And the question is?
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I'm not sure exactly what the question is myself, but I do use an Avid Pulsus. You need to use at least the 60db setting (or even the 70db setting) on the phono stage for a cartridge with that low an output.
I would forget about the "10:00 - 2:00" rule and use what's best for your cartridge. It also sounds like your amp produces more noise than the Pulsus.
I think he is measuring mainly white thermal noise as opposed to 60/120/240 Hz hum (via the tweeter).
I found something similar with the PS Audio GCPH. That pre-amp has a low noise input stage and a gain cell buffer/output stage. The gain on the input stage is determined by a set of fixed gain settings (48/54/60/66 dB) and the gain cells have a volume control. So the input stage is analogous to your the pre-amp and the gain cell similar to the line amp.
I found that the lowest noise (as measured by my Tektronix 7L5 spectrum analyzer) was found at the higher input gain settings, minimizing the output stage gain. In other words, the output buffer gain stages were noisier at higher gains than the input stage was. In fact, with gain increases of 6 dB of the output gain cells, I gained 8-12 dB in noise. But this trend was very non-linear, and this effect occured only over the last 6 dB of gain in the gain cells. Below that, the noise was dominated by input stage gain.
05-22-12: Dhl93449Hi DHL,
I realized that he is not measuring hum. Note that I said "even though no hum occurs." Ground loop effects can cause noise at high frequencies, as well as low frequency hum. And it is conceivable to me that the former could happen in the absence of the latter.
If high frequency distortion components and/or noise that are on the AC line were to couple through stray capacitances onto the chassis and/or the circuit grounds of each component, to a degree that is unequal in the two components, and if the components have their chassis and circuit grounds connected together (or connected together through a low impedance), and if 60 Hz and its lower order harmonics are not able to so couple (to a significant degree), that is what would happen.
See this reference, and keep in mind that the interface between the particular components that are involved is unbalanced:
Note this statement:
The noises originating with the power line are generally described as EITHER [emphasis added] "hum", which is predominantly 60 Hz, or "buzz", which consists of a mixture of high-order harmonics of 60 Hz. These harmonics are the result of power line waveform distortion, which commonly reaches 5% THD and is caused by many types of non-linear power line loads. Because the human ear is much more sensitive to frequencies in the 2 kHz to 5 kHz range at these very low levels, buzz is usually more audible than hum, even though the hum level may be electrically larger.RFI/EMI pickup in the cables is also a possibility, as I indicated.
Thanks for the responses, of interest, and an indirect question, was based on the SNR published for the Acurus RL-11 of 95db A Weighted. The Avid Pulsus on the other hand has a noise spec from -81db to -67db. I have plenty of gain so operating the Pulsus at 48db of gain, even though the XX-2 is 0.28mv, still gives me the “typical” preamp range of 10:00 – 2:00. If strictly going by that rule, I would have set up the Pulsus gain that way. If you look at the SNR in that configuration it is SNRdb = Psignaldb – Pnoisedb. SNRdb = 114db – 60db = 54db. In the Pulsus 60db gain configuration. SNRdb = 114db-52db = 62db. Looking at the specs, Pulsus being noise floor at gain, and Acurus being SNR, it is not apparent to me how you would surmise this result analytically, without actually measuring.
It looks like Dhl93449 hit it on the head when he said
“I found that the lowest noise (as measured by my Tektronix 7L5 spectrum analyzer) was found at the higher input gain settings, minimizing the output stage gain.”
A related question, has anyone actually measured their SNR using this method? I believe I have a quiet system and I also believe 62db is a good result, but what is a reasonable expectation?
The reason the SNR specs for the Pulsus (and most other phono stages) worsen as the gain setting is increased is probably that they are defined relative to a reference level at the output of the phono stage that is the same, regardless of gain setting. Therefore the measurement for higher gain settings will be taken at a lower input signal level, reducing SNR in the circuitry at the front end of the phono stage. That in turn figures to generally be the most critical point in the signal path with respect to overall SNR performance, since signal levels are lowest there.
In contrast, your experiments involved a fixed input signal level. So the SNR differences you obtained for the various gain settings presumably resulted from differences in the relation between signal levels and noise contributors at points in the signal path between the phono stage's gain adjustment provisions and the volume control in the preamp (inclusive). As you indicated, the net result of all of that is probably not analytically predictable.
Regarding comparisons between SNR specs for the phono stage and the preamp, while in general it can be expected that the SNR performance of a good line stage will be superior to that of a good phono stage, I would comment that a direct comparison between the numbers is pretty much meaningless in the absence of specified reference levels. Unfortunately, SNR specs are commonly presented without any indication of what signal levels they are referenced to. That is the case for these particular components, as can be seen here and here.
I understand what you are pointing out, but higher harmonics of the 60 Hz fundamental will be significantly decreasing in amplidtude as you get to higher orders.
One way to determine this experimentally would be to observe the noise at a given volume setting (of the line amp), then disconnect the interconnects and replace them with (a) shorting plugs or (b) better yet shorting plugs with resistors simulating the source impedance of the phono pre-amp. If the noise levels drop considerably in this test, then perhaps you are right and the increased noise levels are due to ground loops or IC issues. If not, they are simply the thermal noise generated by the various stages of amplification.
Everyone tends to assume that line stage noise is insignificant compared to phono stage noise, but depending on the relative gain (and other issues such as source impedances) this may not always be the case.
Also, one needs to abandon rules of thumb based on the rotational position of gain controls. There is no reference standard here, and the gain vs position is highly product design dependent.
Regarding my spectrum analyzer measurements, I was looking at the spectrum from about 50 Hz- 20k Hz. I did not apply any weighting (A or otherwise) to my measurements as the Tek will not do this (it is a vintage 1980's SA but has a S/N/distortion capability of -130 dB).
Re your question of acceptable levels, this is a bit subjective. For me, -60 dB would not be good enough, as I can easily hear this level from my listening position at the highest sound levels I listen to.
You can check out my post "Impressions of the PS Audio GCPH" under the amps/pre-amps section for the actual numbers, but they were on the order of -90 to -100 dB (unweighted), depending on gain, referenced to a 1 volt output. You can add about 6 dB to those numbers since my normal line input levels (to the line amp) is 300-500 mV at max listening SPL. Even in the worst case of GCPH input gain of 66 dB and MAX volume gain control, noise was at -75 dB unweighted. This level is almost inaudible from my listening position.
I emphasize "normal max" listening volumes, which for my JC2 is 9-10 on the volume pot. If I crank up the gain to 12 or full CW rotation (4-5) the -75 dB levels become very audible.
Good points, at 0.00028v with 60db of gain with the Pulsus, gets me to 0.28 volts. My RM-9 MKII Amp at the high gain setting, lowest feedback (which I prefer) puts out max watts (125w/channel) at 0.4 volts (spec in original post). Assuming its proportional, then 87.5 watts = (125*0.28)/0.4. So from 0 – 87.5 watts the Acurus RM-11 preamp is attenuating the signal. From 87.5 watts to 125 watts only an additional 3.1 db is needed. So out of the 17.7db of gain available from the Acurus RM-11 the most that will be used is 3.1db and that is at very high listening levels. So with this configuration 99.999% of the time the signal will be attenuated by the Acurus preamp. There is a lot of discussion on this forum about passive preamps and “possibly” the loss of dynamics with those devices. However, many point out that it is due to the input and output impedance of the passive devices. In my case, I presume, even though the signal is attenuated, the input impedance of the Preamp is still very high (10Kohms) and the output impedance is still low (47ohms). If I recall the RM-9 MK II input impedance is also 10Kohms. So there should be no signal frequency interactions due to impedance mismatching. Therefore even though the signal is attenuated almost all the time I am getting the benefit of an active preamp, would you agree?
If I make the preamp more active, set the pulsus to 48db of gain. Then the 0.00028v cartridge signal exits the Pulsus phono stage at 0.071. Assuming proportionality, then 22 watts = (125*0.071)/0.4. Therefore the Acurus RM-11 preamp is attenuating from 0-22 watts and from 22 watts – 125 watts it is providing 0db – 15.1 db. The preamp range is from 0 – 17.7 db, so on paper these seems the way to go, however, in my Sound Pressure Level measurement, it produced more white noise.
For reference, when cranking up the volume the noise I hear is pink in nature. I won’t call it pink, since it covers a limited frequency spectrum, It seems like all frequencies are being crossed over to the tweeter. I have no buzz or hum, which I am thankful for, since I have spent a lot of time in the past trying to track down those problems in a different system configuration. This noise is inaudible when I play a record or lift up the tonearm at almost all levels from normal listening positions.
Thanks for the reference, I will look up the GCPH post. Yes as far as my Sound Pressure Level test, it was very limited. First the lowest measurement on my SPL meter is 55db, hence I had to crank up the volume to 114db to even get a noise measurement. And there was only one sample 114db. A multi-meter test with multiple samples at the speaker output terminals would be a much better test. Maybe I will do that today, for the heck of it. It would be interesting to calculate the noise level using voltages at normal listening levels and see what the resulting SNR is. That would compute total system performance without the limitations of my SPL meter and the air gap variable.
Therefore even though the signal is attenuated almost all the time I am getting the benefit of an active preamp, would you agree?Yes, certainly. Although there are some rare exceptions, nearly all active preamps have the volume control mechanism at a point in their internal signal path that is "ahead" (upstream) of their output buffer stage. So for nearly all designs turning the control down will have no effect on output impedance, and will not convert the active preamp into a passive one.
Which is not to say that the setting of the volume control won't have any effect on sound quality, particularly if it is set to introduce a lot of attenuation.
When cranking up the volume the noise I hear is pink in nature. I won’t call it pink, since it covers a limited frequency spectrum, It seems like all frequencies are being crossed over to the tweeter. I have no buzz or humSo it sounds like the ground loop effects I indicated as a possibility are probably not a significant factor. A simple way to verify that, though, would be to use a "cheater" (a 3-prong to 2-prong adapter) to temporarily isolate the AC safety ground pin of the power plug of either or both of the components.
Some minor corrections to your math. With respect to the amplifier, what is directly proportional is the relation between output voltage and input voltage. Therefore output power varies in proportion to the square of input voltage. So 0.28V in would result in about 61W out. Also, 125W/87.5W is about 1.5 db, not 3.1 db. 125W/61W is about 3.1 db. For a given load, as you probably realize:
db = 10log(P1/P2) = 20log(V1/V2), where P is power and V is voltage.
Also, keep in mind that 0.28 mv represents the output of the cartridge under specific test conditions, and that figure may reach significantly higher levels on the peaks of some recordings, as well as often being at much lower levels.
Continued good luck in your experiments! Regards,
Yes, I was thinking about your measurements, and at 55 dB SPL you are nearing ambient levels, depending of course on your house and environment. If you are picking up ambient contributions, your baseline noise floor of your audio system may be lower than what your SPL meter indicates. Food for thought.
That is why I like the direct measurement with a SA. You can also make this type of measurement with a distortion analyzer, but the SA has the advantage of showing you graphically exactly what frequencies are contributing to the noise levels, especially useful if you have 60/120/240 Hz line related contributions.
Line noise is particularly insidius due to the emphasis of the RIAA EQ curve. Remember that when you speak of phono pre-amp gain, it is usually referenced to 1 K Hz, which is right in the middle of the EQ curve. 50-60 Hz signals (noise or otherwise) entering the phono pre-amp inputs are amplified by 20 db (a factor of 10x) over the 1 KHz level, so any low frequency noise in this region is amplified considerably. Likewise, very high frequency noise is attenuated by the same 20 dB in the other direction.