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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.

Problems with archiving in JDownloader 2.0

The USB stick may be FAT32. FAT32 drives cannot hold >4GB files irregardless of free space.

If both are true, then your USB stick cannot hold the AVCHD ISO. Try Extract the AVCHD ISO to your PC HD, which should be NTFS (which CAN hold >4GB files).

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

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.

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Working in Premiere Pro and After Effects

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


General Encoding Question from Projects - Scripters opinions wanted.

I think what you want is x264 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

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Restoration tips: HD matrix surround™

     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.


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 , 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).



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.

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tsMuxer and MKVToolNix not working with FLAC?

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.


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.


TServo2049 said:

The sound mix is a major problem with the Blu-Ray. It's been processed to within an inch of its life; the tape hiss removal sounds to have taken a lot of the ambiance and echo with it. I've also read complaints that the directionality was messed up.


The dynamic panning was in the front for the DVD. It's now at the sides/back for the BD. If you do a custom downmix, it would sound about the same, surprisingly (minus the phantom/physical center issues).

EDIT: I'm almost sure, The 2000 DVD, in addition to the reverb that was already there, they added a bit more. The BD they didn't add reverb - the rears sound direct.

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Convert DTS-HD MA to WAV on Mac?

There are many crap to fudge converters lying around.

You should be able to find tools to decode the lossy part of DTSHD for mac. Maybe FFMPEG &/ GUIs.

The only thing that can possibly decode the lossless part of DTSHD for mac would be if it is able to be built for mac, or the commercial DTS quicktime component).

Help with Adobe Premiere Pro Audio

With Premiere CC, you may be able to do basic syncing with multichannel audio. PPRO treats mch WAV files as n x mono (grouped) audio clips. Assuming you have a 6-8ch wav, make a new sequence with "multichannel" 6 or 8 for the audio master.

Then assign your audio channels to the appropriate pairs (1+2, 3+4) in the audio mixer. Don't for get to pan the tracks L & R alternately.

Group the clip with the video & Make your cuts.

Monitoring will be hard.

You should then be able to export in your desired format (SR, bits, #ch).

For 6ch/5.1, monitoring & assigning the channels will be easier.


PPRO's actually bad support for editing files already in 6+ch is bad, which is why I don't use it unless I really have to.

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Hello & thank you Poita & others for sourcing and helping with Fantasia.

I'm interested in Disney audio because recent BD releases leave something to be desired, especially fantasia. I was alerted to this thread by an existing member.

As you know, Fantasound is really 3ch + control tones (possibly for panning & dynamics?). How do you plan on applying that to the captured print or is that already taken care of?

Is there a predetermined audio capture format that you're planning on using/have used (samplerate/depth/channels)?

Would it be possible to upload semi/raw audio, perhaps after panning & dynamics (& sync) but before denoising (& EQ) so that me &/ others can tinker with the audio?

Are there any other changes (perhaps according to audio documents?) that should be applied to the audio to sound proper (ie, EQ curves, ?)


I've had a look at FNT audio from the BDs, and it's relatively easy to make it sound somewhat like the 2000 DVDs by simple channel swapping/mixing, etc.

The phonerip has a gentle roll-off after 8k, and the strings sound strange, which I'm hoping this mag cap will better.

The '89 version by Irwin Kostal (CD) somehow won't sync to the BD, even though it was apparently targeted for theatrical & home release.

I capped a 'selections from fantasia' LP ('70), which ofc won't sync.


I don't have much BW per month so sharing my changes as HQ audio files may be difficult. I can upload small files, though.

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