Output transistor size selection, output stage protection, sound quality, modulation method, anti-electromagnetic interference (emi), lc filter design and system cost must protect the output stage from many potentially dangerous conditions:
Overheating: Although the power consumption of the output stage of class D amplifier is lower than that of linear amplifier, if the amplifier provides very high power for a long time, it will still reach the level of endangering the output transistor. In order to prevent the danger of overheating, a temperature monitoring control circuit is needed. In a simple protection scheme, when the temperature measured by the on-chip sensor exceeds the thermal shutdown safety threshold, the output stage will be turned off and kept cool. In addition to a simple binary indication of whether the temperature exceeds the shutdown threshold, the sensor can also provide other temperature information. By measuring the temperature, the control circuit can gradually reduce the volume level, reduce the power consumption, and well keep the temperature within a limited range, instead of forcing no sound when it is turned off.
Over-current of output transistor: If the output stage and speaker are connected correctly, the output transistor will be in a low on-resistance state, but if these nodes are accidentally short-circuited with another node or positive and negative power supply, huge current will be generated. If unchecked, this current will destroy transistors or peripheral circuits. Therefore, a current detection output transistor protection circuit is needed. In a simple protection scheme, if the output current exceeds the safety threshold, the output stage will be turned off. In a more complicated scheme, the output of the current sensor is fed back to the amplifier, trying to limit the output current to the maximum safe level, while allowing the amplifier to work continuously without turning off. In this scheme, if the current limiting protection is invalid, the last resort is forced shutdown. An effective current limiter can also keep the amplifier working safely when the loudspeaker vibration causes instantaneous high current.
Undervoltage: Most switching output stage circuits can only work normally when the positive power supply voltage is high enough. If the power supply voltage is too low, problems will occur. This problem is usually solved by undervoltage lockout circuit, and the output stage is allowed to work only when the power supply voltage is greater than the undervoltage lockout threshold.
Turn-on timing of output transistors: mh and ml output stage transistors (see Figure 6) have extremely low on-resistance. Therefore, it is very important to avoid the situation that mh and ml are turned on at the same time, because this will create a low-resistance path from vdd to vss through the transistor, thus generating a large impact current. The best case is that the transistor heats up and consumes electricity; In the worst case, the transistor may be damaged. Make-before-make control of transistors prevents inrush current by forcing two transistors to turn off before one transistor is turned on. The time interval when both transistors are turned off is called non-overlapping time or dead time.
Figure 6. Make-before-make switch of output stage transistor
Note: Switch output stage = Switch output stage.
Non-overlapping time = non-overlapping time.
On = on
Off= Off In class D amplifier, several problems must be solved to obtain good overall sound quality.
Click sound: The click sound made when the power amplifier is turned on or off is very annoying. Unfortunately, they are easily introduced into class D amplifiers, unless special attention is paid to the modulator state, output stage timing and lc filter state when the amplifier is quiet or not. Distortion mechanism includes nonlinearity in modulation technology or modulator implementation, and dead time adopted by the output stage to solve the surge current problem.
Information including the amplitude of an audio signal is usually encoded in the output pulse width of a class D modulator. The extra dead time used to prevent the surge current in the output stage will introduce nonlinear timing error, and the distortion it produces in the speaker is proportional to the timing error relative to the ideal pulse width. The shortest dead time to avoid impact is usually the most beneficial to minimize distortion; For detailed design methods to optimize the distortion performance of the switching output stage, please refer to in-depth reading 2.
Other sources of distortion include mismatch between the rise time and fall time of the output pulse, mismatch of the timing characteristics of the gate drive circuit of the output transistor, and nonlinearity of the lc low-pass filter element. In the circuit shown in Figure 2, the power supply noise is almost directly coupled to the output speaker, and there is almost no suppression effect. This is because the output stage transistor connects the power supply to the low-pass filter through very low resistance. The filter suppresses high-frequency noise, but all audio passes, including audio noise. For a detailed explanation of the influence of single-ended and differential switch output stage circuits on power supply noise, please refer to in-depth reading material 3.
If the distortion problem and power supply problem are not solved, it is difficult to achieve psr better than 10 db or thd better than 0. 1%. To make matters worse, thd is often harmful to high-order distortion of sound quality.
Fortunately, there are some good solutions to these problems. It is very useful to use feedback with high loop gain, as used in many linear amplifier designs. The feedback of lc filter input will greatly improve psr and attenuate all non-lc filter distortion sources. The nonlinearity of that lc filt can be attenuated by a speak included in the feedback loop. In the carefully designed closed-loop class D amplifier, PSR >:; 60 db and total harmonic distortion
However, feedback complicates the design of the amplifier, because the loop stability must be satisfied (which is a very complicated consideration for high-order design). Continuous-time analog feedback is also necessary to capture important information about pulse timing error, so the control loop must include analog circuits to process the feedback signal. In the implementation of integrated circuit amplifier, this will increase the chip cost.
In order to minimize the cost of integrated circuits, some manufacturers prefer not to use or use the least number of analog circuit components. Some products use digital open-loop modulators and analog-to-digital converters to detect power supply changes, and adjust the modulator behavior to compensate. See in-depth reading 3 for details. This can improve the psr, but it will not solve any distortion problems. Other digital modulators try to pre-compensate the expected output stage timing error or correct non-ideal modulators. This will deal with at least some distortion sources, but not all. These open-loop class-D amplifiers can be used for processing applications with loose sound quality requirements, but some form of feedback seems necessary to obtain the best sound quality. Class-D amplifier modulator can be realized in many ways, and there are many related research and intellectual property support. This article only introduces the basic concepts.
All D-class power amplifier modulation technologies encode the related information of audio signals into a series of pulses. Generally, the pulse width is related to the amplitude of the audio signal, and the pulse spectrum includes useful audio signal pulses and useless (but inevitable) high-frequency components. In all schemes, the total power of integrated high frequency is almost the same, because the total power of time domain waveform is the same, according to parseval theorem, the time domain power must be equal to the frequency domain power. However, the energy distribution is very different: in some schemes, there are high-energy tones on the low-noise floor, while in other schemes, the high-energy tones are eliminated by energy shaping, but the noise floor is higher.
The most commonly used modulation technique is pulse width modulation (pwm). In principle, pwm compares the input audio signal with a triangular wave or an oblique wave operating at a fixed carrier frequency. This will produce a series of pulses at the carrier frequency. In each carrier period, the duty ratio of pwm pulse is proportional to the amplitude of audio signal. In the example of fig. 7, both the audio input and the triangular wave are centered at 0 v, so for zero input, the duty cycle of the output pulse is 50%. For larger positive input, the duty cycle is close to 100%, and for larger negative input, the duty cycle is close to 0%. If the amplitude of audio exceeds the amplitude of triangular wave, full modulation will occur, at which time the pulse train stops switching and the duty cycle is 0% or 100% in a specific period.
Pwm is attractive because it allows audio band snr of 100 db or higher (low enough to limit the switching loss of the output stage) at a carrier frequency of several hundred kilohertz. Many pwm modulators are stable even if the modulation reaches almost 100%, and in principle, high output power is allowed to reach the overload point. However, pwm has several problems: first, many pwm implementations will increase the inherent distortion (see further reading 4); Secondly, the resonance of pwm carrier frequency will produce emi in AM radio band. Finally, the pwm pulse width is very small near full modulation. This will cause problems in the gate driving circuits of most switching output stages, because their driving ability is limited, and they cannot switch correctly at the extremely fast speed required to regenerate the short pulse width of several nanoseconds (ns). Therefore, in the amplifier based on pwm, it is often impossible to achieve full modulation, and the maximum output power that can be achieved is less than the theoretical maximum, that is, only the power supply voltage, transistor on-resistance and speaker impedance are considered.
The alternative to pwm is pulse density modulation (pdm), in which the number of pulses in a given time window (pulse width) is proportional to the average value of the input audio signal. Its single pulse width is not arbitrary like pwm, but a "quantized" multiple of the modulator clock period. 1 bit σ-δ modulation is a form of pdm.
A lot of high frequency energy in σ -δ modulation is distributed in a wide frequency range, rather than concentrated at the frequency doubling of carrier frequency like pwm, so the potential emi advantage of σ -δ modulation is better than pwm. At the mirror frequency of pdm sampling clock frequency, energy still exists; However, in the typical clock frequency range of 3 mhz~6 mhz, the image frequency falls outside the audio frequency band and is strongly attenuated by the lc low-pass filter.
Another advantage of σ -δ modulation is that the minimum pulse width is one sampling clock period, even under almost completely modulated signal conditions. This simplifies the design of the gate driver and theoretically allows safe operation at full power. Nevertheless, 1 bit σ-δ modulation is not commonly used in class D amplifiers (see further reading 4) because the traditional 1 bit modulator can only stabilize to 50% modulation. In order to obtain sufficient snr of audio frequency band, it needs at least 64 times oversampling, so the typical output data rate is at least 1 mhz, and its efficiency is limited.
Recently, a self-excited oscillation amplifier has been developed, such as the one introduced in Extended Reading 5. This kind of amplifier always includes a feedback loop, and the switching frequency of the modulator is determined by the loop characteristics, not by the externally provided clock. High frequency energy is usually flatter than pwm distribution. Because of the feedback, excellent sound quality can be obtained, but the loop is self-oscillating, so it is difficult to synchronize with any other switching circuit, and it is also difficult to connect to a digital audio source without first converting the digital signal into an analog signal.
The full-bridge circuit (see Figure 3) can use "three-state" modulation to reduce differential emi. In the traditional differential working mode, the output polarity of half-bridge A must be opposite to that of half-bridge B, and there are only two differential working states: high output A and low output B; Output a is low and output b is high. However, there are two other * * * mode states, that is, two half-bridge outputs have the same polarity (both high or both low). One of the two * * * mode states can cooperate with the differential state to generate tri-state modulation, and the differential input of lc filter can be positive, zero and negative. A zero state can be used to indicate a low power level, instead of switching between a positive state and a negative state in a two-state scheme. During the zero state, the differential effect of lc filter is very small. Although * * * mode emi actually increases, differential emi decreases. The differential advantage only applies to low power levels, because the positive and negative states must still be used to provide high power to the speakers. The voltage level change of * * * mode in three-state modulation scheme is a design challenge of closed-loop amplifier. Note: sample audio input = sample audio input.
Pwmeout = PWM output.
Triangle wave = triangle wave
Pwm concept =pwm principle
Pwm example =pwm example
Sine = sine wave
Audio input = audio input
Pulse = pulse
Pwm output =pwm output. The high frequency component of the output of class D amplifier deserves serious consideration. If not properly understood and handled, these components will produce a lot of electromagnetic interference and interfere with the work of other equipment.
Two types of emi need to be considered: signals radiated into space and signals conducted through speakers and power lines. The modulation scheme of class D amplifier determines the baseline spectrum of conducted emi and radiated emi components. However, some board-level design methods can be used to reduce the emi emitted by class D amplifiers, regardless of their baseline spectrum.
A useful principle is to reduce the loop area carrying high frequency current as much as possible, because the intensity related to emi is related to the loop area and the proximity of the loop to other circuits. For example, the layout of the entire lc filter (including speaker wiring) should be as close as possible to the amplifier. Current drive and return path printed lines should be concentrated together to minimize the loop area (twisted pair is very helpful for speakers). Another point to note is that when the gate capacitance of the output stage transistor is switched, a large number of transient charges will be generated. Usually this charge comes from the storage capacitor, thus forming a current loop containing two capacitors. By minimizing the loop area, the influence of transient emi in the loop can be reduced, which means that the storage capacitor should be charged as close as possible to the transistor.
Sometimes, it helps to insert an rf choke in series with the amplifier power supply. Their proper arrangement can limit the high-frequency transient current in the local loop near the amplifier without conducting along the power line for a long distance.
If the gate drive non-overlapping time is long, the induced current of the speaker or lc filter will forward bias the parasitic diode at the transistor end of the output stage. When the non-overlapping time ends, the diode bias changes from forward to reverse. Before the diode is completely turned off, there will be a large reverse recovery current spike, which will cause troublesome emi sources. Minimize emi by keeping the non-overlapping time very short (it is also recommended to minimize audio distortion). If the reverse recovery scheme is still unacceptable, a Schottky diode can be used in parallel with the parasitic diode of the transistor to divert the current and prevent the parasitic diode from always conducting. This is helpful because the metal-semiconductor junction of Schottky diode is basically unaffected by the reverse recovery effect.
The Lc filter with annular inductance core can minimize the influence of stray field power lines caused by amplifier current. A good compromise between cost and emi performance is to reduce radiation from low-cost drum cores through shielding. If attention is paid, this shielding can ensure that the linearity of the inductor and the sound quality of the speaker can be reduced acceptably. In order to save cost and pcb area, lc filters of most class D amplifiers adopt second-order low-pass design. Fig. 3 shows a differential second-order lc filter. Speakers are used to reduce the natural resonance of the circuit. Although the loudspeaker impedance is sometimes similar to a simple resistance, the actual impedance is more complex and may contain a large number of reactive elements. In order to get the best filter design effect, design engineers should always try to use accurate speaker models.
The common purpose of filter design selection is to minimize the decline of filter response under the condition of requiring the highest audio frequency, so as to obtain the best and lowest bandwidth. If the voltage drop below 20 khz is required to be less than 1 db, a typical filter needs to have a Butterworth response of 40 khz (to achieve the maximum flat passband). For common loudspeaker impedances and standard L and C values, the following table gives the nominal component values and their corresponding approximate Butterworth responses:
Inductance L(μH) Capacitance C(μF) Speaker Resistance (ω) Bandwidth -3 dB(kHz)
10 1.2 4 50
15 1 6 4 1
22 0.68 8 4 1
If the design does not include speaker feedback, the speaker thd will be sensitive to the linearity of lc filter elements.
Inductor design considerations: Important factors in designing or selecting inductors include rated current, core shape and winding resistance.
Rated current: The rated current of the selected core should be greater than the expected maximum current of the amplifier. The reason is that if the current exceeds the rated current threshold and the current density is too high, many inductor cores will be magnetically saturated, resulting in a sharp drop in inductance, which is not desirable.
Inductors are formed by winding wires around a magnetic core. If the number of winding turns is large, the resistance related to the total winding length is very important. Because the resistor is connected in series between the half-bridge and the loudspeaker, some output power will be consumed. If the resistance is too high, we should use a thick winding or choose a magnetic core made of other metals, which requires less winding turns to provide the required inductance.
Finally, don't forget that the shape of the inductor used will also affect emi, as mentioned above. What are the important factors that affect the total cost of audio systems using class D amplifiers? How can we minimize the cost?
The active devices of class D amplifier are switching output stage and modulator. The cost of this circuit is about the same as that of an analog linear amplifier. The real tradeoff to consider is the other components of the system.
The low power consumption of Class D amplifier saves the cost (and pcb area) of heat dissipation devices, such as radiators or fans. Compared with analog linear amplifiers, class D integrated circuit amplifiers have smaller package size and lower cost. When driving a digital audio source, an analog linear amplifier needs a digital-to-analog converter (dac) to convert the audio signal into an analog signal. Class-D amplifiers that process analog inputs need the same conversion, but dac functions are effectively integrated for class-D amplifiers with digital inputs.
On the other hand, the main cost disadvantage of class D amplifier is lc filter. Components of lc filter, especially inductors, occupy pcb area and increase cost. Among the high-power amplifiers, the overall system cost of class D amplifier is still competitive, because the large cost saved on the heat sink can offset the cost of lc filter. However, in low-cost and low-power applications, the cost of inductors is very high. In rare cases, such as the low-cost amplifier of mobile phone, the cost of amplifier ic may be lower than the total cost of lc filter. Even ignoring the cost consideration, the pcb area occupied by lc filter is still a problem in small applications.
In order to meet these considerations, the lc filter is sometimes completely cancelled to adopt a filter-free amplifier design. This can save cost and pcb area, although the advantages of low-pass filter have been lost. If there is no filter, the increase of emi and high frequency power consumption will be unacceptable unless the speaker is inductive and very close to the amplifier, the current loop area is the smallest, and the power level is kept low. Although this design is usually used in portable applications, such as mobile phones, it is not suitable for high-power systems, such as home stereo systems.
Another method is to minimize the number of lc filter elements required for each audio channel. This can be achieved by using a single-ended half-bridge output stage, which requires half the inductance and capacitance of a differential full-bridge circuit. However, if the half-bridge output stage needs bipolar power supply, the cost associated with generating negative power supply may be too high unless the negative power supply has other uses or the amplifier has enough audio channels to share the power supply cost. In addition, the half-bridge can also be powered by a single power supply, but this will reduce the output power and usually require the use of a large DC blocking capacitor. All the design problems just discussed can be summed up in a very strict project. In order to save the time of design engineers, analog devices has provided various class D amplifiers ic 1, including programmable gain amplifiers, modulators and power output stages. To simplify the evaluation, analog devices provides demonstration boards for each type of amplifier. The pcb layout and bill of materials of these demonstration boards can be used as practical reference designs, thus helping customers to quickly design a proven, economical and effective audio system without having to do "repetitive work" to solve the main design problems of class D amplifiers.
For example, we can consider using the dual amplifier ic series products of ad 19902, ad 19923, ad 19944 and ad 199655. These products are suitable for each channel with two outputs of 5,10,25 and. Here are some features of these ics:
Ad 1994 d audio power amplifier has two programmable gain amplifiers, two σ -δ modulators and two power output stages, which are used to drive full H-bridge connection loads in home theater, automobile and pc audio applications. The switching waveform generated by it can drive two 25 w stereo speakers or a 50 w mono speaker with an efficiency of 90%. Its single-ended input is applied to a programmable gain amplifier (pga), and the gain of this amplifier can be set to 0, 6, 12 and 18 db to handle low-level signals.
The Ad 1994 integrated protection function can protect the output stage from overheating, overcurrent and surge current. Due to its special timing control, soft start and DC offset calibration, the click associated with mute is very small. Its main performance indexes include the dynamic range of 0.00 1%thd and 105 db, the psr is greater than 60 db, and the gate driver of the output stage has continuous time feedback and optimization of the switching output stage. Its 1 bit σ-δ modulator has been specially enhanced for class D applications to achieve an average data frequency of 500 khz, with high loop gain of 90% modulation and full modulation stability. Independent modulator mode allows driving external high-output field effect transistors (fet).
Ad 1994 uses 5 v power supply for pga, modulator and digital logic, and 8 V ~ 20 V high voltage power supply for switching output stage. Relevant reference designs meet the requirements of FCC emi standards. When the 6 Ω load is driven by 5 v and 12 v power supplies, the static power consumption is 487 mw, the power consumption is 7 10 mw under the condition of 2× 1 w output power, and the power consumption is 0.27mw in standby mode. The ad 1994 is packaged in a 64-lead lfcsp, and the operating temperature range is–-40°c to+85 c.