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Class-D amplifier

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Class-D amplifier

A class-D amplifier, or switching amplifier, is an electronic amplifier in which the amplifying devices (transistors, usually MOSFETs) operate as electronic switches, and not as linear gain devices as in other amplifiers. They operate by rapidly switching back and forth between the supply rails, using pulse-width modulation, pulse-density modulation, or related techniques to produce a pulse train output. A simple low-pass filter may be used to attenuate their high-frequency content to provide analog output current and voltage. Little energy is dissipated in the amplifying transistors because they are always either fully on or fully off, so efficiency can exceed 90%.

The first class-D amplifier was invented by British scientist Alec Reeves in the 1950s and was first called by that name in 1955. The first commercial product was a kit module called the X-10 released by Sinclair Radionics in 1964. However, it had an output power of only 2.5 watts. The Sinclair X-20 in 1966 produced 20 watts but suffered from the inconsistencies and limitations of the germanium-based bipolar junction transistors available at the time. As a result, these early class-D amplifiers were impractical and unsuccessful. Practical class-D amplifiers were enabled by the development of silicon-based MOSFET (metal–oxide–semiconductor field-effect transistor) technology. In 1978, Sony introduced the TA-N88, the first class-D unit to employ power MOSFETs and a switched-mode power supply. There were subsequently rapid developments in MOSFET technology between 1979 and 1985. The availability of low-cost, fast-switching MOSFETs led to class-D amplifiers becoming successful in the mid-1980s. The first class-D amplifier based integrated circuit was released by Tripath in 1996, and it saw widespread use.

Class-D amplifiers work by generating a train of rectangular pulses of fixed amplitude but varying width and separation. This modulation represents the amplitude variations of the analog audio input signal. In some implementations, the pulses are synchronized with an incoming digital audio signal removing the necessity to convert the signal to analog. The output of the modulator is then used to turn the output transistors on and off alternately. Since the transistors are either fully on or fully off, they dissipate very little power. A simple low-pass filter consisting of an inductor and a capacitor provides a path for the low frequencies of the audio signal, leaving the high-frequency pulses behind.

The structure of a class-D power stage is comparable to that of a synchronously rectified buck converter, a type of non-isolated switched-mode power supply (SMPS). Whereas buck converters usually function as voltage regulators, delivering a constant DC voltage into a variable load, and can only source current, a class-D amplifier delivers a constantly changing voltage into a fixed load. A switching amplifier may use any type of power supply (e.g., a car battery or an internal SMPS), but the defining characteristic is that the amplification process itself operates by switching.

The theoretical power efficiency of class-D amplifiers is 100%. That is to say, all of the power supplied to it is delivered to the load and none is turned to heat. This is because an ideal switch in its on state would encounter no resistance and conduct all the current with no voltage drop across it, hence no power would be dissipated as heat. And when it is off, it would have the full supply voltage across it but no leakage current flowing through it, and again no power would be dissipated. Real-world power MOSFETs are not ideal switches, but practical efficiencies well over 90% are common for class-D amplifiers. By contrast, linear AB-class amplifiers are always operated with both current flowing through and voltage standing across the power devices. An ideal class-B amplifier has a theoretical maximum efficiency of 78%. Class-A amplifiers (purely linear, with the devices always at least partially on) have a theoretical maximum efficiency of 50% and some designs have efficiencies below 20%.

The 2-level waveform is derived using pulse-width modulation (PWM), pulse-density modulation (sometimes referred to as pulse frequency modulation), sliding mode control (more commonly called self-oscillating modulation.) or discrete-time forms of modulation such as delta-sigma modulation.

A simple means of creating the PWM signal is to use a high-speed comparator ("C" in the block diagram above) that compares a high-frequency triangular wave with the audio input. This generates a series of pulses of which the duty cycle is directly proportional with the instantaneous value of the audio signal. The comparator then drives a MOS gate driver which in turn drives a pair of high-power switching transistors (usually MOSFETs). This produces an amplified replica of the comparator's PWM signal. The output filter removes the high-frequency switching components of the PWM signal and reconstructs audio information that the speaker can use.

DSP-based amplifiers that generate a PWM signal directly from a digital audio signal (e. g. SPDIF) either use a counter to time the pulse length or implement a digital equivalent of the triangle-based modulator. In either case, the time resolution afforded by practical clock frequencies is only a few hundredths of a switching period, which is not enough to ensure low noise. In effect, the pulse length gets quantized, resulting in quantization distortion. In both cases, negative feedback is applied inside the digital domain, forming a noise shaper which results in lower noise in the audible frequency range.

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