Topic: Audiogate editing/ normalizing

[DavidDSD]

Hi

Got a very basic question:

Just recorded a dj's vinyl set using a Korg MR-1000.  It's a 4-hour 1bit/2.8 Mhz file and, for this particular project, I'll be converting to AIFF or FLAC.  It was recorded at quite a lowish level to safefguard against clipping.

I'm just wondering if I should do the very basic editing (just joining a gap) in Audiogate and normalize with Audiogate, too. Or would it be best to do the editing/ normalizing after it's converted to MP3 with Audacity or Wavelab.

Thanks!

[taperj]

I generally dump the whole set through audiogate as one track then if I need to normalize I take it to my DAW and do it there normalizing to -0.5dB over the whole set. Definitely don't normalize per song(you probably know that already). This isn't to say the normalize wouldn't work in audiogate, it just seems primitive, I have greater flexibility on what level to normalize to in my DAW. I suppose I should just normalize a test track sometime through audiogate to see what it does, the "normalize to average volume" just kinda scares me. As far as editing, I find that much easier to do in a DAW where I can see the waveform and work with multiple track overlays for patching etc. I then use CDWave to track so there are no SBE's and use it to flac as well at level 8, I believe audiogate will use system default which is level 5, audiogate is also not SBE aware so you will have sector boundary errors if you use it for tracking without fixing SBE's with TLH or another such tool.

my usual processing chain for DSD:

Korg MR-1(DSD of whatever flavor) > audiogate(decimate to wav/dither[Korg Aqua]/resample) > DAW(normalize/edits if needed) > CDWave(SBE friendly tracking/flac'ing) > FLAC(Level 8[or whatever other format, be it mp3,ogg,whatnot])

If the audiogate normalize had more functionality I would certainly use it and take the DAW out of the equation for anything I didn't need to do edits on. Not sure if this helped or not, at least it's of an example of one way to go about it.

J

[DavidDSD]

Thanks, that was just what I was looking for!

Couple of things:

Not sure exactly what role CDwave is playing in your chain.  Why not go from Wav > Flac with your DAW?  (What's an SBE?)

Can CDwave also leave markers in a DJ's set, so that it will read as having separate tracks (on a cd player etc), even though it still plays as one continuous, uninterupted mix?

[taperj]

Not sure exactly what role CDwave is playing in your chain.  Why not go from Wav > Flac with your DAW?  (What's an SBE?)

Ok, CDWave for 2 reasons in my case, The less important is #1, because it allows me to set flac Level 8(my DAW just does system default flac level 5). Reason #2 is paramount and leads into your next question. SBE's are Sector Boundary Errors. Simply put, CDs are only capable of supporting track boundaries at 1/75th of a second intervals, audiogate and most DAWs are far more accurate going down to thousands of a second and ticks which, unless there is a setting to be SBE compliant, you will end up with SBE's because you are not cutting your tracks on the 1/75th of a second sector boundaries. CDWave's accuracy for tracking is set to only cut on compliant sector boundaries. How does this translate to the real world? On to your next question.

Can CDwave also leave markers in a DJ's set, so that it will read as having separate tracks (on a cd player etc), even though it still plays as one continuous, uninterupted mix?

SBE's are what will make a cd "skip" at track breaks when on continuous play. This is due to an extremely small blank space in between where it was tracked and where the next sector boundary is which is where the next track begins. This makes the player have to drop the levels to 0 for that thousandth of a second, which makes an audible track "break." If all of the tracks are cut on SBE's, then the show will play seamlessly on continuous play because the tracks truly do join together at the breaks and are continuous.

Hopefully that's detailed enough to explain it?

[DavidDSD]

Thanks for the explanation, man!

Converted the one 4.5 hour Audiogate PRJ file to wav (2.78GB), using Aqua dither, but it seems to be corrupted. I'm wondering if my ageing Athlon +3000 PC is able to handle processing such a large file properly.  Looks like I'll have to convert each of the original 1GB files at a time and then join them together with Wavelab. Pain!

Once done, will also need to decide what normalizing to use.  Wavelab has a level normalizer, loudness normalizer and gain change.  My original vinyl mix was recorded at quite a low level to avoid clipping.

[Shadow_7]

I normally convert it all to 24@96kHz with audiogate.  Although sometimes I DON'T join the individual files at that time.  Why you ask?  Because audacity makes a temp file / copy by default.  Which goes to your main partition which may not have enough space for that.  And everytime that you do an edit it creates another copy.  If you don't have enough ram / HDD space for this, it does a lot of processing for a lot of time and then fails.  If I had more ram and HDD space on that laptop and primary partition I'd probably do 24@192kHz editing.  I keep the higher sampling rate so I can do some EQ edits before rendering out at 16@48kHz (DVD) or 16@44.1kHz(CD).  Once I've done my edits I use sox to rejoin longer clips and other things if I need to. 

I do stereo recording so my L to R perspective is set at the time of capture.  And there might be things that skew towards the L or R so I don't normalize per say.  I use the Amplify function in audacity.  This gives me a numerical number that I can use as a common denominator between individual files.  And this keeps L and R at the relative levels to each other that they were recorded at.  That way one side isn't quieter than the other if someone slams a door during a recital or something.  And when joining multiple files later, they all had the same edit(s) done.

[DavidDSD]

Got ya, Shadow.

Regarding Audiogate, if you want to export to WAV at 179 or 192kHz, what's the most appropriate setting for the DSD filter: sharp roll-off, soft or no filter?

And if I allow Audiogate to do the normalizing, would it still be appropriate to use the Dither function as well (eg Aqua)?

Thanks!

[Shadow_7]

I just use the TPDIF filter.  I might not have the latest version of audiogate.  Although my version does have aqua as an option.  I don't like doing any normalization before I see the waveforms.  Just to see that they are closely matched, no bird chirp / door slam / obnoxious clapper to one side or anything that would otherwise skew a stereo image if normalized (as opposed to amplify to the highest peek for both channels).  i.e. There's a lot of other things that audiogate could eventually do, until then I let it do what it should do well.  Dither and down sample.  Everything else is done in more featureful software (not to imply better).

[it-goes-to-eleven]

Reading the audiogate release notes, I do see that there is a function to normalize the dsd stream.  I would think native normalization would be best. It might be interesting if someone recorded a low level source, like a clock tick, and compared dsd normalization vs. pcm.

I did just read something troubling about audiogate... Apparently you must have a working recorder to activate the software.  So let's fast-forward 5 or 10 years, when my Korg device has stopped working... But I still have lots of DSD recordings I might want to convert.

[Shadow_7]

Audiogate does authenticate that you have the device to use the software on the new version(s).  The mac version had an update so it didn't have to do that EVERY time that you used it.  It's why it DOESN'T run in linux.  Otherwise it probably would run via wine.  The OLDER version that ships with the device doesn't require that auth iirc.  But it's very feature lite.  It does authenticate something with the windows registery, which is why that one doesn't run in linux either.  Although I haven't really tried it recently to see if that's still the case. 

Not that it matters much.  I keep the DSD original and the 24/96 versions.  Disk space is cheap these days.  1TB < $100.  I've got maybe 60GB of DSD files right now after 2+ years.  Weekend warrior though and mostly every other weekend.  In 5 to 10 years DSD should be supported to some degree in non-Korg software and probably OSS software too.  Most of my deliverables are 16/44.1 (CD) or 16/48 (DVD) so even 24/96 is HD to a degree.  As least until 1080p60 with 24/96 or better audio is the minimum standard.  And broadband is last years dialup.

[taperj]

Just a quick testimony on the dither..

 I've been using the Korg Aqua dither since I upgraded and it works quite well also. Admittedly, I haven't done an A/B comparison to see if I can hear any additional noise or whether there's any tonal difference between it and the TPDIF dither but I think either one should serve you well. Just from the presentation of the options I would say Korg is trying to convey that their dither should be better than the "old faithful" TPDIF dither and that the TPDIF dither is there for skeptics. Whether either one is really better I have no idea.

J

[it-goes-to-eleven]

I've been using the Korg Aqua dither since I upgraded and it works quite well also. Admittedly, I haven't done an A/B comparison to see if I can hear any additional noise or whether there's any tonal difference between it and the TPDIF dither but I think either one should serve you well.

I'd be curious how an fft of the resultant pcm looks when you record a sine wave (and some silence), comparing the two types of dither.   Of course noise-shaping dither typically isn't so much about how it *looks* on the fft, as how it subjectively *sounds* from a psycho-acoustic perspective.

[DavidDSD]

I did just read something troubling about audiogate... Apparently you must have a working recorder to activate the software.  So let's fast-forward 5 or 10 years, when my Korg device has stopped working... But I still have lots of DSD recordings I might want to convert.

That would be ironic bearing in mind Korgs marketing blurb about DSD recordings being.... 'futureproof'.

Audiogate doesn't seem to work perfectly in my Windows 7, 64 system- I get weird speed fluctuations.  The latest version of Audiogate was designed for Vista/64 and hasn't been refreshed since about August  .