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  #16  
Old 08-27-2018, 01:04 PM
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Originally Posted by ChuckS View Post
PM sent.....
Ah, I see a download link in the embedded Soundcloud (though not on Soundclouds website). I will check out the embedded download bit rate later on.
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  #17  
Old 08-27-2018, 08:21 PM
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Originally Posted by ChuckS View Post
I appreciate your comments on the recording and the detailed suggestions.

The mic setup was a spaced pair, separated by about 16", set back about 22", one aimed at the bridge (right channel, panned hard right) and one aimed at the 15th fret (left channel, panned hard left). Mics were a pair of Gefell M296S's, Gordon preamp (recently picked up used), and Mytek converters.
And keep in mind I was being very picky. I sit here mixing & editing music all day. So I push towards what I like to hear...what feels really balanced and open to my ears.

The Gefells are very nice mics...and you won't get much better than a Mytek ADC. I'm not familiar with the Gordon pre.

Thanks for sharing - well played & recorded. As I said, I'm just being super pcky.
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  #18  
Old 08-27-2018, 08:25 PM
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Sounds pretty good. Soundcloud uses a low bit rate mp3. You might make the original wav file available for download.
I find that uploading a 24-bit, better than 44.1k file to Soundcloud (I typically go 24/48k) really improves the sound of the converted file. Also keeping the file with a true peak of -1.0 dBFS and -14 LUFS minimizes how much processing Soundcloud applies to get your file to those specs.
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1988 Fender Fretless J Bass
1991 Washburn HB-35s
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1996 Taylor 510c (custom)
1996 Taylor 422-R (Limited Edition)
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  #19  
Old 08-27-2018, 08:50 PM
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I find that uploading a 24-bit, better than 44.1k file to Soundcloud (I typically go 24/48k) really improves the sound of the converted file. Also keeping the file with a true peak of -1.0 dBFS and -14 LUFS minimizes how much processing Soundcloud applies to get your file to those specs.
I believe the file Soundcloud file ends up being a 128 bit rate mp3.
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  #20  
Old 08-27-2018, 10:22 PM
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Played with downloaded wav file.

Sound was a little muffled due to frequencies in about the 150 to 250 hertz range. Cut a few decibels in about that range and then
decided to boost a few decibels around 4000 hertz and above. Boosted volume a little and also added a little reverb.

Results:

http://dcoombsguitar.com/Temp/HerFavoriteAGF.wav
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  #21  
Old 08-28-2018, 07:34 AM
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Quote:
Originally Posted by rick-slo View Post
Played with downloaded wav file.

Sound was a little muffled due to frequencies in about the 150 to 250 hertz range. Cut a few decibels in about that range and then
decided to boost a few decibels around 4000 hertz and above. Boosted volume a little and also added a little reverb.

Results:

http://dcoombsguitar.com/Temp/HerFavoriteAGF.wav
Thanks for taking the time to mix my recording and for providing the file. I listened in the car this morning and will check it out on my monitors when I'm at home.

Would you mind sharing what you did for reverb?

Maybe I'll be able to control the low end somewhat if I pay better attention to mic placement (need to check notes as I did make an adjustment in the process of recording several songs). It may be more critical for this guitar, as it does have quite a dark sound.

Thanks again.
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Last edited by ChuckS; 08-28-2018 at 07:51 AM.
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  #22  
Old 08-28-2018, 08:17 AM
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Would you mind sharing what you did for reverb?
I believe I used one of the Lexicon PCM Native Reverb room reverbs at around 15% wet to dry. Basically I wanted one that keep things clear and brighter
rather than darker and at a percentage to keep the reverb from being very obvious. I do not like to use heavy amounts of reverb.
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  #23  
Old 08-29-2018, 10:05 PM
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Originally Posted by rick-slo View Post
I believe the file Soundcloud file ends up being a 128 bit rate mp3.
Right. And the conversion sounds better if you can avoid hitting it too hot and by using a high sample rate to push the Nyquist filter way out of the range of the final converted frequency response.
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1991 Washburn HB-35s
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  #24  
Old 08-29-2018, 11:32 PM
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Originally Posted by DupleMeter View Post
Right. And the conversion sounds better if you can avoid hitting it too hot and by using a high sample rate to push the Nyquist filter way out of the range of the final converted frequency response.
Interesting. Had to test that.

Following audio is original 16bit/44.1 hertz wav file followed by a 128 bit rate lame mp3 derived from the 16bit/44.1 hertz wav file
followed by a 128 bit rate lame mp3 derived from a 24bit/48 hertz wav file.

Http://dcoombsguitar.com/Temp/ValsCr...av16442448.wav

Ehhh.
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  #25  
Old 09-01-2018, 06:11 AM
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It sounds good Chuck. Nice playing.
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  #26  
Old 09-01-2018, 09:20 PM
DupleMeter DupleMeter is offline
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Originally Posted by rick-slo View Post
Interesting. Had to test that.

Following audio is original 16bit/44.1 hertz wav file followed by a 128 bit rate lame mp3 derived from the 16bit/44.1 hertz wav file
followed by a 128 bit rate lame mp3 derived from a 24bit/48 hertz wav file.

Http://dcoombsguitar.com/Temp/ValsCr...av16442448.wav

Ehhh.
You need to actually start with a higher bit-rate recording. Upsampling a 44.1k file to 48k gives you nothing that wasn't already there.

You also need to make sure you're hitting -14 LUFS, with a true peak (not just a peak, but a true peak...which is calculated with intersampling overs) of -1 dBFS, for the absolute best results.

I actually learned this from Andrew Scheps...and wouldn't you know it...he was right (well..duh, right...I mean the guy has great ears).
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1927 Martin 00-21
1986 Fender Strat
1987 Ibanez RG560
1988 Fender Fretless J Bass
1991 Washburn HB-35s
1995 Taylor 812ce
1996 Taylor 510c (custom)
1996 Taylor 422-R (Limited Edition)
1997 Taylor 810-WMB (Limited Edition)
1998 Taylor 912c (Custom)
2019 Fender Tele
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  #27  
Old 09-02-2018, 12:13 AM
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Quote:
Originally Posted by DupleMeter View Post
You need to actually start with a higher bit-rate recording. Upsampling a 44.1k file to 48k gives you nothing that wasn't already there.

You also need to make sure you're hitting -14 LUFS, with a true peak (not just a peak, but a true peak...which is calculated with intersampling overs) of -1 dBFS, for the absolute best results.

I actually learned this from Andrew Scheps...and wouldn't you know it...he was right (well..duh, right...I mean the guy has great ears).
Iffy theory.

Many plugins (equalization, etc.) use internal upsampling prior to complex calculations to prevent audible aliasing effects.

For example read: https://www.soundonsound.com/techniq...pling-plug-ins

The same principle could well apply to upsampling a 44.1 kilohertz recording prior to mp3 conversion. Or perhaps that's an inherent part of the mp3 conversion process,
or perhaps that is not even necessary, or perhaps it's even detrimental (see article below) to use a sample rates other than 44.1 or 88.2.


An interesting article ( https://www.audiorecording.me/dither...e-study.html/2) where sampling rates of 96 and especially 48 created more artifacts than
a 44.1 sampling rate (at least with lame). Also interesting in this article is that the use of dither can cause problems for mp3 conversion quality.


Who knows what soundcloud does. Perhaps whatever they receive is converted straight into the mp3. Or perhaps they first convert every thing to a certain resolution wav
file prior to mp3 conversion.
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"Reality is that which when you stop believing in it, doesn't go away."

Woods hands pick by eye and ear
Made to one with pride and love
To be that we hold so dear
A voice from heavens above

Last edited by rick-slo; 09-02-2018 at 08:59 AM.
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  #28  
Old 09-02-2018, 09:08 AM
DupleMeter DupleMeter is offline
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Originally Posted by rick-slo View Post
Iffy theory.

Many plugins (equalization, etc.) use internal upsampling prior to complex calculations to prevent audible aliasing effect

That's completely irrelevant to the point of SoundCloud conversions.

The issue is the amount of frequency material stored in the file and how much processing happens on the SoundCloud end.

By providing more data in the file the compressed result is closer to the original. You want to give the compression algorithm as much data to work from as possible. It makes better decisions with more information.

SoundCloud doesn't care if your plugins oversampled to sound better. It is only looking at your final deliverable. And upsampling a 44.1k file to anything else provides no benefit. It's very basic physics. There is no additional info added during upsampling, just more bandwidth for future use. But to take advantage of that you need to fill that bandwidth with data (in our case audio signal). It's all about writing the data up past the audible range while avoiding the Nyquist filter's effects. That's why higher sampling rates sound better. There is less damage being done in the audible range. You're simply pushing the filter artifacts way up the frequency spectrum to where they do less (not none, just less) damage.

The only valid test for this is to start with a higher bandwidth file. You need to have the harmonic information already in the file. This is why iTunes Plus/Mastered for iTunes requires a 24/48k file at minimum (they actually prefer 96k). They figured out that the compressed MP4 sounds better from the extended frequency data.

If you look at your example the 44.1k & 48k have the same sonic fingerprint (us a spectrograph). This is because the 48k file is simply the same info, with extended storage capacity that is empty.

The mp3 shows significantly less data above 12k, which is expected.

I get that this is not easy to wrap your head around. And there is a lot of bad info going around. But for those of us who are actually engineers, who understand the math behind all of this, it actually makes logical sense. In the end it's just math. Which is provable. Myths & heresay are not.
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1927 Martin 00-21
1986 Fender Strat
1987 Ibanez RG560
1988 Fender Fretless J Bass
1991 Washburn HB-35s
1995 Taylor 812ce
1996 Taylor 510c (custom)
1996 Taylor 422-R (Limited Edition)
1997 Taylor 810-WMB (Limited Edition)
1998 Taylor 912c (Custom)
2019 Fender Tele
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  #29  
Old 09-02-2018, 01:14 PM
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DupleMeter,

I would suggest you read "Digital Audio Explained" by Nika Aldrich and some of the myths including the one that sample rate recordings higher than the Nyquist frequency (about 44.1 kilohertz) inherently sound better.

Oversampling is useful during dynamic processing (say compression) of a waveform to prevent aliasing artifacts into the frequency range of human hearing. Depending on what processes Soundcloud goes through to get to the final 128 bit rate mp3 oversampling may not be relevant. I mentioned it because you mentioned it.

Also you are buying into “Mastered for iTunes” marketing. I would suggest a health skepticism. For example read: https://9to5mac.com/2012/02/28/maste...ser-to-the-cd/
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Youtube -> Website -> Music -> Tabs
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"Reality is that which when you stop believing in it, doesn't go away."

Woods hands pick by eye and ear
Made to one with pride and love
To be that we hold so dear
A voice from heavens above
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  #30  
Old 09-02-2018, 09:26 PM
DupleMeter DupleMeter is offline
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Originally Posted by rick-slo View Post
DupleMeter,

I would suggest you read "Digital Audio Explained" by Nika Aldrich and some of the myths including the one that sample rate recordings higher than the Nyquist frequency (about 44.1 kilohertz) inherently sound better.

Oversampling is useful during dynamic processing (say compression) of a waveform to prevent aliasing artifacts into the frequency range of human hearing. Depending on what processes Soundcloud goes through to get to the final 128 bit rate mp3 oversampling may not be relevant. I mentioned it because you mentioned it.

Also you are buying into “Mastered for iTunes” marketing. I would suggest a health skepticism. For example read: https://9to5mac.com/2012/02/28/maste...ser-to-the-cd/
So the 9to5 article is concerning itself with sounding closer to a CD release. That's not what we're talking about. He's simply null testing against a CD release. Anyone with even the slightest audio engineering background knows you will never null a PCM file against a lossy compression version. Heck, I can make a CD file fail to null against a copy of itself by adding even the slightest spectral change to one of them. That's just smoke & mirrors. And adding more confusion to this whole thing.

I have read Nika's work. I don't think you fully understand his stance about sampling rates. He does say there are times that lower sampling rates can sound better. But they are specific cases, with specific criteria. He also avoids getting into the math, which leaves out some important facts in exchange for being more accessible. I recommend reading Bob Katz's "Mastering Audio". Which goes into a lot more detail. it will fill in a lot of the holes.

I feel like I want to clarify what I'm saying with a quick visual. So here are 2 mp3s from the same file. The top is from the a 48k version of the file (original recorded sampling rate). The bottom is a conversion to mp3 after resampling to 44.1. You can see there is less info in the 44.1 file. including some areas of frequency totally missing (meaning the mp3 decoder has to interpolate that each time...with varying degrees of success). Now do the same with 96k (or 192k) vs 44.1 and imagine how much more info the mp3 decoder has to work with and not have to magically pull out of algorithm world? Also, keep in mind that with a spectrogram, the brighter the orange the more info is there. So that's significant too...as you will see differences in the 2 mp3s even in that. And this isn't even zoomed in.



I'm really not interested in arguing this. I don't want to get into a thing. It's just not worth it...I'm sure you have other things to do...I know I do. This would probably be best as a conversation over a few beers. My treat if you're ever in the NYC/CT area.
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1927 Martin 00-21
1986 Fender Strat
1987 Ibanez RG560
1988 Fender Fretless J Bass
1991 Washburn HB-35s
1995 Taylor 812ce
1996 Taylor 510c (custom)
1996 Taylor 422-R (Limited Edition)
1997 Taylor 810-WMB (Limited Edition)
1998 Taylor 912c (Custom)
2019 Fender Tele
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