Skip to main content

Notice

Please note that most of the software linked on this forum is likely to be safe to use. If you are unsure, feel free to ask in the relevant topics, or send a private message to an administrator or moderator. To help curb the problems of false positives, or in the event that you do find actual malware, you can contribute through the article linked here.
Topic: is WAV normalization lossless? (Read 51727 times) previous topic - next topic
0 Members and 1 Guest are viewing this topic.

is WAV normalization lossless?

Reply #75
-

is WAV normalization lossless?

Reply #76
Quote
I wonder WHY CDs are being mastered SO LOUD these days!? Any CD I rip, any genre... if it's in MP3 all of them get clipping and usually displays 100 dB or over. And CD's from mid 80's and early 90's they're not so crazy loud!

Is there any reasonable explanation why CDs are being mastered so loud nowadays?
[{POST_SNAPBACK}][/a]


This link gets posted fairly often in response to the loudness race:
[a href="http://www.prorec.com/prorec/articles.nsf/articles/8A133F52D0FD71AB86256C2E005DAF1C]http://www.prorec.com/prorec/articles.nsf/...6256C2E005DAF1C[/url]

is WAV normalization lossless?

Reply #77
1) encode the WAV files with a lossless audio codec (FLAC, WavPack, etc.)




have i use EAC to extract the tracks in Flac or wave? thanks!

about normalizations with audition:
'The Normalize effect lets you set a peak level for a file or selection. When you normalize audio to 100%, you achieve the maximum amplitude that digital audio allows--0 dBFS.

The Normalize effect amplifies the entire file or selection equally. For example, if the original audio reaches a loud peak of 80% and a quiet low of 20%, normalizing to 100% amplifies the loud peak to 100% and the quiet low to 40%.

To apply RMS normalization, you must use the Group Waveform Normalize command. If desired, you can apply that command to only one file. '

is RMS normalization lossless? 

is WAV normalization lossless?

Reply #78
is RMS normalization lossless? 


Read the thread and find out    The answer is the same regardless of whether you use peak or RMS method.

is WAV normalization lossless?

Reply #79

is RMS normalization lossless? 


Read the thread and find out    The answer is the same regardless of whether you use peak or RMS method.


thanks!
i read the whole thread yesterday and did some tests:
source Beatles-Martha my dear extracted with EAC.
in audition:
to see how much the track need to amplify to 0dB was used amplify/fade
with peak level in 0dB that show me 4.14dB(62,9%) of amplification with lock left/right!
than i:
normalize the track to 100%-  saved as norm100.wav
using 'calculate now' to amplify the track to 0dB- saved as ampl0dB.wav

using the "saved norm100.wav" and normalize again to -4.14dB(62.9%)- saved as finalnorm.wav
using the "saved ampl0dB.wav" and amplify/fade to -4.14dB(62.9%)- saved as finalampl.wav

comparing source ripped with the "finalnorm.wav" and "finalampl.wav",
nothing is change all tracks seems(waveform view) and sounds equals,
for more than i listen or use the zoom tool to compare!
 
please,give me more explanations and tell me if i forgot something!
thanks.

is WAV normalization lossless?

Reply #80
The entire point of this thread is that the changes are always much too small to notice any change with human senses, regardless of how you may wish to label the results. You can try till the cows die of old age and you will achieve nothing. You can MEASURE that there are small changes if you use the right tools. This is absolutely standard, so well established that you might as well worry whether or not hydrogen will forget how to fuse and produce helium as to fret with it any further.

is WAV normalization lossless?

Reply #81
The entire point of this thread is that the changes are always much too small to notice any change with human senses, regardless of how you may wish to label the results.

It depends on the source material. I could make a 16-bit wav with high dynamic range variations on the audio that would be audibly affected by replaygaining it. The replaygain process would reduce the dynamic range in a way that when reversed would result in an audible increase of the noise floor and loss of the lowest level detail. In real world, in most cases this won't happen, but you can't say it won't ever happen.

Now, in case of 24-bit wavs I don't think I could make a wav where an audible difference could result as a consecuence of replaygaining. And 32 bit floating point (and higher) wavs can be replaygained in a in practice lossless way for all purposes.

is WAV normalization lossless?

Reply #82
ReplayGaining .wav files clearly isn't lossless, and isn't reversible.

It's also not as "lossy" as, say, mp3 encoding.

I'm still happy calling it "lossy", but for those who want to draw a distinction, call it "near lossless".

This matches the situation with some "lossless" audio codecs which can be forced to reduced the bitrate by operating in a "near lossless" mode, which sacrifices the accuracy of the LSBs in order to reduce the bitrate - but not nearly as dramatically as, say, mp3.


KikeG just beat me to an important point: if you have a carefully noise shaped dithered 16-bit file with a huge dynamic range, then a slight level change (implemented without dither, or with optimal but spectrally flat dither) could introduce audible noise into the quiet parts, if you turn the volume up when listening.

The chances of it happening are small, even in an intentionally contrived "worse case" situation, but they are there.


I think AndyH-ha is wrong about "the real world". Most people who own a computer will be familiar with the concept of "lossless", even if they can't name it. It means no loss, which implies no change, which implies I can do something as many times as I want without causing any problem what so ever.

So I can zip and unzip documents, copy them from memory to disk to network to where ever, pass them on to friends and let them to the same multiple times - and if so much as a single character changes in the document, I'll say that there's a fault somewhere - because these operations are lossless.

Similarly, if ReplayGaining .wav files was lossless, I could ReplayGain them as often as I want, to whichever loudness I choose. I could pass the files onto friends and let them do the same. The file could pass through the hands of every person in the world, each one changing the gain to a different level, and when I get the file back I could just change it back, and have my original file.

This is clearly nonsense. If I tried this game of ReplayGain Chinese Wispers with a .wav file, I'd be lucky to get any music back at all! However, if I tried it with an mp3 file, there would have to be a fault somewhere to prevent me from recovering my original file at the end of it.

There are "degrees of lossy", and you might argue that some of them are "near lossless" - but there is only 1 "lossless" - it has a very strict, clear definition.

Cheers,
David.
P.S. thank you to those who gave useful answers to the OP - I'd wavegain before burning a CD and not think anything of the "loss".

is WAV normalization lossless?

Reply #83
Hi folks. I recently did some readings to find out whether quality loss caused by normalization is neglibile or indeed harmful (and my personal conclusion aligns with AndyH-ha's opinion here: http://www.hydrogenaudio.org/forums/index....st&p=449552 ). But before coming to this conclusion, I tried to establish what the consensus on this topic would be, and surprisingly enough ... there was none. Even - and particularly - audio professionals didn't agree for whom this is their daily bread, they make their every-day living with.

So I made a collection of information and sorted it by "negative", "positive" and "neutral" opinions. It was an interesting endevour for me and I thought maybe someone would like to read this ... so here we go.

I am aware that there are more current threads about normalization, but I didn't want to hijack an ongoing discussion with this ... uhm, lenghty posting, so I thought, I'd pick an old thread where it wouldn't matter too much.

Kind regards.

------


point of departure:


Quote from: NorseHorse link=msg=0 date=
I do a lot of live on-site recordings. Often, I don't get a significant sound check (if any) and am not always given a clear heads up about the program material. I naturally loathe digital clipping, and I prefer to play-it-safe while tracking. Hence, I record at 24-bits, and for most shows, my levels peak between -12db and -15db. After I import the audio into the DAW (Sonar, in my case), I turn the channel GAIN up before editing so that my highest peak is -1db or -2db.

Quote from: Alan Cassaro link=msg=0 date=
... I have a lot of OTHER tracks from old tape masters that were digitally transferred over to DAT tape by a respected engineer, who recorded them at such a  low level that most of them were about 50 percent of full gain. He said, "No problem, you can raise them later in the PC, it's all digital, and it doesn't matter, you won't be adding any hiss." He wanted to allow himself plenty of headroom, he said.


OK, so here is what I found in terms of (conflicting) answers:

NEUTRAL:

A while back I responded to a "lossy normalization" thread saying it is rather silly to call normalization lossy, and furthermore, most of the professional audio world does not. While there is a permanent change, it does not result in any artifacts or other loss of quality such as  can be the result of lossy compression.


Quote from: sleepwalker link=msg=0 date=
Myth#1: Never normalize, every additional application of math on your digital audio will add distortion, etc.
This just isn't true, especially with normalize. If you don't believe me. Take an audio file and normalize it to -5dB. Save it as #1. Now normalize it to -4dB. Finally, normalize it again back to -5dB. Now save it as #2. These files will null out completely. That means it's a bit for bit copy. THat means NOTHING HAS BEEN DONE TO YOUR PRECIOUS AUDIO. [...] It's just crazy to hear people talk definitively about desctructive changes it makes when a 2 minute test proves the exact opposite.

[later]

... Looks like Adobe Audition's normalize is better than Wavelab's if you did the exact same test and it didn't null. Don't remember if I mentioned it, but I'm using 44.1 and 32bit float files.

However, using more drastic normalization did not null out, so I do stand corrected. Starting with a -10dB Bonham drum track, I normalized to 0db, then to -1, then to -5, then -3, then to -10 again and compared against the original. I get (inaudible) low level noise at -144dB.

Still, I think -144dB is in inaudible range to anybody I know. I just think it's funny that people fret over normalize, but use track automation which in some cases is even stairstepped pretty significantly. Thumbing your nose at inaudible artifacts while introducing real artifacts left and right just doesn't make sense.


Quote from: Corran link=msg=0 date=
Using Sonar, I took a short clip, doubled it on another track, and phase reversed it. Complete cancellation, of course. I normalized both tracks to -1, no output at all. I normalized them again to -20, and then back up to -1. Guess what? No output. Therefore, the normalization has not changed the audio file at all, or, at least has not changed it in a random way which would be akin to distortion. I tried this with the gain increase/decrease function rather than normalization and it worked the same way. If I increase the gain a few db on one track and push the fader up the same amount, I did not get a complete cancellation, but it was far below hearing threshold (-60 or so). Upping the fader on both channels the same amount cancels out completely.

My personal opinion based on these observations, my testing, and my personal experience is that normalization does not "degrade" anything. I think that in normalizing a file, you are adding say 5.347 db of signal, and if you just add 5.3 db on another file of course it won't "cancel out" because it has .047db difference in signal. Most likely, whatever minute difference in the signal that one might suspect was a change due to normalization is not a "degradation" but rather just a slight difference in signal. Somewhere on this thread it was mentioned that each bit is 6.03 db of resolution I believe, meaning adding "one bit" to a signal or some such will increase the gain by just over 6 db. So there cannot be a perfect addition of say 5 db (in terms of just adding a bit) but the math still works out and the actual signal is not changed. I'm sure there will always be endless debate over this but to my ears and my eyes on the signal indicator normalizing does not do anything destructive and I will keep right on using it for various things.

Quote
Quote
My personal opinion based on these observations, my testing, and my personal experience is that normalization does not "degrade" anything.
We know this from digital theory. All you did was prove it :thumbsup:


is WAV normalization lossless?

Reply #84
(NEUTRAL ... continued)

Quote from: chris319 link=msg=0 date=
Quote
at best this has been at the cost of extra noise, distortion or DC offset
Please explain how this distortion is introduced. Of course the noise floor comes up but that's going to happen any time you add gain. A properly-written program will obey the signs of the samples and there will be no DC shift.
[later]
Given a signal with 72 dB dynamic range at 24 bits, distortion due to truncation/rounding error amounts to 1/((2^24)-4096) or 0.000006%. Think you can hear 0.000006% distortion?


Quote from: chris319 link=msg=0 date=
Quote
at best this has been at the cost of extra noise, distortion or DC offset
Please explain how this distortion is introduced. Of course the noise floor comes up but that's going to happen any time you add gain. A properly-written program will obey the signs of the samples and there will be no DC shift.
[later]
Given a signal with 72 dB dynamic range at 24 bits, distortion due to truncation/rounding error amounts to 1/((2^24)-4096) or 0.000006%. Think you can hear 0.000006% distortion?
[later]
OK, I wrote a program which takes a 10-second sine wave generated algorithmically (using Goldwave) at 24 bits and -60 dBFS and loads the samples into two floating-point arrays; one array is the test array and the other is the control array. I then digitally increased the gain of all the samples in the test array by 60 dB (multiplied by 1000) and then reduced the gain of the samples by 60 dB (divided by 1000). 60 dB is much more gain than you would apply to a typical real-world audio file. The increase and decrease of digital gain was repeated for 1,000 cycles. The program then compared the samples in the test array to the samples in the control array. I then normalized all of the samples in both arrays to values ranging from -1 to 1 (divided by 2^24) and repeated the experiment. In both cases there were zero errors in the bit patterns. This demonstrates that there is no degradation caused by applying and removing 60 dB of digital gain at 24 bits over 1,000 increase/decrease cycles. Here is the PureBasic source code: ...
Quote
neat. Is it possible to load in some real recorded audio? Just to see. And do like 25db instead, some off number that won't be an exact multiple.

Wow! Good call! For a 25 dB gain reduction we multiply the sample values by .0562. Using just any old 24-bit music file, there was indeed a .000002% error, with 0.32% of the samples in error. More interestingly, we get the same error with 1, 10, 100 and 1,000 cycles of gain changing. This tells us that the error is not cumulative, provided the same gain multiplier is used. The .000002% error was arrived at by calculating a checksum of the control samples and a checksum of the modified samples. The error percentage was the difference in the two checksums divided by the checksum of the control samples.
[later]
The checksum error creeps up to a whopping .000003% if, after performing the initial 25 dB reduction/increases, we subsequently modify the samples twice more using arbitrary values. So error is introduced but it is so tiny as to likely not be audible. Further, there is zero error provided the gain multiplier is a multiple of 20 dB (1, 10, 100, 1,000, etc.).
[later]
Repeating this test with 1,000 gain modification cycles using random gain scaling factors, the error comes out to .000003% or 3 parts out of every 10,000,000 (three of ten million).

is WAV normalization lossless?

Reply #85
POSITIVE:

Quote from:  link=msg=0 date=0
Normalization is often used when remastering audio tapes for CD production, in order to maximize the signal level while not changing the signal to noise ratio.
[blockquote][blockquote]My question: If the SNR remains unchanged, this means that the noise level is actually amplified by the same amount as the signal level, right? So what is the benefit?
Answers may vary:
Quote from: NorseHorse link=msg=0 date=
The first engineer I ever worked with swore there was a difference. He normalized EVERYTHING. "Doesn't that just increase the noise too?" I asked him. "I can't explain it, but it actually lowers your noise floor," he responded. And as noted above, Lagerfeldt suggests it may actually deteriorate audio quality, and Sleepwalker suggests it does nothing at all.
[/blockquote][/blockquote]

Quote from: themixingbowl link=msg=0 date=
Once you've completed manipulating the audio, I'd recommend running the file through WaveGain. This is based upon the Replaygain standard and will effectively apply gain (volume) adjustments directly by adjusting the scaling of the samples. Unlike Replaygain, this is not a lossless process an cannot be reversed. It basically means the files are adjusted to -89db, which normally results in a smaller file size too, as most people tend to record at too high a level.


Quote from: Mike Caffrey link=msg=0 date=
If you were to set something to normalize to 0dbFS it would search your selection, find the peak and increase gain evenly through the section that peak reached 0dBFS. If your trach had clipped, there'd be no change. If your highest peak was -1.5 dB, it would give you the same results as increasing the gain by 1.5dB. I could see normalizing being useful if you had a track that was recorded too low and you wanted to increase the gain as much as possible without peaking and you didn't want to have to guess several times.


Quote from: David Spearritt link=msg=0 date=
as a final mastering step to properly encode a cold recording for the delivery medium, it is technically responsible and necessary. If done correctly and competently with plenty of math precision it results in no audible degradation. It forms part of a mastering person's required skills. In my field of live classical, sometimes we record cold because we do not get a sound check, or we simply underestimate the SPL in the first place, or the program has very loud contrasting with very soft and we are not able to change gain mid concert, there are lots of reasons for cold source material. Now to not change the level because of some unjustified fear of gain changing in the digital domain makes no sense. High precision quality digital mixing and gain changing sounds superb in my humble experience.


Quote from: NorseHorse link=msg=0 date=
I disagree with those of you who have ruled out normalization as part of the mastering stage. Are you the folks creating the music that I can't hear in my car?! Not every system is capable of infinitely adjustable volume. Listening systems have noise floors too, and every dB of nothingness you leave at the top increases the relative height of the noise floor (because the listener has to turn up their volume). I haven't normalized in years, but I do manually adjust the gain of whole pieces or concerts to make sure I'm not wasting signal. If I can help listeners overcome the shortcomings of various systems by making sure my music peaks around -.5dBFS, I'm happy to. After all, it doesn't cost me dynamic range - it just means they have less hiss in the way.

is WAV normalization lossless?

Reply #86
NEGATIVE:

You generally *don't* want to normalize wav files.  The procedure is not lossless.

Quote from: DeeDrive link=msg=0 date=
Quote
So when do I know it's a good idea to normalize or not? Im a little confused when it comes to this? Little help?
Never, that's when. Normalizing is useless. It does nothing but raise the peak value to a determined level, which is completely based on peak level, and therefore useless. Average level is a much better indicator of volume, and the only way to set the volume of something is to use yours ears.

Quote from: dualflip link=msg=0 date=
Never normalize, period

Quote from: Lagerfeldt link=msg=0 date=
many types of destructive processing (such as normalizing) is performed at 16 bits using truncation, despite the DAW being able to handle at least 32 bits float - another reason to stay away from this type of handling. [...] [in] cases where the peak reference is more relevant than the minor possible deterioration of the sound quality, normalizing can be performed.

Quote from: shotmillions link=msg=0 date=
I dont personally normalize. Normalizing just push's the wav forms to the highest therotical volume 0db i only normalize if i want to have a closer look at the wav form for chopping then i put it back how it was.

Quote from: mixerguy link=msg=0 date=
NEVER normalize.

Quote from: Robert Randolph link=msg=0 date=
Quote from: sleepwalker link=msg=0 date=
Myth#3: Record the hottest possible signal without clipping.
When we were using 8bit converters, that wasn't bad advise. People seem to think that you need to hit 0 dB to get the *full* 16 bits. Not so. The loud stuff is covered in the 1st few bits. By the time you get to the 16th bit we're talking way way way low level stuff here. If you're tracking in 24 or 20 bit, feel free to leave headroom, you'll end up with 16bits of usable data anyway, there's no need to bludgen your AD and make mixing harder for absolutely no improvement in sound.

#3 as described is just plain wrong. [...] if normalization, which is just an amplitude adjust is pefect as in said myth 1. Why does it matter if you record full scale? If you get a full signal that does not clip, and bring it down 24db, by the logic of "dispelling myth #1" the signal will be without degredation. But that's just not true now is it?

is WAV normalization lossless?

Reply #87
(NEGATIVE ... continued)

Quote from: mogWai link=msg=0 date=
Well Myth#1 ... I did exactly what you suggested using 'Audio File Compare' in WaveLab 5. I used a 24bit - 48kHz mix. I also trust WaveLab's internal data integrity. Result - 7482365 differences (albleit inaudible)
But wait - comparing exactly the same file -
Result - can't be done 
so, original file opened in Wavelab, saved under a new name - compare with original.
Result - EQUAL
Control - Original file normalised to -3db and saved. Original file opened again, normalised to -3db and saved to another name.
Result - EQUAL

Conclusions -
1. Making multiple copies of the same file doesnt effect the integrity of the file.
2. Normalising sucks.

Quote from: Bob Olhsson link=msg=0 date=
There's actually a fair bit of literature that suggests transparent gain changes aren't nearly as easy to accomplish as many people assume. Floating point files are the only ones that can POTENTIALLY be "normalized" without damage. Considering that it buys you nothing at best, it's not a good idea unless you KNOW the gain change has been properly implemented. I learned the hard way years ago to never assume that software developers know what they are doing.
[...]
Normalizing means you are multiplying the original numbers and then truncating, rounding or, hopefully, dithering the result down to a number that you can write in a file. The only thing you have gained is level but at best this has been at the cost of extra noise, distortion or DC offset. Should this not be the final level, it's going to require even more noise, distortion or DC offset to produce the final level.In an ideal world, we would record audio and then in one grand computation calculate our final digital audio signal. Adding unnecessary math, noise and distortion to a recording is utterly foolish yet that's precisely what normalizing does.
[...]
The point is that normalizing, exactly like every gain change and other form of digital signal processing, degrades the signal. There are many legitimate reasons to change the gain but just mindlessly applying an extra gain change when you are going to need to turn right around and change it again is pretty silly.


Quote
Prompted by a similar debate elsewhere I put "Myth 1" to the test recently myself.
I did extensive testing in Logic on the normalising question, and I could clearly hear a difference between a normalised file and the original - I could pick one from the other blind every time, easily. I used an analog mixer to match the levels and I made sure to flip channels and D-A outputs to eliminate the possibility of other sonic differences affecting my judgement. Can someone please explain in plain English (as my math is none too hot) that if normalising supposedly 'does nothing to the sound' why I could in fact hear a difference?

Quote from:  link=msg=0 date=0
Well it does waste time and drive space. If you normalize a signal down in level it's possible you might lose some useful information; normalizing up in level probably won't have much effect...unless the DAW does use 16bit precision on a 24 bit file (which would cause me to remove the programmer responsible from the production team)


Quote
The only type of normalization I fine useful is Average level or RMS normalization and only in the broadcast or sample creation field. Peak level normalization is an ancient destructive process that was more of a time saving measure carried over from pre-plugin DAW's.

Quote from: aracu link=msg=0 date=
What normalizing software normally accomplishes normally sounds distorted, no matter what the normal mathematical or normal scientific explanation of it normally is.


... uff, done

is WAV normalization lossless?

Reply #88
There's a profound amount of FAIL in the NEGATIVE comments there, particularly regarding TOS8.

Look. Fundamentally, peak normalization is scalar multiplication. PERIOD. It does not truncate, it does not add distortion. Those are purely effects impinging on it for some signal processing systems. Nothing's preventing you from defining an audio system that operates in 128-bit quad precision with quantization noise levels literally -2000db down from full scale. Nothing's preventing you from defining your input signals entirely algebraically and doing all your processing symbolically, eliminating quantization noise entirely (until you evaluate all the sample values of course).

In terms of the actual operation on the signal, peak normalization is exactly the same as RMS normalization. They're both multiplication by a constant.

You need to look at the quantiztion noise introduced by the normalization operation to gauge the quality loss. You need to look at the internal numeric format for computations vs the output numeric format and the noise/distortion imposed by each of them. You need to look at whether or not dithering/noise shaping is employed at the output stage. You need to look at how overflow/underflow are handled in the internal numeric format.  And if you're talking about audibility, if you don't employ a good DBT, you must look at ALL OF THIS in BOTH the time and frequency domains to analyze audible effects.

Very little of that, beyond what little commonality exists in choice of numeric formats, can be generalized beyond one digital audio application or another, or depending on configuration differences, even between different versions of the same app.

That isn't to say that evaluating normalization issues is somehow unknowable or that we should never care about it - dealing with quantization-related issues is a well-established (if complicated!) aspect of digital signal processing. But generally, once you are computing with a mantissa larger than 16 bits, employ dithering, and have a listening environment far noisier than the dither noise (read: all of them), you have nothing to worry about.

There's more I could say about the role of audio engineers in this debate but I would be saying things which are not nice at all so I'll stop.

is WAV normalization lossless?

Reply #89
You may have mixed several meanings/uses of the word normalization.


Normalizing is applying a gain, yes. But the gain is not the problem itself. The problem is the need of using normalization.

When the average joe talks about normalization, it talks about peak normalization. Pushing everything up doesn't necessarily give any benefit (the SNR is mainained so the noise is increased too). Also, nowadays there's little (if anything) to normalize, because everything is compressed ("clipressed" some would say).

Also, there are several fields of application:
It is not the same to normalize when recording than to normalize on playback.  Using too low or too high gain value when recording can either increase the background noise (decrease SNR), or clip inevitably (in case of recording in digital. With analog you had a safe margin).
Instead, normalizing on playback is basically the same than using a volume. If the user want to attach several volume knobs one after another it's up to him.
And in this case it's where the talk about the exactness of gaining in software or hardware plays a role.

is WAV normalization lossless?

Reply #90
Using too low or too high gain value when recording can either increase the background noise (decrease SNR) ...

what is the optimal gain value then?

is WAV normalization lossless?

Reply #91
“gain staging” is the widely used term. Most recording chains contain a number of devices between microphone and recorder. The trick is to adjust each stage for the best gain-noise compromise, aiming for the final value desired at the end of the chain. Maybe there are some difference practices today, but for many decades, for most professionals, that overall gain goal was that the expected highest peaks would be at -20dBfs on the VU meter. This generally gave plenty of headroom for unexpectedly high peaks.

Most analogue devices have relatively low noise at low gain that gets noticeable worse at some point of increasing gain. For example, my modest microphone preamp is nice and quite as long as the gain knob isn’t rotated too high. Noise starts going up very noticeably at about 75% of the way to maximum rotation. Therefore I would set it at o more than 75% – if I can get enough signal, and not too much, to the soundcard at that setting.

In many recording chains there is at least a mixer, containing a line level preamp, and sometimes several other devices, before the ADC (or some analogue recording device). Each has its own characteristics, and thus its own settings. A particular combinations of devices, for a particular style of recording, or particular type of source, may require different settings on some of the devices than when they are being used for a different recording job.

And to think anything will be harmed in any slightest way by modifying the gain after recording, as necessary to fit the desired mix, is just plain silly.

is WAV normalization lossless?

Reply #92
chrizoo,

I appreciate that you're trying to help, but IMO I think block quoting gearslutz threads on HA isn't helpful. I don't think many of us are interested in other people's subjective unsubstantiated blind-test-free opinions. The "consensus" of a set of such opinions is kind of irrelevant, unless you believe that the wisdom of crowds is a better way to figure something out than to look for verifiable facts. (Please don't delete it, but IMO think twice before doing something like it again!)

It always amazes me how little people on other forums understand audio, but post about it anyway! I guess it's because, if I don't understand something, I either keep quiet, post to ask a question, or preface my opinion with "I don't understand this, but...". Yet the internet is full of people who are happy to pass on their muddled misunderstanding of something as "fact".

Cheers,
David.

is WAV normalization lossless?

Reply #93
Yet the internet is full of people who are happy to pass on their muddled misunderstanding of something as "fact".

Hmm, your "misunderstanding" might be a "fact" to them at that point (or for eternity if they can't grasp the issue). You do understand what you're saying is leading to? One should consider everything that one knows as a fact as false? So, we only know that we don't know?  Not understanding something needs an understanding that you don't understand. I bet that you have some "facts" in your head that are in fact "misunderstandings" (and sometimes say them out loud).  Maybe this is not what you meant?

is WAV normalization lossless?

Reply #94
Uh-oh, internet philosophy…

is WAV normalization lossless?

Reply #95
I appreciate that you're trying to help, but IMO I think block quoting gearslutz threads on HA isn't helpful. I don't think many of us are interested in other people's subjective unsubstantiated blind-test-free opinions. The "consensus" of a set of such opinions is kind of irrelevant, unless you believe that the wisdom of crowds is a better way to figure something out than to look for verifiable facts. (Please don't delete it, but IMO think twice before doing something like it again!)

It always amazes me how little people on other forums understand audio, but post about it anyway! I guess it's because, if I don't understand something, I either keep quiet, post to ask a question, or preface my opinion with "I don't understand this, but...". Yet the internet is full of people who are happy to pass on their muddled misunderstanding of something as "fact".

Honestly I'm a little more annoyed by the thread bumpage, but this brings up a topic which has been very much on my mind recently, which really has nothing to do with the OP. (Mods, feel free to fork threads.)

My perception - both about this specific normalization issue, and about half the crap we rant on about on HA - is that not only do most other forum posters seem to not understand the fundamental audio concepts, but that many and perhaps most practicing audio engineers do not understand the fundamental concepts, going all the way up to what is printed in trade mags and repeated by top-tier professionals. The internet is full of misinformed blowhards when they post about crap outside their area of expertise. For lay professionals in a field to also be blowhards is... really rare.

For instance, you'd be very hard-pressed to find an audio engineer who uses spectrum displays in their work while understanding what the effects of windowing are (much less understanding how to compute it). If you find somebody who dislikes MP3s, it is almost never nuanced in terms of encoder version, CBR vs VBR, the superiority of AAC, or even the intrinsic advantages of lossless coding indirectly related to sound quality. No - what I'd suspect you'd hear is something along the lines of "it's obviously poor quality" or "just listen to MySpace".

I've had an audio engineering student tell me with a straight face his understanding that WAV is higher quality than FLAC (which I quickly corrected albeit perhaps not in the most polite terms).

As I've mentioned before, a shockingly large number of audio professionals (both engineers and producers AFAIK) object to the loudness normalization implementations present in Spotify, last.fm etal - all of which are exactly or materially the same as ReplayGain - because they think it compromises sound quality.


This sort of thing is what led me to post this in exasperation. It's easy to just ride the schadenfreude on this, but to be completely fair, this represents a tremendous gap in education efforts, which the AES ought to be filling.

is WAV normalization lossless?

Reply #96
@Akkurat: good point!

@2Bdecided: I largely agree with you. That's why I picked an old, long forgotten thread, where - as I said above - it wouldn't matter too much. I don't plan on re-doing this, but I found it honestly very interesting to read, and I don't think it was in vain as a couple of people replied. Also note that it wasn't one but various threads on the forum you mentioned which I summarized, plus other sources, too. Many of the quoted users state they are audio professionals (Recording and Mastering Engineers), and as can be seen from Axon's posting above, this sparks interest.

“gain staging” is the widely used term. Most recording chains contain a number of devices between microphone and recorder. The trick is to adjust each stage for the best gain-noise compromise, aiming for the final value desired at the end of the chain. Maybe there are some difference practices today, but for many decades, for most professionals, that overall gain goal was that the expected highest peaks would be at -20dBfs on the VU meter. This generally gave plenty of headroom for unexpectedly high peaks.

Most analogue devices have relatively low noise at low gain that gets noticeable worse at some point of increasing gain. For example, my modest microphone preamp is nice and quite as long as the gain knob isn’t rotated too high. Noise starts going up very noticeably at about 75% of the way to maximum rotation.


AndyH-ha, this is very interesting. Because I always thought that I should do what they called myth#3 above, namely "record the hottest possible signal without clipping". But your posting suggests a completely different approach.
If I follow your logic correctly, what I should rather do is trying different recordings, each at a different gain level, then - on the final recordings - adjust the volume so that they are equally loud (in terms of RMS, right?) in order for them to be comparable and see which recording has the lowest amount of noise. Repeat this process until you narrow it down to the best gain level for your specific hardware, e.g. below 75% for the microphone preamp you mentioned.

What I didn't understand though, is why you first suggest one should aim for "the best gain-noise compromise" but in the very next sentence you said that one should aim for -20dBfs ...
While I understand that this gives you good security with regard to clipping, I don't understand why the "best gain-noise compromise" is necessarily at this value.

is WAV normalization lossless?

Reply #97
Yes, experimenting before recording for real is the way to learn the characteristics of your equipment.

The target recording level (i.e. -20dBfs) is independent of “the best gain-noise compromise.” If the gain that gives good noise figures is not adequate to provide the target input level, then the optimum course is to put another gain stage between that and the ADC (e.g. a mixer in between the microphone preamp and the ADC inputs.) In general, this can give you more gain without as much increase in noise as turning up the microphone preamp’s gain beyond its good working range.

is WAV normalization lossless?

Reply #98
My perception - both about this specific normalization issue, and about half the crap we rant on about on HA - is that not only do most other forum posters seem to not understand the fundamental audio concepts, but that many and perhaps most practicing audio engineers do not understand the fundamental concepts, going all the way up to what is printed in trade mags and repeated by top-tier professionals. The internet is full of misinformed blowhards when they post about crap outside their area of expertise. For lay professionals in a field to also be blowhards is... really rare.
You're spot on, and I agree 110%, except...

Professional audio has created for itself a parallel universe. The rules of science, evidence, objectivity etc don't apply there. There are different realities which people accept.

...and in those realities, they're very knowledgeable. The "best" people there don't lack anything.

If you were being kind, you'd say that it's the intersection of art and science, and why should those on the art side be excepted to understand the science.

Why indeed. Problem is, they believe that they do, and happily spout about it. We, here in the scientific real world, know its unsubstantiated bollocks, but in their world, it's accepted "truth".

Quote
This sort of thing is what led me to post this in exasperation. It's easy to just ride the schadenfreude on this, but to be completely fair, this represents a tremendous gap in education efforts, which the AES ought to be filling.
I think you're being generous to the AES. While there's lots of hard science in there, some people who are proponents of what you criticise are paid up members - and vocal ones at that. The AES as a body doesn't subscribes to TOS 8 / DBTs you know!

Cheers,
David.

is WAV normalization lossless?

Reply #99
My perception - both about this specific normalization issue, and about half the crap we rant on about on HA - is that not only do most other forum posters seem to not understand the fundamental audio concepts, but that many and perhaps most practicing audio engineers do not understand the fundamental concepts, going all the way up to what is printed in trade mags and repeated by top-tier professionals. The internet is full of misinformed blowhards when they post about crap outside their area of expertise. For lay professionals in a field to also be blowhards is... really rare.

I don't think it is particularly rare. You certainly see a lot of this in the practice of medicine.