Oh my, I'm bumping this because I absolutely LOVE this plugin! I am using it on virtually every piece of interview audio I record. Air conditioner noise, refrigerator hum, computer power supply fans... One fader move and they're gone! This is just so much easier than the multi-step way I used to do it!

On their web site they have an add to cart with choices of TDM and Native.

What on earth is a TDM? What is the difference between TDM and Native besides price???

There is no FAQ that I can find to answer my questions.

Video showing example is poor at best! I cannot see any difference in the waveform as they move the slider up and my poor hearing cannot hear any difference.

Sorry but poor webs sites makes me doubt their products.

"TDM" is a format that uses the extra Pro Tools hardware that is a part of a full blown Pro Tools system. "Native" is the one you need for programs like Vegas that do this processing with the CPU.

So you're running the Native version in Vegas, without a wrapper?

Bob.

Worked great on my system. (The website I mean)
JJK

My mistake. It is just a VST version. I have Pro Tools and it installed without using my Fxpansion wrapper so I believe it came with both versions.

The Waves NS-1 plugin is a piece of crap in my ears.

Erni

"The Waves NS-1 plugin is a piece of crap in my ears."

How so? Please explain some details.

Here is what I'm working on right now. It's a video for a project that my church is involved with. These were done with a Sennheiser ME2 into a Beachtek and into my Nikon d5200. There was plenty of room noise before I applied the NS-1 plugin.

http://vimeo.com/49865356

To me, it sounds the same as the Sony NR plugin or iZotope RX for this application. The advantage is it is a heck of a lot faster. I didn't have to transfer the audio to Soundforge, find a bit of room tone to make a noise print, apply the noise reduction, then load the cleaned up audio back to Vegas. I just inserted it as a filter and moved that one fader a little. It doesn't sound better. It's just quicker.

And that sure fits into the "Vegas" way of things.
JJK

Waves makes some great stuff. The L2 Maximizer has probably allowed more wannabe mastering engineers to destroy more good recordings than one can imagine. LOL

It just seemed that they kept their prices too high when others were coming out with the same thing, and it got to where I couldn't afford them anymore.

I hear ya Tim, "If some is good more is better"
I can already see the Waves forum, "Why won't the NS-1 fix my dialog audio.. it was recorded with a 'good' camera mounted mic"

I typically record my interviews in whatever location is available, quiet, and relatively decent looking. I use lapel mics mostly because I work by myself and trying to manage a boom mike, a camera, and ask the questions at the same time is unmanageable. I turn off the AC and get the environment as quiet as possible. There is of course still quite a noise floor in any sort of location interview. This environment is where NR such as the NS-1, Sony NR or iZotope really helps.

Laurence, I use lavs most of the time too (Sanken COS-11) and find that adding the windscreen, even when indoors, helps to get rid of a lot of background noise.
Proper mic placement (a black art all by itself) is another useful thing to experiment with when you have the time.
When I have used a boom mic, I put it on a C-stand, get it in position, set a level and then concentrate on getting good footage.
I have two audio channels on my camera so I'll run one channel a bit hotter than the other one, giving me a choice when I get into the edit suite.

Mike

I don't see how adding a windscreen would have any affect on noise other than by damping the high frequencies a little.

Noise of course comes in all frequencies. The wind screen will never attenuate low freqs, but the high stuff is easy to stop hence a wind screen will help just like a pop filter helps with the voice s and p's.

One trick that sometimes works, if you have a hot enough signa,l is to copy the track, invert it, and reduce the output against the original.. While it dampens the overall signal the voice is still loud enough to work while killing some of the noise floor. This is how xlr connectors reduce noise. The third wire carries and phase inverted signal of the ground.

I believe most noise flloor programs work off the same principle. Just the algorithms do better at preserving the louder part of the signal.

"One trick that sometimes works, if you have a hot enough signa,l is to copy the track, invert it, and reduce the output against the original"

That does nothing to reduce the noise relative to the wanted signal. Both will be attenuated by the same amount.


"This is how xlr connectors reduce noise. The third wire carries and phase inverted signal of the ground."

Not so. With a balanced audio line you have two different signals.

1 is the wanted signal which is the difference between the two signal wires.
2 is the signal between the two signal wires and the ground which has the wanted signal plus any noise induced in the cable.

"I believe most noise flloor programs work off the same principle"

Noise reduction is achieved using a combination of techniques, the simplest being a gate. You'll find noise gates in the FX chain of every audio track in Vegas. All a gate does is attentuate any signal below the set level.
More sophisticated technique use notch filters to attentuate the frequencies of the noise. SF's NR2 can use upto 4,000 such filters, their setting is determined from a noise sample. More advanced techniques use a combination of smart gating and filtering. Noise can also be detected and separated from a wanted signal when the noise is random and the wanted signal is not.

Bob.

I thought XLR worked as follows:

At the signal entry end where the signal is assumed to be noise free, the signal is split. One copy is inverted with respect to the other. The two signals travel in parallel in close proximity to the other end of the XLR. Any noise induced into the cable will be induced nearly equally into each signal. When the signal exits, the inverted signal is inverted again, and the signals are re-combined. The two copies of the original signal reinforce one another. The noise component of each signal are now inverted with respect to one another, and should very nearly cancel one another out, leaving the (very nearly) pure original signal.

Is that incorrect?

This is going off topic here, electrical XLR noise 'reduction' has nothing to do with acoustic noise pickup. The two are completely different issues.
With an XLR system, it is said that the signal is 'balanced'. That means the original signal is sent down the line, both inverted and non-inverted down two wires usually referred to as 'hot' and 'cold. These are always very close together in the cord, sometimes twisted together. There is also a screen or earth which has nothing to do with the signal at all - and can be disconnected unless you need phantom power, which is another subject again.
Any external electrical interference reaching the two wires induce the same amount of interference in BOTH wires at the same point more or less in the same quantity. That is crucial. The interference in both wires is therefor 'in phase' but the original sent signal is sent 180 degrees out of phase.
At receiving end, we use a differential amplifier - which only amplifies signals out of phase - any in phase signals (ie. the interference) is rejected. Result is clean audio that is immune to electrical clicks like light switches going on /off, mains hum, dimmer packs etc..
Microphone pickup noise is just that - microphones doing what they do best, picking up sound waves. And this includes sounds from sources that you may not want. Microphone placement is important, and reducing ambient noises at source should really be done during the shoot. But its not always possible for practical purposes to do that, and thats when noise reduction plugins come in handy - like this one.

Paul.

"Is that incorrect?"

In modern transformless balanced audio circuits, yes. In some as Paul says above there's even "hot" and "cold" and the two active signals don't even seem to be out of phase. My audio days go back a long way to when transformers were the norm for long audio line interfacing and there was no "hot" or "cold" and the ground had little to do with anything.

I still have a pair of expensive 600 ohm isolation transformers in my audio kit. I rarely have to use them these days but when all else fails they provide excellent isolation and noise rejection.

Bob.

Sorry for my bad explanation of how an xlr works. Here is the simpliest explanation:
http://www.ians-net.co.uk/articles/balanced_lines.php

However I in no way intended the comment to mean that would reduce room noise picked up by the microphone. It was only a reference to the way inverting a signal can reduce unwanted artifacts.
Should have never said it.

And yes Bob after I wrote about that old trick. I thought, "Well that never worked, its really no different than pulling down the gain."

But noise reduction algorithms do work by reading the low noise and frequencies and invert/and or remove it to cancel that out while ignoring the higher signal.

Here is the description from one noise reduction alogrithm:
The noise reduction algorithms are developed to remove constant background noise, hiss and hum in audio files with slowly varying backgrounds, such as lecture/conference recordings, static podcasts, etc. They analyze the audio file, classify regions with different noise prints, extract the noise characteristics and then remove the background noise in each region.

I see your point now Tim. Yes indeed, by isolating the noise from within an audio signal, then subtracting that from the original signal will result in audio with less noise. Thats true. But thats the easy part!

The hard part is writing an effective algorithm to select and isolate the noise component. Thats tricky, its not just a gate, although that can work by itself if the wanted source level is high enough compared to the noise floor. Taking this further, it depends of how clever the designers thinking is to tackle the problem. Thats why some NR filters are better than others under varying conditions. At the end of the day its a question of what plugin is best for a given situation. There is not one-to-fix-all algorithm although the izotope RX noise reduction filter seems to be the best at this time. Its algorithms are very complex and the results are staggering. But its expensive!
I will say, if the NR filter does the job, even if thats just a gate, then use it. Speech is fairly easy to deal with and this NS-1 plugin seems to do that well enough, but when you have music with background shouts or otherwise annoying noises - thats when you need to call in the big guns.

Paul.