Korg AudioGate SRC

[russtafarian]

Hey Folks,

This is my first post here at Whatsbestforum, having just learned about this forum. I really enjoy the content and interaction I've read so far.

Does anyone know where I can find tech support for Audiogate? I can't find anything on Korg's website. The program locks my PC up whenever I try to burn a disc. Any ideas on how to fix this or where to look for support? Thanks.

Russ

 

[Bruce B]

Well the first thing I'd do is make sure the burner has the latest drivers. Have you run DPC latency checker? Maybe you're having conficts with other drivers.
Is this a 32bit or 64bit machine? I believe there is a Korg User forum as well.

I've had good luck over at GearSlutz as well. You might want to try HERE as well.

 

[russtafarian]

Thanks Bruce. The Korg forum is probably my best bet.

Russ

 

[stevekale]

Hi Bruce. Having a dabble with Audiogate. The first thing I notice is something is going on with gain despite a setting of zero. Here is the output of SOX's Stats command for the same album first with Audiogate conversion to WAV (stereo interleaved, 24/176.4, soft roll-off, normalise off, dither none) with gain set to zero:

Overall Left Right
DC offset -0.000005 -0.000005 0.000001
Min level -1.000000 -1.000000 -1.000000
Max level 1.000000 1.000000 1.000000
Pk lev dB 0.00 0.00 0.00
RMS lev dB -16.34 -16.38 -16.30
RMS Pk dB -7.89 -8.76 -7.89
RMS Tr dB -124.49 -124.49 -124.49
Crest factor - 6.59 6.53
Flat factor 25.16 24.33 25.79
Pk count 11.3k 10.3k 12.3k
Bit-depth 24/24 24/24 24/24
Num samples 395M
Length s 2239.960
Scale max 1.000000
Window s 0.050

And now from Weiss Saracon again with gain set to zero (dither off/truncate, 24/176.4, smart interleave):

Overall Left Right
DC offset -0.000003 -0.000003 -0.000000
Min level -0.665071 -0.664047 -0.665071
Max level 0.667673 0.651757 0.667673
Pk lev dB -3.51 -3.56 -3.51
RMS lev dB -22.36 -22.40 -22.32
RMS Pk dB -13.91 -14.78 -13.91
RMS Tr dB -242.76 -242.76 -242.76
Crest factor - 8.75 8.72
Flat factor 0.00 0.00 0.00
Pk count 2 2 2
Bit-depth 24/24 24/24 24/24
Num samples 395M
Length s 2239.961
Scale max 1.000000
Window s 0.050

A description of each of these measurements can be found here (scroll down to "stats").

 

[Bruce B]

Each software handles DSD->PCM differently because of the Scarlett Book standard of +3dB

If I do a straight transfer of DSD->PCM, if the SACD is recorded hot, there is mucho clipping. I had to get equipment modified to allow for the overages. In PCM, it's called clipping and in DSD it's called over modulation. You will get different values if you use SBM, Pyramix or DiscWelder.

 

[stevekale]

I understood the headroom that ought to be left in DSD to be 6dB, a fundamental characteristic of DSM.

Here Audiogate is applying gain when asked not to. Also, note the gain being applied is different for the left and right channel.

The Flat Factor and Peak Count would appear disturbing also. No?

BTW here an analysis of the same file processed by Saracon but with gain of 3.5:

Overall Left Right
DC offset -0.000004 -0.000004 -0.000000
Min level -0.995103 -0.993571 -0.995103
Max level 0.998997 0.975182 0.998997
Pk lev dB -0.01 -0.06 -0.01
RMS lev dB -18.86 -18.90 -18.82
RMS Pk dB -10.41 -11.28 -10.41
RMS Tr dB -141.58 -141.57 -141.58
Crest factor - 8.75 8.72
Flat factor 0.00 0.00 0.00
Pk count 2 2 2
Bit-depth 24/24 24/24 24/24
Num samples 395M
Length s 2239.961
Scale max 1.000000
Window s 0.050

 

[swanlee]

Hi
I'm starting to do some slightly crazy stuff with audio gate and was just looking for some advice and opinions.

I have always enjoyed the sound signature of DSD, having a very smooth analog signature compared with PCM. Technically I'm probably mid level and I understand most concepts of audio but for me it always boils down to how something sounds.

I've gotten into collecting high res vinyl rips and collecting high res audio in general.

I've had some fun with my first tries using Audio Gate and my results may not be explainable but I like what I'm hearing.

So I took some 24BIT 96K audio files of an LP rip which contained full spectrum high res info or as much as you can get for an LP . It Sounds great but it sounds like 24bit 96K PCM.

Since I love the sound of DSD I decided to convert it using audio gate to 1 bit 5.2mhz WSD files using no filters all default settings. Then converted those WSD files using Audio gate back to 24BIT 96K PCM audio. I'm using WMA lossless as my output type and it is also my input filetype

Guess what? The new 24BIT 96K files had that nice warm DSD signature sound I liked and I prefer them over the original 24bit 96K files.

I looked at both audio files in Adobe Audition and they both retain the full spectrum freq and are the same audio level but oddly enough the wav forms do look ever so slightly different. It Doesn't look like anything was lost in conversion. And their was no gain loss or addition which I thought happened in DSD converting. This at least confirms that the conversion process did something to the audio that made it slightly different and I prefer the sound of those differences.

My question is what technically happened to the audio that makes it look like a slightly different wavform and what may have happened that might make it more pleasing to listen to? Specially to me DSD is smoother, less fatiguing, more analog sounding, details are more natural and easier for my ears to pick up etc.

Again I fully realize how silly it is to go from PCM ---> DSD--->PCM. Since these are my own files I'm not a professional and I'm not going to sell my results and audiogate is free and I always loved the way DSD sounds I thought I'd play around.

Also any advice on setting with audiogate that can make 24BIT 96K audio sound even better when converted to DSD and then back to PCM?

I understand that

1. The DSD conversion may somehow color the sound
2. I really like what ever it does
3. At first glance this conversion process seems silly and should really not produce anything different but it does to me and is confirmed by the wavforms looking different.

I'm really thinking about converting all my High res audio this way and even my favorite CD quality stuff.

 

[Bruce B]

Well first thought is converting PCM to DSD adds noise. This is probably very pleasing to you. Converting back to PCM still retains this noise. You can see this on spectrum plots.
Why did you convert to WSD? This is a proprietary Korg format. Most people are converting to DSF, in which you can add meta data.
As far as looking at the waveforms, you are probably seeing the added noise. I know that DSD and PCM waveforms do look different in this respect.

 

[swanlee]

Well first thought is converting PCM to DSD adds noise. This is probably very pleasing to you. Converting back to PCM still retains this noise. You can see this on spectrum plots.
Why did you convert to WSD? This is a proprietary Korg format. Most people are converting to DSF, in which you can add meta data.
As far as looking at the waveforms, you are probably seeing the added noise. I know that DSD and PCM waveforms do look different in this respect.

I picked the format at random just playing with the software. No rhyme or reason. My wma audio has metadata and it was retained when I converted back to pcm. Is any format better than another when it is just used as an intermediate format?

So is it the noise that makes DSD smoother and less fatiguing in the audible range? To me DSD has s pretty specific sound signature that I like and the simple act of converting to DSD and then back to pcm added that to the files.

 

[Bruce B]

I picked the format at random just playing with the software. No rhyme or reason. My wma audio has metadata and it was retained when I converted back to pcm. Is any format better than another when it is just used as an intermediate format?

So is it the noise that makes DSD smoother and less fatiguing in the audible range? To me DSD has s pretty specific sound signature that I like and the simple act of converting to DSD and then back to pcm added that to the files.

I don't know if any are "better". I just do it in DFF because it's a universal standard for DSD.

I'm sure the noise has an affect on the audible band.

 

[swanlee]

I don't know if any are "better". I just do it in DFF because it's a universal standard for DSD.

I'm sure the noise has an affect on the audible band.

Something does cause the DSD audio signature on these converted tracks are very obvious.

 

[audioarcher]

I picked the format at random just playing with the software. No rhyme or reason. My wma audio has metadata and it was retained when I converted back to pcm. Is any format better than another when it is just used as an intermediate format?

So is it the noise that makes DSD smoother and less fatiguing in the audible range? To me DSD has s pretty specific sound signature that I like and the simple act of converting to DSD and then back to pcm added that to the files.

An interesting experiment would be take a native DSD file and convert it to PCM and back to DSD and see if there is a similar change.

 

[Bruce B]

An interesting experiment would be take a native DSD file and convert it to PCM and back to DSD and see if there is a similar change.

Depends on what filters/dither you use. Going from DSD to PCM back to DSD will yield a much worse result than PCM to DSD and back again because you're adding DSD noise on top of noise and possibly going out of Scarlet Book spec!

 

[swanlee]

Depends on what filters/dither you use. Going from DSD to PCM back to DSD will yield a much worse result than PCM to DSD and back again because you're adding DSD noise on top of noise and possibly going out of Scarlet Book spec!

Over the weekend I did alot more conversions and the intermediate DSD step is pretty much magic when it comes to giving 24Bit PCM that smooth analog like sound signature. It worked for 16BIT 44K audio as well. I'm at least going to go through all my 24Bit audio and convert it.

The DSD sound signature is pretty addictive, you never realize how fatiguing PCM is until you hear something else.

 

[mep]

The DSD sound signature is pretty addictive, you never realize how fatiguing PCM is until you hear something else.

I think many of us here (alright-some of us here) have always found PCM to be fatiguing compared to analog.

 

[Gumby]

Bruce,

Can you explain this statement from the Audiogate User Manual? I am converting DSD to PCM and suspect I need to adjust the gain down by at least 3dB.

"The conversion algorithm of most PCM <> DSD converters (including AudioGate) is designed so that by default,
0 dBFS = 0 dBSACD."

 

[Bruce B]

Bruce,

Can you explain this statement from the Audiogate User Manual? I am converting DSD to PCM and suspect I need to adjust the gain down by at least 3dB.

"The conversion algorithm of most PCM <> DSD converters (including AudioGate) is designed so that by default,
0 dBFS = 0 dBSACD."

0dB SACD is I believe 50% modulation. Scarlet book allows for up to +3dB SACD which is 71.4% modulation. I have seen some SACD's out of Scarlet book spec going all the way up to +6dB SACD. Doing a 1:1 conversion with a hot SACD like this will result in clipping. So what I do most of the time is attenuate -3dB. Still sometimes I get clipping, but it's few.

 

[bblue]

0dB SACD is I believe 50% modulation. Scarlet book allows for up to +3dB SACD which is 71.4% modulation. I have seen some SACD's out of Scarlet book spec going all the way up to +6dB SACD. Doing a 1:1 conversion with a hot SACD like this will result in clipping. So what I do most of the time is attenuate -3dB. Still sometimes I get clipping, but it's few.

I just use the album (song list) normalization, which brings the levels down to nominal 0db as needed. Works quite well, except it does cost you a bit more time before the actual conversion starts, since it has to scan all the songs first to find the highest level, to bring them all down (or up) equally.

--Bill

 

[Gumby]

Bruce,

Thanks for responding to this. I have had trouble getting AudioGate to accept any gain adjustments. Bblue offers an alterantive approach, which I will need to try.

 

[Gumby]

I just use the album (song list) normalization, which brings the levels down to nominal 0db as needed. Works quite well, except it does cost you a bit more time before the actual conversion starts, since it has to scan all the songs first to find the highest level, to bring them all down (or up) equally.

--Bill

I tried this and some minor clipping was eliminated. I tried it on 4 different conversions, and found the gain adjustment to range between -0.7dB and -2.6dB. Thanks for the tip. As noted previously, I could not get Audactiy to accept a direct gain adjustment...don't know what I was doing wrong there.