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4

Normalization is a very simple process. It finds the highest peak and applies gain (makes everything louder) until the highest peak is at the specified maximum level (-1dB in the picture). There's also the option to remove DC. You can imagine DC as an upward or downward shift in the waveform, making it stay averagely more on the positive or the negative ...


3

+1 for Jurgen's mentioning of the Hilbert~ object in PD. That object is also in Max/msp. I think PD or Max are pretty good environments for exploring this type of thing. Most of the objects have pretty good examples as help files. For example in Max you get given this for Hilbert~ alt text http://s16.postimage.org/5n4wjvp79/Hilbert.jpg You can then use ...


3

unless they are very talented, i think it's a very good idea to not use employees on videos. so your boss is right, but he should also bare the consequence of it and hire an (amateur) actor. you'll get a professional sounding result unlike with any of the options you've already mentioned. just make sure it's recorded and 'directed' properly.


2

1 & 2) You seem to understand the reason your compressor is a part of your signal chain. It sounds as though your compressor is set up to adjust the level it receives from your console to about -3dB. If it is working for you then it is in a good spot IMO. However, I don't think your Topaz console has a master insert. An insert literally places your ...


2

Ogg Vorbis doesn't suffer from much of the horriffic issues that occur with MP3, and generally produces smaller files. Hi Quality Voice Recorder will do what you want. If you need smaller, obviously gsm which is the encoding used for mobile phone calls should be suffficient, but there is less software compatible with this format.


2

You could probably use Melodyne to alter the pitch slightly, and preferably the formants the most. This could make your voice so different that your boss won't recognize it for beeing you, and still don't make your voice into an artificial sounding one.... There's a free 30 day trial version at their website.


1

Protools has a "Split Silence" function which can be used to seperate out all elements where the audio level does not reach a particular threshold. Assuming that most of the audio is actually speech, then this is where I would start. Once you have all the speech segments, you can then 'shuffle' them down to a single region which you can then export. Yes, ...


1

The best solution for this longer term is to use the Opus codec. There isn't anything better in terms of speech/music compatibility and also bitrate flexibility. Compared with mp3 it is a fairly new codec, but support for the codec is growing rapidly and most browsers have support built-in to allow you to play back Opus encoded data from within the browser.


1

Not exactly sure what you're asking for. You don't really explain what the videos are - tutorials, adverts?? The intended audience and expected quality are obviously key factors here. There are text-voice synthesisers out there, but all of them are recognisable for what they are. You also pay for the better of them. There is no effect which will make you ...


1

Smiley Sam, I'm a ""professional"" (note the quadruple air quotes) sound designer/composer/musical type who is particularly finicky about "bang for buck"... and While getting specific details from you is helpful, I have a simple answer that's worked for me and many others: Get an Audio-Technica ATR2100-USB. The price has risen a bit since I got mine, but ...


1

1) This isn't invalid, but it is a little weird. An insert goes before the fader where as the output goes after. Using the insert would give the master fader total control over the output level where as the positioning on the outside of the board means that a) it will protect the output gear from accidentally going too loud and b) it will make the behavior ...


1

Oh jeez. You mean the speaker's handheld wireless runs into the speakers in the ceiling? I bet there's a million of them up there. That's why it's sounding more refelctive. The more sound sources you have in a room (depending on how it's been designed, and I'm guessing this is a fairly normal, rectangular meeting/ballroom) the more sources for ...


1

Two people come to mind when I think of language and sound art. Diana Deutsch and Kurt Schwitters Kurt Schwitters has a famous sound work called "Ursonate". It is phonetically based and may give you some ideas of structure or the way a language can sound. Links to listen to "Ursonate" on this page: http://www.ubu.com/sound/schwitters.html Diana Deutsch ...


1

Tolkien's Middle Earth languages comes to mind, although of course they have been used in the Lord of the Rings films. Also Anthony Burges' A Clockwork Orange uses Nadsat, a conlag based on Russian.


1

Interesting subject material! Firstly, the problem appears more to be a pre-echo problem rather than a post-echo problem. I'm not sure how this has occurred, but the sound of your voice in the recording appears to be preceeded by a pre-echo of it coming through skype. Additionally, the audio is very distorted and has been recorded at a very high level, ...


1

"Is it possible to create a voice from scratch with no recordings." One system I was looking at is from the 1970's called VOSIM. (Just wrote a Java version of the basic algorithm about a month ago. It only makes simple tones, not words.) Yes, it sounds very robotic and artificial. But I think it was (is) possible to make recognizable words, especially with ...


1

It sound like the good version has simply been cleaned up using equalization. The humming you speak of is just bad low frequency 'rumble'. It's a bad recording, but in the cleaner version somebody has removed some of the more troublesome frequencies, and maybe boosted some of the brightness. It would be pretty easy to tidy up the 'bad' version using the ...


1

Try leveling, a much used compression technique. set compressor speed to the fastest setting lower threshold to the lowest setting play with the ratio between 1...2(2is kinda extreme 1.2-1.3 is recommended) At last if you want more volume try a limiter!


1

You're not gonna like this. The only for-sure way of lining up audio files is to have them All the same length to begin with, including the silences. So when you record them, make sure you start from zero with each file, even if it only has one bleep five minutes in! Lining up varying length soundfles is a nightmare otherwise. Yours respectfully Chris


1

Unfortunately boosting the signal you have is unlikely to help - where you don't have the information in the signal you can't really create it. As you have already seen, while DSP's can do a lot to increase the distinction between noise floor and signal, and between unwanted frequencies and speech you are ultimately limited by the quality of the recording. ...


1

The direct answer is very generic: Frequency: Changes in wave frequency relate to pitch/intonation Amplitude: Changes in wave amplitude relate to stress and pronunciation But that is pretty much the answer to every other "What are the characteristics of X that make it different from other X'es, in terms of properties of the sound wave?" The more ...


1

So long as you're doing some sort of transformation on the vocal sound, it can be reversed. A vocoder is probably the the best way to go if you need real-time disguising. If concealing vocal identity is paramount, then you shouldn't be utilizing a voice recording at all. You should instead use a method that totally obfuscates the original vocal signal, ...


1

The best thing I know is to use TRAX software from Flux / IRCAM, for which an astonishing video is posted here. The software enables independant Pitch / Formant / Gender / Age control as well as mapping to subtle LFOs, and ability to add further Formant/Spectral envelopes. The software is designed for both speech and singing voice signals. The easiest way ...


1

some pitch detunes and filters tied to the most random slow and medium fast LFO are very difficult to decode, like i think it would take a typical audio pro a lifetime before he gave up on it, any FX like volume, overdrive, vowel morph, vocoder, just has to be on super random LFO's.


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

Five years later and there are a number of cloud services that offer to solve this challenge. Services https://otter.ai/ and https://cloud.google.com/speech-to-text offer some alternatives for the problem of "speaker diarisation", which identifies when specific people are speaking. Once you have a list of times at which a specific person spoke in a recording,...


1

Summary: Use a noise removal plugin that can learn noise to produce a track with one person "noisereduced" and use this track as sidechain source for a noise gate. Here is how to achieve it with a noise removal plugin that can learn a sample, some EQ and a sidechainable noise gate: Make two empty tracks and four copies of the track and name them like so: ...


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