That's what I do and connect directly to the Okto AES input. Currently it then sends it to computer for XO and EQ via USB and then back to the Okto but once I settle on a particular filter set I'll wrap my brain around the new firmware and it could, theoretically, remove the computer from the equation.
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- Thread starter dualazmak
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That's what I do and connect directly to the Okto AES input. Currently it then sends it to computer for XO and EQ via USB and then back to the Okto but once I settle on a particular filter set I'll wrap my brain around the new firmware and it could, theoretically, remove the computer from the equation.
OP
- Thread Starter
- #182
I can understand the two diagrams (especially the upper one) you shared above look almost the same idea and concept like in my DSP-based multichannel setup shared in my very first post on this thread.
In my case, I use "four HiFi stereo integrated amplifiers" (in way of right-person-in-right-place) instead of "ART 16 channel BAL. Patchbay + Apollon NC8350MP".
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Multicore
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How do you switch between analog and digital? Unplug 16 XLRs and plug in the others?
Nice choice of phono preamp.
Burning Sounds
Addicted to Fun and Learning
Yes, I do prefer the digital x-over/eq done in JRiver to the ASP. The difference is not huge, but it seems more realistic and dynamic to me. The other advantage is, as you say, easy to add any other filters needed because of the room - just one in my case. It's also very versatile in that you can easily isolate each channel for testing. I find using TBPro Audio's DPmeter5 (VST) useful for checking that there is no clipping going on. It's only 6 channels, so doesn't include the tweeter, but the tweeter will never clip anyway.
The analogue setup is already coloured by the time it gets to the ASP so not having any additional correction is less bothersome to me. Playing LPs (vinyl to young people) is more of a nostalgia thing these days, but nevertheless, very enjoyable.
I've just recently updated my Okto to FW 1.6, so haven't explored the new DSP capabilities yet. I'm not sure it will be able to do what JRiver can do, but I'm open to that possibility.
Burning Sounds
Addicted to Fun and Learning
Is there any noticeable latency with a large number of filters? In the digital playback world it's not important, but going from analogue to digital I've experienced putting the tonearm down and then nothing coming out of the speakers for a couple of seconds. Same when the record ends, the tonearm is in the run-out grooves and the music is still playing. Having an analogue system from start to finish avoids this.
Burning Sounds
Addicted to Fun and Learning
No, only need to move 8 xlrs to the empty sockets - the analogue outputs from the Okto are on the bottom left and the outputs from the ASP on the bottom right. Outputs to the poweramp are on the top. The patchbay just sits vertically at the back of the rack and everything is colour coded - takes about 15 seconds - need to turn the power amp off first of course. Get's me out of the chair.
Probably 90% of my listening is digital so moving to analogue doesn't occur that often anyway.
Multicore
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Not many systems here are optimized for ease of use by the whole family. ;}No, only need to move 8 xlrs to the empty sockets - the analogue outputs from the Okto are on the bottom left and the outputs from the ASP on the bottom right. Outputs to the poweramp are on the top. The patchbay just sits vertically at the back of the rack and everything is colour coded - takes about 15 seconds - need to turn the power amp off first of course. Get's me out of the chair.
View attachment 389459
There is potential latency if you use FIR filters. I've never used any that would cause that much delay but I keep my corrections under 400Hz and don't need huge taps. If you check out my earlier post in this thread you'll see 2 separate graphs I use for correction utilizing Hang Loose Convolver Host, the 2nd (more complicated) graph shows the same corrections done in both FIR and PEQ capable plugins, the PEQ version has near zero latency.
OP
- Thread Starter
- #189
Really? I am just wondering why you have so much latency/delay in vinyl TT listening in your setup.
Just for your reference, in my revived vinyl TT setup in my DSP-based multichannel system, JRiver hears TT vinyl in "Open Live" mode through audio-interface TASCAM US-1x2HR and phono-preamp AUDIO-TECHNICA AT-PEQ30, and send the signal into system-wide DSP center EKIO.
In this signal path, I have essentially no such latency/delay in "live play" of TT vinyl sound, as I replied to @MCH in my post #692 on my project thread.
@MAB also joined saying "I have a phono preamp plugged into the ADC of my DSP crossover. I have no delay."; please refer to our communication #693 and #694.
As shared in my post #931, the latest signal path in live TT vinyl listening is shown in the two diagrams;
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Keith_W
Master Contributor
To be more specific, linear phase FIR filters. Minimum phase FIR filters don't have this latency. This is because the impulse is located in the middle of the filter length:
Green = LR4 minphase FIR, red = LR4 linphase FIR.
The latency of the linphase can be calculated with (n-1)/2Fs where n = number of taps and Fs = sampling rate. So for a 48kHz sampling rate and 65536 taps, the latency will be 0.682s. This latency is unavoidable and will be present no matter how powerful your CPU is. The only way to reduce the latency is to use fewer taps or increase the sampling rate.
There are of course other causes for latency, e.g. additional processing with VST's, interface latency, streaming latency, and so on. The latency on my RME is about 20ms, and the latency on my Focusrite is a whopping 100ms. You can measure interface latency by using a loopback cable.
I have no idea how long it takes for my streamer to fetch data from Tidal's servers. It will be a minimum of 10ms (because that's my ping) but the actual latency, I don't know. I am sure there is a way to measure it, but I haven't spent any brain cells attempting to do so yet
EJ3
Master Contributor
Not to mention mine may change hourly, daily or weekly, depending on what gear I want to listen to.
Or it may sit unassembled for long periods, while my interests are elsewhere.
OP
- Thread Starter
- #192
This post is just "REMINDER for SURE" for beginner people in audio hobby world who are not so familiar with "latency" and "relative-group-delay/time-alignment".
I believe almost all the audio-expert people frequently onboard on this thread would well understand the difference between "absolute latency" and "relative-delay/time-alignment".
I have several times noticed, however, that some of beginner people in audio would have confusion, misunderstanding, mix-up, of "(absolute) latency" and "relative-delay/time-alignment".
This is why I wrote, rather redundantly, at the top of my pot #493 on my project thread as follows;
In the digital signal processing, we have so many buffers or latencies; JRiver output buffer, ASIO4ALL's I/O buffers, (VB-AUDIO MATRIX's buffers), EKIO's processing buffer, DIYINHK USB ASIO driver's buffer (for OKTO DAC8PRO), and so on. Consequently, it is not straightforward to exactly measure the "absolute delay" between the JRiver's "shout" and the final air sound kick-up by SP.
I usually set all the buffers in the digital domain in rather large size, so that I should not have any latency or (absolute) delay problems; in our audio setup, we have no problem at all if all the bunch of the digital and analog signal (15Hz - 30 kHz) have identical (common) amount of (absolute) delay time (latency) from the signal origin at JRiver, and this is always the case in our digital (PC based) audio system.
The relative delay between the SP units (SP drivers), or "time alignment" in multiple SPs (SP drivers), however, is always one of the critical issues in audio system, especially the multichannel multi-SP-driver multi-amplifier system, as you may agree.
I have been always take my attention and care on this issue, and in my very early posts #18 through #21, by using REW's wavelet analysis, I briefly checked that all of my SP units (SP drivers), i.e. super-tweeter (ST), Be-tweeter (TW), Be-squawker (SQ) and woofer (WO) have essentially no delay with each other, while (however/on-the-other-hand) my sub-woofer (SW) has 10 - 20 ms delay against the other SP units (SP drivers).
Now, I became really would like to establish my own simple, reliable and reproducible precision method for "time alignment" or "relative delay" measurement, and fine adjustment(s) if needed.
By the way, if you would be interested, please visit my post here #931 on my project thread for the details of the latest (as of June 26, 2024) system setup of my multichannel audio rig.
I believe almost all the audio-expert people frequently onboard on this thread would well understand the difference between "absolute latency" and "relative-delay/time-alignment".
I have several times noticed, however, that some of beginner people in audio would have confusion, misunderstanding, mix-up, of "(absolute) latency" and "relative-delay/time-alignment".
This is why I wrote, rather redundantly, at the top of my pot #493 on my project thread as follows;
In the digital signal processing, we have so many buffers or latencies; JRiver output buffer, ASIO4ALL's I/O buffers, (VB-AUDIO MATRIX's buffers), EKIO's processing buffer, DIYINHK USB ASIO driver's buffer (for OKTO DAC8PRO), and so on. Consequently, it is not straightforward to exactly measure the "absolute delay" between the JRiver's "shout" and the final air sound kick-up by SP.
I usually set all the buffers in the digital domain in rather large size, so that I should not have any latency or (absolute) delay problems; in our audio setup, we have no problem at all if all the bunch of the digital and analog signal (15Hz - 30 kHz) have identical (common) amount of (absolute) delay time (latency) from the signal origin at JRiver, and this is always the case in our digital (PC based) audio system.
The relative delay between the SP units (SP drivers), or "time alignment" in multiple SPs (SP drivers), however, is always one of the critical issues in audio system, especially the multichannel multi-SP-driver multi-amplifier system, as you may agree.
I have been always take my attention and care on this issue, and in my very early posts #18 through #21, by using REW's wavelet analysis, I briefly checked that all of my SP units (SP drivers), i.e. super-tweeter (ST), Be-tweeter (TW), Be-squawker (SQ) and woofer (WO) have essentially no delay with each other, while (however/on-the-other-hand) my sub-woofer (SW) has 10 - 20 ms delay against the other SP units (SP drivers).
Now, I became really would like to establish my own simple, reliable and reproducible precision method for "time alignment" or "relative delay" measurement, and fine adjustment(s) if needed.
By the way, if you would be interested, please visit my post here #931 on my project thread for the details of the latest (as of June 26, 2024) system setup of my multichannel audio rig.
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Here is my schematic.
On the right side is the filter section, I use the windows pc ((acourate convolver)) to create the FIR filters for the crossovers and room correction. (PC Interface - Amp - Microphone). Every ASR member should use measurement equipment.
On the left is the listening section.
Smartphone as controller (bubbleupnp), RPI5 streamer with DSP (Camilladsp), multichannel DAC8X, the amplifiers and the speakers.
As a gimmick, there is also the remote power circuit for the amplifiers via the Raspberry GPIO (SSR, powercon).
On the right side is the filter section, I use the windows pc ((acourate convolver)) to create the FIR filters for the crossovers and room correction. (PC Interface - Amp - Microphone). Every ASR member should use measurement equipment.
On the left is the listening section.
Smartphone as controller (bubbleupnp), RPI5 streamer with DSP (Camilladsp), multichannel DAC8X, the amplifiers and the speakers.
As a gimmick, there is also the remote power circuit for the amplifiers via the Raspberry GPIO (SSR, powercon).
D
Deleted member 77012
Guest
Here’s my Easy Breeze chart again 
Had to add the Phono PreAmp which I forgot in the first edition.
Had to add the Phono PreAmp which I forgot in the first edition.
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Waxx
Major Contributor
This is my office system, the system i listen to the most while doing stuff on my computer. It does the job very well, but for "critical listening" i go to my main system that i posted before in this tread. Amp and DAC are in use in my hosue for over 10 years and never had issues. And both were fairly cheap but sound wonderfull good.
OP
- Thread Starter
- #197
Thank you for sharing your system diagram!
It would be very interesting reference for many people (including myself) that your are using HiFiBerry DAC8x 8-CH DAC having four of dedicated 192kHz/24bit high-quality Burr-Brown DAC chip/processor. I found your nice post here showing inside of RP51+DAC8X box.
If possible, could you please share the details of the two amplifiers and the audio interface, specification and photo of backside, since it is hard to identify them even on the enlarged diagram?
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EJ3
Master Contributor
Me, 2!
But, in 6 months or so, I'll likely see how I can implement at least one of these items into my system.
Maybe a computer that is not available on any network (except a local NAS hooked into it).
the interface is a focusrite 18i20
and the amps are the well known Aiyima A07.
My main system is the same scheme but with an RPI4. The DAC is the discontinued topping DM7 and the 4 channel amplifier IMG Stageline 2000D.
For me, a system has to be flexible and easy to customize. Now that the topping dm7 is no longer available, I need a good multichannel dac. Fortunately, the DAC8X with the rpi5 is now a replacement.