Hello, I am new to this forum but reading already for quite some time interesting questions and answers.
Over the years I developed myself a nice audio system which I really enjoy (system components listed below). I did investigate what is the best option for me to improve my system and I think replacing the analog active filter I use by a DSP based system would be a nice next step. I want to move to DSP based crossover, phase correction, time alignment, room correction and have some questions related. I think PC software based DSP is the best way for me to go for.
I am planning to use Acourate for measurement and creating the filters.
At the moment I am using Roon and I did read Roon can handle these filters so that could be interesting for me. Otherwise JRiver is one of the best option as far as I understood.
I now try to make the right design decisions regarding the hardware and interfaces to use.
Question 1: I am thinking of using an fanless NUC with i5 or i7 and 8GB RAM. Any advice on this?
Question 2: I am planning to add 2 more Metrum DAC’s to my system to create a DSP based 3-way system. What is the best option to connect these DAC’s to the NUC? Use the USB port of the Metrum and connect 3 USB cables from NUC to the 3 Metrum DAC’s? I have the feeling this can generate timing (clock) problems. But is this more theoretical are also audible?
Or is it better to add an audio card to the NUC with for example 3x AES/EBU output. Or is it better to use something like “XMOS USB Digital Interface 32/384khz AES EBU” like the XMOS USB Digital Interface 32/384khz AES EBU TCXO Alimentation 230V Silver - Audiophonics or SINGXER SU-2 USB Digital Interface 32bit 768khz DSD1024 SPDIF AES/EBU I2S HDMI LVDS - Audiophonics ?
Or do you advice a different way to connect the DSP NUC to the DAC’s?
In short my current audio system:
• Magnepan MG20.1 (fully modified, silver foil wiring, no internal crossover, all external with high quality components)
• 2x Velodyne DD12 (I am not using the bass panel of the MG20.1 because it has a negative influence on the mid panel and tweeter. The 2x DD12 replaces the bass panels and using the internal DSP I already improved the low level frequency domain in combination with my room.)
• Metrum Ambre source
• Metrum Adagio DAC (volume control normally not used)
• Audio Research LS26 (modified with ODAM caps)
• Magnepan active analog filter (crossover between bass and mid/high)
• Krell KSA 300S (upgraded version) and Audio Research VS115 (I use 1 of both to power the MG20.1. The Krell is better in controlling the mid panel and the VS115 creates a nicer and bigger soundstage with the big ribbon tweeter)
• Interlinks Transparent Balanced reference and speaker wires self made using Mundorf 44mm foil and Teflon isolator.
Over the years I developed myself a nice audio system which I really enjoy (system components listed below). I did investigate what is the best option for me to improve my system and I think replacing the analog active filter I use by a DSP based system would be a nice next step. I want to move to DSP based crossover, phase correction, time alignment, room correction and have some questions related. I think PC software based DSP is the best way for me to go for.
I am planning to use Acourate for measurement and creating the filters.
At the moment I am using Roon and I did read Roon can handle these filters so that could be interesting for me. Otherwise JRiver is one of the best option as far as I understood.
I now try to make the right design decisions regarding the hardware and interfaces to use.
Question 1: I am thinking of using an fanless NUC with i5 or i7 and 8GB RAM. Any advice on this?
Question 2: I am planning to add 2 more Metrum DAC’s to my system to create a DSP based 3-way system. What is the best option to connect these DAC’s to the NUC? Use the USB port of the Metrum and connect 3 USB cables from NUC to the 3 Metrum DAC’s? I have the feeling this can generate timing (clock) problems. But is this more theoretical are also audible?
Or is it better to add an audio card to the NUC with for example 3x AES/EBU output. Or is it better to use something like “XMOS USB Digital Interface 32/384khz AES EBU” like the XMOS USB Digital Interface 32/384khz AES EBU TCXO Alimentation 230V Silver - Audiophonics or SINGXER SU-2 USB Digital Interface 32bit 768khz DSD1024 SPDIF AES/EBU I2S HDMI LVDS - Audiophonics ?
Or do you advice a different way to connect the DSP NUC to the DAC’s?
In short my current audio system:
• Magnepan MG20.1 (fully modified, silver foil wiring, no internal crossover, all external with high quality components)
• 2x Velodyne DD12 (I am not using the bass panel of the MG20.1 because it has a negative influence on the mid panel and tweeter. The 2x DD12 replaces the bass panels and using the internal DSP I already improved the low level frequency domain in combination with my room.)
• Metrum Ambre source
• Metrum Adagio DAC (volume control normally not used)
• Audio Research LS26 (modified with ODAM caps)
• Magnepan active analog filter (crossover between bass and mid/high)
• Krell KSA 300S (upgraded version) and Audio Research VS115 (I use 1 of both to power the MG20.1. The Krell is better in controlling the mid panel and the VS115 creates a nicer and bigger soundstage with the big ribbon tweeter)
• Interlinks Transparent Balanced reference and speaker wires self made using Mundorf 44mm foil and Teflon isolator.
Looks like Metrum DACs can be configured for use with USB or else with I2S interfaces. With multiple DACs, USB cannot be guaranteed sample accurate between dacs since each DAC is running from different internal clocks. Also, an OS and drivers probably can't be guaranteed to start playing via multiple dacs all exactly at the same time.
OTOH, using I2S interfaces to the DACs could likely solve such problems. However, one would need to know the particular I2S implementation supported by Metrum. In addition, some custom hardware might need to be designed to drive all the dacs synchronously via I2S.
OTOH, using I2S interfaces to the DACs could likely solve such problems. However, one would need to know the particular I2S implementation supported by Metrum. In addition, some custom hardware might need to be designed to drive all the dacs synchronously via I2S.
Hello Markw4, Yhx for your replay!
Yes, my Metrum Adagio has 1x Optical, 2x Coax, 1x AES/EBU, 1x USB or I²S as input. Agree that USb is not the right option. I was told that when you use AES/EUB and use a PC card like the Lynxs AES16e AES16e - Products - Lynx Studio Technology, Inc. the output of the 3 AES channels I need should be that good in sync that you can use the as input for the of the same DAC’s without any timing problems. Somebody told me that timing differences of a speaker being both not exactly the same distance from your ears is bigger.
I2S indeed looks interesting because of the clock signal which is communicated. But until now I did not find a good PC card or external box solution that enables me to send out 3 I2S channels.
Yes, my Metrum Adagio has 1x Optical, 2x Coax, 1x AES/EBU, 1x USB or I²S as input. Agree that USb is not the right option. I was told that when you use AES/EUB and use a PC card like the Lynxs AES16e AES16e - Products - Lynx Studio Technology, Inc. the output of the 3 AES channels I need should be that good in sync that you can use the as input for the of the same DAC’s without any timing problems. Somebody told me that timing differences of a speaker being both not exactly the same distance from your ears is bigger.
I2S indeed looks interesting because of the clock signal which is communicated. But until now I did not find a good PC card or external box solution that enables me to send out 3 I2S channels.
"Somebody told me that timing differences of a speaker being both not exactly the same distance from your ears is bigger."
IMHO that is not necessarily true. The question to me would be one of jitter and its effect on imaging. The ear/brain estimates horizontal virtual sound source position by the difference in time of arrival of a sound wave at each ear. Interaural time difference - Wikipedia That being the case, it is timing difference from where a virtual performer would exist between the speakers, not so much exactly where the listener is relation to speakers. IME, among other problems, jitter widens the perceived virtual source position of an instrument.
However in regard to unequal distance from speakers, when the listener is not centered between the speakers then speaker dispersion non-uniformity can cause issues at the listener position. The best solution I am aware of is to choose speakers with wide uniform dispersion and place them carefully in the room. There are various approaches to speaker positioning, perhaps I could find a link if there is further interest in that topic.
The question of jitter when using SPDIF would depend on how Metrum DACs process SPDIF. Is a PLL used to derive a recovered clock from the SPDIF stream, is ASRC used to resample SPDIF digital audio to a low-jitter master clock, or perhaps is a FIFO used to remove SPDIF/PLL jitter? I ask because you may find there is an audible difference between the different approaches that may matter to you, perhaps depending on how you personally perceive sound.
As an alternative to SPDIF, I2S inputs usually include an external master clock which can be of quite low jitter. My guess would be that could give the best SQ if the clock and I2S distribution system was well designed. Unfortunately there are no known commercial solutions known to me. Results might also depend on how Metrum processing external I2S: Do they use an external master clock to operate the DAC, or do they resample or FIFO the external I2S input?
Because of the above considerations, some people wanting to multi-amp stereo speakers (and or to implement surround sound) look for single DACs boxes that support several synchronous channels at once. Some multi-channel DACs designed for use in recording studios might be good examples of that type.
IMHO that is not necessarily true. The question to me would be one of jitter and its effect on imaging. The ear/brain estimates horizontal virtual sound source position by the difference in time of arrival of a sound wave at each ear. Interaural time difference - Wikipedia That being the case, it is timing difference from where a virtual performer would exist between the speakers, not so much exactly where the listener is relation to speakers. IME, among other problems, jitter widens the perceived virtual source position of an instrument.
However in regard to unequal distance from speakers, when the listener is not centered between the speakers then speaker dispersion non-uniformity can cause issues at the listener position. The best solution I am aware of is to choose speakers with wide uniform dispersion and place them carefully in the room. There are various approaches to speaker positioning, perhaps I could find a link if there is further interest in that topic.
The question of jitter when using SPDIF would depend on how Metrum DACs process SPDIF. Is a PLL used to derive a recovered clock from the SPDIF stream, is ASRC used to resample SPDIF digital audio to a low-jitter master clock, or perhaps is a FIFO used to remove SPDIF/PLL jitter? I ask because you may find there is an audible difference between the different approaches that may matter to you, perhaps depending on how you personally perceive sound.
As an alternative to SPDIF, I2S inputs usually include an external master clock which can be of quite low jitter. My guess would be that could give the best SQ if the clock and I2S distribution system was well designed. Unfortunately there are no known commercial solutions known to me. Results might also depend on how Metrum processing external I2S: Do they use an external master clock to operate the DAC, or do they resample or FIFO the external I2S input?
Because of the above considerations, some people wanting to multi-amp stereo speakers (and or to implement surround sound) look for single DACs boxes that support several synchronous channels at once. Some multi-channel DACs designed for use in recording studios might be good examples of that type.
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Typical time difference for only 1 degree of angular difference between ears is 9000ns, typical jitter in I2S etc is < 1ns. Totally not any kind of issue.
And these signals carry L and R samples interleaved so that jitter is correlated strongly for the left and right signals anyway.
The problem with jitter is the noise and distortion it modulates on top of a signal. Take a 44.1kSPS 16 bit signal of a full-amplitude 11kHz waveform - the successive samples change by 2^15.
Thus jitter of 1/(2^15) times the sample period equates to 1 bit error in the signal magnitude. This is +/-0.7ns of jitter. So theoretically in the face of such high strength HF signals this level of jitter will start to create artifacts above the noise floor of a 16 bit source. Real signals won't be anything like as harsh and thus less sensitive to jitter distortion.
And these signals carry L and R samples interleaved so that jitter is correlated strongly for the left and right signals anyway.
The problem with jitter is the noise and distortion it modulates on top of a signal. Take a 44.1kSPS 16 bit signal of a full-amplitude 11kHz waveform - the successive samples change by 2^15.
Thus jitter of 1/(2^15) times the sample period equates to 1 bit error in the signal magnitude. This is +/-0.7ns of jitter. So theoretically in the face of such high strength HF signals this level of jitter will start to create artifacts above the noise floor of a 16 bit source. Real signals won't be anything like as harsh and thus less sensitive to jitter distortion.
>But until now I did not find a good PC card or external box solution that enables me to send out 3 I2S channels.
Since there's no real connectivity standard for I2S (except within the same PCB layout or between boards in an enclosed device) I suspect those capable of designing such a thing have shied away from the investment expense. Who wants to build something that complex only a DIY'er could use, without guaranteed results?
It seems that the Lynx outfit you linked could, but I see they stick to the standard that anyone can just plug 'n play.
I imagine their card with 3 independent digital stereo outs into 3 DACs with your DSP running on the PC is going to work out pretty well. I believe I5 would provide plenty of power for handling the audio and DSP. I7 would be a waste of cores, IHMO. These things are so damn fast these days, I suspect executing the DSP code is hardly anything for them - they probably blow though a group of samples in a relative instant, load it up into memory - and then wait around for the DAC card to DMA access that chunk and finish clocking it out.
Since there's no real connectivity standard for I2S (except within the same PCB layout or between boards in an enclosed device) I suspect those capable of designing such a thing have shied away from the investment expense. Who wants to build something that complex only a DIY'er could use, without guaranteed results?
It seems that the Lynx outfit you linked could, but I see they stick to the standard that anyone can just plug 'n play.
I imagine their card with 3 independent digital stereo outs into 3 DACs with your DSP running on the PC is going to work out pretty well. I believe I5 would provide plenty of power for handling the audio and DSP. I7 would be a waste of cores, IHMO. These things are so damn fast these days, I suspect executing the DSP code is hardly anything for them - they probably blow though a group of samples in a relative instant, load it up into memory - and then wait around for the DAC card to DMA access that chunk and finish clocking it out.
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Except that it must fold down into time intervals we can hear since it is experimentally audible, at least that seems to be the case with some common sigma delta architectures. Maybe not with old non-OS dacs.
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There is no clock issues using pro hardware.
Aes has two mode regarding clock distribution: either selfclocked ( the clock is embeded within the digital audio stream as Spdif) or (if your hardware have a dedicated bnc socket) wordclock signal.
The first mode is usually used for stereo, if you need multiple channels with different digital gear within a chain ( typical: ad/ digital mixer/ computer/effects/dac) the second option will be used to distribute one and only clock reference to everyone in the chain ( and so all channels being 2 or 128 will all be clocked along the same reference: mandatory for multitrack recording/ mixing). Of course you can use a stereo signal with wordclock too but you'll have to invest in a good wordclock generator ( it isn't cheap, in the same range as good converters) to really have a step up in quality ( if any as some gear could sound best selfclocked!).
The timing problems only arise when you use connection which are not 'clocked' ( usb).
In proaudio you can't waste time with things like that so protocols/norm used are able to lock and run multichanel ( Aes, Adat, Madi, Spdif,...).
Any digital pro out card will be up to the task ( Lynx, Rme,...).
I use an Rme aes to drive a DSP without any glitch for the last 10 years ( and use another RME adat based card 24/7 with a digital desk at a webradio for 5 years now with same outcome...) .
@JeroenD: As can be seen there are a variety of opinions about what is or isn't audible in terms of imperfections in reproduction systems. Its a decades old debate that has never been resolved to the satisfaction of all parties.
It your case what may or may not be audible will depend on how you listen, how picky you may be about perfection, and the degree of perfection in your overall reproduction system. If you would like to discuss in more detail, feel free to PM me.
It your case what may or may not be audible will depend on how you listen, how picky you may be about perfection, and the degree of perfection in your overall reproduction system. If you would like to discuss in more detail, feel free to PM me.
Markw4,
Why discuss this in private, let's talk about it in public. I'm sure it could be nice for the OP to have different pov to make his own informed choice.
Why discuss this in private, let's talk about it in public. I'm sure it could be nice for the OP to have different pov to make his own informed choice.
Hello Markw4, Mark Tillotson, jjasniew, krivium, and thank you for your explanation, examples and proposals. That really helps me in defining what hardware to use.
I think going the I2S way is not a good option. The only suitable hardware I found is the MiniDSP MCHStreamer Kit. I have no problem to create a nice boxed solution with good connectors etc. but I have some doubts about the quality. I am aiming for really high quality also because my audio system is able to reveal al the smallest details and changes. So spending a little more and have a high quality and robust solution is my approach (indeed more a perfectionist looking for a good balance between cost and result ).
I also contacted a designers at Metrum and asked him what the best interface to use regarding my use case. He indicated that I2S could be a good solution but that off the shelf solutions are rare and he is not sure if the MCHStreamer Kit would work with a Metrum DAC. He also warned to use a robust and stable solution that has a proven track record avoiding streaming problems, glitches etc. He suggested to use the AES/ EBU interface with a professional computer card or external solution that has a good internal clock to guarantee that the 3 AES/ EBU output signals, I want to send to the Metrum DAC’s, are in sync. He indicated that this way the DAC’s will perfectly process the digital signal in sync like supplied by the computer.
So back to my questions:
Answer to question 1: A computer with i5 and 8GB RAM will do the job.
Answer to question 2: Based on the feedback in this forum and from Metrum I think the best and most stable solution is to go for a professional PC card or external device (like the Merging Hapi). It looks like a professional PCIe card is cost wise the best option without any quality loss.
I now have 2 option on my list for a suitable professional PCIe card with minimal 3 AES/ EBU channels out.
• Lynxs AES16e AES16e - Products - Lynx Studio Technology, Inc.
• RME HDSPe AES HDSPe AES - RME Audio Interfaces | Format Converters | Preamps | Network Audio & MADI Solutions
At the moment I think the Lynxs AES16e is the best option because:
• Cheaper
• Needs only 1 PCIe card slot
Until now I could not find a good reason to choose the RME HDSPe AES over the Lynxs AES16e.
Can you advice on this?
Am I missing other good options in my list?
I think going the I2S way is not a good option. The only suitable hardware I found is the MiniDSP MCHStreamer Kit. I have no problem to create a nice boxed solution with good connectors etc. but I have some doubts about the quality. I am aiming for really high quality also because my audio system is able to reveal al the smallest details and changes. So spending a little more and have a high quality and robust solution is my approach (indeed more a perfectionist looking for a good balance between cost and result ).
I also contacted a designers at Metrum and asked him what the best interface to use regarding my use case. He indicated that I2S could be a good solution but that off the shelf solutions are rare and he is not sure if the MCHStreamer Kit would work with a Metrum DAC. He also warned to use a robust and stable solution that has a proven track record avoiding streaming problems, glitches etc. He suggested to use the AES/ EBU interface with a professional computer card or external solution that has a good internal clock to guarantee that the 3 AES/ EBU output signals, I want to send to the Metrum DAC’s, are in sync. He indicated that this way the DAC’s will perfectly process the digital signal in sync like supplied by the computer.
So back to my questions:
Answer to question 1: A computer with i5 and 8GB RAM will do the job.
Answer to question 2: Based on the feedback in this forum and from Metrum I think the best and most stable solution is to go for a professional PC card or external device (like the Merging Hapi). It looks like a professional PCIe card is cost wise the best option without any quality loss.
I now have 2 option on my list for a suitable professional PCIe card with minimal 3 AES/ EBU channels out.
• Lynxs AES16e AES16e - Products - Lynx Studio Technology, Inc.
• RME HDSPe AES HDSPe AES - RME Audio Interfaces | Format Converters | Preamps | Network Audio & MADI Solutions
At the moment I think the Lynxs AES16e is the best option because:
• Cheaper
• Needs only 1 PCIe card slot
Until now I could not find a good reason to choose the RME HDSPe AES over the Lynxs AES16e.
Can you advice on this?
Am I missing other good options in my list?
Hi,
As pointed by Markw4 there is multiple options possible.
I'll talk about the one i know which is from pro studio.
I have used both cards Rme and Lynx in pro context ( as well as Motu, Protools HD with Apogee converters, Prism audio, Antelope,...) both are same quality to me: stable, transparent.
As i am in EU Rme are widely and cheaply availlable second hand and i like their routing utility 'totalmix' (which may be or not of importance for your own use), so used them.
Lynx cards go well with their Aurora converters. A second hand 8 chanel should be fairly affordable now and is a very respectable converter imho. Same league as Lavry Black/blue, Apogee, Mytek.
If you want absolute transparency Prism sound ADA8 XR but the price is hard to swallow imho.
Anyway with Accourate and a Lynx bundle you'll have a killer system imho.
Otherwise they have recent version you can connect via usb to computer. The sound quality is pristine and afaik those are stable too ( they have too or won't be used!). But they are around 3000euros. Still hard to swallow for me even if only half an ADA8...
As you plan to use Acourate may i suggest you to read this?:
Accurate Sound Reproduction Using DSP : Barnett, Mitch: Amazon.fr: Livres
As pointed by Markw4 there is multiple options possible.
I'll talk about the one i know which is from pro studio.
I have used both cards Rme and Lynx in pro context ( as well as Motu, Protools HD with Apogee converters, Prism audio, Antelope,...) both are same quality to me: stable, transparent.
As i am in EU Rme are widely and cheaply availlable second hand and i like their routing utility 'totalmix' (which may be or not of importance for your own use), so used them.
Lynx cards go well with their Aurora converters. A second hand 8 chanel should be fairly affordable now and is a very respectable converter imho. Same league as Lavry Black/blue, Apogee, Mytek.
If you want absolute transparency Prism sound ADA8 XR but the price is hard to swallow imho.
Anyway with Accourate and a Lynx bundle you'll have a killer system imho.
Otherwise they have recent version you can connect via usb to computer. The sound quality is pristine and afaik those are stable too ( they have too or won't be used!). But they are around 3000euros. Still hard to swallow for me even if only half an ADA8...
As you plan to use Acourate may i suggest you to read this?:
Accurate Sound Reproduction Using DSP : Barnett, Mitch: Amazon.fr: Livres
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I don't know anyone using Accurate or Audiolense for active crossovers that uses multiple consumer grade DACs, no matter how perfect they are for 2 channel operation. There's a reason pro gear uses external clocking for multiple interfaces, because otherwise each DAC is de-jittering to a different internal clock. This is especially important if you want to use linear phase crossovers.
For your proposed budget on two new DACs there's a significant number of mastering studio grade interfaces to choose from with 8 or more channels of balanced output.
Hi,
You could do it with 'consumer' 2 chanel dac : by using Spdif out and a digital out card as RME Digiface as TNT pointed.
But you have to be sure the optical receiver in the consumer DAC is good quality ( i have no doubt about RME side).
I must say i've been tempted to do that: a digiface and some SMSL Sanskrit 10th or Topping units could make a nice sounding affordable system.
But except if you want to be cheap i don't see the point if you can afford a pro 8x dac unit.
About the concern with computer, i run on a secondary system an older pc with XP, 2x4ghz proc, 3gig ram and old pci pro multiple out soundcard for FIR without issue (except latency (but you'll have it anyway and whatever hardware you use if using fir)and max sampling frequency limited by soundcard).
For Acourate follow recommendations given.
You could do it with 'consumer' 2 chanel dac : by using Spdif out and a digital out card as RME Digiface as TNT pointed.
But you have to be sure the optical receiver in the consumer DAC is good quality ( i have no doubt about RME side).
I must say i've been tempted to do that: a digiface and some SMSL Sanskrit 10th or Topping units could make a nice sounding affordable system.
But except if you want to be cheap i don't see the point if you can afford a pro 8x dac unit.
About the concern with computer, i run on a secondary system an older pc with XP, 2x4ghz proc, 3gig ram and old pci pro multiple out soundcard for FIR without issue (except latency (but you'll have it anyway and whatever hardware you use if using fir)and max sampling frequency limited by soundcard).
For Acourate follow recommendations given.
Thinking about a peasant version of what the OP is targeting.
Would the difference between a PCIe>AES>DAC system and a USB>DAC system (e.g MOTU.com - Overview ) realistically be audible given the weakest link is the speaker?
Would the difference between a PCIe>AES>DAC system and a USB>DAC system (e.g MOTU.com - Overview ) realistically be audible given the weakest link is the speaker?
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RME seem to be good stuff. I have the Toslink version and my 2 DAM 1121 (one for each channel balanced) never sound this good. I think it is not the receiver itself that is critical but rather how the clock handling is made on the other side... After finding out that one can inhibit the clock adjustment in the DAMs by entering the control mode on the RS232 i/f its very good indeed. And these 2 DACs have no sync line between them - I run them very close to each speaker directly via a Nc400...
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Doubtful to me that multi-dac SPDIF, including the AES variant, can be truly be top notch (for reasons previously stated). IME there are probably no commercial products that will do what the OP wants if highest possible sound quality is the goal.
First of all, there is no such thing as a perfectly transparent dac. It doesn't matter how much one spends. Not Chord DAVE with prescaler, not Holo May, not the most exotic dCS, and not Prism. They all sound different if the rest of the reproduction system is good enough to audibly resolve the differences.
What is a good enough system? Probably only one with electrostatic speakers. The best are made by Sound Lab. They are huge and they need a good, suitable room dedicated to their use. Most power amps can't do them justice so choice of amps is critical too. Benchmark AHB2 doesn't cut it; tried it already, the AHB2 now sits in a corner unused. Box speakers are never going to be as close to perfect for a true perfectionist.
The downside to getting the best or very close to it is high cost, especially for someone with little or no diy skills. I could go on, but will stop unless the OP wants to know more.
First of all, there is no such thing as a perfectly transparent dac. It doesn't matter how much one spends. Not Chord DAVE with prescaler, not Holo May, not the most exotic dCS, and not Prism. They all sound different if the rest of the reproduction system is good enough to audibly resolve the differences.
What is a good enough system? Probably only one with electrostatic speakers. The best are made by Sound Lab. They are huge and they need a good, suitable room dedicated to their use. Most power amps can't do them justice so choice of amps is critical too. Benchmark AHB2 doesn't cut it; tried it already, the AHB2 now sits in a corner unused. Box speakers are never going to be as close to perfect for a true perfectionist.
The downside to getting the best or very close to it is high cost, especially for someone with little or no diy skills. I could go on, but will stop unless the OP wants to know more.
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I have heard some... and my view is that it can. All the expensive gear looks impressive but luckily, if one can snap out of being impressed by the bling, one can build an equal or better system for 1/10 of the cost of the "big bling"... but it takes some skills yes and if you dint have it - pay up.
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My idea behind using multiple “consumer grade” DAC’s is the quality of conversion and thus sound quality. I have to admit that I am less experienced with pro audio multi DAC quality. Maybe you can help me with this. The Metrum Adagio and Pavane are NOS DAC’s with a real high quality conversion for reasonable cost. My Magnepan 20.1 (modified) speakers are able to produce unveil small changes in my system. So having a software DSP including crossover and 3 Metrum DAC’s behind this should provide in a very good DA conversion. My amps are also very good although and able to drive the Magnepan very well.
What options are there regarding multichannel DAC’s (at least 3x stereo) that can compete with a DAC like the Metrum Adagio?
Are these pro audio multichannel DAC’s also made to achieve the best audiophile level of audio quality or are they more focust on very good audio quality, robustness, stability etc?
What options are there regarding multichannel DAC’s (at least 3x stereo) that can compete with a DAC like the Metrum Adagio?
Are these pro audio multichannel DAC’s also made to achieve the best audiophile level of audio quality or are they more focust on very good audio quality, robustness, stability etc?