Flat Bugle Mod? (ie no RIAA eq-ing)

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RonP

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Flat Bugle Mod? (ie no RIAA eq-ing)
« on: 13 Sep 2011, 05:22 pm »
Hello,

I hope this hasn't been covered before. Google didn't turn up anything.

Would it be feasible to build/have built the Bugle *without* the RIAA circuit
(components)? If so, how to go about this?

My intention is to record flat and use software to do the RIAA eq depending on
my source LP. I want a *clean* flat phono amp that won't break the bank.

Ideas/Comments/Suggestion are very welcome.

Thanks



ps. I'm a pretty sharp guy, but circuits designs are something I know nothing about.


Brinkman

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Re: Flat Bugle Mod? (ie no RIAA eq-ing)
« Reply #1 on: 14 Sep 2011, 02:42 am »
My guess would be to omit R13 & R14 and C2 & C3. Probably could omit R9 & R11 as well; without the shunt EQ the voltage division isn't really required.

Poty could tell you for sure.

poty

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Re: Flat Bugle Mod? (ie no RIAA eq-ing)
« Reply #2 on: 14 Sep 2011, 10:40 am »
I'm sure Brinkman know the answer very well.  :)
In short: It's possible, but with several bad outcomes.
More explanation:
About the circuit itself:
If you don't mind rised distortion and noise you can use only the first stage (one opamp) to achieve your goal. Certainly you'll have to recount R5, R6 (R34, R24) to get enough amplification (1+R6/R5). Then you can leave only R5, R6, R12, R17, U1 and corresponding parts for the right channel. The J1 in the case would be connected to U1 output through R7. To get AC coupling you can save also C1 (C4) between the R7 and the U1 output. Of course we save power supply filtering too (C7, C8, C10, C11).
The better approach - use two stages. First stage (U1) you can left untouched, the second - strip almost all frequency-dependent parts (R10, C3, R8, R13, C2) and connect R7 either to the output of U2 or to the C1 (DC or AC coupling). To maintain adequate load for the first stage and input impedance to the second stage you should short-circuited the R10 and C3 (connect R14 to the output of the U1). R3 and R4 should be calculated to achieve the nessesary amplification.
In my opinion 3 stages is too much for the task.
I hope I haven't overlooked something important. :)
... and I'd prefer to use Piccolo in the case.
About the idea:
I should say at once that I don't like the idea. If it somehow against your thoughts you can stop reading further. :)
I've already mentioned earlier that EQ correction (RIAA or many others) affects the way noise and distortions distributed across the amplification gear. For example, recorded on the vinyl signal pre-equalised according to the RIAA has almost -20dB fall for 20Hz comparing to 1kHz and 25dB rise at 20kHz comparing to 1kHz.
What's happen in classic audio (simplifyed to be short, all is related to the 1000Hz):
A signal (20Hz- x dB; 1000Hz- y dB; 20kHz- z dB)
was recorded on vinyl as (20Hz- (x-20) dB; 1000Hz - y dB; 20kHz - (z+25) dB)
to the input of the phonostage the signal comes with some added noise, say ndB to each strip
from the output [s.-signal, n.-noise] (20Hz- x dB s. + (n+20) dB n.; 1000Hz- y dB s. + n dB n.; 20kHz- z dB s. + (n-25) dB n.)
If we digitize the signal for CD quality (assuming equal power for each strip) all bands will get about 14-bit. The interconnect and internal circuits adds some almost equal noise to each band.
Your idea path:
to the input of the modified pre the signal comes with some added noise, say n dB to each strip
the same layout (as in input) would be from the output
if we digitize the signal, the low-end band should take 20 дБ less in amplitude bits, middle - equal number, high - 25 dB more. There also be equally added noise to each band. So we have less information in low-end and excessive information in high end. The low-end will suffer from more distortions because of lack of digital levels, filters after ADC will have errors on high level of high-freq. - the distortions also. More than that - you'll going to amplify low-end noise of interconnects and ADC input (including problematic 60Hz) together with distorted signal to +20dB! The amplification "in digital" will bring additional distortions to the signal, because it can't add interlevel values correctly.
So... :roll:

RonP

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Re: Flat Bugle Mod? (ie no RIAA eq-ing)
« Reply #3 on: 14 Sep 2011, 09:24 pm »
Let's try this again... Note to self: if the preview is fubar, the post will be too
(even though your text appears down below in the edit box)

About the idea:
I should say at once that I don't like the idea. If it somehow against your thoughts
you can stop reading further. :)

I certainly don't mind hearing from naysayers when they've gone into so much details
as to "why not"! Thanks for the both of the replies!

ok so I see (based on your explanation of the 3 strips @ freq/db amplification) where the
high end might get distorted. 40db typical gain on the bugle + the extra 25db that hasn't
been padded down via the RIAA curve. But I still dont "get" why the low end would have
problems as well.

[more info in general for all]

I've seen the software end discussed here (thank you google search), but not the
hardware end. Thus the thread. I guess the following links would give a bit more
background of the overall process:

The process: http://www.tracertek.com/cms-display/newway.html

their hardware: http://www.tracertek.com/ctp-1000-flat-phono-preamp

another vendors hardware: http://www.channld.com/seta/     (crazy pricing here!)

My thoughts here on the process are: "Hey this makes sense... I should pursue the
feasibility of it." But I wasn't really thrilled by either preamp. One looks too cheap and
the other too expensive. A $100 spend on a Hager kit would be a in a different class
versus spending $100 on a pre-assembled unit available on the market.

So, are we basically concluding here their method may be more technically accurate in
the EQ-ing, but noisier?

your thoughts?

poty

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Re: Flat Bugle Mod? (ie no RIAA eq-ing)
« Reply #4 on: 16 Sep 2011, 03:12 pm »
Sorry in advance for very-very long post!
... I see (based on your explanation of the 3 strips @ freq/db amplification) where the high end might get distorted. 40db typical gain on the bugle + the extra 25db that hasn't been padded down via the RIAA curve. But I still dont "get" why the low end would have problems as well.
It's amazing - I think questions would be about high end, not low. :) The amplification you are talking about is just relative to the total power of the signal. The total power should be chosed in the way the peak of the signal does not exceed max input signal capability of the ADC. In that point of view the distribution of the power amongst bands is not important. So "25dB up in high freq." means  only that the signal would have more high frequencies power and less middle and even more less - low frequencies. Taking into account that:
- the ADC (even the best) has finite analog part's noise; (then you will digitize the high frequencies with excellent signal-to-noise values, middles - worse than usual (part of middles in the total power is less as we agreed before), lows - the worst)
- the ADC (even best) has finite conversion noise; (then after the conversion the low-band conversion noise will rise according to the RIAA correction)
- the ADC (even best) has finite digitizing error; (that explains more distortions in the low-end: the less "power" fit the low band - the more the finite digitizing error affect the signal in the band)
- most ADCs has statistically optimised digital filter as part of the digitizing process; (that explains problems in high end - typically the amount of high frequencies in the signal is much lower then low, the filter is optimised for that situation, in our case it would be almost equal which adds the filter errors due to non-optimum setup)
- you cannot get more information (in digital) then you have after digitizing (second problem for low frequencies - it will use tiny number of bit-depth comparing to "normal" situation; the tiny bit-depth will have small amount of "information" after the digital amplification you'll save "rudeness" of low-informative lows).
My thoughts here on the process are: "Hey this makes sense... I should pursue the feasibility of it."
I've already written a lengthy answer, I'm afraid if I analyse the whole idea it takes several pages to read. First of all the idea is not new. It has been used mainly for restoration of old records in the low-signal end for decades and for high-signal end - more than 10 years in many receivers. Initially, the idea go back to multiband equalisers - at the time - analog. And it has the same problems as the initial idea: more distortions and phase variations.
To be more precise I copy the list of problems of "old approach" they give us:

•Analog EQ's produce Noise because of the op amps used.
•The left and right channels track poorly because of component variations in terms of tolerance.
•Aging has a substantial effect on capacitors, so what you hear today will be different than what you hear a year or two from now from an analog system.
 •Analog circuits pick up some hum because of the physical loop areas which can not be avoided in the circuit layout.
 •Analog circuits display crosstalk due to stray capacitance which can not be eliminated between the channels.
 •Analog circuits are somewhat microphonic picking up low levels of room sound or feedback.
They use the same "op amps" in their RIAA-less devices. Most of the analog stages are the same in the "new" and "old" circuits... including caps. Pure analog circuits doesn't have problems with high digital frequencies which lay nearby the analog circuits, so pick up rather less junk than the combined ones. And you don't have to add ADC-DAC process which is not ideal at all!!!

•Analog EQ's have sloppy frequency and phase calibration because of resistor and capacitor tolerances.
ADC-DAC conversation is more vulnerable to several variations including jitter, variations in driving oscillator... Many studies says the amplitude variations is not so bad comparing to other type (distortions, jitter ...) inconsistencies.

 •Analog components such as resistors, capacitors and transistors exhibit a temperature dependency referred to as tempco (temperature co-efficient). Therefore, the performance of an analog circuit (gain and break point frequencies) will change as a function of ambient temperature.
For such circuits you can use precise parts with low or stable pps. On the other hand increasing gain (because before ADC there is "temperature dependent analog circuits") in the conversion process to digital could lead to clipping easily.

 •Physical capacitors exhibit such anomalies as DA (Dielectric Absorption), ESR (Effective Series Resistance) ESL (Effective Series Inductance), Voltage dependent capacitance or incremental capacitance (dC/dV) (which creates non-linear capacitance vs. signal level), and leakage resistance. All of these parasitics cause them to behave in a less than ideal manner when used in an analog circuit.
Scarecrows! Very-very big! :) If I start to list all problems in ADC-DAC conversion they will occupy many-many pages!

RonP

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Re: Flat Bugle Mod? (ie no RIAA eq-ing)
« Reply #5 on: 16 Sep 2011, 10:27 pm »
Wow. thanks for your efforts! I think that pretty much kills the "flat" idea  :thumb:

So I think I'll just step up to a the piccolo. I'll use the white noise / pink noise test tones to create some correction/calibration eq curves for my system. That should do it.

I''l wait until I'm ready to buy whole system before I buy the phono amp. Maybe the wizards at hagtech.com will have an upgraded version of it by then  :D

poty

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Re: Flat Bugle Mod? (ie no RIAA eq-ing)
« Reply #6 on: 19 Sep 2011, 07:57 am »
I'd like that you understood me right. Every approach has its good and bad sides. I'm not opponent of digital world at all. In many posts I defended the digital devices. But there is no universal all-sides-best approaches and from that paradigm - to each "way of thinking" you should add engineering skills and attention to the small details.

heartm8

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Digital RIAA. I've done it.
« Reply #7 on: 29 Sep 2011, 03:41 am »
Hi,

It all depends on what kind of cartridge, microphone preamp, and impedance settings on the mike preamp you're going to use. Tread very carefully lest you fry your gear.
 The good news, you can buy a mike preamp of quality far surpassing home audio for a fraction of what audiophile outfits charge. You don't need a trumpet or bugle, just GAIN, balanced outs, and the right impedance of your mic pre. You need very very clean gain and the proper mic impedance depending on your cartridge. My first choice was the Steinberg 816 mic pre with 80dB of gain, but it's got input impedance of 3500 ohms, so it wont work. I wish they had variable impedance, and sampled at 192kHz, this would be perfect.  Also it's sampling rate of 96/24 is not the highest, not that we could hear the difference. I find that higher freq and bitrate helps most when applying effects to the sample. So here's what I did.

I bought two used Presonus Eurekas, because they have variable impedance but less gain (52dB) than the Steinberg . I had upgraded Jensen transformers and Linear Tech OP-AMPs factory installed. This is known as the "hot rod" version. One per channel. One 192/24 bit DAC works for both units. All this for less than $1000. I use a Transformer step up (about 30dB) with balanced outs right into the Eurekas +28-30dB, with the gain set about its DAC into my Macintosh. Using half of the Eurekas gain keeps my op amps functioning in a conservative range with lots of headroom. You don't want clipping.

Please don't believe the garbage about OP-AMPs being noisy. Maybe those in the 1970's -1990's. National semiconductor and Linear Devices make stuff for $2 that will blow away most anything today, provided you keep within their guidelines for use. NS engineers are very very smart.

I just bought some NS op amps for a headphone output that are dead flat WAAAY out of the normal range of hearing. For $1.69 each. Shipping was $8.00. LME49720N, with .00005% distortion, almost unmeasurable with test equipment. It could be 1000x more and you still wouldn't even hear it anyway. I have a recording studio with about $70,000 worth of gear. I master stuff through my headphones mostly to get dead flat freq response. If you don't like op amps, I have news for you, almost everything you hear recorded today was mastered with them and heavily processed. The shipping actually cost more then the op amps, and performance is superb.

Back to our situation. First take your cartridge, the Denon DL-103. It needs about 70dB of gain, but Pure Vinyl states that their software takes 10-12 less dB. That's 58-60. So boost the signal about 30dB using a Transformer, and another 30dB in the mic pres. If you're using a MM like Bugle, just subtract the gain from 58-60dB and thats how much gain your mic pre needs. Also, you have to go line to balanced to go into mic pres. For this you need to make sure your mic pre loads the signal right, with the proper impedance. So if Jim made a trumpet without RIAA, and about 80dB of clean gain, that would bethe perfect device for this application.

hagtech

Re: Flat Bugle Mod? (ie no RIAA eq-ing)
« Reply #8 on: 29 Sep 2011, 07:01 am »
I'm with poty on this one.  I am very much against the EQ in software approach.  It's just sooo wrong in my book for many reasons.  But I'm not going to elaborate.  I really don't want to get into it.  Try it if you must, but I am passionate enough about this topic to refuse helping anyone convert a BUGLE over to straight gain.  It is easily done, however.

Sorry to be a jerk about this, but I feel very strongly about certain things.

jh

poty

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Re: Digital RIAA. I've done it.
« Reply #9 on: 29 Sep 2011, 08:49 am »
I bought two used Presonus Eurekas... I use a Transformer step up (about 30dB) with balanced outs right into the Eurekas
The Eureka has a selection of 2.5k, 1.5k, 600, 150, 50 Ohms input impedances. You use 30dB transformer (voltage gain), it gives 1000 times for the impedance ratio. Usual load for MM cartridge is 47k, using the given aspect ratio we should have input impedance in 47M or so. For MC cartridges with typical load values of 100 Ohm, it should be 100k respectively. The Eureka doesn't have either input impedance possibilities.
Please don't believe the garbage about OP-AMPs being noisy... I just bought ... LME49720N, with .00005% distortion...
The numbers looks great, but not for the small-signal applications and not for audio. Let's analyse the mentioned OpAmps for the RIAA-free application.
We will use not IMD you mentioned, but THD+N as far as the param is more relevant to "noisy" nature. We'll use "nominal" value 0.00003% (the max value is 3 times more). According to the recommendations of NS the optimal gain for one stage is 40dB. We need about typically 45dB for the MM cart and 25dB more - for MC, so we divide the needed gain into two stages for 35dB each. One more number for counting - output voltage at about 750mV.
- 750mV - 35dB = 13mV output from the 1-st stage and input for the 2-nd stage. Then we have (looking at page 5 of the datasheet http://www.national.com/ds/LM/LME49720.pdf) 0.008% THD+N for the first stage + 0,0001% for the second = 0,0081%. At the low-frequencies in the future we need to increase the amplitude to 20dB comparing to the middle-frequencies and to (20+25=45)dB to the high frequencies. The "anchor point" for most measurements is 1kHz, so let's determine the amplitude of THD+N comparing to the frequency. So 0,0081% + 20dB = 0,081%. We already don't have HiEnd digits - yeah? And we had not taking into account ADC (not DAC as you incorrectly mentioned) noise and errors in digital processing. For example of errors in digital processing, for low frequencies to be able to have digital gain we should use 20dB less bits (24bit -20dB ~ 20bit). Actually even less, because some low-order digits will be occupied with noise. It's difficult to estimate the number of "music" bits at the end, because it greatly depends on spectral distribution of power across the frequency.

P.S. By the way - Eureka has as far as I know discrete input, not OpAmps.