Lectrosonics DC1 User Manual

DC1

AUTOMATIC PRE-PROCESSOR

WITH ADAPTIVE PROPORTIONAL GAIN ALGORITHM*

(Version 2.0 and higher)

OPERATING INSTRUCTIONS

* US Patent Pending

LECTROSONICS, INC.

Rio Rancho, NM

INTRODUCTION

The DC1 is a sophisticated, microprocessor-based audio signal processor which will add the capability of automatic
mixing to any standard mixing console. The DC1 provides the capability to automate up to 4 standard channels.
Since more than one DC1 may be used together, any number of channels may be automated. The DC1 is
intended for use in any system that already has a standard mixing console, but where automation on some
channels is helpful. Examples might be TV and radio broadcast studios, large churches, etc.

The DC1 uses Lectrosonics’ unique "Adaptive Proportional Gain" mixing algorithm rather than hard switching to turn
channels off and on. This results in totally inaudible automatic action. Complete control over all operational
parameters is afforded by the DC1. The actual automatic mixing algorithm is implemented in software, and as a
result much greater control and flexibility may be exercised than would normally be the case in a purely analog
system.

TABLE OF CONTENTS

INTRODUCTION .......................................... 1

THEORY OF OPERATION ................................... 2

INSTALLATION ........................................... 4

FRONT PANEL DESCRIPTION ................................ 5

REAR PANEL DESCRIPTION ................................. 6

OPERATING INSTRUCTIONS ................................. 7

SETUP PROCEDURE ....................................... 8

AUTOSET DESCRIPTION ..................................... 10

AUTOSET PROCEDURE ..................................... 11

DC1 Function Summary and Default Settings ...................... 12

SPECIFICATIONS ......................................... 13

SERVICE AND REPAIR ..................................... 14

RETURNING UNITS FOR REPAIR ............................. 14

WARRANTY ........................................ Back cover

Note: This manual contains instructions which are specific to version 1.4 of the firmware.

THEORY OF OPERATION

RFI

FILTER

FROM CHANNEL 2
FROM CHANNEL 3
FROM CHANNEL 4

FROM CHANNEL 4

FROM CHANNEL 3

FROM CHANNEL 2

TO CHANNEL 2

TO CHANNEL 3

TO CHANNEL 4

FRONT PANEL LEDS, LCD DISPLAY,

FROM CHANNEL 2
FROM CHANNEL 3
FROM CHANNEL 4

TO CHANNEL 4

TO CHANNEL 3

TO CHANNEL 2

SLAVE

MASTER

NC
NC

NC

NC

EXPANSION OUT

EXPANSION IN

CHANNEL 1

(CHANNELS 2-4 IDENTICAL)

BALANCED

INPUT

BALANCED

OUTPUT

LOGIC

OUTPUT

CHANNEL

ON

LOGIC OUT

PROCESSING

AND

OPTO-COUPLER

ATTENUATION

DISPLAY

HEADROOM
DISPLAY

PHANTOM

POWER

PROGRAMMABLE

MICROPHONE

PREAMP

VOLTAGE

CONTROLLED

AMPLIFIER (VCA)

RELAY

BYPASS

PROGRAMMABLE

ATTENUATOR

SPEECH

FILTER

LOG

CONVERTER

PRE-VCA

OUTPUT

POST-VCA

OUTPUT

11 CHANNEL

A/D

CONVERTER

8 CHANNEL

D/A

CONVERTER

NON-VOLATILE

MEMORY

uPROCESSOR

MC68HC705

PRE-VCA OUT
POST-VCA OUT
NOM OUT
VAR THRESHOLD
NOM TOTAL
EXT PRIORITY

GND

PRE-VCA IN
POST-VCA IN
NOM IN
VAR THRESHOLD
NOM TOTAL
EXT PRIORITY

GND

MEMORY
PRESET 1

MEMORY
PRESET 2

MEMORY
PRESET 3

AND PUSHBUTTONS

Please refer to the block diagram of the DC1 for the following discussion.

The DC1 has two major subsystems; the analog signal processing and the digital control section.

The audio signal processing includes an ultra-low noise microphone preamp, a high-quality Voltage Controlled
Amplifier (VCA), and an output transformer to prevent ground loops and RF interference. In addition, speech filters
and precision log amplifiers detect audio signals over a wide dynamic range. The headroom detection circuitry
monitors internal signal levels and gives a front panel indication of the amount of signal headroom left. The
attenuation detection circuitry monitors the control voltage to the VCA and gives a front panel indication of the
instantaneous gain reduction in the channel.

The heart of the digital control section consists of a Motorola 68HC705 micro-controller. The micro-controller is
interfaced to the LCD display and front panel buttons, internal analog-to-digital and digital-to-analog converters, and
relay drivers. The automatic mixing algorithm is completely software controlled, which provides a level of
performance difficult to obtain with purely analog systems.

The DC1 is a mic level in, mic level out device. The microphone preamp in each channel may be set (via front
panel control) to 40dB (Input Type: High) or 60dB (Input Type: Low) to accommodate all types of microphones. 48
volt phantom power is also available (via front panel control) on a per channel basis. After the signal is amplified, it
passes through the VCA. From there, the signal is buffered and transformer coupled to a passive attenuation
network, which brings the signal back down to mic level.

Figure 1 - DC1 Block Diagram

2

THEORY OF OPERATION (cont’d)

The DC1 audio signal path is designed to eliminate any signal degradation from the DC1 itself. State of the art
microphone preamplifier and voltage controlled amplifier (VCA) ICs are used to process the audio signals. When
the Bypass mode of operation is selected from the front panel, all VCAs are set to unity gain and microphone signal
pass through unaffected. A hardware bypass capability is provided by relays which switch all three pins of the input
and output XLR connectors. If power to the DC1 is lost for any reason, the relays will automatically drop into the
hardware bypass mode, giving fail safe operation. A setup mode exists on the DC1 where the actual input levels of
the microphones are displayed in real time so that differences in microphone sensitivity may be determined.

The automatic mixing algorithm used by the DC1 is called "Adaptive Proportional Gain" (patent pending). The
channel sense level (consisting of a combination of Pre-VCA and Post-VCA signals) from each channel is
continuously compared to the system sense level (the sum of all channel sense levels). Each channel is then
attenuated based on the difference, in dB, between its sense level and the overall sense level. System gain is
automatically shifted toward the microphone(s) with the greatest signal level. As a result, no "threshold" is
necessary to determine when to turn a channel on or off, and the gain allocation is not affected by changes in
ambient noise in the room.

In addition to preamp gain and phantom power, there are a number of other operational parameters that may be
adjusted on the DC1 to optimize the performance for any application. Each channel has two modes of operation,
Auto and Direct. Auto mode sets the channel for automatic operation. Direct turns the channel on under all
circumstances.

Level Match allows the apparent channel signal level to be increased or decreased by up to 30dB, in 1dB
increments. The actual audio signal path gain is unaffected by changes in the Level Match value. The Level Match
function is used to minimize variations in sensitivity (and as a result, access to the system) which might result from
the use of microphones with different sensitivities in the system. In addition, the Level Match function may be used
to give a soft talker access to the system equivalent to louder talkers.

The off attenuation of each channel is also adjustable, from 6dB to 20dB, using the Max Attenuation function. This
allows inactive channels to be attenuated to a greater or lesser degree, depending on the application.

The Auto Gain Skew function provides the capability to minimize the interruption of the active microphone by non­speech signals such as coughs, paper rattling, etc. Auto Gain Skew dynamically changes the relative proportion of
Pre-VCA to Post-VCA signals which comprise the sense level for a channel. In this way, channels which are active
for a significant amount of time will tend to dominate inactive microphones. Auto Gain Skew also helps eliminate
"bleed-over" of a talker into adjacent closely spaced microphones.

An AutoSet function is included to simplify the process of setting the important parameters of the DC1. The
AutoSet function allows different types of microphones (typically with different sensitivities) to be used with the DC1.
AutoSet automatically calculates the sensitivity differences in the microphones and sets internal parameters for
optimum operation.

3

INSTALLATION

"Master"

"Slave"

Master DC1 Slave DC1 Slave DC1

OUT IN OUT IN OUT IN

Installation of the DC1 is quite straightforward. To use one DC1 by itself, simply plug the microphone cables from
up to four microphones into the XLR inputs of the DC1. Connect the XLR outputs of the DC1 to the XLR inputs of
the mixing console.

If more than one DC1 is to be used, all but one must be set to the slave mode. Slave mode strapping is
accomplished by removing the top cover of the DC1 and moving the two jumpers in the upper right hand corner of
the DC1 from the forward position (i.e. toward the front panel) to the rear position (i.e. toward the rear panel). In
addition, the DC1’s should be connected using 9 pin subminiature D type cables as shown in Figure 3.

Figure 2 - DC1 Jumper Location and Settings

Figure 3 - Master/Slave Connections

4