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5

In addition to a little reverb (convolution reverbs are critical tools for sound for picture, IMHO), it will still sound like soundtrack music with extra reverb unless it is filtered. You could run it through a speaker emulator or guitar or bass amp plug-in, or you could do high and low pass filters along with a little high shelf. Instead of low pass + high ...


5

This is called a sibilance, or sibilant articulation. http://en.wikipedia.org/wiki/Sibilant You can avoid it in the recording process using a pop filter, but in post production there are a handful of de-essers available for taming those sounds. The process involves compressing just the frequency range that contains the sibilant consonants through ...


3

This is all a bit over-simplified, so please forgive me if you know already... 'volume' in digital terms has an absolute maximum, unlike analog which you can 'push a bit' without it getting unpleasant. As Jim Mack says, checking for 'clipping' [squared off waveform tops] is an indication you have passed this limit. If, when you analyse the file visually, ...


2

Noise in the louder parts is probably clipping, since louder sounds usually mask 'floor noise' -- you're more likely to hear actual noise in quieter parts. I assume Audition will let you view the waveform. Look at the upper and lower tips of the waveform, moderately expanded in time. If you see the tips squaring off against the limits, you're clipping. ...


2

I suggest you should use multi-band compression, but as expansion: You set each band with the threshold slightly under the level of the sound you want to amplify and then set a positive range. Everything that crosses the threshold will be amplified, the noise floor stays where it belongs. Since you use a multi-band compressor, you will amplify the parts of ...


2

Mess around with EQ. Try lowering the treble a bit to muffle the sound. Normally, you would put a low pass filter on to make it sound a bit muffled. As for talking to him\herself, Increase the bass a bit, and do not apply reverb to it. In film, we typically have room tone, reverb and shave off most of the low frequencies to give it the sound of being in a ...


2

You need to add reverb and reflection similar to that room. Think convolution reverb with an impact that fits the restaurant space and eq with low pass if there are obstruction in the way of the sound. It might help to tweak panning and stereo spread to fit the location of the camera


2

An equalizer is a combination of filters with different shapes that you adjust in some constrained way. The FFT filter is based on the Fast Fourier Transform, which is a different way of manipulating signals. It can be viewed as having a direct control of the amplitudes of a selected number of bands (e.g. 2048 bands) in the frequency domain. In other words, ...


2

There's Way too much to go into here, about leveling audio, and as I haven't heard the audio clip, I will keep it short. The speech leveller is probably causing the problem and you may not be using the hard limiter properly. I would suggest scrapping the speech leveller and instead using the single band compresser with 'instant' attack(0ms) ,a short ...


2

Yes, there will be a difference. Mp3 is a lossy codec that deteriorates with every generation. It's bad practice to use mp3 (or lossy compression in general) at any stage where the audio may yet be edited or reencoded. Some programs are smart enough to simply copy the original audio stream without reencoding if possible, but I can't say whether this applies ...


1

It seems you've searched the keyboard shortcuts so the next step is either some type of macro recorder within AA(?Like Cubase has?) or an external keyboard macro recorder. You could then have a single shortcut key to kick off this list of shortcuts you posted. I'm not sure about deleting the markers, but surely the in/out markers would just relocate on the ...


1

Surprised nobody here has mentioned actual denoising algorithms. If you can't remove the source of the noise or can't re-record what you already have, look into effects such as izotope rx and edison. Even Audacity, which is free, has a noise reduction algorithm. You've highlighted a section of the pure noise, which is what is fed to the algorithm as an ...


1

You need to use EQ/Equalization on the dry voice and boost resonant frequency with a narrow notch filter. It might match closer if you then apply a basic reverb afterward, but it will take some trial and error. The important thing about modal frequencies is that practically they are just a higher amplitude of that frequency. If you can identify one or two ...


1

A "speech volume leveler" is just a fancy name for some proprietary algorithm they've cooked up to try to make things simpler for the amateur user. The basic, building block tools to control your levels here are a compressor and an expander. You'll probably need both. Apply the expander first. If you set the threshold and other attributes correctly, ...


1

First of all, you are making a fundamental error which is resulting in a higher noise level. This error is with your recording levels. The level of your voice is too low during recording. Secondly, you have DC bias(offset) occurring somewhere in your system. This lowers the highest level your voice can be recorded at. Again, the result is a higher noise ...


1

Mute the track you are recording to and make sure to turn off the microphone in windows playback.


1

Do some noise reduction before your dynamics. I usually try to cut all the bad stuff (in your case, noise) before I go boosting anything (volume or frequency). I'm not sure what your material is, but I like to use McDSP's mulitband limiter to increase volume and do some downward expansion. I've had good success with it for dialog.


1

Also with these programs (audition etc) you can generally find a noise reduction plugin. You can then possibly reduce some of the noise (hopefully without artifacts. As to limiter, personally I'd compress rather than limit, but it also depends on the amount of dynamic range you're talking about... If it's not hugely dynamic then maybe even using an expander ...


1

Just link the pan of the stereo channel to the level of left channel before mixing down to a mono one. Try turning it a bit so that when left level is not to high, the pan stays at left, and when it gets peaked/clipped, the pan drives to the right. That would sound more natural and smooth than suddenly switching channels at times. EDIT: An example. FL ...


1

Nothing which is going to do it for you, but it's not complicated to mix it up yourself. You can use the channel mixer (in effects > amplitude and compression) to get two mono files, one left and one right. Then you can do a mix in the multi-track view using volume automation to level each out. Pan if you like or mix down to mono.


1

Adobe Audition can do exactly what you want! In the Edit view, go to Window -> Marker list (Alt-8) Select the marker regions you want. At the bottom there's an icon that looks like a save button (I added a red circle around it): As you can see from the screen-shot, you can save them as individual FLAC files.


1

You could use ProTools software. Just use the strip silence tool to remove the silence between the sounds. Then ProTools will put all your snipers into your Region list. Then you simply need to collect all audio files in the Region list and press Export Regions as Files. Then you can decide in which Format and at wich Location to bounce. Hope this helps! ...


1

An "FFT filter" is not an established term. The main purpose of FFT is to speed up convolution with an ongoing signal (cf "overlap-add" and "overlap-shift" algorithms), so I expect the "FFT filter" to just be an implementation of a long-response FIR filter that is rather efficient at the price of considerable time lag. For offline processing, the time lag ...


1

The algorithm is different. Typical eq works in a simmilar way that analog circuits does, but in a digital domain. There are FIR (finitive impulse response) and IIR (infinitive impulse response) filters, not going to much in technical details. FFT filter makes fourier transforms of the sound to frequency domain, where you can easily acces distinctive ...


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