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Hagerman Bugle3 Phono Stage Review

Rate this phono stage:

  • 1. Poor (headless panther)

    Votes: 3 2.6%
  • 2. Not terrible (postman panther)

    Votes: 39 33.9%
  • 3. Fine (happy panther)

    Votes: 67 58.3%
  • 4. Great (golfing panther)

    Votes: 6 5.2%

  • Total voters
    115
This is substantially better than the older sample as you can see in our ranking table:
View attachment 273007

Thanks for the review Amir. Your ranking table isn't showing the Cambridge Audio Duo MM/MC phono pre-amp you reviewed back in February 2019. Looks like that one got lost somewhere along the way. Quite understandable considering the huge amount of reviews you publish. :)
 
Thanks for the review Amir. Your ranking table isn't showing the Cambridge Audio Duo MM/MC phono pre-amp you reviewed back in February 2019. Looks like that one got lost somewhere along the way. Quite understandable considering the huge amount of reviews you publish. :)
I took it out on purpose because its test conditions are different than what I use now.
 
I would have wanted to see higher tolerance of input spikes as to lower the chances of clicks and pops clipping:

This is due to a single rail DC power supply and a rail splitter, resulting in a bit less than half the power supply voltage (assuming it's ~12V). So +/-6V which tallies quite well with the 60mV you measured. 40dB gain (100 times) the circa 60mV =6000mV or 6V. The overload performance at 60mV is absolutely atrocious and equally as bad as the worst I have ever measured. It means that on the 64dB gain setting for MC, it will overload at ~3.8mV. For a 0.3mV LOMC, that is only a 22dB overload margin.

Usually this is highly frequency dependent but is not in this case:

That is a clue the RIAA network is passive and not in the feedback loop like most implementations. A glance at the schematic confirms that.
 
This is due to a single rail DC power supply and a rail splitter, resulting in a bit less than half the power supply voltage (assuming it's ~12V).
The switching supply is 24 volts.
 
I do vinyl for nostalgia.
In my past home for years I played vinyl only with a vacuum tube riaa pre-amp and a vacuum tube SET power amplifier. That is as good as it gets.
In the big picture vinyl play back is limited by noise; noise from every step along the path.
In recent years I have moved on to much cleaner electronics including op-amp riaa pre amplifiers. I use 2 12 volt gel cell batteries for a power supply.
Still for nostalgia nothing beats vinyl.

Thanks DT
 
You guys seem to be very knowledgeable about phono preamps. Do you know of any diy freely available design for MM known to measure fine and have good headroom and all the specs you like to see in a phono stage? Or is there anything against diying one? (ie. too tight tolerances required or stuff like that. No clue, just wondering)
Thanks!
I'd forgotten this interesting thread: https://www.audiosciencereview.com/forum/index.php?threads/openamp-mm-phono-preamplifier.38059/
 
as an 80ies kid I'd rather have a tape deck lol
From a gadget POV so would I, I still have my Revox, but the pre-recorded tapes of the era were rapid duplicates and not as good as LPs for SQ.
I Only used mine to make recordings myself but the DAT recorder I bought was markedly better still, so it has been for replay of old tapes only for about 30 years!

It still looks better than any other source though.
 
From a gadget POV so would I, I still have my Revox, but the pre-recorded tapes of the era were rapid duplicates and not as good as LPs for SQ.
I Only used mine to make recordings myself but the DAT recorder I bought was markedly better still, so it has been for replay of old tapes only for about 30 years!

It still looks better than any other source though.

I used to make "mix tapes" recording radio. with radio allready beeing badly processed you can imagine to quality lol. It was also a copy of another tape that I would let record the radio show haha
I remember having to manualy fade the endings because the radio man would start to talk.
 
Now I have looked at the schematic, the reasons for low overload are clear and my above 'calculations' were quite wrong as I had no idea of the supply voltage and I made some bad :facepalm: quick calculations. I can see why Hagerman have done what they have done, but it doesn't excuse the overall performance (or lack thereof).

It's an interesting phono stage. Close, but no cigar.
 
I have one of these I bought pre-assembled a few years back. It is audibly noisy. I guess the solution is to have one custom made by the designer? Wonder what would happen to the $200 price tag for a custom build?
 
@amirm, thanks for covering the Bugle 3! Much better performance than the kit Bugle 2 I had sent in.

In all honesty, I think the Bugle 2 review should be taken down since, as I had noted in the Bugle 2's review thread, when the unit was returned I noticed a number of soldering defects which likely affected performance so it wasn't a fair assessment to the actual performance of the unit. It was, however, a great assessment of my piss-poor soldering skills back when I had built it.
 
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Thank you for the re-test! The Bugle2 and Bugle3 are virtually the same circuit and layout, with a few DIP switches added to the latter for adjustable gain and loading. They should measure exactly the same! Please take down the invalid results for the Bugle2, as that unit was clearly faulty (and not built by factory).

Regarding a few other questions...

Input capacitance is very low at 26pF, so does not affect MM loading greatly. I always preach minimal capacitance, no matter how determined a customer is to add more. The 10Hz low frequency rolloff is absolutely intentional, as I do not believe amplifying such sub-audible noise is beneficial. The 60Hz blip in the frequency plot does not come from the Bugle3 itself, but rather from some external source. Being housed in plastic does make the unit a bit more susceptible to such external influences, and proper placement and orientation within the audio system can eliminate that minor blip.

Overload is another matter. If all one cares about is noise floor or extra high headroom, they are both clearly solved via application of negative feedback. That is however, a huge tradeoff! I chose a split-passive EQ filtering method instead, because it results in a much more natural sonic presentation. It also rides through pops and tics much better, as such transients often cause EQ feedback circuits to go open loop, causing a much greater impact, with the circuit desperately slewing back to correction. Pick your poison. There is a lot more to a good-sounding phonostage than mere measurements.

The rise in EQ response is due to an added 3.18us zero. Intentional. Explained in my paper here:

 
Hey Jim, thanks for joining in on the discussion and explaining your design logic. I am an avid DIYer and encourage others to explore your designs and learn how to wield a soldering iron :)

While I haven’t built any of your designs, I commend you on your support for the DIY community. It’s the DIY spirit that makes this a great hobby.

I see a Bugle in my future…
 
As I mentioned, the Bugle2 and Bugle3 are essentially the same thing...

B2-B3.jpg
 
I chose a split-passive EQ filtering method instead, because it results in a much more natural sonic presentation.

Are you sure about this. Isn’t it rather one of the audiophile myths that is supported by popular magazines and popular audio forums. IME, if listeners have to make a choice between blackboxes (NFB EQ x passive EQ), such assumption does not work, provided the units have competent design.
 
Are you sure about this.

Actually, yes I am.

I do not know if this has been popularized or not by forums and magazines, only that I performed experimentations myself. Keep in mind my original Bugle is over 20 years old, so likely precedes many of these "myths" you may be reading about. Also, I've been getting tons of customer feedback over the years, comparing my equipment to many others (which I cannot afford to do by myself), letting me know where I stand amongst competition. I catch a lot of clues that way.

There are also technical reasons. As I tried to point out earlier, having a lot a frequency dependent feedback can push an opamp into desperate territory. Not during steady state conditions (which is where most of the above measurements exist), but transients. In my opinion time-domain response is more important sonically than steady-state frequency response. The best way to do overload measurements is with pulse / chirp style short duration (music-like) transients. This is also the domain of pops and tics. If your global negative feedback circuit sees too fast or too great of an input signal, it will go open loop, meaning all true music information is lost during slewing and recovery of loop. It's quite annoying.

Another good example is an amplifier using NFB powered by a regulated supply that also uses NFB. Here you have an amplifier tugging on a supply at the same time the supply is trying to regulate itself. Basically, you have two NFB networks with different timing characteristics working against each other. You can't imagine how common this is! The result is an overly bright, accentuated, somewhat "technicolor" presentation. It sounds fantastic for a minute or two, and in a short comparison, the listener will often select it as better. However, long term listening brings truth to light. And that is where my passive split-EQ comes in. It does not have excess sparkle, but instead stays true to nature (and the sound of a real voice or instrument). The best way to measure this is over time. How long can you listen to something without fatigue? How long can you crank something up without feeling discomfort or pain? If the circuit is behaving well, you will never have to turn the volume down.

It is one of my main listening tests (besides cranking just the noise floor through headphones). How long can you crank the shit out an LP without feeling like you need to turn it down?
 
I am a DIY EE and was, in younger days, a professional electronics repair technician.
I built my own phono stages and with experimentation I found out that is better to have them as close as possible to the cartridge. Inside the player was the best location.
Connection with the coil was done with shielded twisted pair cable, one cable per each channel. Ground only one side of the shield.
RCA cabling is suboptimal IMO for this high impedance coil load.
I also believe that capacitance loading should be minimal or as prescribed by manufacturer.

Also I totally agree that steady state measurements show only part of the story.
They are foundation of audio quality, but not the final answer.
I had people here attacking that oppinion, I think it's result of not understanding the underlining of audio measurements and a bit of fanboy mentality about those measurements.
 
Actually, yes I am.

I do not know if this has been popularized or not by forums and magazines, only that I performed experimentations myself. Keep in mind my original Bugle is over 20 years old, so likely precedes many of these "myths" you may be reading about. Also, I've been getting tons of customer feedback over the years, comparing my equipment to many others (which I cannot afford to do by myself), letting me know where I stand amongst competition. I catch a lot of clues that way.

There are also technical reasons. As I tried to point out earlier, having a lot a frequency dependent feedback can push an opamp into desperate territory. Not during steady state conditions (which is where most of the above measurements exist), but transients. In my opinion time-domain response is more important sonically than steady-state frequency response. The best way to do overload measurements is with pulse / chirp style short duration (music-like) transients. This is also the domain of pops and tics. If your global negative feedback circuit sees too fast or too great of an input signal, it will go open loop, meaning all true music information is lost during slewing and recovery of loop. It's quite annoying.

Another good example is an amplifier using NFB powered by a regulated supply that also uses NFB. Here you have an amplifier tugging on a supply at the same time the supply is trying to regulate itself. Basically, you have two NFB networks with different timing characteristics working against each other. You can't imagine how common this is! The result is an overly bright, accentuated, somewhat "technicolor" presentation. It sounds fantastic for a minute or two, and in a short comparison, the listener will often select it as better. However, long term listening brings truth to light. And that is where my passive split-EQ comes in. It does not have excess sparkle, but instead stays true to nature (and the sound of a real voice or instrument). The best way to measure this is over time. How long can you listen to something without fatigue? How long can you crank something up without feeling discomfort or pain? If the circuit is behaving well, you will never have to turn the volume down.

It is one of my main listening tests (besides cranking just the noise floor through headphones). How long can you crank the shit out an LP without feeling like you need to turn it down?

And I am saying that there does not exist an advantage of passive EQ vs. NFB EQ. Similarly as you, I have had 50 years+ experience with audio circuits, EE degree, and hundreds of electronic boards designed. So we have statement against statement. So who would arbitrate? Users? You will find plenty in both camps. DBT? Do you do them, or do you consider them "flawed"? Do you need to sell and support the design approach? I do not.

It is trivial to design the feedback system to have the circuit free of overload recovery issues. It is trivial to test with transient/impulse signals. All those thoughts about "steady state testing" issues are just fairy tales to bring something interesting to the laymen audience. They will eat almost everything.
 
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