Analog Vs. Digital Consoles

ADVANCED RECORDING PRIMER

BY ROBERT DENNIS

ANALOG VS. DIGITAL CONSOLES

The Front End

The "front end" of the console refers to the microphone preamplifiers of the console. The large studio console front-ends have significantly less sound coloration and extremely lower noise when they are operated without overload. Small format consoles often have too much noise and will restrict full fidelity at high output levels - most noticeable restrict fidelity at the lower-bass frequencies.

When rating microphone preamplifiers, how the unit sounds when it is overloaded is part of the rating. You want the extra harmonics generated as overload artifacts to be pleasant to the ear. Often you want the preamplifier to slightly compress the output near or at the overload point. Both of these types of characteristics are most noticeably found in "tube" preamp circuits and in many "Class-A" solid-state designs. As a result the "lunch Box" of 4 to 8 channels of outboard microphone preamplifiers are often found in use at professional sessions.

There is no inherent difference in the microphone preamplifiers found in digital or analog consoles, but there is usually a big difference between the preamplifiers found in small and large-format consoles. The preamplifiers in large-format consoles are usually of the highest quality.

Signal Flow and Algorithms

In the digital console the output of the microphone preamplifier is sent to an A/D converter that changes the signal into a digital audio signal. All mixing, level changes and signal processing is done by the central processor, the "brain" of the console. The brain will simply change the number values of the digital words in order to affect the changes needed. An algorithm will tell to processor which change to make first, which second, etc. Since consoles usually use ultra-low error 32-bit processing, the number of changes made to the signal does not, for all practical purposes, diminish the audio quality of the signal.

In an analog console changes are accomplished by running the signal though device that will affect the changes. If the signal is to be reduced, it is run through a fader; if the signal is to be equalized, it is run though and equalizer; if the signal is to be compressed, it is run through a dynamics processor; if the signal is to be mixed with other signals, it is routed to a buss. The order of the processing and changes is determined which device is first, which second, which third, etc. in the signal flow.

In analog consoles, amplifiers are used to boost the signal for any loss incurred in any of the signal processing or mixing functions. There will be an amplifier associated with the console channel, one for the equalizer, one for the dynamics processor, one for the mixing buss and possibly one for the fader of the channel. The recording engineer is concerned with a thing called "level-structure" where the signal is kept at the correct level in each stage of the signal's travels through the console to prevents adding unacceptable levels of noise or distortion to the signal.

Analog Console Types

The Original "Split" Console

Originally Consoles were built with discrete sections of controls. Each section had a different function and was physically located in a different place. Such a console was eventually called the "Split Console." The common sections are:

1. The Input Section refers to the controls that determine which signals will be accepted by the main console channels. The input section controls adjust the signals’ tone and level and send the signals out of the console alone or mixed together with other signals.

2. The Output Section refers to the set of controls on the console that set levels just before signals go out of the console. There may also be switches or controls that determine which signals go to which output.

3. The Monitor Section refers to the set of controls that adjust what signals the console operator (recording engineer) hears and at what volume. Making adjustments in this section does not affect what is being recorded.

4. The Cue Section refers to the controls on the console which send signals to the musicians/performers through headphones. On live sound mixing consoles this section is often referred to as the "Foldback Section" or "Stage Monitor Section".

5. The Echo Section refers to the controls that adjust the total amount of signal being sent and received back from an echo or reverberation chamber.

The "In-Line" Console

In the 1970’s a new console type called the "in-line" console gained popularity. There are no longer discrete "sections" of controls in different physical locations. The main component of the in-line console is the "input-output" module. Each input-output module has two distinct and separate channels running through it. One channel is the "main" channel" and the second channel is the "monitor channel" Each module also usually has controls for the recording buss level (an "output section" control) and sends for reverberation and cue levels. The result was that this one module has controls for all sections of the console.

The idea behind the in-line console was to put all of the controls (both monitor & input) in front of the engineer. Controls that were used often were placed close to the engineer and controls used left often were placed away from the engineer (usually at the top of the module).

The input-output module is distinct in appearance because there are usually two faders on the module. The large fader toward the bottom of the console is designed as the main channel fader. A smaller fader is placed above the large fader and is designed as the level control for the monitor channel.

The functions of the channels and the operation of controls are pretty much the same for an in-line console - it is just that the physical placement of controls are different. The engineer still sets input "section" controls, output "section" controls and monitor "section" controls in the same manner as the split console.

In the original split console and in the in-line console, the "main" channels would be used for all signal adjustments during recording and mixing. These main channels would be set to "mic input" during recording and the output of the channels would either be assigned to a direct out or a recording buss to get the signals to the multitrack tape machine. During mixdown, the main channels would be set to accept "line" inputs from the tape machine and be assigned to the main "stereo outputs" to send the mix to the 2 track master recorder.

Two Fader Modules

The inline console has two separate audio channels contained in one strip. Each "strip" is called a module because it unbolts and can be removed as one unit. Each the two module channels have a fader. During Recording, one channel is designated in the "monitor channel" that is used to listen to the track signals coming back from the multitrack tape recorder. On the SLL 4000 series console, as with many consoles of this class, the monitor channel fader is shorter than the main channel's fader.

The large, main channel faders contain an amplifier that is "voltage-controlled." Moving the position of the fader changes the amount of a control voltage sent to the amplifier. As the control voltage fed to the amplifier increases the amplifier will in turn increase the level of signal out of it. This type of fader (Called a VCA fader) allows some other part of the console to determine the channel gain. One example of this is when a fader is assigned to a "group" the group master determines the maximum control voltage sent to the individual fader. When a fader is assigned to a group, the group master and the individual fader's position control the channel gain. There is a small switch on each fader that allows it to be assigned to a VCA group.

Figure 1-1a shows a regular audio fader where moving the fader up and down directly controls the audio signal strength in the channel. In the drawing, the fader is at the "-6 db" point so only 50% of the input audio voltage is at the output of the fader. This kind of fader is also referred to as a "manual" fader.

Figure 1-1b shows a VCA fader where the fader's slide is fed a 10-volt control signal. The audio signal from the channel is fed through a voltage-controlled amplifier (VCA). The control voltage out of the slide is fed into a separate control input of the amplifier that is in the fader's case. Since the control voltage into the amplifier determines its gain, moving the slide up and down controls the channel gain. In the drawing, the fader is at the "-6 db" point so only 50% of the input control voltage is at the output of the fader (5 volts).


Figure 1-1a	Figure 1-1b

Figure 1-1c shows what happens when two faders are assigned to a group master fader. The group master fader's slide feeds both of the slides of the individual faders. Since the group master fader is at "-6dB", only five volts of control voltage are fed to each of the individual VCA faders. The slide of the fader for channel 1 is again at "-6dB" so 50% of the reduced of the control voltage (only 2.5 volts) is feed to the amplifier for channel #1. The slide for channel fader #2 is at "-20dB" so, therefore, its amplifier receives 0.5 volts (10% of 5 volts). When individual channel VCA faders are assigned to a group, both the group slide and the individual slide determine the channel's gain.

Figure 1-1c

The Shortest Path

Shortest Path: A technique in recording that routes the signal through the least amount of active (amplified) devices during recording.; Further Information: Each time the signal travels through an amplifier, there is an opportunity for an increase in noise and distortion. Many experienced engineers maintain that although there may not be a noticeable difference in fidelity caused by sending a signal through one device, there is a very noticeable difference in recording quality when several amplifiers are removed from the signal path used for recording; thus the adoption of the policy of always using the shortest path will, in general, increase fidelity.

The type of noise and distortion introduced by amplifiers include the following:

Noise

The input of an amplifier is, for all practical purposes, the place that noise is generated. If the signal is too low at the input of any amplifier, noise will be added to the signal. This is the reason that the recording engineer is concerned with level structure in analog consoles.

Harmonic Distortion

Severe harmonic distortion is caused by "clipping" of an amplifier. But amplifiers, operated near full output, also tend to alter the waveform to a smaller degree. Harmonic distortion of signals below clipping is kept extremely low in the higher-class microphone preamplifiers of the large-format console.

Phase Distortion

Phase distortion is the delaying of one set of frequencies more than another set of frequencies. This delay causes the transients (high bursts of energy caused by the attack of the sound) to come though at a different time than the body of the sound. Phase distortion causes the harmonic content of the waveform to be out-of time with the fundamental frequencies. Although these time errors are slight, they do adversely affect the sound. Descriptions used to describe the audio with phase distortion include "recorded" (as opposed to "live"), "less-real," "false," and "two-dimensional" (as opposed to "three-dimensional"). The transients often become rounded and the resulting sound can be described as "muffled," "blurred," "fuzzy" or "dull."

Phase distortion is most severe in equalization circuits. An equalizer causes the gain of the circuit to be different for one set of frequencies compared to other frequency signals. In analog circuits, equalization is accomplished by the use of reactive elements that oppose one set of frequencies differently than another set of frequencies. A by-product of these reactive elements is to delay one set of frequencies more than another set of frequencies, causing the phase distortion shown in figure 1.2 below


Figure 1-2A - Fundamental with Harmonic in Time	Figure 1-2B - Fundamental with Harmonic Delayed (Phase Distortion)

Figure 1-2 - Phase Distortion

The Shortest Path

As we have mentioned earlier the channel strip of an analog console may contain many amplifiers. Avoiding just one of these amplifiers during recording can eliminate 24 amplifiers used in the recording, because we are using up to 24 channel strips in the multitrack recording process. Although eliminating only one amplifier does not make a noticeable improvement in the audio quality, avoiding 24 would definitely make a difference.

Avoiding extra amplifiers associated with recording busses is particularly important because the mixing resistors associated with the buss may bring down the signal as much as 40 dB only to be boost the signal back up with a boost amplifier. This extreme changes of gain makers the addition of noise and distortion much more likely.

Attaining the Shortest Path

The shortest path is attainable by paying attention to signal flow through the console. Some rules to follow are as follows:

Keep Unnecessary Circuits Switched "Out"

A channel strip in a console has an "in" button for signal processing devices like the equalizer or dynamic processor. It is usual for the signal to totally by-pass the amplifier associated with the signal processing when the processor is switched to the "out" position. Having the devices "in" button activated when the device is not being used will, therefore, increase the number of amplifiers that the signal goes through in the channel strip.

Patching Around Busses

When recording one microphone to one track, you should not be using a recording buss. The buss is designed to mix two or more signals together. Consoles have a "direct out" system for getting one microphone to one track - usually the track that has the same number as the console channel. Using this "direct out" function avoids the boost amplifier that is associated with the recording buss.

When faced with the situation that one microphone signal is to get to one track, but to a track with a different number, the buss could be used to route the signal. Sending one signal on a buss, however, is about as sensible as driving a 52 passenger vehicle to get one person to work. There will be a patch point on the console that will allow the channel signal to be patched directly into the multitrack tape recorder track input. Using this patch point greatly helps keep the signal path shorter.

When outboard microphone preamplifiers are used, it is common for their outputs to be patched directly to the tape recorder inputs to avoid unnecessary signal patch length.

Flip Status

We are using the SSL 4000 series console as an example of Flip Status. The flip status of the SSL is called VCA to Monitor Status.

The main channels of the SSL 4000 series console have faders with a Voltage Controlled Amplifier (VCA) in the fader case. The VCA is used for computer control of channel gain during automated mixdown and for controlling the gain of groups of channels (also during mixdown). Thus the large faders on the SSL are referred to as VCA Faders. The small faders (monitor channel faders) on the SSL 4000 do not contain VCA’s.

The SSL console has a "status" button called "VCA to Monitors." This status switches the function of the two faders. The large VCA faders are placed in the monitor channel and the small faders are placed in the main channels. There are three very good reasons that engineers use this "VCA to Monitor" or "Flip" status during recording:

1. If the small faders are used during recording, each main channel has one less amplifier. Thus the signals being recorded on the multitrack tape go through less amplifiers. Since each amplifier causes some noise and distortion, the signal that is recorded has less noise and distortion. This technique of using the smallest number of amplifiers possible before the tape recorder is referred to as using the "Shortest Path."

2. During recording the main channel fader us usually set close to 75% up and not moved during recording. Thus a large fader is not needed. The monitor faders may, however, need to be moved during recording, especially during the final overdubs of the reproduction. Thus the large faders would work "better" as monitor faders. The VCA to Monitor status allows the engineer to use the best fader for the job.

3. After recording comes mixing. When changing from Record status (with VCA to Monitor activated) to Mix status, the large faders are in position for starting the mixdown. The levels you used for the monitor mix will already be established to start the mixdown.

The Shortest Path In Digital Consoles

By their very nature of having the "brain" of the console perform all changes to the audio, the digital console inherently has only one path length. Adding additional processing devices and using recording busses does not increase the path length and does not add to noise or distortion because additional amplifiers are not needed to affect these changes.

About the only way the patch length is increased is by use of analog "inserts" in the console channels. To have an insert that allows the signal to be routed through an analog outboard unit, the signal has to be converted back to analog to route to the in the "insert send" jack and needs to be converted back to digital when the signal is received by via the "insert return" jack. When operating a digital console, the recording engineer should not activate any channel insert jacks unless they are needed.

Phase Distortion Control and Delay In Analog & Digital Consoles

Because digital consoles apply signal processing by computer alteration of the digital words that make up the digital audio signal, there are no "reactive elements" necessary to achieve such things as equalization, and the highs, mids and lows do not get out of time with each other. There is, therefore, no phase distortion.

On the other hand digital consoles have to "process" the signal before spitting out the audio. If the signal has to be converted back to analog at the output, this has to be processed. As a result, there is a very slight delay of the signal by passing though a digital console (less than 1 millisecond). This slight delay does not cause phase cancellation because it is an even delay at all frequencies. If however, you were to mix the input signal of a digital console with the output of that console, you would get some phase cancellation. As a result, this type of situation is carefully avoided.

Persons who have only used analog equalization often first complain that the digital equalizers don't "sound" the same as analog equalizers. Hobbyists who only have used digital equalizers have similar complaints about analog EQ. This is due to the lack of phase cancellation in digital equalization. The professional engineer is used to both and knows and can work with the differences.

The phase distortion of analog console is considerably less at lower settings of gain change and at lower Q values (wider bandwidths). Try to restrict the amount of boost or cut and work with the lowest workable Q value to reduce the amount of phase distortion.

GO TO CHAPTER 1 - PART 2, "SSL: 400G INPUT/OUTPUT STRIP"

RETURN TO ADVANCED RECORDING PRIMER CONTENTS