Amplitude or Normalize?

What is the difference between using Normalize or Amplitude in the Process menu to adjust the volume level of an audio file in Waveform Editor? It looks like the same thing to me.

You might also try limiter on the Output track (down at bottom of tracks display). Limit to the volume you want. Compressor will affect all sound levels by essentially raising up the quiet levels toward the louder content. Limiting just limits the peaks without making the quieter content sound too different. I usually add compression to individual track (to help fatten them or make quiet parts more audible), then I add limiter to the master output track set to -10dB for broadcast and around -1dB for DVD or Computer output.
Most pro's I know use the compressor levels (and auto levels) controls for adjusting track main volume rather than the normal track volume which should be used for just changing the ups and downs of the various content in the track. I hope that makes sense.

Similar Messages

Normalize grayed out in Audition CC

I am trying to use the normalize effect in Audition creative cloud but it is grayed out under both Effects/Amplitude & Compression/Normalize and Favorites/Normalize. How can I get access to the Normalize effect?

It will be greyed out if you try to use it in Multitrack view. It's a process 'effect' (not really an effect at all) and therefore only runs in Waveform view.

How to do a time normalization on my data?

I am trying to "time normalize" my data.Labview offers a way to normalize(ie.the amplitude).But how do I time normalize?
To explain more:We have different kinds of movement(several movements inside one data file) starting and ending at different times-we would like to time normalize the data, so that each individual movement starts and ends at the same time-so we can compare it for our analysis.
Thanks,
Sujani.

Well let me say that I appreciate the life you spare to help us!
And judging by the following post, the "father of LV" appreciates it enough to thank you publicly!
http://exchange.ni.com/servlet/ProcessRequest?RHIVEID=101&RPAGEID=135&HOID=506500000008000000B7D70000&USEARCHCONTEXT_CATEGORY_0=_49_%24_6_&USEARCHCONTEXT_CATEGORY_S=0&UCATEGORY_0=_49_%24_6_&UCATEGORY_S=0
If you click on his link, it looks like he registered on the Dev-Exchange just to acknowledge you.
Dennis, you have now accomplished the super-human twice over!
Ben
Ben Rayner
I am currently active on.. MainStream Preppers
Rayner's Ridge is under construction

How to normalize audio (streaming .mov output)

We are trying to figure out how to normalize audio. We're using v. 3.0.5. Tried Peak Limiter, and tried the highest setting, and the highest amplitude samples of the waveform did indeed reach 100%, however the noise floor was raised significantly more in proportion. We are looking for a function that will take the guesswork out of the audio settings -- i.e., the typical normalization function you find in most audio editors. Set the percentage of digital full scale that you wish the loudest peak to reach (e.g., 97%), then the entire waveform is increased an equal amount. Even better yet if there is RMS normalization, but I doubt that is in the app. Thank you.
Chris

In FCP you can select your audio, choose Modify -> Apply Normalization Gain and set the threshold. In some cases it will raise the noise floor, but so will any automatic plugin.

Play Waveform Express vi: How to disable "Normalize waveform"?

I am an EE professor. My students are using LabView to develop simple music synthesizers. How can we disable the "Normalize Waveform" function within the "Play Waveform" express vi, so that we can control volume with amplitude?

Since you know that Normalize Waveform.vi is inside the Express VI, you have apparently converted Play Waveform.vi to a standard VI and opened its block diagram. You can either remove Normalize or, probably better, add a boolean control and a case structure around Normalize from the Play Waveform and save with a different name.
While you are at it, change the signal inputs and outputs to arrays of waveforms (as it uses internally) and get rid of the evil Dynamic Data Type. You can use the process to teach your students better programmig pracitces while getting the Vi to perform as you want.
Lynn

How is calculated the amplitude in the FFT function?

Hello,
I have a time signal, for example an 1KHz tone. I use the FFT complex function to calculate its Fourier Transform, and I want to Know, which is the relation beetween the amplitude of this FFT and my time signal.
I have an example of a sine of 1KHz with an amplitude of 1V. When I calculate the FFT the result is one delta at the correspondient frecuency, but with an amplitude of 12149. My signal has 65536 samples.
Could you help me please?

With the complex FFT, I do believe you have to normalize the by half the size of your sample set. This should give you back the original magnitude of your signal. Take a look at this screenshot.
Attachments:
fft_magnitude.jpg ‏65 KB

How do I get the correctly scaled amplitude of a power spectrum?

I am taking data off of a afm probe with a resonant frequency. I want to determine the amplitude of the signal at the resonant frequency. I've tried using AutoPowerSpectrum (FFT(x)FFT*(x)/n^2) and just plain FFT, but neither are scaled correctly when I input a sine wave at a given amplitude and frequency. What do I need to do to scale my spectrum, so that I can tell the actual amplitude of the signal at the resonant frequency?

The Help topic for the SpectrumUnitConversion Method describes ENBW as, “equivalent noise bandwidth of the selected window. You can use this value to compute the power in a given frequency span.” CoherentGain is defined as, ”peak gain of the window relative to the peak gain of the Rectangular window. Use this value to normalize peak signal gains to that of the Rectangular window.”
The Help topic also says, “The ENBW and CoherentGain parameters are output by the ScaledWindow function. You need this input only when you use the spectral density output formats (the last four DisplayUnit selections).” In other words, you only need to use these inputs if you have used the ScaledWindow method and are setting the DisplayUnit to 4, 5, 6, or 7. From what you and Caroline have discussed, I assume you will be setting the DisplayUnit parameter to 1.
Regards,
Eric

Audio Gain and Normalize - different in CS4 to CS3 - problem

Before in CS3, if an audio clip was peaking into the red I'd just right click, select Audio Gain, then click Normalize and then OK - the audio would be normalized and no more peaking into the red. Now in CS4, the right click Audio Gain brings up 4 options. Say if the Peak Amplitude is -0.5dB, which of the 4 options do I select to normalize, and what dB number would I put in to stop peaking into the red? I've tried lots of things but the sound meter still shows peaking! This used to be super easy in CS3—what is method now for CS4?
Any help please!!

One thing to consider is that if your Audio has already been recorded and you're peaking, you can Normalize, or otherwise process your Audio, but you're already clipping signals. It takes a lot to try and restore that "clipped" signal and you can only come close. The -3db/-5db is great for when the recording is being done. If the levels were exceeded at the time of the recording, and that is all that you have, then most processes in post will only be working with the clipped signal.
I usually shoot for -5db in the recording process and then maintain something in the -5db to -3db range when editing. This way, I am working with the full, un-clipped signal.
Also remember that with most "normalization" processes, you just establish the max for that signal, clipped or not, and then bring up the quieter passages to relate to that. Is that what you really want to do? Better to record the Audio at the proper level, with no clipping to begin with.
Good luck,
Hunt

The Levels bar into the red after normalization

Sorry if this is a daft question, I am new to home recording and need to submit my audio via specific guidelines. I'm trying to get into audiobook production. I have a strong theatrical background but i don't know much about mastering audio so this has been challenging...
In the raw file, the levels were primarily in green, occasionally in yellow.
I used "match volume" to get my RMS to a certain level. Then I used a limiter to set to -3 db.
Now the file goes into the red. According to Amplitude statistics. there are 0 "possibly clipped samples" so why then do the levels keep going into Red? I thought that was indicative of clipping?
That's my first question, my second is does anyone know of any basic audio mastering classes for voiceover using Adobe Audition? Less about how to edit audio, more about anatomy of audio files and the meaning of each of these parameters and how they relate to one another.

Thanks for your response. I probably did a terrible job of explaining.
I did originally try the "normalize" option and then the limiter but my RMS stats were too high to submit the audio book. They recommend normalizing peaks at -6db, I found this video which states that you can normalize via RMS levels by match volume: How to Normalize Audio Peak and RMS in Adobe Audition CS 5.5 Tutorial - Normalization - YouTube which I thought would at least bring that RMS number down so I was within their required range. It did so all my amplitude specs meet their specs now but I'm just a little perplexed by the red in the level bar.
This is all very new to me and I need to educated myself. I don't even really understand the difference between normalization and compression and why one is favored over the other. This is all really kind of over my head. I think I can learn it but I need someone to explain it to my in layman's terms.
When I do a raw recording, my Total RMS is always way over the recommended RMS the audiobook production company suggests a pre-mastered file should have. Lowering the gain makes the peak amplitude too low. That should be at -12 max while the RMS should be no greater than -28. RMS is around -40. So in a sorta related question, while recording, how do i lower my RMS without killing the peak amplitude.

Why is it that an amplitude-scaled sound does not play louder than the original sound using Play Waveform?

I don't know if I have posted this on a correct location (kinda new here).
I am trying to demonstrate the effect of multiplying a constant to a waveform. I have a sound file read by Sound File Read Simple.vi. Then I am trying to scale the amplitude of the signal by a constant number. I've tried three methods for this: the Numeric Multiplication, the express VI Scaling and Mapping, and the scaling and conversion VI of Signal Express. The waveform shows that the amplitude of the scaled signal is indeed scaled, but when played thru the play waveform VI, the sound level is still the same, no matter how much I am scaling the input. Does anyone why is this so? Also, I really wanted to show that scaling an audio waveform results in a louder waveform. How do I do this? As much as possible, I've wanted to use simple VIs as the class is an introduction to signal processing and the students (and so am I) are still finding our way into LabVIEW (we used to use MATLAB in the course).
Attachments:
scaling.vi ‏69 KB

JaymeW wrote:
Hi kirkbardini,
According to this article about Low Volume From Sound VI, you can use a scalar multiple to increase the volume of a sound saved to a .wav file, so you should be able to do this before playing it. Does that not work for you?
That information is for Labview 7.x and hence outdated. As mentioned before. Labview do a normalization before sending data to the sound card. As you know sound cards are most often 16 or 24 bits. And in most cases data input to a sound card is in DBL type. After normalization the data is converted so the floating point value 1 represent full scale output on data sent to the the sound card DA. To get any further I suggest kirkbardini take the time needed to send us his/hers current code. Then we may solve things from there
Besides which, my opinion is that Express VIs Carthage must be destroyed deleted
(Sorry no Labview "brag list" so far)

What's the difference between "Normalization" and "Change Gain" ?

So I've got a bunch of tracks that I've mixed down and bounced to 24-bit stereo AIFF files. Next, I'll be mastering those bounced files with the Linear Phase EQ, followed by the Multipressor (if needed) and ending with the AdLimiter.
Needless to say, I'd like for all the tracks to have the same relative volume before starting the mastering process.
-3db is my desired level as it will give me a little room to play with during the mastering.
Anyway, I figured that I could simply normalize all the bounced files to -3db in the sample editor before mastering. But while looking into normalizing in the manual, I noticed the change gain function listed right below it.
However, unless I'm missing something, I don't see a real difference between the two.
Could any of you experts out there tell me if there is a real difference bewteen them? Are there clear advantages to using one over the other?
Thanks,
Ken
17" MacBook Pro "SABRINA": 2.16 GHz, 2GB RAM, 256MB VRAM, 100GB/7200 RPM HD Mac OS X (10.4.7)
Logic Pro 7.2 User

Changing gain = changing gain. There's no functional difference, as I understand it, regardless of how the function is labeled. For example, if you have a waveform that peaks at -3 dB and you normalize it, the entire waveform's amplitude is recalculated so that now the -3 dB point is at a 0dB. This means that every other sample in the waveform had to also have its amplitude multiplied by some factor that results in an increase of 3 dB for every part of that waveform, proportionally. This means that your track will not sound more dynamic. It will have exactly the same dynamics as before, but the overall volume is greater. It's like turning up the volume control. Done.
If you used "change gain" instead of normalize and you changed the gain +3 dB, you'd be performing exactly the same function as described above, yes?
To take this one step further... if you didn't bother to analyze what level the waveform peaked at but used Change Gain to create an arbitrary +3 dB gain change, the waveform may clip if its peak amplitude was already pretty hot (Logic will warn you of this, but doesn't prevent you from creating a clipped waveform). On the other hand, the Normalize function performs an "auto peak search" before it sets out to increase the amplitude of the overall waveform so that the peak resides at 0 dB. Indeed, if you try to normalize an already normalized waveform, Logic will issue an alert message saying something like "already at maximum volume" or something like that, and no processing takes place.
So in a nutshell, Normalize will increase (and only increase) the gain of a waveform by first searching for the peak in that waveform and then increasing the overall gain of the entire waveform such that the peak resides at 0 dB. Change Gain allows you to increase or decrease the gain of a waveform (or a selected part of a waveform) by whatever amount YOU set it to be.
Kenmitch, I'm not convinced that your formulaic approch to mastering is going to give you the best sonic results. For one thing, pre-normalizing your tracks is apt to create headroom problems. I wouldn't do it.

Can not change the number of decimal places in the normalization of result

dear all
    i want to see the proportion of some data, for example, the income of May is 300, and the total income is 1000, i need to display it like 33.33% . so i set the
Calculate single values as normalization of result, and then it display 33.333%, i like to display only two number of decimal places, so i set the number of decimal places as 0.00, but i doesn't work, it still display three decimal numbers.
    maybe you say i can use the percentage function like %CT %GT %RT, but i need to allow external access to my query, so the i can not use those functions.
    can somebody helps me ? your advice is appreciated.

hi,thanks for your advice, but that doesn't suit for my problem.
before i set the normalization of result, i can change the decimal values. After that i cann't.
In your thread, someone proposes use the T-code OY04. but this wouldn't help. As i change to other key figure, such as user quantity, when i set normalization of result, it still display 3 decimal values.
i think the point maybe lie in the normalization of result. please advise... thanks...

SNC - Sistema de Normalização Contabilistica - Portugal

Bom dia,
No decorrer deste ano foi aprovado em concelho de ministros o novo Sistema de Normalização Contabilistica, que entrará em vigor no inicio do ano de 2010 em Portugal.
Alguem tem a noção dos impactos e que metodologias se deverá aplicar para que a transição para esta nova realidade se efectue de forma pacifica?
Terá a SAP alguma estratégia para a alteração para esta nova realidade? (tenho conhecimento da ferramenta SLO - system landscape optimization, para a alteração do plano de contas, mas esta nova abordagem não se resume á alteração do plano de contas, mas sim a um conjunto de principios contabilisticos que deverão ser repeitados pelas empresas).
Qual a vossa opinião sobre este assunto?

Oi Jackson
Sugiro você dar uma olhada nas best practices.
http://service.sap.com/bestpractices
Selecione baseline packages > Portugal > Technical > Content Library
Ali você vai achar o guia de configuração, fluxo do processo e procedimento de negócio para todos módulos e processos de negócio suportados.
* vale também para outros países inclusive o Brasil.
Abraço
Eduardo Chagas

Converting CAP file to JCA using normalizer

Hi, I'm trying to generate 2.2.1(or 2.2.2) JCA file from 2.2.1(or 2.2.2) CAP file using JCDK 3.0 normalizer. I generated class files successfully but when using converter from javacard-kit 2.2.2 i get following error messages
unsupported class file format of version 50.0.
Is there any way to solve the error? I also like to know if there exists any other tool other than normalizer to convert cap file to jca file,

javac -g -source 1.2 -target 1.2 -classpath $JC_HOME/lib/api.jar:$JC_HOME/lib/installer.jar src/xu/test/*.java -d binYou can have a look at the man file for javac in the section source/target and try the different options. I'm using sun-java6-jdk with these compiler parameters and have no problem when it comes to converting the files.
Hope this can help you.
Cheers

Restrict % symbol in normalization BI 7.0

Dear Friends,
I have two quick questions.
1. In Bex, while selecting variable values, how do i do setting to read the values either from Master data table or from the cube (data target).
2. I have normalized a key figure. It shows normalization with symbol %. How to i restrict display of the % symbol.
Your earliest help will be highly appreciated.
Regards
PK

Hi,
you can not insert directly.
I can suggest you this options:
1) Workbook introducind in MS-Excel "%" symbol.
2) Introduce, like attribute of a IO Char in ODS, an IO that contains only %. Then in query use Formula Variable to take value % like attribute from that IO. Not sure but you can try.
Ciao.
create either calculated key figure or formula. Select your key figure, percentage share (%A) out of section "percentage functions" and 100 so formula will show up like this
'key figure' %A 100
If you created formula, hide your key figure from output to display result only once.
Regards,
Ravi.

Amplitude or Normalize?

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