PC Based SQ DECODING - ALMOST DONE!

[mandel]

I've run winopeners script over one of my 24/96 SQ vinyl recordings. The result is very nice and I am left with a huge 6 channel wave file. Firstly this has two empty streams, and I can't see in Audition how to reduce it to a 4 channel wave file. Secondly FLAC reports "ERROR: unsupported compression type 65534" when I try to encode it. FLAC should support multichannel waves so not sure why this is. Any advice welcomed.

 

[winopener]

I've run winopeners script over one of my 24/96 SQ vinyl recordings. The result is very nice and I am left with a huge 6 channel wave file. Firstly this has two empty streams, and I can't see in Audition how to reduce it to a 4 channel wave file. Secondly FLAC reports "ERROR: unsupported compression type 65534" when I try to encode it. FLAC should support multichannel waves so not sure why this is. Any advice welcomed.

Audition CANNOT save such a file, just open it and then save it as as sequence of mono files. So, better to have 4 single mono files and get rid of the 6channel.
Don't know anything about flac.

 

[mandel]

Ta. So then I need something else to convert the 4 mono wavs. Also, like Armyofquad I am using 96khz files so need to change SR 44100. Do I also need to change all the lines like "10: 22050"? (looks like half sample rate to me).

Sorry for all the questions, you've created a godsend :banana:

 

[neil wilkes]

That error of 65534 in FLAC looks suspiciously close to a 16 bit file's amount of possible amplitude levels.....
Perhaps FLAC does not like 24/96 files and is looking for 16/44.1..???

 

[mandel]

FLAC is fine with 24/96 files. I am in the process of archiving all my stereo vinyl as 24/96 FLACs.

I stupidly only posted half the error message... I lose at the internet :slap:
It also says:
WARNING: found non-standard 'fmt ' sub-chunk which has length = 40

I suspect this is because Adobe Audition is using Extensible Wave format and FLAC doesn't like it (though FLAC does support multichannel files somehow). Ideally I want another piece of software that can turn the 4 mono wave files (dropping the empty centre and LFE ones) into a quad wave file. No look finding anything yet though...

 

[imploder]

FLAC is fine with 24/96 files. I am in the process of archiving all my stereo vinyl as 24/96 FLACs.

I stupidly only posted half the error message... I lose at the internet :slap:
It also says:
WARNING: found non-standard 'fmt ' sub-chunk which has length = 40

I suspect this is because Adobe Audition is using Extensible Wave format and FLAC doesn't like it (though FLAC does support multichannel files somehow). Ideally I want another piece of software that can turn the 4 mono wave files (dropping the empty centre and LFE ones) into a quad wave file. No look finding anything yet though...

I use Steinberg Nuendo for that. It can handle multichannel wave files and redirect the channels to an ASIO compatible sound card. ASIO4ALL works good to make some onboard device compatible...

The promised homepage is more work than I thought. Just some more days...

Imploder

 

[mandel]

Ta. So then I need something else to convert the 4 mono wavs. Also, like Armyofquad I am using 96khz files so need to change SR 44100. Do I also need to change all the lines like "10: 22050" to "10: 48000"? (looks like half sample rate to me).

Hate to pester you again, wonder if you missed this last time I asked? Just want to make sure I'm doing it right before I start archiving stuff. Thanks

Edit: No worries, figured where this comes from after playing about a bit.

 

[Lucanu]

@winopener

I tried at last your script for decode sq material I own and I must say it's amazing.
Now the problem is...

Can you post a script for qs decoding as well?

 

[Dylan Berichon]

I've played with the settings a bit, and I can get some fairly decent results.

Is there a way to enhance front to back separation? The back channels seem to separate out a lot better from the front, then vise versa.

 

[BananaSlug]

I have a question for those of you who understand this process better than I. I have been using Winopener's script and getting good results from SQ cd sources. So good, in fact, that I had my parents dig out my old turntable and bring it up on their last visit. So, now I'm working with vinyl. After running the script I have been normalizing as opposed to plain raising the .wav volume. I am concerned that this may raise some the volume disporportionately. Say the rears were mixed to be lower and I normalized the fronts and backs to 0 db - does this seem like a legitimate concern? Or is it sort of personal preference (people change the setting on their speakers at the receiver to suit their setup), or could there be some other negative consequences? I like the idea of normalizing - it's easy and requires no thinking on my part (except for all this thinking regarding the possible problems). Maybe I should just relax ( ) and have a listen to what I've accomplished so far...

There certainly is an allure to the mechanics of vinyl, but I really wish the record company bastards would just do the work for me and put out these mixes on SACD/DVD-A. I will get the money and buy the discs, I promise!!!

 

[Sonik Wiz]

So good, in fact, that I had my parents dig out my old turntable and bring it up on their last visit. So, now I'm working with vinyl. After running the script I have been normalizing as opposed to plain raising the .wav volume. I am concerned that this may raise some the volume disporportionately. Say the rears were mixed to be lower and I normalized the fronts and backs to 0 db - does this seem like a legitimate concern?

You are correct, it would alter the level away from what was intended. Try this approach:

Analyze the stats for both front & rear. Let's say front has the loudest peak but is still at -5dB (or whatever). And say you would usually normalize to -1dB peak level. Now just amplify both front and rear by 4dB & you've got the peak level where you want it and maintained the original balance.

So, what discs are you working on?

See Ya Bye!
R. Scott

 

[BananaSlug]

I had some fun with ebay and picked up Herbie Hancock Sextant and Thrust as well as Omega - 200 years after the War (read on a web site that the quad mix was quite good, though I'm not familiar with the band (http://www.iyte.edu.tr/~thomasbechteler/quad-rating.html)). Thanks for the tip, that does make more sense - I could just feel myself starting to think in circles. I've saved my work along the way, so it shouldn't be too much of an issue to correct.

 

[Lucanu]

I think normalize is a good thing, but not always.
Let's say you have a peak due to a crack that you can't recover with Audition. The result would be a "false" normalize between fronts and rears.
It's best that you apply an amplify function differently to front and back channels.
Look in the graphic of your samples, see at what value is the peak (i.e. -6 dB), then apply the amplification to reach -1 dB.
For the rears, it usually sounds quiter than the front, so you need to add 3dB to the values used for the front channels.
3db is the minimum value your ears could notice on a change of volume in sound. If you apply less than 3dB you couldn't notice nothing changing at all.

Another thing, for SQ decoding, it was said initially that the front channels (once decoded) should have been amplified 3dB more than the rear channels, so maybe you don't need this trick if you think that the balance is good enough.

Instead, in Qs decoding script what you get is the volume levels that were REALLY intended to be hear, but if you need to gain fronts or rears it s your choice

Bye

 

[BananaSlug]

I think normalize is a good thing, but not always.
Let's say you have a peak due to a crack that you can't recover with Audition. The result would be a "false" normalize between fronts and rears.
It's best that you apply an amplify function differently to front and back channels.

I had thought of that, but I manually fixed the large pops so I hoped that wouldn't be a problem. Winopener said he raised the fronts only by 3 db. I had thought that was a personal preference, but it may be to compensate for the decoding process. When I checked the peak levels I found that the rears were louder by 1 to 1.5 db. Since we aren't sensitive to anything less than 3 db I guess it's not something I should worry too much about. Thanks for all the input. I will raise the levels so that the peaks are just under 0 db (maybe a little lower in the rears) and then have a listen and find what works best for my system.

 

[Dylan Berichon]

I've played with the settings a bit, and I can get some fairly decent results.

Is there a way to enhance front to back separation? The back channels seem to separate out a lot better from the front, then vise versa.

I decided to do a bit of experimenting, and so far I've found that bumping up the Amplitude Band Width in the Center Channel extraction tool has helped increase seperation considerably.

It can make things a bit muddy if you go to far with it though.

 

[Dylan Berichon]

Has anyone here gotten a hold of the new version of Audition and tried this?

Is the Center Channel Extraction filter any better?

 

[Bob Romano]

To my ears, the new Audition CCE is smoother and sounds cleaner but I havent tried it really extensively.

 

[Bob Romano]

I am trying to use the script for SQ using Audition 2.0 I am having a problem. Whgen I start the script, it stops immediately. I have the 3 wavs made (all are 96/32float). The folder where the waves are is indeed
F:\sqdata\

I have made all the changes to the script to point to that folder. So what am I missing? Does the script work with 96/32 files?



Collection: SQdecoding_7_AK
Title: _ SQ_enc_1_f:\SQData\ _96000_
Description:
Mode: 1
Undo: 0
cmd: Channel Both
cmd: Command 1100
1: f:\SQData\temp.wav
2:
Selected: 0 to 518400000 SR 96000
Freq: Off
cmd: Channel Left
Selected: 0 to 518400000 SR 96000
Freq: Off
Comment: Filters\Graphic Phase Shifter
cmd: {FC0778B7-D686-4B7C-B40F-5F112504D8CC}
1: 262144
2: 2
3: 0
4: 900
5: 4096
6: 900
7: 1
8: 0
9: -180
10: 180
11: 3
Freq: Off
cmd: Channel Right
Freq: Off
Comment: Filters\Graphic Phase Shifter
cmd: {FC0778B7-D686-4B7C-B40F-5F112504D8CC}
1: 262144
2: 2
3: 0
4: 2700
5: 4096
6: 2700
7: 1
8: 0
9: -180
10: 180
11: 3
Freq: Off
cmd: Channel Both
Freq: Off
cmd: Copy
1: 0
Freq: Off
cmd: Command 1100
1: f:\SQData\rear.wav
2:
Freq: Off
Comment: Amplitude\Channel Mixer
cmd: {4EB62A8D-32F7-4515-8C90-919A17A50EE5}
1: 0
2: 1
3: 1
4: 0
5: 1
6: 0
7: 0
Freq: Off
cmd: Paste Special
1: 1
2: 1
3: 3
4: 1
5: 0
6: 0
7:
8: 0
9: 0
Freq: Off
Comment: Filters\Center Channel Extractor
cmd: {2A0B716A-4C11-4871-8C81-36BAB58A47B6}
1: 1
2: 20480
3: 0
4: 16
5: 6
6: 2
7: -48
8: 90
9: 0
10: 22050
11: 180
12: 1124
13: 8
14: 4
15: 52
16: 0
17: 0
18: 0
Freq: Off
Comment: Filters\Center Channel Extractor
cmd: {2A0B716A-4C11-4871-8C81-36BAB58A47B6}
1: 1
2: 20480
3: 0
4: 16
5: 6
6: 2
7: -48
8: 90
9: 0
10: 22050
11: 90
12: 1124
13: 8
14: 4
15: 52
16: 0
17: 0
18: 0
Freq: Off
Comment: Filters\Center Channel Extractor
cmd: {2A0B716A-4C11-4871-8C81-36BAB58A47B6}
1: 1
2: 20480
3: 0
4: 16
5: 6
6: 2
7: -48
8: 90
9: 0
10: 22050
11: 270
12: 1124
13: 8
14: 4
15: 52
16: 0
17: 0
18: 0
Freq: Off
cmd: Command 1100
1: f:\SQData\front.wav
2:
Freq: Off
Comment: Filters\Center Channel Extractor
cmd: {2A0B716A-4C11-4871-8C81-36BAB58A47B6}
1: 1
2: 20480
3: 0
4: 16
5: 6
6: 2
7: -48
8: 90
9: 0
10: 22050
11: 180
12: 1124
13: 8
14: 4
15: 52
16: 0
17: 0
18: 0
Freq: Off
Comment: Filters\Center Channel Extractor
cmd: {2A0B716A-4C11-4871-8C81-36BAB58A47B6}
1: 1
2: 20480
3: 0
4: 16
5: 6
6: 2
7: -48
8: 90
9: 0
10: 22050
11: 90
12: 1124
13: 8
14: 4
15: 52
16: 0
17: 0
18: 0
Freq: Off
Comment: Filters\Center Channel Extractor
cmd: {2A0B716A-4C11-4871-8C81-36BAB58A47B6}
1: 1
2: 20480
3: 0
4: 16
5: 6
6: 2
7: -48
8: 90
9: 0
10: 22050
11: 270
12: 1124
13: 8
14: 4
15: 52
16: 0
17: 0
18: 0
Freq: Off
Comment: Amplitude\Amplify/Fade
cmd: {03EA5F5A-8046-4D8F-95E0-4387A5A4289D}
1: 1.4126
2: 1.4126
3: 1.4126
4: 1.4126
5: 1
6: 0
7: 0
8: 0
9: 1
10: 0
11: 0
12: 0
13: 0
14: 0
Freq: Off
End:

 

[Dylan Berichon]

I am trying to use the script for SQ using Audition 2.0 I am having a problem. Whgen I start the script, it stops immediately. I have the 3 wavs made (all are 96/32float). The folder where the waves are is indeed
F:\sqdata\

I have made all the changes to the script to point to that folder. So what am I missing? Does the script work with 96/32 files?

What happens if you use your modified script in Audition 1.5?

I haven't gotten a chance to check out the new version of Audition yet, so I don't know if it might just be a script incompatibility between versions.

It shouldn't matter what the sample or bit rate is at. It's never mattered for me with Audition 1.5.

 

[Bob Romano]

I believe that it is an incompatibility between this script, 1.5 and 2.0. It worked fine with 1.5.