode Averaging" (unless you stretch the audio, and that by itself is successful). Any audio that is not in perfect sync will not polarity inverse cleanly, this is where FFT-based tools come in handy.

RE FFT & sources, Other language tracks are less useful unless they align 100% (or if they are surround and have mostly isolated music in some channels). You may need to combine FFT with polarity inversion to get BGM if you have stereo only.

If you really have only 2 languages  in stereo and they don't align (and you don't treat it because it's generally hard to fix), you will get poor results , because 2 sides are pulling the same vocal frequencies. See pic for an example when we want to keep the music only, but we only have 2 voiced sources.

If you want to break through and get stuff done (and know how to use an actual DAW instead of audacity), I will suggest tools like REAPER + Quikquak mashtactic or R-Mix (FFT based imager) to remove music/sfx.

I have completed a project where I mostly removed the music from a mono version of the Animax card captor sakura dub, and replaced it with my own fixed surround version of the BGM derived from the JP 5.1 track [using the tools i mentioned above. You can probably do this with kn0ckout VST (or other FFT) in audiacity but it would be at least 4x slower & more finicky].

I'm in the middle of a similar project replacing the mono bgm of the Streamline Kiki's Delivery svc dub with surround bgm. But I don't really have that much time nowdays.

Post
#785783
Topic
Working in Premiere Pro and After Effects
Time

Is the bottleneck your CPU or HD? Check in resource monitor.

If it's your HD, reencode your file to make it smaller.

If it's your CPU, try upgrading to PPRO CS6+ which includes mercury engine & enable it https://www.google.com/search?q=premiere%2Bmercury%2Bengine&ie=utf-8&oe=utf-8

or

https://www.google.com/search?q=premiere%2Bmercury%2Bengine+on+windows&ie=utf-8&oe=utf-8

Post
#781383
Topic
General Encoding Question from Projects - Scripters opinions wanted.
Time

I think what you want is x264 ZONES.

https://en.wikibooks.org/wiki/MeGUI/x264_Settings#zones

Where you set an overall CRF normally, and then specify another CRF for a zone.

===

Yes you can encode __MAIN using CRF and then the credits separately using 2pass, both with the --stitchable option, and join using MKVnix. Any acknowledgement would be purely up to the user. Or you can write a script to try & try again to match an overall size. I think this is retarded and not worth pursuing. After all, __MAIN is the bulk of your filesize, not the creds. Any dealbreaker for filesize typically would have long passed before the credits.

===

1pass CRF is the highest quality mode x264 has to offer, better than 2pass, because the encoder is not concerned about filesize issues.

===

you can use x262 for encoding MPEG2, and it apparently supports zones

https://www.videolan.org/developers/x262.html

Post
#779326
Topic
Restoration tips: HD matrix surround™
Time

     What is HD matrix surround™?
    It's a discrete multichannel track, originally matrixed in a stereo sound, and then hardware decoded (hence the HD) using the best, and rarer, matrix hardware decoder.

It is matrixed, and therefore not discrete.

Matrix encoding does not allow one to encode several channels in fewer channels without losing information: one cannot fit 5 channels into 2 (or even 3 into 2) without losing information, as this loses dimensions: the decoded signals are not independent. The idea is rather to encode something that will both be an acceptable approximation of the surround sound when decoded, and acceptable (or even superior) stereo.

via https://en.wikipedia.org/wiki/Matrix_decoder

But will it sound better than, for example, Dolby Pro Logic II?
Albeit DPL II is a great matrix surround decoder, some older technology *may* sounds better, or in a different way... at the end:

Reading the info at https://en.wikipedia.org/wiki/Dolby_Pro_Logic , I can conclude it will ONLY sound better (or more discrete) than DPL2 if :

(It was thouroughly tested and checked on a DPL1 system to make sure that the artefacts and intent sound acceptable OR no discrete surround mix exists) AND (Only during section when the steering/gating system is active)

I am assuming that the decoder is single-band. A multiband decoder will go somewhat towards DPL2

When the gating system is not active (ie, cannot decide btw M/S and L/R dominance), a DPL2 decoder would do better as it can steer effectively 128 to 1024 (or more) bands depending on the FFT size.

there are some interesting plugins for popular software like Foobar2000, that decode somehow a matrixed track;..., but it *seems* hardware decoders produce decoded tracks of better quality.

Software decoders are typically free, you can make one fairly quickly (based on consumer DPL1), but would not sound goot/discrete. Some Proprietary software decoders will sound better than the free ones, because they put in more effort to make it sound goot (FFT/active matrix, etc). FFT takes much more effort to figure out what is right, and how to do it, etc.

I'll quote DTS Neural upmix (used FFT) as one of the best surround upscalers. It is proprietary (&paid), but easily available. It works as well upscaling normal stereo as on matrixed stuff. I also typically put on additional DSP to process the back to move transients back to the front, to give more discreteness (QQ Mashtactic).

===

E3:

I beleive it is possible to make a DPL1 Pro decoder in software, but I have never come across documentation comprehensive enough to determine the gating thresholds/etc. OTOH, I can probably figure out the thresholds for DPL2 decoding because I know  FFT & DTSnu well enough.

Post
#775247
Topic
tsMuxer and MKVToolNix not working with FLAC?
Time

FLAC is not in the BD spec, so...

1) Extract the FLAC using MKVEXtractGUI2

2) Convert the FLAC to WAV/(L)PCM or DTS/AC3. Depending on the length of the files, it may be over the WAV 4GB limit. Reduce bitdepth/channels/don't use PCM/use a tool that ignores the header length.

3) Remux & author.

Issues===

Depending on if you want to join the mkvs, i'm not aware that tsmuxer can mke a BDMV from 2 mkvs unless they're joined. If you want to join them, make sure they're exactly the same codec & settings (and even then it may not work).

Joining 2 FLACs via mkv and then demuxing that will yield a broken file. convert to another codec 1st.

We assume your mkvs' H264 track conforms to BD profile & settings. If not, you need to convert that too.