You could use the Normalize function in Sound Forge, it can be set to normalize for average levels, which is what you want. 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. 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 Dymanic Compression." -20db is a good starting point, you will have to play with this value for your project. Once you have determined the appropriate RMS level, use that same value for everything and the loudness of each segment will be matched throughout your entire project.
Or, if that seems a bit too involved, you can use a free, stand-alone program called "Levelator." It may or may not do what you want:
If the original sound file is mp3, you can use free "MP3 Gain", and that will set all the files to the same "playback volume". You can convert to mp3 in soundforge.
Then when you bring them back into Vegas, they will all be the same volume with NO CLIPPING.
Note: MP3Gain does nothing to the waveform, only the playback volume. The program default volume is 89db, but you can adjust it up or down if any files want to clip. It's very simple.
"MP3 Gain" is a less-than-ideal solution. You will have to convert your pristine uncompressed audio files to lossy MP3. Also, the default volume of 89db is a completely arbitrary number. 89db with reference to what?
I use the John Cline RMS Norm method with varied program material with great success. Good enough for CD production if used sensibly. Quick and easy once you've done it a couple of times. (Hint: If the normalization seems to have no effect on the waveform, the RMS level is set too low to affect that clip.)
I have used Levelator for meeting archives, but not for mission critical audio.
The "Normalize" function in a lot of programs, including Vegas and probably Real Player, only adjusts the peak audio level to a specific value. Unfortunately, peak levels are meaningless when it comes to determining how "loud" your final product is. The human ear doesn't determine loudness by the peak level, it determines it by the average (or RMS) level. The "normalize" function in Vegas is useless because it only makes adjustments based on peak levels and that's not they way we hear things.
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.
Audio 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. But like I said, peak levels are virtually meaningless (well, as long as they don't exceed 0db.)
Typically, I set all my projects to an RMS "average" loudness of -20db using Sound Forge. First, I go to "Process" > "Normalize" and scan the levels, noting the RMS level. Let's say it's -27.4db with a few peaks to -2db, I subtract 20 from that and end up with 7.4. I then get into the Waves L2 Maximizer (or Sony WaveHammer) and set the threshold for -7.4db and let it do it's thing. This raises the loudness to -20db and lops off a few peaks in the process. I end up with a program that has an average loudness level of -20db. In case the RMS level has ended up louder than -20db, I merely use the volume process and turn it down. The -20db number isn't etched in stone, I do make exceptions depending on various factors, but -20db works 95% of the time.
Of course, as far as the level relationships between VO, nat sound, effects and music, mixing is an art and I have simply used my ears and used the meters to make sure I haven't gone over 0db at any point in the process. :)
I wasn't suggesting that this is an exclusive club only for professionals, rather that people coming here would expect an answer that's industry best practice by default. If they want something lesser or they don't have the funds for the best there is I'd expect them to say so rather than leave us to probably make the wrong assumption and possibly end up offending one way or the other.
No, that'll just compress the final mix, a quite common but crude technique that is used a lot.
One really good warning to keep in mind about compression. Compression ratios multiply. So if you compress something 5:1 and you send to out to a radio or TV station and they use 2:1 compression your mix will be heard with 10:1 compression which is getting to be a bit savage.
You could apply a compressor to each track, that will possibly work, it depends on the content of each track.
If each track contained say the same person talking then even normalising the tracks would very likely work, assuming there was no errant bumps or thumps.
In my opinion and without any further information really the correct answer is, you cannot do it. You simply cannot make the twitter of a small bird sound as loud as an explosion. You can play the birds song at an ear piercing 110dBA and the explosion at a very quiet 60dBA and listeners will still say the bird sound is quiet (an perhaps stuck in their ear) and the explosion is loud (but far away).
Another problem in these discussions is that words like Loudness, Volume and Level are somewhat arbitary in their meaning.
I still think MP3Gain is a good solution because it maintains the waveform, and a high bitrate VBR encode has minimal loss.
I was reading about mp3 compression, and the first thing it does is take out the inaudible parts of the sound, then of course thins out the resolution.
I know there are pure venues like a movie theatre, large screen projector presentation, or live band accompaniment where mp3 would never be suitable, but most of the time the listening volume is never loud enough to expose the lowered quality.
And if someone has to ask about how to do it, that's a tipoff that they wouldn't be helped by a wheelbarrow full of arcane audio jargon, so that's why I threw it out there.
Myself, I read all the responses and enjoy the various methods people throw out, and I certainly don't cheat myself of variety in the thought process, it's all good.
I still think MP3Gain is a good solution because it maintains the waveform, and a high bitrate VBR encode has minimal loss.
The only time this might be a good solution is if you have a very specific purpose for encoding to this format.
PCM (i.e. WAV) and AC-3 are the two formats most discussed here.
PCM is what you would use if you were making an audio CD.
It's also occasionally used for the audio portion of a DVD.
AC-3 is the format most used though and is a part of the DVD spec while MP3 is not.
If you were to encode to MP3, DVD Architect would re-encode it to AC-3 anyway, resulting in even more compression, so there's no point in even considering an MP3 encode.
If you're going to use Acid, start with a file that has the volume you want and leave the master buss set there, that's your reference.
Then, adjust the preview volume BEFORE adding each track and it will go onto the next track at that volume, similar to Vegas, but Vegas has no preview.
You can use the metronome as an audible reference too.
=====================================================
I don't have Sound Forge but I have ACID XMC which came with V7.
I'll try to see if it has a Normalize function!
In my scenario mp3 would be used as a proxy before importing to Vegas.
Example: .wav to .mp3, adjust volume, then .mp3 to .wav, then import to Vegas and proceed as usual.
Sorry to belabor this, but I think this process is handy, and can be a quick fix for non-pro projects.
===================================================
PCM is what you would use if you were making an audio CD.
It's also occasionally used for the audio portion of a DVD.
AC-3 is the format most used though and is a part of the DVD spec while MP3 is not.
No, in your scenario mp3 is used as an intermediate, not a proxy.
Since using a perceptually lossy codec as an intermediate doesn't enjoy much favor here from either a practical or quality sense, I don't think it's an argument you are likely to win. But no one is saying you can't use it if you want . . .
It's been more than a week. Isn't it time for someone to post an outrage that SCS-encoded MP3s always have silence at the beginning and end and won't loop seamlessly? Or something like that?