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#1 |
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diyAudio Member
Join Date: Sep 2004
Location: Yahoo, USA
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Let me start this topic with a discussion of two from the many possible modulation schemes for non clocked class d amplifiers. For both schemes the power topology will be assumed to be a standard totem pole switching stage feeding a single output inductor. Linear feedback of current from this output inductor can be used to drive the power stage into oscillations, either by the application of hysteresis, sufficient phase shift, or a combination of the two. Thus, these common types of feedback driven, self oscillating class d amplifiers are really just high power variations of standard phase shift or hysteresis signal oscillators.
Hysteresis control alone suffers a troublesome reduction in oscillation frequency as the amplifier is driven into deep saturation (stopping altogether if the amp is railed). This is not so much a problem for the phase shift technique. Also, with hysteresis control, the period of oscillation varies inversely with rail voltage (since output inductor current slews faster when driven with higher voltage). These effects can be mitigated by making the hysteresis band a function of rail and output voltages and by filtering the current feedback signal. This latter step not only increases noise immunity when the hysteresis band is squashed at high output levels, but it also introduces phase shift into the negative feedback path. In fact, if the current signal is routed through two or more low pass sections of sufficient size, the mode of oscillation will become phase shift dominated. This happens when the phase shift around the feedback loop reaches 360 degrees at a frequency lower than the natural hysteresis frequency (a phase shift oscillator requires exactly 360 degrees of phase shift around its feedback loop). Starting around the loop from the high level switched output from the power stage, it is then immediately low pass filtered by the output inductor (yields 90 degrees of phase shift in current). If this were directly fed back to the error amplifier there would only be an additional 180 degrees phase shift (from negative feedback) - not nearly sufficient to cause natural oscillations. When there is little or no audio modulation, power stage duty cycle remains near 50 percent so that the comparator / modulator provides no phase shift input-to-output and, in a phase shift oscillator, the additional 90 degrees of phase shift required for oscillation by the Nyquist criterion typically comes from two additional high frequency poles in the feedback path (often placed as error amp compensation and as the comparator noise filter). The placement of these high frequency poles controls the frequency shift characteristics of the phase shift type amplifier, with the simplest case being two coincident high frequency poles placed at the intended nominal switching frequency. With this type of oscillator, loop gain, through the mechanism of power stage saturation, is automatically ensured to be exactly unity at the point of 360 degrees phase shift. When the phase shift class d amplifier is driven to very near saturation (greater than 90 percent duty cycle), the modulation process produces close to 90 degrees of phase shift on its own due to the sawtooth shape of the carrier that appears at the comparator’s input (duty cycle is now a pulse rather than a square wave), so that 360 degrees total phase shift occurs at a 3 to 4 times lower frequency without much contribution from the two additional high frequency poles. A free running class d amplifier's phase shift oscillator / modulation scheme allows loop gain for audio signals to come within a factor of two or three of the theoretical maximum possible bandwidth of one half the carrier frequency (which varies from the switching frequency at saturation to double the switching frequency at zero output). This is a Good Thing. High inner loop gain cascaded with outer loop gain really drives down unavoidable, open loop deadtime and switch on voltage distortion effects and makes for an inherently very high rejection of power supply noise. Hopefully this exposition was clear enough to convey my thoughts on how the typical non clocked class d amplifier oscillates and how it leads, not only to a very simple circuit, but a very high performing one as well. In addition to further thoughts on this subject, I am interested in learning about any other variable frequency class d amplifier schemes that are out there. Thanks for your time and your comments. -- analog(spiceman) |
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#2 |
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diyAudio Member
Join Date: Dec 2002
Location: Slovenia
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Hi,
only problem with your modulation scheme is that we need voltage generator to drive loudspeakers, not a current one. You inevitably need outer feedback loop to compensate for varynig loudspeaker impedance and filter capacitor current that becomes significant part of totem pole output current at higher audio frequencies. On the other hand, if one directly integrates totem pole voltage instead of difference between supply voltage and output voltage (as you do with output inductor), then we have all three known self oscillating voltage output topologies: -hysteretic oscillator -phase shift oscillator with feedback from rc filtered totem pole output voltage (COM patented by B&O) -phase shift oscillator with feedback from output LC filter (UcD patented by Philips) I am sure that someone with better insight will post a better comment. Best regards, Jaka Racman |
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#3 | ||
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diyAudio Member
Join Date: Sep 2004
Location: Yahoo, USA
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Quote:
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I also think that patent issues may be circumvented by using a hysteresis scheme with "heavy filtering" such that the result really becomes a hysteresis / phase shift combo modulator. The bulk of audio class d patents drive me crazy because most of their ideas are borrowed from or at least follow after well documented techniques from the switched mode power supply and motor control fields. Oh well, call it an audio circuit and suddenly you are a genius. |
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#4 | |
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diyAudio Member
Join Date: Apr 2004
Location: BE/NL/RW/ZA
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#5 | ||||
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diyAudio Member
Join Date: Apr 2004
Location: BE/NL/RW/ZA
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Quote:
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There is no practical reason why a class D amplifier should be equipped with an output filter of order exceeding 2. To use only the output filter to hold the state variables limits the available loop order to 2. Higher orders of integration would by necessity require a higher order output filter. If anything, small-signal integrators are cheaper than power inductors. Furthermore, the use of linear integrators (as opposed to saturatable ones like coils) will help reduce distortion from the coils whereas the coils themselves, used as integrators, will be ineffective at cancelling their own distortion! Quote:
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![]() Cheers, Bruno |
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#6 |
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diyAudio Member
Join Date: Dec 2002
Location: Slovenia
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Hi Bruno,
I do not know how familiar are you with currernt mode control which has been workhorse of SMPS industry for the last 20years. In comparison with SG3524 modulator it provided two advantages: reduction of one pole in transfer function and free pulse by pulse current limiting. Phase shift modulation would stand no chance there because nobody would buy power supply with 400mV ripple at the output. So I understand Analogspiceman's motives when he proposed his scheme. His original proposal would certainly not fall under UcD patent. If any, then original Bose patent for current mode control would be in place, but I think it has already expired. Also I think that "substantially free of hysteresis" is there because of certain T... patent Analogspiceman, I agree with you that current mode control is the best way to deal with multipole output filters. But Bruno had already explained why more than second order is not necessary, and you can always use multiphase ripple cancelation technique with second order filter. The only advantage of current mode control in class D amplifier I see is inherent current limit protection. Best regards, Jaka Racman |
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#7 | |
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diyAudio Member
Join Date: Apr 2004
Location: BE/NL/RW/ZA
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Quote:
![]() Current mode control of class D output filters has also been described by Karsten Nielsen but he stopped at 2nd order, probably because that was where the output filter stopped. For a class D amp, it's much more efficient to perform current mode control on the output cap instead of the coil. Controlling cap current has all the advantages of controlling coil current except that the loop does not heed or control actual output current (thus maintaining a low output impedance inherently). Because of that, controlling capacitor current will result in vastly lower output impedance than can ever be achieved through traditional inductor current mode control. The reasoning extends to higher order filters. At each tap in the filter, you attach a PD (R + C parallel) control takeoff which are summed to form the control signal. Presto: same loop gain around the switching power stage, lower output impedance. Yes, I am saying this is a better method than leapfrog. |
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#8 | ||
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diyAudio Member
Join Date: Sep 2004
Location: Yahoo, USA
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Quote:
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This tip brought to you by the school of hard knocks. -- analog(spiceman)
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#9 | |
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diyAudio Member
Join Date: Apr 2004
Location: BE/NL/RW/ZA
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#10 | |||
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diyAudio Member
Join Date: Sep 2004
Location: Yahoo, USA
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I know audio is just waking up to an alarm that went off more than twenty years ago for power supplies and dc motor control. There, techniques such as current mode control, self oscillation, phase staggered paralleled output stages, linear-switcher combos are old hat. Don't the patent guys pay attention to related fields? Or is their attention more relating to supplementing their patent pay? [/RANT] For these reasons, several years ago I decided to put leapfrog and whatever else I've come up with into the public domain via the internet in forums such as this one and various engineering newgroups. Quote:
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Regards -- analog(spiceman) |
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