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5.1 surround sound on Despecialized Editions?

Anyways, I think using the Dolby Pro Logic IIx 6.1 decoder will yield better results even for mono surrounds than Dolby Pro Logic II 5.1 or Dolby Pro Logic 4.0.

Not necessarily. Decoders/upscalers decode to whatever output config you select.

The upscalers have NO KNOWLEDGE of the config of the source material, all they get is 2ch something. And they WILL upscale to any number of speakers you select.

DPL2x is 6.1 OR 7.1 (Again, whatever you select). DPL2x will (generally) NOT increase the discreteness of your upscale by a large amount, because there is only so much info you can store in (matrixed) 2ch (in fact, the most sortove discrete ch you can store in 2ch is 3ch with the best DPL2 decoding, you can only get more with extremely controlled synthetic conditions.). Using professional 5.1>7.1 upmixers, what’s in the back 4 is basically the same as the previous 2.

You can>: decode to 5.1 (DPL2) and mono the rears. OR decode to 4ch (DPL1) and stereoize the back.

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5.1 surround sound on Despecialized Editions?

Moth3r said:

That said, when listening to hairy_hen’s purist mix for Harmy’s despecialized ROTJ, I noticed a significant amount of bleeding of dialogue from the centre channel into the front left and rights. I say significant because, if I play the 1993 mix through my receiver in Pro Logic II mode, this bleed is much reduced. I thought that the 5.1 mix was created from a Pro Logic II upmix in software, so this is surprising.

Most FREE DPL decoders in software are of the passive consumer non-steering type :dolby surround: , which are inferior to the PRO PL1 (with primitive allband steering), & obviously all inferior to PL2. I’m guessing he used PL1/DS things.

As of now, there are FREE PL2 things, only one I know of is freesurround (the foobar implementation being easiest to use imo), which should have much less C ch bleed. (hint, use old defaults preset for maximum discreteness)

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DTS audio preservation .... UPDATE 07 May 2015 ... Work In Progress

Turisu said:

I’ve now had the chance to sample the Matrix theatrical DTS and compare it to the BR audio. To me the two sound very similar with any perceived difference possibly being due to levels as hairy_hen mentioned.

That being said, there’s something about knowing I’m listening to the original theatrical audio that gives me a warm fuzzy feeling and an outstanding job has been done syncing it to the BR so this will remain my preferred track for watching the movie. Many thanks Jetrell Fo for making this available. 😃

Seeing how the BD & CINEMA tracks are ± identical, and keeping in mind cinema DTS is presumably worse than consumer DTS, which is about half the quality as AC3 at the same bitrate, it is incredibly daft to listen to the cinema track when BD AC3 is available because you’re listening to much reduced quality for no benefit.

Oh, and thanks hairy_hen for explanation/comparison.

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Audio Isolation Using Per-Sample (or near per sample) Mode Averaging

TylerDurden389 said:

I’m wondering if this will help me with my edit Of Terminator 2. I need to isolate and extract all the dialogue, as it’s the only original audio my edit will have. All other sounds are being rebuilt from the ground up. Gunshots, explosions, the T-1000’s noises, ambience, and of course, the score.

No it won’t. You won’t have a 2ry audio source. If T2 is in surround, you can try fiddling with the center chan, if it isn’t satisfactory or 2ch, try dialogue isolate in izotope Rx6.

Audio Isolation Using Per-Sample (or near per sample) Mode Averaging

My 2017 reply.

My thought on this project is based on the “My Neighbor Totoro” original english DUB - I’d love to hear the vocal talent from that release grafted onto the higher quality soundtrack backing the Japanese and Disney DUB. Of course, this only even theoretically works if the audio is in perfect sync.


Usually when you combine audio or video, you use a regular mean average. Sometimes when you’re capturing video and want to remove artifacts, you use a median average that eliminates outliers. What you don’t see a lot is MODE averaging, where the most commonly occurring pixel or sample is used.


I’m proposing a potential breakthrough in movies with dubbing at least, using a MODE average process to compare and combine audio tracks on a Per-Sample basis (or 1ms basis… just something very small).

No, mode is probably not going to work because probably not perfectly sync. YOu can try, but you will likely not get what you want.

If you wanted to do something roughly using average modes, it wouldn’t work in the time domain, you’d need to use FFT. And it would be eaxsiet to use 2 languages only, to do a ‘minimum of’ bins in the frequency domain, instead of mode. This tool does not exist yet. Lets’ take a look again at the motive:

I’d love to hear the vocal talent from that release grafted onto the higher quality soundtrack backing the Japanese and Disney DUB.

Okay. So if this is ur motive, you can do center channel removal of a stereo track, and then place it behind the mono EN dub.

Or, take the side channel of any stereo track, and then place it behind the mono EN dub.

Drawbacks of both methods: may be too much or too little content, and unstable imaging.

NB: I haven’t tried any of these things btw.

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Good Hi-Fi VHS vs. 192kbps AC3?

VHS is TAPE< and so that’ll have a gentle rolloff staring from ~12k down to 18.

DVD 192kps has a ~18kco.

All tings equal, I’d say the AC3 should sound better, assuming that the Ac3 has the same original source as the tape.

NOW, to do EQUAL comparison, make sure that the VHS & DVD are the SAME VOLUME.

m2ts to mkv

A few things: By editing the AC3 audio in AUdacity, AUdacity can only reencode, so you’ll take a quality hit. For lossless editing, you can try mkvtoolnix or delaycut to add silence & cut etc.

AUdacity cannot make DTS. try Surcode or DTS master audio suite.

some audio advice please

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.

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On MAC: Convert DTS-HD to wav/flac

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

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.

File sizes and codec help

Is the 22gb the same res as the 8gb? If so, the Lags output might be similar at around 70-80GB. Don’t bother with encoding that down and then reencoding that with Lags, as the final filesize will probably be the same, UNLESS you apply some DSP such as: denoising, colour reduction, which makes the frames easier to predict.

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