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Experiences with Prism Titan or Orpheus?


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Jussi,

I know you like to use a non-synchronous in/out when you claim to use Acourate for filter measurents. I've actually used a few different DSP calibration tools. I usually defer to the software's designer. Ask Uli which is better: synchronous or non-synchronous measurent. He has an opinion that's worth knowing.

Michael.

 

Well, I can only think of disadvantages of having both in the same, because you will end up with quite restricted set of playback devices and other unavoidable compromises.

 

I don't see any advantage of using one unit, as long as the units are not grossly misdesigned and have flat frequency and phase response exceeding frequency response of the loudspeakers.

 

I chose Focusrite Forte as measurement device because it had best specs for the job and was otherwise suitable as being portable and reasonably priced.

 

If you want, RoomEqWizard can playback and record using different devices for the measurement. I don't remember right now if Acourate was able to do the same. But with the open source DRC-FIR you can also do this without issues.

 

 

IOW, in all cases I've tried, the difference between two subsequent measurements using the same hardware is so much bigger that telling two different pieces of converter different has been impossible.

THINK OUTSIDE THE BOX

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And? Still looks like an all-in-one with a simple PEQ; Similar to a DEQX.

 

Did you see this: Illusonic Room Equalization - ILLUSONIC

 

and

 

IAP 16 - ILLUSONIC

 

and

Music in the Round #64 | Stereophile.com

 

Illusonic Immersive Processing

 

From Facebook page:

 

We just released Immersive Audio Processor (IAP 16) firmware update Version 3.0:

 

These are the main novelties and improvements:

 

– breakthrough room equalization system, including the new IAP Calibration software

 

– improved IAP Controller software

 

– more sound settings presets with more settings, including bass/treble control (with gain and frequency)

 

– virtual inputs for more flexibility and easier usage

 

– up to 2 subwoofers are supported

 

– added new infrared control codes

 

– remote control is faster and more reliable

 

– major HDMI firmware update

 

All for free for IAP 16 owners!

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Just making sure Mike. This is your area (room correction) and I wanted your considered judgement.

 

Illusonic is more targetted to commercial cinema and home cinema so I was not sure how well it addressed MCH audio for audiophiles.

 

When I visited them in Feb, they were adjusting all kinds of parameters by PC.

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I know you like to use a non-synchronous in/out when you claim to use Acourate for filter measurents. I've actually used a few different DSP calibration tools. I usually defer to the software's designer. Ask Uli which is better: synchronous or non-synchronous measurent. He has an opinion that's worth knowing.

 

I've also used many calibration tools and written quite a bunch myself. I'm every now and then talking with Uli. But "synchronous" vs "asynchronous" measurement is completely unrelated again. I can do "synchronous" measurement by playing back and recording both through Focusrite Forte and then use the generated filter for playing back through any other DAC as long as it's in the same loudspeaker system. I do understand how these algorithms work, as I've been working 20+ years on DSP software, including DRC. If you want to limit your performance to a performance of combo-unit, I don't mind.

 

As long as the result is impulse response of the system, it's all fine. Acourate has a little software module that plays and records sweep. Later on the software needs to find the sweep in the recording and calculate impulse. It cannot expect those to happen in sync, because there is more or less unknown amount of latency caused by:

1) device drivers

2) buffering in the actual converter (especially when you use something latency-pathetic as USB)

3) DAC and ADC chips used in the converters

4) biggest of all, the delay between amplifier input and the sound reaching your microphone which largely depends on the distance between microphone and loudspeakers, this delay is worth many samples' time

 

If you use anything that uses asynchronous USB, it certainly isn't going to be synchronous, no matter if AD and DA are in the same device or not. If you want something synchronous you MUST use something like Firewire.

 

If the software expects zero latency between playback and recording, it is complete crap and I don't want to be anywhere near it. It will systematically fail always.

 

As long as your recording of the sweep contains the sweep you are fine. If you are worried about the converters affecting the results, you would need to proper calibration data for AD and DA separately, as well as for the mic-amp and microphone. If you are worried about phase, I have not yet seen any microphone calibration data contain phase component for the microphone, only frequency response variations. None yet that would contain also phase response data...

 

 

P.S. Plus you can samplerate sync two or more converters using Word Clock I/O like is done in studios...

Signalyst - Developer of HQPlayer

Pulse & Fidelity - Software Defined Amplifiers

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Jussi,

In another thread you mentioned you were using the exa for out and the focus rite for IN during measurent. Now, I understand you are using one device for IN/OUT measurement and another for playback. One of the disadvantages with using one device to record/measure and another for playback is that part of the correction does take into consideration the device itself. I know these are relatively small differences but a difference nontheless.

 

For example, I used to own a Totaldac which had a nonlinear FR because it was a true NOS DAC. I had to use separate ADC/DAC to measure for my correction due to the asynchronous FIFO buffer inside the TOTALDAC. This setup produced a poor result. I know the gear you are using is much more accurate but there may still be differences.

 

Also, you seem to imply that somehow there's better sounding MCH DACs available from so called audiophile manufacturers such as exasound. I know you own one and like how it sounds. Ive had an e28 to test in my system and thought it sounded great too. However, it's truly impossible to compare different DACs using the same correction file for the reasons stated above; they each weren't in the measurement chain from the beginning. IMO, theres no way one could even opine with any conviction that one DAC is "better" than another if the DAC was never in the measurent chain to begin with.

 

If I had to guess, I would say the differences in DACs (like the ones mentioned in this thread) would be totally insignificant compared with the improvement the software can bring to the table.

 

The other disadvantage with using 2 devices is the added unnecessary expense.

 

 

I've also used many calibration tools and written quite a bunch myself. I'm every now and then talking with Uli. But "synchronous" vs "asynchronous" measurement is completely unrelated again. I can do "synchronous" measurement by playing back and recording both through Focusrite Forte and then use the generated filter for playing back through any other DAC as long as it's in the same loudspeaker system. I do understand how these algorithms work, as I've been working 20+ years on DSP software, including DRC. If you want to limit your performance to a performance of combo-unit, I don't mind.

 

As long as the result is impulse response of the system, it's all fine. Acourate has a little software module that plays and records sweep. Later on the software needs to find the sweep in the recording and calculate impulse. It cannot expect those to happen in sync, because there is more or less unknown amount of latency caused by:

1) device drivers

2) buffering in the actual converter (especially when you use something latency-pathetic as USB)

3) DAC and ADC chips used in the converters

4) biggest of all, the delay between amplifier input and the sound reaching your microphone which largely depends on the distance between microphone and loudspeakers, this delay is worth many samples' time

 

If you use anything that uses asynchronous USB, it certainly isn't going to be synchronous, no matter if AD and DA are in the same device or not. If you want something synchronous you MUST use something like Firewire.

 

If the software expects zero latency between playback and recording, it is complete crap and I don't want to be anywhere near it. It will systematically fail always.

 

As long as your recording of the sweep contains the sweep you are fine. If you are worried about the converters affecting the results, you would need to proper calibration data for AD and DA separately, as well as for the mic-amp and microphone. If you are worried about phase, I have not yet seen any microphone calibration data contain phase component for the microphone, only frequency response variations. None yet that would contain also phase response data...

THINK OUTSIDE THE BOX

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In another thread you mentioned you were using the exa for out and the focus rite for IN during measurent. Now, I understand you are using one device for IN/OUT measurement and another for playback. One of the disadvantages with using one device to record/measure and another for playback is that part of the correction does take into consideration the device itself. I know these are relatively small differences but a difference nontheless.

 

Except that the measurement software is completely unable to tell what part of the change comes from INPUT side of the chain and what comes from OUTPUT side of the chain. And thus, without separate calibration data for both, the point is moot.

 

For example, I used to own a Totaldac which had a nonlinear FR because it was a true NOS DAC. I had to use separate ADC/DAC to measure for my correction due to the asynchronous FIFO buffer inside the TOTALDAC. This setup produced a poor result.

 

Sure, if the responses are not flat, then the result is not good. But when you run the measurement and playback at 192/24 PCM with equipment without large errors you are as good as it gets. As I said, the differences between multiple subsequent measurements done after each other are so big, that swapping in another piece of AD/DA doesn't make any notable difference compared to random variation between subsequent measurement.

 

Also, you seem to imply that somehow there's better sounding MCH DACs available from so called audiophile manufacturers such as exasound. I know you own one and like how it sounds. Ive had an e28 to test in my system and thought it sounded great too. However, it's truly impossible to compare different DACs using the same correction file for the reasons stated above; they each weren't in the measurement chain from the beginning.

 

As long as the measurement chain doesn't know all the errors coming purely from the ADC side it doesn't really matter. Result with some other DAC could actually result in more correct output...

 

The other disadvantage with using 2 devices is the added unnecessary expense.

 

I can use the same measurement system with multiple different playback systems, as it is portable. So it is actually hugely beneficial.

Signalyst - Developer of HQPlayer

Pulse & Fidelity - Software Defined Amplifiers

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Regarding measurements with exaSound e28 and other DACs having S/PDIF input. If you want to include DAC into the measurement chain, you can use interface like Fireface UCX to route output through S/PDIF to the DAC and then use the built-in microphone inputs for the measurement mic.

 

Then just measure at 192/24 or similar rate to make sure the responses are at least flat within 0 - 20 kHz.

 

After measuring, you can then take the measurement interface out of the system and feed DAC straight trough USB with corrections.

Signalyst - Developer of HQPlayer

Pulse & Fidelity - Software Defined Amplifiers

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Spdif is only 2CH. The exa could benefit from an analog input or at least a word clock input.

 

 

Regarding measurements with exaSound e28 and other DACs having S/PDIF input. If you want to include DAC into the measurement chain, you can use interface like Fireface UCX to route output through S/PDIF to the DAC and then use the built-in microphone inputs for the measurement mic.

 

Then just measure at 192/24 or similar rate to make sure the responses are at least flat within 0 - 20 kHz.

 

After measuring, you can then take the measurement interface out of the system and feed DAC straight trough USB with corrections.

THINK OUTSIDE THE BOX

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Spdif is only 2CH. The exa could benefit from an analog input or at least a word clock input.

 

Well, the measurement is anyway one channel at a time, so it is not an issue. One channel is enough.

 

S/PDIF can serve as word clock input like on many pro-audio interfaces you can select if you want to source word clock from S/PDIF or WC BNC input. But the firmware doesn't allow syncing on external clock. However, the sweep is so short in time, that the drift between two clocks in that time should be way less than period of a single sample at 192k. Correction should use something like 1/6th octave smoothing on the response which would smooth out effects of much bigger drifts. You probably won't be able to sit still enough to have less doppler effect due to your head movement... :)

 

If you play back through S/PDIF then you don't need word clock, because S/PDIF provides same functionality for you already (being synchronous interface).

 

Mytek DAC has all the possible inputs and sync options you could ever imagine needing for this kind of purpose. :) Of course it is stereo-only, but it at least supports DSD which is hard minimum-requirement for me.

Signalyst - Developer of HQPlayer

Pulse & Fidelity - Software Defined Amplifiers

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The OP wants to do a multiway crossover with target based room correction. He will need to measure a lot more than 1 channel at a time. In my case, I need to measure 3 channels at a time.

 

Btw, the Lynx Hilo meets all requirements including DSD. Multiple Hilo can be connected together over thunderbolt.

image.jpg

 

 

Well, the measurement is anyway one channel at a time, so it is not an issue. One channel is enough.

 

S/PDIF can serve as word clock input like on many pro-audio interfaces you can select if you want to source word clock from S/PDIF or WC BNC input. But the firmware doesn't allow syncing on external clock. However, the sweep is so short in time, that the drift between two clocks in that time should be way less than period of a single sample at 192k. Correction should use something like 1/6th octave smoothing on the response which would smooth out effects of much bigger drifts. You probably won't be able to sit still enough to have less doppler effect due to your head movement... :)

 

If you play back through S/PDIF then you don't need word clock, because S/PDIF provides same functionality for you already (being synchronous interface).

 

Mytek DAC has all the possible inputs and sync options you could ever imagine needing for this kind of purpose. :) Of course it is stereo-only, but it at least supports DSD which is hard minimum-requirement for me.

THINK OUTSIDE THE BOX

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The OP wants to do a multiway crossover with target based room correction. He will need to measure a lot more than 1 channel at a time. In my case, I need to measure 3 channels at a time.

 

That's another thing... I'm still doing ordinary multichannel with passive cross-overs inside speakers. 3-way 5.1 channel system would be already 18 channels, doable with studio gear. But then I would build a system with 24 or 32 channels of DACs and with just stereo ADC. Combination not necessarily being available as single box.

 

Btw, the Lynx Hilo meets all requirements including DSD. Multiple Hilo can be connected together over thunderbolt.

 

Quite many firewire, PCIe and ethernet based studio systems allow connecting multiple converters. If DSD64 is enough, you could use some multi-AES interface and connect stack of Mytek DACs too (or any other DAC that supports DoP over AES or S/PDIF).

 

I think with dCS DACs you could also achieve DSD128 this way since it support dualwire-AES and DoP?

 

But Hilo is still just DSD64? Although it's CS4398 DAC chips would allow DSD128 too.

Signalyst - Developer of HQPlayer

Pulse & Fidelity - Software Defined Amplifiers

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I've had a Prism Orpheus for several years now (older FW version) and have been happy with the SQ. I use it for recording and playback and run it 2Ch into a pair of Genelec 8020As. I did add a Goldpoint volume control, and I prefer the SQ running the Orpheus full out into the analogue volume control over using its digital volume control. The headphone amp in the Orpheus is just ok. I'd recommend a separate headphone amp. Also, Prism's customer service has been great.

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I've had a Prism Orpheus for several years now (older FW version) and have been happy with the SQ. I use it for recording and playback and run it 2Ch into a pair of Genelec 8020As. I did add a Goldpoint volume control, and I prefer the SQ running the Orpheus full out into the analogue volume control over using its digital volume control. The headphone amp in the Orpheus is just ok. I'd recommend a separate headphone amp. Also, Prism's customer service has been great.

 

Hey Gary.

Thanks for the input. The Orpheus is, as far as I can tell, the closest thing to the Titan. And the fact you prefer to add an external volume control is a cautionary notice for me - it would be hard fir me to do so in an 8-channel setup. I appreciate the candid feedback!

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Btw, the Lynx Hilo meets all requirements including DSD. Multiple Hilo can be connected together over thunderbolt.

[ATTACH=CONFIG]15201[/ATTACH]

 

Michael,

Had not thought about that! Now two Hilos together vs a single Titan look a lot more tempting. It would allow me to get one Hilo, crossover the subwoofers and mains immediately. As soon as I get the amp/s for the bass I could still use one Hilo, and when I'm ready for the next amp/s for the treble I could get another Hilo...interesting!

 

The one thing that still bugs me a bit is the dependency on a software volume control. Maybe I just need to get over it.

 

Do you know what is needed to get two Hilos linked as described? You mentioned Thunderbolt. I currently use USB out of the PC. Should I get the first Hilo with the Thunderbolt option, or could it be the second one?

 

Uli's endorsement of the Titan is certainly meaningful. Yet the Hilo also looks as a great option.

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Hello,

 

I've had an Orpheus for many years. I originally bought it for studio use, but often it would do double duty in my listening room for use with room correction filters and digital crossovers. I've compared the Orpheus to many audiophile DACs, including the Meitner MA-1, Playback Designs MPD-5, Berkeley Alpha series 2, and many pro sound DACs (but not the Hilo). There are a lot of great sounding DACs out there, and a number of less good sounding DACs. I've never considered selling the Orpheus, because it sounds great, works great, and is very simple to use in regards to room correction workflow.

 

If I were looking for a two channel DAC to put into my system without running any correction filters, the Orpheus would probably not be my first choice because I would be finding a DAC with an exact flavor that fit my system and taste at the time. However, for use with room correction filters, I want a DAC that does well in all areas of the frequency spectrum, and does not impart any edginess or glare. In this manner, it should be able to respond to what I feed into it via my chosen target curve. The Orpheus does so very well. It has excellent bass response, transparency without imparting a lack of body, and excellent response in the high frequencies as well.

 

I read above that one Orpheus owner does not like the volume control, but I think the volume control on the Orpheus sounds better than using JRiver's "internal" volume control. As well I have a few preamps to choose from, and the Orpheus does very well driving long interconnects without a preamp. The Orpheus outputs are switchable between +4 and -10 dB. At the -10 setting and with room correction filters, I do not generally have to attenuate very much.

 

I would like to hear the Lynx Hilo. I've heard good things about its sound quality from another Orpheus owner, but he never said one sounded distinctly better than the other. Having to divide the output channels between main, monitor and headphone outputs is probably not ideal, but if it sounds good who cares.

 

Alan

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  • 6 months later...

Hello, i am going full active with 4-ways fronts and so for multi channel audio i need at least 12 audio channels... I am Now using Lynx aes 16 pc and two Lucid converters

I would line to i prove DA conversion, so i need a top 16 channels converter...

Prism ADA8xr with aes and 16 analog outs is around 12k euros, so that is IZ ADA...

Among more affordablle solutions there are Lynx aurora, Antelope Orion 32 and Focusrite rednet 2

Are those latter a real improvement to the Lucid??? Which one? I am tempted by network based Focusrite (i don't know why but merging HAPI is not really tempting me...)

 

Thanks

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If you go AES67, AVB or Ravenna, be careful that there is an ASIO driver for it and the ASIO driver will be Steinberg compliant; will work properly with 3rd party softwares like REW, Acourate, Audiolense, DIRAC, rephase, etc. Pro gear is made for pro 3rd party apps not necessarily for the apps you may need to get proper delay/phase/crossovers for active speakers.

 

IF you are going with an ethernet capable converter, make sure it also has the USB option as well. I think the prism gear will soon have ethernet and I think their USB driver is very good. I don't think Lynx Hilo will ever be ethernet capable but the USB driver is solid and its not difficult to find USB/ethernet extenders like the startech.

 

Hello, i am going full active with 4-ways fronts and so for multi channel audio i need at least 12 audio channels... I am Now using Lynx aes 16 pc and two Lucid converters

I would line to i prove DA conversion, so i need a top 16 channels converter...

Prism ADA8xr with aes and 16 analog outs is around 12k euros, so that is IZ ADA...

Among more affordablle solutions there are Lynx aurora, Antelope Orion 32 and Focusrite rednet 2

Are those latter a real improvement to the Lucid??? Which one? I am tempted by network based Focusrite (i don't know why but merging HAPI is not really tempting me...)

 

Thanks

THINK OUTSIDE THE BOX

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  • 5 months later...

Hi,

 

I'm looking at the same sort of stuff. Planning on doing 3,5 way active so in need of 8-channel DAC. I bought a Motu Hybrid MKIII (MOTU.com - UltraLite-mk3 Hybrid Overview) to get a working prototype. The unit has 8-in, 8-out and a Mic preamp so all you need is in one box. It has USB + FW. For the money not a bad dac at all.

 

Once it's all up an running I'd be looking at a Sabre based multichannel dac. Units that interest me are the Motu 24Ao, 16A, Focusrite Rednet 1, Universal Audio Apollo (not sure this has Sabre converters). What I like about the Focusrite (Dante) and Motu's (AVB) is that they (can) interface via Ethernet (audio + control over IP). I just want the best bang for buck and not pay for 32 mic and headphone inputs etc which I won't be using. Looking forward to see which other candidates might pop-up. I'd really want to stay below €2K.

 

Reds,

 

Bukem

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The Motu maybe the best option. You may want to find the Motu thread on Jriver forum and talk to Mojave about his Motu. He's a fount of information and very friendly.

 

 

Hi,

 

I'm looking at the same sort of stuff. Planning on doing 3,5 way active so in need of 8-channel DAC. I bought a Motu Hybrid MKIII (MOTU.com - UltraLite-mk3 Hybrid Overview) to get a working prototype. The unit has 8-in, 8-out and a Mic preamp so all you need is in one box. It has USB + FW. For the money not a bad dac at all.

 

Once it's all up an running I'd be looking at a Sabre based multichannel dac. Units that interest me are the Motu 24Ao, 16A, Focusrite Rednet 1, Universal Audio Apollo (not sure this has Sabre converters). What I like about the Focusrite (Dante) and Motu's (AVB) is that they (can) interface via Ethernet (audio + control over IP). I just want the best bang for buck and not pay for 32 mic and headphone inputs etc which I won't be using. Looking forward to see which other candidates might pop-up. I'd really want to stay below €2K.

 

Reds,

 

Bukem

THINK OUTSIDE THE BOX

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The Motu maybe the best option. You may want to find the Motu thread on Jriver forum and talk to Mojave about his Motu. He's a fount of information and very friendly.

 

Thanks for the tip. Will certainly do so. What made you steer towards the Motu?

 

I was planning on doing all the EQ inside my DAW (Logic X or MainStage) via the Apple or Fabfilter plugins using iTunes as source. From memory Jriver is not on OSX. I will check.

 

Cheers,

 

Bukem

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I use a lynx Hilo. I don't have the Motu. I have a partially digital active system. I may switch to fully digital active in the future which would require 6 out. If/when I do that, I'll probably switch over to a prism. I'm very happy with the Hilo. It just doesn't have a channel count and level control I would want for more than 4CH digital active.

 

 

Thanks for the tip. Will certainly do so. What made you steer towards the Motu?

 

I was planning on doing all the EQ inside my DAW (Logic X or MainStage) via the Apple or Fabfilter plugins using iTunes as source. From memory Jriver is not on OSX. I will check.

 

Cheers,

 

Bukem

THINK OUTSIDE THE BOX

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  • 8 years later...

Like the eight channel Exasound and Motu, does the Prism Titan use the ESS DAC chips?

Unfortunately, at least how ESS chips are typically configured by the user-along with output stage design-they do nothing to offset intersample overs distortion, as rigorously discussed here. https://benchmarkmedia.com/blogs/application_notes/intersample-overs-in-cd-recordings?_pos=1&_sid=0eeb1f150&_ss=r 

 

And especially here

https://gearspace.com/board/mastering-forum/1401406-intersample-clipping-audible-19.html#post16787090

 

 

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