EQ And Compression Techniques For Vocals, Acoustic Guitar

This article talks some of the basic Vocal and Acoustic GTR  EQ and Compressor studio recording skills. Even though this is talking about studio recording but still can learn different something from this article

For the vocal Compressor
 He likes to set the hardware compressor’s attack settings at around 30 ms and release settings at about 1 second and my compression ratio to 3:1. Then, he plays with the threshold making sure that at the hottest parts of the vocal, the gain reduction is at a maximum of -3 dB. This leaves me room in the mix to compress further using software compression plug-in.

For the Vocal EQ
If the sound too muddy or boomy vocal –he tends to pull a dB or two at around 200 Hz. This also has the effect of making the vocal cut through a mix better or sound brighter.
He’ll pulls a few dB at around 3 kHz if the sound too piercing or painful. This tends to take the edge off of the vocal without removing any of the clarity.

For the Acoustic GTR
Compare with Vocal, he likes to have a bit more aggressive with attack and release settings than you would with a lead vocal. Medium fast attack (approximately 22 ms) and release (approximately 500 ms) on a hardware compressor with a ratio of 3:1 and then play with the threshold knob until, again, the gain is attenuated by about -3 dB.

For the EQ
Pull 1 or 2 dB at 125 Hz to remove the boominess and adding air or breathe by using shelving EQ that boosts all frequencies above 12.5 kHz. Bringing out the low end in a vocal – add 1 or 2 dB of gain at between 80 Hz and 100 Hz.

Here is my question when I am doing FOH Engineer and the mix sound is perfect but during the artist solo part with some of background music, how could I suddenly change the EQ setting? Should I setup the “Scene and using fade out?” or have other ways can do it? 


http://www.prosoundweb.com/article/eq_and_compression_techniques_for_vocals_acoustic_guitar/

The bigger picture on equalization

I felt very confused after finish this article. They did not give us too much information therefore it makes me felt frustrate. First of all they talk about measure loudspeaker frequency not only “on –axis” response but also concern off-axis response. They suggest average every audience sit position and tune the speaker as flat as possible. However when we only turn one speaker on to measure the speaker frequency instead of turn both left and right on.  It maybe sound good when we just tune only one speaker on, what about turn both speakers on at same time? Do I have to tune the EQ again? Or what should I do? The second question is the phase issue by looking at the 3 dimensional radiation balloon.  I know a properly gathered balloon will reveal the anechoic response of the loudspeaker at all listener angles but I don’t understand when the article compare between figure one and figure two pictures. I understand the F1 picture but why the Figure 2 will come into phase at two-off axis position? What is meaning “at two-off axis position”?   

I felt the more I learn the more I don’t understand.

 

Digitally Steerable Arrays

The Digitally Steerable Array speaker is one of the newest technologies during this year. This type of loudspeaker is relative infancy compared with other sound reinforcement technology. Therefore it still can’t easy be find in the big live concert or arena.  According to this article, this unique audio animal is primarily intended for small-to-medium size commercial and house-of-worship applications where priorities include intelligibility, controlled dispersion and aesthetics. What’s the benefit of  “Steerable” function?  According to the last week article I posted. When we use more then one speaker in the same room at same time will cause comb filter happen and depend on the position you stand will have will listen frequency lose.

That is why have a lot of tricks to prevent comb filter happen.

Ex: Line array especially when we use Sub.  We use the highest frequency that the Sub has and known that highest frequency wavelength and multiply 2/3, that will be the distance between speaker and speaker you should put.

The other method is by using End fired array

The last speaker set “0” delay time and the speaker before the last speaker increase the delay time and so on. The purpose is put all sound together and become very big sound wave.

These skills are too difficult for consumers but if we have Digitally Steerable Array speaker that will be very easy. We can remote the position where we want without moving the cabinet.  (Think of it like your head is facing forward and is stationary, but your eyes are shifting up and down and back and forth to focus your vision on different areas of a room)

Most DSA have DSP system and amp inside of the speaker. Here is a company just release the kind of product

Developments In Cardioid Subwoofers

Developments In Cardioid Subwoofers

It is very often hear audiences complain sub is not strong enough however the artists some time will yelling too much low frequency especially in this line array world. This happen because most of subwoofer is Omni directional  and cause the sound towards the stage, performers, and then reflecting off rear walls to produce late arrivals in the main listening areas. That reflection cause delay sound and west many of speakers energy. Because of that reason, the manufacture start create more powerful low-frequency speakers. The  Multi-kilowatt dual-18 enclosures are now most common sub in the industry and most of them also provide rigging hardware and software((Electro-Voice LAPS, d&b audiotechnik ArrayCalc, Meyer Sound MAPP, Martin Audio Viewpoint, etc.) for set up the cardio subwoofers.

Cardio subwoofer setup is basically by reverse one or two speakers and set up delay as back reflection time and decrease the sound wave. The revers speaker cause the Omni-directional become Cardio pattern.