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 Normal input impedance values
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 20th February 2013, 10:20 PM #1 diyAudio Member     Join Date: Sep 2005 Location: Montréal, Canada Normal input impedance values I'm building an LM3886 chipamp, non-inverting. It has a 10k resistor (Rin) from the non-inverting input to ground. This will be fed by a first-order RC low-pass filter to keep out the RF garbage. I'm looking at a 2 MHz corner frequency. As Rseries gets large, and C gets small, then the loading of Rin will have an effect on the RF filter, which makes analysis more complicated, but for this to be a concern, Rseries will have to be so large as to cause undesirable attenuation in the audio band. As a rough approximation, the C//Rin combination alone will determine the phase between the input current and input voltage at 20 kHz. For a given corner frequency, as Rseries decreases, C must increase, which increases the phase angle between the input current and voltage. Is a line-in generally expected to be resistive? What amount of phase angle is considered normal at 20 kHz? When can I expect problems? At 20 kHz: 82 pF // 10k = 6 deg 100 pF // 10k = 7 deg 150 pF // 10k = 11 deg
 21st February 2013, 03:35 AM #2 diyAudio Member   Join Date: May 2008 "Line in" simply means that an input level of one volt is expected. That's one volt approximately, very approximately. All audio inputs are expected to be resistive, or more correctly to have impedance, which is the AC version of what DC calls resistance. Resistance does not vary with frequency (at least not in the audio band), but impedance does, and that's the difference. The general rule is low output impedance (source) feeding high input impedance. Input impedance approximately 10 times output impedance. However, you usually have no idea what output impedance a device might have, so the rule of thumb is go large. Without false modesty, you might do worse than to take a look at the input circuit of my own li'l mixer-amp. Just look at input Line1a: http://www.diyaudio.com/forums/attac...d_3b_iview.jpg Here the source sees an input impedance of 24k, which is the value of Ra parallel to Pot1a (neglecting the reactance of C1a, which is OK in this case). Wiper position doesn't affect the impedance seen by source. It does affect the impedance seen by the LM3886, but in this case that doesn't matter. Frequency response is flat 20-20k Hz. Phase error is very near zero degrees 20-20k Hz. If you choose to use the circuit, replace the data sheet's Rin with my network of Ra, C1a, and Pot1a. The wiper of Pot1a then directly feeds the data sheet's RB, which is a protective component and can be left as is. Done dealie. All of this is assuming that you'll have a line level input to the amplifier, which again means something very vaguely in the vicinity of a volt. If not a preamp stage will be needed. Nearly everything that's self-powered (battery or plug-in) can be assumed to have a line-level output, usually including headphone outputs. A 24k input impedance will deal very nicely with all of these. PS: Capacitors and inductors (coils of wire) display reactance, which is once again their AC version of resistance, and varies with frequency. Resistors do not, in the audio band, display reactance, nor does their impedance change. You never speak of resistor "impedance," you always simply say resistance, stated in ohms.
 21st February 2013, 05:06 AM #3 diyAudio Member     Join Date: Sep 2005 Location: Montréal, Canada I used the term "resistive" to mean that the input impedance is mostly real; that is, dominated by resistance. How did you choose the 47k input coupling capacitor drain resistor value? The 47k is in series with the 50k pot, and so with the 22 uF capacitor, gives a 2.1 second time constant for draining. Is that a general guideline?
 21st February 2013, 02:10 PM #4 diyAudio Member   Join Date: May 2008 << How did you choose the 47k input coupling capacitor drain resistor value? >> It's semi-arbitrary. The idea is to keep resistor values low (to stay away from Johnson Noise), but keep input impedance high. These contradictory goals are, in my opinion, served somewhere around the 24k point. I wouldn't say anybody was wrong if they cut this to around 10k or even to 5k or perhaps lower. Always bearing in mind that you want an input impedance around ten times source's output impedance--and you usually don't know source's output impedance. << The 47k is in series with the 50k pot, and so with the 22 uF capacitor, gives a 2.1 second time constant for draining..Is that a general guideline? >> Ra and Pot1a are actually in parallel. Hence the 24k input impedance: 47k in parallel with 50k. That is, they're parallel as far as the audio signal is concerned, since it passes freely (in theory) through capacitor C1a. DC is not significant, of course, since C1a is there to block DC in the first place. In either/any case, no it's not a general guideline for anything. Except that in my view it's a workhorse input circuit that will serve generally except for piezo or magnetic instrument pickups, which want to feed an input impedance of something like half a meg, or better still a meg. But don't go by me. To an old tube head like me there's no such thing as a drain resistor anyway. To me Ra is a load resistor. It's there for insurance because, again, you usually don't know anything about source's output circuitry. Ra insures that there is a path to ground, therefore there will be current flow, therefore there will be voltage at the top of Ra, and this is the voltage that will be amplified. At the same time the 24k impedance is high enough not to be a significant load on source...we hope and assume. The actual time constant to drain C1a is, in my view, not significant. When the amp is turned off the capacitor will discharge in any case. Or actually, even if it didn't no harm done.
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Quote:
 Originally Posted by pmbrunelle I'm building an LM3886 chipamp, non-inverting. It has a 10k resistor (Rin) from the non-inverting input to ground. This will be fed by a first-order RC low-pass filter to keep out the RF garbage. I'm looking at a 2 MHz corner frequency. As Rseries gets large, and C gets small, then the loading of Rin will have an effect on the RF filter, which makes analysis more complicated, but for this to be a concern, Rseries will have to be so large as to cause undesirable attenuation in the audio band. As a rough approximation, the C//Rin combination alone will determine the phase between the input current and input voltage at 20 kHz. For a given corner frequency, as Rseries decreases, C must increase, which increases the phase angle between the input current and voltage. Is a line-in generally expected to be resistive? What amount of phase angle is considered normal at 20 kHz? When can I expect problems? At 20 kHz: 82 pF // 10k = 6 deg 100 pF // 10k = 7 deg 150 pF // 10k = 11 deg
I think that you will want a cutoff frequency of just a few hundred kHz. I would keep the series R as low as is practical; maybe 100 Ohms to 500 Ohms (?). I am not sure that the phase angle between the current and voltage would matter. What might matter would be differing phase angles for different frequencies, for the voltage (or current).

Make sure that the filter is close to the input pin.

You should also consider the filtering of all of the other inputs, i.e. output and power. Everything is an input, for RF. If your decoupling caps are large enough, then the impedance of the power rails might form a sufficient low-pass filter, for each of the DC power inputs. The output should probably have a small resistor and a parallel inductor.

These matters are discussed fairly well in chapter 7 of Walt Jung's (et al's) book, Op Amp Aplications Handbook, available as a free download from Analog Devices' website, at:

ADI - Analog Dialogue | Op Amp Applications Handbook

Last edited by gootee; 22nd February 2013 at 06:17 AM.

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Quote:
 Originally Posted by picbuck Wiper position doesn't affect the impedance seen by source. It does affect the impedance seen by the LM3886, but in this case that doesn't matter.
A TL071-family opamp is a FET opamp, so its inputs don't draw much current. Quite different than an LM3886.

The LM3886 bipolar inputs can draw up to 1 uA. Through a 50k resistance, that's up to 50mV of DC offset at your input, which will diminish to zero as the wiper touches ground!

Quote:
 Originally Posted by gootee I am not sure that the phase angle between the current and voltage would matter. What might matter would be differing phase angles for different frequencies, for the voltage (or current).
I thought maybe the phase between the voltage and current might matter, that a music source might not like driving a capacitive load, but I guess it's not really a problem.

I just looked at some of Rod Elliot's amplifiers; many of them have a 22k//220pF at their input.

At 20 kHz, that typical Rod Elliot input impedance has the current leading the voltage 30-ish degrees, way more capacitive than what I was considering above.

I brought up the subject because it's possible to have quite different voltage-current phase seen by the music source, while having a similar output/input transfer function.

Normally, I'd have a lower RF low-pass corner frequency, but this chipamp is destined for speaker testing duty! Thus, it is preferable to keep the phase shifts and magnitude changes in the audio band to a minimum. And whatever non-flat frequency response there is must be repeatable (i.e. not change with the volume control). DC offset that changes with the volume control is also not desired, but that's mostly for anal-retentive reasons.

You have to be careful with a low-pass that uses a small series resistor, because if you have a 100R series resistor with a say 100 kHz corner frequency, that corner frequency becomes 10 kHz once you connect an el-cheapo 1k output impedance music source, so using a larger series resistor reduces sensitivity to that sort of thing.

That's quite an e-book!

This is what the LM3886 speaker tester amplifier is looking like so far:

I'm going with the datasheet recommended output inductor + resistor for now, since I don't understand enough to change these parts.
Attached Images
 Amplifier-1536.jpg (239.9 KB, 191 views) PSU-1536.jpg (234.6 KB, 187 views)

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Well, while I'm at it, I might as well show the PCB layout, and the rest of my speaker tester schematic, so the PCB will make some sort of sense.

I'm working with the freebie version of EAGLE, so that's why some things are drawn in MS-Paint; the free version is limited in terms of board area, so some components aren't there, pads just represented with vias. And the mounting holes also look like vias; this is so their position gets transferred to the copper when I do the toner transfer. Soooo, excuse the hackjobbery!

It will be dual layer (my first), but there are some limitations, such as no thru-plated holes, and the bare copper traces on the top have to be routed where they won't accidentally short component leads, etc.

With the current layout now, the ceramic PSU caps are ~13 mm away from the chip pins. And that's not including the lead lengths! Is leaving potentially ineffective ceramic caps a don't care thing (if they do no harm, I'll leave them for the feel-good feeling), or is it a bad recipe for resonance issues with the large electrolytics, in which case, they won't be included.

The focus on loop areas is really a convenient trick. At the end of the day, the only things that can affect a wire are the electromagnetic fields at the wire. However, there's Stokes' Theorem from calculus, which relates what's happening within the loop area to what's happening along the closed loop. Saves us some brain-scratching, and gives us a simple rule of thumb to follow (minimize loop area).

I get that's there's the rule of thumb to minimize loop areas. My problem is a fundamental lack of understanding of electricity and magnetism. This makes reducing loop areas difficult. My speaker tester here has maybe ~30 nodes (I didn't bother to count them). With 30 nodes, I could probably draw over 100 unique current loops, maybe more? I'm not sure which loops are important, and which ones are not. Minimizing the area of one loop increases the area of another...

So which loops are important? I realize I won't gain my own judgement for this overnight...

In general though, I tried to arrange so that the sum of currents in any wire bundle is zero.

I just did the rough PSU ripple calculation, assuming the capacitors get recharged at 120 Hz (60 Hz full-wave rectified), and that the diodes only conduct for a short time at the peaks.

If I'm drawing a constant current (makes for the nice linear discharge curve, yay for high-school math!) of 7A (LM3886 max current limit), then I'll have 5.2 V peak-to-peak of ripple (on one rail). It's a lot of ripple, but that's not a normal operation condition, that's an abuse condition that should just be semi-survivable, and the 2x 4A ripple current rated electrolytics should handle that. Plus I have the EAGLE board real-estate problem Oh well, can't complain, given what I paid for it.

I looked at your transformer spreadsheet, and since I'm not going to understand it anytime soon (a lot of numbers, everywhere), I'll just post now instead...

Description of the speaker tester (for use with Speaker Workshop, but possibly other software):

Signals from a computer sound card output are buffered by an LM3886 power amplifier.

Two basic functions:

1. Measure the impedance of speakers and stuff using a voltage divider with a known series resistance. The two unknown voltage divider nodes are measured by the sound card's left and right inputs, so the unknown impedance can be solved.

2. Take acoustic measurements. While playing a signal through a speaker, capture the acoustic response with a microphone (I have a separate pre-amp). The left soundcard input measures the amplifier signal, and the right input measures the acoustic signal. Comparing the two, the speaker's acoustic response can be measured.

Note: there are three unlabelled solder pads near the middle bottom of the board. This is for the 100 ohm pot.
Attached Images
 Speaker Tester Layout.png (188.2 KB, 162 views) Speaker Tester Layout - Labelled.png (311.2 KB, 161 views) Speaker Tester-1536.jpg (255.6 KB, 158 views)

Last edited by pmbrunelle; 23rd February 2013 at 08:46 AM.

 23rd February 2013, 09:07 AM #9 diyAudio Member   Join Date: Nov 2006 Location: Indiana Blog Entries: 1 The important loops to minimize are: 1. The AC mains pair 2. The secondary pairs 3. The rectifier output pair(s) 4. The input signal and ground pairs See Faraday's Law (Maxwell's Equations). The first three will transmit and the fourth one will receive.
 23rd February 2013, 09:58 PM #10 diyAudio Member     Join Date: Sep 2005 Location: Montréal, Canada With some thought, Ive decided that 11200 uF per rail wasn't really adequate. Too much droop, especially given the (low) +/- 20 V rails, so I revised this to 30000 uF per rail (3X 10000 uF). The board dimensions grew slightly. 20 V sounds low, but it's in case I decide to connect a 2 ohm speaker, or decide to apply a rail-to-rail square wave to any of the resistors connected to the LM3886's output; the resistor power ratings necessary become excessive. The idle heat dissipation also goes down with lower voltage supplies. I modified the traces around the bridge rectifier to reduce loop area there. I suppose a one-piece rectifier is better in terms of stray magnetic fields, but after a certain current level, they need a heatsink, which is a pain in the butt. With the exposed surface area of 4 discrete diodes, you can rectify more current before needing to worry about heatsinking.

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