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I need an oppinion

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Re: I need an oppinion

Postby jpn » Sun Feb 08, 2015 1:07 pm

Thank you all for your oppinions.
Thank you ianc for the link (very nice article).

Some times asking oppinions is a shortcut to increase "personal enlightenment" especially in a delicate matter like this...
all information count .... at least for me.
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Re: I need an oppinion

Postby Sheikyearbouti » Sun Feb 08, 2015 2:58 pm

Than there is also the question - how do you filter (LPF) and how is this or that particular Nebula program made?

From what I know, usually you will have 2 filters at your ADC stage (one analog for the oversampled signal and one digital linear phase for the actual sample rate at which you record for when the oversampled signal is than downsampled to go to 192, 96, 88, 48 or 44). Please correct me if I am wrong or missing something, but it is what I remember from reading books back in the day... Quite a lot of thoughts below and as you can see a bit chaotic.. and a lot to discuss and check really.

Some developers swear in native sampling (so separate sampling session for 44, 96, etc.). I personally don't see the point of this. Least to say, you waste a lot of time sampling the same device more than once. Than you also lose consistency.

OK, if you sample at, say, 192 and than downsample, you would have the (digital) filtering twice (in fact 3 times if we count the analogue filer in the converter too). And if you native sample, it will be 1 filter (2 with the analogue). So in theory it may be better to native sample. But than in practise we are talking about a linear phase filter applied at a very different frequency range of the audio spectrum (so if there is ripple from the first filter, the question is - is it not something we will remove with the second completely?). And also you miss the fact that when you downsample offline (SRC), you can chose the best configuration of the filer for the particular program, and not use the default fixed filter in your ADC (so you can chose to have steeper filer, or to lose some of the VHF content and make the filter less steep by adjusting the cut-off frequency, or adjust pre-ringing parameters, and this depending on the actual aliasing created when you sample - so for a clean program you can have one setting, for a dirtier another).

You also have the well known Lavry argument that at sample rates higher than 88/96 the converter delivers worse performance. If you check the Lavry Gold DAC you will find that the highest sample rate is 96k. And it is a 5k device for 2 channels!

There is another argument, which I kind of favour - and it needs more testing on my side - that at 88/96 your converter works better than at 44/48 and thus you think that the recording is better at 88/96 than at 44/48 (well, in fact, it may also be that the recording slightly better, but not only). This can easily be tested in a blind listening test with HF rich signal that is record at say 48 and 96. Than the 48 is upsampled to 96 and the two 96 versions compared with the original 48 one. If the upsampled 48 sounds better than the original 48, than clearly it is your converter that sounds better - not the recording.

Also to put it this way - when I record, say, distorted brass (yeah, I love putting ribbons on brass and than cranking a copy on another channel on the N**e console to distort it and do parallel distortion to add some highs) I am 100% that I get cleaner sound at 96 than at 44 - my explanation is aliasing. But when I record clean piano, this HF content is not there to create aliasing.

For now to me 96k is a very good safe area to work at. I am doing tests to see if 192 is better for NAT sampling with additional offline SRC.

If I have to downsample to 44/48, I would FIRST process in Nebula (so preamps, some basic EQ, etc.) and than downsample. The idea is that you use the higher res Nebula presets on the higher res audio and than you go down to 44/48 to finish your mix. But I also like to commit to processing and process to tape when I record, so this approach works well for me - for others it won't.

Than there are people who go for DSD (or modern equivalents of it) to avoid the filtering at all (no need for anti-aliasing filter there - at least in the first stage of recording). But in the end, you still need a filter to make the program usable in NAT/Nebula. So it is not such a massive benefit, if there is any at all - but again, I've no real opinion, as I have never tested it myself.

And than the last option is to SRC in Nebula, which I have no opinion on too, but would generally avoid, exactly because it is an unknown process for me that I'd rather test first.

For example SADiE was notoriously famous for bad SRC (fixed in the latest version). And SADiE and Prism are one company really. So even the they had it wrong until recently.
In any case - all these will be relatively minor differences and of course there are lot more important factors to consider - what is the source you record (player, instrument, room), where do you put that microphone, what is the gear you use for the recording. Only than comes Nebula and its sample rate and the way the programs are done...

And BTW, see, I do not mention anything about hearing above 20k, which I think is marketing bull. Although recently I have been hearing poeple from serious companies talking about interaural time differences in localisation that can be detected above 20k time span - i.e. the effect of sample rate on transients (how come we can hear if we can't hear??? ahaha). I think that if implemented perfectly 44k should be just enough for us humans (the questions I ask are related to - is it implemented perfectly?).

Yes, there are tons of amazing records released at 44k/16bit. Still, my job as an engineer is to deliver the best I can. Thus I try to stay at 96 for as long as I can in the production process, ideally until the final point when I have to prepare the audio for CD quality release. I even bounce, freeze and commit to achieve this.
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Re: I need an oppinion

Postby david1103 » Sun Feb 08, 2015 7:19 pm

Thanks for the great post Sheikyearbouti!

I have just done a whole lot of tests and am certain that my soundcard sounds worse recording at 44.1 than 88.2 and 96! I am sure it has nothing to do with higher rates being 'better' but how the ADC works in my sound card. I would also expect a better ADC to sound better at 44.1.

Conclusion, there is no escaping the fact you need to try all the sample rates and decide which is sounding best ON YOUR SETUP!

I have proven beyond a doubt that I get a much better sound recording at 88.2 or 96 (doesn't matter which) and then using Sox at highest quality settings in 100% linear phase mode to downsample to 44.1

http://sox.sourceforge.net/

Its not that subtle, I am only testing on my recorded voice. At 44.1 native recording it sounds boxed in and lacking air.

It's important to note I tried using Reapers SRC and the results were only a fraction better than native 44.1. You need to use Sox downsampling in the right mode to get the better sound.
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Re: I need an oppinion

Postby richie43 » Sun Feb 08, 2015 7:43 pm

david1103 wrote:and then using Sox at highest quality settings in 100% linear phase mode to downsample to 44.1


I am a smart guy (I like to think so at least...ha ha), but I have never been abke to wrap my head around SoX. I even am familiar with command line, Linux, etc. But I have never been able to make SoX work.....
David (or anyone else): Would you be willing to help me at least to the point of getting SoX to do some simple conversion? Once I get that far, I am sure I can dig deeper on my own.....Please.......?
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Re: I need an oppinion

Postby david1103 » Sun Feb 08, 2015 7:57 pm

richie43 wrote:
david1103 wrote:and then using Sox at highest quality settings in 100% linear phase mode to downsample to 44.1


I am a smart guy (I like to think so at least...ha ha), but I have never been abke to wrap my head around SoX. I even am familiar with command line, Linux, etc. But I have never been able to make SoX work.....
David (or anyone else): Would you be willing to help me at least to the point of getting SoX to do some simple conversion? Once I get that far, I am sure I can dig deeper on my own.....Please.......?


Hi Richie,

Are you on a PC? I am using the Foobar Sox plugin:

http://www.hydrogenaud.io/forums/index. ... opic=67373

It's very easy to use. If you are using Reaper then this reascript might be the best solution:

http://forum.cockos.com/showthread.php?t=145240

You need Python installed on your machine to run the script.

Hope one of those options works for you :) I am sure there are many different ways to use Sox other than the command line.
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Re: I need an oppinion

Postby richie43 » Sun Feb 08, 2015 11:48 pm

Thanks David. There are some versions people have made over the years that come with an actual GUI, but they are very old versions of SoX. I am on PC, and I do use Reaper, and already have Python installed...so maybe I am on my way...Thanks!
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Re: I need an oppinion

Postby david1103 » Sun Feb 08, 2015 11:58 pm

richie43 wrote:Thanks David. There are some versions people have made over the years that come with an actual GUI, but they are very old versions of SoX. I am on PC, and I do use Reaper, and already have Python installed...so maybe I am on my way...Thanks!


Also Audacity now has Sox built in, forgot that!

I am trying to get that rescript to run and having a few problems. I am hoping the creator will reply soon, I can see he is online :)

Here is the thread:

http://forum.cockos.com/showthread.php?t=145240
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Re: I need an oppinion

Postby richie43 » Mon Feb 09, 2015 2:04 am

I saw the thread, thanks.
So far I have not figured it out....

Maybe I will try Audacity...It's been years since I messed with that!
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