Right. SY, highly representative of the average audiophile. NOT.
OK, you've forced an admission from me...
Yes, but you're comparatively sane.
OK, you've forced an admission from me...
Yes, but you're comparatively sane.
Given I have just taken delivery of a miniDSP for X-over duties I could very easily be swayed. But can't require me to fire up a PC to listen to a record.
Not for the majority of my collection. although I do let the telarc 1812 on the turntable 🙂
Yes, there is the hassle of running it all thru a PC. But a miniDSP or similar could just be like any other box.
FWIW, I did post impulse files if someone wants to try them. They work. However, we may be better not using the convolution engine in JRIver or another player. Leave convolution for room correction or other stuff like crossovers. From what I've read, the PEQ settings can be hand edited in JRiver. That would be the ticket. Not much latency, very accurate curve. Just need to figure out where the settings for the PEQ are stored.
FWIW, I did post impulse files if someone wants to try them. They work. However, we may be better not using the convolution engine in JRIver or another player. Leave convolution for room correction or other stuff like crossovers. From what I've read, the PEQ settings can be hand edited in JRiver. That would be the ticket. Not much latency, very accurate curve. Just need to figure out where the settings for the PEQ are stored.
Is someone going to actually try the digital RIAA here, I forget?
A friend and I did a blind test last weekend that was pretty interesting. Blind A/B test with an input selector between straight analog signal from vinyl to amp vs. vinyl (with analog RIAA) into creative soundcard's digital in at 24bit/96k. After level matching, the difference was very obvious.
Through the soundcard's input it sounded like any other digital media except it had a high noise floor. Creative soundcards are crap but it was an interesting test because it instantly exposed the sound of the soundcard's dac.
I will say it was very cool to see the vinyl in some spectrogram software through the computer on a 4k TV. Old meets new... imagine taking that rig back to the 70's and blowing some minds.
A friend and I did a blind test last weekend that was pretty interesting. Blind A/B test with an input selector between straight analog signal from vinyl to amp vs. vinyl (with analog RIAA) into creative soundcard's digital in at 24bit/96k. After level matching, the difference was very obvious.
Through the soundcard's input it sounded like any other digital media except it had a high noise floor. Creative soundcards are crap but it was an interesting test because it instantly exposed the sound of the soundcard's dac.
I will say it was very cool to see the vinyl in some spectrogram software through the computer on a 4k TV. Old meets new... imagine taking that rig back to the 70's and blowing some minds.
Last edited:
The ins and outs of my little M-Audio USB cards aren't perfect, but pretty darn transparent. Most people can't hear it in the signal path.
I'd give the digital RIAA a whirl, if my system were not all packed up in moving boxes. 🙁
I'd give the digital RIAA a whirl, if my system were not all packed up in moving boxes. 🙁
If the noise floor was higher than on the original vinyl, something was definitely not right - at least not the gain structure.
Indeed. We had those capabilities back in the 70's, but only in really big research labs, as it was a major investment. Now we can all have it on our computers in our living rooms or workshops.
Nah, it had a high noise floor compared to digital audio files is what I was saying. I'm pretty sure an 8 core PC showing high resolution waveform and spectrogram on a 4k flatscreen TV would command the respect of any top lab from the 70's. Crappy soundcard and all.
OK, that makes sense.
Absolutely - what we had in the 70's was waveform display and spectrograms, but mostly done in the analog domain. Digital music was really primitive - you would enter the parameters and let the computer crunch overnight, and if you were lucky, you had a (very low-resolution) sound file in the morning. Then the Synclavier and the Fairlight came along...
90 posts on digital RIAA and you guys still haven't mentioned Metric Halo's excellent (hardware & software) implementation?
https://www.mhsecure.com/mhdirect/product.php?productid=76&cat=0&page=1
Using the Metric Halo ULN-8/LIO-8 as a phono preamp | Steve Hoffman Music Forums
https://www.mhsecure.com/mhdirect/product.php?productid=76&cat=0&page=1
Using the Metric Halo ULN-8/LIO-8 as a phono preamp | Steve Hoffman Music Forums
My goal here is to achieve a perfect digital RIAA on the DIY side, of course.
Most of us here have high quality (cheap, more or less) digital stuff (digital to analogue stuff), so it seems to me very interesting to implement a cheap high quality digital RIAA.
The FIR way is the one i like.
Even if I still don't understand how to do it, there are here several good suggestions to do it, so now it's time to try them all.
So, I have a basic question: If you're using miniDSP, or the Hypex DLCP, to do crossover/EQ, can you incorporate an RIAA transform in there as well, without additional hardware?
This seems to be a unique mod done to the box for MM cart inout. Is it just this one guy, or are other people using the Metric Halo box for RIAA?
Did some searching found jiitteepee's new site, RIAA at the bottom you will find the IIR coefficients in the form that most programs take. miniDSP inverts the sign on two of these (see their guide). If you have a 96K miniDSP you can use those and sample at 24/96.
Gary Galo published an article somewhere pointing out that you do NOT want to use an FIR filter for RIAA. There was a response to this from a developer at
Audacity that was hilarious. Analog filters are IIR and an IIR filter can approximate both amplitude and phase to a high degree. The fact that just about everything uses floating point math eliminates early problems with DSP round off and overflow
In any case this whole issue seems still debated hotly. Channel D was at RMAF with their whole setup getting rave reviews (even from Fremer). While several others have tried "everything" and still say it's crap compared to analog with the usual instantly obvious comments thrown in.
Gary Galo published an article somewhere pointing out that you do NOT want to use an FIR filter for RIAA. There was a response to this from a developer at
Audacity that was hilarious. Analog filters are IIR and an IIR filter can approximate both amplitude and phase to a high degree. The fact that just about everything uses floating point math eliminates early problems with DSP round off and overflow
In any case this whole issue seems still debated hotly. Channel D was at RMAF with their whole setup getting rave reviews (even from Fremer). While several others have tried "everything" and still say it's crap compared to analog with the usual instantly obvious comments thrown in.
Can you say more, fine sir?
'Cause I was just about to 'upgrade' from my miniDSP (ADAU1701):
miniDSP (ADAU1701)
http://www.analog.com/static/imported-files/data_sheets/ADAU1701.pdf
to my miniSHARC (ADSP21369)
miniSHARC (ADSP21369)
http://www.analog.com/static/imported-files/data_sheets/ADSP-21367_21368_21369.pdf
miniDSP (ADAU1701)
vs miniSHARC (ADSP21369)
... and go alll ... FIRy 😉
"Finite impulse response (FIR) filters
An FIR filter requires more computation time on the DSP and more memory. The DSP chip therefore needs to be more powerful. miniDSP products that support FIR filtering include the OpenDRC and the miniSHARC kit.
FIR filters are specified using a large array of numbers. In the case of the OpenDRC, there are 6144 coefficients (or "taps") per channel. In the case of the miniSHARC, there are a total of 10240 taps assignable to all input and output channels. Generation of this large array of numbers must be done in a separate program, such as rephase, Acourate, and others.
FIR filtering has these advantages over IIR filtering:
- It can implement linear-phase filtering. This means that the filter has no phase shift across the frequency band. Alternately, the phase can be corrected independently of the amplitude. See examples below.
- It can be used to correct frequency-response errors in a loudspeaker to a finer degree of precision than using IIRs.
However, FIRs can be limited in resolution at low frequencies, and the success of applying FIR filters depends greatly on the program that is used to generate the filter coefficients. Usage is generally more complicated and time-consuming than IIR filters.
from miniDSP's :FIR vs IIR filtering
Thanks and cheers,
Jeff
Last edited:
miniDSP
DSP 28/56 bit DSP Engine / Double precision processing
Host Processor 48MHz microcontroller
Sample rate 48kHz
ADC/DAC Data resolution 24 bit
miniSHARC
Digital Signal Processor 32bit Floating point Analog / 333M
MCU On board MCU for ind conf, CNT
Sample rate 48kHz
SDRAM 128Mb SDRAM
Whew, I h8t c/p from dtashit!
Thanks and Cheers,
Jeff
DSP 28/56 bit DSP Engine / Double precision processing
Host Processor 48MHz microcontroller
Sample rate 48kHz
ADC/DAC Data resolution 24 bit
miniSHARC
Digital Signal Processor 32bit Floating point Analog / 333M
MCU On board MCU for ind conf, CNT
Sample rate 48kHz
SDRAM 128Mb SDRAM
Whew, I h8t c/p from dtashit!
Thanks and Cheers,
Jeff
Last edited: