[SlimDevices: Audiophiles] Re: Best Hard Drive For Music Quality?
snarlydwarf Wrote: The catch is that there is no clock in that data. There isn't a seperate line saying okay, here comes the next bit! which would get rid of the jitter argument completely. And because even the crystals that control timing are not -exactly- accurate, you can see errors where it tries to derive a clock. If the sender is sending bits at 44,100 bits per second, and the receiver's clock is just a slight bit off at 44,098 bits per second, it is going to misread some.This isn't quite correct. The receiver does not generate its own clock from an oscillator. It extracts the clock from the SPDIF signal coming from the transport. What you have described is data errors. Jitter is not data errors. Jitter is variations in the timing between adjacent clock pulses. Sometimes jitter can be so bad that it causes data errors, but that's very unusual and is not what most people are concerned about in digital-to-analog converters. The problem that jitter creates in a DAC is additional noise at the analog output. A digital sample is meant to represent Voltage = V(n) at time = T(n) where n is the number of the sample within the sequence. Generally the samples are equally spaced in time, so that T(101) - T(100) is the same as T(1001) - T(1000). The samples are sent to the DAC in sequence at a constant rate, along with a clock signal with one upward pulse per sample. Like snarlydwarf described for the hard drive, the data for V(100) is sent slightly before the 100th clock pulse to make sure it reaches the DAC before the clock pulse. When the clock pulse arrives, BAM. The DAC reads the data on its input pins and immediately changes its output level to the voltage that matches V(100). It holds that value while the 101st data sample arrives, then switches the output to V(101) when the next (101st) clock pulse arrives. Then the output is filtered to connect the dots of the stair-step output voltage and make a smooth curve that matches the original wave that was captured by the analog-to-digital converter in the studio. That's great, and if the clock pulses really are equally spaced then you should perfectly reproduce the original wave (assuming the analog to digital converter also works perfectly). But what happens if there is jitter in the clock -- meaning that there is some variation in the time between clock pulses? Imagine the DAC is reproducing a signal that is rising an equal amount from one sample to the next. With a regularly spaced clock, you can draw a straight line through the corners of the stair-stepped output. However, what if one of the clock pulses arrives early? The DAC with switch to the next voltage level early, so one of the step corners will be shifted over to the left of the straight line. After the filter smooths out the corners, that section of the output signal will be slightly higher than it should be. So, for a rising signal, an early clock pulse results in output that is too high. A late clock pulse results in output that is too low. For a falling signal, the results are just the opposite. The end result is that noise in the timing of the clock pulses gets translated by the DAC into voltage noise in the output signal. Why is it especially a problem with SPDIF? well, the more noise that you pick up in the clock signal, the more jitter will result. Ideally, you'd like to generate the clock right next to the DAC so that you've got the least possible chance to pick up additional jitter. Then you can send the clock signal all the way back to the transport to tell it how fast to send out the data. The transport doesn't care about jitter, it's just moving bits from one place to another. The exact timing doesn't matter except at the DAC. But with SPDIF, the clock is supplied by the transport and sent over a coaxial cable along with the data, and then separated by the receiver and sent on to the DAC. There is plenty of opportunity to pick up jitter there. In an ideal world, the receiver would be able to filter the heck out of the clock signal it receives and restore it to exactly equally-spaced pulses. In the world of consumer audio, that doesn't seem to be the case, and jitter that comes from the transport or is induced by noise in the cable driver, the cable, or the receiver, arrives at the DAC and creates corresponding voltage noise in the output signal. I don't know whether the amount of jitter present in most systems is enough to produce measurable differences in the output voltage, or for that matter, discernable differences in the sound. However, what I've described above is the theory, and the mechanism by which jitter *can* cause problems in DAC (or ADC) circuits. -- John Stimson John Stimson's Profile: http://forums.slimdevices.com/member.php?userid=218 View this thread: http://forums.slimdevices.com/showthread.php?t=24616
[SlimDevices: Audiophiles] Re: Is SB2 (or 3) a source of RF interference?
My Squeezebox2 definitely interferes with my FM radio (around 92MHz). It doesn't seem to affect my record player (moving magnet type cartridge). -- John Stimson John Stimson's Profile: http://forums.slimdevices.com/member.php?userid=218 View this thread: http://forums.slimdevices.com/showthread.php?t=22274 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Anyone recommend decent speaker cable?
What I'm using is actually fire alarm cable. It's 14 or 16 gauge solid copper wire with Teflon insulation, in a twisted pair with polyethylene or Teflon outer sleeve. It was labeled as New York City fire alarm cable, but since then it seems that NYC has slackened their specifications. That cable isn't offered any more. The closest option right now is Belden 6220FK, which has a PVC outer sleeve and a foil shield. You could of course strip those off. -- John Stimson John Stimson's Profile: http://forums.slimdevices.com/member.php?userid=218 View this thread: http://forums.slimdevices.com/showthread.php?t=21750 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: SPDIF question - error correction and interconnect quality
Andrew L. Weekes Wrote: It represents an impedance discontinuty that will give rise to reflections that are serious enough to impact the performance of SPDIF. It's specific location is largely irrelevant here - a mismatch is a mismatch is a mismatch.The closer the impedance anomaly is to the transmitter or receiver, the closer the reflection is to the incident edge. In a 100ns pulse, a reflection at a 1ns delay may not even be picked up, but a reflection at a 50ns delay would probably cause problems. -- John Stimson John Stimson's Profile: http://forums.slimdevices.com/member.php?userid=218 View this thread: http://forums.slimdevices.com/showthread.php?t=21415 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: SPDIF question - error correction and interconnect quality
Patrick Dixon Wrote: But the pulse width is not that important in an S/PDIF connection, it's the transitions that are. Reflections / impedance mismatches are rarely serious enough to cause actual data corruption (in S/PDIF), but they can affect the timing information contained in the rising and falling edges.If the reflection occurs close enough to the source or receiver, it can blend into the transition. Switching isn't instantaneous, and SPDIF isn't that high in frequency. The timing of a reflection can determine whether you see blurring of the edge/jitter in the clock, or visible artifacts/bit errors. -- John Stimson John Stimson's Profile: http://forums.slimdevices.com/member.php?userid=218 View this thread: http://forums.slimdevices.com/showthread.php?t=21415 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: SPDIF question - error correction and interconnect quality
dorkus Wrote: and btw, reclocking does not eliminate jitter, it just attenuates/shifts it.I think what you mean by reclocking here is what is commonly referred to as clock reconstitution, which is done by chips like the Crystal Semiconductor chip I referred to earlier. That takes the incoming clock signal and filters it to produce a clock signal with less jitter or with a different jitter spectrum. True re-clocking would involve storing the incoming data in a memory buffer, and then clocking the data out with a locally generated clock that isn't based on or otherwise influenced by the transport clock. If you do that, then there is no way for the jitter from the source to affect the timing of the data coming out of the buffer. You are in effect starting fresh, with only the jitter present in the local clock. That jitter can be very, very, very low if you take the care to do it right. -- John Stimson John Stimson's Profile: http://forums.slimdevices.com/member.php?userid=218 View this thread: http://forums.slimdevices.com/showthread.php?t=21415 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Active Crossover
jonheal Wrote: There used to be a really good record store in Claremont, the name of which I can't remember.That would be Rhino Records, owned by Chuck Oken, the drummer for Djam Karet. The recommendation for a good temperature controlled soldering iron was a good call. I use a Weller WTCPT myself, about $110 retail. Robin, the difference between us is that you make the effort to include command-line options and thus make your scripts usable by someone other than the author, then put them on the web for others to use. I'm too lazy and grouchy to do tech support or entertain feature requests, which is why I'm the evil twin ;) -- John Stimson John Stimson's Profile: http://forums.slimdevices.com/member.php?userid=218 View this thread: http://forums.slimdevices.com/showthread.php?t=20907 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Active Crossover
Hi Jon, The one change I would make is to eliminate the big loop that goes around the world from pin 2 (op-amp output) to pin 6 (op-amp -in). Just run a short wire between the two pins on the bottom side of the board. Another suggestion: get an 8-pin DIP socket for the op-amp so that you can remove it. That way you can replace it if it gets fried, or experiment with other op-amp models if you feel like doing so. By the way, did you know that I'm Robin's evil twin? We both write the same perl scripts and don't tell each other what we've done ;) -- John Stimson John Stimson's Profile: http://forums.slimdevices.com/member.php?userid=218 View this thread: http://forums.slimdevices.com/showthread.php?t=20907 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: SPDIF question - error correction and interconnect quality
75R? Do you mean 75 Ohm, or 75 Ohm right angle? 1) RCA connectors are used for SPDIF and composite or component video in just about all consumer-grade A/V gear. I would expect at least the high-end units to use something else if the RCAs didn't work. Plus, you'd be able to see visible ringing on high-quality monitors (I can easily spot a cheapo VGA cable used with a computer monitor). 2) It seems that there are several manufacturers of 75 Ohm RCA connectors (Canare, et al) who would be highly vulnerable to false advertising suits if their product did not meet their published specifications. 3) The closer an impedance anomaly is to the sender or receiver, the less effect it will have. Since the RCA connector is generally placed very close to the sending or receiving circuit, that may provide the answer to #1, but not to #2. -- John Stimson John Stimson's Profile: http://forums.slimdevices.com/member.php?userid=218 View this thread: http://forums.slimdevices.com/showthread.php?t=21415 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: SPDIF question - error correction and interconnect quality
According to what I've read, SPDIF does not include re-clocking the data at the receiver, and even the high end Crystal Semiconductor (now Texas Instruments) only filter out clock jitter at high frequencies above the audio band. So unless you are sure that your DAC buffers and re-clocks the data with a low-jitter clock, it's probably a good idea to use a high-bandwidth cable with the proper impedance. Fortunately, high-bandwidth cable with the proper impedance isn't expensive or difficult to find. One of the Squeezebox guys pointed me to http://www.bluejeanscable.com/ which is a company that assembles RCA cables using high quality cable and connectors. You can also buy the parts yourself from http://www.zackelectronics.com/ . I made my own cables using Belden 1505F (or was is 1505A?) cable and Canare RCAP-C4F connectors. Boutique cable with freaky designs is likely to perform worse than this, because they're unlikely to have impedance that's as consistent and preceisely matched, or to support as much bandwidth. -- John Stimson John Stimson's Profile: http://forums.slimdevices.com/member.php?userid=218 View this thread: http://forums.slimdevices.com/showthread.php?t=21415 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Active Crossover
1) The grounds should all be tied together, preferably at the same point. 2) For short connections, I trim a piece of the resistor leads to bridge between the connection points. You can bridge with just solder, but I feel like having the wire in there is a more robust connection. Use insulated wires for longer connections or if you have to go around another connection point. 3) I typically use 1/4 W resistors since they are easy to find. These resistors are not carrying much current due to their high resistance. 1/8W is probably okay. Don't use high power resistors becase many of those use coiled wire, which is inductive. 4)Ideally, the film capacitor should be as close as possible to the IC. Its job is to handle the high speed transients that the electrolytic capacitor can't respond to. You want to minimize the resistance and inductance between it and the IC. The electrolytic capacitors can be wherever you want on the board. -- John Stimson John Stimson's Profile: http://forums.slimdevices.com/member.php?userid=218 View this thread: http://forums.slimdevices.com/showthread.php?t=20907 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Active Crossover
jonheal Wrote: This particular board has two rows of holes down the middle, the copper pads of which are connected. I was going to use those rows as my ground. There are also numerous groupings of two and three holes, the copper pads of which are also connected. I was going to install the parts in these group hole pads, and then run jumpers of the bus wire between the group hole pads to complete connections.This sounds reasonable. The long rows of connected holes are typically meant for power and/or ground. It's too bad that there aren't three of them... the connection might require three little jumpers (six solder connections), and might run a total distance of and inch and a half. Will this distance/soldering result in too much resistance/inductance? That probably won't cause problems, since audio frequency circuits really aren't that sensitive to wire lengths. On the other hand, I like to make my connections as short as possible anyway. One thing to be avoided is making large loops formed between your signal wire and your ground. Those can pick up electromagnetic interference, for instance from your power supply or fluorescent lights. Does insulated jumper wire lower inductance, or just prevent shorts?It is for avoiding shorts. Shorts suck. I don't even understand how the capacitors effect the condition of the power supply, as they seem (from the schematic) to be off to the side of the circuit bewteen the power supply and the IC.You can imagine that your power distribution is like a plumbing system. Voltage is like pressure, and current is like flow. The power supply is like a pump that is trying to pressurize the pipes to a fixed level. It has a maximum flow that it can generate before it can't keep up and the pressure starts to drop. The capacitors are like big water tanks that fill up to a certain level due to the pressure coming from the pump. Once filled, they can provide extra water flow when the pump can't keep up. The pressure of the water at the outlet of the tank is a function of the height of the water inside the tank. A small valued capacitor is like a skinny tank. It will fill up to the same height as a wide tank, but it will drain faster and therefore the pressure will drop faster. A large valued capacitor is like a wide tank. It takes a longer time to drain, so it can maintain the pressure in the pipes without the pump for a long time. -- John Stimson John Stimson's Profile: http://forums.slimdevices.com/member.php?userid=218 View this thread: http://forums.slimdevices.com/showthread.php?t=20907 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: piano solos
Hi Sean, most of the really well recorded piano-oriented stuff I can think of is accompanied. There are a few really nice solo pieces on Bruce Hornsby's -Spirit Trail-. The Esbjorn Svensson Trio (EST) is a really excellent piano-centered jazz trio, with great recordings. -- John Stimson John Stimson's Profile: http://forums.slimdevices.com/member.php?userid=218 View this thread: http://forums.slimdevices.com/showthread.php?t=20611 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Where is the incredible sound?
Weren't you arguing the merits of disk space and future support of the format? It doesn't matter to me that you get off on particular rituals and I think it's great that you realize that the point is to enjoy the music, but what does that have to do with choosing between MP3 and FLAC? The act of playing an MP3 track is much the same as the act of playing a FLAC track. You sit down, press a few buttons on your Squeezebox remote, and the music starts. Why use FLAC? - FLAC exactly preserves the original digital recording. - FLAC is available as source code. If there is never a new version of FLAC released, you will still be able to use the current one. If you buy a new computer with different architecture, you can re-compile FLAC and it will continue to work. The only case in which that would not be true is if the C programming language were not available for your chosen hardware. - Once you've ripped to FLAC, you never have to re-rip your collection again to change formats. With MP3 you will either have to re-rip or accept the loss of quality. Converting from MP3 to another lossy format compounds the loss of quality -- different formats throw out different information. If you decide to switch from MP3 to AAC to Vorbis, you'll have to re-rip every time or suffer generational losses. If you decide to switch from FLAC to Apple Lossless to Monkey Audio to some future lossless format, direct conversion from one format to the next involves no generational loss and can be automated. So what's the downside of FLAC? - Disk space. But disks are cheap. Cheap cheap cheap. If you don't want noisy disks near where you relax or work, don't put them there. Put them somewhere else. - Not as compatible with portable players, in terms of format and in terms of space. However, it's not hard to automate the process of duplicating your entire collection into some other format from the FLAC originals and store it in a separate directory, ready to use with your portable devices. So that's really back to the disk space issue. Did I mention that disks are cheap? -- John Stimson John Stimson's Profile: http://forums.slimdevices.com/member.php?userid=218 View this thread: http://forums.slimdevices.com/showthread.php?t=21173 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Active Crossover
Skunk Wrote: I'm thinking you meant 'match up with the low pass filter on the sub', not high, but maybe I'm completely confused again.Yeah, that's what I meant. I get them confused sometimes, maybe I should just refer to them as high-cut and low-cut filters. -- John Stimson John Stimson's Profile: http://forums.slimdevices.com/member.php?userid=218 View this thread: http://forums.slimdevices.com/showthread.php?t=20907 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Active Crossover
jonheal Wrote: 1. Is the OPA134 a good choice? Digi-Key has several varieties (OPA134UA-ND, OPA134UA/2K5-ND, OPA134PA-ND)Yes, in fact I have been using the dual version (OPA2134) in my active crossover. The OPA2604 is also not bad. 2. I included the capacitor and resistor in the circuit because I saw no harm in blocking DC voltage, UNLESS it will have an adverse effect on signal quality. Will it?It won't, but tied together like they are, the averaging resistors and the highpass filter interact with each other. The main effect is that the 21.5k resistor cuts your gain by more than half. DC isn't a big deal going through a single buffer stage. The power amp for the sub probably has a DC blocker on its input. If you want to include a DC filter, put in a second buffer amp. Also, the page I borrowed most of this circuit from specified a cutoff frequency 7.4Hz based on a function that included capacitance and resistance as factors. Is there any advantage to increasing the value of the capacitor and decreasing the value of the resistor (or vice versa) to achieve the same cutoff frequency? In general, increasing the resistor value and decreasing the capacitor value will make the filter an easier load to drive but increase the sensitivity of the filter's behavior to the input impedance of the op-amp. Going the other way will draw more current from the preamp outputs but be less affected by the op-amp. 3. Are metal film resistors and polypropylene capacitors the best to use?I think that they're the most cost effective for DIY circuits. You can get some pretty fancy capacitors from DIY audio shops like the wondercap and auricap and such, but I have used the Panasonic polypropylene film capacitors for my own circuits. 4. Will the voltage I apply to the OpAmp make a difference? (I think the specifications said 15V, nominal.)You want to have the voltage supplies exceed the maximum signal you wish to reproduce by about 2V. Otherwise, anything within the specified range for the op-amp is fine. +/-9V is easy to achieve with batteries. Which reminds me -- you will want some sort of power supply reservoir capacitors between +V and ground and between -V and ground. Something like a 100uF electrolytic capacitor in parallel with a 1uF film capacitor should work for just one op-amp. -- John Stimson John Stimson's Profile: http://forums.slimdevices.com/member.php?userid=218 View this thread: http://forums.slimdevices.com/showthread.php?t=20907 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Active Crossover
Skunk, full range speakers don't generally have a high-pass filter to cut off the bass for the woofers. The mechanics of the woofer and the enclosure do that naturally. It's better to have an electronic or pure electrical (passive) crossover to do that. It gives you control and knowledge of the exact frequency, slope, and tuning of the filter so that you can match up with the high-pass filter on the subwoofer. It also eliminates wasted motion of the woofers in your main speakers. Especially in a sealed enclosure the woofer still moves when excited by the frequencies below the bass cutoff, but not enough to produce useful sound. That's a potential source of distortion and/or wasted energy. Jon, the slope of the cutoff for a sealed box is 2nd order. The slope for a ported box is 4th order. -- John Stimson John Stimson's Profile: http://forums.slimdevices.com/member.php?userid=218 View this thread: http://forums.slimdevices.com/showthread.php?t=20907 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Get the balance right
Are you saying that the VU meters on the Squeezebox show that the left channel is louder? I would expect those to register the levels in the digital data, and it would be very odd indeed if the Squeezebox were digitally boosting the level of one channel and not the other. A bug in the firmware or the slimserver, or perhaps the level bias is intrinsic to the tracks that you're testing with. What version of the firmware are you using? What version of slimserver are you running? Have you examined those files in an audio editor? What format are the files you're trying to play? -- John Stimson John Stimson's Profile: http://forums.slimdevices.com/member.php?userid=218 View this thread: http://forums.slimdevices.com/showthread.php?t=20964 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Active Crossover
Yep, that's it. The resistor and capacitor in that circuit are not necessary, they are just there to remove any DC voltage. Without them, you would just connect your source directly into the + input of the op-amp. -- John Stimson John Stimson's Profile: http://forums.slimdevices.com/member.php?userid=218 View this thread: http://forums.slimdevices.com/showthread.php?t=20907 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Get the balance right
But left and right should be calculated the same way, so you'd expect them to match if the levels match. Fezco, have you measured the analog outputs with a meter? Have you tried playing a test tone? Sometimes absorption or reflection in your listening room can make the audio sound skewed to one side. Sometimes the recording really is skewed to one side. I've certainly experienced moments of doubt that caused me to go get my multimeter and test CD... -- John Stimson John Stimson's Profile: http://forums.slimdevices.com/member.php?userid=218 View this thread: http://forums.slimdevices.com/showthread.php?t=20964 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Active Crossover
Hi Jon, You asked about my homebuilt crossover on the photos thread. So as not to clutter up that thread with unrelated chatter, I'm responding here. I designed my crossover based on the system concepts I read about in an article in -Speaker Builder- magazine. I used IC op-amps instead of the op-amps made of discrete components that were used in the article. The basic information is 'here' (http://www.idsfa.net/~john/crossover.html). Be sure to review the references in the link at the bottom of the page. If you know or can learn a little bit about op-amp circuit design, you should be able to make your own. It sounds like you'll need a high-pass section for your main amp, and a low-pass section and a summing amp (to sum the left and right channels) for the subwoofer. -- John Stimson John Stimson's Profile: http://forums.slimdevices.com/member.php?userid=218 View this thread: http://forums.slimdevices.com/showthread.php?t=20907 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Active Crossover
Because if a bass sound is hard-panned to one side or the other, you may miss it completely. Bass may be fairly nondirectional but that's no guarantee that the left and right channels of all recordings will be identical in the 20-80Hz range. A quick dirty way to blend the channels would be to put a 1kOhm resistor in series with each channel before going into a Y-connector. If your preamp can safely drive a 1kOhm load, then this will average the two signals, with a small amount of attenuation. The fancy way is to use larger resistors that tie together to feed the input of the buffer amp, which eliminates the problem of attenuation and allows you to present a higher (more friendly) load impedance to your preamp. -- John Stimson John Stimson's Profile: http://forums.slimdevices.com/member.php?userid=218 View this thread: http://forums.slimdevices.com/showthread.php?t=20907 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: Active Crossover
Nah, those are more like the quick dirty way. They're specific to 1/4 plug audio cables too. I assume that you're going to be using RCA cables. In that case, you want to put the resistor in series with the center conductor, and leave the outer conductor intact. The wiring for the center conductor is: R---(1kOhm)---\ ---Sub L---(1kOhm)---/ R and L are the subwoofer outputs of your crossover. The fancy way would replace the two 1kOhm resistors with, say, 47kOhm, and insert an active buffer circuit between the Y and the subwoofer. -- John Stimson John Stimson's Profile: http://forums.slimdevices.com/member.php?userid=218 View this thread: http://forums.slimdevices.com/showthread.php?t=20907 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: External DAC is making me grin
I migrated to the Squeezebox from a CAL CL-2500DVD player. At one point, a friend brought over a CAL Sigma (sibling to the Alpha with 1 tube instead of 2). We listened to the Sigma driven by the CL-2500DVD, by a CAL DX-1, and to each player individually. We found the Sigma to sound about the same driven by either source. The Sigma sounded better than the DX-1 alone, and not quite as good as the CL-2500DVD alone. This was the original Sigma DAC, not the Sigma II or the 24/96 version. Later on, I had a SB2 and a MSB Nelson Link III DAC. The MSB in non-upsampling mode sounded a little better than either the SB2 or the CL-2500DVD alone. Upsampling was slightly better than non-upsampling. A friend brought over a Musical Fidelity A3.24 DAC to compare to the MSB. For high female vocals, the A3.24 was sweet and sounded more pleasing. For everything else, including better imaging and keeping sounds distinct, the MSB was better. -- John Stimson John Stimson's Profile: http://forums.slimdevices.com/member.php?userid=218 View this thread: http://forums.slimdevices.com/showthread.php?t=20615 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
[SlimDevices: Audiophiles] Re: FLAC onboard decoding v. server side in SB2
void Wrote: I'm not an expert, but the jittery data signal has to pass the 'master clock gate' at the DAC. What comes out of the gate is jittery/noisy again. A flip-flop gate has a data input, a clock input, and a data output. Here's how it operates: any time the clock input switches from low to high, the data output is switched to match the level of the data input at that instant. The data output remains at that level until the next low-high clock transition. The time that the data arrives on the input pin does not affect the time that the output pin switches. The same clock signal is used to drive the clock input of every gate in the circuit, so the jitter between gates does not influence the timing of the signal at the final output -- only the jitter on the clock. Basically, the only thing that affects the signal that you hear is the jitter in the clock at the DAC output. If the clock signal is generated right there, then jitter in earlier stages doesn't cause jitter at the output. The reason that with SPDIF, jitter in the transport induces jitter in the DAC, if because with SPDIF the clock is generated in the transport. If jitter at the data input of a circuit influences the timing of the output data, then something is corrupting the clock -- excessive impedance in the power and ground planes, bad signal routing, poor grounding scheme, etc. -- John Stimson ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/audiophiles
