Forum: n-Track Studio Discussion Forum
Topic: 16 Bit vs 24 Bit Audio
started by: dannyraymilligan

Posted by dannyraymilligan on Oct. 07 2010, 6:36 AM
Okay, moving this over from another thread, I'd like to pick up where TomS and I left off.

The discussion began because Tom was concerned about my use of 16 bit plugins such as blueline in processing. As I noted to him in a response, even if you have a system capable of processing the information in 64-bit mode (which, as I understand it, isn't really 64-bit but two 32-bit floating point registers), the very moment you mix the track down to cd audio, lossless, or anything else that uses PCM, you go back to 16-bit, and defeat the whole purpose of using 24 bit, in the first place.

Now, my soundcard is not 24 bit, only 16. Like all 16 bit soundcards, it has a decibel response MAXIMUM of 96db. 24-bit sound cards have a signal to noise ration of 128db. Now this is a monstrous difference in bandwidth, as you might guess. However, aside from RedBook and Super Audio CD format (Audiophile equipment) there is NO WAY to listen to your audio in 24 bit unless you have a card with 24 bit. There are no 24 bit cd players. They are all 16-bit, 44.1 khz PCM.

In this case, you could listen to a song you recorded in 24 bit and get the full response of your audio, but you cannot transfer that to CD and keep the FULL song. It will be converted by the system to 16 bit, and there's nothing any of us can do about that. I'm not up enough on the technology, but as I understand it, Analog was not subject to the limitations of Digital Audio, so the frequency responses were much better in Analog than in Digital. I imagine there are still recording studios out there that will do completely analog, but the same limitations exist for them that exist for us: the moment the music is converted over to where it can be put on a CD, it is rendered as 16-bit, and you're back to where you started.

:whistle:

So now, knowing that we cannot get our music out to the public in any format EXCEPT 16-bit, I fail to understand where it is an improvement to record something in AWESOME QUALITY if no one will ever hear it but myself? All that would do is bring me down, I think  :heart-break:

At this time, I'm planning my next computer, which will be an AMD 6 core, 64 bit processor, with probably 8 gigabytes of RAM. Why? So I can work on my music without having to mixdown in stages just so I can still get all the tracks onto one song. I dream of being able to run a true 48 track studio out of my home with as many effects as I want on each track. I could probably run 24 tracks now, as long as I used no effects, but the moment I add reverb or eq plugins to a track (let's not even go INTO what pitch shift does to your cpu load, lol!) then my machine would bog down on me.

Any ideas? Criticisms? Disagreements? Here's the place for them  :)

-Danny
Posted by dannyraymilligan on Oct. 07 2010, 6:44 AM
Question: Would changing my preferences help me get a better end result? Right now, I'm recording in 44.1 khz. What if my wavs and mixdowns were done in 96 khz format? Would my resultant 320 kbps mp3's sound better than the ones done from 44.1 khz master files?

Hmmmmm, time to play with my computer, lol!
Posted by nick on Oct. 07 2010, 12:47 PM
My understanding is that the advantage of 24 bits is that it gives you the ability to record with more headroom when recording signals with uncontrolled levels like live musicians. If you record with peak levels of say -12dB in 16bit resolution you are actually only using 14 bits to represent your sound and things are getting a bit marginal. With 24 bit you can record and peak at say -18dB and you are still getting 21 bits of available resolution.

Some people think the digital "mix bus" sounds better with signals peaking around -18dB and if you sum a number of tracks your overall level will increase anyway, (there's no need when working in 24 bit to try and get each individual track peaking near to zero.)

Once you are dealing with a controlled signal level and you can safely peak to near 0dB then 16 bit is probably good enough for most people.
Posted by dannyraymilligan on Oct. 07 2010, 3:07 PM
Okay, Rick, that made sense to me, lol!

There is still the issue of the mixdown, however. Whether you make the end product into mp3, flac or wma for the web distribution, or you burn it to CD, it's still going to be 16-bit, so your headroom is gone. Another thing I just thought of. Probably 80% of all computer soundcards are 16-bit. Only gamers and musicians are buying the 24-bit soundcards, the general public could care less what theirs is. So when you're playing a DVD on a computer, you're not getting the 24-audio then, either, the soundcard has to squeeze it into 16-bit to play it, right?

-Danny
Posted by bbrown on Oct. 07 2010, 5:02 PM
I found this wikipedia article on quantization which makes some points about recording audio in 24-bit. < http://en.wikipedia.org/wiki/Quantization_(sound_processing) >
Posted by dannyraymilligan on Oct. 07 2010, 5:07 PM
Quote: (dannyraymilligan @ Oct. 07 2010, 8:07 AM)

Okay, Rick, that made sense to me, lol!

There is still the issue of the mixdown, however. Whether you make the end product into mp3, flac or wma for the web distribution, or you burn it to CD, it's still going to be 16-bit, so your headroom is gone. Another thing I just thought of. Probably 80% of all computer soundcards are 16-bit. Only gamers and musicians are buying the 24-bit soundcards, the general public could care less what theirs is. So when you're playing a DVD on a computer, you're not getting the 24-audio then, either, the soundcard has to squeeze it into 16-bit to play it, right?

-Danny

oops, Sorry Nick, I wrote your name as rick, lol
Posted by woxnerw on Oct. 07 2010, 5:33 PM
Hi Gents:
 I'd like to add my .02 cents here.. for what it's worth. 

 I'd like to think that the resolution of a .wav and their specs. can be measured in two ways..

 Vertical amplitude is referred to as bit resolution..  e.g.  16-bit   24- bit..   and....    Horizontal amplitude is referred to as Sample resolution..  e.g. 44.1 kHz.  48khz.   and so-on up to 96 kHz.  for .wav files..  mp3 files   the horizontal resolution..  e.g.  at variable bit rate or 320 kHz.  .etc...    

 My easy understanding of these specifications goes something like this..

  Take a photo using an old analogue camera with film..   e.g.   place the negative in an enlarger to enlarge the image from say 5" x 7", to say, 8" x 10"..   A good camera with a good lense cam produce an image on the film that shows as well at a small enlargement as the same negative when enlarged to any size..   without appearing as a grainy reproduction..  

 A digital .wav file with poor A/D conversion specs, e.g. poor or low bit-and-sample specs will be grainy when plugs are added to the file when editing and mastering..   The more editing-and-mastering the poorer the file will sound as the number of times the file is manipulated.. till it finally gets to the CD for distribution..  Then the buyer says..  that sounds good..  or..  the buyer says what happened to the sound on that CD ????    

 On the other hand..   Each time the .wav file is edited-and-rendered, it looses it's original pristine reproduction..    Then again....  It's best for the .wav file to start off with the best analogue electronics e.g.   Good Dynamics, as it (the signal) enters  the A/D converters, then return to analogue via the D/A Converter with the best analogue reproduction possible..

 Even the quality of the digital converters is important, to the process..

 That makes for good audio hearing.. In My Opinion..    

        Bill..
Posted by Paco572 on Oct. 07 2010, 6:22 PM
The key word here is "dither" a complex process of converting 24bit to 16 or even 8bit audio.

For example if you record at 16bit 41,000khz, you can mix down your song within n-track without the need to "dither" and take it straight to CD. However if you record at a higher bit rate such as [email protected],000khz you'll need to enable dithering from the mixdown window options, provided you have done no dithering at this point. Although this is just a very basic method, it will process the audio.

From what I understand and this is just a basic understanding, the wave form of a 24bit audio, say a line that goes up and then down will be made up of series of many points tightly packed together (24bit tight), when you covert to 16bit without dithering, it removes every other point from the 24bit wave form leaving a sixteen bit wave form with gaps left behind. These gaps have no audio, resulting in the sample being lower in volume and quality.

With dithering, those empty gaps/points are replaced with noise, thus allowing the audio to maintain it's volume with very little loss of audio or quality. There are many mastering/dithering programs in which different dither methods for different situations but the whole idea as I understand it is fill the gaps with the right kind of noise or delete them.
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For example taken from within OZONE 3 various methods using the OZONE 3 app, (I don't understand a good portion of this stuff but it does give me an the idea behind it)

Type:
MBIT+: This is a proprietary iZotope word length reduction technology that reduces quantization distortion with minimal perceived noise. While this might sound like a paradox, MBIT+ is a very smooth, quiet and almost "analog sounding" technology.

Type 1: Dither is applied using a "rectangular" distribution function. While this provides a dither noise source with a low amplitude, the dither noise can become modulated by the audio signal and vary in level, which is undesirable in many situations. Also, the non-linear quantization distortion is not completely suppressed in some situations with this low dither amplitude.

Type 2: Dither is applied using a "triangular" distribution function. This dither is larger in amplitude and completely suppresses the non-linear quantization distortion.


Shape

By shaping the dither noise, it is possible to provide more effective and transparent dithering by shaping the dithered noise spectrum. There are several different methods for shaping noise so that it is less audible yet still effective. Please refer to our online mastering guide for more information as to the technology behind these methods and how to apply them effectively.


Type 1 or Type 2 Shapes

None: No noise shaping is applied

Simple: High pass filtering is applied to the dithered noise.

Clear: The noise is shifted towards the Nyquist frequency, near the upper limit of our hearing.

Psych 5: A fifth order psychoacoustic shaping is applied to provide dither across the spectrum. The shaping is designed to move the noise away from frequencies that are heard as "louder" at low levels.

Psych 9: A more complex ninth order psychoacoustic shaping is applied.

In general, the "Clear" option is a safe bet for complex program material, although auditioning the dither against the Psych 5 and Psych 9 shapes may be more desirable in some cases.

Please note that Psych 5 and Psych 9 shapes are specifically designed to be used on audio with a 44.1 kHz sample rate. For other samples rates, use None, Simple or Clear shaping, or MBIT+ mode which is designed for effective word length reduction at any sampling rate.

MBIT+ Shaping

The MBIT+ dither technology also provides options for noise shaping. You can control the aggressiveness of this shaping, ranging from None (no shaping) through Ultra (roughly 14 dB of audible noise suppression).


Bit Depth

This is the target bit depth for the audio. For mastering for a CD, for example, you would want this set to 16.

Note that Ozone does not perform the actual conversion of the audio. After processing a mix with Ozone, it is necessary to then actually convert the audio to the desired bit depth in the host application. For example, if you have a 24-bit audio file, you can use Ozone to dither down to 16 bits. The remaining 8 bits are "padded" as zeros. Your file is still a 24-bit audio file, there's just not anything but zeros in the lowest 8 bits. So when you then convert to a 16-bit file in the host app, the 8 bits (that didn't have any audio in them) are discarded.



With this process in mind:

1) Do not perform any processing to the audio after it has been dithered with Ozone. You may perform level adjustment with the output gain sliders in Ozone (those come before the dither) but do not change any levels in the host app or with other plug-ins. Almost all host apps have their master faders after the effects slot, so any level adjustment in the host app will destroy the dither.

2) Do not put any plug-ins after Ozone if you are dithering with Ozone. The dither must be the last thing that touches the audio.

3) Turn off dithering in the host app. Basically, you just want to truncate (throw away) the bits, because they're just zero anyhow.


Num Bits or Dither Amount

This sets the number of bits or amount of dither that will be  used for the dither source. For Type 1 and Type 2 dither, in most cases 1 bit will be sufficient, but in some situations the "over-dithering" obtained by setting Num Bits to 2 can be useful. In MBIT+ mode, the dithering amount can be varied from None (noise shaping only) to High. No dithering or Low dither amount can leave some non-linear quantization distortion or dither noise modulation, while higher settings completely eliminate the non-linear distortion at the expense of a slightly increased noise floor. In general, the Normal dither amount is a good choice.


Auto-blanking

Selecting this option instructs Ozone to completely mute dither output (i.e. dither noise) when the input signal is completely silent (0 bits of audio) for at least 0.7 seconds.

Limit Peaks

Dither noise is random in nature and has a very low amplitude. However, after noise shaping, especially in aggressive dithering modes, the high-frequency dither noise is significantly amplified, and the overall dither signal can show spurious peaks up to -60 dB FS. If such high peaks are undesirable, you can enable the Limit Peaks option to effectively suppress the spurious peaks in the noise-shaped dither.

Suppress Harmonics

If, for some reason, any dithering noise is undesirable, simple truncation remains the only choice. Truncation results in harmonic quantization distortion that adds overtones to the signal and distorts the timbre. In this case you can enable Suppress Harmonics option to slightly alter the truncation rules, moving the harmonic quantization distortion away from overtones of audible frequencies. This option doesn't create any random dithering noise floor. Instead it works more like truncation, but with better tonal quality in the resulting signal. This option is applicable only in the modes without dithering noise and without aggressive noise shaping.
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I hope this gives you an idea of what it takes to enjoy your 24bit audio in a 16 bit format.

PACO
Posted by bax3 on Oct. 07 2010, 7:05 PM
Warning: Nurd alert!  This can cause your eyes to cross . . .
The photo analogy works pretty well, but you can do the comparison in digital as well.
As I understand it:
Think of digital music as a camera that is taking pictures of the sound:
The 44,100 or 96000 is the number of times the computer "takes a picture" of the sound being produced - the number of times the sound is "sampled" in a second.  So, if you have a very complicated sound, an orchestra for example, you can raise the sample rate (the number of pictures" taken) to get more detail - a movie at 16 frames per second or at 35 frames per second, the slower frames per second may be a bit "blurry" , but it doesn't matter so much if the subject is not moving very fast.  Neither may truly be fast enough to capture everything that is happening, but the human mind fills in the spaces.
The bit depth is the "resolution" of each sample - the amount of information that can be stored in each "picture."  More information is stored in each picture.  This means that the recording can handle a wider range of information in each picture (more head-room is the analog term), the chances of "overloading the sample" is reduced.  So, recording at 24 bit can avoid clipping because it has a wide dynamic range before there is more information than can fit on the picture. *1
The question of how much all this matters is a long standing debate.  The advantage of 24 bit is pretty easy to see, it makes recording easier and when you dither down to 16 bit you are throwing away sounds that are not in the hearing range of humans.  So, does it matter if you record in 24 bit - well, it does have the advantage of avoiding clips and some people believe they can hear the difference - I can't hear the difference, but I record in 24 bit sometimes to avoid clips.  Recording at 96k, well, it make huge files and again I can't hear the difference, but I mostly record singers and guitar players.
As as been mentioned, CD quality is 44,100 at 16k.  That is pretty good for most everything I have ever recorded.  If I was going to use DVD that will take much higher rates, and that may be the future.

*1 Digital sound is made of 0s and 1s. Think of the sum as the amount of energy that is being created.  Comparing to an electrical circuit: (I have no idea the actual comparisons) think of 16 bit as a 15 amp circuit, 24 bit as a 60 amp circuit. If too much information (sound/ Energy) is sent to the digital math the sum comes to more than the computer is set up to handle, and we get a "clip" - we have "overloaded the circuit" and an unpleasant noise is produced.
confused yet?
Bax
Posted by TonyR on Oct. 07 2010, 8:13 PM
I want brain-to-brain Bluetooth.
Posted by TinaM on Oct. 07 2010, 8:56 PM
i want a brain!


TinaM
Posted by bax3 on Oct. 07 2010, 9:10 PM
The short answer is - unless or until you find that it is not sounding the way you want - 16 bit 44,100k will do the job.
All of the other stuff we are talking about can improve a mix OR really screw it up.  
So, to get good everyday recordings standard 16 by 44,100 work great!  In general, the less you do to a mix past re-balancing the volumes and panning is nuance or more to the mastering side of recording.
Posted by TonyR on Oct. 07 2010, 9:17 PM
"Bang bang Baxwell's sivler hammer....."
:laugh:
Posted by bbrown on Oct. 07 2010, 9:17 PM
I would have to agree with bax... everything I have read so far about converting from one bit-depth to another involves some form of dithering (as Paco mentioned) and therefore its possible for the integrity of the original track to be degraded by the conversion process.
Posted by dannyraymilligan on Oct. 08 2010, 12:08 AM
Did audio techs and home recording enthusiasts have to worry about all this back in the days of analog???

OY VEY!!!!
Posted by TonyR on Oct. 08 2010, 12:49 AM
Aye, Danny. Stick on a bit of Leadbelly or Ottlie Patterson, Motown....
Still sounds magical:-)
Posted by bax3 on Oct. 08 2010, 3:02 AM
I think we worry about all this technical stuff because now we have the capacity to mess with it.  You can do some great stuff with digital recording and I love it -HOWEVER,
With analog, you went for the performance, that was the first measure of the recording and should still be!  Think about a recording session in about 1930: everyone is in one room and the recording is literally cut into the record as the performance is done.  You get it right or you do it again.  
I was a performer in the late 50's and 60's.  We went to studios and paid good money to record on mics that were not as "good" as today's, with 2 tracks if we were lucky - some of that stuff sounds great!  Not pristine like a digital, but the performance is there and if it wasn't there, then little or nothing we could do about it, except record it again.
All that said - I still love what I can do with NTrack to make a "professional" sounding recording.
Bax
Posted by Paco572 on Oct. 08 2010, 3:51 AM
Quote: (TinaM @ Oct. 07 2010, 1:56 PM)

i want a brain!


TinaM

You can have mine, I'm getting a new one next week that can,

A: Drink without getting a hangover.
B: Remember.
C: Know how to change the bobbin on the weed whacker.
D: Has interchangeable parts that can be upgraded.


PACO :whistle:
Posted by TomS on Oct. 08 2010, 4:31 AM
Quote: (bbrown @ Oct. 07 2010, 2:17 PM)

I would have to agree with bax... everything I have read so far about converting from one bit-depth to another involves some form of dithering (as Paco mentioned) and therefore its possible for the integrity of the original track to be degraded by the conversion process.

Only when going from higher to lower.
Posted by TomS on Oct. 08 2010, 4:38 AM
Quote: (dannyraymilligan @ Oct. 07 2010, 5:08 PM)

Did audio techs and home recording enthusiasts have to worry about all this back in the days of analog???

OY VEY!!!!

Yes - tape alignment!  Also, gain structure, tape degradation, wobble, spindle fold and mutilation...

I'm not sure the discussion up to this point addressed my original concern.  Even if you record at 16 bits, when the DAW processes it, it's at a much higher bit rate, e.g., 64 bits.  Why?  This is needed because so many computations are going on that the numbers quickly get really big.  And all of those numbers represent audio information.  Now, imagine you have info at 64 bits going into a plugin that truncates everything after 16 bits.  All of that other information is just lost.  This results in degradation in audio.  And it is totally audible.  I think it was MDA who had a plugin that tracked the actual bits used in other plugins.  

In sum: you may be recording at 16 bits, but the calculations in the program use a much longer bit rate, to preserve the audio.  If no calculations were done ever with the signal as it came in right after AD conversion, it wouldn't matter, but lots of calculations are done, and hence it matters if a 16 bit bottleneck is inserted in the middle of them.
Posted by dannyraymilligan on Oct. 08 2010, 4:39 AM
lol Tom, going from lower to higher is useless too, ya know? Imagine taking some old 64 kbps audio file (anyone remember real audio format? :p ) and trying to resample it as FLAC... all you'll get is a lossless copy of the old crappy track :p

"No matter how long you polish a turd, it will never shine"

:laugh:

-Danny
Posted by TomS on Oct. 08 2010, 4:41 AM
So, back to the blueline plugins.  If you like what they sound like, then use them!  But it might be interesting to compare them side-by-side with some newer ones.  I dunno how the newer ones would work for you, given how far along things have come since the version of n-Track you are using, but you should absolutely be able to hear a difference.  Perhaps not one you like, but then again perhaps one you would.
Posted by TomS on Oct. 08 2010, 4:43 AM
Quote: (dannyraymilligan @ Oct. 07 2010, 9:39 PM)

lol Tom, going from lower to higher is useless too, ya know? Imagine taking some old 64 kbps audio file (anyone remember real audio format? :p ) and trying to resample it as FLAC... all you'll get is a lossless copy of the old crappy track :p

"No matter how long you polish a turd, it will never shine"

:laugh:

-Danny

Different issue.  There is no loss of information going lower to higher, so no dither is needed.  But - if you go lower to higher and then do something with that new file, and then go back to lower (which is what is going on in the DAW) then you will necessarily lose information.  

Anyway, no one here records turds!  We are brilliant!
Posted by dannyraymilligan on Oct. 08 2010, 4:45 AM
Quote: (bax3 @ Oct. 07 2010, 8:02 PM)

I think we worry about all this technical stuff because now we have the capacity to mess with it.  You can do some great stuff with digital recording and I love it -HOWEVER,
With analog, you went for the performance, that was the first measure of the recording and should still be!  Think about a recording session in about 1930: everyone is in one room and the recording is literally cut into the record as the performance is done.  You get it right or you do it again.  
I was a performer in the late 50's and 60's.  We went to studios and paid good money to record on mics that were not as "good" as today's, with 2 tracks if we were lucky - some of that stuff sounds great!  Not pristine like a digital, but the performance is there and if it wasn't there, then little or nothing we could do about it, except record it again.
All that said - I still love what I can do with NTrack to make a "professional" sounding recording.
Bax

I often think about that, Bax. What would "Let It Be" or "Hey Jude" have been if the Beatles had possessed the technology we have now?

Can you imagine Elvis Presley toting around a laptop back in 1959, and noodling around on a guitar in some motel room while he tried to figure out which plug-in sounded best on the guitar for "Hound Dog"?

Some of the greatest records in the world were cut in a matter of a month or two, altogether, and are still LOVED 30 years later, yet bands will go in the studio now and spend a YEAR making a record, and not one memorable song or anthem in the lot :( ....

Anyone remember Hotel California, Back In Black, or Dreamboat Annie?

-Danny
Posted by TomS on Oct. 08 2010, 4:52 AM
I dunno, it's not like the Beatles didn't have really fine equipment.  I will never own a Fairchild compressor.  Sadly.   Or a really good room.  Or have their talent.  

Personally, I think that if they had had all this great stuff, they would have made records that were even better.  If possible.   :)

Mix magazine has a review of a new version of the Fairchild 670 for only 19000 usa dollars.  here:

< http://mixonline.com/gear/reviews/analoguetube_at101_limiter/ >
Posted by dannyraymilligan on Oct. 08 2010, 6:19 AM
To be honest with you, Tom, I believe the Beatles talent lay in songwriting skills. Musically (as in instrumental proficiency)... well, I won't bash anyone's deities, but the Beatles weren't the Gods they thought they were. True, it's quite possible that the technology of the time limited the scope of what they could do, yet I believe that the idolatry of this group proceeds from the fact that they were really the first to do what they were doing at the time.

As an example, mind you, I point to my favorite group of all time, KISS. I love their early records, and am the first to admit that Ace Frehley is the reason I first picked up a guitar. However, when I look at it objectively, I realize that I can play circles around Ace as a musician, but I can recall back in the early 1980's when I would have beat the snot out of anyone who suggested that Eddie Van Halen was better than Ace Frehley.... pleaseeeeeeeeeeeeeeeee!!!

I think my attitude when I was younger reflects a lot of what people think about groups such as the Beatles and Led Zeppelin. Because they influenced those of us who came later, somehow we've put them on a pedestal, yet, if you actually look at what they did, it wasn't earth-moving compared to what we can do now.

Tom, I've listened to a great deal of your own music, and let me tell you this, in all candor: I would buy a CD of yours much quicker than I would anything of the Beatles catalog.

Still, one wonders what the Beatles might have been, had they grown up with the influences that we grew up with, and had the techniques and technology available to them that we have, or others of our heroes... what would Rock Guitar be now if Jimi Hendrix had stumbled across tapping, instead of Edward Van Halen?

-Danny
Posted by TomS on Oct. 08 2010, 4:55 PM
Danny, I totally agree, I've been saying for years that this was not an overly talented group of musicians, not technically!  Paul by far the best.  But what musicality!  

KISS - when we put on the paint - and, yes, we did! - my buddy Jim was always GS and I was always AF. We gave concerts on top of the dog house in the back yard.  

Back in Black and those other two songs are, for better or worse, etched in my mind like an epitath on a gravestone...
Posted by TomS on Oct. 08 2010, 4:56 PM
Also- that is a really nice comment about the music and stuff.
Posted by bax3 on Oct. 08 2010, 5:03 PM
An interesting discussion for sure - maybe not the origional reason for the post, but interesting.
The whole thing has got me thinking. Maybe we are really talking about two different reasons to record music: one reason might be to reproduce as faithfully as possible the art of the performance; another reason might be to record a performance with the recording itself an integral part of the art. The difference between, let's say, a well-done photograph of a beautiful scene that captures as well as enhances the beauty of what is actually there. The other an oil painting that comes not just from reality but what the artists can add with technique and imagination.
The Beatles have been mentioned a great deal here. As I look at their career, I think that one of the reasons they may have quit live performance was that they became enmeshed in the art of recording. They used every trick in the book and wrote some pages that others had not seen before. I don't believe that the technology of the time would've allowed them to reproduce some of their more innovative productions on a live stage. So not only was their music creative, inventive and of the time, what they did in the recording studio became an integral part of Beatles music.  
When I was a young feller on the road with an acoustic guitar a few folk songs and a few I made up, the only thing I carried into a club (coffeehouse) was my guitar and me. Most of the better places supplied a Shure D35 microphone (if we were lucky) and a home entertainment sound system. More than likely the stage lighting consisted of some colored yard lights that we can operate with a foot switch. My friends that still perform all own at least a van so that they can carry all the equipment that they require to sing a few songs. And that's if they're working as a single!
Bax
Posted by TonyR on Oct. 08 2010, 6:14 PM
Fascinating.
Brian Eno isn't a musician - but, he plays a mean studio.
Posted by TomS on Oct. 08 2010, 6:48 PM
Entirely agree, Bax.  I try hard to do things that can actually be played, even though I'm often just step-programming keys, or whatever.  

Tony - Eno plays keys, and guitar, doesn't he?   Here Come the Warm Jets was one of the most important albums for my in my salad days.  Still a great work.
Posted by TonyR on Oct. 08 2010, 7:55 PM
I agree, too. My life long drummer mate, sadly no longer with us, hated anything to do with drum machines, but would insist that programming should reflect what can actually be done - I took that on board.
I quite like Eno's playing but he doesn't. He doesn't regard it as his thing.
Posted by dannyraymilligan on Oct. 09 2010, 12:21 AM
I haven't a clue who brian eno is, lol, but I will tell you one thing, I think music has become too polished, and that's also bad for us home musicians. We're trying to compete with Evanescence and the like, huge theatrical orchestrations, etc. They spend MILLIONS on making a record, and most of us spend TIME.

Don't get me wrong, I think each one of us is obligated to make the best product we can, but when it comes down to it, what is music? IT'S ENTERTAINMENT. Some of the most beautiful songs in history were done in a few hours inside a ratty old garage studio, and yet every human being on earth knows them by heart. Sit down sometime and really listen to Simon & Garfunkel do "Bridge Over Troubled Water". The Entire Hotel California Album was done in less than two months, and Glenn Frey said most of that was spent partying... now, I find myself obsessing that my music isn't on a production footing that is equal with someone like Creed or Evanescence, when the truth is that I have the technology available to me right now to do BETTER than anyone did back in the 50's, 60's or even most of the 70's. If I render each track down once I'm happy with it, I can easily run 24 tracks on this computer, pristine digital. John, Paul, Ringo and George didn't have anywhere close to that.

But I'm not competing with the Beatles, am I? LOL! When someone listens to a song I record now, they compare it against Alice In Chains, or Creed, or Evanescence, or other modern musicians, and needless to say, my output doesn't measure up to theirs.

I will say this, however: I know that it can measure up, if I work at it hard enough, because I listen to what you others do with yours, and much of your work is on a par with the pros, so that gives me hope :)
Posted by bax3 on Oct. 09 2010, 12:45 AM
Danny,  I am not sure that we need to be competing whit the "big Money productions." I shall continue to maintain that it is the performance that matters.  listen to what Johnny Cash did when he took some of the hard metal songs and did them with an acoustic in his home studio, while he was dying.
You give several examples of the music winning out over the technology - go with that!  If I somehow end up with my radio tuned to a "country" station I am blown away with the interchangeability of the music - the words are different but the approach to the music is uniform in it's over-production. I'm pretty sure it's the same in every genre.
Posted by TonyR on Oct. 09 2010, 1:17 AM
There's a moto that I can't quite remember, it goes something like; Song, Sound, ___, ?
Posted by bax3 on Oct. 09 2010, 1:53 AM
I don't know the quote - could the other word be soul?  A good friend of mine and a great song writer wrote, "Sing for the song, boy . . ."
Posted by Bubbagump on Oct. 13 2010, 1:36 AM
So much plain sideways almost right but not quite information in one thread....

Here we go...

Bit depth (not bit rate as is used in MP3s) is a measure of dynamic range. Not bandwidth. So yes, 16 bit has a dynamic range of 96db and 24 bit 124db. Nick was the most correct in his post. The benefit here is that you A) don't need to record things as hot to get great resolution B) Differences in dynamics are more accurate as you have more "steps" with which to measure volume. These additional "steps" are an advantage at mix time too. The final mix is a complex calculation of everything in the project. 16 bit is like doing your taxes by 10 dollar increments rather then to the actual cents. At the end of all the tax calculations, you'll be much more accurate had you used all those pesky decimals for cents.

Sample rate: sample rate simply determines the maximum frequency you can record. The Nyquist theorum states that at a given sample rate, the highest frequency that can be recorded is one half the sample rate itself. Therefore, in a perfect converter, at 44.1 khz, the maximum frequency you can record is 22.05khz which is well above what a human can hear. The problem is finding a perfect converter. Converters have filtering on the high end and crappy filters can effect frequencies much lower that ARE in the audible spectrum. The advantage to higher sample rates, especially on less than great gear, is that any filtering that happens can be done way up in the inaudible spectrum and any lower frequencies effected by the filter are still well above the audible range. A great 44.1khz converter will still sound better than a crappy 96khz converter.... but recording at 96khz can help make some crappy converters sound less crappy.

Dither: I won't get into the theory, but will get into the rules. Dither is necessary when truncating from any FIXED bit depth source to a lower fixed bit depth format. Therefor dithering when going to MP3 is worthless as it is not a fixed bit depth format. Dither when going to a 16 bit wav/FLAC... otherwise forget it. You can dither coming out of the 32 bit mix buss to 24 bit, but it is hardly worth it as any truncation artifacts are well below the threshold of hearing.
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