Thanks. I considered using transformers but the cost for good transformers like the ones Sowter or Lundahl make just for that purpose is more than I want to spend.
I am not trolling anyone. What we have is a failure to communicate caused by a failure to agree on what the definitions of the terms being used even mean. Read this please and I think you will understand why I do not think you are correct.
"RIAA equalization and inverse RIAA equalization are complementary processes used in vinyl record recording and playback to compensate for physical limitations of the medium and achieve a flat frequency response. RIAA equalization is applied during recording, reducing low frequencies and boosting high frequencies. Inverse RIAA equalization, on the other hand, is applied during playback to reverse this process, restoring the original, flat frequency response." This is the correct use of the terms as defined by any electronics handbook.
If RIAA equalization cannot be applied to a signal that is less than say 2V, that is a different issue but I have not seen any mention of that until you just did, If that is correct, then the entire discussion could have ended just by saying that in the beginning.
The owner of PS Audio:
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This will of course ruin your goal of least intervening stuff between the source and the rest of the circuit. Putting a reverse RIAA and a RIAA circuit in between is just the opposite of your stated goal.
It all doesn't make much sense except as a nice obstacle course, but the end result will not sound better than what is the usual setup; quite the opposite.
But hey, you can do what you want!
Jan
I would take that as an insult. His English is pretty perfect, but the content only makes sense to you if you understand the matter at hand.
Jan
If you don't buy the highest quality coupling transformers, I would abandon the idea of using them...
I have an opamp in my audio chain and it is besfect, the best and only one on earth, nothing comes close ...
Place 3 or four in series, it still sings
I have an opamp in my audio chain and it is besfect, the best and only one on earth, nothing comes close ...
Place 3 or four in series, it still sings
One can only hope your need to be right in the face of decades of accepted terminology in an arena you appear to know precious little about will stand you in good stead going forward.
LOL. What I am trying to do is really not much different than what happens when i put an LP on my TT and play it. The difference is that the LP is RIAA equalized it leaves the record plant. I am merely trying to add the RIAA equalization to the signal output by my DAC and play that through the same phono stage as I play LPs through. I have no intention to add a reverse RIAA circuit anywhere in the chain. The only one that would n used is the one that is already in my phono preamp.
I certainly did not intend it as an insult. His comment was edited. Before it was edited, it was difficult for me to understand. He has actually helped me more than almost anyone else on this thread. In any case, there is nothing wrong with someone speaking English as a second or third or more language, In fact, that is admirable.
You need to add the inverse RIAA before you enter the phono preamp to cancel the RIAA correction that is in the phono preamp. That inverse RIAA is the response that is added to the signal in the LP pressing plant. You must do the same to your DAC output to get a flat output from the RIAA preamp. Such a circuit is called a reverse RIAA, because it is the reverse of what the phono preamp does, but you are right it is actually the RIAA processing done at the LP plant.
You want to minimize stuff between your DAC output and the amplifier. But with your plan, you introduce 1) a RIAA phono stage that is a a high gain non-linear response building block that is one of the most non-linear audio blocks around, because of that. To add insult to injury, you must add 2) an inverse RIAA stage to undo the nonlinear frequency and phase characteristic of the phono preamp. It will be hard to find another way to add more stuff to the signal path, exactly the opposite of what you want. The shortest path in your case would be the DAC output, direct voltage or from an I/V, through a volume control to the power amp input.
I can understand your initial idea, it seems logical. But if you look at the actual implementation, it's a very bad idea.
Scratch it, move on and find an idea that actually improves your sound.
Jan
You want to minimize stuff between your DAC output and the amplifier. But with your plan, you introduce 1) a RIAA phono stage that is a a high gain non-linear response building block that is one of the most non-linear audio blocks around, because of that. To add insult to injury, you must add 2) an inverse RIAA stage to undo the nonlinear frequency and phase characteristic of the phono preamp. It will be hard to find another way to add more stuff to the signal path, exactly the opposite of what you want. The shortest path in your case would be the DAC output, direct voltage or from an I/V, through a volume control to the power amp input.
I can understand your initial idea, it seems logical. But if you look at the actual implementation, it's a very bad idea.
Scratch it, move on and find an idea that actually improves your sound.
Jan
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With that quote, it seems, you have misunderstood the whole concept of "RIAA".
There´s no "difference" in "The LP is riaa equalized", and what the reverse riaa would do to the "signal that comes out of your DAC".
When a LP is mastered, the 20Hz signal is lowered by -20dB going up to 0dB at 1KHz, and then increased to reach +20dB at
20KHz. This is called an "inverse/reversed riaa curve", and this is what´s in the groove of a LP.
This enables you to play a LP through riaa phono input with the opposite curve: +20dB
at 20Hz and -20dB at 20KHz (which is what ALL phono preamps do), giving you the result of a "flat" curve from your phono amp.
I have no idea how to explain this any better, so if your statement "I have NO intension to add a reverse riaa circuit anywhere" stands,
your project of using a phono input is NOT doable.
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Given the constraints that the circuit needs to consist of resistors and capacitors only, so no inductors allowed, you could try something like this:
You can use the attached spreadsheet to calculate the values. In the first three rows, you have to fill in R1, R4 and the corner frequency above which the RIAA recording curve should not be approximated anymore. I suggest a value of about 35 kHz for that, because it results in about as much high-frequency loss in the audio band as the third-order Bessel filter with a -3 dB frequency of about 30 kHz that is usually used in Philips CD players. As far as I know, the digital interpolation filter corrects for that. (The suppression of ultrasonic images is of course worse than with the original Bessel filter, but still better than many audiophile DACs.)
Example:
R1 = 10 Ω
R4 = 49.9 Ω
Corner frequency 35 kHz
R3 = 726.3018389 Ω
C2 = 103.26 nF
R2 = 9093.321658 Ω
C1 = 349.71 nF
You have to include the MC input parallel resistance in R4, so please fill in the parallel value of the resistor you want to use and the MC input shunt resistance.
The very same spreadsheet can also be used for RIAA recording curve networks that are driven from a voltage source, you then just fill in R1 = 0.
You can use the attached spreadsheet to calculate the values. In the first three rows, you have to fill in R1, R4 and the corner frequency above which the RIAA recording curve should not be approximated anymore. I suggest a value of about 35 kHz for that, because it results in about as much high-frequency loss in the audio band as the third-order Bessel filter with a -3 dB frequency of about 30 kHz that is usually used in Philips CD players. As far as I know, the digital interpolation filter corrects for that. (The suppression of ultrasonic images is of course worse than with the original Bessel filter, but still better than many audiophile DACs.)
Example:
R1 = 10 Ω
R4 = 49.9 Ω
Corner frequency 35 kHz
R3 = 726.3018389 Ω
C2 = 103.26 nF
R2 = 9093.321658 Ω
C1 = 349.71 nF
You have to include the MC input parallel resistance in R4, so please fill in the parallel value of the resistor you want to use and the MC input shunt resistance.
The very same spreadsheet can also be used for RIAA recording curve networks that are driven from a voltage source, you then just fill in R1 = 0.
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Actually audioxcel just uses a different definition of RIAA and reverse RIAA than most people here. I switched to calling it RIAA recording and playback equalization to solve ambiguities.
I fear it may run a little deeper than that. Whatever the terminology a quick look at the respective response curves would put one on the right track.
I bet it would sound very 'hifi', unbelievable, high contrast and resolution...
Just think one second about it.
You are going to send an unbuffered signal through wires straight into a RIAA inverse which has two opamps...
Then you will have to use a volume control to reduce the signal to lowest as possible not to overload the Phono.
Moving magnet inputs are signals max of around 3mV, and Moving Coils .2mV.
Then it will pass through two more tubes sections with more filters then pre amp, then power amp.
So in total you did this to the signal:
used 4 band pass (which will absolutely screw up the phase, no phono is perfect) filters
Two more opamps
Two more triode sections
You will have raise the noise floor by at least 40db with tube noise, and volume control.
And greatly reduce dynamics.
You will have amplified the signal by 30db then reduced it by 30db , then even more, before it even goes to preamp.
It will sound like some of the best diy gears, a real revelation.
CDs are already compressed from factory to ensure you will not jump and run for the control volume non-stop.
Vinyl are just the master without compression because the signal is naturally compressed, and naturally gets smoothed out in the stylus and all the amplification.
So... you probably in multiplication factors have 10 000x times more noise, and 10 000x less definition (losing bits of data).
Enjoy 3 bits of hifi sound!
Just think one second about it.
You are going to send an unbuffered signal through wires straight into a RIAA inverse which has two opamps...
Then you will have to use a volume control to reduce the signal to lowest as possible not to overload the Phono.
Moving magnet inputs are signals max of around 3mV, and Moving Coils .2mV.
Then it will pass through two more tubes sections with more filters then pre amp, then power amp.
So in total you did this to the signal:
used 4 band pass (which will absolutely screw up the phase, no phono is perfect) filters
Two more opamps
Two more triode sections
You will have raise the noise floor by at least 40db with tube noise, and volume control.
And greatly reduce dynamics.
You will have amplified the signal by 30db then reduced it by 30db , then even more, before it even goes to preamp.
It will sound like some of the best diy gears, a real revelation.
CDs are already compressed from factory to ensure you will not jump and run for the control volume non-stop.
Vinyl are just the master without compression because the signal is naturally compressed, and naturally gets smoothed out in the stylus and all the amplification.
So... you probably in multiplication factors have 10 000x times more noise, and 10 000x less definition (losing bits of data).
Enjoy 3 bits of hifi sound!
https://www.ebay.com/itm/176660264496
something like this to install into the cd player, very easy. You are a pioneer 🙂(
something like this to install into the cd player, very easy. You are a pioneer 🙂(
Where do you gather all that from? You just disconnect the TDA1541 from the transimpedance amplifiers, connect it to circuits as shown in post #72 and connect those to a moving-coil preamplifier. The signal-to-noise ratio and image suppression will get worse than with the original CD player circuitry and you will have to wire things up very carefully not to get hum, but that's all.
Remember that Pre-amp which made your system sound the same?
As soon as you placed it in the chain your system had this certain sound? It was a no feedback tube design 🤣
As soon as you placed it in the chain your system had this certain sound? It was a no feedback tube design 🤣
Thanks. After I went to bed last night, I thought about the definition problems we were having that were hindering progress on what circuit I was asking for help designing. I had decided to propose language that would hopefully get everyone on the same page.
I hope everyone will agree to use your proposed language going forward. Henceforth, please use the terms "RIAA recording" (for the LF -20dB and HF +20dB EQ that is applied to LPs) and "RIAA playback" (for the LF +20dB and HF -20dB EQ that a typical phono preamp stage applies) in this discussion.
That said, what I am asking to help with is a schematic of an RC circuit for adding "RIAA recording" EQ to the signal that is output by a TDA1541A DAC (after that signal has been I/V converted from its typical 4ma PP to whatever voltage would result from a resistor I/V conversion).
Please do not get into the pros and cons of this kind of circuit at this point. I am willing to consider any alternative circuits except those that contain IC op amps at a later point. One of my goals if to definitely eliminate the IC op amp circuits in my CD880 player.
FYI: The CD880 player outputs a non-inverted signal but I think you are correct about how that happens,
TDA1541A DAC https://www.dutchaudioclassics.nl/img/info/tda1541/philips-TDA1541A-datasheet.pdf
NE5532 Op Amp https://www.ti.com/lit/ds/symlink/ne5532a.pdf?ts=1745132145204
Here is a thread that discussed the inverting subject. Interestingly, none of the Lampizator discussions I have ever read mention anything about polarity even though they all eliminate the op amps.
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