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guitar junkie Offline
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PostPosted: Sat Aug 18, 2007 8:02 am    Post subject: Reply with quote

Dude you can discuss the war here... but telling everybody that you have the whole thing figured out and why it HAS to be that loud is not what the board is about... I like to have healthy tracks myself... But at the same time I have heard some that needed to be mastered a lot lower than they were... I have a volume knob on my player.. I can turn it up when I need to but I hate having to jerk around real fast and turn it down at three in the morning to keep the other people in the house from waking up!

And my guess is that you have not been around here long enough to know much about Harris and his tunes... If you have not heard them then take a trip over to his site... I am sure that you have not yet touched the stuff he has been doing for the last 15 years.

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PostPosted: Sat Aug 18, 2007 12:31 pm    Post subject: Reply with quote

guitar junkie wrote:
Dude you can discuss the war here... but telling everybody that you have the whole thing figured out and why it HAS to be that loud is not what the board is about...

And my guess is that you have not been around here long enough to know much about Harris and his tunes... If you have not heard them then take a trip over to his site... I am sure that you have not yet touched the stuff he has been doing for the last 15 years.


I'm not a "dude", I'm a girl. bad girl. Very Happy
I'm sure that most of the ppl here do music more than 10 years...I myself started with computer music since I was 12, before that I was playing the piano since age 6+- and than a synthesizer and a guitar.

These days (and since 8 years already) the music I do is not just notes....it's all about production.
This is why I will to learn more and more.

I learn new things day by day...an even if I wrong about something it leads me to many other brilliant ideas.


Back on topic, what I been thinking, and still does somehow...,
when you make it loud as much as possible without ruining important dynamics at 24bit , you don't raise the noise floor, you actually save better details that you don't want to integrate with the noise floor after you save as 16 bit.

Now please don't be sarcastic because it make lots of sense to me, explain me if I wrong.
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guitar junkie Offline
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PostPosted: Sat Aug 18, 2007 1:54 pm    Post subject: Reply with quote

aye its a chick! Well then I suppose you should team up with Jaimee Harris and see how things go from there....
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pglewis Offline
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PostPosted: Sat Aug 18, 2007 2:51 pm    Post subject: Reply with quote

dreamzcatcher wrote:
Back on topic, what I been thinking, and still does somehow..., when you make it loud as much as possible without ruining important dynamics at 24bit , you don't raise the noise floor, you actually save better details that you don't want to integrate with the noise floor after you save as 16 bit.


Really, with 24-bit source files you just need a "healthy" level. Just stay well above the noise floor, which should be very easy, even with average home recording equipment.

16 bits (as I'm sure you are aware) gives you much less dynamic range to work with vs. 24 bits. If you're down at, say, -18dB then you've lost 3 bits off the top and you are now working with 13 bits of audio resolution. Also, at this point, you will NOT REGAIN the lost resolution by making that 13 bit audio file louder. The source is of poorer resolution and you cannot get the resolution back by simply making it louder after the fact. Just as you cannot take a poor quality image and magically make it high quality by simply increasing the image resolution.

I still think you would find it difficult or impossible to tell the difference between that example (losing 3 bits) and a file which maintains as high a level as possible when both are played back at the same volume. I continue to urge you to setup a proper double-blind test, it may be very... umm... ear-opening Smile. We listen to music with our ears, not digital audio theory or screen-shots of waveforms.

But this knowledge is something to keep in mind, especially when it comes to fade-ins and fade-outs. Fades do indeed really get down into the lower resolution and the signal approaches the noise floor. So do all fades and all processing at 24 bits and dither down to 16 bits as the VERY last stage of your processing (to preserve as much detail as possible). This is solid advice you will find all over the forums, it is not news.

When working with 24 bits of resolution, however, you'd have to lower the level 48dB before you've even dropped down to 16 bits of resolution. That is TONS OF HEADROOM. You will waste your time sweating useless details if you worry too much about keeping the level as high as possible with 24 bits of resolution to work with. It is also potentially bad advice to newbies who start thinking "OMG, I've gotta keep everything incredibly loud or I'll lose resolution!".

There are a billion factors that will degrade your audio quality long before bit resolution in 24 bit source files becomes an issue. Worry about those billion things first.

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pglewis Offline
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PostPosted: Sat Aug 18, 2007 3:14 pm    Post subject: Reply with quote

dreamzcatcher wrote:
how sad

37 years old musician/engineer being sarcastic to a 24 years old artist.
you don't like your job that much i guess...


Careful about assumptions, Chris is lawyer Very Happy. He's also quite a talented musician and songwriter, and he has an uber-talented daughter who is quite a bit younger than you. They make some great recordings.

I think it's a shame that no one else seems to follow what you're saying because I think it's just making you more stubborn Smile.

But it kinda reminds me of a crazy guy who goes on and on about saving gasoline by removing his antenna. He'll insist his facts are solid and go on an on about aero-drag, show you mathematics and wind-tunnel experiments to prove that the antenna increases drag and will therefore lower the vehicle's gas mileage. He'll preach to the world about solving the energy crisis by removing their antennas too, as he drives off dragging a boat anchor on a rope.

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PostPosted: Sat Aug 18, 2007 5:06 pm    Post subject: Reply with quote

dreamzcatcher wrote:


how sad

37 years old musician/engineer being sarcastic to a 24 years old artist.
you don't like your job that much i guess...


Look, sorry if I hurt your feelings, but I'm usually more impressed with peoples' recordings than with their claims of fame. The internet is a funny place where you can say anything you like about yourself, and I'm pretty distrustful of people in general, so when I see somebody post pictures about playing in festivals in front of hundreds of thousands of people, it pushes all of my buttons.

I appreciate the kind words that some here have said about my own recordings, but the truth is I'm a rank amateur and I make no bones about it. I toured for years and didn't have the drive to keep going on the road, so I turned my passion for music into recording. In an amateur fashion. I'm no pro, and if you are, more power to you.

Post some recordings...let's be friends and talk about bits and stuff.

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M.Brane Offline
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PostPosted: Sat Aug 18, 2007 6:03 pm    Post subject: Reply with quote

dreamzcatcher wrote:


Back on topic, what I been thinking, and still does somehow...,
when you make it loud as much as possible without ruining important dynamics at 24bit , you don't raise the noise floor, you actually save better details that you don't want to integrate with the noise floor after you save as 16 bit.

Now please don't be sarcastic because it make lots of sense to me, explain me if I wrong.


Ah, but the noise floor even at 16bit is way below even the best analog gear. Cranking up your mic pres beyond their useable range just to make use of the upper bits only creates poor sound, and a higher noise floor.

Many truly great recordings were made on analog medium that had around only 50-60db of useable dynamic range. The difference is tape itself has a relatively high noise floor compared to even 16bit digital, and tape compression from driving levels hot into the deck is often a desired sonic effect. Clipping like that in digital is a death sentance for your audio.

The noise floor of 24bit is near the atomic level. Much lower than any mic, pre, or A/D converter. Leaving bit headroom during tracking/mixing makes it easier to mix without having to constantly worry about clipping busses, and plug-ins. Any leftover headroom after the mix is done can then be compensated for with gain in mastering for maximum impact, and clarity.

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PostPosted: Sun Aug 19, 2007 3:48 pm    Post subject: Reply with quote

pglewis, I like to read your replies.
You are a smart person Smile

You made it more clear for me. thanks.

ps - Chris, I will post some stuff later on. you seems to be more friendly to ppl you know more . Smile
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pglewis Offline
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PostPosted: Mon Aug 20, 2007 9:01 am    Post subject: Reply with quote

dreamzcatcher wrote:
ps - Chris, I will post some stuff later on. you seems to be more friendly to ppl you know more . Smile


lol... if you see how he treats his friends you'll know that you don't want to be his enemy Very Happy. I kid, I kid. I love Chris, we're both becoming cranky old farts at a similar pace.

To sum up, here's what I'd keep in mind on these things:

*) At a large festival there is too much environmental noise for truly critical listening. It's certainly not going to reveal any flaws that are not revealed in a decent monitoring room.

*) 24 bits gives you plenty of resolution. Even if your level is averaging way down at -48dB, it still averages a full 16bits of resolution. Record at 24 bits. Process internally at as high a resolution as your software and computer can handle. If you're mastering yourself, render your mix at 24-bits, do your fades and mastering at 24-bits, and at the very end dither down to 16 bits and do not process that file any further. 16 bits is more limited for processing, so avoid it when at all practical. If someone else is mastering, give them the 24-bit stereo mix and don't ever touch 16-bits.

*) Like MBrane said, Leave yourself some headroom when tracking. Getting the level as high as possible is fine if you don't go over, but remember that all your tracks will be added together. If they're all fully normalized, you're just gonna end up bringing the faders down anyway to keep from peaking as you add more tracks. In a complex project it can be tricky to find that one buss that's clipping and ruining your whole recording. Remember: if you're just going to bring the faders down anyway, it's the same thing as tracking a little lower.

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PostPosted: Thu Aug 23, 2007 9:46 am    Post subject: Reply with quote

Slashdot joins the discussion:

http://science.slashdot.org/article.pl?sid=07/08/23/1219205&from=rss

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PostPosted: Mon Aug 27, 2007 6:41 am    Post subject: Reply with quote

Just jumping in from a practical side of things. The address below is for the listening room of my military band.

http://www.usma.edu/band/recordings/bcdmain.htm

If you follow the page down you will see a six volume collection of CDs recorded over a four year time frame. We used the same room, mics, placement, mic-pres, cables et al. The only factor that changed over the process was how much headroom we gave to the original recording. In volume one and two we did push much of the material right up to 0 dbfs. In volumes 3-5 we used -12 dbfs as a target point for the peaks. The end result was the mastering phase in volumes 3-5 allowed for more freedom and room for final master. The sound files are all in MP3 but I think you might be able to hear the change. Frankly have not listened to them in a while but it seemed like these might be good sonic examples.

I nearly always when doing current recordings shoot for -12dbfs as the cut off point.
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GT Offline
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PostPosted: Tue Aug 28, 2007 10:20 am    Post subject: Reply with quote

dreamzcatcher wrote:
pglewis, I like to read your replies.
You are a smart person Smile

You made it more clear for me. thanks.

ps - Chris, I will post some stuff later on. you seems to be more friendly to ppl you know more . Smile


OH, the old devide and conquer ploy!!

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PostPosted: Sun Oct 14, 2007 8:33 am    Post subject: Reply with quote

I had a really interesting conversation with an ME last month - discussing this whole topic ...

What I learned is that when you record at 16-bits, it is more than likely going to be beneficial to use compression on the input signal - this is to ensure that you encode as many steps as you can and is strictly a limitation of the nature of 16-bit digital.

dreamz - coming back to your picture of the waveforms earlier - I think it shows exactly the disastrous effects of not dithering! What I believe happened in your experiment is this: when you lowered the sound to -80dB, you were roughly 3 bits away from the 16-bit noise floor (-96dB). So when you saved the file to 16-bits without dithering, you ended up with a whole lot of quantization distortion, which became much more noticeable when you normalized the track to 0dB.

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PostPosted: Tue Dec 11, 2007 12:34 pm    Post subject: Reply with quote

I just got to this discussion -- sorry I missed it.

What no one pointed out is ... in digital ... there is no resolution -- no deeper clarity of sound through bits. More bits makes a lower noise floor possible, hence increased dynamic range -- but it's only that -- nothing else. There is no heightened sensitivity to either frequency or volume.

Since "resolution" isn't in the digital world of audio relative to bit depth, it is really irrelevant to this whole discussion. Bit depth allows detailed mapping of volume and volume alone ... nothing more. Volumes have been written on this.

One has to remove correlating the concept of "resolution" with bit depth from their heads. It's not resolution.

I'll see if I can locate a link to an online, in depth discussion of the matter.

Best,

Kev-

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PostPosted: Tue Dec 11, 2007 2:10 pm    Post subject: Reply with quote

Got it ...

http://recforums.prosoundweb.com/index.php/m/4131/0/?srch=bit+depth#msg_4131

It's a HUGE online discussion, spanning 19 pages and a full year's worth of information, but -- it's probably one of the most informative by folks with impressive knowledge.

Best,

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PostPosted: Thu Dec 13, 2007 2:39 pm    Post subject: Reply with quote

All depends on the sampling rate. At 16bit, yes a little bit mattered you needed to be up in -4 to -8 range. At 24 bit you can comfortably mix at -12. At 96 bit.. do what ever you want, you will still have great sound.
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PostPosted: Thu Dec 13, 2007 5:59 pm    Post subject: Reply with quote

96 bits? At 6db per bit (linear PCM) that would give you 576 db of dynamic range. That's awful loud at full-scale.

Samplerate is what determines your bandwidth. Bitrate gives you your amplitude.

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PostPosted: Thu Dec 13, 2007 6:30 pm    Post subject: Reply with quote

M.Brane wrote:
96 bits? At 6db per bit (linear PCM) that would give you 576 db of dynamic range. That's awful loud at full-scale.

Samplerate is what determines your bandwidth. Bitrate gives you your amplitude.


Exactly correct.

And for folks that question why there is no such thing as "resolution" in digital audio ... the answer, in long form ... lies there. Digital audio is not analogous to video.

Like I said above. Read the other forum. There is a lot of crap in in from folks who don't know shit. But the ones that REALLY do impart HUGE volumes of "need to know" information. Nika Aldrich writes books on the subject, and knows exactly what is going on.

Kev-

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PostPosted: Thu Dec 13, 2007 7:54 pm    Post subject: Reply with quote

For me it's as simple as that: I hear 3db in gain reduction by a compressor or limiter, but I don't hear a thing when I turn the volume down by much more.
So I will only use compression only when I want the effect of an compressor. Once I get too hung up on the details I loose the big picture. There are more important things than resolution and noise-floor.

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PostPosted: Thu Dec 13, 2007 11:00 pm    Post subject: Reply with quote

K-dub wrote:
And for folks that question why there is no such thing as "resolution" in digital audio ... the answer, in long form ... lies there. Digital audio is not analogous to video.


What you say may be technically correct, but people talk about resolution in digital audio all the time and get understood, so for me it's a useful term cuz it communicates successfully. But if 96K is not a 'higher res' sample rate than 44.1K, then what adjective and noun do you use to talk about recording settings that capture more pieces of information per unit of time than other settings?

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PostPosted: Thu Dec 13, 2007 11:20 pm    Post subject: Reply with quote

Capturing more samples may or may not be beneficial depending on the converters you're using.

There are many other factors that come into play in conversion beyond bit, and sample rate. Ironically it's the analog components in conversion that often are the culprits in poor quality conversion. There is also clocking (timing of the samples AKA jitter).

In recording these things are generally secondary considerations to input source however. I.E. unless your input sources are of the utmost quality there is little to be gained from having top quality conversion.

The advantage to using higher sample rates with low-end converters is it pushes artifacts higher up in the frequency spectrum which can (hopefully) be dealt with better ITB with good SR conversion.

Personally I'd rather use a better quality converter at 16/44.1 than a cheap one at 24/96, but YMMV. You can always add empty bits to the bottom of a file ITB for processing without sonic penalty, but if your audio is hosed on the way in you are screwed indefinitely.

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PostPosted: Thu Dec 13, 2007 11:23 pm    Post subject: Reply with quote

I woulda thought most dedicated soundcards did a decent enough job of conversion so that you don't have to worry about that stuff much?
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PostPosted: Thu Dec 13, 2007 11:28 pm    Post subject: Reply with quote

dobro wrote:
I woulda thought most dedicated soundcards did a decent enough job of conversion so that you don't have to worry about that stuff much?


Well that's what the marketing departments would like you to believe. Laughing

For most folks the average modern soundcard will be great. For those of us who grew up with analog the modern digital world has just as many warts as the old analog world did. They are just different warts.

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PostPosted: Fri Dec 14, 2007 9:48 am    Post subject: Reply with quote

M.Brane wrote:
dobro wrote:
I woulda thought most dedicated soundcards did a decent enough job of conversion so that you don't have to worry about that stuff much?


Well that's what the marketing departments would like you to believe. Laughing


Laughing

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PostPosted: Fri Dec 14, 2007 10:31 am    Post subject: Reply with quote

Actually, it was touched on above. "Resolution" isn't the term. It's inappropriate.

Greater "Bandwidth" (freq response) and "Dynamic range" (amplitude response) are the right terms, resulting respectively from higher sample rates and deeper bit depths.

K-

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