For Those Naysayers...

Julf

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so it's all in the mastering then? The same guy who did a great 16/44.1 can't top himself with more bits and samples at his disposal from recording all the way down to mastering? It's already obvious how much less peak limiting he'll need for the latter, that is something you can't fix once it's there.

Well, with most modern recordings we are unfortunately nowhere near pushing the limits of even 44/16.
 

mep

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Certainly!

You and Mr.Cantankerous consistently try and bait others using miss-quotes; fictional content, or condescending flaims. If silly assumptions and childish ridicule are considered valid debating "skills" around here ... well ... I can only conclude that whatever I, or anyone else with a differing opinion may claim ... won't really matter.

Therefore your "because I want to make sure I understand you before I respond." was hilariously ironic.

tb1

I never misquote people intentionally nor do I "bait" anyone because I don't like to fish. I also don't make silly assumptions. You asked me a question (questions actually) that I wanted to make sure I understood before I replied so that I could give you a thoughtful answer. You chose to make a joke about it and laugh and never clarified the intent of your questions. I see now that it was merely an attempt to impugn my integrity and try and have a little fun at my expense by accusing me of things that I'm not guilty of. I hope you enjoyed yourself.
 

MylesBAstor

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JackD201

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Well, with most modern recordings we are unfortunately nowhere near pushing the limits of even 44/16.

Like I said. Work flow. There is a reason for this, the $$$$$$$$$ of studio time. You'll have to keep your mic gain very low to avoid hitting 0. That brings problems on both the high (peak) and low side (SNR). You will either end up using a peak limiter or rely heavily on normalization in post. If you want to really push it that takes hours of takes (that translates to $$$$$$) to anticipate what the artists will do and calibrate gain accordingly. Musicians aren't drum machines. All it takes to overload is an artist to get a bit excited. Want an artist to lose his game and probably fire you? Make him play over and over again.

Use 24bit while recording and those limits are easily reached while cutting time by removing potentially destructive safeguards for 16bit that will imbedded for all time. You'll probably get a better performance as well from your happy musicians.

Now given the choice, would you rather get the 24bit recording or a 16bit fold down for the same price give or take a dollar or a few cents? Price wise this would be the case of a CD delivered to your door vs a 24/44.1 or 24/48 downloaded.
 

Julf

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Use 24bit while recording and those limits are easily reached while cutting time by removing potentially destructive safeguards for 16bit that will imbedded for all time. You'll probably get a better performance as well from your happy musicians.

I agree that it is better to record in 24 bit to have the headroom, but once the recording is done and normalized, 16 bits is unfortunately more than enough for most modern, "produced" music with very limited dynamic range - that was my point.

Now given the choice, would you rather get the 24bit recording or a 16bit fold down for the same price give or take a dollar or a few cents? Price wise this would be the case of a CD delivered to your door vs a 24/44.1 or 24/48 downloaded.

I would like to re-frame the question - given the choice, would I prefer a 16-bit file of a well-recorded, well-produced piece of music, or a 24-bit file of the same piece of music, compressed and limited to death? You can probably guess what my answer would be...
 

JackD201

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On the first point, why would you want a normalized track at all when you can have one with none (gain just ridden) within the context of a quiet listening environment? I'm not saying that that is what is all that is out there now, just that that is the promise and there are examples of such, mostly classical.

On the second, yes that is obvious. Unfortunately, that was not the question.
 
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Julf

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On the first point, why would you want a normalized track at all when you can have one with none within the context of a quiet listening environment? I'm not saying that that is what is all that is out there now, just that that is the promise and there are examples of such, mostly classical.

But why wouldn't you normalize it to use the full dynamic range? I am not talking about compression, I am talking about adjusting the gain/amplitude so that you use the bits you have in the most optimal way.

On the second, yes that is obvious. Unfortunately, that was not the question.

My answer to your original question is yes, I would prefer the 24-bit version if there was no significant cost disadvantage. That would give me the choice of keeping it in 24 bits or reduce it to 16 bits, depending on if I feel there is any benefit to it.
 

JackD201

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Let's start where we agree. The headroom of 16bit is more than enough. We're good there. Again I said work flow. With 16bit you have to record, for safety, expediency and cost at levels typically between minus 20 and 16 dbfs. Smack down the middle we are only using about 13bits. So yes you would normalize to 0 dbfs thereby using the full dynamic range but this has its own set of potential problems nor would it in either 16bit or 24bit still obviate the requirement of some judicious compression on a case to case basis. These problems like the potential of adding artifacts from too much processing can be avoided by recording to 24bit at levels where you are equivalent to the usable dynamic range of 16bit. No need for too much added processing. I think we are still in agreement at this point. What would be crazy is to try and utilize the full dynamic range of 24bit for obvious reasons, no system can probably handle it but more importantly we humans wouldn't and couldn't. Here I am SURE we are in full agreement.

My recording background isn't in music, it's in cinema and TV, but the tools are the same. The final outcomes are also the same in that the material is prepared for an intended use. So I am not on the anti compression or anti normalization band wagon as they are useful tools if the goal is delivery of the material to the most common denominators. In cinema attention is paid to the quiet rather than the peaks compared to music as we add tons of low level elements that viewers pay no attention to like foley elements. Take out the rustling of the clothes, the footsteps, the wind or even the room tone and the experience becomes weird and unnatural. To a certain degree this is true of music too.

Wait, where don't we agree again? :D

I think I am now seriously off topic since this is a DSD thread and I am a DSDunce. I only ever recorded with PCM and the highest I have used is 24/88.1. The greater majority of the time was at 24/44.1.
 

Julf

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Wait, where don't we agree again? :D

Not sure :), I think we are saying the same thing. The headroom of 24 bits is useful when recording (and processing), but the finished end result will "fit" in 16 bits.
 

JackD201

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Word for word I agree. Personally I still would want as little post processing as possible as this can affect the quality of the original product. If I can get the original and not have it subject to the whims of the guy doing the downconversion, I'll take the original. Like you said, you can do this yourself. Now I'll only have myself to blame!

One thing for sure, for things I did record myself, the 24bit always sounded better. The sampling rate? The differences were not night and day, at least on the monitoring platforms we were using. As for what I buy, it's hit and miss with any format. Why that is, I can't say. I wasn't there.
 

Bruce B

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But why wouldn't you normalize it to use the full dynamic range? I am not talking about compression, I am talking about adjusting the gain/amplitude so that you use the bits you have in the most optimal way.

normalize the music you also normalize the noise. just use correct gain-staging to begin with and use the volume control! ;)
 

Julf

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normalize the music you also normalize the noise.

Normalizing doesn't change the signal-to-noise ratio.

If you amplify the music, you also amplify the noise, but the listener can turn down the volume by the same amount, lowering the noise from his/her listening gear.

Is there any rational reason not to take advantage of all the bits available in the final digital file? Increasing gain (as long as you don't exceed 0 dB) is pretty much the most benign digital operation you can perform - totally linear, without any temporal or spectral implications.
 

JackD201

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http://www.youtube.com/watch?v=4SxClRg-JOU

3:10 - Noisefloor

6:30 - Errors (some other potential errors are not mentioned BTW)

Somwhere in between are two types of Normalization. Here we have only discussed Peak because I never applied RMS. I could never find a good use for it.

I'm too lazy to type :)
 

Julf

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3:10 - Noisefloor

"If you have an audio file that is *very low* in gain value [...] so the noise floor will come up *as well*".

6:30 - Errors (some other potential errors are not mentioned BTW)

Talks about *excessive* amplitude manipulation, and dangers of not having enough precision *while processing*.

Somwhere in between are two types of Normalization. Here we have only discussed Peak because I never applied RMS. I could never find a good use for it.

RMS normalisation has the risk of producing clipping. Peak normalization that ensures peaks never exceed 0 dB with a margin for inter-sample peaks is safe.
 

Bruce B

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Normalizing doesn't change the signal-to-noise ratio.

If you amplify the music, you also amplify the noise, but the listener can turn duown the volume by the same amount, lowering the noise from his/her listening gear.

Is there any rational reason not to take advantage of all the bits available in the final digital file? Increasing gain (as long as you don't exceed 0 dB) is pretty much the most benign digital operation you can perform - totally linear, without any temporal or spectral implications.

Well record it correctly to begin with so you don't need to normalize
 

Julf

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microstrip

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Normalizing doesn't change the signal-to-noise ratio.

If you amplify the music, you also amplify the noise, but the listener can turn down the volume by the same amount, lowering the noise from his/her listening gear.

Is there any rational reason not to take advantage of all the bits available in the final digital file? Increasing gain (as long as you don't exceed 0 dB) is pretty much the most benign digital operation you can perform - totally linear, without any temporal or spectral implications.

Although we could expect that an high digital level would not to affect sound quality, IMHO once you are in real world it can affect. The distortion spectra of most analog circuits depend strongly on level and often with poorly designed equipment you have saturation effects at high digital levels, some times due to the poor performance of the electronic switches used in the attenuators at high level.
 

Julf

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Although we could expect that an high digital level would not to affect sound quality, IMHO once you are in real world it can affect. The distortion spectra of most analog circuits depend strongly on level and often with poorly designed equipment you have saturation effects at high digital levels, some times due to the poor performance of the electronic switches used in the attenuators at high level.

I agree that analog circuits are non-linear and thus level-sensitive, but I would like to see some references to the digital saturation effects - and what level is the "perfect" level? I would assume that that would vary from one device to another, so that there is no "right" answer, and thus no "right" level to aim for.
 

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