Phono preamp headroom - why?

[atmasphere]

Is the frequency dependence of many of these stages a function of the topology? That is, is this the result of RIAA in the feedback loop, or RIAA between two gain states, or something else entirely?

This sort of dependence seems to be a recurring theme among many stages tested here.

If the RIAA de-emphasis is done through feedback, you might run into a problem if the circuit lacks the Gain Bandwidth Product to support that feedback. That could result in rising distortion with frequency. You might also run into a problem where the input of the phono section (depending on topology) isn't inside the feedback loop. So if it also has a limited Voltage swing, might be easily overloaded by an ultrasonic (or RF) signal 20 or 30dB higher than the output level of the cartridge which can be a thing, depending on if your cartridge is high output (MM with high inductance; 20dB peak) or LOMC (low inductance with high Q value, so 30dB or so peak in the RF region).

 

[cgallery]

If the RIAA de-emphasis is done through feedback, you might run into a problem if the circuit lacks the Gain Bandwidth Product to support that feedback. That could result in rising distortion with frequency. You might also run into a problem where the input of the phono section (depending on topology) isn't inside the feedback loop. So if it also has a limited Voltage swing, might be easily overloaded by an ultrasonic (or RF) signal 20 or 30dB higher than the output level of the cartridge which can be a thing, depending on if your cartridge is high output (MM with high inductance; 20dB peak) or LOMC (low inductance with high Q value, so 30dB or so peak in the RF region).

If one is using a MC cartridge into a step-up and then a MM preamp with limited (let's say 35db) gain, is the ultrasonic/RF concern somewhat mitigated? Or does the ultrasonic/RF get stepped-up by the step-up transformer?

 

[atmasphere]

If one is using a MC cartridge into a step-up and then a MM preamp with limited (let's say 35db) gain, is the ultrasonic/RF concern somewhat mitigated? Or does the ultrasonic/RF get stepped-up by the step-up transformer?

Yes to the former and no to the latter. Stepup transformers don't have the bandwidth to pass a 1 to 5MHz signal. I've no doubt this is why so many people say LOMC sounds better though a stepup rather than direct (if the gain is there to support it). But if the phono section is immune to the RFI problem caused by LOMC cartridges then its a different matter.

 

[JP]

Perhaps someone with flat conversion could look at the spectral content of some clicks/pops? @JP?

These are 96kHz if you want to count samples.

Dropbox

www.dropbox.com

Dropbox

www.dropbox.com

 

[atmasphere]

Y'all are not looking at the basic problem.

Its not the tick or pop that's the issue. This issue is if the resonance that exists can overload the phono section. If yes, then it can generate a tick or a pop.

In the case of a LOMC cartridge, the energy of the cartridge tracking can set the high Q peak into oscillation. The resulting RFI can really mess with a phono section and one result is ticks and pops.

So we can have two very different phenomena that a have very similar symptom (ticks and pops); in both cases the ticks and pops are not on the LP surface or are much lower amplitude than how the phono section reacting to the overload suggests.

 

[JP]

These ticks and pops are on the LP. Have never experienced the phenomena you claim is so prevalent. Would love for there to be enough information given to reproduce it, look at the spectral content, etc. I believe I've asked before.

 

[markstein]

Y'all are not looking at the basic problem.

Its not the tick or pop that's the issue. This issue is if the resonance that exists can overload the phono section. If yes, then it can generate a tick or a pop.

In the case of a LOMC cartridge, the energy of the cartridge tracking can set the high Q peak into oscillation. The resulting RFI can really mess with a phono section and one result is ticks and pops.

So we can have two very different phenomena that a have very similar symptom (ticks and pops); in both cases the ticks and pops are not on the LP surface or are much lower amplitude than how the phono section reacting to the overload suggests.

From my understanding this "oscillation" does only happen when the phono preamp is clipping. Which is the reason we discussed:

1) How much headroom above a reference point (5cm/s velocity) typically is required to avoid that pops on the record clip the preamp.

2) What is the typical spectral nature of these pops caused by dust, dirt and scratches on the records, i.e. how much headroom the preamp should have at different frequencies.

Looking at the pops shown in previous posts and the relatively high energy across the frequency band, I would like to see sufficient headroom across the entire frequency range, also at high frequencies, in a phono preamp.

 

[markstein]

Addition:

3) It should be also measured and evaluated how "favorable" the preamp clips, should this happen unintentionally. Meaning, does it recover fast or create a bunch of distortion or oscillate?

Obviously @atmasphere has a lot of background knowledge and may contribute to developing a test methodolgy. @Michael Fidler is also encouraged to provide his feedback.

 

[Michael Fidler]

In terms of overload behaviour: it's not so much resonances or anything that trails the clipping, more the fact that the RIAA curve features almost 40dB of gain difference from the top to the bottom of the audio band. Therefore, any clicks and pops that push the preamplifier into overload above 10kHz where cartridge resonances and high-velocity surface anomalies exist create intermodulation/distortion artefacts that will be equalised and boosted 20-30dB or so.

While the anomalies on the surface themselves are quite innocuous, as they exist mainly above 10kHz viewing a flat waveform, they are usually assymetrical and provoke strong overload artefacts below 3kHz where we are much more sensitive with our hearing. This is especially true for preamps showing premature overload for higher frequencies (of which there are many tested on this forum) either due to an overly low RIAA feedback network impedance (causing the driver amplifier to go into current-limiting at HF as the impedance drops), or just passive RIAA designs pulled out of op-amp datasheets by the 'design gurus' of any particular audio company who are really just marketing men.

 

[markstein]

@Michael Fidler:
- Which headroom (dB) above 5cm/s do you regard sufficient, based on your experience?
- If I understand you correct, you are saying that - if a preamp's headroom does not drop with rising frequency and is sufficiently high - the "clipping behaviour" does not matter. Is that correct?

 

[Michael Fidler]

@Michael Fidler:
- Which headroom (dB) above 5cm/s do you regard sufficient, based on your experience?
- If I understand you correct, you are saying that - if a preamp's headroom does not drop with rising frequency and is sufficiently high - the "clipping behaviour" does not matter. Is that correct?

Based on my own experience and various articles I've read upon the subject, it's my view that overload margin should be at least 10 times the nominal cartridge output level, preferably plus 3dB all the way to 23dB or so. I aim for at least 75mV at 1kHz, and 350mV at 10kHz in my own MM designs, and make sure that no further HF constraints appear against an RIAA-pre-emphasised signal appear until at least 25kHz.

If the headroom drops with increasing frequency, then I would say that this falls under the category of 'clipping behaviour'. In discrete designs, or impedance-overloaded IC ones, this can be assymetrical which is even more undesirable as the likelihood of LF artefacts becomes higher even with programme material. Rail-sticking and phase inversion is also possible, but with proper audio devices such as the NE5532/NJM8802/LM4562/NJM2068/NJM4580 etc. it doesn't happen. If you have HF instability or other artefacts appearing when the device clips, then this is a sign of poor engineering. I always double-check this under various conditions to make sure clipping is just peak limiting and nothing worse. Some op-amps, such as the TL072, actually invert the phase in certain overload conditions, which produces a very loud crackling sound - most unpleasant!

EDIT - a couple of very good sources are Tomlinson Holman's work in the 1970s (well before Self - who seems to be awarded authorship of such ideas due to his prolific contemporary writing). See attached...

 

[markstein]

Michael, thank you for sharing your knowledge!

 

[levimax]

This thread has turned into a great source of information. Thanks to everyone for contributing. I have a question regarding RIAA done digitally vs analog in the phono pre-amp. I am using a MIC interface (preceded by correct impedance balanced transmitter) and then do the RIAA using FIR filters. It seems to work flawlessly and after reading through some of the information in this thread it seems like doing RIAA digitally solves a lot of potential issues. Outside of some latency, what are the potential disadvantages of "digital RIAA" and what do you need to watch out for to avoid problems?

 

[Michael Fidler]

This thread has turned into a great source of information. Thanks to everyone for contributing. I have a question regarding RIAA done digitally vs analog in the phono pre-amp. I am using a MIC interface (preceded by correct impedance balanced transmitter) and then do the RIAA using FIR filters. It seems to work flawlessly and after reading through some of the information in this thread it seems like doing RIAA digitally solves a lot of potential issues. Outside of some latency, what are the potential disadvantages of "digital RIAA" and what do you need to watch out for to avoid problems?

Just make sure that there's enough headroom in the system at HF for it to work well (probably best controlled by switched gain on the flat preamp stage). As already stated, if this stage or the ADC clips, then LF artefacts will be greatly amplified. It should be easy to monitor when this is happening as ADC overload. If you have latency, you could even include a look-ahead peak limiter before the final gain is applied in the digital domain to bring the subjective level up to contemporary digital recordings.

If you are concerned with phase and group delay, make sure the RIAA equalisation actually approximates a first order mathematical function in the time domain (rather than a flat group delay 'graphic EQ' style curve. In addition to that, I would recommend a 4th order filter at 20Hz to remove subsonic disturbances, as well as a much higher order filter at 25kHz as well (if the sampling rate is high) as MC cartridges can output quite a lot of HF distortion products that may cause trouble further down the signal path if your first-order RIAA function flattens after 20kHz...

For a microphone preamplifier, you want to make sure the loading is 50k/100pF to keep the response at HF flat. Many mic preamps have a low impedance of 10k or less, which will cause the response of an MM cartridge to droop by as much as a decibel at 5kHz.

 

[levimax]

For a microphone preamplifier, you want to make sure the loading is 50k/100pF to keep the response at HF flat. Many mic preamps have a low impedance of 10k or less, which will cause the response of an MM cartridge to droop by as much as a decibel at 5kHz.

Thank you for your response. For this reason I use a "balanced transmitter" like this https://sound-au.com/project176.htm with a 47K input impedance between the TT and the interface. This also allows a short cable run from TT to the balanced transmitter and then a balanced run without any noise or location issues to the MIC interface.

 

[Thomas_A]

My stage (Muffsy) clips above 100 mV at 1 kHz and 43 dB gain. I was thinking of increasing voltage from 15V to 17.5V to the opamps and decreasing gain to 40 dB but I don’t think I have any problems with clipping as it is configured now.

 

[Michael Fidler]

Thank you for your response. For this reason I use a "balanced transmitter" like this https://sound-au.com/project176.htm with a 47K input impedance between the TT and the interface. This also allows a short cable run from TT to the balanced transmitter and then a balanced run without any noise or location issues to the MIC interface.

I'm not sure that circuit is a good idea, as the input impedance is set by a network of other resistors, instead of the first resistor to ground... If you like I can share my own circuit that runs off phantom power.

 

[atmasphere]

In terms of overload behaviour: it's not so much resonances or anything that trails the clipping, more the fact that the RIAA curve features almost 40dB of gain difference from the top to the bottom of the audio band. Therefore, any clicks and pops that push the preamplifier into overload above 10kHz where cartridge resonances and high-velocity surface anomalies exist create intermodulation/distortion artefacts that will be equalised and boosted 20-30dB or so.

While the anomalies on the surface themselves are quite innocuous, as they exist mainly above 10kHz viewing a flat waveform, they are usually assymetrical and provoke strong overload artefacts below 3kHz where we are much more sensitive with our hearing. This is especially true for preamps showing premature overload for higher frequencies (of which there are many tested on this forum) either due to an overly low RIAA feedback network impedance (causing the driver amplifier to go into current-limiting at HF as the impedance drops), or just passive RIAA designs pulled out of op-amp datasheets by the 'design gurus' of any particular audio company who are really just marketing men.

Did you mean to state something else? The RIAA pre-emphasis of a mastering system boosts high frequencies, the de-emphasis curve rolls them off. But usually the input circuit is not able to take advantage of this fact since the EQ usually occurs slightly later in the signal path.
IME the problem is that the input circuit is not equalized and often is not in a feedback loop if one exists. Opamps are the exception and my experience mirrors the advise you stated in a later post.

 

[levimax]

I'm not sure that circuit is a good idea, as the input impedance is set by a network of other resistors, instead of the first resistor to ground... If you like I can share my own circuit that runs off phantom power.

That would be great, you can post it here if you like as I am sure others may be interested or if not you can send me a PM.

Also I am actually using an older version of this project with the schematic below. I changed R102 to 47K. To me this looks like it should set input impedance to 47K but maybe I am not looking at it correctly?

 

[Michael Fidler]

That would be great, you can post it here if you like as I am sure others may be interested or if not you can send me a PM.

Also I am actually using an older version of this project with the schematic below. I changed R102 to 47K. To me this looks like it should set input impedance to 47K but maybe I am not looking at it correctly?

The version you've built is better for your application than the one listed on the website, but does suffer a few noise penalties in the higher input noise current of the 5532 (not a great match for MM cartridges) as well as the noise generated by the second stage's relatively high feedback resistors (ideally should be about 2k). Other than that, it looks OK. You might want to attach a capacitor of about 100pF between pin 3 of U1A and ground to load the cartridge, as well as reduce RFI through the series action of R101 (which isn't particularly useful on its own).

Here's my version of a 48V flat-level phono preamp to microphone circuit. It has a gain of about 6dB, runs off phantom power alone, and can tolerate an input of 1V RMS while driving a 3k external microphone load. Quite a high component count, but it's absolutely bomb-proof as a result. It requires phantom power through both stereo outputs to work correctly. The output isn't fully balanced, but the mic-preamp will be able to reject common-mode noise due to impedance balancing...