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24bit vs 16bit, the myth exploded! - Page 77

post #1141 of 1938

I guess you listened to that htg podcast. I tried too but closed the window after he said that his special stuff is also full of tubes.. oh yeah and that his studio is the best he's ever seen (or something like that).

post #1142 of 1938

Thanks a lot for the help biggrin.gif.

post #1143 of 1938

Wow, I did not realize this thread was still going. I totally want to be a part of this. Has anyone brought up ripple distortion on the digital reconstruction of the signal? I couldn't read through all the comments but in what I saw, it didn't seem like there was anything mentioned. 

post #1144 of 1938
Quote:
Originally Posted by krtzer View Post

Wow, I did not realize this thread was still going. I totally want to be a part of this. Has anyone brought up ripple distortion on the digital reconstruction of the signal? I couldn't read through all the comments but in what I saw, it didn't seem like there was anything mentioned. 

Ripple distortion?

post #1145 of 1938

Perhaps he is talking about the Gibbs effect (http://en.wikipedia.org/wiki/Gibbs_effect).

post #1146 of 1938
Quote:
Originally Posted by krtzer View Post

Wow, I did not realize this thread was still going. I totally want to be a part of this. Has anyone brought up ripple distortion on the digital reconstruction of the signal? I couldn't read through all the comments but in what I saw, it didn't seem like there was anything mentioned. 

 

It was discussed already, just not called "ripple distortion". Basically, with any reasonable implementation of filtering at 44.1 kHz, the effect is too short and is at too high frequency to be audible. Also, human hearing does not really care about what waveforms "look" like, since it senses bands of frequencies, rather than a simple time domain signal like in a WAV file. There is an example of a real reconstruction filter in a DAC here, which shows that the filtering only significantly affects frequencies above 20 kHz, and that the pre- and post-ringing have a duration of only about +/- 1 ms. Additionally, there were some ABX tests on the forum that could be used to compare (among others) different reconstruction filters, but the positive results are lacking so far.

post #1147 of 1938

Or perhaps he's talking about ripple in the frequency domain:

 

 

In which case there's no need to worry either, because the ripple is usually too low to make an audible difference.

post #1148 of 1938
Quote:
Originally Posted by xnor View Post

Or perhaps he's talking about ripple in the frequency domain:

 

 

In which case there's no need to worry either, because the ripple is usually too low to make an audible difference.

xnor, yes that was exactly what I was talking about. The ripple in the passband. I think that is one of te main arguments for 96khz recordings; to try to reduce the amount distortion. It's been a couple of years since my DSP class so I'm trying to refresh myself. 

post #1149 of 1938

Frequency response ripple in the passband is only an issue in outdated or very low end DAC chips (such as older onboard codecs, some portable players, etc.). It is easy to reduce to well below the threshold of audibility with a reasonable FIR filter design; very low ripple and very high stopband rejection can be achieved by using the right window function, at the expense of requiring a somewhat longer impulse response. The DAC example linked above does not have any significant ripple either.


Edited by stv014 - 5/13/13 at 6:13am
post #1150 of 1938
Quote:
Originally Posted by jaddie View Post

Then, at least one high-rate proponent (Bill Schnee, Bravura Records) claims he had to have a "special" A/D converter built before he could hear the difference (his ProTools gear simply wasn't good enough), and even then he claims you need 192KHz before the benefits are readily apparent. If you reduce his claims to specifics, it's his "special" converter at 192/24, his "special" console, his "special" mix to 2 tracks, and then it's all wonderful, even if down sampled, though if that's done at least some of the "wonder" goes away.  Of course, he doesn't actually sell anything, at least yet.  When asked what, specifically, causes the improvement, he doesn't claim that it is not just the additional bandwidth, though, it's something else.  But unfortunately, he places the improvement into the "mysteriously unmeasurable" zone.  

 

That would imply that garden-variety 192/24 ADCs mask the benefits of that bit rate.  So files we can buy online won't contain the full high-rate glory.  And you pretty much need...well...his entire studio and him to get those kind of dramatic results.  That means there's no way for the rest of us to hear this stuff outside of the Christmas music downloads on his site.  He didn't call DACs into question, and I'm sure I have no idea why not, some would. 

 

I'm not saying I agree with any of this, but I find it interesting that someone of that stature would focus the issue that tightly and try to build a record company on that basis.

 

Barry Diament has made similar observations about recording at 24/192 and his Soundkeeper Recordings is built on delivering his hi-res recordings. He has full songs in different resolutions available for download and comparison:

 

http://www.soundkeeperrecordings.com/format.htm

post #1151 of 1938
Quote:
Originally Posted by weirdo12 View Post

....He has full songs in different resolutions available for download and comparison...

At least one of the 16/44 wavs has a bad header or corrupted wav container so won't play in some players. I downloaded the Maria samples from Americas. I remember the same thing a few months ago with the sample. I can't remember if I tried the other offerings. Anyway if your player chokes on the 16/44 wav you can just "convert" it losslessly to a new wav file which will have identical pcm audio but will actually play in any player/device/app that supports wav. ffmpeg will do it with "ffmpeg -i input.wav -c:a copy output.wav". You could also use foobar2k's convert utility.
Edited by julian67 - 5/13/13 at 9:12am
post #1152 of 1938

Not sure if this has been discussed yet or not. Apologies if it has. So, I listened to my first hi-res 24bit/88.2kHz recording the other day. At the time to be honest it sounded pretty normal to me. Then I realised that I was listening to it on my secondary system, which the DAC only has 3 sample rates, 32 kHz, 44 kHz and 48 kHz. I first thought, oh that must be why I can't hear any noticeable difference in quality. What I really should have been wondering was, why is this working on this DAC? Yes the sample rate was a multiple of 2 of 44.1 kHz and indeed that is what the DAC seemed to lock on to (a red LED told me so). But is this it? Did the DAC just average the two bits instead or one? Any explanation on why this worked would be really appreciated. I can see how it might be possible, but assumed the DAC would be more fussy than that.

post #1153 of 1938

My guess is you're using Windows with DirectSound. If the sample rates don't match the sound engine will resample it accordingly.

post #1154 of 1938

Sorry, I should have explained my setup. I'm streaming the music over ethernet from my music server to a squeezebox, then from the squeezebox to the DAC. I'm pretty sure that the Squeezebox server is not resampling on the fly, but I suppose it could be.

post #1155 of 1938

I just posted this reply in the topic 24/96 files and I thought it might be of interest here as well.

 

Quote:

I think you quote me a bit out of context.

I am talking about the recording process,

i.e.multi track audio being recorded and mixed.

That is were the benefit of i.e.24/96 resolution is to be gained.

At the source.

When recording in 24/96 we are at mix down using less equalization and the reverb tails just sound so much better,than at 16/44 or 24/44.

 

Frans and I were the engineer/producer team behind all 10 of the official Carmen Gomes inc.albums.the 8 previous albums were recorded for the now no longer existing Byton/Via label.

These recordings were done in 24/44 and maybe even the first 2 in 16/44.

But when we went to 24/96 for the "Torn'',and the ''thousand Shades of Blue'' albums,everything(mixing and mastering just got so much easier).

The samples on our site from the different recordings we have made, are just simple mp3's,yet people comment on how good they sound,that is because the source was recorded at 24/96 MULTITRACK.

If I was to take one of the songs we are selling and do the test you suggest,I'm sure I would fail,even with a 320 or 256 mp3 thrown in for good fun, I'd probably fail.

But if I was to go back in the studio, bring back in the 4 musicians,put back up the 3 drum mic's,the double bass mic,the guitar mic,the vocal mic,the two room mic's.take just as much care when placing the microphones,that is listen, record,move the mic or mic's,listen record,repeating the process many times,driving the musicians crazy,then record the whole song in 16/44,mix it down in 16/44,and then compare the two,then I would tell the 24/96 file immediately.

I know this for sure because we did so.

We actually were very skeptical to the whole higher resolution thing in the beginning.It was also a question of money,to record higher resolution(24/96) we had to upgrade the computer we were using, if we wanted to record more than 8 tracks simultaneously,and being a very small company that was a expense we could do with out at that moment.But boy are we happy with the investment.(the musicians too)

 
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