Rx 7 Voice De-noise

[Michelle Benitone]

Imtrying to use RX 10 Advance voice de-noise on a 5.1 Group. I Would like for it to process only the center channel. However I am unable to select the center channel in my input configuration in the plugin window.

Have you considered/is it an option, to use a mono instance earlier up the chain? If you only need denoise on 1 channel, the best thing is always to edit the source channel before it goes to a 5.1 bus.

If that is not possible why not split the 5.1 source (a file? This is unclear to me) and only treat the center and combine the 5 other channels to different tracks.

Yes no problem I can definitely work around it. I just saw that I am unable to use it on a 5.1 track and wanted to know if I was doing something wrong.

For this super noisy project, I wanted to send all my noisy dialogue tracks to Dialogue Group/Bus and filter compress de-noise the whole lot in one go. But in when you send a mono group to 5.1 you lose your individual panning on the tracks if all goes to a mono group. Hence wanting to send all to a surround Group and treat that. Goyo, Acon etc all works fine on a surround bus. I just saw that RX was not happy as an insert on 5.1 group

Many plugins are only mono or Stereo. I remember making a spreadsheets a while back with the channel configurations for the plugins I own. I do not, nor ever intend to own, any Izotope plugins. Too buggy.

Have you considered/is it an option, to use a mono instance earlier up the chain? If you only need denoise on 1 channel, the best thing is always to edit the source channel before it goes to a 5.1 bus.

Voice De-noise is an intuitive, zero latency de-noiser that offers high quality results on a variety of material.

Voice De-noise can intelligently analyze speech signals and determine the best noise threshold for your signal. In a DAW, this feature can be used to write automation in case you need to override the automatic settings and correct the noise threshold by hand.

ADAPTIVE MODE: Analyzes the incoming signal and adjust the noise threshold automatically to compensate for changes in the noise floor. This can be useful for removing noise from recordings with variable noise floor and continual noisy sections, and works well for almost any recording of dialogue and spoken word.

Adaptive mode considerations

The noise threshold settings in Adaptive Mode may be different from the settings achieved by running Learn to set the noise threshold manually.

Because the adaptive noise threshold is continually being adjusted, it is set lower to prevent artifacts from occurring.

THRESHOLD: The master Threshold control allows you to offset all Threshold Node values by the same amount. If you find that processing is too aggressive or processing is affecting audio you want to leave unprocessed, try adjusting this control.

Voice De-noise has been specifically designed to provide high efficiency, zero latency adaptive noise removal when inserted on a track in your DAW or NLE. The Spectral De-noise plug-in is far more resource intensive and uses higher latency.

I extract audio clips from a video file for speech recognition. These videos come from mobile/other handmade devices and hence contain a lot of noise. I want to reduce the background noise of the audio so that the speech that I relay to my speech recognition engine is clear. I am using ffmpeg to do all of this stuff, but am stuck at the noise reduction phase.

If you are looking to isolate audible speech try combining a lowpass filter with a high pass filter. For usable audio I have noticed that filtering out 200hz and below then filter out 3000hz and above does a pretty good job of keeping usable voice audio.

In this example add the high pass filter first to cut the lower frequencies then use the low pass filter to cut the higher frequencies. If needed you could run your file through this more than once to clean up higher db frequencies within the cut frequency ranges.

Either of these should be much better than highpass / lowpass, unless your only noise is a 60Hz hum or something. (Human speech can still sound ok in a pretty narrow bandpass, but there are much better ways to clean up a broadband noise background hiss.)

ffmpeg doesn't have any decent audio filters for noise-reduction built in. Audacity has a fairly effective NR filter, but it's designed to be used with 2-pass operation with a sample of just the noise, and then the input.

The comments at the top of explain how it works. (basically: suppress every FFT bin that's below the threshold. So it only lets signals through when they're louder than the noise floor in that frequency band. It can do amazing things without causing problem. It's like a band-pass filter that adapts to the signal. Since the energy of the noise is spread over the whole spectrum, only letting through a few narrow bands of it will reduce the total noise energy a LOT.

See also Audio noise reduction: how does audacity compare to other options? for more details of how it works, and that thresholding FFT bins in one way or another is the basis of typical commercial noise-reduction filters, too.

Porting that filter to ffmpeg would be a bit awkward. Maybe implementing it as a filter with 2 inputs, instead of a 2-pass filter, would work best. Since it only needs a few seconds to get a noise profile, it's not like it has to read through the whole file. And you SHOULDN'T feed it the whole audio stream as a noise sample, anyway. It needs to see a sample of JUST noise to set thresholds for each FFT bin.

So yeah, a 2nd input, rather than 2pass, would make sense. But that makes it a lot less easy to use than most ffmpeg filters. You'd need a bunch of voodoo with stream split / time-range extract. And of course you need manual intervention, unless you have a noise sample in a separate file that will be appropriate for multiple input files. (one noise sample from the same mic / setup should be fine for all clips from that setup.)

The cb (conjoined-burgers) model is the one I found most impressive and versatile. I also found this filter pretty efficient (doesn't seem to use more CPU than the loudnorm filter for instance).

This filter accepts stereo input and produce surround (3.0) channels output. The newly produced front center channel have enhanced speech dialogue originally available in both stereo channels. This filter outputs front left and front right channels same as available in stereo input.

As an aside, the reason this is using ffmpeg to pass the output to ffplay is because I don't have a version 5 copy of ffplay, thus the dialoguenhance filter would be missing, if I tried to use it directly.

Reaper audio recording software is awesome, and is what I recommend to everyone. I use it every day. And I also seem to learn something new about it every day. It has an incredible noise reduction tool built right in that I just discovered recently.

The noise usually comes from a combination of stuff happening in the space/room where the recording takes place, and the electronics of the microphone and other gear involved. Recordings sound much better if you can reduce the noise, and that is what noise reduction tools are designed to do.

In order to do this, the software has to know what noise looks like so it can separate it from the signal (voice). So you have to highlight a section of the recording where there is ONLY noise, and no voice, and feed that sample to the noise reduction tool. Once it has the noise profile, it can do its thing.

My favorite recording program, Reaper, is a digital audio workstation (DAW), which is a fancy way of saying full-featured multi-track recording program. For more information on why I love Reaper so much, see my article, Why Reaper Rocks As A DAW.

ReaFIR is a EQ and dynamics plug-in that includes an FFT spectrum analysis window. Amongst other things it can be used as a precision EQ, a gate, a fast attack/release precision compressor, a noise reduction tool.

So see it in action in the video above. But here is is a written summary of how it works. Once you have recorded some audio onto a track, click the FX button in the track control panel. Then select VST: ReaFIR from the Cockos collection of FX plug-ins.

Then click on the Toggle Repeat button (down by the Play and Stop buttons). This is important because it will prevent any of the actual voice signal being played when sampling the noise for ReaFIR. With that area still highlighted, click on the FX button again to open the ReaFIR window.

The louder the noise and the more varied the noise (if it contains lots of frequencies and intermittent clicks, pops, etc.), the more likely you are to have that swirly artifact left over after noise reduction.

Awesome! Works very well. I was doing a album video-review and I kind of forgot how reducing noise with Reaper worked. So I quickly searched for some guide on the internet and stumbled upon this guide. the instructions in the video were very easy to follow and I learned something new as well, I think I never used ReaFir. So thanks a lot!

exactly what I was looking for. Very effective for voice recording with cheap microphones. For me, the noise is reduced so much that it sounds unnatural. So I reduced the effect level to about 70% (70% wet). Thank you for the very useful tip

Worked great to pull tape hiss out of a rip of a 40 year old cassette. Some audible artifacts in very quiet spots, but way less annoying than the hiss. Expected to have to work way harder to do it. Thanks for posting.

This is a great tool! Thanks for the instructions. I have just used the tool to remove noise from two songs on an old tape recording (perfectly), but a 3rd, from the same session on the same tape is playing back with a subtle noise stutter throught the song, as though noise has been removed in chunks about a half second apart & bits of noise have been left behind. Any idea why that might be? I tried using noise samples from both before & after the song for the subtraction process. Or are there other ways to attempt to de-noise?

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