The channel mixer is composed of several sections: section inputs (in which I establish the type of signal to bring in the channel) amplifier section (with which port the input level to a standard level), equalizer section (by which I change tone signal), section pre fader aux (to manage the monitoring of the musicians), the fader section (which will establish the amount of signal to carry out), section aux post fader (with which I manage the special effects), section panel pot (which will establish the amount of signal to bring the output left and right), section assignment (which settles on how many and which outputs carry the signal). 
We have two entrances, one cannon to the microphone signal, (signal decoding software)and a jack for a dial tone. On the mic we 48v phantom power to supply power to the microphone. The Phantom is an internal tension to the mixer, which is resting on the microphone signal. The 48v engages with the negative mass of the microphone signal, and with the positive on the positive and negative of the microphone signal. In this way I send power to the microphone using the microphone cable itself. By ohms law the voltage propagates to both the microphone to the mixer, and in both cases meets a transformer unbalance. Transformers see the 48 st as a disorder, and is eliminated in the reconstruction of the signal. To power the microphone take the 48v before it gets to transformer imbalance of micro (and is therefore canceled) and bring the circuit being connected to the capacitor. The phantom power is applied only on the microphone signal, is not harmful to tools that do not require, and is used to power condenser microphones. The next device in the signal path is the impedance adapter, capable of restricting the mainsails level signal line (lowers the signal line because the preamplifier could not handle it). The mic / line signals are presented to the switch, by which I choose which to use using the appropriate button. To the selector follows the phase inverter, it has a button to turn the phase of 180 °, thus reversing positive and negative. You use this button when you have any problems of phase, for example when a source register with more microphones, these are located at different distances from the source the same, then some frequencies are to be in phase opposition because they have different wave frequencies (arrive in phase opposition to the mic). It happens then that feeling the most overlapping channels, have an abnormal sound, although individually every recording sounds perfectly.(go2DECODE) Are two possible solutions: either try to reverse the phase of one of the two channels, or change position of the microphones. This button can be useful for problems of feedback (whistling on stage). Inverter phase follows the unbalancing.
The preamplifier section is used to adjust the level of the input signal to the standard level of the mixer, corresponding to zero dBu (if the signal arrives top is lowered and vice versa). It 'the first section with the active components, and so it is very important, as a poor quality of the components can give rise to problems of stamp or puff. A preamplifier little can change the tone and add noise. The first component is encountered by the signal attenuator, face to dampen the signal because the preamplifier itself is not able to do so. It follows the preamplifier: in general all the microphone signals are amplified because they are under the 0,755v. The knob of the gain in the preamp there are values, and strangely this dial is turned to the right more these values ​​are negative. The values ​​shown on the gain knob Indian the level of the input signal when the meter read zero dB (I carry forward the potentiometer until the meter read zero dB, and when I reached him, the value indicated on the dial is the real level of the signal in entrance). Before being sent to the equalizer the signal goes to a hi pass filter, which has a technical action designed to remove unwanted frequencies that do not affect the timbre of the instrument (for example if the hi-hat microphone picks up the sound coming from the bottom, I can eliminate low frequencies for the clean sound of charleston.
To introduce the equalizer must open a parenthesis about the filters. 
The filters are devices which hold apparently spectrum and make pass the remainder. The elimination of a portion of the spectrum is equivalent to an attenuation less than infinite. On the cutoff frequency is already an attenuation of 3 dB.
The slope of the operation depends on the number of filter poles: the first-order filters have a slope of 6 dB per octave filters of the second order of 12 dB per octave filters of third order of 18 dB per octave, and filter of the fourth order of 24 dB per octave, and then a fourth order filter has a slope greater than one of first order. The fact that the filter cuts to minus infinity differentiates it by the equalizer.
The main filters are high-pass, low pass, the filters are band-pass and notch compounds. 
The classic application for the filters is the passive crossover, in which the spectrum is divided depending on the frequency and then be sent separately to the tweeters and woofers. In active crossovers are active filters that allow you to gain the signal amplitude through the preamplifiers.
And 'the control panel of the recording studio and its front panel I manage in total freedom the signal path. The various devices of the study are not connected in a direct way with each other, but via the patch bay, so that the engineer can change the path of the signal from only one panel, instead of by the various outputs of the devices themselves. It is therefore a personalized device of each study, as the connections between all the devices and the patch bay are fixed. You can therefore define the patch bay as an extension of the rear panels of various machines. E 'consists of a panel of aluminum to which are mounted the females of bantam, therefore a second advantage is that the connections are made via only one type of connector. The connections during setup in the patch bay are conducted on the basis of two principles:
The logic: suggests to place in the vertical connections that have a logic signal path between the upper row and the lower row, or it must do so in the upper row are positioned outputs that are in relation with the underlying inputs. For example, if the top row are positioned aux post out, the bottom row will be appropriate to the revenue they are the effects, if the above are the outputs of the master shall be positioned below the revenue of the two track recorder, and if the top row are the direct outputs of the mixer, the bottom row will logically put revenue multitrack.
The optimization: is divided into two further principles, namely the optimization of the logic blocks and the one of the spaces. The first suggests that the blocks (or groups of related inputs or outputs) are nearby places if they are related to each other. For example, if the top row were placed the aux out and bottom in the efx, the neighboring blocks are efx out the top row and line in the bottom row. The optimization of space suggests instead u savings of empty holes, or whether advance of the females bantam not used in the top row (in which it is usual to lay the outputs) and you need to make space for the inputs, the inputs should be put in the top row rather than use a second patch bay.
For correct setting of the patch bay must respect three rules: 
1) Place the outputs and inputs on the top row in the lower (exceptions) 
2) Always write numbers and letters in / out and arrow standardization 
3) Do not add it detracting from the list of proposed material (quandoi I get the list to do the project I have to stick to the list)