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Pulse Width Modulation (PWM) is popular for its high efficiency and ease of use. In the past, it was generally considered to be suitable only for power or digital devices, but not for high-sensitivity devices such as audio. However, in recent years, many well-known audio amplifier manufacturers have begun to produce a series of PWM audio amplifiers, initially subwoofer amplifiers, and now cover the entire audio spectrum from 20Hz to 22kHz. This article will explore how to use PWM digital technology to achieve the performance of traditional analog audio equipment.
Classification of amplifiers
Power amplifiers are generally divided into four categories: A, B, AB, and C.
The simplest amplifier has only one active device, such as a transistor. The transistor is biased, so no matter how large the input signal is, it is never possible to turn it on completely or completely. This non-off/non-conducting region is a so-called linear region. The amplifier output in the linear region has very low output distortion, but its efficiency is also low. It is a class A amplifier.
Class B amplifiers consist of two transistors that push and pull each other, one for the output current and the other for the current. Suppose you want to amplify a positive-negative half-cycle sine wave with zero-symmetry, then one transistor amplifies the upper half of the sine wave (the part above zero) and the other amplifies the lower half (the part below zero). In other words, the amplification is done by two transistors in turn, so the efficiency of the Class B amplifier is higher. The problem with such an amplifier is that there is a non-linear region, that is, the small region where the sine wave just passes through the zero point. At this time, one transistor has just turned off and the other just turned on. Since the transistor turns on requires a short transition time, it will cause distortion due to the nonlinear state.
Class AB amplifiers are a combination of Class A and Class B amplifiers. It is much like a Class B amplifier, but uses a circuit that provides a small bias current to each transistor, so each transistor does not completely turn off. It consumes as much power as a Class A amplifier, but with much lower distortion. It is also like a Class B amplifier. Two transistors work together to complete the task, so the overall performance is better.
Class C amplifiers are typically used for RF or oscillators because distortion is not an issue at this time and is not discussed in depth here.
Class D amplifier with PWM technology
Class D amplifiers use PWM technology, which controls the duty cycle of a fixed-frequency square wave and represents the input value by duty cycle. Because PWM can achieve higher efficiency, it is often used in high power devices. The power amplifier used in electric vehicles is a class D amplifier, and the wind generator is also a class D amplifier for returning current. So, can this industrial technology be used to process music?
In terms of amplifiers, Class D amplifiers are indeed very efficient (typically up to 90%). Since transistors are almost always in either the on state or the off state, they only enter the linear region when moving from one state to another, so they consume much less power than linear amplifiers. In a linear amplifier, the transistor has a large portion of its time in the linear region.
For Class D audio amplifiers, the load is placed in the middle of the H-bridge (see Figure 1). This has the advantage that the output can be either positive or negative, which can greatly increase the power to four times that of an A or B amplifier.
From a practical point of view, as long as the PWM has sufficient accuracy and frequency, it is possible to obtain acceptable control characteristics and good audio effects. The accuracy should be 16 bits (or greater) and the PWM carrier frequency should be no less than 12 times the audio bandwidth, preferably 25 times. As with other audio devices, it is important to improve the accuracy of the dynamic range. The accuracy of a standard CD player is 16 bits.
Filter removes high frequency harmonics
The PWM carrier must be removed from the audio before hands-on design.
If you want to design a subwoofer class D amplifier, its typical bandwidth is 20Hz to 500kHz. This requires an oversampling frequency of at least 6 kHz, preferably 12.5 kHz. In a simple application, the audio codec can be used as an input to the DSP, and the digital output can be used to drive the on-board peripherals of the PWM, in many cases without any processing.
To remove the PWM carrier from the audio output, the task can be accomplished with a suitable filter. The structure of the filter—that is, the cutoff frequency and order—is determined by the oversampling frequency or PWM frequency. The higher the PWM frequency, the lower the filter order and the simpler it is. In Figure 1, there is one speaker between the two second-order LC filters and one filter per half bridge. These filters remove carrier and other harmonics from the output.
Dead-band distortion is the second problem to be solved by filters. It takes time for the large power transistors that make up the H-bridge to turn on and off, so some time must be allocated to prevent one transistor from being turned on while the other is on. If this happens, the so-called "breakdown" phenomenon will occur. To avoid this, the controller must ensure that the upper and lower transistors of each pin are turned off for a period of time before being turned on. This period of time is called the dead band time, which causes distortion similar to a Class B amplifier. This distortion problem can be solved with a filter.
It is generally sufficient to use a Butterworth or Bezier filter, and the passbands of both are relatively flat. Bessel filters also have the advantage of linear phase.
The H-bridge in Figure 1 has two filters, one on each speaker foot. If you are used to single-ended filter design, changing them to equalization filters is also a trivial matter. A simple calculation of the filter with a half-rated load can then be used to obtain the resulting L and C values.
About the Author:
Don Morgan is a senior engineer at the Ultra Stereo Lab, a senior consultant with 25 years of experience in signal processing, embedded systems, hardware and software. He is also the author of "C Language Digital Methods for DSP Systems", "Practical DSP Modeling, Techniques and C Programming" and "Digital Methods for Embedded Systems." You can contact him by email.
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