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goldenpiggy

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  1. First, many thanks to @gmgraves for this fantastic article. Due to starting/running my own business, I've been inactive on Computer Audiophile for many years (that's how long it's been -- missed the name change) and not had time to listen to music...until now. I could not be happier rediscovering my SACD collection, no less since I never got into ripping them. So thank you! I bought the same HDMI-> I2S converter box (with the green LED) and a Topping D90 MQA (hopefully Denefrips or Holo someday.) I believe the box seems to use the PS Audio I2S pinout on the HDMI connector, and that's the default setting on the D90. It's plug-and-play with the D90 and a stock Oppo 205 on the audio HDMI output of the player. Just use the default settings on the D90's I2S settings. (I didn't check if L/R are reversed though.) No need for external 5V power supply. In fact, it did not work if I connect an the 5V power supply. It's also plug and play on my ancient Sony BDP-S790 Blu-ray player, which has dual HDMI. I did need to go into the menu of the Sony to set audio out on the 2nd HDMI. No need for external 5V power supply. It did not work at all on my Oppo BDP-83 (original, non-SE, but Exemplar-modded) which has a single HDMI. If I connected the loop out HDMI to a TV, then the box powered up. If I connected an external power supply, the box powered up. But no matter what the settings on the Oppo and D90, nothing. I noticed on the TV that it would say PCM when playing SACD and sound was coming out of the TV. This leads me to believe the BDP-83 will modify the audio format over HDMI based on the TV's EDID. I didn't try another TV. I have an Esoteric UX-3SE universal player that has HDMI out, but not able to try it since drawer won't open. I could not tell much difference between the Oppo 205 vs. Sony BDP-S790 as a transport, so I was a bit disappointed in that regard. The Oppo is slightly faster in loading and advancing tracks, but not by much. There is a loud pop heard through the speakers when the DSD stream stops (when ejecting disc.) It was worse on the Sony than the Oppo. Sonically, the D90 is fantastic for what it costs when used in a HDMI->I2S setup. It is super clean, has amazing bass extension and control, fast, coherent, rhythmic, and has convincing highs. Amazing details that I never heard before, not even out of my 205. Alas it is not a warm sounding DAC by any means, nor does it sound "holographic," but neither does the Oppo 205. The D90 sound can be described as the sound out of mastering DACs, like Benchmark, Lavry, or Prism Sound (but Prism is warmer.) I previously owned an Accuphase DP-700 SACD player which I believe used 4x AD1955 delta-sigma DAC chips per channel. The Accuphase, if I can "recall" what it sounded like, was much warmer and more holographic, like a proper high-end piece. Compared to my Esoteric UX-3SE (I think it used top-spec BB PCM1704-K chips), I do feel the D90 via I2S and stock Oppo 205 best it in every way, maybe except for warmth. I do like th I2S sound of out the D90 better than the internal Oppo 205 when playing SACD. It seems smoother, more natural to me. Both slam hard and have great bass extension and control, with a nod to the D90's AK4499EQ. However, I like the male vocals on the 205's ESS9038Pro better. There's just a bit too much abrasive glare on male vocals with the D90 in my setup. (FWIW, I'm in the pro AV business and take advantage pro audio gear value proposition and being able to buy some gears at dealer cost. My preamp is a retired-from-show-use APB Dynasonics Pro Rack House mixer --analog warmth and full of body. Amp is a LEA Professional Connect 352 class-D commercial install amp, one of the best kept secrets in home hi-fi class D amplification. (Disclosure: I'm an LEA dealer.) Speakers are 12 year old Gallo Reference 3.5. Sub is who knows how old Hsu Research VTF-3 HO. Interconnects are all standard pro-grade mic cables like Pro Co.) For those that like the convenience of popping in physical SACDs but don't have a "high end" SACD player, this HDMI -> I2S converter that @gmgraves reviewed, along with any one of the inexpensive DSD-capable AK4499/ES9038/R2R DACs with I2S (Audio-GD, Gustard, Topping, Denefrips, Holo, etc.) can take your system to another level for the cost of snake-oil power cords.
  2. The PS Audio DLIII is a great sounding DAC, but it's USB implementation is the standard, non-async one which limits you to 24bit/48Khz. It does have balanced outs though.
  3. S/PDIF impedance is specified as 75 Ohm. BNC is better than RCA because BNC connector is a controlled impedance connector -- it's made to match the 50 or 75 Ohm impedance of coax cables such as RG6U and RG59. You would get the 75 Ohm version of the connector for audio. RCA connector's impedance can be all over the map depending on the design and material used. Other than that, BNC is a locking connector so you know when it mates perfectly.
  4. I have the same problem in my shallow but wide living room. I brought one speaker forward and adjusted the balance on the preamp slightly. Seems to work OK. The more dispersive the speakers, the less you'll need symmetry and equidistant between the two sides.
  5. If you go AES/EBU, realize that you don't need boutique cables or even dedicated AES/EBU cables. Good old cat5e works great and is what's commonly used in pro audio applications -- its impedance is within that of the 110 Ohm +/- 20% spec'ed for AES/EBU. Save yourself money and get XLR to RJ45 baluns instead of AES/EBU cables with XLR connectors. 100' AES/EBU cables will cost some pretty pennies. Cheers, JR
  6. From my personal experience, a carpet-covered slab flooring (such as yours) should not exhibit much in terms of room coupling and inciting resonance. That 4000psi concrete floor is solid! So I would be surprised if you hear much difference putting the speakers on spikes. The effect most prominent may be from elevating/angling of the drivers when you use spikes. I've had problems with older crawl space and basement homes that use floor joists made of 2x4 trusses and solid 2x8 or 2x10. The flooring flexes easily and bass can become boomy. My last house had laminate flooring over TruJoist I-beams and had no issue. I would worry more about speakers close to walls and windows. Wall coupling can alter your low freq response tremendously, especially if you have ported/vented speakers. In live sound, I take advantage of corner and floor coupling to get more slam out of subs, at expense of accuracy, of course. If you are really serious about this, try to borrow an RTA or rent a speaker processor such as Driverack 260 from a PA rental place (maybe around $25 a day). You'll need the RTA mic with also. Generate some pink noise and see how the spikes effect the freq response. Granted it's only 31-band but you could might be able to see some peaks due to resonance. FWIW, I removed the soft gel bottom from my Gallo speakers and now the cast aluminum base rests on laminate flooring. Bass output improved and it hasn't gotten worse in terms of boomy or muddy sound. Cheers, JR
  7. Please check the front input trims/levels ...Julf is right that there should be no need for an attenuator. Are you sure you don't have the preamp gain turned up on the Emu? Is there a software-controlled switch for -20dB pad for line-level inputs? Not sure what kind of attenuator you're using, but beware that you could end up with channel-to-channel imbalance. Two attenuators (one for left, one for right) will not have same component tolerance. So you might have to check the input meter and adjust the gain on one channel.
  8. FWIW There is a technique to create frequency-accurate, low distortion waveforms using DSP called DDS -- direct digital synthesis. Instead of a PLL or VCO, it uses.. guess what, a D/A converter. One normally finds this on arbs and function generators. I believe modern mobile phones use this method for their RF modems. I did a search and lone and behold an AC power supply that utilize DDS does exist. Here's one... http://www.behlman.com/p1352.htm I have no idea how good or the cost, but 1% THD and 0.1% regulation is pretty darn good, approaching linear power supply territory. 10A output might not be enough for many amps though. Regards, JR
  9. Absolutely. All professional equipment that accepts balanced inputs will accept unbalanced inputs. The better ones usually convert unbalanced into balanced anyway. You don't really get the benefit of balanced signal when you run unbalanced source by connecting the RCA connector's signal pin to both the hot and cold of the balanced connector (be it TRS or XLR). The reason is the output impedance at the RCA end is high, and you get impedance mismatch. Besides the output of the turntable is usually very low level (be it MC or MM) and would need a phono preamp to suitably drive balanced inputs (which operate at +4dBU). There is not enough "juice" to drive balanced inputs to get the increased CMRR benefits. The whole point of balanced signaling is that the shield is not carrying current. If you don't transformer-isolate the signal, at the source end the return is still the shield, and the whole darn thing is your AC outlet ground. Here is good reading about the balanced/unbalanced connection. http://www.jensen-transformers.com/an/an003.pdf I don't think the OP needs to worry about converting unbalanced RCA to balanced. As long as the distance between the components is 10ft or less and he is in a typical home environment (versus a live venue with tons of noise), he should not experience noise. Just get a good RCA to 1/4" TS mono cable. Avoid at all costs those cheap HOSA cables (and the cheapest LiveWires made by HOSA). They have poor shielding. Bigger problem may be 60Hz (or 50Hz depending on your locality) buzz. Cheers, JR
  10. Fear not...almost every professional audio interface I know of and have used accepts unbalanced inputs. All RME interfaces accept unbalanced connection no problemo. It doesn't matter what the physical connector is -- unbalanced is unbalanced. RCA is unbalanced. You should get an RCA male to -1/4" mono (TS or tip-sleeve) male cable. You DO NOT need to get an RCA to 1/4" stereo (tip-ring-sleeve - TRS) or XLR cable -- they provide NO added benefit and just costs more. It will not "balance" your unbalanced signal. To truly convert an unbalanced signal into a balanced one, you need a transformer, not just changing the connector. If you must convert to balanced, get something like a Whirlwind Director DI box, which has very good Jensen transformer that won't color the sound much if at all. (This box has 1/4" unbalanced input and XLR balanced outs, so you still need RCA-1/4" TS cable.) Cheers, JR
  11. I've been using the 3.5 for almost a year now. These replaced the excellent NHT Classic 4's. Unfortunately I can't comment about using with Peachtree. These speakers are not the warmest sounding , but neither are they bright. They excel at giving a very "live" presentation where the speakers seem to disappear from the room. There is no sweet spot no matter where you sit in the room, even in front of one speaker. The soundstage is quite deep and wide. At times it sounded like there was extra reverb added to the music. These are no doubt due to the 300 degree cylindrical tweeter. Be forewarned that the Gallos require a large, wide room to sound good with plenty of distance between walls. Also figure on at least 20ft to the listening position. Clarity is another virtue...it is a transparent and very revealing. When coupled to a quiet preamp and good DAC (I'm using Metrum Octave), you hear things buried deep in the recording for the first time. There is almost no coloration and no dip between drivers -- there is no crossover whatsoever between midrange and tweeter. You do hear this because music is pure and timing is great. There is a single order LPF to the woofer, but that's it. If you ever wanted a crossover-less speakers that sound great across entire spectrum, these fit the bill. Transient is another good trait. Pianos sound true and realistic. At first I felt treble was a bit subdued, but as I got used to it, it was "just right." Decays of things like hi hats and cymbals are natural and excellent. These speakers are not the warmest sounding, but neither are they bright. Midrange is good and intimate, but doesn't quite have the "she's standing right in front of me" presence for vocal performance. Violin and strings sound real good, if you're into classical music. Bass is taut and does go down surprisingly deep for the size, but you're not going to rock the house foundation with these. You can say the bass is "just enough." They are fast and dynamic, but not the most efficient, so require amps with high current delivery. Use a wimpy amp and bass really suffers. I drive each with a class A amp in bridge mode, about 180W/ch of very high current. Design-wise I think it is pretty cool looking, but you have to see them for yourself if they fit in with the decor -- they are not normal looking speakers. They are very small floor standers in terms of footprint and height. I have laminate flooring and two small kids that like to climb and ride them like a horse, so I removed the antiresonant gel bottom. IMO I don't think you can go wrong with these. Truly excellent sounding speakers for the money. I would not hesitate to buy another pair (although their new Classico floor standers could give the 3.5s a run for the money.) Good luck, JR
  12. Ignoring the fact that Apple products are mass-marketed consumer electronics and high-end audio is targeted, specialized niche products, I would vote for the Japanese company Accuphase on these grounds: 1) All their products are expensive, like Apple 2) All their products have superb fit and finish and look great. No cheap flexing plastic. 3) All their products work the way they should and are very reliable. Good user experience. Great sound. 4) State of the art (in-house SACD transport, AAVA volume control, gold plated teflon PCB, etc.) 5) High resale value Cheers, JR
  13. I believe SRC means reclocking data, and if you have a very stable master clock, it is in effect a de-jitter. Also, when one goes from 16 bit to 24 bit, this tends to lessen or redistribute quantization error. These should have audible effects. Those of us who were into sampling from the stone age (remember the 8-bit Ensoniq Mirage or original Emu Emax?) can attest to what quantization error does to sound quality. Going from 8 to 12 then to 16 bit sampling tremendously improved sound quality. So you can imagine from 16 to 24 bits. Regards, JR
  14. Protocol-wise, sample rates above 192Khz is not a problem with AES/EBU -- it can theoretically support any sample rate as long as clock can be recovered from the data. Bandwidth is usually not an issue as it is common to run AES/EBU over good old 75 Ohm RG6U coax or cat 5e (350MHz) cables. The current spec allows for maximum of 24 bit audio data though. You do have some issues: the more common 110 Ohm characteristic impedance allows a pretty loose +/-20% tolerance. The common XLR connector is not impedance matched. At faster rates, impedance matching between transmitter and receiver is a must. To keep the eye opening bigger and jitter low, faster rise time is needed, so one must also consider EMC/radiated emissions at higher rates given the biphase signaling (not very efficient and quite a noise polluter.) Commercial audio equipment has to still pass FCC class A certification. AES/EBU is almost always transformer-coupled, so at higher rates it will require better and more expensive isolation. In the pro audio world where AES/EBU is prevalent, 96KHz/24bit is common these days, although you'll often see 192KHz in the studio environment. I don't think I've ever seen anything higher though. Probably not a whole lot of equipment that handles LPCM at greater than 192KHz sample rate anyway. AES/EBU was designed to handle long distance transmission in less than ideal environment, rather than sheer transmission speed. With an active equalizer (distribution amp), you can easily go 2-3 football fields over cat 5.
  15. Generally oversampling takes place before the DAC, by an oversampling digital interpolating filter chip. 8x is a common ratio. The incoming 44.1KHz samples get zero-padded samples placed in between adjacent original samples (they are literally samples with LPCM data set to 0). Some kind of interpolation is done to those added samples to give them values. The additional data/higher sample rate is required to minimize artifacts generated during digital filtering (namely the lobes or ringing caused by an FIR filter -- from the sinc function.) Without additional data, one would hear what's known as pre-echoes or transient smears. It is the cost of using the sinc() function which is the ideal brick wall filter. As a side, these ringings can cause amplitude to go above 0dB, which is of course clipping in digital. Hence the reason you use higher-bit DACs with oversampling (e.g. 18 or 20 bit DACs even though original data is only 16 bits.) Of course these days using higher bit DACs has additional benefits such as lowering noise floor by dithering. Oversampling has the effect of moving aliases or higher harmonic images of the samples to higher freq band, well outside of audio band. With most of the high freq aliasing noise gone or way above audible freq, it is then as you said: the analog reconstruction filter can then have a gentler slope, minimizing phase shifts. It's also cheaper (less components) If the original LPCM data is of a higher sampling rate to begin with (e.g. 176.4KHz), then you don't need as many zero padded samples and interpolation for the digital filtering process. This should sound better than 44.1KHz. I'm not sure if in oversampling schemes the filter chip performs decimation (data reduction) before feeding the DAC chip. I believe up-sampling can be done the same way as oversampling (using an oversampling digital interpolating filter chip) or using an asynchronous sample rate converter chip. Difference is upsampling goal is to get the 44.1KHz up to 192Khz (or maybe 384KHz) and feed the higher sample rate directly to the DAC. Cheers, JR
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