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Raummode

2022-03-03T20:37:50Z

87.156.174.222: Created page with "'''Raummode''' (von {{enS|room mode}}, dort von {{laS|modus}}; Plural: '''Raummoden''') ist ein Fachbegriff der Akustik. Er beschreibt Eigenschaften Stehende Welle|stehe..."

'''Raummode''' (von {{enS|room mode}}, dort von {{laS|modus}}; Plural: '''Raummoden''') ist ein Fachbegriff der [[Akustik]].
Er beschreibt Eigenschaften [[Stehende Welle|stehender]] [[Schallwelle]]n mit einer [[Eigenfrequenz]] in geschlossenen Räumen, wobei vor allem die Auswirkung auf den [[Hörereignis|Höreindruck]] der darin befindlichen Menschen von Interesse ist.
[[Datei:Onde stationnaire vitesse tuyau ouvert trois modes.svg|mini|Raummoden zwischen zwei harten Wänden. An den Wänden muss dabei immer maximaler [[Schalldruck]] herrschen, was an den dortigen Druckbäuchen zu erkennen ist.]]

Eine Raummode ist eine den Raum ausfüllende [[Moden|Eigenform]] der Luft, während sie mit einer von mehreren Eigenfrequenzen schwingt. Die Schwingung pendelt dabei zwischen zwei gegensätzlichen [[Auslenkung]]szuständen. Die Raummoden zeigen also, wo im Raum sich Schwingungsknoten und -bäuche bei bestimmten Eigenfrequenzen ausbilden.

Für den Beobachtungszeitraum wandert die Welle nicht mehr durch den Raum, sondern hat feste [[Amplitude]]n-Maxima und -Minima. Die Schwingungsknoten sind [[Nullstelle]]n der Amplitude, d. h. an der Stelle, an der ein Knoten auftritt, gibt es keine Auslenkung. In der Praxis bedeutet dies, dass sich z. B. für Wohnräume mit [[Hifi-Anlage]]n der Höreindruck mit der Position der Person im Raum ändert.
Abhängig von der [[Raumakustik]] bilden sich speziell bei üblichen Wohnraumabmessungen einige Wohnraummoden im tiefen [[Frequenz]]bereich aus, die sehr störend wirken können.
Von vorrangiger Bedeutung sind jene Moden, die am stärksten ausgebildet sind. Bei Räumen gibt es sechs [[Freiheitsgrade]] für [[Eigenschwingung]]en, was zu einer mehrdimensionalen Zusammensetzung der möglichen Eigenfrequenzen und deren Schwingungsformen führt.

Grundsätzlich wird die Anzahl der maximal möglichen Freiheitsgrade durch die herrschenden [[Zwangsbedingung]]en wieder reduziert. Wenn man die ganzzahligen [[Harmonische]]n ausklammert, gibt es je Freiheitsgrad eine Eigenschwingung. Die Freiheitsgrade für Moden in Räume lassen sich für Berechnungen in guter Näherung auf drei begrenzen.

== Anregung von Eigenformen ==
Während kleine Räume ausgesprochen [[Linienspektrum|diskret]]e Eigenfrequenzen aufweisen, überlagern sich bei großen Räumen wie bei Kirchen alle Moden zu einem [[Kontinuum (Physik)|Kontinuum]] – es tritt verstärkt [[Nachhall|Hall]] auf. Bei Räumen spiegeln die Raummoden, wie der [[Klang]] eines Raums verfärbt wird, weil bestimmte Töne besonders hervortreten und eine ungleichförmige Energieverteilung innerhalb des Raums haben. Treten diskrete [[Resonanzfrequenz]]en auf, so sind diese auffälliger als wenn mehrere [[Resonanz]]en gleichmäßig im [[Frequenzspektrum|Spektrum]] verteilt sind.

Eine bestimmte Resonanzfrequenzverteilung ist eine physikalische Eigenschaft des Raumes, die von seinen Abmessungen abhängt. Nur bestimmte [[Frequenz]]en werden angeregt. Bei diesen Resonanzeffekten spielen sowohl der erhöhte [[Schalldruckpegel|Pegel]] als auch die zeitliche Fortdauer des [[Ton (Musik)|Tons]] eine Rolle. Die Amplitude einer akustischen Mode hängt von der Position im Raum ab. Der Grad der Klangverfärbung ist daher von Ort zu Ort verschieden.

[[Datei:ANI Stehende Welle.gif|mini|Eine stehende Welle. An den Enden (den Raumbegrenzungen) erscheint jeweils ein Druckbauch als Maximum.]]

== Schröderfrequenz ==
Kennt man die [[Nachhallzeit]] <math>T</math> eines Raumes (in Sekunden) und sein Volumen <math>V</math> (in <math>\mathrm{m^3}</math>), so kann mit Hilfe folgender [[Zahlenwertgleichung]] die [[Schröderfrequenz]] oder Großraumfrequenz <math>f_\text{s}</math> bestimmt werden, die bei den meisten Räumen um 300 Hz liegt:<ref>Thomas Görne: ''Tontechnik.'' 2008, ISBN 3-446-41591-2, S. 72 ({{Google Buch |BuchID=LJkHImqC9HsC |Seite=72}}).</ref>

: <math>f_\text{s} = 2000 \, \sqrt {\frac {\text{T}} {\text{V}}}\text{Hz} \approx \frac{1}{\pi}\sqrt {\frac {c_S^3{T}} {V}} = \frac{c_S}{r_H\pi} </math>
Nahezu das Gleiche ergibt sich, wenn man die Schallgeschwindigkeit c<sub>S</sub> bzw. den [[Hallradius]] r<sub>H</sub> einsetzt und das Ergebnis durch π teilt.

Unterhalb der Schröderfrequenz können akustische Moden des Raums wahrnehmbare Klangverfärbungen bewirken. Da diese besonders die tiefen Töne betreffen, werden sie als ''Dröhnen'', ''Booming'' oder ''Ein-Noten-Bass'' empfunden. Oberhalb dagegen verursachen sie in Wohnräumen keine hörbaren [[Verzerrung (Akustik)|Verzerrungen]] der Wiedergabe, weil die Moden in Form von dichten [[Reflexion (Physik)|Reflexionen]] und Nachhall ineinander übergehen.

== Berechnung ==
Vorrangig werden drei Arten stehender Moden berechnet, die in einem typischen [[quader]]förmigen Hörraum vorkommen. Dieses sind axiale (longitudinale), tangentiale und diagonale Moden (auch Obligue- oder Schrägmoden genannt). Die axialen Moden dominieren deutlich.

{{"|Die Raummode erster Ordnung tritt bei einer Frequenz auf, deren halbe Wellenlänge dem Abstand zwischen den beiden Wänden entspricht. […]
Die Eigenfrequenzen <math>f_\text{n}</math> des vom betrachteten Wandpaar eingeschlossenen eindimensionalen Raumes berechnet sich aus

: <math>f_\text{n} = \frac {c_0 \cdot n} {2 \cdot d}</math>

Dabei ist
* <math>c_0</math> die [[Schallgeschwindigkeit]],
* <math>d</math> der Abstand zwischen den beiden Wänden und
* <math>n</math> die Ordnung der Raummode, die auch gleichzeitig der Anzahl der Schalldruckminima […] entspricht.
Die an zwei parallelen Wänden angestellten Überlegungen lassen sich auf dreidimensionale quaderförmige Räume übertragen. Dabei treten zusätzlich zu den beschriebenen, als axial bezeichneten Moden zwischen zwei gegenüberliegenden Wandpaaren auch Moden auf, deren Pfade sich in zwei und drei Dimensionen des Raumes bewegen. Man bezeichnet diese im zweidimensionalen Fall als tangentiale und im dreidimensionalen Fall als oblique Moden.

Die Berechnung aller Eigenfrequenzen <math>f_\mathrm{n_x / n_y / n_z}</math> eines quaderförmigen Raumes kann mit der bereits 1896 von [[John William Strutt, 3. Baron Rayleigh]] beschrieben Formel erfolgen:

: <math>f_\mathrm{n_x / n_y / n_z} = \frac{c_0}{2} \sqrt{\left(\frac{n_\text{x}}{l_\text{x}}\right)^2 + \left(\frac{n_\text{y}}{l_\text{y}}\right)^2 + \left(\frac{n_\text{z}}{l_\text{z}}\right)^2} \,</math>

Dabei ist
* wiederum <math>c_0</math> die Schallgeschwindigkeit,
* <math>l_\text{x}, l_\text{y}</math> und <math>l_\text{z}</math> sind die Abmessungen des Raumes, also Länge, Breite und Höhe, und
* <math>n_\text{x}, n_\text{y}</math> und <math>n_\text{z}</math> bezeichnen die Ordnungen der Moden in den jeweiligen Richtungen. […]
Aus der Überlagerung aller Moden eines Raumes setzt sich die räumliche [[Schalldruck]]-[[Schallschnelle]]-Verteilung und damit das dreidimensionale Feld komplexer Schallfeldimpedanzen zusammen. Raummoden sind resonanzfähige Systeme. |Autor=Stefan Weinzierl}}<ref>Stefan Weinzierl: ''Handbuch der Audiotechnik.'' 2008, ISBN 3-540-34300-8, S. 284–285 ({{Google Buch|BuchID=OLgY0QpXD0YC|Seite=284}}).</ref>

== Ordnungszahl ==
Die Frequenzen und Eigenschwingungsformen werden nach ihrer Ordnungszahl (Nummer) benannt, also:
# Die erste Eigenschwingungsform oder Grundform stellt sich bei einer Schwingung mit der ersten Eigenfrequenz, der [[Grundfrequenz]], ein.
# Die zweite Eigenschwingungsform schwingt mit der zweiten Eigenfrequenz.
# usw.
Ist die Zusammensetzung der Eigenfrequenzen komplexer, etwa bei Räumen, so wird die Ordnungszahl mehrstellig oder durch Komma getrennt in Klammern angegeben.

== Minimierung der Auswirkungen ==
Ein Raum mit harten Wänden zeigt markante Spitzen bei bestimmten Raumresonanzfrequenzen. Durch Maßnahmen zur [[Schallabsorption]] kann das geändert werden. Je nach Menge und Position der absorbierenden Materialien in einem Raum werden diese markanten Ausprägungen gemindert. Es gibt inzwischen eine Vielfalt an akustischen Absorbermaterialien, die geeignet sind bestimmte Frequenzbereiche bevorzugt zu dämpfen. Mikroperforierte Deckenpanele, Spezialfolien mit Perforationen, und konventionelle Absorberpanele können verwendet oder kombiniert werden um [[Raumakustik]] für den jeweiligen Einsatzbereich zu optimieren.

{{Zitat
|Text=Für Abhängehöhen zwischen 200 und 600 mm, wie sie in der Praxis häufig vorkommen, liegt das Wirkungsmaximum dieser neuartigen Akustik-Decke im so wichtigen Frequenzbereich zwischen 125 und 500 Hz, wo bei der heute üblichen kargen Möblierung mit durchweg schallharten Oberflächen die Schallabsorption der Decke dringend benötigt wird. Bei Frequenzen zwischen 500 und 2000 Hz, wo die Decke weniger stark schluckt, ist i.a. Schallabsorption durch Teppiche, Vorhänge und die Personen selbst vorhanden. Dies führt zu einer relativ ausgeglichenen Nachhallzeit über der Frequenz und einem geringeren Schallpegel in den Räumen.
|ref=<ref>{{Webarchiv |url=http://www.ibp.fraunhofer.de/Kompetenzen/akustik/raumakustik/ |wayback=20121021005452 |text=Raumakustik, Forschungsrichtung: „Mikroperforierte Metallkassetten als Unterdecke“.}} Fraunhofer-Institut für Bauphysik:</ref>}}

Auch passive und aktive [[Helmholtz-Resonator#Als Absorber in der Raumakustik|Resonanzabsorber]] kommen zum Einsatz. Besteht die Möglichkeit die Raumgeometrie in der Planungsphase zu ändern, so kann man günstige Proportionen erreichen<ref>{{Internetquelle |url=http://www.zehner.ch/lab/faq.html#11 |titel=Geeignete Raumdimensionen |abruf=2020-03-27 }}</ref>. In Kombination mit geeigneten Schalldämmungsmaßmahnen kann die Raumakustik für den Anwendungsbereich weiter optimiert werden. Auch die Art der Wandkonstruktion hat Einfluss auf die Raumakustik, Leichtbauweise führt meist zu einem geringeren Bedarf an zusätzlichen Maßnahmen. Teppichboden oder schwere Vorhänge verändern die Raumakustik jedoch in einen Bereich der nicht unbedingt gedämpft werden soll. Auf tiefe zum Raumdröhnen führende Raummoden haben diese so gut wie keinen Einfluss.

=== Sound-Systeme ===
Einige Anbieter bieten seit 1990 aufwendige sog. Raumkorrektursysteme mit Messmikrofonen und nutzen aktuelle Möglichkeiten digitaler Filterung, um den notwendigen Ausgleich für Raummoden zu implementieren. Angesichts der hohen Kosten für diese Systeme gibt es eine Kontroverse über den relativen Wert der Verbesserung in normalen Räumen. Optimale Nutzung erfordert Grundkenntnisse der akustischen Zusammenhänge vom Betreiber und eine umfangreiche Datenerfassung am Aufstellungsort, die automatisiert in der Einstellungsphase der Geräte vorgenommen werden muss. Die Kompensation und Entzerrung über den Frequenzgang des verwendeten Sound-Systems sind von begrenztem Nutzen, da zeitlich ablaufende Vorgänge nicht beeinflusst werden, wie die Nachhallzeit und Ein-Ausschwingvorgänge. Die Entzerrung passt nur für eine bestimmte Hörposition und kann bei falscher Anwendung dazu führen, dass andere Hörpositionen sogar verschlechtert werden. Weder die Lautsprecher- noch die Messmikrofonplatzierung darf in einem Knotenpunkt erfolgen, denn die akustisch bedingte Auslöschung einer Frequenz kann nicht durch erhöhte Verstärkung derselben Frequenz kompensiert werden. Eine solche Überkompensation würde die Lautsprecher ohne nennenswerten Nutzeffekt übersteuern.

Raumkorrektursysteme erfassen den Ist-Zustand-Frequenzgang am Messort, der Benutzer kann meist eine Zielkurve auswählen oder selbst gestalten und das Korrektursystem generiert eine Entzerrungskurve, die die Differenz von Zielkurve und Messkurve ausgleichen soll. Die Gestaltung der Zielkurve erfordert Kenntnisse und Beachtung der lokalen Gegebenheiten, eine vereinfachend eingesetzte linealgerade Frequenzgang-Vorgabe wäre ein typischer Anfängerfehler. Wenn auch ein im reflexionsfreien Raum gemessener Lautsprecher noch einen linearen Frequenzgang zeigte, ist beim praktischen Einsatz in Räumen neben den Raumresonanzen auch mit Reflexionen an Boden, Decke und Seitenwänden zu rechnen, die zeitverzögert eintreffen. Der resultierende gemessene Frequenzgang kann nicht mehr linear sein, weil das Zeitfenster der Messung groß genug sein muss, um den Aufbau der Raumresonanzen nicht auszuschließen.

Ein alternativer Kompromiss ist, die bekannten Raumresonanzfrequenzen möglichst vollständig durch den Einsatz digitaler [[Kammfilter]] oder [[Kerbfilter]] bei der Beschallung des Raumes zu unterdrücken. Eine weitere Möglichkeit ist, die Anzahl der Tieftonlautsprecher und Verstärkerkanäle zu erhöhen und diese über aufwendige digitale Verarbeitung gezielt bei den betroffenen Resonanzfrequenzen gegenphasig und unter Berücksichtigung der Signallaufzeiten durch den Raum anzusteuern und somit Reflexionen zum Teil auszulöschen, wobei der Aufstellungsort der Lautsprecher im Raum von besonderer Bedeutung ist. Damit gelingt zum Teil bei größeren Räumen und unter Vernachlässigung der Raummoden in tangentialer und diagonaler Ausrichtung eine Verbesserung der Wiedergabe.

== Siehe auch ==
* [[Normalschwingung]]
* [[Shuntimpedanz]]

== Literatur ==
* {{Literatur
|Autor=J. Krüger, M. Leitner, P. Leistner
|Titel=PC-Instrumente zur Messung und Prüfung akustischer Parameter
|Sammelwerk=IBP-Mitteilung
|Band=331
|Nummer=25
|Datum=1998
|Sprache=de
|Online=https://www.ibp.fraunhofer.de/content/dam/ibp/de/documents/Publikationen/IBP-Mitteilungen-optimiert/331.pdf
|Format=PDF
|KBytes=}}
* H. V. Fuchs, C. Häusler, X. Zha: ''Kleine Löcher, große Wirkung.'' In: ''Trockenbau-Akustik.'' 14, Nr. 8, 1997, S. 34–37.
* H. V. Fuchs, M. Möser: ''Schallabsorber.'' In: Gerhard Müller: ''Taschenbuch Der Technischen Akustik.'' Springer, 2003, ISBN 3-540-41242-5, S. 247 ({{Google Buch|BuchID=99_004Seq0AC|Seite=247}}).
* H. Kuttruff, E. Mommertz: ''Raumakustik.'' In: Gerhard Müller: ''Taschenbuch Der Technischen Akustik.'' Springer, 2003, ISBN 3-540-41242-5, S. 331 ({{Google Buch|BuchID=99_004Seq0AC|Seite=331}}).

== Weblinks ==
* [http://www.trikustik.at/rechner/rechner-raummoden.html Raumeigenmoden-Rechner. Für rechteckige Räume ermittelt dieser Rechner die Raumeigenmoden mit den 20 niedrigsten Eigenfrequenzen und stellt sie in aufsteigender Reihenfolge dar.]
* [http://www.sengpielaudio.com/Rechner-raum-moden.htm Berechnung der drei Raummoden von Rechteck-Räumen (Stehende-Wellen-Berechnung)]
* [http://amroc.andymel.eu/ Umfassender Raummoden Rechner (3D Ansicht der Druckverteilung jeder Mode, Audio Wiedergabe, Bonello-Diagramm, Bolt-area, Schröderfrequenz in Abhängigkeit von der Nachhallzeit uva.)]
* [http://www.sengpielaudio.com/StehendeWellen.htm Der Unterschied zwischen den Moden als Schalldruckverteilung in Räumen und den Moden der Saitenschwingungen]
* [http://masterclass-sounddesign.com/content_bonusmaterial/raumakustikanalyse.htm Vorgehensweise bei der Raumakustikanalyse - Praktische Anweisungen zum Feststellen der Raumakustik]
* {{Webarchiv |url=http://www.pia-alfa.de/de/anim.htm |wayback=20110811004912 |text=Simulation der Anregung kleiner Räume bei tiefen Frequenzen, IBP, Fraunhofer Institut Bauphysik}}

== Einzelnachweise ==
<references />

[[Kategorie:Akustik]]
[[Kategorie:Wellenlehre]]
[[Kategorie:Nachrichtentechnik]]

Microphone

2022-03-03T20:36:47Z

87.156.174.222:

[[File:Shure mikrofon 55S.jpg|thumb|[[Shure Brothers]] microphone, model 55s, Multi-Impedance "Small Unidyne" Dynamic from 1951]]
[[File:SennMicrophone.jpg|thumb|right|A [[Sennheiser]] dynamic microphone]]

A '''microphone''', colloquially named '''mic''' or '''mike''' ({{IPAc-en|m|aɪ|k}}),<ref>{{cite news|url=https://www.nytimes.com/2010/08/01/magazine/01-onlanguage-t.html?_r=1|title=How Should 'Microphone' be Abbreviated?|last=Zimmer|first=Ben|date=29 July 2010|work=[[The New York Times]]|accessdate=10 September 2010}}</ref> is a device – a [[transducer]] – that converts [[sound]] into an [[electrical signal]]. Microphones are used in many applications such as [[telephone]]s, [[hearing aid]]s, [[public address system]]s for concert halls and public events, [[motion picture]] production, live and recorded [[audio engineering]], [[sound recording]], [[two-way radio]]s, [[megaphone]]s, [[radio]] and [[television]] broadcasting. They are also used in computers for recording voice, [[speech recognition]], [[Voice over IP|VoIP]], and for non-acoustic purposes such as ultrasonic sensors or [[Automatic Performance Control|knock sensors]].

Several types of microphone are used today, which employ different methods to convert the air pressure variations of a [[Raummode]] to an electrical signal. The most common are the [[dynamic microphone]], which uses a coil of wire suspended in a magnetic field; the [[condenser microphone]], which uses the vibrating [[Diaphragm (acoustics)|diaphragm]] as a [[capacitor]] plate; and the [[contact microphone]], which uses a crystal of [[piezoelectric]] material. Microphones typically need to be connected to a [[preamplifier]] before the signal can be [[Sound recording and reproduction|recorded or reproduced]].

==History==
In order to speak to larger groups of people, a need arose to increase the volume of the human voice. The earliest devices used to achieve this were acoustic [[megaphone]]s. Some of the first examples, from fifth century BC Greece, were theater masks with horn-shaped mouth openings that acoustically amplified the voice of actors in [[amphitheater]]s.<ref name="Montgomery1959"/> In 1665, the English physicist [[Robert Hooke]] was the first to experiment with a medium other than air with the invention of the "[[lovers' telephone]]" made of stretched wire with a cup attached at each end.<ref>{{cite web|last1=McVeigh|first1=Daniel|title=An Early History of the Telephone: 1664–1866: Robert Hooke's Acoustic Experiments and Acoustic Inventions|url=http://oceanofk.org/telephone/html/part1.html|archiveurl=https://web.archive.org/web/20030903082530/http://www.oceanofk.org/telephone/html/part1.html|archivedate=2003-09-03|date=2000}}</ref>

In 1861, German inventor [[Johann Philipp Reis]] built an early sound transmitter (the "[[Reis telephone]]") that used a metallic strip attached to a vibrating membrane that would produce intermittent current. Better results were achieved in 1876 with the "[[#Liquid|liquid transmitter]]" design in early telephones from [[Alexander Graham Bell]] and [[Elisha Gray]] – the diaphragm was attached to a conductive rod in an acid solution.<ref>MacLeod, Elizabeth 1999 Alexander Graham Bell: an inventive life. Kids Can Press, Toronto</ref> These systems, however, gave a very poor sound quality.

[[File:David Edward Hughes.jpg|thumb|200x200px|[[David Edward Hughes]] invented a [[carbon microphone]] in the 1870s.]]
The first microphone that enabled proper voice telephony was the (loose-contact) [[carbon microphone]]. This was independently developed by [[David Edward Hughes]] in England and [[Emile Berliner]] and [[Thomas Edison]] in the US. Although Edison was awarded the first patent (after a long legal dispute) in mid-1877, Hughes had demonstrated his working device in front of many witnesses some years earlier, and most historians credit him with its invention.<ref>{{cite book|url=https://books.google.com/books?id=e9wEntQmA0IC|title=Oliver Heaviside: The Life, Work, and Times of an Electrical Genius of the Victorian Age|author=Paul J. Nahin|authorlink=Paul J. Nahin|year=2002|publisher=JHU Press|page=67|isbn=9780801869099}}</ref><ref>{{cite web|url=http://telephonecollecting.org/DavidHughes.html|author=Bob Estreich|title=David Edward Hughes}}</ref><ref name="Huurdeman 2003"/><ref>{{Cite web|url=http://www.britannica.com/EBchecked/topic/274915/David-Hughes|title=David Hughes|accessdate=2012-12-17}}</ref> The carbon microphone is the direct prototype of today's microphones and was critical in the development of telephony, broadcasting and the recording industries.<ref>{{Cite web|url=http://www.angloconcertina.org/files/HughesforWebsite.pdf|title=David Edward Hughes: Concertinist and Inventor|accessdate=2012-12-17|archiveurl=https://web.archive.org/web/20131231000041/http://www.angloconcertina.org/files/HughesforWebsite.pdf|archivedate=2013-12-31}}</ref> [[Thomas Edison]] refined the carbon microphone into his carbon-button transmitter of 1886.<ref name="Huurdeman 2003"/><ref name="microphone-data"/> This microphone was employed at the first ever radio broadcast, a performance at the New York [[Metropolitan Opera House (Lincoln Center)|Metropolitan Opera House]] in 1910.<ref>{{cite web |title=Lee De Forest – (1873–1961) |url=http://www.smart90.com/deforest|publisher=Television International Magazine |date=2011-01-17 |accessdate=Dec 4, 2013 |archiveurl=https://web.archive.org/web/20110117104308/http://www.smart90.com/deforest |archivedate=2011-01-17}}</ref><ref>{{cite web|url=http://www.smart90.com/deforest|title="Radio Boys" & "The SMART-DAAF BOYS"|last1=Cory|first1=Troy|date=2003-01-21|archiveurl=https://web.archive.org/web/20030121065153/http://www.smart90.com/deforest|archivedate=January 21, 2003|url-status=unfit}}</ref>

[[File:Bogart Bacall AFRS.jpg|thumb|left|Jack Brown interviews [[Humphrey Bogart]] and [[Lauren Bacall]] for broadcast to troops overseas during World War II.]]
In 1916, E.C. Wente of Western Electric developed the next breakthrough with the first [[#Condenser microphone|condenser microphone]].<ref>Fagen, M.D. A History of Engineering and Science in the Bell System: The Early Years (1875–1925). New York: Bell Telephone Laboratories, 1975</ref> In 1923, the first practical moving coil microphone was built. The Marconi-Sykes magnetophone, developed by [[H. J. Round|Captain H. J. Round]], became the standard for [[BBC]] studios in London.<ref>Hennessy, Brian 2005 The Emergence of Broadcasting in Britain Devon Southerleigh</ref><ref>{{citation |title=The Marconi-Sykes Magnetophone |url=http://www.coutant.org/magnetophone/ |access-date=2018-06-18}}</ref> This was improved in 1930 by [[Alan Blumlein]] and Herbert Holman who released the HB1A and was the best standard of the day.<ref>{{cite web |last1=Robjohns |first1=Hugh |title=A Brief History of Microphones |url=http://microphone-data.com/media/filestore/articles/History-10.pdf |website=Microphone Data Book |archiveurl=https://web.archive.org/web/20101125131858/http://microphone-data.com/media/filestore/articles/History-10.pdf |archivedate=2010-11-25 |date=2001}}</ref>

Also in 1923, the [[ribbon microphone]] was introduced, another electromagnetic type, believed to have been developed by [[Harry F. Olson]], who essentially reverse-engineered a ribbon speaker.<ref>{{cite web |title=1931 Harry F. Olson and Les Anderson, RCA Model 44 Ribbon Microphone |url=http://mixonline.com/TECnology-Hall-of-Fame/olson-anderson-rca-090106/ |publisher=[[Mix Magazine]] |date=Sep 1, 2006 |accessdate=10 April 2013 |archiveurl=https://web.archive.org/web/20080324111751/http://mixonline.com/TECnology-Hall-of-Fame/olson-anderson-rca-090106 |archivedate=2008-03-24 }}</ref> Over the years these microphones were developed by several companies, most notably RCA that made large advancements in pattern control, to give the microphone directionality. With television and film technology booming there was a demand for high fidelity microphones and greater directionality. [[Electro-Voice]] responded with their Academy Award-winning [[shotgun microphone]] in 1963.

During the second half of 20th-century development advanced quickly with the [[Shure]] Brothers bringing out the [[Shure SM58|SM58]] and [[Shure SM57|SM57]].<ref>{{cite web |publisher=Shure Americas |url=http://www.shure.com/americas/about-shure/history |title=History – The evolution of an audio revolution |accessdate=13 April 2013 |archiveurl=https://web.archive.org/web/20120915095300/http://www.shure.com/americas/about-shure/history |archivedate=2012-09-15}}</ref> The latest research developments include the use of fibre optics, lasers and interferometers.

==Components==
[[File:MicrophoneSymbol.png|thumb|Electronic symbol for a microphone]]
The sensitive transducer element of a microphone is called its ''element'' or ''capsule.'' Sound is first converted to mechanical motion by means of a diaphragm, the motion of which is then converted to an electrical signal. A complete microphone also includes a housing, some means of bringing the signal from the element to other equipment, and often an electronic circuit to adapt the output of the capsule to the equipment being driven. A [[wireless microphone]] contains a [[radio transmitter]].

== Varieties ==
Microphones are categorized by their [[transducer]] principle, such as condenser, dynamic, etc., and by their directional characteristics. Sometimes other characteristics such as diaphragm size, intended use or orientation of the principal sound input to the principal axis (end- or side-address) of the microphone are used to describe the microphone.

{{Anchor|Condenser microphone}}

=== Condenser ===
[[File:Oktava319-internal.jpg|thumb|left|upright|Inside the Oktava 319 condenser microphone]]

The '''condenser microphone''', invented at Western Electric in 1916 by E. C. Wente,<ref>{{cite web |url=http://www.stokowski.org/Development_of_Electrical_Recording.htm |title= Bell Laboratories and The Development of Electrical Recording |publisher=Stokowski.org (Leopold Stokowski site)}}</ref> is also called a '''capacitor microphone''' or '''electrostatic microphone'''—capacitors were historically called condensers. Here, the [[Diaphragm (acoustics)|diaphragm]] acts as one plate of a [[capacitor]], and the vibrations produce changes in the distance between the plates. There are two types, depending on the method of extracting the [[audio signal]] from the transducer: DC-biased microphones, and radio frequency (RF) or high frequency (HF) condenser microphones. With a '''DC-biased microphone''', the plates are [[Voltage bias|biased]] with a fixed charge (''Q''). The [[voltage]] maintained across the capacitor plates changes with the vibrations in the air, according to the capacitance equation (C = {{fract|Q|V}}), where Q = charge in [[coulomb]]s, C = capacitance in [[farad]]s and V = potential difference in [[volt]]s. The capacitance of the plates is inversely proportional to the distance between them for a parallel-plate capacitor. The assembly of fixed and movable plates is called an "element" or "capsule".

A nearly constant charge is maintained on the capacitor. As the capacitance changes, the charge across the capacitor does change very slightly, but at audible frequencies it is sensibly constant. The capacitance of the capsule (around 5 to 100 [[Farad|pF]]) and the value of the bias resistor (100 [[ohm|MΩ]] to tens of GΩ) form a filter that is high-pass for the audio signal, and low-pass for the bias voltage. Note that the time constant of an [[RC circuit]] equals the product of the resistance and capacitance.

Within the time-frame of the capacitance change (as much as 50 ms at 20 Hz audio signal), the charge is practically constant and the voltage across the capacitor changes instantaneously to reflect the change in capacitance. The voltage across the capacitor varies above and below the bias voltage. The voltage difference between the bias and the capacitor is seen across the series resistor. The voltage across the resistor is amplified for performance or recording. In most cases, the electronics in the microphone itself contribute no voltage gain as the voltage differential is quite significant, up to several volts for high sound levels. Since this is a very high impedance circuit, only current gain is usually needed, with the voltage remaining constant.

[[Image:AKG C451B.jpg|thumb|right|[[AKG Acoustics|AKG]] C451B small-diaphragm condenser microphone]]
'''RF condenser microphones''' use a comparatively low RF voltage, generated by a low-noise oscillator. The signal from the oscillator may either be amplitude modulated by the capacitance changes produced by the sound waves moving the capsule diaphragm, or the capsule may be part of a [[resonant circuit]] that modulates the frequency of the oscillator signal. Demodulation yields a low-noise audio frequency signal with a very low source impedance. The absence of a high bias voltage permits the use of a diaphragm with looser tension, which may be used to achieve wider frequency response due to higher compliance. The RF biasing process results in a lower electrical impedance capsule, a useful by-product of which is that RF condenser microphones can be operated in damp weather conditions that could create problems in DC-biased microphones with contaminated insulating surfaces. The [[Sennheiser]] "MKH" series of microphones use the RF biasing technique. A covert, remotely energised [[The Thing (listening device)|application of the same physical principle]] was devised by Soviet Russian inventor [[Léon Theremin]] and used to bug the US Ambassador's Residence in Moscow between 1945 and 1952.

Condenser microphones span the range from telephone transmitters through inexpensive karaoke microphones to high-fidelity recording microphones. They generally produce a high-quality audio signal and are now the popular choice in laboratory and [[recording studio]] applications. The inherent suitability of this technology is due to the very small mass that must be moved by the incident sound wave, unlike other microphone types that require the sound wave to do more work. They require a power source, provided either via microphone inputs on equipment as [[phantom power]] or from a small battery. Power is necessary for establishing the capacitor plate voltage and is also needed to power the microphone electronics (impedance conversion in the case of electret and DC-polarized microphones, demodulation or detection in the case of RF/HF microphones). Condenser microphones are also available with two diaphragms that can be electrically connected to provide a range of polar patterns (see below), such as cardioid, omnidirectional, and figure-eight. It is also possible to vary the pattern continuously with some microphones, for example, the [[Røde]] NT2000 or CAD M179.

A [[valve microphone]] is a condenser microphone that uses a [[vacuum tube]] (valve) amplifier.<ref>{{cite web|last=Institute BV Amsterdam| first=SAE|title=Microphones |url=http://www.sae.edu/reference_material/audio/pages/Microphones.htm|publisher=Practical Creative Media Education|accessdate=2014-03-07}}</ref> They remain popular with enthusiasts of [[tube sound]].

==== Electret condenser ====
{{Main|Electret microphone}}
[[Image:US Patent 3118022 - Gerhard M. Sessler James E. West - Bell labs - electroacustic transducer - foil electret condenser microphone 1962 1964 - pages 1-3.png|thumb|left|First patent on foil electret microphone by G. M. Sessler et al. (pages 1 to 3)]]

An electret microphone is a type of condenser microphone invented by [[Gerhard Sessler]] and [[James Edward Maceo West|Jim West]] at [[Bell laboratories]] in 1962.<ref>{{cite journal
|first=G.M. |last=Sessler |author2=West, J.E.
|title=Self-biased condenser microphone with high capacitance
|journal=Journal of the Acoustical Society of America |volume=34 |year=1962 |pages=1787–1788
|doi=10.1121/1.1909130
|issue=11|bibcode=1962ASAJ...34.1787S }}</ref>
The externally applied charge used for a conventional condenser microphones is replaced by a permanent charge in an electret material. An [[electret]] is a [[ferroelectric]] material that has been permanently [[electric charge|electrically charged]] or ''polarized''. The name comes from ''electr''ostatic and magn''et''; a static charge is embedded in an electret by the alignment of the static charges in the material, much the way a [[permanent magnet]] is made by aligning the magnetic domains in a piece of iron.

Due to their good performance and ease of manufacture, hence low cost, the vast majority of microphones made today are electret microphones; a semiconductor manufacturer estimates annual production at over one billion units.<ref>{{citation |url=http://www.national.com/nationaledge/dec02/article.html |archive-url=https://web.archive.org/web/20100819045334/http://www.national.com/nationaledge/dec02/article.html |archive-date=August 19, 2010 |url-status=dead |title=Integrated Circuits for High Performance Electret Microphones |author=Arie Van Rhijn |publisher=National Semiconductor}}</ref> They are used in many applications, from high-quality recording and [[lavalier microphone|lavalier]] (lapel mic) use to built-in microphones in small [[sound recording]] devices and telephones. Prior to the proliferation of MEMS microphones,<ref>{{cite journal |title=The Evolution of Integrated Interfaces for MEMS Microphones |author1=Piero Malcovati |author2=Andrea Baschirotto |doi=10.3390/mi9070323 |pmid=30424256 |pmc=6082321 |journal= Micromachines|volume=9 |issue=7 |pages=323 |year=2018 }}</ref> nearly all cell-phone, computer, PDA and headset microphones were electret types.

Unlike other capacitor microphones, they require no polarizing voltage, but often contain an integrated [[Microphone preamplifier|preamplifier]] that does require power (often incorrectly called polarizing power or bias). This preamplifier is frequently [[phantom power]]ed in [[sound reinforcement]] and studio applications. Monophonic microphones designed for [[personal computer]]s (PCs), sometimes called multimedia microphones, use a 3.5 mm plug as usually used, without power, for stereo; the ring, instead of carrying the signal for a second channel, carries power via a resistor from (normally) a 5 V supply in the computer. Stereophonic microphones use the same connector; there is no obvious way to determine which standard is used by equipment and microphones.

Though electret microphones were once considered low quality, the best ones can now rival traditional condenser microphones in every respect and can even offer the long-term stability and ultra-flat response needed for a measurement microphone. Only the best electret microphones rival good DC-polarized units in terms of noise level and quality; electret microphones lend themselves to inexpensive mass-production, while inherently expensive non-electret condenser microphones are made to higher quality.

{{Anchor|Dynamic microphone}}

=== Dynamic ===
[[Image:Patti Smith performing in Finland, 2007.jpg|thumb|[[Patti Smith]] singing into a [[Shure SM58]] (dynamic cardioid type) microphone]]

The '''dynamic microphone''' (also known as the '''moving-coil microphone''') works via [[electromagnetic induction]]. They are robust, relatively inexpensive and resistant to moisture. This, coupled with their potentially high [[gain before feedback]], makes them ideal for on-stage use.

Dynamic microphones use the same dynamic principle as in a [[loudspeaker]], only reversed. A small movable [[induction coil]], positioned in the [[magnetic field]] of a [[permanent magnet]], is attached to the [[diaphragm (acoustics)|diaphragm]]. When sound enters through the windscreen of the microphone, the sound wave moves the diaphragm. When the diaphragm vibrates, the coil moves in the magnetic field, producing a varying [[current (electricity)|current]] in the coil through [[electromagnetic induction]]. A single dynamic membrane does not respond linearly to all audio frequencies. For this reason, some microphones utilize multiple membranes for the different parts of the audio spectrum and then combine the resulting signals. Combining the multiple signals correctly is difficult; designs that do this are rare and tend to be expensive. On the other hand, there are several designs that are more specifically aimed towards isolated parts of the audio spectrum. The [[AKG Acoustics|AKG]] D112, for example, is designed for bass response rather than treble.<ref>[http://www.akg.com/site/products/powerslave,id,261,pid,261,nodeid,2,_language,EN.html "AKG D 112 – Large-diaphragm dynamic microphone for bass instruments]"</ref> In audio engineering several kinds of microphones are often used at the same time to get the best results.

=== Ribbon ===
{{main|Ribbon microphone}}
[[Image:Edmund Lowe fsa 8b06653.jpg|thumb|upright|left|[[Edmund Lowe]] using a ribbon microphone]]

[[Ribbon microphone]]s use a thin, usually corrugated metal ribbon suspended in a magnetic field. The ribbon is electrically connected to the microphone's output, and its vibration within the magnetic field generates the electrical signal. Ribbon microphones are similar to moving coil microphones in the sense that both produce sound by means of magnetic induction. Basic ribbon microphones detect sound in a [[#Bi-directional|bi-directional]] (also called figure-eight, as in the [[#Microphone polar patterns|diagram]] below) pattern because the ribbon is open on both sides. Also, because the ribbon has much less mass it responds to the air velocity rather than the [[sound pressure]]. Though the symmetrical front and rear pickup can be a nuisance in normal stereo recording, the high side rejection can be used to advantage by positioning a ribbon microphone horizontally, for example above cymbals, so that the rear lobe picks up sound only from the cymbals. Crossed figure 8, or [[Blumlein Pair|Blumlein pair]], stereo recording is gaining in popularity, and the figure-eight response of a ribbon microphone is ideal for that application.

Other directional patterns are produced by enclosing one side of the ribbon in an acoustic trap or baffle, allowing sound to reach only one side. The classic [[RCA Type 77-DX microphone]] has several externally adjustable positions of the internal baffle, allowing the selection of several response patterns ranging from "figure-eight" to "unidirectional". Such older ribbon microphones, some of which still provide high-quality sound reproduction, were once valued for this reason, but a good low-frequency response could be obtained only when the ribbon was suspended very loosely, which made them relatively fragile. Modern ribbon materials, including new nanomaterials,<ref>{{cite journal
|url=http://www.bizjournals.com/masshightech/stories/2008/02/11/story8.html
|title=Local firms strum the chords of real music innovation
|journal=Mass High Tech: The Journal of New England Technology
|date=February 8, 2008
}}</ref> have now been introduced that eliminate those concerns and even improve the effective dynamic range of ribbon microphones at low frequencies. Protective wind screens can reduce the danger of damaging a vintage ribbon, and also reduce plosive artifacts in the recording. Properly designed wind screens produce negligible treble attenuation. In common with other classes of dynamic microphone, ribbon microphones don't require [[phantom power]]; in fact, this voltage can damage some older ribbon microphones. Some new modern ribbon microphone designs incorporate a preamplifier and, therefore, do require phantom power, and circuits of modern passive ribbon microphones, ''i.e.'', those without the aforementioned preamplifier, are specifically designed to resist damage to the ribbon and transformer by phantom power. Also there are new ribbon materials available that are immune to wind blasts and phantom power.

=== Carbon ===
{{Main|Carbon microphone}}
[[File:Western Electric double button carbon microphone.jpg|thumb|[[Western Electric]] double button carbon microphone]]
The [[carbon microphone]] was the earliest type of microphone. The carbon button microphone (or sometimes just a button microphone), uses a capsule or button containing carbon granules pressed between two metal plates like the [[Emile Berliner|Berliner]] and [[Thomas Edison|Edison]] microphones. A voltage is applied across the metal plates, causing a small current to flow through the carbon. One of the plates, the diaphragm, vibrates in sympathy with incident sound waves, applying a varying pressure to the carbon. The changing pressure deforms the granules, causing the contact area between each pair of adjacent granules to change, and this causes the electrical resistance of the mass of granules to change. The changes in resistance cause a corresponding change in the current flowing through the microphone, producing the electrical signal. Carbon microphones were once commonly used in telephones; they have extremely low-quality sound reproduction and a very limited frequency response range but are very robust devices. The Boudet microphone, which used relatively large carbon balls, was similar to the granule carbon button microphones.<ref>{{cite web
|url=http://www.machine-history.com/Boudet%20Microphone
|title=Boudet's Microphone
|publisher=Machine-History.com
|access-date=2009-12-09
|archive-url=https://web.archive.org/web/20150822100052/http://www.machine-history.com/Boudet%20Microphone
|archive-date=2015-08-22
|url-status=dead
}}</ref>

Unlike other microphone types, the carbon microphone can also be used as a type of amplifier, using a small amount of sound energy to control a larger amount of electrical energy. Carbon microphones found use as early [[repeater|telephone repeaters]], making long-distance phone calls possible in the era before vacuum tubes. Called a Brown's relay,{{cn|reason=Apprently for [[Sidney Brown]] but I can't find a supporting ref that indicates he worked on this problem or that it was given his name.|date=January 2020}} these repeaters worked by mechanically coupling a magnetic telephone receiver to a carbon microphone: the faint signal from the receiver was transferred to the microphone, where it modulated a stronger electric current, producing a stronger electrical signal to send down the line. One illustration of this amplifier effect was the oscillation caused by feedback, resulting in an audible squeal from the old "candlestick" telephone if its earphone was placed near the carbon microphone.

{{Anchor|Piezoelectric microphone}}

=== Piezoelectric ===
[[Image:Astatic crystal mic.jpg|thumb|Vintage [[Astatic Corporation|Astatic]] crystal microphone]]
A '''crystal microphone''' or '''piezo microphone'''<ref>{{cite journal|last1=Seung S. Lee |first1=Woon Seob Lee |title=Piezoelectric microphone built on circular diaphragm |journal=Sensors and Actuators A |volume=144 |issue=2 |year=2008 |pages=367–373 |url=http://www.pitt.edu/~qiw4/Academic/ME2080/ZnO%20circular%20microphone.pdf |accessdate=3 March 2017 |url-status=bot: unknown |archiveurl=https://web.archive.org/web/20130717185137/http://www.pitt.edu/~qiw4/Academic/ME2080/ZnO%20circular%20microphone.pdf |archivedate=17 July 2013 |doi=10.1016/j.sna.2008.02.001 }}</ref> uses the phenomenon of [[piezoelectricity]]—the ability of some materials to produce a voltage when subjected to pressure—to convert vibrations into an electrical signal. An example of this is [[potassium sodium tartrate]], which is a piezoelectric crystal that works as a transducer, both as a microphone and as a slimline loudspeaker component. Crystal microphones were once commonly supplied with [[vacuum tube]] (valve) equipment, such as domestic tape recorders. Their high output impedance matched the high input impedance (typically about 10 [[Ohm|megohms]]) of the vacuum tube input stage well. They were difficult to match to early [[transistor]] equipment and were quickly supplanted by dynamic microphones for a time, and later small electret condenser devices. The high impedance of the crystal microphone made it very susceptible to handling noise, both from the microphone itself and from the connecting cable.

Piezoelectric transducers are often used as [[contact microphone]]s to amplify sound from acoustic musical instruments, to sense drum hits, for triggering electronic samples, and to record sound in challenging environments, such as underwater under high pressure. [[Pick up (music technology)#Piezoelectric pickups|Saddle-mounted pickups]] on [[acoustic guitar]]s are generally piezoelectric devices that contact the strings passing over the saddle. This type of microphone is different from [[Pick up (music technology)#Magnetic pickups|magnetic coil pickups]] commonly visible on typical [[electric guitar]]s, which use magnetic induction, rather than mechanical coupling, to pick up vibration.

=== Fiber-optic ===
[[Image:Optimic1140 fiber optical microphone for wiki.jpg|thumb|right|The [[Optoacoustics Ltd|Optoacoustics]] 1140 fiber-optic microphone]]

A [[optical fiber|fiber-optic]] microphone converts acoustic waves into electrical signals by sensing changes in light intensity, instead of sensing changes in capacitance or magnetic fields as with conventional microphones.<ref>{{cite journal |first=Alexander |last=Paritsky |author2=Kots, A. |title=Fiber optic microphone as a realization of fiber optic positioning sensors |journal=Proc. Of International Society for Optical Engineering (SPIE) |volume= 3110 |year=1997 |pages=408–409 |url=http://proceedings.spiedigitallibrary.org/proceeding.aspx?articleid=928593 |doi=10.1117/12.281371|series=10th Meeting on Optical Engineering in Israel |bibcode=1997SPIE.3110..408P }}</ref><ref>{{US patent reference | number = 6462808 | y = 2002 | m = 10 | d = 08 | inventor = Alexander Paritsky and Alexander Kots | title = Small optical microphone/sensor}}</ref>

During operation, light from a laser source travels through an optical fiber to illuminate the surface of a reflective diaphragm. Sound vibrations of the diaphragm modulate the intensity of light reflecting off the diaphragm in a specific direction. The modulated light is then transmitted over a second optical fiber to a photodetector, which transforms the intensity-modulated light into analog or digital audio for transmission or recording. Fiber-optic microphones possess high dynamic and frequency range, similar to the best high fidelity conventional microphones.

Fiber-optic microphones do not react to or influence any electrical, magnetic, electrostatic or radioactive fields (this is called [[Electromagnetic interference|EMI/RFI]] immunity). The fiber-optic microphone design is therefore ideal for use in areas where conventional microphones are ineffective or dangerous, such as inside [[Gas turbine#Industrial gas turbines for electrical generation|industrial turbines]] or in [[magnetic resonance imaging]] (MRI) equipment environments.

Fiber-optic microphones are robust, resistant to environmental changes in heat and moisture, and can be produced for any directionality or [[impedance matching]]. The distance between the microphone's light source and its photodetector may be up to several kilometers without need for any preamplifier or another electrical device, making fiber-optic microphones suitable for industrial and surveillance acoustic monitoring.

Fiber-optic microphones are used in very specific application areas such as for [[infrasound]] monitoring and [[Noise-canceling microphone|noise-canceling]]. They have proven especially useful in medical applications, such as allowing radiologists, staff and patients within the powerful and noisy magnetic field to converse normally, inside the MRI suites as well as in remote control rooms.<ref>{{cite web|url=http://www.rt-image.com/Case_Study_Can_You_Hear_Me_Now_Technology_for_better_communication_in_the_MRI_su/content=9004J05E48B6A686407698724488A0441 |title=Case Study: Can You Hear Me Now? |work=rt-image.com |publisher=Valley Forge Publishing |first=Susan |last=Karlin |archiveurl=https://web.archive.org/web/20110715212557/http://www.rt-image.com/Case_Study_Can_You_Hear_Me_Now_Technology_for_better_communication_in_the_MRI_su/content%3D9004J05E48B6A686407698724488A0441 |archivedate=2011-07-15 |url-status=dead }}</ref> Other uses include industrial equipment monitoring and audio calibration and measurement, high-fidelity recording and law enforcement.<ref>{{cite web|last1=Goulde|first1=Berg|title=Microphones For Computer|url=https://microphonetopgear.com/microphones-for-computer/|website=Microphone top gear|accessdate=3 March 2017}}</ref>

=== Laser ===
{{Main|Laser microphone}}

[[Laser microphone]]s are often portrayed in movies as spy gadgets because they can be used to pick up sound at a distance from the microphone equipment. A laser beam is aimed at the surface of a window or other plane surface that is affected by sound. The vibrations of this surface change the angle at which the beam is reflected, and the motion of the laser spot from the returning beam is detected and converted to an audio signal.

In a more robust and expensive implementation, the returned light is split and fed to an [[interferometer]], which detects movement of the surface by changes in the [[optical path length]] of the reflected beam. The former implementation is a tabletop experiment; the latter requires an extremely stable laser and precise optics.

A new type of laser microphone is a device that uses a laser beam and smoke or vapor to detect [[sound]] [[vibration]]s in free air. On 25 August 2009, U.S. patent 7,580,533 issued for a Particulate Flow Detection Microphone based on a laser-photocell pair with a moving stream of smoke or vapor in the laser beam's path. Sound pressure waves cause disturbances in the smoke that in turn cause variations in the amount of laser light reaching the photodetector. A prototype of the device was demonstrated at the 127th Audio Engineering Society convention in New York City from 9 through 12 October 2009.

=== Liquid ===
{{Main|Water microphone}}

Early microphones did not produce intelligible speech, until [[Alexander Graham Bell]] made improvements including a variable-resistance microphone/transmitter. Bell's liquid transmitter consisted of a metal cup filled with water with a small amount of sulfuric acid added. A sound wave caused the diaphragm to move, forcing a needle to move up and down in the water. The electrical resistance between the wire and the cup was then inversely proportional to the size of the water meniscus around the submerged needle. [[Elisha Gray]] filed a [[Patent caveat|caveat]] for a version using a brass rod instead of the needle.{{when|date=February 2019}} Other minor variations and improvements were made to the liquid microphone by Majoranna, Chambers, Vanni, Sykes, and Elisha Gray, and one version was patented by [[Reginald Fessenden]] in 1903. These were the first working microphones, but they were not practical for commercial application. The famous first phone conversation between Bell and Watson took place using a liquid microphone.

=== MEMS ===
{{main|Microelectromechanical systems}}

The [[Microelectromechanical systems|MEMS]] (MicroElectrical-Mechanical System) microphone is also called a microphone chip or silicon microphone. A pressure-sensitive diaphragm is etched directly into a silicon wafer by MEMS processing techniques and is usually accompanied with an integrated preamplifier. Most MEMS microphones are variants of the condenser microphone design. Digital MEMS microphones have built-in [[analog-to-digital converter]] (ADC) circuits on the same CMOS chip making the chip a digital microphone and so more readily integrated with modern digital products. Major manufacturers producing MEMS silicon microphones are Wolfson Microelectronics (WM7xxx) now Cirrus Logic,<ref>{{cite web |url=http://www.marketwatch.com/story/cirrus-logic-completes-acquisition-of-wolfson-microelectronics-2014-08-21|title=Cirrus Logic Completes Acquisition of Wolfson Microelectronics |publisher=MarketWatch.com |accessdate=2014-08-21 }}</ref> InvenSense (product line sold by Analog Devices <ref>{{cite web |url=http://www.analog.com/en/about-adi/news-room/press-releases/2013/10_14_13_adi_to_sell_microphone_product_line_to_in.html|title=Analog Devices To Sell Microphone Product Line To InvenSense |publisher=MarketWatch.com |accessdate=2015-11-27 }}</ref>), Akustica (AKU200x), Infineon (SMM310 product), Knowles Electronics, Memstech (MSMx), NXP Semiconductors (division bought by Knowles <ref>{{cite web |url=http://investor.knowles.com/phoenix.zhtml?c=252194&p=irol-newsArticle&ID=1884042|title=Knowles Completes Acquisition of NXP's Sound Solutions Business |publisher=Knowles |accessdate=2011-07-05 }}</ref>), Sonion MEMS, Vesper, AAC Acoustic Technologies,<ref>{{cite web |url=http://seekingalpha.com/article/157790-mems-microphone-will-be-hurt-by-downturn-in-smartphone-market |title=MEMS Microphone Will Be Hurt by Downturn in Smartphone Market |publisher=[[Seeking Alpha]] |accessdate=2009-08-23 }}</ref> and Omron.<ref>{{cite web |url=http://www.omron.com/media/press/2009/11/c1125.html |title=OMRON to Launch Mass-production and Supply of MEMS Acoustic Sensor Chip -World's first MEMS sensor capable of detecting the lower limit of human audible frequencies- |accessdate=2009-11-24 }}</ref>

More recently, since the 2010s, there has been increased interest and research into making piezoelectric MEMS microphones which are a significant architectural and material change from existing condenser style MEMS designs.<ref>{{cite web|url=http://www.eetimes.com/document.asp?doc_id=1324827|title=MEMS Mics Taking Over|work=EETimes}}</ref>

=== Speakers as microphones ===
A [[loudspeaker]], a transducer that turns an electrical signal into sound waves, is the functional opposite of a microphone. Since a conventional speaker is similar in construction to a dynamic microphone (with a diaphragm, coil and magnet), speakers can actually work "in reverse" as microphones. [[Reciprocity (engineering)|reciprocity]] applies, so the resulting microphone has the same impairments as a single-driver loudspeaker: limited low- and high-end frequency response, poorly-controlled directivity, and low [[sensitivity (electronics)|sensitivity]]. In practical use, speakers are sometimes used as microphones in applications where high bandwidth and sensitivity are not needed such as [[intercom]]s, [[walkie-talkie]]s or [[Voice chat#Voice chat in gaming|video game voice chat]] peripherals, or when conventional microphones are in short supply.

However, there is at least one practical application that exploits those weaknesses: the use of a medium-size [[woofer]] placed closely in front of a "kick drum" ([[bass drum]]) in a [[drum set]] to act as a microphone. A commercial product example is the Yamaha Subkick, a {{convert|6.5|in|adj=on}} woofer shock-mounted into a 10" drum shell used in front of kick drums. Since a relatively massive membrane is unable to transduce high frequencies while being capable of tolerating strong low-frequency transients, the speaker is often ideal for picking up the kick drum while reducing bleed from the nearby cymbals and snare drums.<ref>{{cite magazine |url=http://recordinghacks.com/reviews/tapeop/yamaha-subkick/ |title=Yamaha SubKick – The Tape Op Review |author=Larry Crane |magazine=[[Tape Op]] |date=July 2004 |access-date=2020-06-12}}</ref>

Less commonly, microphones themselves can be used as speakers, but due to their low power handling and small transducer sizes, a [[tweeter]] is the most practical application. One instance of such an application was the [[Standard Telephones and Cables|STC]] microphone-derived 4001 super-tweeter, which was successfully used in a number of high-quality loudspeaker systems from the late 1960s to the mid-70s.

== Capsule design and directivity ==
The inner elements of a microphone are the primary source of differences in directivity. A pressure microphone uses a [[Diaphragm (mechanical device)|diaphragm]] between a fixed internal volume of air and the environment and responds uniformly to pressure from all directions, so it is said to be omnidirectional. A pressure-gradient microphone uses a diaphragm that is at least partially open on both sides. The pressure difference between the two sides produces its directional characteristics. Other elements such as the external shape of the microphone and external devices such as interference tubes can also alter a microphone's directional response. A pure pressure-gradient microphone is equally sensitive to sounds arriving from front or back but insensitive to sounds arriving from the side because sound arriving at the front and back at the same time creates no gradient between the two. The characteristic directional pattern of a pure pressure-gradient microphone is like a figure-8. Other polar patterns are derived by creating a capsule that combines these two effects in different ways. The cardioid, for instance, features a partially closed backside, so its response is a combination of pressure and pressure-gradient characteristics.<ref>{{cite web | last = Bartlett | first = Bruce | title = How A Cardioid Microphone Works | url = http://www.prosoundweb.com/article/church_sound_how_a_cardioid_microphone_really_works/ }}</ref>

{{anchor|Microphone polar patterns|patterns}}

== Polar patterns ==
Microphone polar sensitivity. Microphone is facing towards the top of the page in diagram, parallel to the page.<ref>{{cite web |url=http://microphonegeeks.com/different-microphone-polar-patterns/ |title=Understanding different microphone polar patterns |date=March 28, 2015 |access-date=2020-04-04}}</ref>
<gallery widths="100px" heights="100px">
Image:Polar pattern omnidirectional.svg|<center>Omnidirectional</center>
Image:Polar pattern figure eight.svg|<center>Bi-directional or Figure of 8</center>
Image:Polar pattern subcardioid.svg|<center>Subcardioid</center>
Image:Polar pattern cardioid.svg|<center>[[Cardioid]]</center>
Image:Polar pattern hypercardioid.svg|<center>Hypercardioid</center>
Image:Polar pattern supercardioid.svg|<center>Supercardioid</center>
Image:Polar pattern directional.svg|<center>Lobar</center>
</gallery>

A microphone's directionality or polar pattern indicates how sensitive it is to sounds arriving at different angles about its central axis. The polar patterns illustrated above represent the [[locus (mathematics)|locus]] of points that produce the same signal level output in the microphone if a given [[sound pressure level]] (SPL) is generated from that point. How the physical body of the microphone is oriented relative to the diagrams depends on the microphone design. For large-membrane microphones such as in the Oktava (pictured above), the upward direction in the polar diagram is usually [[perpendicular]] to the microphone body, commonly known as "side fire" or "side address". For small diaphragm microphones such as the Shure (also pictured above), it usually extends from the axis of the microphone commonly known as "end fire" or "top/end address".

Some microphone designs combine several principles in creating the desired polar pattern. This ranges from shielding (meaning diffraction/dissipation/absorption) by the housing itself to electronically combining dual membranes.

===Omnidirectional===
An '''omnidirectional''' (or nondirectional) microphone's response is generally considered to be a perfect sphere in three dimensions. In the real world, this is not the case. As with directional microphones, the polar pattern for an "omnidirectional" microphone is a function of frequency. The body of the microphone is not infinitely small and, as a consequence, it tends to get in its own way with respect to sounds arriving from the rear, causing a slight flattening of the polar response. This flattening increases as the diameter of the microphone (assuming it's cylindrical) reaches the wavelength of the frequency in question. Therefore, the smallest diameter microphone gives the best omnidirectional characteristics at high frequencies.

The wavelength of sound at 10 kHz is 1.4" (3.5 cm). The smallest measuring microphones are often 1/4" (6 mm) in diameter, which practically eliminates directionality even up to the highest frequencies. Omnidirectional microphones, unlike cardioids, do not employ resonant cavities as delays, and so can be considered the "purest" microphones in terms of low coloration; they add very little to the original sound. Being pressure-sensitive they can also have a very flat low-frequency response down to 20 Hz or below. Pressure-sensitive microphones also respond much less to wind noise and plosives than directional (velocity sensitive) microphones.

Areas of application: studios, old churches, theatres, on-site TV interviews, etc.<ref>[https://micspeech.com/types-of-microphones/ Types of microphones.] Micspeech.</ref>

An example of a nondirectional microphone is the round black ''eight ball''.<ref>[http://lloydmicrophoneclassics.com/mic_history.html History & Development of Microphone.] {{Webarchive|url=https://web.archive.org/web/20080704172711/http://lloydmicrophoneclassics.com/mic_history.html |date=2008-07-04 }} Lloyd Microphone Classics.</ref>

===Unidirectional===
A unidirectional microphone is primarily sensitive to sounds from only one direction. [[#Microphone polar patterns|The diagram above]] (shotgun) illustrates a number of these patterns. The microphone faces upwards in each diagram. The sound intensity for a particular frequency is plotted for angles radially from 0 to 360°. (Professional diagrams show these scales and include multiple plots at different frequencies. The diagrams given here provide only an overview of typical pattern shapes, and their names.)

{{Anchor|Cardioid}}

===Cardioid, hypercardioid, supercardioid, subcardioid===
[[Image:Us664a microphone.jpg|thumb|right|University Sound US664A dynamic supercardioid microphone]]

The most common unidirectional microphone is a '''cardioid''' microphone, so named because the sensitivity pattern is "heart-shaped", i.e. a [[cardioid]]. The cardioid family of microphones are commonly used as vocal or speech microphones since they are good at rejecting sounds from other directions. In three dimensions, the cardioid is shaped like an apple centred around the microphone, which is the "stem" of the apple. The cardioid response reduces pickup from the side and rear, helping to avoid feedback from the [[Foldback (sound engineering)|monitors]]. Since these directional [[transducer]] microphones achieve their patterns by sensing pressure gradient, putting them very close to the sound source (at distances of a few centimeters) results in a bass boost due to the increased gradient. This is known as the [[Proximity effect (audio)|proximity effect]].<ref>[http://www.tonmeister.ca/main/textbook/node473.html Proximity Effect.] {{Webarchive|url=https://web.archive.org/web/20071016054151/http://www.tonmeister.ca/main/textbook/node473.html |date=2007-10-16 }} Geoff Martin, ''Introduction to Sound Recording''.</ref> The [[SM58]] has been the most commonly used microphone for live vocals for more than 50 years<ref>{{cite web|url=http://www.shure.com/americas/about-shure/history/index.htm |title=History – The evolution of an audio revolution |publisher=[[Shure]] |accessdate=2013-07-30}}</ref> demonstrating the importance and popularity of cardioid mics.

The cardioid is effectively a superposition of an omnidirectional (pressure) and a figure-8 (pressure gradient) microphone;<ref>{{Cite book|url=https://books.google.com/books?id=KApvYcQkY_gC|title=Eargle's The Microphone Book: From Mono to Stereo to Surround – A Guide to Microphone Design and Application|last=Rayburn|first=Ray A.|date=2012-11-12|publisher=Taylor & Francis|isbn=9781136118135|language=en}}</ref> for sound waves coming from the back, the negative signal from the figure-8 cancels the positive signal from the omnidirectional element, whereas, for sound waves coming from the front, the two add to each other.

By combining the two components in different ratios, any pattern between omni and figure-8 can be achieved, which comprise the first-order cardioid family. Common shapes include:

* A '''hyper-cardioid''' microphone is similar to cardioid, but with a slightly larger figure-8 contribution, leading to a tighter area of front sensitivity and a smaller lobe of rear sensitivity. It is produced by combining the two components in a 3:1 ratio, producing nulls at 109.5°. This ratio maximizes the [[directivity factor]] (or directivity index).<ref name=":0">{{Cite journal|title=On the Design and Implementation of Higher Order Differential Microphones – IEEE Journals & Magazine|issue=1|pages=162–174|journal=IEEE Transactions on Audio, Speech, and Language Processing|volume=20|language=en-US|doi=10.1109/TASL.2011.2159204|date=January 2012|last1=Sena|first1=E. De|last2=Hacihabiboglu|first2=H.|last3=Cvetkovic|first3=Z.}}</ref><ref name=":1">{{Cite book|url=https://books.google.com/books?id=tym4kKVPfaAC|title=Study and Design of Differential Microphone Arrays|last=Benesty|first=Jacob|last2=Jingdong|first2=Chen|date=2012-10-23|publisher=Springer Science & Business Media|isbn=9783642337529|language=en}}</ref>
* A '''super-cardioid''' microphone is similar to a hyper-cardioid, except there is more front pickup and less rear pickup. It is produced with about a 5:3 ratio, with nulls at 126.9°. This ratio maximizes the ''front-back ratio''; the energy ratio between front and rear radiation.<ref name=":0" /><ref name=":1" />
* The '''sub-cardioid''' microphone has no null points. It is produced with about 7:3 ratio with 3–10 dB level between the front and back pickup.<ref>{{cite web|url=http://www.uaudio.com/webzine/2005/december/text/content2.html |title=Ask the Doctors: The Physics of Mid-Side (MS) Miking |author=Dave Berners |publisher=[[Universal Audio]] |work=Universal Audio WebZine |date=December 2005 |accessdate=2013-07-30}}</ref><ref>{{cite web|url=http://hyperphysics.phy-astr.gsu.edu/hbase/audio/mic3.html#c2 |title=Directional Patterns of Microphones |accessdate=2013-07-30}}</ref>

===Bi-directional===
"Figure 8" or bi-directional microphones receive sound equally from both the front and back of the element. Most ribbon microphones are of this pattern. In principle they do not respond to sound pressure at all, only to the ''change'' in pressure between front and back; since sound arriving from the side reaches front and back equally there is no difference in pressure and therefore no sensitivity to sound from that direction. In more mathematical terms, while omnidirectional microphones are [[Scalar (physics)|scalar]] transducers responding to pressure from any direction, bi-directional microphones are [[Gradient vector|vector]] transducers responding to the gradient along an axis normal to the plane of the diaphragm. This also has the effect of inverting the output polarity for sounds arriving from the back side.

{{anchor|Shotgun and parabolic microphones|shotgun}}

===Shotgun===
[[Image:shotgun microphone.jpg|thumb|right|An Audio-Technica shotgun microphone]]
[[File:Interference Tube.jpg|thumb|right|The interference tube of a shotgun microphone. The capsule is at the base of the tube.]]

'''Shotgun microphones''' are the most highly directional of simple first-order unidirectional types. At low frequencies, they have the classic polar response of a hypercardioid but at medium and higher frequencies an interference tube gives them an increased forward response. This is achieved by a process of cancellation of off-axis waves entering the longitudinal array of slots. A consequence of this technique is the presence of some rear lobes that vary in level and angle with frequency and can cause some coloration effects. Due to the narrowness of their forward sensitivity, shotgun microphones are commonly used on television and film sets, in stadiums, and for field recording of wildlife.

===Boundary or "PZM"===
Several approaches have been developed for effectively using a microphone in less-than-ideal acoustic spaces, which often suffer from excessive reflections from one or more of the surfaces (boundaries) that make up the space. If the microphone is placed in, or very close to, one of these boundaries, the reflections from that surface have the same timing as the direct sound, thus giving the microphone a hemispherical polar pattern and improved intelligibility. Initially, this was done by placing an ordinary microphone adjacent to the surface, sometimes in a block of acoustically transparent foam. Sound engineers Ed Long and Ron Wickersham developed the concept of placing the diaphragm parallel to and facing the boundary.<ref>({{Cite patent|US|4361736}})</ref> While the patent has expired, "Pressure Zone Microphone" and "PZM" are still active trademarks of [[Crown International]], and the generic term [[boundary microphone]] is preferred. While a boundary microphone was initially implemented using an omnidirectional element, it is also possible to mount a directional microphone close enough to the surface to gain some of the benefits of this technique while retaining the directional properties of the element. Crown's trademark on this approach is "Phase Coherent Cardioid" or "PCC," but there are other makers who employ this technique as well.

== Application-specific designs ==
A [[lavalier microphone]] is made for hands-free operation. These small microphones are worn on the body. Originally, they were held in place with a lanyard worn around the neck, but more often they are fastened to clothing with a clip, pin, tape or magnet. The lavalier cord may be hidden by clothes and either run to an RF transmitter in a pocket or clipped to a belt (for mobile use), or run directly to the mixer (for stationary applications).

A [[wireless microphone]] transmits the audio as a radio or optical signal rather than via a cable. It usually sends its signal using a small FM radio transmitter to a nearby receiver connected to the sound system, but it can also use infrared waves if the transmitter and receiver are within sight of each other.

A [[contact microphone]] picks up vibrations directly from a solid surface or object, as opposed to sound vibrations carried through air. One use for this is to detect sounds of a very low level, such as those from small objects or [[insect]]s. The microphone commonly consists of a magnetic (moving coil) transducer, contact plate and contact pin. The contact plate is placed directly on the vibrating part of a musical instrument or other surface, and the contact pin transfers vibrations to the coil. Contact microphones have been used to pick up the sound of a snail's heartbeat and the footsteps of ants. A portable version of this microphone has recently been developed. A [[throat microphone]] is a variant of the contact microphone that picks up speech directly from a person's throat, which it is strapped to. This lets the device be used in areas with ambient sounds that would otherwise make the speaker inaudible.

[[File:Sony parabolic reflector.jpg|thumb|upright|A Sony parabolic reflector, without a microphone. The microphone would face the reflector surface and sound captured by the reflector would bounce towards the microphone.]]
A [[parabolic microphone]] uses a [[parabolic reflector]] to collect and focus sound waves onto a microphone receiver, in much the same way that a [[parabolic antenna]] (e.g. [[satellite dish]]) does with radio waves. Typical uses of this microphone, which has unusually focused front sensitivity and can pick up sounds from many meters away, include nature recording, outdoor sporting events, [[eavesdropping]], [[Police|law enforcement]], and even [[espionage]]. Parabolic microphones are not typically used for standard recording applications, because they tend to have a poor low-frequency response as a side effect of their design.

A stereo microphone integrates two microphones in one unit to produce a stereophonic signal. A stereo microphone is often used for [[broadcast]] applications or [[field recording]] where it would be impractical to configure two separate condenser microphones in a classic X-Y configuration (see [[microphone practice]]) for stereophonic recording. Some such microphones have an adjustable angle of coverage between the two channels.

A [[noise-canceling microphone]] is a highly directional design intended for noisy environments. One such use is in [[aircraft]] cockpits where they are normally installed as boom microphones on headsets. Another use is in [[live event support]] on loud concert stages for vocalists involved with [[Concert|live performances]]. Many noise-canceling microphones combine signals received from two diaphragms that are in opposite electrical polarity or are processed electronically. In dual diaphragm designs, the main diaphragm is mounted closest to the intended source and the second is positioned farther away from the source so that it can pick up environmental sounds to be subtracted from the main diaphragm's signal. After the two signals have been combined, sounds other than the intended source are greatly reduced, substantially increasing intelligibility. Other noise-canceling designs use one diaphragm that is affected by ports open to the sides and rear of the microphone, with the sum being a 16 dB rejection of sounds that are farther away. One noise-canceling headset design using a single diaphragm has been used prominently by vocal artists such as [[Garth Brooks]] and [[Janet Jackson]].<ref>[http://www.crownaudio.com/pdf/mics/136368.pdf Crown Audio. Tech Made Simple. ''The Crown Differoid Microphone''] {{webarchive |url=https://web.archive.org/web/20120510070314/http://www.crownaudio.com/pdf/mics/136368.pdf |date=May 10, 2012 }}</ref> A few noise-canceling microphones are throat microphones.

== Stereo microphone techniques ==
{{main|Microphone practice}}
Various standard techniques are used with microphones used in [[sound reinforcement]] at live performances, or for recording in a studio or on a motion picture set. By suitable arrangement of one or more microphones, desirable features of the sound to be collected can be kept, while rejecting unwanted sounds.

== Powering ==
Microphones containing active circuitry, such as most condenser microphones, require power to operate the active components. The first of these used vacuum-tube circuits with a separate power supply unit, using a multi-pin cable and connector. With the advent of solid-state amplification, the power requirements were greatly reduced and it became practical to use the same cable conductors and connector for audio and power. During the 1960s several powering methods were developed, mainly in Europe. The two dominant methods were initially defined in German DIN 45595 as [[:de:Tonaderspeisung]] or T-power and DIN 45596 for [[phantom power]]. Since the 1980s, phantom power has become much more common, because the same input may be used for both powered and unpowered microphones. In consumer electronics such as DSLRs and camcorders, "plug-in power" is more common, for microphones using a 3.5 mm phone plug connector. Phantom, T-power and plug-in power are described in international standard IEC 61938.<ref>{{cite journal|title=Multimedia systems – Guide to the recommended characteristics of analogue interfaces to achieve interoperability|journal=Webstore.iec.ch|volume=IEC 61938:2013|url=https://webstore.iec.ch/publication/6142|accessdate=3 March 2017|language=en}}</ref>

== Connectors ==
[[File:Yeti-USB-Microphone.jpg|thumb|upright|A [[Blue Microphones|Blue]] Yeti with a USB connector (not visible)]]

The most common connectors used by microphones are:
*Male [[XLR connector]] on professional microphones
*¼ inch (sometimes referred to as 6.35 mm) [[Phone connector (audio)|phone connector]] on less expensive musician's microphones, using an unbalanced 1/4 inch (6.3 mm) TS (tip and sleeve) phone connector. Harmonica microphones commonly use a high impedance 1/4 inch (6.3 mm) TS connection to be run through guitar amplifiers.
*[[Phone connector (audio)#Miniature size|3.5 mm]] (sometimes referred to as 1/8 inch mini) TRS (tip, ring and sleeve) stereo (also available as TS mono) mini phone plug on prosumer camera, recorder and computer microphones.
*[[USB]] allows direct connection to PCs. Electronics in these microphones powered over the USB connection performs preamplification and ADC before the digital audio data is transferred via the USB interface.

Some microphones use other connectors, such as a 5-pin XLR, or mini XLR for connection to portable equipment. Some lavalier (or "lapel", from the days of attaching the microphone to the news reporter's suit lapel) microphones use a proprietary connector for connection to a wireless transmitter, such as a [[radio pack]]. Since 2005, professional-quality microphones with USB connections have begun to appear, designed for direct recording into computer-based software.

=== Impedance-matching ===
Microphones have an electrical characteristic called [[electrical impedance|impedance]], measured in [[ohm]]s (Ω), that depends on the design. In passive microphones, this value relates to the impedance of the coil (or similar mechanism). In active microphones, this value describes the load impedance for which its amplifier circuitry is designed. Typically, the ''rated impedance'' is stated.<ref name="autogenerated1">International Standard IEC 60268-4</ref> Low impedance is considered under 600 Ω. Medium impedance is considered between 600 Ω and 10 kΩ. High impedance is above 10 kΩ. Owing to their built-in [[Electronic amplifier|amplifier]], condenser microphones typically have an output impedance between 50 and 200 Ω.<ref name=Eargle2002>{{cite book |title=Audio Engineering for Sound Reinforcement |last=Eargle |first=John |authorlink=John M. Eargle |author2=Chris Foreman |year=2002 |publisher=Hal Leonard Corporation |location=Milwaukee |isbn=978-0-634-04355-0 |page=66 |url=https://books.google.com/?id=YWzZe6z4xdAC }}</ref>

If a microphone is made in high and low impedance versions, the high impedance version has a higher output voltage for a given sound pressure input, and is suitable for use with vacuum-tube guitar amplifiers, for instance, which have a high input impedance and require a relatively high signal input voltage to overcome the tubes' inherent noise. Most professional microphones are low impedance, about 200 Ω or lower. Professional vacuum-tube sound equipment incorporates a [[transformer]] that steps up the impedance of the microphone circuit to the high impedance and voltage needed to drive the input tube. External matching transformers are also available that can be used in-line between a low impedance microphone and a high impedance input.

Low-impedance microphones are preferred over high impedance for two reasons: one is that using a high-impedance microphone with a long cable results in high-frequency signal loss due to cable capacitance, which forms a low-pass filter with the microphone output impedance{{citation needed|reason=Unless there is an IEEE specification that states specific audio input and output impedances for all equipment, this is an over-reaching statement that might not always (or frequently) be true.|date=September 2015}}. The other is that long high-impedance cables tend to pick up more [[mains hum|hum]] (and possibly [[Radio frequency interference|radio-frequency interference]] (RFI) as well). Nothing is damaged if the impedance between microphone and other equipment is mismatched; the worst that happens is a reduction in signal or change in frequency response.

Some microphones are designed ''not'' to have their impedance matched by the load they are connected to.<ref>[http://www.shure.com/ProAudio/Products/us_pro_ea_imepdance] {{webarchive|url=https://web.archive.org/web/20100428071840/http://www.shure.com/ProAudio/Products/us_pro_ea_imepdance|date=April 28, 2010}}</ref> Doing so can alter their frequency response and cause distortion, especially at high sound pressure levels. Certain ribbon and dynamic microphones are exceptions, due to the designers' assumption of a certain load impedance being part of the internal electro-acoustical damping circuit of the microphone.<ref>Robertson, A. E.: "Microphones" Illiffe Press for BBC, 1951–1963</ref>{{Dubious|date=April 2010}}

=== Digital microphone interface ===
[[File:Neumann D-01 IBC 2008.jpg|thumb|150px|Neumann D-01 digital microphone and Neumann DMI-8 8-channel USB Digital Microphone Interface]]
The [[AES42]] standard, published by the [[Audio Engineering Society]], defines a digital interface for microphones. Microphones conforming to this standard directly output a digital audio stream through an XLR or [[XLR connector#XLD keyed variant|XLD]] male connector, rather than producing an analog output. Digital microphones may be used either with new equipment with appropriate input connections that conform to the AES42 standard, or else via a suitable interface box. Studio-quality microphones that operate in accordance with the AES42 standard are now available from a number of microphone manufacturers.

== Measurements and specifications ==
[[Image:Oktava319vsshuresm58.png|thumb|left|A comparison of the far field on-axis frequency response of the Oktava 319 and the [[SM58|Shure SM58]]]]

Because of differences in their construction, microphones have their own characteristic responses to sound. This difference in response produces non-uniform [[phase (waves)|phase]] and [[frequency]] responses. In addition, microphones are not uniformly sensitive to sound pressure and can accept differing levels without distorting. Although for scientific applications microphones with a more uniform response are desirable, this is often not the case for music recording, as the non-uniform response of a microphone can produce a desirable coloration of the sound. There is an international standard for microphone specifications,<ref name="autogenerated1" /> but few manufacturers adhere to it. As a result, comparison of published data from different manufacturers is difficult because different measurement techniques are used. The Microphone Data Website has collated the technical specifications complete with pictures, response curves and technical data from the microphone manufacturers for every currently listed microphone, and even a few obsolete models, and shows the data for them all in one common format for ease of comparison.[https://web.archive.org/web/20070210000128/http://www.microphone-data.com/]. Caution should be used in drawing any solid conclusions from this or any other published data, however, unless it is known that the manufacturer has supplied specifications in accordance with IEC 60268-4.

A [[frequency response]] diagram plots the microphone sensitivity in [[decibel]]s over a range of frequencies (typically 20 Hz to 20 kHz), generally for perfectly on-axis sound (sound arriving at 0° to the capsule). Frequency response may be less informatively stated textually like so: "30 Hz–16 kHz ±3 dB". This is interpreted as meaning a nearly flat, linear, plot between the stated frequencies, with variations in amplitude of no more than plus or minus 3 dB. However, one cannot determine from this information how ''smooth'' the variations are, nor in what parts of the spectrum they occur. Note that commonly made statements such as "20 Hz–20 kHz" are meaningless without a decibel measure of tolerance. Directional microphones' frequency response varies greatly with distance from the sound source, and with the geometry of the sound source. IEC 60268-4 specifies that frequency response should be measured in ''plane progressive wave'' conditions (very far away from the source) but this is seldom practical. ''Close talking'' microphones may be measured with different sound sources and distances, but there is no standard and therefore no way to compare data from different models unless the measurement technique is described.

The self-noise or [[equivalent input]] noise level is the sound level that creates the same output voltage as the microphone does in the absence of sound. This represents the lowest point of the microphone's dynamic range, and is particularly important should you wish to record sounds that are quiet. The measure is often stated in [[dB(A)]], which is the equivalent loudness of the noise on a decibel scale frequency-weighted for how the ear hears, for example: "15 dBA SPL" (SPL means [[sound pressure]] level relative to 20 [[micropascal]]s). The lower the number the better. Some microphone manufacturers state the noise level using [[ITU-R 468 noise weighting]], which more accurately represents the way we hear noise, but gives a figure some 11–14 dB higher. A quiet microphone typically measures 20 dBA SPL or 32 dB SPL 468-weighted. Very quiet microphones have existed for years for special applications, such the Brüel & Kjaer 4179, with a noise level around 0 dB SPL. Recently some microphones with low noise specifications have been introduced in the studio/entertainment market, such as models from [[Georg Neumann GmbH|Neumann]] and [[Røde]] that advertise noise levels between 5–7 dBA. Typically this is achieved by altering the frequency response of the capsule and electronics to result in lower noise within the [[A-weighting]] curve while broadband noise may be increased.

The maximum SPL the microphone can accept is measured for particular values of [[total harmonic distortion]] (THD), typically 0.5%. This amount of distortion is generally inaudible,{{citation needed|date=February 2012}} so one can safely use the microphone at this SPL without harming the recording. Example: "142 [[dB SPL]] peak (at 0.5% THD)". The higher the value, the better, although microphones with a very high maximum SPL also have a higher self-noise.

The clipping level is an important indicator of maximum usable level, as the 1% THD figure usually quoted under max SPL is really a very mild level of distortion, quite inaudible especially on brief high peaks. Clipping is much more audible. For some microphones, the clipping level may be much higher than the max SPL.

The dynamic range of a microphone is the difference in SPL between the noise floor and the maximum SPL. If stated on its own, for example, "120 dB", it conveys significantly less information than having the self-noise and maximum SPL figures individually.

[[Sensitivity (electronics)|Sensitivity]] indicates how well the microphone converts acoustic pressure to an output voltage. A high sensitivity microphone creates more voltage and so needs less amplification at the mixer or recording device. This is a practical concern but is not directly an indication of the microphone's quality, and in fact the term sensitivity is something of a misnomer, "transduction gain" being perhaps more meaningful, (or just "output level") because true sensitivity is generally set by the [[noise floor]], and too much "sensitivity" in terms of output level compromises the clipping level. There are two common measures. The (preferred) international standard is made in millivolts per pascal at 1 kHz. A higher value indicates greater sensitivity. The older American method is referred to a 1 V/Pa standard and measured in plain decibels, resulting in a negative value. Again, a higher value indicates greater sensitivity, so −60  dB is more sensitive than −70 dB.
{{Clear}}

== Measurement microphones ==
[[Image:AKG C214 condenser microphone with H85 shock mount.jpg|thumb|200px|right|An [[AKG Acoustics|AKG]] C214 condenser microphone with [[shock mount]]]]
Some microphones are intended for testing speakers, measuring noise levels and otherwise quantifying an acoustic experience. These are calibrated transducers and are usually supplied with a calibration certificate that states absolute sensitivity against frequency. The quality of measurement microphones is often referred to using the designations "Class 1," "Type 2" etc., which are references not to microphone specifications but to [[sound level meter]]s.<ref>IEC Standard 61672 and ANSI S1.4</ref> A more comprehensive standard<ref>IEC 61094</ref> for the description of measurement microphone performance was recently adopted.

Measurement microphones are generally scalar sensors of [[pressure]]; they exhibit an omnidirectional response, limited only by the scattering profile of their physical dimensions. [[Sound intensity]] or sound power measurements require pressure-gradient measurements, which are typically made using arrays of at least two microphones, or with [[Hot-wire anemometry|hot-wire anemometers]].

===Calibration===
{{Main|Measurement microphone calibration}}

To take a scientific measurement with a microphone, its precise sensitivity must be known (in [[volt]]s per [[Pascal (unit)|pascal]]). Since this may change over the lifetime of the device, it is necessary to regularly [[Calibration|calibrate]] measurement microphones. This service is offered by some microphone manufacturers and by independent certified testing labs. All [[measurement microphone calibration|microphone calibration]] is ultimately traceable to [[primary standard]]s at a national measurement institute such as [[National Physical Laboratory (United Kingdom)|NPL]] in the UK, [[Physikalisch-Technische Bundesanstalt|PTB]] in Germany and [[NIST]] in the United States, which most commonly calibrate using the reciprocity primary standard. Measurement microphones calibrated using this method can then be used to calibrate other microphones using comparison calibration techniques.

Depending on the application, measurement microphones must be tested periodically (every year or several months, typically) and after any potentially damaging event, such as being dropped (most such microphones come in foam-padded cases to reduce this risk) or exposed to sounds beyond the acceptable level.

== Arrays ==
{{Main|Microphone array}}

A microphone array is any number of microphones operating in [[tandem]]. There are many applications:

*Systems for extracting voice input from [[ambient noise level|ambient noise]] (notably [[telephone]]s, [[speech recognition]] systems, [[hearing aid]]s)
*[[Surround sound]] and related technologies
*Locating objects by sound: [[acoustic source localization]], ''e.g.'', military use to locate the source(s) of artillery fire. Aircraft location and tracking.
*[[High fidelity]] original recordings
*3D spatial [[beamforming]] for localized acoustic detection of [[Subcutaneous tissue|subcutaneous]] sounds

Typically, an array is made up of omnidirectional microphones distributed about the [[perimeter]] of a space, linked to a [[computer]] that records and interprets the results into a coherent form.

{{Anchor|Microphone windscreens}}

== Windscreens ==
[[File:Microphone and cover.JPG|thumb|Microphone with its windscreen removed.]]
{{see also|Pop filter}}
'''Windscreens''' (or windshields – the terms are interchangeable) provide a method of reducing the effect of wind on microphones. While pop-screens give protection from unidirectional blasts, foam "hats" shield wind into the grille from all directions, and blimps / zeppelins / baskets entirely enclose the microphone and protect its body as well. The latter is important because, given the extreme low-frequency content of wind noise, vibration induced in the housing of the microphone can contribute substantially to the noise output.

The shielding material used – wire gauze, fabric or foam – is designed to have a significant acoustic impedance. The relatively low particle-velocity air pressure changes that constitute sound waves can pass through with minimal attenuation, but higher particle-velocity wind is impeded to a far greater extent. Increasing the thickness of the material improves wind attenuation but also begins to compromise high-frequency audio content. This limits the practical size of simple foam screens. While foams and wire meshes can be partly or wholly self-supporting, soft fabrics and gauzes require stretching on frames or laminating with coarser structural elements.

Since all wind noise is generated at the first surface the air hits, the greater the spacing between the shield periphery and microphone capsule, the greater the noise attenuation. For an approximately spherical shield, attenuation increases by (approximately) the cube of that distance. Thus larger shields are always much more efficient than smaller ones.<ref >{{Cite web
|url=http://www.ips.org.uk/files/03_Blasted_Microphones.pdf
|title=Blasted microphones
}}</ref> With full basket windshields there is an additional pressure chamber effect, first explained by Joerg Wuttke,<ref >{{Cite web
|url=http://www.filmebase.pt/Wind.html
|title=Joerg Wuttke – Microphones and Wind
}}</ref> which, for two-port (pressure gradient) microphones, allows the shield/microphone combination to act as a high-pass acoustic filter.

Since turbulence at a surface is the source of wind noise, reducing gross turbulence can add to noise reduction. Both aerodynamically smooth surfaces, and ones that prevent powerful vortices being generated, have been used successfully. Historically, artificial fur has proved very useful for this purpose since the fibers produce micro-turbulence and absorb energy silently. If not matted by wind and rain, the fur fibers are very transparent acoustically, but the woven or knitted backing can give significant attenuation. As a material, it suffers from being difficult to manufacture with consistency and to keep in pristine condition on location. Thus there is an interest (DPA 5100, Rycote Cyclone) to move away from its use.<ref >{{Cite web
|url=http://rycote.com/microphone-windshield-shock-mount/cyclone/
|title=Rycote Cyclone
}}</ref>

[[File:Pop filter.jpg|thumb|Singer and disc [[pop filter]] in front of a large-diaphragm condenser mic]]
In the studio and on stage, pop-screens and foam shields can be useful for reasons of hygiene and protecting microphones from spittle and sweat. They can also be useful colored idents. On location, the basket shield can contain a suspension system to isolate the microphone from shock and handling noise.

Stating the efficiency of wind noise reduction is an inexact science since the effect varies enormously with frequency, and hence with the bandwidth of the microphone and audio channel. At very low frequencies (10–100 Hz) where massive wind energy exists, reductions are important to avoid overloading of the audio chain – particularly the early stages. This can produce the typical “wumping” sound associated with wind, which is often syllabic muting of the audio due to LF peak limiting. At higher frequencies – 200 Hz to ~3 kHz – the aural sensitivity curve allows us to hear the effect of wind as an addition to the normal noise floor, even though it has a far lower energy content. Simple shields may allow the wind noise to be 10 dB less apparent; better ones can achieve nearer to a 50 dB reduction. However, the acoustic transparency, particularly at HF, should also be indicated, since a very high level of wind attenuation could be associated with very muffled audio.

<gallery widths="150px" heights="120px" perrow="4">
File:Schulze Brakel windshields 1 IBC 2008.jpg|Various microphone covers
File:Ecoacoustics recording in Rural Illinois, USA.jpg|Two recordings being made—a ''blimp'' is being used on the left. An open-cell foam windscreen is being used on the right.
File:Dead cat Dead Kitten.JPG| "Dead cat" and a "dead kitten" windscreens. The dead kitten covers a stereo microphone for a DSLR camera. The difference in name is due to the size of the enclosure.
</gallery>

== See also ==
{{Portal|Electronics}}
* [[Geophone]]—transducer for sound within the earth
* [[Hydrophone]]—transducer for sound in water
* [[Ionophone]]—plasma-based microphone
* [[Microphone blocker]]—computer accessory for disabling internal microphone.
* [[Microphone connector]]
* [[Microphone practice]]—examples of usage
* [[Nominal impedance]]
* [[Shock mount]]—Microphone mount that suspends the microphone in elastic straps

==Further reading==
* Corbett, Ian. ''Mic It!: Microphones, Microphone Techniques, and Their Impact on the Final Mix''. CRC Press, 2014.
* Eargle, John. ''The Microphone Book''. Taylor & Francis, 2004.

==References==
{{Reflist|refs=
<ref name="Montgomery1959">{{cite journal|last1=Montgomery|first1=Henry C|title=Amplification and High Fidelity in the Greek Theater|journal=The Classical Journal|date=1959|volume=54|issue=6|pages=242–245|jstor=3294133}}</ref>
<ref name="Huurdeman 2003">{{cite book |last=Huurdeman |first=Anton |year=2003 |title=The Worldwide History of Telecommunications |publisher=John Wiley & Sons }}</ref>
<ref name="microphone-data">{{Cite web|url=http://www.microphone-data.com/media/filestore/articles/History-10.pdf|title=A brief history of microphones|accessdate=2012-12-17}}</ref>
}}

==External links==
{{Commons category|Microphones}}
* [http://www.coutant.org/contents.html Info, Pictures and Soundbytes from vintage microphones]
* [http://www.sengpielaudio.com/calculator-transferfactor.htm Microphone sensitivity conversion—dB re 1 V/Pa and transfer factor mV/Pa]
* [http://recordinghacks.com/microphones Searchable database of specs and component info from 1000+ microphones]
* [https://web.archive.org/web/20091016023242/http://arts.ucsc.edu/ems/Music/tech_background/TE-20/teces_20.html Microphone construction and basic placement advice]

{{music technology}}
{{Basic computer components}}

{{Authority control}}

[[Category:Microphones| ]]
[[Category:Computing input devices]]
[[Category:History of television]]
[[Category:Sound recording]]
[[Category:American inventions]]
[[Category:19th-century inventions]]