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Mad_Cat
01-16-2010, 07:38 AM
found this in my travels thought i'd share



Processing initial sound file

1 )DC offset removal and frequency response limiting to 20 Hz-20 kHz

DC offset is a shift of sound form from the baseline. The shift itself means nothing, but if you are going to edit sound file, first thing you should do is to remove it. Otherwise any further file processing will be done incorrectly and result in distortion. DC offset removal can be done by Remove DC offset command of the sound editor.

Limiting frequency response to 20 Hz-20 kHz (i.e. to audible range) is necessary for reduction of distortion appearing during file processing and its subsequent DA conversion. This can be done by applying Paragraphic Equalizer of the sound editor.

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2 )Frequency range equalization and special processing

Frequency range equalization of sound file should be done very carefully. This editing field is fraught with appearance of artifacts and distortion. The best results while processing go for Waves Paragraphic EQ plug-ins. For bass enhancement (if the sound file has normal low frequencies) can be used Waves MaxxBass plug-in.

Appearance of Steinberg FreeFilter plug-in kicked up a row in the world of PC sound editing. This unique plug-in is an intellectual equalizer able to remember reference phonogram and apply its frequency characteristics to your own sound file. Thus you can mix into one file different music pieces or match bad CD to your favorite sound pattern.

For stereo enhancement and sound space processing you can apply Waves StereoImager+ plug-in. The best effect so far produces its Spatial Enhancer preset.

For realization of recent fancy QSound technology you can apply QTools plug-in. With its help you can get effects of 3D sound, Pseudo Stereo, Expanded Stereo, etc. However it is necessary to note that yet becoming nice the sound acquires some affectation.

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3 )Channels balancing, normalizing and sound form density increasing

Channels balancing is done by increasing or decreasing volume in one of the channels by Volume option of the sound editor. To apply this you should first play the file and mark average difference between channels in dB as per volume meter, then set this value in Volume option.

For normalization, increase of sound density and audible resolution you can apply Waves L1-Ultramaximizer+ plug-in. This is the most effective plug-in for such operation. A similar Steinberg LoudnessMaximazer plug-in is ineffective. However, incompetent use of Waves L1-Ultramaximizer+ may result in sharp sound distortion as well. This operation should be applied only at the final sound processing stage.

MP3 pro club

DJ Hero
01-16-2010, 08:49 AM
you're better off cutting BELOW 20hz, and ABOVE 20hz...I typically will cut 19hz and 21 hz.

Brandon
01-16-2010, 10:32 AM
Here's a nickle's worth of free advice: render to -2db and get a copy of oZone

Puff Powers
01-16-2010, 11:40 PM
^ what he said!

Botrytis
01-19-2010, 06:25 PM
Do the DAWs have good conversion? i.e. logic ableton cubase