So-called “digital microphones” have entered the market place during the last six or so years.
Strictly speaking, these models are not actually “digital” in the specific sense of directly generat-ing a digital output code from the diaphragm. Rather, they make use of traditional DC bias and preamplification of the analog signal at the diaphragm. It is only after this stage that analog-to-digital conversion takes place.
The advantage of these microphones is that certain problems in digital processing can be dealt with earlier, rather than later, in the audio chain. For example, the useful signal-to-noise ratio of a well-designed 25 mm (1 in) condenser diaphragm can be in the range of about 125 to 135 dB.
An ideal 20-bit system is capable of a signal-to-noise range of 120 dB, and in a traditional record-ing system this will require truncation of the available dynamic range of the microphone by about 10 dB. In and of itself, this may or may not be a problem, depending on other electrical and acoustical considerations in the actual studio environment.
In the beyerdynamic model MCD100 series, the capsule looks into a 22-bit conversion system directly when the acoustical level is high (greater than 124 dB LP). For normal studio levels (less than about 100 dB LP), –10 or –20 dB padding can be inserted ahead of the digital conversion stage in order to optimize the bit depth. Sophisticated level control prevents the system from
(b)
Details of beyerdynamic digital microphone system: view of microphone (a); signal flow diagram (b). (Data courtesy of beyerdynamic.) fIgure 4.27
A circuit for paralleling the output of two microphones. (Data after Shure Inc.)
97 going into digital clipping. The microphone and associated signal flow diagram is shown in Figure
4.28(a) and (b).
The Neumann Solution-D uses two 24-bit A-to-D converters operating in parallel and offset by 24 dB.
These two digital signals are seamlessly recombined in the digital domain to produce a single digital output signal with a net resolution of 28 bits (Monforte, 2001). Figure 4.29(a) shows a view of the Solution-D microphone, and a signal flow diagram is shown in Figure 4.29(b).
Both of these microphone systems have additional digital features, including variable sampling rates, various interface formats, some degree of built-in digital signal processing, and the ability to respond to certain user commands via the digital bus. The Audio Engineering Society (AES) is actively pursuing interface standards for this new class of products.
Non-linear network
High level
ADC gate
DSP +
Low level 24dB Gain ADC
16x Mic capsule
(b)
–
+
fIgure 4.29
Details of Neumann Solution-D digital microphone system: view of microphone (a); signal flow diagram (b). (Data courtesy of Neumann/ USA.)
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references
Hibbing, M., 1994. High Quality RF Condenser Microphones, Chapter 4 in Microphone Engineering Handbook, ed. M. Gayford. London: Focal Press.
Monforte, J., 2001. Neumann Solution-D Microphone, Mix Magazine.
Perkins, C., 1994. Microphone Preamplifiers – A Primer, Sound & Video Contractor 12, no.2.
The third section of this book “Preamplifiers, Mixers, & Interconnects” gets to grips with the part that small-signal electronics plays in sound systems.
In the first chapter, Francis Rumsey and Tim McCormick give a thorough and comprehensive account of mixing console principles and recording methods, diving in at once with a description of a simple six-channel mixer that has on each channel microphone and line inputs, an input gain control, bass and treble EQ controls, a pan control, a PFL button and a slide fader with 10 dB at the top. Its master section has output faders, two output level meters, and a monitor volume control. This is representa-tive of millions of little mixers, and they can be seen in the window of most music shops.
Each feature of this mixer is carefully examined, with special sections on the faders, the pan-pots with their special control laws, and the PFL (Pre-Fade Listen) system.
From here we move on to the needs of multitrack recording, for which a more sophisticated mixing console is required, because of the need to play back existing tracks at the same time as recording new ones, so the new material can be laid down in synchrony. Twice the number of signal paths are there-fore needed; one path from microphone to recorder, and one return path from recorder to the monitor system.
These two paths can be arranged in two different ways, which brings us to the next section of this chapter, the distinction between split and in-line console configurations. The split console is the more obvious of the two; it puts the input channels on one side (normally the left), a master control section in the middle, and the replay monitor mixer on the right. In many ways there are two consoles in one chassis. Since there must be as many replay monitor channels as there are recorder tracks, this section takes up a lot of room, but it has the great advantage that it is conceptually simple.
The in-line configuration is quite different. Now the replay monitor sections are integrated with the input channels, so they can share resources such as EQ and auxiliary sends. This tends to give a console which is a little deeper, due to the increased channel length, but much less wide. The only real draw-back is that it is less intuitive and takes a bit of learning.
The next section deals with channel grouping, and is illustrated by a picture of the Soundcraft Sapphyre (yep, one of mine). Both sorts of grouping—mixing several channels to a group and controlling that, and VCA group control—are covered.
Rumsey and McCormick then move on to examine more closely the features of a sophisticated recording console, such as EQ, dynamics sections, monitor controls and effects returns. The chapter concludes by looking at basic operational techniques, technical specifications, metering, and console automation.
Some readers may be aware that I spent a good number of years engaged in mixing console design, as Chief Engineer at Soundcraft Electronics, and I think this chapter is the best concise description of mixer operation that I have come across.
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Audio Engineering Explained
Copyright © 2009 by Elsevier Inc. All rights of reproduction in any form reserved.
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2010
Mixers
Sound and Recording by Francis Rumsey and Tim McCormick
This chapter describes the principles and basic operation of audio mixers. It begins with a description of a simple system and moves on to consider the facilities of large-scale multitrack systems. Because many design and layout concepts of analog mixers find a place in more recent digital mixers, these aspects are covered here in a fairly generic way. Those features found more commonly only in digital systems are described towards the end of the chapter.
In its simplest form an audio mixer combines several incoming signals into a single output signal.
This cannot be achieved simply by connecting all the incoming signals in parallel and then feeding them into a single input because they may influence each other. The signals need to be isolated from each other. Individual control of at least the level of each signal is also required.
In practice, mixers do more than simply mix. They can provide phantom power for capacitor micro-phones; pan control (whereby each signal can be placed in any desired position in a stereo image);
filtering and equalization; routing facilities; and monitoring facilities, whereby one of a number of sources can be routed to loudspeakers for listening, often without affecting the mixer’s main output.
A SiMPlE Six-ChAnnEl AnAlog MixER overview
By way of example, a simple six-channel analog mixer will be considered, having six inputs and two outputs (for stereo). Figure 5.1 illustrates such a notional six-into-two mixer with basic facilities.
It also illustrates the back panel. The inputs illustrated are via XLR-type three-pin latching connectors, and are of a balanced configuration. Separate inputs are provided for microphone and line level signals, although it is possible to encounter systems which simply use one socket switchable to be either mic or line. Many cheap mixers have unbalanced inputs via quarter-inch jack sockets, or even “phono” sockets such as are found on hi-fi amplifiers. Some mixers employ balanced XLR inputs for microphones, but unbalanced jack or phono inputs for line level signals, since the higher-level line signal is less susceptible to noise and interference, and will probably have traveled a shorter distance.
On some larger mixers a relatively small number of multipin connectors are provided, and multicore cables link these to a large jackfield which consists of rows of jack sockets mounted in a rack, each being individually labeled. All inputs and outputs will appear on this jackfield, and patch cords of a meter or so in length with GPO-type jack plugs at each end enable the inputs and outputs to be interfaced with other equipment and tie-lines in any appropriate combination.
(The jackfield is more fully described in “Patchfield or jackfield,” below.)
The outputs are also on three-pin XLR-type connectors. The convention for these audio connections is that inputs have sockets or holes, outputs have pins. This means that the pins of the connectors “point”
in the direction of the signal, and therefore one should never be confused as to which connectors
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are inputs and which are outputs. The microphone inputs also have a switch each for supplying 48 V phantom power to the microphones if required. Sometimes this is found on the input module itself, or sometimes on the power supply, switching 48 V for all the inputs at once.
The term “bus” is frequently used to describe a signal path within the mixer to which a number of sig-nals can be attached and thus combined. For instance, routing some input channels to the “stereo bus”
conveys those channels to the stereo output in the manner of a bus journey in the conventional everyday sense. A bus is therefore a mixing path to which signals can be attached.