To reflect on your first question, Randy is referring to a psychoacoustic principle that falls under something called "scene analysis." I don't know that he's necessarily read a ton on psychoacoustics...but whether or not he has, the different effects are something that you begin to pick up on after a certain amount of experience. Basically, what's going on ...
It really depends.
Sometimes you need a global volume boost because the audio is too low anyway to correctly listen to the noise reduction you're applying. Sometimes there's a 50 or 60Hz rumble or any other fixed unwanted frequency that you could remove before denoising...
But generally speaking "the sooner the better". And before any processing other than ...
The first is that he said that one of his huge trade secrets in sound design are sounds that continually change in pitch just slightly. This way, they clash less with music and dialog and can be heard better through the mix because they're constantly changing. This makes a tone of sense and is genious. The question is...what would you use to do that?
Are you mixing the project as well? If not, you should consult the mixer first before you denoise anything, as that is traditionally the mixer's responsibility, and chances are they have their own specific ways they go about it. The lines are getting blurry, though.
My personal workflow is as follows:
1) Do all dialogue editing without any denoising.
Put it in a different room!
Professional studios do this by having a control room with all of the more noisy equipment, so in a home studio you're going to have to try to approximate that environment as best you can.
On a MacBook, you may be able to limit CPU usage in the Energy Saver preference pane. I can't tell you specifically since I'm on an iMac, but ...
The ReaFir plugin that comes with Reaper is pretty excellent. And you don't HAVE to have Reaper to use it, because you can download all of their VST plugins free. ReaPlugs Download
Insert the plugin
Change to "Subtract" mode
Select a range of audio that you can profile the noise to be removed and enable "Repeat" (so that it loops over the noise over and ...
You should be able to work some magic with those tools with a bit of patience. Apart from getting your hands on a CEDAR or ADR-ing it, you might get something half decent to work with.
Heres what I did in 3 mins using RX: http://soundcloud.com/andrewjohnlewis/noisy-dialogue-2
Very rough, but with some finesse you could get somewhere there I'm sure.
I usually have my Dial editors only do very basic level adjustments. Basically lower any really loud pfx or a line that is dramatically out of sorts with the rest of the scene. With noise reduction, I never want my editor to do any broadband NR or notch filtering. You can get into a lot of unnecessary processing that might not be needed once everything ...
It depends on the agreements made between the dialogue editor/supervisor and the sound supervisor. Mixers could prefer to not have volume automation, but this differs.
Why do you ask? Are you editing dialogue? Or are you mixing the movie?
In addition these spectral characteristics for clicks/pops, in your audio, localize some of your pop/crackle sounds by ear and scrub over them in the timeline, while looking at the spectrogram. You'll soon be able to pick out the anomalies visually.
You may also need to tweak your "spectrogram" view settings to show these artifacts more clearly.
I can see two reasons why you'd use a gate while recording instead of in editing :
1. You're broadcasting live.
2. You're using outboard equipment.
A gate should, generally, be first in your audio chain but in your case you might want to put it after your noise reduction. You never want to put it after the compressor!
If you do it in editing you have ...
Yup, that's definitely switching–power-supply bursts. (Has nothing whatsoever to do with hard drives, though it may sound somewhat similar.)
How exactly this gets in the signal path I don't know, but you might be able to eliminate it by changing your grounding configuration. One thing to try is to directly connect the interface to ground, for instance by ...
Adobe Audition has this feature:
The Noise Reduction/Restoration > Noise Reduction effect dramatically
reduces background and broadband noise with a minimal reduction in
signal quality. This effect can remove a combination of noise,
including tape hiss, ...
Try a parametric EQ, or a band-reject filter (essentially the same thing).
Setup your audio so that you can loop the section with the problem sound.
In your EQ or filter, adjust the controls so that your band is narrow and deep. That is, a small range of frequency is impacted, but that band is almost entirely attenuated.
slowly sweep the filter from low ...
Without knowing what you are recording:
If the main problem is rumble sound you could use a high-pass filter - that is if what you are recording isn't using a low frequency range. Try with values such as 60-80 Hz as a start point. In you record vocals/voice you can try starting from 100-200 Hz.
Try then to apply a low-pass filter to remove noise in the ...
What is your use case scenario? Field recordings or vocals?
You can if you have the advanced version of 2 or 3. You could also use the 'envelope' function to create a frequency curve that allows you to attenuate noise in a specific frequency range. Highly useful function especially with complex material (with loud low or midrange sounds), because it makes '...
It does depend but most will apply it early in the chain especially if it's an offline process like Izotope Denoiser. It's much easier for the software to work if the noise floor isn't moving around and being compressed, expanded, clip gained, volume automation etc.
Welcome to SD.
I think you might be placing too much importance on what this can do for you. Looking at what you have there, a lot of the noise floor is just broadband noise, I assume that came from the transfer medium. Clicks will show up as vertical lines when zoomed in, but pops and crackles are less easy to spot - especially amongst all that noise.
Use RX 5 Audio Editor.
You will probably get better results than with Audacity.
RX 5 allows you to recover certain parts that are missing.
Many quality presets are provided and you'll also find exhaustive documentation on the net.
'Sensitivity' is a measure of 'how much electric it puts out for a given level of sonic input'
It has nothing whatsoever to do with its ability to separate wanted from unwanted sound.
Microphones cannot do that, they have no brain, only ears.
Some are designed to pick up everything around them, known as an omni pickup pattern - i.e. it hears equally ...
There are automated algorithms like those you tried in iMovie, the ones in Audacity might be worth a try but if you want better results you should individually edit phrases.
Software like izotope RX can help in the restoration process.
It really depends on the material; are these sounds that you want to remove truly peaking at a lower level than every bit of the speech? How about in loudness or 'energy'?
You need to look into the kind of sounds you want to remove and see how they differ from the ones you want to keep. This is key to being able to separate them.
Are they of a shorter ...
Firstly, do any and all processing in WAV files. If your source material is on CD, then rip to 44.1kHz/16bit WAV and process at 24-bit in your software. Do not transcode via mp3.
Tracks like 104b should be discarded. There is nothing you can do with files like this. The studio has not done their job properly. They have recorded the transfer with input ...
Try doing a varispeed on a section of the file, slowing it down by 50%. Then run that through spectral analysis to find the pitch of the tone (or sweep it with an EQ). Then, simply notch out the actual tone (existing at 2x that frequency) from the original 100% speed. If everything's within the 48k domain you should be fine since your within the Nyquist ...
Well if it is inaudible, unless you plan on pitch shifting the audio down, it sounds like it doesnt need to be removed?
It is probably there because when it was transferred to mag, it may have transcoded frequencies from the mag above the our audible range and when you scrub, your pitching it down in varispeed which would cause it to fold down higher ...