rjp

Wow, that looks so cool!

Thanks for the link to the Palladium review. Very interesting. Beautiful too! Clearly Klipsch makes some amazing high priced stuff. I may try some of the Heritage line in the future especially if I come across a good used deal. But what about the Reference series? Are these even worth trying for 2 channel listening? I have search for opinions online and it seems rare to find anyone using the RP line towers ( 250F, 260F, 280F) for stereo. All reviews are geared toward HT. But I think they sound better than any speaker I've auditioned so far. If I want to spend no more than 1800 for two speakers to use with a 35wpc tube amp, what would you recommend? I can still return the RP-260F if there is something better. I could also just keep these for a year while I learn more about what I want in speakers, then upgrade next year. I think I just don't know enough to make the best decision now. Which I guess is why I'm asking other people what speakers to buy What did you mean by "It's take a lot of work to get the best out of a horn set up."? What work should I be doing? Are you referring to room treatment? Thanks for any help

I have been using a subscriber Spotify account at highest quality and no normalization as my sole audio input source to my VTA70 tube amp and Klipsch RP-260F system. Is this good enough? I have read a little discussion of others saying so other digital formats are much better and I would like to learn more. As an EE I do understand the mathematics of sampling rate and the effect of bit length of samples (resolution) on SNR. I am also fairly familiar with compression algorithms, at least in concept. I guess the first question is, what exactly am I getting from Spotify streaming? Clearly it must be a compressed format, and I would assume it is a lossy compressed format., Correct? Is it MP3? What is it's average bitrate at highest quality level? Is this good enough for HiFi 2.0 listening? Is there a streaming source for lossless formats for comparison? Also, what about the quality of the D/A in the iPhone. Is this good enough? I have read testing results claiming that(beginning with the 5s) the specs as far as IMD, SNR, and freq response are as good as any stand alone unit I could buy. I think this DAC is limited to 16 bit formats. I assume it is more important to have a perceptualy lossless compression algorithm, high sampling rates, and good filters than to increase the sample resolution from 16 to 24 bits.

I agree. Klipsch (even my modest RP-260F series) sounds a lot like live music to me. The dynamics and clarity seem to be the reasons why. I now understand why Best Buy Magnolia Sound center does not have any Klipsch setup in their high end listening room. Once you hear Klipsch everything else sounds bland. I imagine B&W and the other pricey vendors will not allow it. It is hard to get someone to spend 10K on a pair of speakers when a 1K model is kicking its butt. My reason for buying the B&W's is to see if after more extended listening would I still prefer the Klipsch sound or would the less aggressive sound of the B&W start to seem more pleasant. I notice the serious audiophile reviewers all seem to shun Klipsch as not even being worthy of consideration. Why is this? Are they just not expensive enough? I find it hard to believe these audiophiles really think the sound is no good. So I'm thinking, what's wrong with me? Why don't I too like the more expensive "serious" speakers?

Last night I bought a pair of B&W 683 S2 for comparison. I've been listening to them for hours now. At first I thought they sounded very nice, then today I hooked up my remote A/B speaker line switch so I could flip between the Klipsch and the Bowers and Wilkins instantly. When I first switched to the Klipsch I thought maybe I wired them out of phase because the imaging changed so drastically from the B&W, but the wiring was correct. The sound stage on the B&W was focused in the center whereas the Klipsch was spread all over the room. After listening to both for a while I think the Klipsch was much better actually. The B&W sounded confined to the center. With the Klipsch I could hear all sorts of different instruments and subtleties in separation that all got blended together in the B&W. So I would say imaging and separation are clearly better on the Klipsch. I guess this is what's called soundstage. The B&W did have a more substantial midrange. At first I thought it sounded boxier but I got to like it after listening for a while. Maybe it is the fact that the B&W are 3-way. The Klipsch does seem to be missing these calm smooth mids. However, as far as high frequency harshness (2K - 6k) I think both speakers have about the same degree. Maybe the B&W a slight bit less. These are two very different sounding speakers, but so far I can't say I like one clearly better than the other. They each have their good points. I love the clarity and imaging of the Klipsch (and the dynamic sort of live energy!), but I also love the mellowness of the B&W mids that are there to support the highs and fill in the picture (i.e., a drum sounds like the body of the drum and not just a sharp smack on its head). Not sure I could ever give up the playful imaging of the Klipsch though even if the B&W do have a more refined overall sound. It's hard to go back after hearing that excitement. I see why people are hooked on these. More listening required.

Thanks twk123. Yes, always highest bitrate streams and no normalization. Regarding the DAC, I am using either my iPhone 5s or my macbook pro analog outputs. From what I've read, the DAC in the iPhone beginning with 5s is as good or better than most stand-alone DACs I could buy. I was surprised to read this but the article had a lot of technical distortion measurements to support this claim. If I can buy a better DAC I will. Have you any experience with the iPhone DAC? Wall treatments are definitely a good idea as are the blankets over couches.

I recently bought a pair of Klipsch RP-260F for 2 channel stereo listening with my Dynaco VTA70 tube amp and spotify streaming audio and LP. At first I thought they were the best speakers I had ever heard. I was hearing music like I'd never heard before, but after about a week my ears started becoming more and more sensitized to a certain harshness in the sound that I did not like. Sorry to use the "H" word but I'm not sure how else to describe it. I feel like I want to love these speakers. They do have a lot of great qualities, and I might be able to temper them with a bit of EQ, but was wondering if I should just try something else while they are still eligible for return. I already tried the Polk RTI a7, but in an A/B comparison the Klipsch were so much more clear and dynamic that it was hard to like the Polk even though they did present totally without any harshness. Sure, the harshness was gone, but so also was the excitement and the clarity. They sounded dull by comparison. I could also hear the resonance of the box. Is there a speaker that might be a little softer on the ears but still clear, dynamic, and inspired sounding like the RP-260? Oh, and not too much more expensive. Maybe this is asking the impossible. The room is kind of "hard" I'd say. About 28 x 16. 10 ft high. Area rugs and leather furniture, 6 windows and small curtains. I realize this is a difficult room. I have tried the miniDSP 2x4 hd (an amazing product btw) and REW with the UMIC-1 measurement microphone to try to EQ out the harshness, but it is not been as effective as I'd like. I have better success with a simple analog Loki tone control box.

This weekend I had a chance to more thoroughly evaluate the miniDSP 2x4 hd and it definitely introduces some loss of fidelity, but not much. I did this using a line level A/B switch to allow me to listen to the same source either though the miniDSP or direct. I set all filters in the miniDSP off. My goal was to hear only the effects of the miniDSP's signal path when it is not trying to change the sound. I figure this is a baseline of the best it can do as far as preserving fidelity. I could hear a slight loss of high end clarity and spaciousness through the miniDSP path compared to the direct path. My conclusion from this is the the miniDSP's A/D, D/A, and signal path circuitry introduce a slight loss in fidelity. Of course this must be the case, but I wasn't sure if it would be audible. It is. The loss of fidelity is slight, and I could easily see how the benefits of equalization it brings could outweigh this. So pretty much as expected, it is a trade-off as to which is more important in the particular situation and the particular listener.

Thanks I just copied what I liked from other pictures I saw online. Too bad I'll never get to see it once the bottom's on though

Btw, I had the amp on the bench for some sweep testing last night and I am amazed at the performance I am seeing. The frequency response is perfectly flat from 20-20K. In fact, it's pretty much flat from near DC to over 30Khz. The NFB seems to be implemented perfectly. 10KHz square wave presentation is textbook perfect. No ringing, nice and square. 40Hz square wave looks as good as the advertised specs for the new ST70 gen3. The slopes are only slightly tilted down. This thing can really sustain some low end energy! I don't have a distortion analyzer, but I am seeing a solid 38 watts RMS per channel with both channels driven at 1KHz into 8 Ohms before any clipping is visible. At this level I see no signs of crossover distortion or caving of the sinusoid shapes. I think it's a keeper! Here is a pic. The purple wires are for a volume control pot I installed. May move to a stepped attenuator later. PIO caps just arrived from Ukraine so I'll be putting them in soon. This was so much fun to build

How are you liking the combination of miniDSP and VTA70? I have both as well and am still getting used to what the miniDSP can do. I am only running 2 ch system so no crossover functions enabled, just PEQ. After a week or so of evaluating I have found some pretty good room corrections, but I can't help wondering if the introduction of all this digital processing is degrading the sound more than it is helping the frequency response. I suppose it is always a trade off. It is nice that it is possible to send digital digital input to the miniDSP directly from my computer/iPhone as this eliminates one stage of D/A and A/D conversion. I wonder if the final D/A in the miniDSP is as good as the iPhone. In any event, it is an extraordinarily powerful device and a lot of fun to play with. I have the 2x4 HD version and the UMIC-1 with REW.

Got some pretty good sounding EQ settings last night. I decided to explore the "generalized room correction" approach a bit more. I made a collection of measurements at 8 listening positions and averaged them together and then filtered at 1/12 octave. I then constructed 10 PEQs over full range to match this. I downloaded the 90 degree mic cal file for this and used the mic pointing at the ceiling. The results sound pretty good. Fixed some bass resonances I didn't even realize were there. I'd still like to soften up the mids a bit more. Overall I found that I tend to get better results (to my ears) with the PEQ's cutting rather than boosting. If I allow the PEQs to have unlimited positive gain and position the target response in the middle of the recorded data the predicted response is nearly perfectly flat, but to my ears it sounds a little better if I instead set the target response a little lower and restrict the PEQ gains to no more than +3dB (or even 0dB). I believe this forces the algorithm to do the best it can with cuts only.

Well just as an update. I now have the miniDSP 2x4 HD and the UMIC-1 and REW. I have to say that as an engineer myself, I am in heaven with all this adjustability and measurement. I have spent about 5 hours tweaking so far and so far haven't found any settings that sound good to me, but it is still a lot of fun. I see each PEQ is actually implemented as a 5 coefficient IIR filter. How neat! And I see there is even an option for a FIR filter with up to 2048 taps. This is an amazing little box! One thing I wish it had is a way to bypass the entire PEQ set in a single click for A/B comparison rather than having to bypass each of the 10 PEQ's one at a time. Is there a way? I have been reading the help documents on the miniDSP site and I am impressed with the technical detail they include. The article made it really clear why I should be interested in the slope of the excess phase (They call this excess group delay) as an indicator os which portions of the spectrum are likely to be "EQ-able" without making a mess of the phase response. Honestly, this device and REW EQ are so technical I wonder how a non electrical engineer could even understand it well enough to use it. It is really easy to mess things up reall bad if you don't know what you are doing. (and I barely know what I am doing at this point) So at this point, one day into the fun, I would say I love this product and appreciate what it can do, but I can't seem to find a way to use it to improve the sound in my room. I could use some pointers on how to proceed. So far I tried two basic approaches: (1) Measuring just one speaker at 1M and EQing for that and applying to both speakers. I have observed that the options for smoothing, freq range to treat, and maximum individual gain make a huge difference in the results. (2) MEasuring the "room" by taking 6 to 10 mic position measurements and averaging them, smoothing, and EQing. (3) Just manually playing around with a single PEQ to find annoying frequencies and cut them out. This has been remarkably effective actually. The method I use is to generate a single PEQ with high gain and narrow BW and move it around in frequency until I find the spot that sounds the absolute worst, then change the gain to negative and Bam! it is gone. I find it is much easier to find what sounds bad than what sounds good. And I'm fairly used to this technique from working sound for bands back in the day so I guess it comes easier. But I want to get the measurement method to work better. Any tips appreciated. Thanks!

Chris, Mathematically it seems the ideal phase would be any linear function of frequency since this would give a single constant time shift to the entire signal, thus preserving TOA across the board. It shouldn't matter what the slope of this phase is as long as it's linear. Does this hold true for audio systems as well? You mentioned group delay above as a figure of merit. Is this simply the derivative of phase as a function of frequency? If so then constant group delay is the goal I assume?

Great. Thanks Chris. Regarding the time alignment. Is this just for aligning the time of arrival from physically different drivers, i.e., a bi-amped woofer and a tweeter fed by a time-adjustable crossover, or can it be used to correct for frequency dependent TOA in a speaker system fed by a single amp? Does that make sense? The latter would be much more complicated if I'm explaining it correctly.

Hi Chris, Excellent write up. This tool looks really interesting. Hard to believe it is free! I have been playing around with the application on my computer for a while and am amazed at the tools and visualization this software provides. I just ordered the UMIK-1 and would like to know if you could recommend a good source where I could learn to use it effectively. There is so much information online I scarcely know where to begin. My primary interest at this point is for a 2 channel setup with no sub. I have a pair of Klipsch RP-260F in a room about 28x16. I plan to experiment with a miniDSP 2x4 HD to see what it can do to produce the best sound possible with these speakers in that room. The mic should arrive tomorrow. Is there a special procedure for equalization of 2 Ch systems? Seems so much is geared towards home theater and sub integration. Some basic questions: Do I measure one speaker at a time? I would think trying to measure both with a monaural input signal would cause an interference pattern that would ruin the measurements. Where should I place the mic? The freq response and PEQ correction seems pretty straight forward, but what is all this about time/phase correction? Is the treatment of time of arrival errors separate from the PEQ generation shown in your examples above or is this already built into the PEQ filters generated? I assume the goal is to have the wavefront of signals at all frequencies arriving at the listener at the same time, so I would need some sort of frequency based time correction. Thanks for any help RIch

^^^ This. Sure sounds like a speaker issue to me.

My understanding is that damping factor works a bit differently. The output impedance of a good audio amplifier is frequently less than 1 ohm. It makes no difference how many resistors the signal has passed through on the way through the amp. An amplifier with a very low output impedance by definition has a very high damping factor, since DF is by definition inversely proportional to the output impedance. High DF is good because it means the amp can make the speaker do exactly what it wants, prevents overshoot, even stop it dead if necessary. If a resistor is added to the speaker line then there can be a voltage drop making it more difficult for the amp to control the speaker. The amp will not be able to "damp" as well, but this is only due to the voltage drop and has nothing to do with the signal being degraded by passing through a resistor.

??? Pzannucci, Did you read the PWK article Deang posted back on Page 5?

Yes, that is exactly correct. Any resistance in the output path interferes with the amp's ability to "hold" the speak exactly where it wants to (Damping factor). If the amplifier itself has a very low output impedance (high damping factor) as in modern SS designs this will probably be more noticeable than an older tube amp I think.

I don't know how PWK felt about resistors in general, but in this particular article he shows that the resistor network (L-pad) attenuated the peaks less than the troughs in a frequency domain plot of the mid range horn he was using. He attributed this to higher order harmonics. The autotransformer attenuated them equally. This was why he chose the transformer over the resistor as I read it.

That's the good part about resistors. Any undesired signal is turned into heat rather than coming out of the speakers. It is perfectly discarded forever. Where does the extra treble energy go when the tweeter level is reduced by a transformer Back into the rest of the circuit.

Great read!

Yes. Interestingly, even this resulting network of two fixed resistors is still called an L-pad sometimes, which can be confusing.

That has been my opinion as well. It seems pretty hard to mess up a resistor. Using a transformer is interesting though. Whereas the resistor dissipates the unwanted signal energy as heat, the transformer changes the voltage and current ratio. It seem that if a driver volume is reduced by a step down transformer then the current flow through the crossover must be reduced. With a resistor the current flow is unchanged. I wonder if changing the current flow through the crossover effects the sound? I would think it must if there is any internal resistance at all.