Tom: I couldn't agree with you more on the efficacy of DIN audio signal and speaker connectors. Decades ago I owned a complete QUAD system (33/FM3/303/ESL57) . While they at least used 4mm banana jacks on the amp output and speakers, they used the 5 pin DIN for signals - very efficient use of real-estate on compact equipment like the 33 pre-amp is about the nicest thing I can say about them. The stock cords were notorious for poor strain relief, which meant rewiring. I almost always just cut the end off and started over with a fresh plug, rather than try to unsolder / resolder .
Henry: The amp shown is for LXmini with is an active bi-amped speaker system for which the miniDSP provides the XO, as well as capacity for PEQ and perhaps even Room Correction with something like REW. For those with a bit more coin, there's also the DDRC-24 which adds DIRAC Live to the mix.
Henry: The amp shown is for LXmini with is an active bi-amped speaker system for which the miniDSP provides the XO, as well as capacity for PEQ and perhaps even Room Correction with something like REW. For those with a bit more coin, there's also the DDRC-24 which adds DIRAC Live to the mix.
It is with some trepidation that I post this as these discussions tend to get heated. Should you find yourself getting spooled up about this, I suggest taking a deep breath and contributing to one of the threads on this topic in the Lounge section.
The Sonos has analog I/O and digital out. You could run your own DAC with it if you aren't happy with the built-in DAC.
Back in the 1940ies, Shannon and Nyquist proved mathematically that a sine wave can be reconstructed from samples without loss of fidelity as long as the sampling frequency is at least twice the signal frequency. The Sonos supports 48 kHz sampling, hence, is theoretically capable of reproducing a 24 kHz sine wave. Allowing for the anti-aliasing filters, the practical limit is approx. 22 kHz.
You can find a bunch of audio myths debunked here: Testing audiophile claims and myths
I am not trying to start a shouting match. I'm just making you aware that you may be making your decision based on marketing rather than science.
Tom
The Sonos has analog I/O and digital out. You could run your own DAC with it if you aren't happy with the built-in DAC.
Back in the 1940ies, Shannon and Nyquist proved mathematically that a sine wave can be reconstructed from samples without loss of fidelity as long as the sampling frequency is at least twice the signal frequency. The Sonos supports 48 kHz sampling, hence, is theoretically capable of reproducing a 24 kHz sine wave. Allowing for the anti-aliasing filters, the practical limit is approx. 22 kHz.
You can find a bunch of audio myths debunked here: Testing audiophile claims and myths
I am not trying to start a shouting match. I'm just making you aware that you may be making your decision based on marketing rather than science.
Tom
Tom's second anniversary message reminds me that I really ought to get my amps completely before I have had the boards 2 years. I have decided to go 5 pin DIN and Speakon for my connectivity for 2 reasons
1. have them in my box o bits
2. Will offend someone's audio sensibilities, which is good
Of course some always have to go one better https://www.naimaudio.com/product/range/super-lumina. Julian must be turning in his grave.
1. have them in my box o bits
2. Will offend someone's audio sensibilities, which is good
Of course some always have to go one better https://www.naimaudio.com/product/range/super-lumina. Julian must be turning in his grave.
I had read the about the relation of sampling frequency with signal frequency. It made me think all the Hi-Rez stuff is snake oil.
I was just curious about the Sonos and apparently got the wrong impression that it required a DSP. The only thing I want to learn is the bit rate of the Sonos DAC. I'll check out the Sonos site. I know the Node 2 has a 32 bit DAC and I've read most good DACs, like the Sabre, are 32 bit. Don't know if that's also snake oil.
I'm not planning to buy either. Just curious about all this, new to me, technology. I'm old school. Never had a separate DAC and only started streaming movies via Wi Fi about a year ago. I even think my Meridian 506.20 is still a very good sounding CDP.
I was just curious about the Sonos and apparently got the wrong impression that it required a DSP. The only thing I want to learn is the bit rate of the Sonos DAC. I'll check out the Sonos site. I know the Node 2 has a 32 bit DAC and I've read most good DACs, like the Sabre, are 32 bit. Don't know if that's also snake oil.
I'm not planning to buy either. Just curious about all this, new to me, technology. I'm old school. Never had a separate DAC and only started streaming movies via Wi Fi about a year ago. I even think my Meridian 506.20 is still a very good sounding CDP.
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Poseidon's Voice Modulus 86
Hi fellas,
To commemorate the second anniversary and inception of the Tom Christiansen's excellent Modulus 86, here is my build.
Poseidons Voice Modulus 86 Slideshow by nycavsr2000 | Photobucket
The speed of the slideshow link above can be modified by scrolling your cursor over the bottom of your screen.
A few points:
1. Although there are 2 power transformers, there is only 1 mains fuse! Rest assured, another 'in line' fuse will be added in the future.
2. The cotton sleeving that covers the input and output signal wiring has been fire treated.
3. On the main Modulus 86 board, you will see that there are Vishay resistors. I purchased these as I have a colleague in the industry who furnished them to me at an excellent "low" price along with 0.1% matching. I believe they are unnecessary. Just go with Tom's BOM recommendations and you will achieve excellent performance. The only way to know if my build objectively makes any improvement whatsoever is to measure my board with Tom's expertise and equipment.
4. The Cardas XLR's, output connectors, Furutech IEC inlet, etc...are there solely for looks. I do not believe they make one iota of a sonic difference.
5. The use of (2) 200VA Antek electrostatically shielded toroids along with steel covers are totally unnecessary in my application. The original plan was to build a bridged set of Modulus 86 R1.0's, but I sold the R1.0's and moved on to R2.10. In hindsight, I never would need the output power levels of a bridged Modulus 86 design. If I were to design the chassis today, I would build it with Tom's new SMPS-86 supply given that my main speakers are 96 dB efficient.
6. The design of the front panel is still under consideration.
A full objective and subjective review will be posted in the future comparing the Modulus 86 with my NCore 400 mono blocks which Tom measured as well. The review will include both vinyl and digital sources. I will post a link when it is complete.
Thanks again Tom for your brilliance!
Best,
Anand.
Hi fellas,
To commemorate the second anniversary and inception of the Tom Christiansen's excellent Modulus 86, here is my build.
Poseidons Voice Modulus 86 Slideshow by nycavsr2000 | Photobucket
The speed of the slideshow link above can be modified by scrolling your cursor over the bottom of your screen.
A few points:
1. Although there are 2 power transformers, there is only 1 mains fuse! Rest assured, another 'in line' fuse will be added in the future.
2. The cotton sleeving that covers the input and output signal wiring has been fire treated.
3. On the main Modulus 86 board, you will see that there are Vishay resistors. I purchased these as I have a colleague in the industry who furnished them to me at an excellent "low" price along with 0.1% matching. I believe they are unnecessary. Just go with Tom's BOM recommendations and you will achieve excellent performance. The only way to know if my build objectively makes any improvement whatsoever is to measure my board with Tom's expertise and equipment.
4. The Cardas XLR's, output connectors, Furutech IEC inlet, etc...are there solely for looks. I do not believe they make one iota of a sonic difference.
5. The use of (2) 200VA Antek electrostatically shielded toroids along with steel covers are totally unnecessary in my application. The original plan was to build a bridged set of Modulus 86 R1.0's, but I sold the R1.0's and moved on to R2.10. In hindsight, I never would need the output power levels of a bridged Modulus 86 design. If I were to design the chassis today, I would build it with Tom's new SMPS-86 supply given that my main speakers are 96 dB efficient.
6. The design of the front panel is still under consideration.
A full objective and subjective review will be posted in the future comparing the Modulus 86 with my NCore 400 mono blocks which Tom measured as well. The review will include both vinyl and digital sources. I will post a link when it is complete.
Thanks again Tom for your brilliance!
Best,
Anand.
True that!
You can probably ask Sonos that directly. Their "chat with us" box pops up pretty quickly on their website.
16 vs 24 bits is not snake oil. You can actually pick up the difference in the time domain with an audio analyzer. See any of Stereophile's measurements of a DAC. They usually show a low-level signal (say 1 mV) at 16 bit vs 24 bit. The difference is obvious.
Is it audible? That's a good question.
24 vs 32 bits is mostly marketing. You can get a good sense of how much marketing by dividing the peak-to-peak output voltage by 2^N, where N is the number of bits. You'll quickly find that you're dealing with nV or pV per bit.
One exception to this is DSPs, ASRCs, etc. where math gets involved. in that case the added bit depth of 32, 64, 80, etc. bits is more likely to result in measurable differences.
An alternative to the various streaming players is the Apple Airport Express. It has TOSLINK optical out and you can stream directly to it from iTunes. There are other players available as well, such as Audirvana. The Airport Express is about $100 as far as I recall.
Tom
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Oh, very cool!! Thank you very much for posting. That is an incredibly good looking build. I agree with your observations on the "bling factor". Then again, should you decide to sell the amp at some point, it is guaranteed to have a high resale value.
I hope you don't mind me featuring your slide show on my website.
I rather like the simplicity of the current one.
Thanks again Tom for your brilliance!
You're very welcome.
Tom
Without intending to flame anyone or start a forest fire, I mention two things:
1. Sampling frequency is not the only criterion with respect to "high resolution" audio formats. The other is bit depth. Having personally recorded at home at 16 bits (effectively 14 because of the poor quality of the equipment), and then with true 16 and then 24 bits, the greater bit depth allows much more information to be recorded. Tom's discussion about crest factor provides an insight into this issue of number of bits actually available to encode the signal. If you assume that some margin is needed to ensure that you do not have a "digital over," then the full 16 bits are not being used to encode the signal. Since audiophile music is not heavily compressed, it means that most of the music is being sampled/played back with much less bits - or much less different levels available. This results in less resolution for regular music. You can verify this for yourself on CDs that are mastered at a low level, such that you have to turn up the volume. The lack of resolution should be pretty clear - esp in quiet sections and reverb tails (which fall off fast). My favorite CDs to demonstrate this are Oregon's first two CDs. They were recorded in NYC at 15 ips two track (mixing in the board to two channels and then directly to tape. The viny CDs are good. The Mobile Fidelity vinyls were incredible. The Vanguard CDs issued in the 1990s - a travesty in part because they are mastered at a low volume. And do not think I am talking about compression here - the CDs still have range, but now the softest parts are way, way down and the lack of resolution is pretty bad. Had Vanguard/Welk Group mastered them so that the MOL was near the top of the CD 16 bit range- like the Mercury Living Presence CDs done by Wilma Covert and bob Fine, they would have been incredible.
2. The Nyquist theory assumes perfect filtering upon decoding, and does not take into account the need for Red Book CD players to include steep anti-aliasing filters with "claimed" audible effects into the audio passband. Increasing the sample rate allows the filters to be moved up so that there is no impact on the passband. In sum, within the range of audio signals, Nyquist theory works and I do not question it. It is the imperfections of implementation that drive higher sampling rates. Yes, oversampling is a way to get around this, but think about it - oversampling is creating fake samples at a higher frequency. Needed if you are limited by the Red Book and existing CD players. Not an limitation if you are talking about downloadable audio files, or DVD Audio/SACD, so doesn't it make more sense to simply record "real data?" if the hardware allows this to be done accurately? Having said that, do I think that 192kHz is needed? No way. 96 kHz is more than enough.
YMMV Please accept my apologies in advance if anyone is offended.
I agree there are audible differences between 16 and 24 bit. I think there are also differences between 16 and 20 bit. I had my Meridian 506 upgraded, years ago to the highest level then available for it, to 20 bit and there was an audible improvement. I must say that I think my OPPO with the 32 bit DACs sounds a tad better.
I must also say that, since I started listening to the combination of Meridian 506.20 to Aragon 18K MkII to Mod86 to custom outboard xovers to tweaked Sound Dynamics 300Ti speakers, I haven't turned on the system with the OPPO source.
That system is where the Parallel86 is going. It will be OPPO to Audible Illusions M3A to Parallel86 to custom outboard xovers to B&W N803s. This system also has the Dual TT and Magnum Dynlab FT101a FM tuner and I've missed listening to both of those sources
I must also say that, since I started listening to the combination of Meridian 506.20 to Aragon 18K MkII to Mod86 to custom outboard xovers to tweaked Sound Dynamics 300Ti speakers, I haven't turned on the system with the OPPO source.
That system is where the Parallel86 is going. It will be OPPO to Audible Illusions M3A to Parallel86 to custom outboard xovers to B&W N803s. This system also has the Dual TT and Magnum Dynlab FT101a FM tuner and I've missed listening to both of those sources
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Just a thought for other builders. I built the Mod86 and am very pleased with the construction results...like screws, washers nuts, wiring, ect.. Now that I'm about to build a Parallel86, I really wish I had taken pics of my mod86 build process, with components insitu, and kept records of were I bought the stuff. Guess I'll take the cover off the Mod86 for a refresher. You all probably already do that. I wish I had thought to do it.
I've posted about building an A B speaker selector switch and included some sketches. I was going to put the switch in a small metal enclosure. The enclosure would require input and output speakon connectors. The Par86 and switch box would require a total of 8 connectors.
I'm thinking of putting the switch in the Par86 chassis with a toggle on the front or rear panel. I think that would require only 4 speakon connectors, reduce the number of solder joints and eliminate another small box to deal with. Am I missing something?
I'm thinking of putting the switch in the Par86 chassis with a toggle on the front or rear panel. I think that would require only 4 speakon connectors, reduce the number of solder joints and eliminate another small box to deal with. Am I missing something?
No argument there. All I'm really trying to convey is that I'm not aware of any scientific evidence that supports the claims of the HiRez crowd. I have heard anecdotal evidence that 96 kHz recordings are "better". Not because 96 kHz sampling is "better" but because the recording, mixing, and engineering is better.
There's quite a de-bunking of HiRez here: 24/192 Music Downloads are Very Silly Indeed
Warning: Scientific content.
Tom
tomchr;4818408 There's quite a de-bunking of HiRez here: [url=http://people.xiph.org/~xiphmont/demo/neil-young.html#toc_1bv2b said:
The site has a good discussion of sampling frequency, and I largely agree with it, esp. the aspect of 192 kHz.
However, the short discussion on bit depth at that site, while complete accurate and technically correct as an abstract matter, does not mean that 16 bit recording and playback is sufficient because the author does not reflect the realities of audio recording, production, and mastering. The author implicitly assumes that: (i) the recording is using the full bit 16 bit depth in the Red Book standard, and (ii) the CD is mastered to take advantage of the full 16 bit depth. As I mentioned in my post, at the recording end, headroom to remove digital overs end up forsaking some of the bits. However, competent engineers and a judicious use of limiter can avoid to much loss there. However, if you have a poor mastering job, then you have a further truncation of the bit levels available. This, in my experience is where the CD process is most vulnerable.
Finally - and something I now remember: having more bits makes it easier to use digital processing software with more significant digits. Digital processing of EQ, limiting, hiss and click reduction, and other special effects involves many mathematical operations and if the extra bits are thrown away, then you end up losing the theoretical resolution even if the recording uses the full 16 bits available. I've read several engineers (Bernie Grundman? Bob Ludwig?) talking about the early days of digital recording - and in particular the 3M digital system, and the very destructive audio processing done in the digital realm by the 3M digital processing mixer. It might take me some time to find these discussions, however. Of course, you can record in 16 bit, process in 24 or 36 bit, and then convert back down to 16 to permit Red Book CD mastering. I readily admit that if this is done, then one can be sure that the full capability of 16 bit audio quality as described at the site is not compromised by the processing step. It still leaves, however, the issue of poor mastering on the table, not to mention the assumption that the D/A converter and filters are also performing perfectly. This may not be realistic so, once again, more bits (20 or 24) again provides a margin to account for reality.
Nice build, how come you chose to spend all that money of naked foil resistors? Tom do you have any opinion on their use?
Like I said, they are not required at all. I was doing a bulk buy for some other projects and included the set I needed for the Mod86. I will admit, they are "pretty" looking! My build is an example of what can be done, but not what should be done.
Tom's opinion would be that they are not required at all. Just stick with the BOM!
Best,
Anand.