![]() Normalizing peak level maximizes your volume up to a point where its highest sample value doesn't exceed 100% (highest sample value possible), or anything less than that, which you can set (I use -0.2dB peak level, or about 98%). In digital audio, all the sample values are defined, and if the volume goes above the maximum value possible, it simply gets "cut off", where anything above it remains at that same maximum level (this is called saturation), leaving a straight line (making it a DC or 0Hz, which, if lasted longer would produce no sound, as sound is defined as a wave, 20Hz to 20kHz for humans), and causing an audible distortion, which sounds pretty much like an amplifier with volume turned all the way up, and the audio sounds "broken". Normalizing peak level simply adjusts the volume of your whole audio file in order to bring its maximum value up to the specified value. What John is refering to as "average level" is actually called loudness, and the "Normalize" function in most audio editors and CD rippers usually sets what is commonly refered to as "Peak level". OK I lied, I want a wide-screen plasma the size of a football field, but unless all y'all want to make donations, that won't happen -) And I'm not buying another TV until the digital/HD/whatever standards become mandatory. I don't know if they still make them, though. I used to have a Magnavox TV with "smart sound" that would keep the sound at the same level, no matter what. It just seems a bit strange that I can see on the waveform that it is a low "volume" soundtrack, but I could spend more time trying to fiddle with the Db+ on it and only have a small visual indicator of how "loud" the sound is. Then I just sub the result file for the main sound track in Vegas. wav file and bring it up in my sound editor that I use for converting LP's, cassettes and Cd's to MP's and use their normalize function. What I do now (a bit slow, but it works) is I render the project as a. Once you have determined the appropriate RMS level, use that same value for everything and the loudness of will be matched throughout your entire project. Use the Normalize function in Sound Forge and set it to normalize using "Average RMS power" and normalize to -20db and select "If clipping occurs apply Dynamic Compression." -20db is a good starting point, you will have to play with this value for your project. Render out the audio from the entire timeline and pull it into Sound Forge, then mark each different clip or section that needs to be level matched. You could use the Normalize function in Sound Forge, it can be set to normalize for average levels, which is what you want. But like I said, peak levels are virtually meaningless (well, as long as they don't exceed 0db.) If you're watching a movie on TV and it has some relatively quiet dialog and then a commercial comes on, the commercial sounds louder because it has been heavily compressed in order to raise its average level (and get your attention.) The fact of the matter is that the movie and the commercial probably had the same peak level, it just that the commercial has a much higher average level.Īudio compression and limiting is an art form and it takes a lot of experience to do it "correctly." There are no hard and fast rules to determine the appropriate average level, you'll just have to play it by ear. The "normalize" function in Vegas is useless because it only makes adjustments based on peak levels and that's not the way we hear things. The human ear doesn't determine loudness by the peak level, it determines it by the average (or RMS) level. Unfortunately, peak levels are meaningless when it comes to determining how "loud" your final product is. This will adjust the peak level to a certain value. A lot of people will use the "Normalize" function in Vegas.
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