Well as balanced as a phono cart can be! but yes I am peeling back the ground table on the case so the coils are floating.
Phono carts could (and really should) be perfectly balanced, except that the hifi industry hasn't chosen to go that way (I guess originally for historical cost reasons and then audiophile luddite-ism).
I'm on mobile, so it's not so easy to elaborate the answers.
Anyway, my point is:
of course I know there's no human earing above 20KHz ( and the actual limit is even lower).
I would like to emphasize the possibility that the signal - if any - above the 20KHz could be responsible of an in-band - read: <20KHz - reflection, per the theory of modulation.
Do you know how the radio works?
This is possible, and there were many threads on this topic.
I will try to re-discovery those threads.
When you say it can not affect the music content because there is no human earing above 20KHz ii's like you won't understand what my point is.
Anyway, i'm sorry if it seems that i have got the truth, it is not at all.
Really don't understand what do you mean with confirmation bias.
Peace 🙂
Yes, I am familiar with intermodulation, and also how the radio works (even if not all radios rely in intermodulation).
Anyway, intermodulation effects are distortion, and as they are not in harmonic relation to the actual music, are much more harmful than harmonic distortion, so yes, intermodulation products can be audible, but it is not a desirable thing to have.
Agree! 🙂
Philips did do some 5 wire MM cartridges in the past, but that didn't last. There is a limit to the benefits to gain, esp with MC as the coil will act as an antenna and that pickup cannot be cancelled out. Sorry pulling this a little OT here.
Well it's not entirely OT as ADC are differential or balanced in. I have my cart wired that way, but adapted to unbalanced at the end at the moment. If I ever do build a gain stage for it I would like to be balanced in and out to the ADC.
I'm using 4 pin XLR. I should have used 5 pin with a pin dedicated to the shield.
I'm using 4 pin XLR. I should have used 5 pin with a pin dedicated to the shield.
This is an opinion, not a fact.
What I want to say is: here we are all the audio-ish to find the best way to listen to music, infinite disputes on the vinyl is better than digital etc...
Maybe the fact that some records can content music above 20KHz and, per destructive, constructive, intermodulation, analog noise shaping or whatever, could be the reason why subjectively the vinyl sound 'better'?
If it is possible, why to cut tout court the frequency response at 20KHz, when today is possible to extend the band up to, say, 50KHz without trouble?
Even if 1 vinyl record up to 10 shows content above 20KHz I think it's an argument to be debated if we plan to listen to vinyl through a dsp.
I know wikipedia is not the way, but in this case it evidences there are some cases vinyls have got frequencies above 20KHz:
Comparison of analog and digital recording - Wikipedia
That disharmonic intermodulation products are dissonant, and not part of the original signal?
If the discussion is about what distortion or coloration you happen to prefer, sure. I, in the other hand, am interested in the original aim of "HiFi" - high fidelity. As in accuracy, transparency and fidelity to the original recording. As verified objectively.
Sure, but if it is coloration you prefer, why not use something like this for your DSP:
Vinyl Overview | The ultimate lo-fi weapon
Nothing wrong with Wikipedia, as long as references are checked. Anyway, from that page: " The high frequency response of vinyl depends on the cartridge. CD4 records contained frequencies up to 50 kHz." - exactly what I wrote earlier. Most cartridges don't go much above 20 kHz, and 4-channel matrix stuff that need a higher frequency response require special gear and circumstances.
Why does this really matter? Why does one size have to fit all in DIY? I can not hear anything above 14 Khz anyway, not even subliminaly, I've tested it. Can't hear it sighted, blind or ABX. So I would chose not not bother with it.
If you can hear or sense those frequencies and it adds benefit to you, then it's easy enough to retain them with high sampling rates.
If you can hear or sense those frequencies and it adds benefit to you, then it's easy enough to retain them with high sampling rates.
They are part of the original signal, in that we want to listen to vinyl, not to master.
Again, we want to listen to vinyl.
It's a matter of media, we want to ear the 'analog' sound.
Otherwise, hdtracks is an excellent choice.
I'm not talking about coloration of sound, still it seems you want not to understand what I mean.
But it's ok, it's a free world (?).
This is not the same to say 'no vinyl go above 20KHz'.
It actual says 'some vynils go above 20KHz'.
But
I think it's an endless topic, so I will stop here and now with this argument.
Better to find a good 32bit adc to i2s board, maybe with a moddable input stage - yes, I think a triode would be beautiful there.🙄
Who are "we"?
Sure, some people want to listen to the equipment, not the music - for that, the DSP plugin I suggested is perfect - it allows you to make the sound even more "vinyl".
What do you think "analog sound" means, in your opinion?
I think it is you who don't understand that any deviation from the original signal is coloration. Sure, some people might prefer it, but it is still coloration.
Some 'vynils' might, but I would love to see you present some evidence (as long as we aren't talking about special surround sound recordings).
Yes, a triode is a good way to add coloration and distortion, but no existing ADC gets even close to 24 bits - not that that matters, as vinyl is limited to an equivalent of 11 bits or so.
Thank you iperv! I guess I'll try with something around 36dB. For my first trials I'd rather leave a few bits of dynamic range unused than risk clipping on the input.
Well it's reassuring to know that I'm not the only one who has this habit.
A balanced flat phono board is something I have been looking for for a while now. Never found one, that's why I'm now hacking up this NE5534A buffer to give it the gain I need.
It's somewhat similar with me: Not looking to change the world, I'm just poking around to satisfy my own curiosity.
I have this tower of hardware that I use for vinyl playback. It isn't pretty and it takes up quite some space.
Right now I use
- a phono preamp
- a Denon PCC 1000 stereo crosstalk canceller
- a Marantz SX-72 scratch noise suppressor
- a dbx NX 40 expander for my dbx encoded discs
- a parametric EQ
- and a subsonic filter inspired by Doug Self's devinylizer
My dream is to recreate all of these on one small DSP board so that I put the big ugly boxes in storage. I have implemented a first draft of all of these components in SigmaStudio and have confirmed that I could run them at once on the cheap ADAU1701 as far as hardware performance is concerned.
In the coming weeks I want to be able to get the signal from my cart into my DSP in a usable form. After that I'll implement basic RIAA filtering and then I hope to get to try the more zany ideas I have like crosstalk cancelling and declicking the audio at runtime. Who knows, I might get something going in the next five years or so 😉
I can list the guilty parties who keep tempting me with ideas so you know to avoid them!
I may have a spare. Couple of weeks and I will know for sure. I built two and now the person I built one for doesn't want it.
[/QUOTE]
I'd love a big tower of stuff but space and kids do not allow it, so I am forced to reduce boxes. I do have a focusrite scarlett to allow me to go via a PC first for additional processing and experimentation and general heresy points.
Right now I'm a little confused on low frequency roll off though. Although as I have stated on here I think LF grunge is best dealt with at source if I can bung a steep filter in before RIAA makes sense for belt and braces. BUT I think you have to play some tricks to get numerical accuracy below 50Hz with miniDSP biquads so need to check that.
🙂 I, after a lifetime of gas-torch use, can't hear beyond 4 kHz reliably...however I can consistently pick a great recording from a mediocre or poor one to the chagrin of my younger colleagues.
Quantity is no substitute for quality..😉
An early (18-30yrs of age ) occupation in acoustic and electronics evaluation has been my ongoing joy (as I lose response and gain 'crickets') to have the skills to evaluate both sound and reproduction equipment.
Too often I see posts where the poster claims that as they can't hear past 'x'kHz it won't matter what equipment /sample rate/ they choose to employ "because they won't hear it" .... (my emphasis 🙂...)
People....back yourself..believe in yourself.... you are only unable to judge good from bad once you are below ground...😛
I got the preamp/buffer going today. At first I wanted to use a chinese NE5534 buffer board and modify that to give me some gain but I have abandoned this idea. The op-amps looked like they had been sanded off and relabeled with something else. They were probably fake. I lost my confidence in the whole design at that point.
With the VSPS for some guidance I made my own circuit - my first one ever. I actually got a board with dual genuine TI NE5534A going 😀 Gain is 101, with Rin at 220 Ohm and Rf at 22K. Similar to the VSPS I used 2.2uF output coupling capacitors. Next time I'll try going the DC bias route but I need to keep it simple right now as I'm already way out of my comfort zone.
I powered it all up with a cheap XL6019E1 regulator board set to provide +-15V. I listened to the output with an active speaker and I got what I expected: Line level output without RIAA EQ. What does worry me though is a loud pop when the mute relay of my turntable activates and deactivates.
I managed to upload the 48kHz biquad from the appendix from scott's article in Linear Audio 10 to the Sure DSP. Only then did I find out that the breakout cable for the audio I/O is not included with the board. I have ordered some cables and will have to wait until they have arrived. In the mean time I have some extra time to find out why the loud bang occurs when my turntable mutes the output...
Yes I vaguely remember reading something about inaccuracy with the miniDSP below 20Hz. What I do not remember is whether this was a problem with the specific miniDSP software implementation or the problem was with the underlying ADAU DSP chip. If the former is true then I'll be fine when using SigmaStudio. If the latter is true I'll need to do a lot more measurements to get the subsonic filtering right. IIRC the problem was not that filters below 20Hz would not work - it was that filters below 20Hz did not behave in the way that the software predicted, right?
With the VSPS for some guidance I made my own circuit - my first one ever. I actually got a board with dual genuine TI NE5534A going 😀 Gain is 101, with Rin at 220 Ohm and Rf at 22K. Similar to the VSPS I used 2.2uF output coupling capacitors. Next time I'll try going the DC bias route but I need to keep it simple right now as I'm already way out of my comfort zone.
I powered it all up with a cheap XL6019E1 regulator board set to provide +-15V. I listened to the output with an active speaker and I got what I expected: Line level output without RIAA EQ. What does worry me though is a loud pop when the mute relay of my turntable activates and deactivates.
I managed to upload the 48kHz biquad from the appendix from scott's article in Linear Audio 10 to the Sure DSP. Only then did I find out that the breakout cable for the audio I/O is not included with the board. I have ordered some cables and will have to wait until they have arrived. In the mean time I have some extra time to find out why the loud bang occurs when my turntable mutes the output...
Yes I vaguely remember reading something about inaccuracy with the miniDSP below 20Hz. What I do not remember is whether this was a problem with the specific miniDSP software implementation or the problem was with the underlying ADAU DSP chip. If the former is true then I'll be fine when using SigmaStudio. If the latter is true I'll need to do a lot more measurements to get the subsonic filtering right. IIRC the problem was not that filters below 20Hz would not work - it was that filters below 20Hz did not behave in the way that the software predicted, right?
I'm going to get a frequency sweep vinyl like this to adjust my digital RIAA for my cartridge.
Cardas - Frequency Sweep & Burn In LP version 2 | Shop Music Direct
Are they trustful? Or any better choice?
Cardas - Frequency Sweep & Burn In LP version 2 | Shop Music Direct
Are they trustful? Or any better choice?