Strange flac issue - oddity (I think)

QuadraphonicQuad

Help Support QuadraphonicQuad:

This site may earn a commission from merchant affiliate links, including eBay, Amazon, and others.

JonUrban

Forum Curmudgeon
Staff member
Admin
Moderator
Since 2002/2003
Joined
Mar 2, 2002
Messages
17,724
Location
Connecticut
First off, let me just say that I am no expert in .flac files, tagging, and all this "new fangled stuff", but I am I believe smarter than the average old fart when it comes to this stuff. That being said:

I've been .flac'ing and tagging stuff for the car, and it's been going fairly smoothly. Thanks to Garry for updating the Music Helper program he wrote I no longer have to manually open every quad file in Sound Forge and add a blank center and LFE channel to make them compatible to the bizarre decoder in my Acura.

So, I decided to do a few Chicago Quadio's. I did Chicago (Second Album), VII, and IX. I figured that would be good stuff on my USB stick. So I already had the MKV files on my PC, so that saved me a step. I used AudioMuxer and got them easily to .flac. I then ran Garry's utility to add the empty C and LFE, and then I used Foobar to convert the 24/196 .flac files of unknown compression value to 24/48 zero level compression for the Acura. Not bad. Spent the time tagging them. Got the quad album graphics to put in there, and boom. Done.

WRONG! The car would not play them. Got the tags, got the quad LP covers, but no audio - file format not recognized. OK, I screwed up somewhere. Went back to SF, opened a file to look-see. Perfect 6 CH file, empty center and LFE, SF reports 24/48, no issue there. I actually had no known way to "see" the compression level, but I am sure I had that set to '0' in Foobar upon conversion because it's a saved setup. So what went wrong?

I tried everything but nothing worked. So I thought about everything else I did that worked. The only difference was that everything else I've converted was sourced from a .wav file. OK, I'll try that. Converted all of the .flac's to wav, then converted them back to .flac.

Now it worked. WTF???

I have no clue, but I did all of IX that way and now those files play. No way I'm doing II and VII today, AGAIN. Burned out, ya know?

Anyway, just another stupid story in the world of senior citizens playing with modern shit! :phones
 
WRONG! The car would not play them. Got the tags, got the quad LP covers, but no audio - file format not recognized. OK, I screwed up somewhere. Went back to SF, opened a file to look-see. Perfect 6 CH file, empty center and LFE, SF reports 24/48, no issue there. I actually had no known way to "see" the compression level, but I am sure I had that set to '0' in Foobar upon conversion because it's a saved setup. So what went wrong?

Is that the same thing that happened before you started encoding the FLACs at level 0? I can't remember if your original files were giving you an error message or just silently failing.

If it's exactly the same thing, I wonder if Foobar is doing something odd when it downconverts. Maybe there's a bug that crops up when you're trying to resample and create FLAC at the same time.

This is another case where having command line tools and the necessary amount of masochism to use them might help. For example, it might be useful to know what happens if you have FLAC test one of the problem files, e.g.,
Code:
 flac -t FileThatPissedMeOff.flac
Of course, we already know that you can have perfectly standard-compliant FLAC files that the Acura doesn't like, but this would at least rule out a major issue like encoding so wrong that FLAC itself complained. If it passed that test but still wouldn't play, you could just tell FLAC to re-encode it as a level 0 file just to see if it produced something different (and hopefully playable!).

I know I'm always ranting about obscure command line stuff, but it's actually because I'm lazy. Constructively lazy, but still lazy. I only want to have to figure this stuff out once, then stick it in a script and never have to use what's left of my "brain" on it again. :)
 
So, I decided to do a few Chicago Quadio's. I did Chicago (Second Album), VII, and IX. I figured that would be good stuff on my USB stick. So I already had the MKV files on my PC, so that saved me a step. I used AudioMuxer and got them easily to .flac. I then ran Garry's utility to add the empty C and LFE, and then I used Foobar to convert the 24/196 .flac files of unknown compression value to 24/48 zero level compression for the Acura. Not bad. Spent the time tagging them. Got the quad album graphics to put in there, and boom. Done.

The Quadio 4-channel tracks on the disc are natively 196 kHz/24bit PCM, when ripped from disc?
 
192, 196, Doesn't really matter. My concern was that the sample rate is natively *that* high? The raw audio ripped from the discs is 192/24 PCM? Or is that from some conversion you've done?
 
That's how it came out via MakeMKV to a file, then "Extract Audio from MKV file" in AudioMuxer. These files came out that high. I had nothing to do with it.
 
The Chicago Quadio Blu-ray set is 24/192. I'm not complaining, but probably a bit of overkill given the original source was analog.

All original sources are analog. (If recorded from live intruments - computer generated music not)

(Fimiliar point of discussion follows:) Iā€™m yet to hear any difference between sample rates. 96 - 192 etc. The usual proviso: I now have 60yo ears.

When I rip to FLAC I never go higher than 96kHz or 88.2 kHz, double the sample rate - double the file size; For no increase in audible quality (to me)
 
All original sources are analog. (If recorded from live intruments - computer generated music not)

True, but I was referring to the analog recording media that was somewhere in the chain of creating the Blu-ray.

The highest frequency analog recording media could capture is about 20 khz with max 72 dB of dynamic range; which is easily captured by a 16/44 CD or for multi-channel, a 16/48 DVD.

I should add that many prefer the "warmth" added by analog tape or vinyl and there is nothing wrong with that. I'm just commenting on what would be required to digitally capture the warmth and fidelity of those analog tapes produced in the 70's and 80's.
 
Sounds like a bug situation JonUrban. The app in question can convert from wav to flac but if you try going from flac -> sample rate conversion -> flac, there's a bug.

True, but I was referring to the analog recording media that was somewhere in the chain of creating the Blu-ray.

The highest frequency analog recording media could capture is about 20 khz with max 72 dB of dynamic range; which is easily captured by a 16/44 CD or for multi-channel, a 16/48 DVD.

I should add that many prefer the "warmth" added by analog tape or vinyl and there is nothing wrong with that. I'm just commenting on what would be required to digitally capture the warmth and fidelity of those analog tapes produced in the 70's and 80's.

It's not "warmth added" to analog formats. It's warmth lost in some digital copies. The most glaring tinny sounding examples are not even a shortcoming of the lowly 16 bit CD format either. It's intentionally produced for portable devices (iThings & other phones). Squashed with compression so it can be boosted and then treble eq turned up to 11. HD sample rates will clean up a little distortion in the high end from the sample rate bleeding into the program (it's RIGHT next to the top of the data band in 44.1k) which can make more budget converters sound harsh at SD sample rates. But even this is really subtle compared to the intentional bludgeoning that CD & mp3 copies are subjected to as SOP.

Use 24 bit and a sampling rate above 60k (so... something standard like 88.2k or 96k) and converters with class A analog stages in the front end and you can capture ANYTHING any tape can contain. If your digital copy is compromised it's operator error, not a format limitation. You DO need decent converters! You can shuttle ones and zeros around all day long but the conversion in/out is kind of all or nothing.
 
Last edited:
This side discussion is not germane.

So, Jon, if I read you right, you start with

4-channel 24/192 PCM WAV rips-->
then losslessly compressed to FLAC-->
then losslessly compressed *again*, but at '0' setting (which does not change the size) , and *also* converting to 24/48 at the same time?
 
So, Jon, if I read you right, you start with

4-channel 24/192 PCM WAV rips-->
then losslessly compressed to FLAC-->
then losslessly compressed *again*, but at '0' setting (which does not change the size) , and *also* converting to 24/48 at the same time?

So, Sully, I see where you are going with this query. Perhaps the MKV process is adding lossless compression and we know the new Acura sound system doesn't like compressed flac.
 
IMO 24/192 is always overkill when applied to home audio applications...
Strongly agree!
96k puts the sampling frequency miles away from the top of the audio band and thus cleans up that little bit of high end distortion some budget converters are prone to. And if we're talking about budget converters, their clock might just be more jittery at 192k than 96k and adding distortion. Just to give you something else to worry about. :D Sorry for the digression as well!
 
I was trying to parse what Jon wrote:

" So I already had the MKV files on my PC, so that saved me a step. " 1) ripped via MKV at some point in the past, format unknown...assume PCM .wav @ 192/24 due to mention of that later

" I used AudioMuxer and got them easily to .flac." 2) Converted to FLAC, i.e., losssless compression, uknown level (btw , I was wrong; Level 0 *does* still compress, a little. I've never used it, it's always Level 8 for me. )

" I then ran Garry's utility to add the empty C and LFE," 3) some channel layout reformatting tool unknown to me, converts 4.0 to 5.1 , but which apparently has worked in the past with an Acura system?

" and then I used Foobar to convert the 24/196 .flac files of unknown compression value to 24/48 zero level compression for the Acura." 4) resampled 5.1 192-->48 with foobar, because it seems the Acura system prefers that SR. Don't really know what 'zero level compression' is intended to mean here, so I was guessing another FLAC round? That would be way over-complicated, if so. Or maybe it just means, resampled?

Jon seems to say this process worked in the past, but isn't working now...but I'm not clear what if anything is different between 'before' and 'now.
 
I then ran Garry's utility to add the empty C and LFE," 3) some channel layout reformatting tool unknown to me, converts 4.0 to 5.1

This is Music Media Helper. For the channel remixing tool MMH is just a User Interface for Sox that actually does the conversion. For Quad to 5.1 it just inserts two empty channels without altering the original tracks.
 
The highest frequency analog recording media could capture is about 20 khz with max 72 dB of dynamic range; which is easily captured by a 16/44 CD or for multi-channel, a 16/48 DVD.
You're right on for dynamic range; the digital equivalent of the best analog tape is about 13 bits, or ~78dB. As for frequency response, the best analog tape recorders and formulations smoothly rolled off from around 20kHz to 40 or 50kHz.
 
Back
Top