Phantom Power; Phase Reversal; Input Sensitivity; Sound Adjustment - Monacor DRM-884DTM Instruction Manual

7.6.2 Phantom power

[10] IN1 – 8 only: phantom power, 48 V
Caution: Only activate the phantom power when the audio sources con-
nected expressly require phantom power . Phantom power may damage any
audio sources with unbalanced outputs . To prevent loud switching noise,
only use the button when the input has been muted (Mute [38]) .

7.6.3 Phase reversal

[11] "Polarity", phase reversal switch: 180° = signal inverted

7.6.4 Input sensitivity

[12] IN1 – 8 only: "Microphone Sensitivity" to adjust the microphone
preamplifier

7.6.5 Sound adjustment

To adjust the sound, each input channel is provided with a high-pass
filter, a low-pass filter and five parametric equalizers . High-pass filters
can be used in inputs, for example as impact sound filters to suppress
low-frequency noise . In output channels, they can be used as cross-
over networks, also in combination with low-pass filters . Parametric
equalizers can be used, for example, to match the sound to the room
acoustics .
[13] curve representation of the amplitude frequency response
Each of the five parametric filters has a reference point (small
square containing the filter number) and a frequency curve of a
separate colour . A red curve shows the resulting frequency re-
sponse of all active filters . The reference points of the high-pass
filter (HPF) and the low-pass filter (LPF) are circles .
The filter parameters can partly be adjusted graphically: Use the
mouse to shift the reference point:
– change of frequency by horizontal shifting
– change of gain by vertical shifting
– change of quality factor/bandwidth by horizontal shifting of
one of the two lateral squares which are displayed when the
mouse pointer is moved over the reference point
[14] "Freq": cut-off frequency for the high-pass filter
[15] "Type": type of high-pass filter
Indication
Bypass
BW...
BES...
LK...
The number (6 /12 /18 / 24 / 30/36/42 /48) after the filter type indi-
cates the slope in dB / octave .
[16] "Freq": cut-off frequency for the low-pass filter
[17] "Type": type of low-pass filter (description
[18] "Frequency": frequency for the parametric equalizers EQ1 – 5
[19] "Qfact": quality factor for the parametric equalizers EQ1 – 5
Note: It depends on the filter type selected if this parameter is available .
[20] "Gain": gain/attenuation for the parametric equalizers EQ1 – 5
[21] "Type": filter type for the parametric equalizers EQ1–5
Peak: bell filter
Low: low shelving filter
High: high shelving filter
[22] "Bypass" to deactivate the individual parametric equalizers
EQ1 – 5
[23] "Flat" function to reset all filter parameters
[24] "Bypass" to deactivate all filters
Filter type
—
Butterworth
Bessel
Linkwitz-Riley
☞
[15])

7.6.6 Dynamic compressor

A compressor is used to reduce the dynamics and to attenuate the
signal level above an adjustable level threshold . This is necessary when
the dynamics of the audio signal is higher than permitted by the am-
plifier system or by the listening situation (e . g . in case of background
music) . It is also possible to reduce differences in level (for example
with varying microphone distances) or to attenuate signal peaks in
order to obtain a higher gain setting and thus a higher average volume .
[25] graphic presentation of compressor settings
(horizontal axis = input level, vertical axis = output level)
[26] indication of signal attenuation caused by the compressor
"G . R ." = gain reduction
[27] "Gain" to compensate volume loss caused by compression
[28] "Threshold" to define the level from which the gain will be at-
tenuated
[29] "Attack" to define the time it takes the compressor to attenuate
the gain once the sound level has exceeded the threshold level
[30] "Ratio": compression ratio
A compression ratio of 10 :1, for example, means that, above the
threshold, the output level will only increase by 2 dB when the
input level increases by 20 dB .
When "Limit" is selected, the compression ratio is ∞ :1 and the
signal level is limited to the threshold defined .
[31] "Release" to define the time it takes the compressor to let the
gain return to its original value once the sound level has fallen
below the threshold level
[32] "Bypass" to deactivate the compressor
[33] "Flat Comp" to reset all compressor parameters

7.6.7 Delay

Each output signal and input signal can be delayed individually,
e . g . to compensate varying distances of microphones or speakers .
[34] "Bypass" to deactivate the delay
[35] delay time in milliseconds
[36] distance covered by the sound, as an alternative input option
to [35]: The delay time is automatically calculated on this basis
(speed of sound: 343 m / s) .

7.6.8 Mixer

The lower section of the view can also be found in other views . It shows
a mixer with volume controls for the inputs and outputs .
[37] name of input channel; the name of the channel currently indi-
cated is highlighted in green
To go to the settings of another input, click the name field of
the input . To rename the input, double-click the field (
the end of this chapter) .
[38] "Mute" function to mute the input channel; when the button is
red, the channel has been muted
[39] control / input field to set the gain for the input channel
[40] VU meter for the input channel
[41] switchover option to the other input channels
[42] name of output channel
To go to the settings of an output, click the name field of this out-
put (view "Output DSP Channel",
output, double-click the field (
☞
box at
☞ chapter 7 .8) . To rename the
☞ box at the end of this chapter) .
37