Automixer

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Microphones at a press conference being processed through a Dugan E-1 automixer which has been placed on top of the regular audio mixer. San Francisco mayor Gavin Newsom is speaking at a lectern, while golfers Fred Couples and Greg Norman are seated on stage. Six of eight automixer inputs have been muted and are showing red LEDs. The active input is showing full gain with a ladder of green LEDs. DuganE-1 Newsom,Couples,Norman.jpg
Microphones at a press conference being processed through a Dugan E-1 automixer which has been placed on top of the regular audio mixer. San Francisco mayor Gavin Newsom is speaking at a lectern, while golfers Fred Couples and Greg Norman are seated on stage. Six of eight automixer inputs have been muted and are showing red LEDs. The active input is showing full gain with a ladder of green LEDs.

An automixer, or automatic microphone mixer, is a live sound mixing device that automatically reduces the strength of a microphone's audio signal when it is not being used.

Contents

Automixers reduce extraneous noise picked up and comb filtering effects when several microphones operate simultaneously. Automixers uses a variety of methods that allow increased gain before feedback for live sound reinforcement. [1]

Automixers are typically used to mix panel discussions on television talk shows and at conferences and seminars. They can also be used to mix actors' wireless microphones in theater productions and musicals. Automixers are frequently employed in settings where it is expected that a live sound operator won't be present, such as courtrooms and city council chambers.

Use

Automixers balance multiple sound sources based on each source's level, quickly and dramatically adjusting the various signal levels automatically. [2] Automixers are used in live sound reinforcement to maintain a steady limit on the overall signal level of the microphones; if a public address system is set up so that one microphone will not feed back, then, in general, the automixer will prevent feedback when multiple microphones are used. The equivalent number of open mics (NOM) present at the output of the automixer is kept low, regardless of the actual number of mics in use. [3]

While a skilled mixing engineer can greatly enhance the performance of a sound reinforcement system he cannot anticipate with perfect accuracy which participant will speak next in a spontaneous discussion. Sudden interjections by panelists may be lost completely, or the beginning of a word may be absent (or upcut [4] ) because the operator does not respond quickly enough. [5] A properly adjusted automixer can prevent losing words or phrases due to upcut mistakes or lapses of attention. [6]

Priority ducking is used to lower the level of a microphone or program material, based on a signal at a source microphone, for the duration of the signal present at the source microphone. It restores the original level once the source microphone signal has ceased. This is useful when program material needs to be attenuated in order to accentuate the voice of a narrator or if one microphone is used by a chairman or master of ceremonies and needs to have priority over other mics or program material, or if a paging mic must attenuate all other signals.

Automixers can be oriented toward live reinforcement applications or permanent installation. Live applications typically use XLR connectors and external controls. For permanent installations euroblock connectors and tamper-resistant internal controls may be used. Channel configuration switches select sensitivity (dynamic mic, condenser mic, or line level), enable or disable phantom power, set priority, and select always-on or auto mode (always-on being useful for uninterrupted music playback). Auto mixers are frequently used in boardrooms but are also useful in theatrical productions and town hall meetings where several wireless microphones are in use. Auto mixers may be connected directly to wired mics and wireless mic receivers or may be connected to the insert of a mixing console. Integration of an auto mixer with manual mixing console permits more precise input gain control, absolute mic muting, low cut filter and other equalization and individual mic channel foldback control.

History

Frank J. Clement and Bell Labs received a patent in 1969 for a multiple-station conference telephone system that switched its output to the loudest input. [7] The next year, Emil Torick and Richard G. Allen were granted a patent for an "Automatic Gain Control System with Noise Variable Threshold", an adaptive threshold circuit invention with its patent assignation going to Columbia Broadcasting System. [8]

Some systems using electro-mechanical switching for microphone activation were engineered in the late 1960s and early 1970s. Peter W. Tappan and Robert F. Ancha devised a system of seat sensors that would activate one of 350 hidden microphones at the Seventeenth Church of Christ, Scientist in Chicago in 1970. [9] From approximately 1968, Ken Patterson and Diversified Concepts developed a hardware system that could detect the number of open microphones (NOM) and attenuate the master output by an amount which increased with a higher number of microphones in use. This latter system was public domain. [10]

In 1971, Gregory Maston of Bell Labs filed for a patent involving a circuit that could switch between several audio sources based on their levels. The loudest one was latched into the mix. This system did not ramp switched signals smoothly in and out and did not maintain a constant ambient noise level. It was intended for speakerphone conferencing applications. [11] In 1972, Keith A. T. Knox with the British Post Office Corporation developed an adaptive threshold gate circuit intended for speakerphone usage. The system used a second microphone somewhat near the first to sense ambient noise level. [12]

Dan Dugan's first automixers Dugan automixers - Dugan Model A8 and A6, and prototype.jpg
Dan Dugan's first automixers

Dan Dugan showed his first Adaptive Threshold Automatic Microphone Mixing System in 1974 at the 49th Audio Engineering Society (AES) meeting in New York, [13] and was granted a patent for a control apparatus for sound reinforcement systems which sensed ambient sound level in the environment of a theater to control each microphone's level. [14] In 1976, Dugan was granted a patent for an automatic microphone mixing process whereby the total gain of the system remains constant. [15] He began manufacturing his first automixer system, the Model A, based on his two patents. Dugan built 60 units, with the first, hand-assembled one taken to Bell Labs to be installed in their conference room for Harvey Fletcher. [16] The algorithm was simple and effective: "Each individual input channel is attenuated by an amount, in dB, equal to the difference, in dB, between that channel's level and the sum of all channel levels." [17] Dugan licensed the system to Altec which released several automixer models including the 1674A, -B and -C series and the 1684A, all focusing on speech applications. [18] [19] The earliest Altec product implementation was regarded as inferior within the commercial audio contractor industry, [1] and other manufacturers began to design their own automixer products.

In 1978, Richard W. Peters of Industrial Research Products (IRP) was granted an improvement patent entitled "Priority mixer control". [20] IRP released the Voice-Matic series of 4 × 1 and 8 × 1 automatic mixers using "Dynamic Threshold Sensing" that weighed a combination of the amplitude and history of the signal to determine channel access. The master output was attenuated at the rate of 3 dB for every doubling of NOM. [21] This master output reduction was the solution used by Yamaha Pro Audio two decades later in their DME series of digital signal processing (DSP) products, incorporating an automixer function which was otherwise an 8- or 16-channel noise gate. [22]

Eugene Campbell and Terrance Whittemore of Colorado were granted a patent in 1982 for an automatic microphone mixing algorithm that allowed for musical performance mixing that would not be dominated by the loudest vocalist or instrumentalist. [23]

Graphic user interface for a digital automixer that uses a gain-sharing algorithm. Controls include threshold, depth, polarity inversion and muting for each input, as well as volume controls for the four inputs, the four individual outputs and the full mix output. Automix 4 graphic user interface.png
Graphic user interface for a digital automixer that uses a gain-sharing algorithm. Controls include threshold, depth, polarity inversion and muting for each input, as well as volume controls for the four inputs, the four individual outputs and the full mix output.

Stephen D. Julstrom of Shure Brothers Incorporated was granted a patent in 1987 for a teleconferencing system that used special directionally gated microphones mixed automatically and sent to a distant party via telephone line. The return signal from the distant party was compared in strength to the local mix to determine which party was to be most prominent in the overall mix. Any interrupting party was given priority. [24] Four years later, Shure would introduce the AMS4000 and AMS8000 automixers for sound reinforcement; mixers which required the use of special directional condenser microphones of the Shure AMS Series. [25]

In 1985 Innovative Electronic Designs (IED) introduced the circuit card frame-based auto mic mixing system featuring combine-separate function and programmable gain control (PGC) modules. Combine-separate functionality is useful for ballroom applications with movable partitions that permit portions or all of a room to be used for one program or multiple programs. PGC modules compensate individual mic channel gain for soft-spoken or loud presenters, by slightly boosting or cutting channel gain, respectively. [26]

At the 87th AES Convention in 1989, Dugan introduced the idea of using an automixer inserted within target channels on a professional audio mixer. Each microphone's signal would be interrupted inside its channel strip and sent to a variable gain circuit contained within the automixer. The signal would then be returned to the mixer at a level consistent with the Dugan algorithm. [27] This became the Dugan Model D automixer. [28]

In 1991, Dugan's patent expired. Competing manufacturers began to bring the Dugan algorithm directly to their product designs. In 1993, Travis M. Sims, Jr. of Lectrosonics (Rio Rancho, New Mexico) was granted a patent for a sound system with rate controlled, variable attenuation of microphone inputs, including the Dugan algorithm as well as loudspeaker zone attenuation when in close proximity to an active microphone. [29] The loudspeaker zone part of the patent cited a 1985 patent for proportional amplification by Eugene R. Griffith, Jr. of LVW Systems of Colorado Springs, a commercial audio contractor. [30] In 1995, Sims and Lectrosonics gained another patent for an "Adaptive proportional gain audio mixing system" which incorporated a number of ideas including the Dugan algorithm for maintaining a constant total gain of all the inputs. [31]

In 1996, Dugan came out with the Model D-1, a speech-only economy model that did not offer the music system of the Model D. [16]

In 1997, John H. Roberts of Peavey Electronics was granted a patent for an automatic mixer priority circuit, enabling a hierarchy of logic weighting that allowed selected signals to push forward in the mix when they are in use, while still maintaining the useful constant unity, gain-sharing relationship first described by Dugan. The hierarchy enabled a host, moderator or chairperson to speak over other participants and retain control of the discussion. [32] Peavey's Architectural Acoustics division used three levels of hierarchy in their 1998 "Automix 2" product, placing the first- and second-most influentially weighted sources at inputs 1 and 2, respectively. [33]

Dan Dugan licensed his system to Protech Audio (Indian Lake, New York) in 1997, yielding the Protech 2000 model series. [34]

In 2004, the first standard audio mixer incorporating an eight-channel automixer section was released by Peavey in their Sanctuary Series, [35] and in 2006 the similar HP-W was introduced by Crest. [36] Both mixers were aimed at the House of Worship market, adding functions that ease the process of audio mixing to religious organizations.

In 2007, Mark W. Gilbert and Gregory H. Canfield of Shure (Niles, Illinois) were granted a patent for a digital microphone automixer system that used time of arrival as its main decision-making criteria. [37] [38]

Generations of Dugan's insertable automixers Dugan automixers - D, D2-3, E-1, and Dugan-MY16.jpg
Generations of Dugan's insertable automixers

In February 2011, Dugan announced an automixer card to plug in the accessory slot of a Yamaha digital mixing console such as the LS-9, M7CL or PM-5D. This card, the Dugan-MY16, can mix 16 channels of microphone inputs at 44.1–48 kHz or 8 channels at 88.2–96 kHz sampling rates. Channels to be automixed are assigned in the mixer's graphic user interface, and can then be controlled by a common web browser interface affecting only the Dugan-MY16 card, allowing remote control with an iPad, touchscreen computer or laptop over wireless network. [39]

See also

Related Research Articles

<span class="mw-page-title-main">Microphone</span> Device that converts sound into an electrical signal

A microphone, colloquially called a mic, or mike, is a transducer that converts sound into an electrical signal. Microphones are used in many applications such as telephones, hearing aids, public address systems for concert halls and public events, motion picture production, live and recorded audio engineering, sound recording, two-way radios, megaphones, and radio and television broadcasting. They are also used in computers and other electronic devices, such as mobile phones, for recording sounds, speech recognition, VoIP, and other purposes, such as ultrasonic sensors or knock sensors.

<span class="mw-page-title-main">Audio feedback</span> Howling caused by a circular path in an audio system

Audio feedback is a positive feedback situation that may occur when an acoustic path exists between an audio output and its audio input. In this example, a signal received by the microphone is amplified and passed out of the loudspeaker. The sound from the loudspeaker can then be received by the microphone again, amplified further, and then passed out through the loudspeaker again. The frequency of the resulting howl is determined by resonance frequencies in the microphone, amplifier, and loudspeaker, the acoustics of the room, the directional pick-up and emission patterns of the microphone and loudspeaker, and the distance between them. The principles of audio feedback were first discovered by Danish scientist Søren Absalon Larsen, hence it is also known as the Larsen effect.

<span class="mw-page-title-main">Mixing console</span> Device used for audio mixing

A mixing console or mixing desk is an electronic device for mixing audio signals, used in sound recording and reproduction and sound reinforcement systems. Inputs to the console include microphones, signals from electric or electronic instruments, or recorded sounds. Mixers may control analog or digital signals. The modified signals are summed to produce the combined output signals, which can then be broadcast, amplified through a sound reinforcement system or recorded.

<span class="mw-page-title-main">Dynamic range compression</span> Audio signal processing operation

Dynamic range compression (DRC) or simply compression is an audio signal processing operation that reduces the volume of loud sounds or amplifies quiet sounds, thus reducing or compressing an audio signal's dynamic range. Compression is commonly used in sound recording and reproduction, broadcasting, live sound reinforcement and some instrument amplifiers.

<span class="mw-page-title-main">DI unit</span> Audio signal conversion device

A DI unit is an electronic device typically used in recording studios and in sound reinforcement systems to connect a high output impedance unbalanced output signal to a low-impedance, microphone level, balanced input, usually via an XLR connector and XLR cable. DIs are frequently used to connect an electric guitar or electric bass to a mixing console's microphone input jack. The DI performs level matching, balancing, and either active buffering or passive impedance matching/impedance bridging. DI units are typically metal boxes with input and output jacks and, for more expensive units, “ground lift” and attenuator switches.

<span class="mw-page-title-main">Public address system</span> Electronic system for amplifying sound

A public address system is an electronic system comprising microphones, amplifiers, loudspeakers, and related equipment. It increases the apparent volume (loudness) of a human voice, musical instrument, or other acoustic sound source or recorded sound or music. PA systems are used in any public venue that requires that an announcer, performer, etc. be sufficiently audible at a distance or over a large area. Typical applications include sports stadiums, public transportation vehicles and facilities, and live or recorded music venues and events. A PA system may include multiple microphones or other sound sources, a mixing console to combine and modify multiple sources, and multiple amplifiers and loudspeakers for louder volume or wider distribution.

<span class="mw-page-title-main">Sound reinforcement system</span> Amplified sound system for public events

A sound reinforcement system is the combination of microphones, signal processors, amplifiers, and loudspeakers in enclosures all controlled by a mixing console that makes live or pre-recorded sounds louder and may also distribute those sounds to a larger or more distant audience. In many situations, a sound reinforcement system is also used to enhance or alter the sound of the sources on the stage, typically by using electronic effects, such as reverb, as opposed to simply amplifying the sources unaltered.

<span class="mw-page-title-main">Noise gate</span> Audio processing device

A noise gate or simply gate is an electronic device or software that is used to control the volume of an audio signal. Comparable to a limiter, which attenuates signals above a threshold, such as loud attacks from the start of musical notes, noise gates attenuate signals that register below the threshold. However, noise gates attenuate signals by a fixed amount, known as the range. In its simplest form, a noise gate allows a main signal to pass through only when it is above a set threshold: the gate is "open". If the signal falls below the threshold, no signal is allowed to pass : the gate is "closed". A noise gate is used when the level of the "signal" is above the level of the unwanted "noise". The threshold is set above the level of the "noise", and so when there is no main "signal", the gate is closed.

Shure Inc. is an audio products corporation headquartered in the USA. It was founded by Sidney N. Shure in Chicago, Illinois, in 1925 as a supplier of radio parts kits. The company became a consumer and professional audio-electronics manufacturer of microphones, wireless microphone systems, phonograph cartridges, discussion systems, mixers, and digital signal processing. The company also manufactures listening products, including headphones, high-end earphones, and personal monitor systems.

<span class="mw-page-title-main">Yamaha M7CL</span> Mixing console

The Yamaha M7CL is a digital mixer that was introduced by Yamaha Pro Audio in 2005. Two models with onboard analog input exist: the M7CL-32 and M7CL-48. These models have 40 - and 56 -input channels respectively, counting mono channels. Mixes, masters, groups, DCAs and individual channels can then be routed to an output via any number of the board's 16 configurable output XLR ports. The eight faders of the master control section can control multiple functions by way of "layers" in the same manner as the Yamaha PM5D. The board features Yamaha's "Selected Channel" technology, and Centralogic, unique to the M7CL. It can be augmented with more inputs or outputs via expansion cards, and can be fitted with third-party cards such as ones made by Aviom (A-Net), AuviTran (EtherSound), Audinate, AudioService (MADI), Dan Dugan (automixer), Riedel Communications (RockNet), Waves Audio, and Optocore. In 2010, the M7CL-48ES joined the line-up with built-in EtherSound for digital networking using EtherSound stage boxes.

<span class="mw-page-title-main">Live sound mixing</span> Blending of multiple sound sources for a live event

Live sound mixing is the blending of multiple sound sources by an audio engineer using a mixing console or software. Sounds that are mixed include those from instruments and voices which are picked up by microphones and pre-recorded material, such as songs on CD or a digital audio player. Individual sources are typically equalised to adjust the bass and treble response and routed to effect processors to ultimately be amplified and reproduced via a loudspeaker system. The live sound engineer listens and balances the various audio sources in a way that best suits the needs of the event.

The term microphone preamplifier can either refer to the electronic circuitry within a microphone, or to a separate device or circuit that the microphone is connected to. In either instance, the purpose of the microphone preamplifier is the same.

<span class="mw-page-title-main">Yamaha Pro Audio</span> Professional audio system manufacturer

Yamaha Pro Audio, Inc. is the Pro Audio Division division of Yamaha Corporation that offers a complete line of beginner professional audio products for the live sound and sound reinforcement markets. Their lineup includes a number of world-standard mixing consoles, signal processors incorporating industry-leading DSP technology, power amplifiers based on energy-efficient drive technology, and an extensive range of speakers used for live sound or commercial installations. It has a long history of introducing significant products for the professional audio market such as the PM-1000 modular mixing console, the REV1 and SPX90 digital signal processors, the NS-10 studio monitors, and the 01v, 02R, 03D, PM1D, PM5D, QL5, M7CL, CL5, and PM10/7 Rivage digital mixing consoles.

<span class="mw-page-title-main">Stage monitor system</span> Sound reinforcement for performers

A stage monitor system is a set of performer-facing loudspeakers called monitor speakers, stage monitors, floor monitors, wedges, or foldbacks on stage during live music performances in which a sound reinforcement system is used to amplify a performance for the audience. The monitor system allows musicians to hear themselves and fellow band members clearly.

<span class="mw-page-title-main">Audio mixing (recorded music)</span> Audio mixing to yield recorded sound

In sound recording and reproduction, audio mixing is the process of optimizing and combining multitrack recordings into a final mono, stereo or surround sound product. In the process of combining the separate tracks, their relative levels are adjusted and balanced and various processes such as equalization and compression are commonly applied to individual tracks, groups of tracks, and the overall mix. In stereo and surround sound mixing, the placement of the tracks within the stereo field are adjusted and balanced. Audio mixing techniques and approaches vary widely and have a significant influence on the final product.

<span class="mw-page-title-main">Dan Dugan (audio engineer)</span> First sound designer; inventor of the automixer

Dan Dugan is an American audio engineer, inventor, and nature sounds recordist. He was the first person in regional theatre to be called a sound designer, and he developed the first effective automatic microphone mixer: the automixer. Dugan's sound design work was acknowledged in 2003 with a Distinguished Career Award by the United States Institute for Theatre Technology, and in 2020 with an Emmy Award for technology relevant to remote working. In 2021 he was awarded Fellowship in the Audio Engineering Society.

In live sound mixing, gain before feedback (GBF) is a practical measure of how much a microphone can be amplified in a sound reinforcement system before causing audio feedback. In audiology, GBF is a measure of hearing aid performance. In both fields the amount of gain is measured in decibels at or just below the point at which the sound from the speaker driver re-enters the microphone and the system begins to ring or feed back. Potential acoustic gain (PAG) is a calculated figure representing gain that a system can support without feeding back.

<span class="mw-page-title-main">Matrix mixer</span> Audio device that routes multiple audio signals

A matrix mixer is an audio electronics device that routes multiple input audio signals to multiple outputs. It usually employs level controls such as potentiometers to determine how much of each input is going to each output, and it can incorporate simple on/off assignment buttons. The number of individual controls is at least the number of inputs multiplied by the number of outputs.

<span class="mw-page-title-main">Keyboard amplifier</span>

A keyboard amplifier is a powered electronic amplifier and loudspeaker in a speaker cabinet used for the amplification of electronic keyboard instruments. Keyboard amplifiers are distinct from other types of amplification systems such as guitar amplifiers due to the particular challenges associated with making keyboards sound louder on stage; namely, to provide solid low-frequency sound reproduction for the deep basslines that keyboards can play and crisp high-frequency sound for the high-register notes. Another difference between keyboard amplifiers and guitar/bass amplifiers is that keyboard amps are usually designed with a relatively flat frequency response and low distortion. In contrast, many guitar and bass amp designers purposely make their amplifiers modify the frequency response, typically to "roll-off" very high frequencies, and most rock and blues guitar amps, and since the 1980s and 1990s, even many bass amps are designed to add distortion or overdrive to the instrument tone.

<span class="mw-page-title-main">Professional audio store</span> Retail business

A professional audio store is a retail business that sells, and in many cases rents, sound reinforcement system equipment and PA system components used in music concerts, live shows, dance parties and speaking events. This equipment typically includes microphones, power amplifiers, electronic effects units, speaker enclosures, monitor speakers, subwoofers and audio consoles (mixers). Some professional audio stores also sell sound recording equipment, DJ equipment, lighting equipment used in nightclubs and concerts and video equipment used in events, such as video projectors and screens. Some professional audio stores rent "backline" equipment used in rock and pop shows, such as stage pianos and bass amplifiers. While professional audio stores typically focus on selling new merchandise, some stores also sell used equipment, which is often the equipment that the company has previously rented out for shows and events.

References

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