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Sam Lord

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  1. The best thing you can do in any system is put whatever speakers you buy into the best position in the room that you can visually and ergonomically tolerate. Position the tweeter as close as possible to ear height. Get them AWAY from corners. Start with them an even distance from your ears and pointed straight ahead. You might need to tilt them forward or backward, since many don't come with proper time alignment. You might want to toe them in none, a little, or some, but listen for what gives the best soundstage. PA speakers are usually a bad choice for a small room, but speaker positioning matters more than anything. If the speakers HAVE to flank the tv, put them flush with the tv if possible, nearly touching it. Put something soft along the left wall to reduce slap echo. It'll sound way better after these changes.
  2. Ballistic ventilation is a cultural practice in my region. One sees its salutary effects on signs, mailboxes, and fixed pieces of lawn art.
  3. The Asgard 3 has a stronger output than most "good fits" for the Senns, delivering power in fact between the latest versions of the Schiit Valhalla and Mjolnir into 300 Ohms. But tube voltage swings and compression behavior do make a dense, highlighted sound. I agree that the A3 is not the best fit, but should be a decent fit.
  4. First, be sure the speakers are time-aligned. Raise them to be vertical and aimed at the correct part of your head for correct alignment. The ideal, cheap method is large bricks or paving stones. Use a _very_ thin sticky layer to affix the speakers to the stones. If you can't or refuse to raise the speakers, tilt them until aligned correctly. Use live percussion for this (my favorite is single tom tom): when alignment is ideal the drums sound natural and image well. Be sure to get maximum absorption of the table relections, use whatever you can live with. Heavy and thick foam are best, given your placement on the table. Try starting with zero toe-in, then add toe-in as needed. In midfield and farfield setups zero is typically best. In most situations it's much superior to toed-in setups, especially the ubiquitous equilateral triangle arrangement. The imaging and soundstage should improve a _lot_. You can even put them flush with your monitor if it's a very wide one. Don't be too close to that back wall, but get something absorptive on the back wall behind your head to address those reflections, that will reduce the strong distortion from that reflection. Good luck!
  5. I suggest that you consider powered speakers and a separate headphone amp/preamp combination. I have Edifier 1280 series (current model is now R1280DB, includes a DAC) powered speakers and their quality is amazing for their very low cost: $130USD or perhaps 150GBP excl. VAT. Micca PB42x are also excellent but have fewer features. Then you could get a Schiit Magni 3+ or Magni 3 Heresy (available soon) for 110GBP incl. VAT, or an older, used HP amp for much less. All of the Magni 3 amps, older and newest (currently out of stock in the UK), have preamp outputs. Look here: Schiit UK Headphone Amps. Good luck!
  6. I think my setup is ideal for this, look at my profile photo to view it. I use extremely heavy and rigid stands, each with an ATC SL20 passive on the top shelf, a stereo amp for vertical biamping on the lower shelf, then a Velodyne DD12+ on the base of the stand. Every transducer is time-aligned by ear for my preferred listening position. (Write me for instructions.) Each channel was positioned by rule of fifths wrt room depth and width. (It's a narrow room.) For stands I have custom-built Sound Anchor ADJ2s, around $1.5k/pr and about 130 lbs each. Each shelf mounts with 2 x 3/8" bolts, monstrously rigid. I run my ATCs full range and supplement with the subs. I roll the subs in at 55Hz, slope around 18dB/oct IIRC.
  7. Well yeah, partly because those adapters aren't wired correctly! See Rane notes #110 (Diagram #17, IIRC) about grounding. All off-the shelf rca-xlr adapters can easily cause ground loops: the shields are attached at both the rca ring and xlr pin 1 and thus, usually, to chassis ground = bad. The shield should *float* on the rca end, whose ring should only connect to xlr pin 3 (inverting input). In the case where the rca ring *is* connected to earth internally on the source, then the loop can be fairly free of hum, but that means the source is a bad design anyway.
  8. Vibration control is simple to understand: get the motion of components to frequencies well below the audio band. The main goal is to minimize displacement of speakers from the motion of their transducers, and to minimize acceleration of all other electronics. Like anything, It can be difficult to implement to the nth degree. Energy problems are easy to solve, but momentum problems are harder. The kinetic energy of objects can be converted into heat. But their momentum, being a vector, can never be stopped, only slowed or summed. Treating momentum problems can require effort, space, and mass or mechanical devices to reduce resonant frequencies to be very subsonic. The development of high-quality bearings for speakers (like Aurios) have been a big advance: they sum the horizontal momentum vectors for a wonderful effect, and settle at an ultra-low frequency. For all the other components, which need relief from mostly-vertical displacement, mass loading with sorbothane does a great job. If you shell out for fancier mechanical devices you can do better still, but mounting pieces on heavy, rigid bases like paving stones and then placing carefully-chosen sorbothane feet beneath those bases can do wonders for a fair price.
  9. Glad to hear it! FWIW, I have a good DAC but am in a similar situation; I'd like to try a different DAC, my current is an Oppo UDP-205. OTOH, My hearing has been very damaged in the last 2-3 years. I still want to keep native DSD payback ability (the Oppo does DSD512) because I've had good experiences with SACDs done from tapes, so want to keep the DSD ability. I'm looking at something very good but not terribly costly, like a Holo Spring KTE. Side note: I wish the industry would finally settle on PCM because the majority of sound engineers who actually use hi-res well prefer it, and there is always an absolute level indication. DSD could be delivered with metadata to provide absolute level every few ms, but currently that's not part of the process--silly.
  10. Always list your other gear when asking questions like this. I found this: ----------------------- "My current computer setup- Mac Mini----JRiver Media Player-----ISO Regen/external power supply----Schiit Yiggy-----my only source is Flac files stored on an external Thunderbolt drive. The rest---Pass Labs-Martin Logan Montis's.....I'm actually thrilled with the current audio performance, but I'm thinking it can be even better. Which leads to my question.....Aurender N100H Network Music Streamer with USB Output and 4TB HD IF I wanted to invest another 3K...is the above a good option? If not what would you do with a 3K budget?" --------------------- So what isn't pleasing about your system? You have a nice vinyl rig, but understand that vinyl is _usually_ a very musical and forgiving source if it's well set up. But digital might need help in your system: have you tried tube gear? If you want some real magic, try a tube preamp and/or tube DAC like a Lampizator. I suggest you try out a Schiit Freya, the new "+" version and with the better Tung-Sol tubes (~$1k total), and you might find that a lot more stimulating. Tubes make a much bigger (pro or con) difference than a DAC change, typically. I wouldn't recommend a tube amp unless you really want much more of tubes than sources/preamps can deliver.
  11. 1. Yes: "Ragnarok 2 is modular—it has two input module spaces that can be filled with RCA inputs, or with an MM phono preamp and Multibit DAC module." 2. It's a digital input for their MB DAC card, which only takes USB. 3. There should be reviews around for the MB DAC card, but I haven't seen any. Currawong tested a Lyr 3 but without the card on Youtube.
  12. These look very exciting, especially the $899 Freya 2. The new line has many enhancements. The new solid-state versions should be very good too, but good tube gear at these prices is nearly unobtanium. And they provide very thorough and impressive tests on the premier AP measurement unit: Schiit Freya 2 Preamplifier
  13. There are several approaches, but I encourage you to listen to Shunyata Research gear. You can try them from the Cable Company for a substantial fee. They are unique in using environmental noise reduction. Those parts use iron in silica pellets ("FeSi"), converting nearly all RF noise into heat, unlike a monolithic ferrite ring which is quite reactive.
  14. Since all of you good folks are captive now, waiting for Bruno's response, I'll spew some debris about my experience with class D, in fact an older and lesser variety: 1st gen Icepower in Bel Canto amps. 1) Class D has very low output impedance. If you want to hear them at their best, USE IT! Don't use a stereo amp, use monoblocks with very short speaker cables. Or do what I do, vertically biamp with a stereo amp on each side. This prevents cabling from halving your amp's output impedance, or worse! 2) As above, use that low output Z to maximum benefit: bi-amp if possible, or at least bi-wire, because the difference between 2ft and 6ft of shared speaker cable makes a measurable and audible difference in performance. Separate circuits make an even bigger jump. Bruno emphasizes biwiring for the same reasons, but any IMD between transducers is eliminated with multi-amping. 3) Class D circuits, moreso than Class AB, ramp up in distortion in an abrupt way, usually around 1/2 of the power at clipping (1% THD). If you listen at very high levels, as I do (I have very inefficient ATC monitors), Get at least 2x the amp power you expect to need. Then your loudest passages will truly be pristine.
  15. IIRC the phase shift at 20kHz for nCore was not consequential...below 10 degrees. Bruno is very conscious of phase shift in both amps and speakers, I spoke with him at length about this (many years ago). No Class AB amplifier can be modeled analytically: all have numerous discontinuous behaviors. Crossover notch distortion is often the largest but is far from the only one. Transistor behavior itself, even within an fairly linear range, is discontinuous. I can't find any metric which supports that view, when comparing measured performance of Bruno's Class D designs and any Class AB designs. Note that The Benchmark AB1 is a Class H design, not Class AB. Bruno was not only the principal designer at Hypex, but the reason the company was created. No, UcD, no Hypex. Interesting! Monolithic ceramic caps with the C0G/NP0 rating are superb in both coupling and bypass duties. Also X7Rs are great in spite of microphony: properly potted and/or paralleled they have no faults. However, many have magnetic leads which are not known to their own manufacturers, whose names I won't state here (Kemet). I think that was a very old statement from him. Anyway, one explanation is that damping factor of his Class D circuit might have been much higher. Bruno has stated that his nCore circuits have *very* low HF components either radiated *or* transmitted onto the AC line. His extremely adept use of differential signals in both the SMPS and amp module are the reason. But still, any garbage on the line is bad and minimizing it is important. Recently Bruno demonstrated with Shunyata, with whom I worked under contract from my old company for a couple years (no financial ties now), and their non-reactive, environmental HF reduction is a great antidote to line- and component- generated noise. Really good point Shadders! Many of the measurement improvements disappear when one adds another stage to replace the lost gain! Are the Purifi modules really an improvement over nCore? Another 13dB of gain would knock S/N down by about 4-5dB, right where the NC500 measured. However, the concept of lowering amp gain makes sense, given the excessive output of many high-end DACs. On the third hand, 13 dB is too low. On the fourth hand, you'll *never even nearly* clip the amp when your DAC sends a maximum signal. But fifthly, I'm guessing that an opamp *is* meant to be added to the front of this circuit.
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