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#171 |
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diyAudio Member
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It matters not whether there is a much of a correlation between the input signal and the signal doing the modulating - it'll still mix 'em. I realize we're not looking at a full Gilbert cell here, just a simple dual-quadrature mixer but the math tells you what the outcome should be. Gilbert Cells
Single sine waves are one way to probe the workings of such mixers but a music signal will result in a much more complex interaction. The output passes on only the differential to the load so the common-mode modulation frequency isn't there - but the sidebands will be there as far as I can tell. Anyhow, it may not matter, the IM products may be low enough at low power and under the control of the NFB that it's a non-issue or it may even enhance the sound. There may be a sweet spot in terms of how it's set up regarding idle currents.
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"The test of the machine is the satisfaction it gives you. There isn't any other test. If the machine produces tranquility it's right. If it disturbs you it's wrong until either the machine or your mind is changed." Robert M Pirsig. |
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#172 | |
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diyAudio Member
Join Date: Sep 2006
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Quote:
It works in multiple and subtle ways. If you look at the performances of the "raw" amplifier, you can expect the main distortion product, H3 to be 70dB down. Normal for a three stage amplifier without gain enhancement or fancy techniques, using crappy output transistors. When the tail modulation is present (~6%pp at the full power output), it will add a further -71dB of H3. Thus, will the resulting distortion be -64 to -65dB H3? The modulation current is in fact an error signal from the output. It tells the previous stages to increase the conductance of the output stage. This is the closed-loop part. But this signal can also serve as a "helper" in an open-loop, feedforward fashion, and it has the right phase to compensate, at least partially the compression, gain droop etc of the "noble" path, including the small and medium-signal part: the transfer of a differential pair is not a linear function, but a tan h function. In the end, the result of the compensation will be a composite H3 level of ~-90dB. Other harmonic and IM products will follow the same trend, because the compensation doesn't work in the frequency domain, it globally straightens the transfer function of the amplifier in the amplitude domain. The compensation is not as accurate and as deterministic than in the Unigabuf f.e., but it does provide valuable improvements. Another important aspect to understand is the way the modulation signal affects the functionning of the amplifier, in particular its class. Let us establish the convention that in order to let +1A of current flow through the load, the upper device must provide +1A, and for -1A output, the lower device has to deliver +1A. In a class B push-pull, when the current to the load is +1A, the current is provided by the upper transistor, and the lower one is off. In fact, it is "more" than off: it provides a virtual current of -1A. Or at least, it is biased that way by the driver and the preceding stages. But because it is a unidirectionnal device, it is simply off. Not let us see what happens in the Circlophone under the same conditions. The real time modulation of the current will make sure the lower device passes a minimum of 100mA; the upper transistor will have to provide 1.1A, and the resulting current to the load is again 1A. But the instantaneous quiescent current is now 0.6A. This is class A. That is why I said the circuit is a quasi-class A: it is an adaptative class A, in fact it never works in class B or class AB, it just gives that impression. But it has the attributes of class A, in particular the "grip" both output devices have at all times on the load: it never works in a pure source or sink, open collector mode. There is one more side benefit to this mode of operation: the excursion seen by the amplification chain is not comprised between A- and A+, total 2Ap.p. (clipped at the output), but |0 to A|, amplitude Ap.p. with the supplement provided "parametrically" by the modulation (pump) current that dynamically shifts the operating point. The raw linearity improvement brought by the modulation current is ~20dB, but there is more to it than a mere improvement in figures.
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#173 | |
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diyAudio Member
Join Date: Sep 2006
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Quote:
Here they are again: The silkscreen file is too large for the forum: http://dl.dropbox.com/u/5430178/circ...Silkscreen.pdf
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#174 | |
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diyAudio Member
Join Date: Feb 2005
Location: Watertown, NY
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#175 | |||
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diyAudio Member
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Hope you don't mind my curiosity !
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"The test of the machine is the satisfaction it gives you. There isn't any other test. If the machine produces tranquility it's right. If it disturbs you it's wrong until either the machine or your mind is changed." Robert M Pirsig. |
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#176 | |||
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diyAudio Member
Join Date: Sep 2006
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Quote:
The current servo has to react when the current drive is insufficient to keep both output devices simultaneoulsy active. This is partly caused by the transconductance of the (relatively) inactive device being minimal, gain compression in the active device, and tan h distortion in the driver LTP. This means that the current-servo error signal is heavily correlated to gain errors in the amplification chain. That's why it is used to create an open-loop correction, that acts by modulating the gain, increasing it for both positive and negative excursion. As with most open loop servo's, there is a limit to the level of correction that can be achieved: there are accuracy issues, and the shaping is not optimal, but as long as the polarity is correct and the amplitude is more or less in the right range, the improvement in linearity is valuable. The only serious drawback of the scheme is the possible leaking of the correction signal into the noble path, due to the bias current of the input transistors combined with unequal impedances seen by the bases. That's why I warned the impedance at the input has to be under control. Quote:
I say it is improved compared to standard class AB: If the quiescent current is fixed at some optimal value, say 100mA for example (thus not class A), with the dynamic modulation completely turned off, the distortion shoots up to ~0.03%, which is a normal value for such an amplifier. I didn't make any tests, but I suppose that if the amplifier was operated in full, static class A, the distortion would be comparable to or lower than that of the Circlophone, but of course, you would then need much larger heatsinks... Quote:
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#177 |
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diyAudio Member
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I tried running the simulation file, but it's looking for device models I don't have. This isn't a big issue as I can substitute with other models, but if you were to re-post it with the device models included that would allow people to reproduce your results. The way I do this is copy the text from the device model onto the 'clipboard', switch to spice, hit 'S' and past it in so that the model is directly in the schematic.
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"The test of the machine is the satisfaction it gives you. There isn't any other test. If the machine produces tranquility it's right. If it disturbs you it's wrong until either the machine or your mind is changed." Robert M Pirsig. |
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#178 | |
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diyAudio Member
Join Date: Sep 2006
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I think all other semi's are in the original LTspice libraries.
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#179 |
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diyAudio Member
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Thanks ! -- I have it up and running.
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"The test of the machine is the satisfaction it gives you. There isn't any other test. If the machine produces tranquility it's right. If it disturbs you it's wrong until either the machine or your mind is changed." Robert M Pirsig. |
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#180 |
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diyAudio Member
Join Date: Nov 2008
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Elvee:
I looked for it in all these posts (179, I almost went blind), but could not find it: The Circlophone produces about 20+ watts as shown in the current schematic. Changing the output power requires what changes to the circuit? I believe 25/50/100 watt versions would be of interest to the Circlophonites. Could you post suggestions to the parts value changes? Thanks, E |
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