Normalizing on MR8

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Laura C

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I am recording a few demos on my MR8 with just acoustic guitar and vocals. I've noticed that -- though I *think* I'm being consistent in vocal and strumming intensity, the levels wax and wane (I suppose I may move toward the mic a bit unknowingly, or get carried away on my strum during intense moments). Is there a way to normalize the levels so the volume is consistent throughout?
 
Okay, so I'm a newbie. Compressing it is.

Now, can you tell me how to standardize the volume levels while compromising quality as little as possible?
 
You might consider normalizing first, to get them closer to begin with, and then compression for portions where levels are lower due to performance and not the song.

If you can look at each track in an WAV edit view, it may also become apparent where things tracked up and down. That might make it easier to detrermine what compression or adjustments to apply and where. Sometimes that helps, and sometimes it does not, but it's usually worth trying. Compression does change the actual dynamics of the recorded material and you want to change that as little as possible.

Ed
 
Laura C said:
Now, can you tell me how to standardize the volume levels while compromising quality as little as possible?
Click to expand...
Normalization is NOT the way to go, especially if you're trying to preserve sonic integrity -- you needlessly lose a digital generation because compression/limiting will do in one step what normalization followed by compression does in two steps.

Contrary to what Ed Dixon's seems to believe, normalization is not a commonly used process (except by novices who don't yet know enough to know any better.)
 
Laura C said:
Okay, so I'm a newbie. Compressing it is.

Now, can you tell me how to standardize the volume levels while compromising quality as little as possible?
Click to expand...

Just a thought... Are you aware that data compression is not the same as "signal" compression? Some machines compress data to fit more stuff onto the hard drive. Not good for quality.

Your MR8 uses full 16 bit 44.1khz uncompressed data. Nice.

However, if you were to buy an external compressor that would go between your mic (preamp) and the MR8, you would find your vocals would even out a bit, without any discernable loss of quality. The RNC (Really Nice Compressor) by FMR is a fine unit that almost doesn't seem like it's on. It treats the sound going into the MR8, similar to external EQ, but it doesn't have to mean a sacrifice in quality.
 
Just to throw in my two cents worth here...

You need to compress your vocals. I'd tend to recommend getting an external tube preamp (the ART units are pretty cheap) and a compressor - I'd hate to recommend one because I'm using a real cheap setup and don't have serious experience with voice channel and midrange compressor solutions (I use a Bellari RoundSound tube preamp and a DOD gated compressor/limiter). Crank up the gain on the tube pre and set the compressor for a relatively low compression setting to get a decent signal level going in for recording.

Once you have the gear you'll have to spend a few days playing with it to get the best match between the setting and your singing/playing style. About the only thing I'd recommend is making sure you don't over-compress, but it's pretty obvious when you do (no dynamics left to the sound).

Good luck!
 
Thanks for each and every reply. I really appreciate learning from all of you. Home recording is so freaking complicated! All I want to do with this little setup is write songs and make decent demos!

I will look into the equipment mentioned above... I had no earthly idea that I needed something more. Is vocal compression a standard practice in most recording studios? If not, how then are the levels standardized?

Lastly is the equipment you recommended expensive?
 
The tools you have already are probably sufficient for what you need. With an MR8, PowerTracks on PC, and mics, you can make very high quality recorded products.

I suggest working with that first to learn the overall process and see what works best for you. Your own ears are probably your best resource. You will probably find that you need no other gear.

The MR8 does a good job of recording the raw audio source in a 44KHz 16-Bit format. Processes like compression actually change the character of the recorded file by altering the actual dynamic range of the audio. The same is true for EQ when you expand/contract certain frequency ranges that we'nt present in the original material. Sometimes that is a good thing and sometimes it is not. Again your ears are your best resource as to what works best for you and the material you are trying to record.

Processes like volume and normalization do not change the character of the audio but raise/lower the level of the raw audio material (Blue Bear actually has it backwards). The mix down process does exactly this by combining what you have at different levels for the final result. Again your ears are your best source of determining what final result sounds best.

The MR8 is a great piece of gear, and when used with PC mixing software can produce CD quality products with ease.

Ed
 
Laura C said:
Thanks for each and every reply. I really appreciate learning from all of you. Home recording is so freaking complicated! All I want to do with this little setup is write songs and make decent demos!

I will look into the equipment mentioned above... I had no earthly idea that I needed something more. Is vocal compression a standard practice in most recording studios? If not, how then are the levels standardized?

Lastly is the equipment you recommended expensive?
Click to expand...

If you got a Studio Projects VTB1 preamp (also has phantom power, should you ever use a condensor mic) it would run you $120. The FMR RNC (Really Nice Compressor) goes for about $169. The VTB1 is a very quiet, single channel preamp that allows you to add some tube "warmth" if you want it.

While you can't go wrong with these purchases, you may want to spend some time on your technique in terms of vocal and guitar levels. You can adjust the levels during mixdown, if you don't do it too obviously. Make sure you're getting a good strong signal into the MR8, good high levels, just below clipping. It's always better to reduce levels when bouncing down than it is to have to boost a weak signal that was recorded too low. The latter approach adds noise.

Are you transferring to PC then mixing? If so, the adjustments you can make on the PC are almost limitless. Still, it's always best to record the best signal you can before having to tweak.

The latest issue of EQ magazine has a whole article on how especially good vocals were done. I believe they all used at least some compression. But, go get a copy of Janis Ian's "Breaking Silence". No compression was used. Great recording -- something to aim at.

Keep at it, you'll be a pro before you know it. And all the hard work will make you feel like you've really earned/achieved something.
 
Ed Dixon said:
Blue Bear actually has it backwards
Click to expand...
Ya shore don read so gud....... :rolleyes:

I have nothing "backwards" -- my point was regarding loss of audio quality due to round-off error in one versus two steps of digital processing.... the order was irrelevant to my point, but thanks for playing anyways...
 
Blue Bear is right.

And to add my two cents, if you are transferring your tracks to your pc and using a program that will accept VST plugins you need to try endorphin. It is a good softknee compressor and will do a good job on separate tracks or on the whole mix. You can get it here free: www.digitalfishphones.com
 
Actually not.

There are lots of good definition sources on the web for techniques like compression, limiting, and normalization. Looking up some of those might be a good place to start.

An originally recorded WAV file includes a set of audio samples over time. When you move that to the PC, you still have the same audio sample set. That set of samples is the only accurate representation of the original material that was recorded. It includes a specific dynamic range that includes a lot of the character of the original recording.

The normalization process attempts to adjust the volume of the original material without either introducing any distortion or changing the original dynamic range.

The compression process actually does just what the name sounds like, it compresses the dynamic range of the original. That changes the character of the original material. Limiting does similar things to the original material.

There are advantages and disadvantages to each of these processes. There are also various reasons when and where each are used in various audio areas. There is however no “One size fits all” approach that works. Each audio engineer looks that the sources they have, the objectives at hand, and then makes the choices as they see them.

Blue Bear likes to use the term “Sonic Integrity”, which is not really a common term. It does have a nice ring to it. If a process changes the dynamic range of a recorded track, one could easily determine that the “Sonic Integrity” had been altered.

The MR8 is a good piece of gear. Power Tracks is a good PC mix down tool and includes a variety of software based tools for these processing options. Each user needs to see what they have to work with, and then use those tools that work best for them.

Ed
 
Ed -- you really don't know your digital theory, do you...... I think it's time you picked up the digital bible -- JOhn Watkinson's The Art of Digital Audio and start reading, because you've spouted off some significant errors with regard to digital theory lately.

The following is a fact, and Ed can try and talk around it all he wants but he is dead wrong:

EVERY single process that you perform on a waveform requires mathematical calculation. Since the math involved is done with floating point precision, and the wordsize (24/32/64-bit) is limited by the hardware or software involved, there IS round-off error that WILL occur. That round-off error is precisely what causes the sonic degradation. The first process may not result in noticeable changes, but it IS cumulative for every DSP you apply.

Now I'm not going to argue any further with you until you learn more about the subject, so I suggest you start reading.
 
Blue Bear, you're preaching to the choir. I think my Ph.D in mathematitcs pretty much qualifies me in digital theory.

Ed
 
Then explain the nonsense you posted previously regarding "no loss/degradation" due to normalization...
 
theory

1. general principles of a subect
2.plausible of scientifically acceptable explanation
3.judgment, guess, or opinion.
The information in one book by one aurthor is opinion not law
i compress the vocals, then use normilization to get equal volume between 8 or so songs on a cd so they all have the same level.
 
Blue Bear Sound said:
Then explain the nonsense you posted previously regarding "no loss/degradation" due to normalization...
Click to expand...

The term "no loss" only appears in one post on this page, yours.

If you're going to ask questions, at least get the quote right.

Ed
 
dave in toledo said:
1. general principles of a subect
2.plausible of scientifically acceptable explanation
3.judgment, guess, or opinion.
The information in one book by one aurthor is opinion not law
i compress the vocals, then use normilization to get equal volume between 8 or so songs on a cd so they all have the same level.
Click to expand...
:rolleyes:

Digital recording principles are well-defined and not "theory" in a literal sense.

The Art of Digital Audio is considered oneof the premier reference books on the subject.
 
That is not an answer to the question. Please get the quote right if you want to discuss a subject.

Ed
 
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