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Class D Switching Power Amplifiers and Power D/A conversion

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Old 30th November 2002, 12:15 AM   #21
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Default Filters and Processors

Quote:
Originally posted by Rarkov
Hi,
I was reading your reply...My point about the output filter is that for a high power amp (100W in the case of the zetex), it needed about 6 or 8 amps current handling through a 20uH inductor (I'm trying to remember these figures off the top of my head though). These don't exist. Upon searching the internet for them - I came accross a 40A (!) Switched Mode power supply with these exact ratings. I got the cores and wound them myself...It required about ten wires in parallel to handle the current!

I said it was difficult to design with low distortion and easily sourced parts...That wasn't something I considered to be either. If the TI does have a far higher smapling rate, a smaller inductor will be used - and therefore - higher current handling...Smiles all round. So maybe the morel of the story is not to go off Class D - but to avoid Zetex...As I said earlier...
You're right that the output inductors are a little trickier to source. This is one of the final details I'm sorting out before I order my boards.

TI recommends a shielded bobbin type inductor. Most 384KHz Equibit amps with BTL configured outputs require a 10uH inductor coming off of each half-bridge. This gives a total of 20uH for each channel, the same as for the Zetex parts you were using. I'm guessing they must have had a similar carrier frequency. Their 30W reference design uses a Taiyo Yuden LHFP13BB100M, which just happens to be the largest value available for this part.

Once you get up to a certain power level (say >50VA) there are enough variables with transformers and inductors that it isn't practical for manufacturers to offer off the shelf parts. Instead they have a selection of standard cores and do custom winding. Many of these places are set up to do low volume samples.

I'm also considering using output inductors wound on torroid cores.

The other part I'm finding slightly tricky (at least in small quantities) are the helically trimmed MELF0204 resistors required for the power supply snubbers. I'll do a more complete search for these once I settle on the values I require.

Quote:
This is turning out to be quite interesting! Have you tried looking at the DSP Guide. It is a free online book of DSP. Very useful!
Yes, I have this book. It's excellent!
While it isn't specifically for audio, it's probably the best single DSP book I've found. Even though it's offered free on the website, I highly recommend buying the printed copy and supporting the author, Steven Smith.

Quote:
My other problem is implementing DSP functions...Microchip are starting to bring things out - but they are not upto audiophile grades yet...More instrumentation etc...
Microchip's dsPICs should be really usefull for digitally generating test signals and tuning a system, but they don't have sufficient horsepower for much processing of the actual audio signal. I've got the development system for these, although I haven't started using it yet.

In case you didn't see it, there was a thread that included a discussion of audio processors:
Digital speaker phase correction?

I'm planning to start with TI's TAS3103 Digital Audio Processor, which is a preprogrammed DSP for audio. The program can't be changed on this part. The only thing that can be programmed are the filter and gain coefficients. This makes it much easier to get going with this part than with a general purpose DSP. You can start with all of the default values and just use it as a volume control. The other functions can then be experimented with one at a time when you're ready.

The only shortcoming that I see with the TAS3103 (and this is true of most other DSPs), is that it doesn't have the capability of doing massive custom FIRs. Once I'm ready to experiment with these, I plan to use an FPGA. A high powered DSP (such as the TMS320DA610, when it becomes generally available) would be another option. Other than the FIRs, the TAS3103 has the processing bandwidth and features to do everything else I anticipate wanting to do.

Regards,
Brian.
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Old 30th November 2002, 12:54 AM   #22
deandob is offline deandob  Australia
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Default Thoughts on switching frequency

A number of opponents to digital amplifiers call out the switching frequency of current generation of class D amplifiers as a problem. Discussions about distortion caused by the output filter, difficulty in getting the output filter correct, its heat dissipation, high switching frequencies > 2 Mhz needed to be truly load invariant are some of the issues discussed. From Brian's previous explainations of the technologies involved (Equibit with its skewed PWM timings, Tripath with its spread spectrum & adaptive technologies), is it fair to assume that you cannot compare implementations based on switching frequency alone?

However, in general the higher the switching frequency, the easier it would be to design a high performance output filter. The TI chips @ 384Khz look to be at a disadvantage compared to some of the other designs. With most things in audio, its not the theory that counts, its the implementation which is why it would be great to evaluate the sonic differences between the different implementations. I'm looking forward to reading Brian's comments on his experements with the TI chips as they sound like they have great potential

Comments?

Regarding varying the power supply for volume, a thought that comes to mind to do this simply is to use a Varac which will save the complexity of a variable voltage switching power supply.

One other thing that requires careful consideration is the amount of digital hash created by dumping large currents at high frequencies, which can radiate lots of RF noise as well as pollute the mains supply. Careful layout & filter design is needed here.

Dean
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Old 30th November 2002, 02:14 PM   #23
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Default Re: Thoughts on switching frequency

Quote:
Originally posted by deandob

is it fair to assume that you cannot compare implementations based on switching frequency alone?
I would think so.
Quote:
However, in general the higher the switching frequency, the easier it would be to design a high performance output filter. The TI chips @ 384Khz look to be at a disadvantage compared to some of the other designs.
There are two advantages to a higher switching frequency:

Greater dynamic resolution and smaller low pass reconstruction filters.

By playing tricks with the timing, Equibit achieves a much higher resolution for a given PWM frequency than would standard fixed PWM.

In a BTL configuration, Equibit can use fully symetrical, balanced filters. Many Class D amps use unbalanced filters. The use of balanced filters greatly reduces distortion, and also makes filter design somewhat easier.

A lot of the problems Class D has arise from the feedback loop. The circuitry can only make one 'correction' per PWM pulse. This greatly limits the response bandwidth. There are also issues with the way the feedback circuitry interacts with the output filter. Higher switching frequencies help with these issues, but then cause other issues with power disipation, EMI, and other types of distortion. As I mentioned previously, an output device has pretty much fixed turn-on and turn-off times. Higher switching frequencies mean that the required underlap time will be a greater percentage of the total time. This increases distortion.

Tripath deals with the 'one feedback correction per pulse' limitation by using an adaptive / predictive filter. It learns from its past mistakes and doesn't strictly rely on instantanious feedback.

Equibit completely avoids these issues by running open loop.

*****

Equibit amps do have load sensitivity with regard to load impedance. Lower impedance speakers will cause the individual current spikes through the output FETs to be greater. One way to deal with this is to overrate the output devices to handle the worst-case load. To adjust an amp for an individual speaker, the main parameter is the supply voltage. The optimum voltage would be set so that the speaker achieves its maximum output at around 80% to 90% PWM duty cycle. (A lower voltage wouldn't allow for maximum output - but could be used for volume control. A higher voltage would simply increase the current spikes in the FETs.)

Placing the amps near the speaker is desirable to eliminate long speaker wire runs. It's better to physically distribute while the signal is still digital.

There are many good reasons to design an audio system with multiple Equibit amps - one per speaker driver.

1.) Digital crossovers with amplitude and phase correction for each driver.

2.) Optimized supply voltage for each driver.

3.) Elimination of passive crossovers.
Besides all of the advantages of digital crossovers, this allows the amp to have much better control of the driver. Equibit amps have an excellent damping factor. Any back-EMF from the driver gets directly shunted to the power supply. (A passive crossover reduces the coupling to the amp and can also allow back EMF from one driver into another.) Also, a passive crossover usually has a greater voltage drop than the output devices in the amp. Because an Equibit amp has such high efficiency to begin with, elimination of the passive crossover is one of the major opportunities to further increase efficiency.

4.) It's easier to build multiple lower power output sections than it is to build a single higher power section.

Quote:
One other thing that requires careful consideration is the amount of digital hash created by dumping large currents at high frequencies, which can radiate lots of RF noise as well as pollute the mains supply. Careful layout & filter design is needed here.
You're certainly right about that.

In my other life, I work on industrial motor drive design. One of the little jokes we have is that it is a good idea never to get a reputation for being good at dealing with EMI issues. If you get identified as being some type of EMI guru, you will be forever sentenced to chasing the little demons.

I'm relying heavily on TI's reference design for the board layout of the output section. They clearly have paid close attention to power routing. There are little RC snubber circuits sprinkled all over the place. The component values for many of these get fine tuned after the board is built. The main power input to the TAS5110 has a reactive LC filter using controlled board traces for the inductors.

I don't think I would have even considered attempting to build one of these if I didn't have the reference design as an example.

Regards,
Brian.
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Old 1st December 2002, 02:05 PM   #24
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Hi there,
I'm considering building one of these amps myself, with the TI devices and descrete output switching FETs. I haven't got that far into the design yet though, but it's beutiful to see your nowledge here!

A few questions for you guys (mostly Brian so far?) which has gotten a little bit further in designing:

1) Why not consider the higher spec parts TAS5015 and TAS5182? Cos they are not available/sampling yet? Layout should be similar to the parts on the EVM board.

2) The soldering of these parts will not be trivial with the 0.65mm spacing. Do you have professional equipment at hand or do you have other smart ways of dealing with it? Otherwise I can see this becoming a potential showstopper here....

3) Brian, as I understand it you are not buying the EVM, but designing you own PCB? Are you considering posting you PCB's files here? I could use a starting/branching point...

4) It seems like the TacT Millenium uses quite large Jensen air core foil inductors placed close to the speaker terminals. I think it could be a good choice if the distance from the switching stage is not a big issue. They seems bigger than some 10uH though?
(pics)
Jensen

5) What are you considering for high quality volume control? Are you all planning a digital attenuation (potentially decreasing the resolution) or is anybody have ideas for a H.Q. variable power supply?

6) My plan is a totally digital amp with the reciever chip CS8420, which can use an external precision oscillator for clock generation. Any comments on the need for an very stable clock, and its placement?

cheers all,
__________________
Andrej
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Old 2nd December 2002, 06:40 PM   #25
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Quote:
Originally posted by Strummer

1) Why not consider the higher spec parts TAS5015 and TAS5182? Cos they are not available/sampling yet? Layout should be similar to the parts on the EVM board.
The TAS5015 is a substantially more complicated part to use, especially in terms of the gate drive, protection circuitry, and power supply requirements. The layout for this will be very different than the EVM. Also, when I first started my design, it was only available to certain Equibit licensees. (It appears to be available for general consumption now.)

The TAS5182 isn't available yet. I plan on using it for future designs.

Quote:
2) The soldering of these parts will not be trivial with the 0.65mm spacing. Do you have professional equipment at hand or do you have other smart ways of dealing with it? Otherwise I can see this becoming a potential showstopper here....
I use a Metcal SP200 soldering iron with an old used Leica binocular scope.
I use ChipQuick for removal of high lead density parts.
I also have a small oven that is useful for preheating or reflowing boards.

Technique is the most important thing when working with fine pitch surface mount components, much more than specialized equipment. In a pinch, I've gotten excellent results from a cheap 15W soldering iron with the tip filed down to a small blade.

For me, getting the right dose of coffee is critical. Not enough and I can't concentrate. Too much and I get jittery.

The bottom line is to get lots of practice. Learn to be quick and don't be willing to settle for anything less than an ideal joint: fully wetted with no peaking.

A good stereo microscope is probably the best place to spend limited money. I bought mine used for a couple hundred bucks. I can do fine pitch parts without one, but usually get a headache. Also, I'll mention that I find magnifying lenses to be worse than nothing at all. I have yet to find one with a steady enough holder.

Anybody that wants to play with new technology components has to come to terms with working on fine-pitched devices. There is virtually nothing new being introduced with old-style packages.

My next challenge is figuring out how to deal with BGA and chip-scale packages at home.

Quote:
3) Brian, as I understand it you are not buying the EVM, but designing you own PCB? Are you considering posting you PCB's files here? I could use a starting/branching point...
I'll have to answer that with a definite Maybe...
First I want to get something that works to my satisfaction.
Frankly, I suspect that someone wanting to do anything other than an exact copy of one of my boards would do just as well to start with TI's reference layout.

Quote:
4) It seems like the TacT Millenium uses quite large Jensen air core foil inductors placed close to the speaker terminals. I think it could be a good choice if the distance from the switching stage is not a big issue. They seems bigger than some 10uH though?
(pics)
Jensen
Thanks for pointing this out.
I looked at Jensen's website. I think their inductors would be useful for an Equibit output with discrete transistors. I'm afraid they have too much series resistance for the integrated TAS5110.

Their four pole electrolytics also look promising for later higher powered designs.

Quote:
5) What are you considering for high quality volume control? Are you all planning a digital attenuation (potentially decreasing the resolution) or is anybody have ideas for a H.Q. variable power supply?
I discussed this somewhat on earlier posts in this thread. I currently am doubting that digital volume control will cause any meaningful decrease in resolution. To experiment with this, I'm still designing my first board so that I can vary the supply level.

Quote:
6) My plan is a totally digital amp with the reciever chip CS8420, which can use an external precision oscillator for clock generation. Any comments on the need for an very stable clock, and its placement?
Preference should be given to locating the clock near the Equibit modulator (i.e. TAS5012). Think of it as a DAC. I would still try to get the routing as direct and close to the CS8420 as possible.

Regards,
Brian.
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Old 18th December 2002, 01:04 PM   #26
Rookie is offline Rookie  Serbia
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Brian,

I have one more question to you, because it seems that you are the smartest guy on this thread.

One more thing about TAS5015 is not quite clear to me.
According to datasheet TAS5015 requires an external PLL, but when operating in master mode does this realy need to be a PLL or can it be a crystal oscillator? My understanding of this is that when operating in slave mode the high frequency clock signal on pin 6 must be phase locked to LRCLK, but when operating in master mode there shouldn't be any specific phase relationship between HFCLK and any other clock signal, because the LRCLK, SCLK and MCLK are outputs derived from HFCLK.
Is this correct?

I'm interested in master mode operation because I would like to use TAS5015 with AD1896 asynchronous sample rate converter.

Best regards,
Dejan
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Old 18th December 2002, 01:40 PM   #27
Rarkov is offline Rarkov  England
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Quote:
Originally posted by Rookie
I have one more question to you, because it seems that you are the smartest guy on this thread.
A clever way to get help from one person at the expense of anyone else participating!
Gaz
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Old 18th December 2002, 01:45 PM   #28
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I'd like to throw in my 2 pence about the tradeoffs regarding the switching frequency of class-d amps (either digital like Equibit or the classic analog PWM).

It is indeed true that for a given filter the carrier supression is better with a higher switching frequency.
OTOH magnetic core materials that can handle high frequencies with low losses are less widely available than those for lower switching frequencies. I was once thinking about air-cored output filters (the Tact actually uses such) but then you have additionall EMC issues. A compromise would be the use of air-gaps as we did within a PWM amplifier 10 years ago.
The MOSFET and driver switching losses also increase with frequency.
A given absolute timing error means a larger relative timing error for a higher switching frequency and therefore increased distortion. OTOH a higher switching frequency allows for a higher unity-gain point and therefore more NFB.

Once it was thought that the complete (whatever that is) removal of the carrier is important but noone cares that much about that nowadays anymore, to which I can agree as well (our amp had a carrier suppression of > 80 dB).


Regards

Charles
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Old 18th December 2002, 03:32 PM   #29
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Quote:
Originally posted by Rarkov


A clever way to get help from one person at the expense of anyone else participating!
Gaz
I may have written the most in this thread so far, and I'll have to admit that I've probably gone off the deep end with the amount of time that I've spent on this project the last half year, but remember I'm still working on my first attempt. It'll be a major milestone for me just to get the thing functional with some sort of sound coming out. Only then can I start playing around with options to hopefully furthur improve the sound.

I really appreciate and want to hear from others with switching amplifier experience, or that are experimenting with or thinking about digital amps.

Before I recently joined this forum, I had assumed that there would be many people who had already used TI's Equibit chips. Digikey has been selling them for over a year now! (Which I've found to be one of the best indicators that a new component or technology has become accepted.) It's really surprised me that I haven't been able to find a single person (even outside this forum) who has already built a home-brew version of one of these.

It's certainly not because a digital amp is more ambitious than other projects people are doing here. I see other people doing far more advanced things taking even greater efforts.

Perhaps it's because this is still kind of new. I'm very optimistic about the potential that digital amps offer for achieving new levels of sonic performance. I would think many people would want to try it out.

So please, speak up!
Better yet, start playing with this stuff.

Regards,
Brian.
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Old 18th December 2002, 04:39 PM   #30
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Quote:
Originally posted by Rookie
Brian,

One more thing about TAS5015 is not quite clear to me.
According to datasheet TAS5015 requires an external PLL, but when operating in master mode does this realy need to be a PLL or can it be a crystal oscillator? My understanding of this is that when operating in slave mode the high frequency clock signal on pin 6 must be phase locked to LRCLK, but when operating in master mode there shouldn't be any specific phase relationship between HFCLK and any other clock signal, because the LRCLK, SCLK and MCLK are outputs derived from HFCLK.
Is this correct?

I'm interested in master mode operation because I would like to use TAS5015 with AD1896 asynchronous sample rate converter.
This is very correct.
I'd only realized this point a couple of weeks ago.

Using the AD1896 ASRC allows the digital amp to be a clock master, so as you point out it's possible to use an oscillator instead of an external PLL with the TAS5015. The TAS5015 can then generate all of the other audio clocks in Master mode.

I noticed something else as well:

HFCLK (p6 on TAS5015) is NC on TAS5012.

All of the pins associated with the internal PLL (p3, p4, p45, p46) on the TAS5012 are NCs on the TAS5015.

This means I can lay my first board out to use either the TAS5012 or the TAS5015! I can place oscillator footprints right next to p6 (for the TAS5015) and p1 (MCLKIN on the TAS5012).

Normally, for a 96KHz or 192KHz sampling rate:
The TAS5012 requires a 12.288 MHz oscillator.
The TAS5015 requires a 98.304 MHz oscillator.

The 12.288 MHz is a reasonably standard value. I'm going to use a CTS CB3LV-3C-12.2880-T ($5.08 from Digi-Key). This part is spec'd to have <1pS jitter.

98.304 MHz isn't a standard value. This leaves three options:
1.) Use a custom part (not now for me, at least).
2.) Use a programmable oscillator (I don't know of any with as good jittter specs - this defeats some of the purpose of what we're trying to achieve).
3.) Operate the system at a non-standard frequency (the AD1896 makes this possible).

Option 3 seems like the clear choice to me, but there's a problem:
The closest standard oscillator frequency to 98.304 MHz is 100MHz. This would produce a system fs of 97.656 KHz or 195.313KHz, with an SCLK frequency of 12.5MHz. The TAS5015's data sheet specifies 12.288MHz as the maximum SCLK. I'm guessing that this is listed because it's a standard value and that the absolute limit of the chip is somewhat higher. As long as the other setup and hold time requirements are met by the other components in the system, my guess is that it shouldn't be a problem. I guess it all depends on whether or not you're ready for membership in Ov&rcl@ckers Anonymous.
The next standard frequency down in ultra-low jitter oscillators is 90MHz. I think this might be a significant audio quality hit to fs, but maybe not.

So that's where it's at with me. This is one of the things I plan to play around with, take measurements, and listen to.

Regards,
Brian.
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