Audio power amplifier
An audio amplifier is an electronic amplifier that amplifies low-power audio signals (signals composed primarily of frequencies between 20 - 20 000 Hz, the human range of hearing) to a level suitable for driving loudspeakers and is the final stage in a typical audio playback chain.
The preceding stages in such a chain are low power audio amplifiers which perform tasks like pre-amplification, equalization, tone control, mixing/effects, or audio sources like record players, CD players, and cassette players. Most audio amplifiers require these low-level inputs to adhere to line levels.
While the input signal to an audio amplifier may measure only a few hundred microwatts, its output may be tens, hundreds, or thousands of watts.
History
The audio amplifier was invented in 1909 by Lee De Forest when he invented the triode vacuum tube. The triode was a three terminal device with a control grid that can modulate the flow of electrons from the filament to the plate. The triode vacuum amplifier was used to make the first AM radio.[1]
Early audio amplifiers were based on vacuum tubes (also known as valves), and some of these achieved notably high quality (e.g., the Williamson amplifier of 1947-9). Most modern audio amplifiers are based on solid state devices (transistors such as BJTs, FETs and MOSFETs), but there are still some who prefer tube-based amplifiers, due to a perceived 'warmer' valve sound. Audio amplifiers based on transistors became practical with the wide availability of inexpensive transistors in the late 1960s.
Design parameters
Key design parameters for audio amplifiers are frequency response, gain, noise, and distortion. These are interdependent; increasing gain often leads to undesirable increases in noise and distortion. While negative feedback actually reduces the gain, it also reduces distortion. Most audio amplifiers are linear amplifiers operating in class AB.
Filters and preamplifiers
Historically, the majority of commercial audio preamplifiers made had complex filter circuits for equalization and tone adjustment, due to the far from ideal quality of recordings, playback technology, and speakers of the day.
Using today's high quality (often digital) source material, speakers, etc., such filter circuits are usually not needed. Audiophiles generally agree that filter circuits are to be avoided wherever possible. Today's audiophile amplifiers do not have tone controls or filters.
Since modern digital devices, including CD and DVD players, radio receivers and tape decks already provide a "flat" signal at line level, the preamp. is not needed other than as volume control. One alternative to a separate preamp. is to simply use passive volume and switching controls, sometimes integrated into a power amp. to form an "integrated" amplifier.
Further developments in amplifier design
For some years following the introduction of solid state amplifiers, their perceived sound did not have the excellent audio quality of the best valve amplifiers (see valve audio amplifier). This led audiophiles to believe that valve sound had an intrinsic quality due to the vacuum tube technology itself. In 1972, Matti Otala demonstrated the origin of a previously unobserved form of distortion: transitory intermodulation distortion (TIM), also called slew rate distortion. TIM distortion was found to occur during very rapid increases in amplifier output voltage.[2] TIM did not appear at steady state sine tone measurements, helping to hide it from design engineers prior to 1972. Problems with TIM distortion stem from reduced open loop frequency response of solid state amplifiers. Further works of Otala and other authors found the solution for TIM distortion, including increasing slew rate, decreasing preamp frequency bandwidth, and the insertion of a lag compensation circuit in the input stage of the amplifier.[3][4][5] In high quality modern amplifiers the open loop response is at least 20 kHz, canceling TIM distortion. However, TIM distortion is still present in most low price home quality amplifiers.[citation needed]
The next step in advanced design was the Baxandall Theorem, created by Peter Baxandall in England.[6] This theorem introduced the concept of comparing the ratio between the input distortion and the output distortion of an audio amplifier. This new idea helped audio design engineers to better evaluate the distortion processes within an audio amplifier.
Applications
Important applications include public address systems, theatrical and concert sound reinforcement, and domestic sound systems. The sound card in a personal computer contains several audio amplifiers (depending on number of channels), as does every stereo or home-theatre system.
References
- ^ http://nobelprize.org/educational_games/physics/transistor/history/ The Transistor in a Century of Electronics
- ^ "Circuit Design Modifications for Minimizing Transient Intermodulation Distortion in Audio Amplifiers", Matti Otala, Journal of Audio Engineering Society, Vol 20 # 5, June 1972
- ^ Distribution of the Phonograph Signal Rate of Change, Lammasniemi, Jorma; Nieminen, Kari, Journal of Audio Engineering Society, Vol. 28 # 5, May 1980.
- ^ "Psychoacoustic Detection Threshold of Transient Intermodulation Distortion", Petri-Larmi, M.; Otala, M.; Lammasniemi, J. Journal of Audio Engineering Society, Vol 28 # 3, March 1980
- ^ Discussion of practical design features that can provoke or lessen slew-rate limiting and transient intermodulation in audio amplifiers can also be found for example in chapter 9 in John Linsley Hood's 'The Art of Linear Electronics' (Butterworth-Heinemann, Oxford, 1993).
- ^ "Audio power amplifier design", Peter Baxandall. Wireless World magazine, February 1979
See also
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