My system sounds wonderful when playing well recorded jazz, classical, or "audiophile approved" material. Unfortunately, mass market pop frequently sounds horrible, with screechy splashy highs. It's obviously recorded with a built in bias to be played on car radios or lo-fi mp3s. What can I add to my system to tone-down the highs on this sort of material? Sure, there's plenty of well recorded material to listen to, but there are plenty of pop rock bands I'd really like to explore if the recordings could be made a bit more listenable.
Lot's of frustrated closed-minded individual stuck in the snobbish audio-prison!
Who cares of one's solution for attaining enjoyment from his sound system. Cables, tone controls, equalizers, secret magical stones placed on speakers during a full moon, who cares ?
If it gets you there, do it! And pay no attention to those who put down others if they tinker with "purity". Sometimes a pure, unaltered signal can make your ears bleed in the wrong room, the wrong walls, the wrong surroundings furniture, hell, the wrong ears!
So go ahead, fight back with whatever technology or means at your disposal, as long as you enjoy the end results!
That's what this hobby should be about, enjoyment. So, remove those cobwebs and go for it?
Shakeydeal, you might want to read the recent review of the DSPeasker Dual Core in TAS. He addresses your blind bias towards equalization in very thoughtful way. He points out that the professionals that engineered the recordings you buy use EQ extensively. If it's good enough for them, it's good enough for audiophiles to adjust the sound of bad (too bright), or older (lean bass) recordings to improve their sound.
If not you would still have to use the DSPeaker in an analog processor/tape loop, since its only digital input is optical.
I'm using a MSB Digital Director to convert from Coax to Optical into my Dual Core. I have the same issue with Coax only from my 47 Labs Flatfish transport. MSB doesn't make them any more so you have to find a used one, about $100.00. The Digital Director is 12v and can be powered by a 12v battery power supply if desired. Also the Behringer SRC 2496 can be used as well to convert from coax to optical. See how it is used here, http://forum.audiogon.com/cgi-bin/fr.pl?ddgtl&1329307228&read&keyw&zzdspeaker. Scroll down to the response by Ginov.
I think you guys are misjudging what he means when he says the material sounds like crap. The OP listens to music that's recorded hot with a ton of processing. Yes the DSPeaker will help but it won't fix the problem.IC's and speaker cable won't do jack for his problem, the recordings are simply bad. Which indictates to me that he listens to music he likes, not audiophile crap that's recorded to perfection (boring)with the life stripped out of the music.
How do I know? Because I had the same problem with what I listen to. If he has M/L's that will only make the problem worse. Too many speaker designers/builders seem to have lost their hearing between 2500-5000htz. Most speakers are just plain too hot/emphsized in that range. That's a lot of the reason women can't stand to listen to most speakers. Their just too hot in that area.
Personally it drives me nuts, but there are speakers out their that will help in his pursuit of happy listening, though some are considered out of the normal audiophile approved speakers. Speakers that throw most all of the sound in your face are not helpful in this situation. Sometimes going against conventional thinking of what speaker to use can make it much more tolerable to the OP.
Of course I don't think he's willing to change speakers but those crappy sounding recordings he'll take into consideration a little more in the future when he looks/listens to speakers. Just for kicks take a look at the fr of the Apogee Stages in Stereophile on-line. The way it drops in the rising fr is how I can get around the problem. Some people are also very sensitive to the fr between 2500-5k hertz range. There are other speakers that will help his problem besides the Stages though. So the DSpeaker unit should help out I would think. Just a thought.
I just noticed a thread asking about some Perreaux stuff, an Amp, Preamp and the TC (short for tone controls). I put this in my SP10 in the 80's to do just what you are trying to do. Its really transparent and will work well in your pre-amp tape loop. It has 3 tone control, low medium, and high. It also has high and low filters, a defeat switch and a headphone amp. It was intended to be used in-line between amp and pre-amp and works well there. If you are interested let the guy know. I suspect its worth about $150 +/-. Cost about $350 new. Contact the guy if you're interested.
The transport is also equipped with and AES/EBU output, and seems to be driven along with the coax. I know since I was able to do some A/B testing between the Enkianthus and the Musical Fidelity DACs.
Good! The rear panel photo I was looking at earlier must have been misidentified, but I now see some photos showing the AES/EBU output.
So that would provide you with a means of using the Behringer unit. I can't tell for sure from the photos and writeups I've found if the transport provides an optical output as well, though. If not you would still have to use the DSPeaker in an analog processor/tape loop, since its only digital input is optical.
Swanny, thanks very much for the info about the input impedance of the DSPeaker. I've been considering giving it a try in my system as well, at some point in the next few months, to deal with a room-related suckout I have in the 40 to 50 Hz area. Still a little concerned about the effects on transparency of introducing A/D/A conversions into the main signal path, however, although the very positive comments from Kal, REG at TAS, Roscoeiii here, and others, leave me very tempted.
Almarg --- the transport is also equipped with and AES/EBU output, and seems to be driven along with the coax. I know since I was able to do some A/B testing between the Enkianthus and the Musical Fidelity DACs.
Newbee --- moving the speakers is always worth a try. In my initial setup I had them toed in more, pointing to the listening position. To my ear, that resulted in too much high energy, which tended to confirm the recommendations made in the Martin Logan manual to have the inner 1/3rd of the curvilinear panel pointing to the listening position. Opening up the spacing would be easy to try out.
To add to Almarg's always helpful post above--About the DSPeaker--it looks like you can special order input impedance to whatever value you like for slight up charge (not specified). Standard is 10k ohm. I have been kicking the tires on that unit and looked at the owners manual yesterday. Hope this is helpful.
Bama, Nice post. A couple of comments on speaker/listening chair location/set up. Keep an open mind on this because some of it is not intuitive and conrtary to customary set up.
1) Set the speakers up so that the speakers/listening chair form an equallateral triangle. In a room with you room's dimensions I would keep the distance from the back wall at 5'+, but I would seperate the speakers substantially more apart. Say 10' with the listening chair back to about 10 - 11'. This will give you a much larger soundstage and, in my experience contribute to reducing congestion which is not unusual when the speakers are too close together and listened for far back.
This will also place the speakers closer to the side walls but due to the nature of those speaker's radiation pattern side wall reflections are not a big issue (remember figure 8 pattern). The only thing it might affect is the bass response linearity a bit, maybe a little boost between 100 and 200 hz. But then all speaker position will do that Often its a matter of choice of options, not perfection.
2)I would toe the speakers in so the axis of the speakers pointed at the listening chair and the back wave bounced off the side wall behind them. This will reduce the need for acoustic treatment on the wall behind the speakers substantially and quite possibly bring the stereo image into greater focus.
This is going to look and sound different than your present set up substantially I think. It will also require some tweaking after initial set up so give it a chance. If it works it will save you money and grief and perhaps as I suspect enhance your listening experience. If not just put them back to where you have set them now. Invest a couple of weeks (at least in this project). I discovered all of this the hard way and it took me an embarrasing long time :-)
BTW, flat bass response set with meter or by ear is going to be impossible, close perhaps. But I do not let flat bass response dictate where I set my speakers unless it is gross. It's the mid's and highs that are essential. Usually the location of the listening chair is as, if not more, important for getting a good bass response anyway.
OK, I'm done. Sorry for the length of my posts. Good luck and let us know how it works out for you.
Impedence issues between the CJ preamp and Pass amp were (and maybe still are) a concern. I raised this issue a while back on another forum and received feedback from several folks (including Kal Rubinson) confirming that the pairing meets the guideline that "the input impedance should exceed the output impedance of its source by 10x or more".
Like you, I am uncertain. The 10x factor should be applied to the worst case (maximum) output impedance of the preamp at any audible frequency. For a tube preamp that will commonly be at 20 Hz, due to the output coupling capacitor most (but not all) tube preamps use. Per Stereophile's measurements the original version of the C-J 16LS had a worst case output impedance (at 20 Hz) of 1.8K, and from their comments in the addendum to the review it sounds like the upgrades that were incorporated in the Series 2 version would not have affected that. The manual for your X250 indicates that the input impedance of its balanced inputs is 22K, while not indicating an input impedance for the unbalanced inputs which I presume are provided and you are using. Conceivably the unbalanced input impedance could be half the value of the balanced input, or 11K, which would result in a SLIGHT rolloff of the deepest bass when driven by an output impedance that rises to 1.8K at 20 Hz, while having significantly lower values at higher frequencies.
Re the use of digital equalizers: I did some further checking, and it appears that the two digital outputs of your transport are both 75 ohm coaxial outputs, on an RCA connector and a BNC connector. It also appears that neither the Behringer DEQ2496 or the DSPeaker will accept coaxial digital inputs. Also, the DEQ2496 does not provide unbalanced analog inputs or outputs, so it could not be inserted into a processor/tape loop on your preamp without converting or adapting the signals from unbalanced to balanced form, and back, which would introduce additional cost and/or possible sonic compromise depending on how it is implemented.
The DSPeaker device provides unbalanced analog i/o's, as I and others indicated earlier, and could be readily introduced into one of the processor/tape loops that is provided on your preamp. The one possible issue I see with respect to that approach is that the output impedance of the preamp's processor/tape outputs is not specified, and the input impedance of the DSPeaker's analog inputs appears not to be specified. You might want to contact the two manufacturers to ask them if they can supply those numbers. Although given that the DSPeaker would only be in the signal path when you are listening to the problematic low quality recordings, a less than ideal impedance match may not matter anyway.
Finally, in case you want to consider other kinds of digital equalizers, which could accept the signal from one of the transport's coaxial outputs, I would be cautious about assuming that both outputs can be used simultaneously, unless the manual or other literature explicitly indicates that they can be. It seems conceivable to me that the RCA and BNC connectors might simply be jumpered together inside the rear panel of the transport (rather than being driven by separate output stages), the expectation being that only one of them would be used. In that situation, using both at once would result in a severe impedance mismatch, which would undoubtedly degrade sonics for both outputs.
Again, thanks for all the responses. In answer to a few of the comments, let me offer the following:
.... Youthful, high acuity hearing does not appear to be the issue - I'm a proud member of the boomer generation. My hearing still tests well, but not like it was back in my 20s.
.... Impedence issues between the CJ preamp and Pass amp were (and maybe still are) a concern. I raised this issue a while back on another forum and received feedback from several folks (including Kal Rubinson) confirming that the pairing meets the guideline that "the input impedance should exceed the output impedance of its source by 10x or more".
.... The photo of my system in my profile is obsolete, since we've moved to a new home. The new arrangement is similar, in that a group of windows (with some drapery panels) lies behind the speakers. The room is dedicated to the audio system however, so I do have as much flexibility as needed to try acoustic treatments. Rough dimensions are 14 ft wide by 32 ft long, with the speakers firing down the long dimension. Speakers are roughly 7.5 ft apart (center to center) and about 5.5 ft from front wall (to speaker front side). The listening position is about 13.5 ft from the speaker front. Slight toe-in (to cross behind the listening position). For what it was worth, I used a sound pressure meter in an attempt to position the listening position away from any obvious low frequency nodes.
.... Previous cables were Purist Audio Elementa. To my ears, the Nordost was an improvement in speed and clarity.
.... In a previous system, I used a hybrid tube/solid state ARC preamp (LS2), with a solid state ARC amp (D200) and dynamic speakers (Theil 2 2s). It seemed to be a good system at the time, but I'd say that crummy recordings still sounded crummy. The current system beats the previous one in air and clarity on good recordings but, sadly, still sounds crummy on the others.
I appreciate the added info on the DSPeaker Dual Core. I'll do more research on it. Christmas is coming!
Lastly, I'm appreciative of all suggestions. I was merely trying to focus the conversation. Perhaps the added info in this post provides more background.
Bvdiman, Normally I would agree with your sentiments. In fact in my first post I completely avoided comments of any type on other possible contributions to the problems he was experiencing and a couple of possible solutions. None what so ever. As I said, he had good stuff and said he didn't want to change the sound of his system and I thought 'nuf said.
However since that post, he has recieved all sorts of recommendations, some fleshed out, some not. For example, the toe in recommendation. Now toe in with electrostats/panel speakers can be a huge issue because of the back wave, a problem distinctly different from toe in with boxes. Box speakers have a broad dispersion pattern vs the narrow dispersion pattern (think fan shape v a typically figure 8 shaped) of panels & electrostats. I looked at his set up and his speaker placement was very typical for boxed speakers, not necessarily for panels or electrostats, obviously IMHO. I learned that the hard way - no one told me. It was free advise that someone might just find useful, and it costs nothing to implement. Then why not? I saw no reason to assume he would be offended.
Ditto, Nordost cabling. It has a particular reputation in the market regarding its sound that differentiates it from a lot of other cable. Was Bama aware? If so, no harm. If not, well it gives him something to think about.
His post on 11/28 was somewhat interesting I thought. On the one hand he thought suggestions by myself and others, to put some type of an equalizer in the tape loop were very good ones, yet he neither closed the correspondence nor sought further guidance in an area with limited potential for resolution beyond that done. He just complained about the rec's that were beyond his original request. Complaining about free advise is always curious to me.
This suggested to me that perhaps, even against his comments to the contrary, that fleshed out recommendations might cause him to re-think his problems and potential solutions. I thought I'd take a try at spelling it out a bit. If he already knows - OK. If he doesn't know, he learns. What's at risk, his ego? Why would/should we think that? I work on the assumption that only your friends tell you your fly is open. Others like to just stand back and giggle.
Why do I take the time to write two posts and even explain myself to you? It is because I am (and I would like to say we are) trying to help him improve his sound, even if we have to go beyond his request. That is what most of us would want if we were in the posters position. Pride disappearing in the face of constructive critism or honest observations is not an attribute I normally ascribe to folks who need help, validation perhaps. After all, at the worst we are all a bunch of assholes and he'll never correspond with us again, or perhaps he had a hole in his knowledge base that we have helped him fill.
Bad recordings should sound, well.. bad, played through an honest system. If its tailored to sound acceptable or good reproducing them, then a compromise or two must have been made somewhere to compensate this system's fidelity ie. system is suppress to lying.
To me, the OP has clearly stated right from the get go (then repeatedly) that his main concern is ONLY that of the lesser (bad) recordings, and NOT of the system's set-up nor its general performance (which he seems to be already very happy and contend with!).
Thus, I assume he knows fully well the capability of this system, and had identified the symptom/s and only looking for advice as to what medication best to remedy it.
Let's not undermine the OP's experiences and also of his hearing acuity. Just take it at that and shoot simple suggestions as to what we think best whether it be aspirin, paracetamol or..??--to get that annoying pain off his head as requested, instead of re-analyzing over his illness, and giving him more of it. :P
Bama, FWIW, you can lower the volume of distortion in the upper-mids and highs, but you still get distortion only on a slightly reduced level. Maybe not so good.
But, how do you know without trying first. Perhaps an equalizer or tone control, digital or analog, will solve the problem. Maybe not. It is the cheapest possible solution though. And for me at least, the choice would be the simplist one to use, i.e. it has the flattest learning curve and very simple to use. After all, the recordings are poor in the first place, what would be lost?
I have not challenged the sound quality of the equipment or the set up being used. (You didn't ask and it is good stuff.) But, FWIW, I think the synergy itself may produce a set up with overly emphasized upper mids/highs which passes amoungst many audiophiles as evidence of what the sound should be, but rarely that it is too bright, because it is so smooth when playing high quality recordings you accept the HF emphasis as being correct or consonant with live music.
Obviously when the OP then plays a lower quality recording the poor highs etc stand out like a sore thumb. This is a problem I think created by the industry professionals in making and marketing stuff. Everyone is demanding high quality resolution and the professional's response is often little more than elevating the sound of the high(er) frequencies either by using frequency responses or rise/fall times, i.e. speed, changing the natural resolution of the sound.
A couple of things about your system which caused me to bring this up. The speaker set up looks good but you have a lot of glass and hard wall surfaces behind your speakers which can overemphasize backwave sound and cause distortions.
I would put reflective/absorbant materiels on those walls/glass. Also you can change the direction of the back wave by toing in the speakers so that you are listening on axis. Then the back wave is bouncing off the backwall (or in a narrow room, the side wall) at an angle which will make its arrival at the listening position much more distinct from the first signal (I did that very successfully with Quad 63's).
I would also try some different cable, something not known (in a general sense) for its 'wonderful' delivery of the high frequencies. I used Nordost cables for a time - loved the 'clarity' until found that it really only over emphasized the extant highs of my system.
If you did those things, your first impression might be that you have just dulled down the overall sound of your system, and you just well might have done so, but consider whether or not the new 'sound' is dulled down or actually more realistic, i.e. sounds like real live music, not just the sound of high end audio.
FWIW I went through high quality analog tone controls and equalizers several times many years ago for just the problem you are trying to solve. I haven't used them in years. Their demise started with the conclusion that the sound I was experiencing was more because of my need to have 'face', by buying and implementing really high end stuff, inadequately I think, than it was to produce a sound that was consonant with what good sounding music could sound like in my home. I recall then getting a lot of admiration (of equipment) and a concurrent inability to produce what I considered high quality sound. (Now I get excellent sound and little appreciation. :-)
I don't know if this is of any value to you at all, but what the hell....and its free! :-)
"This is what happens when you place music before equipment."
No music without equipment unfortunately.
You might still need digital processing to get what you want out of certain recordings, but best to have the right foundation first. That can go a long way towards enabling more musical enjoyment more easily over the long term, as opposed to spending time tweaking for every less than optimal recording encountered.
Digital equalization might help if treble level is the issue, but it is not clear to me from what has been stated so far that is the issue. Certainly, if the treble is irritating, making for less of it can only probably help, but there could be other reasons why treble is not good. Distortion would be the most likely. BEst to address that if the case. Almarg presented one scenario that might lead to distortion in the treble. Another might be the pairing of the tube pre-amp with the Pass amp. I am not sure the Pass amp is necessarily designed to work best with a tube preamp with <30K input impedance? That could be creating distortion as well that might be reduced with a better pre/amp impedance match perhaps. Nothing against the CJ or Pass gear certainly, just not sure that the pairing is optimal from a distortion perspective. Maybe others can chirp in on that. If both pre-amp/amp impedance matching and Almargs issue identified with the amp/wire/speaker combo were both in play, that could be a double whammy that accounts for what you are hearing.
I have heard similar MLs run off a SS Krell integrated (amp/preamp impedance matching should not be an issue with an integrated amp) and Krell digital source. There was no irritating treble! Quite the opposite! So I am confident that what you are hearing need not be the case with teh right foundation of equipment in place. If it were me I would want to hear your gear with either a better match between tube pre-amp and amp, or with good all SS amplification in place. Then see what recordings sound good or bad and go from there.
Bama, also I am curious how old you are? Younger people in their teens and 20's tend to hear higher frequencies better and are more affected by "bad treble" than their elders, so it is a useful thing to know.
This is what happens when you place music before equipment. Shame on you!
BTW, I can deeply relate. I rank Todd Rundgren as my favorite songwriter of the r'n'r era and his records are usually borderline unlistenable. I'd agree that digital EQ is the best solution for your problem, particularly because it appears that you use all digital sources. Design choices of ARC, CJ, etc. nothwithstanding, the impact of an additional device in the digital signal path is not IME likely to create any meaningful problem. OTOH, digital EQ can go a long way toward making bad sound acceptable.
Or you could use my own solution: listen to Todd in the car. Or on Sonos. Just not (often) on the good system.
Overall, I'm in line with your thinking that one should always design a system around best of recordings. And once you are there, not to have them later compromised by tweaking to accommodate those less than stellar ones--because then, we will be like chasing the dog's tail.
I feel that you are on the right track, as well your thinking of the best solution into addressing the problem (wherein it lies within the source). Just use a 'corrective device' that's able to jump in and do the job well when needed, and totally OUT of it (signal path) otherwise. Good luck!
*I'd like to add.. The need for one (tone controls)--I suppose also largely depending on the extent of genres and diversity of ones' music collection. Besides jazz--classical, r&b, pops, disco, reggae, traditional, techno, lounge, (you name it, I love them all!). I'm also heavily into popular music of the '60-'80s (again, various genres). Audiophile stuffs, probably only 20% of whole.
Say, about 20-30% of my collection probably don't really need correction. Around 50% of them definitely could be improved upon (if I'm in the mood to tinker. If not, still pretty much enjoyable). Now, that remaining 20% of the worst ones, actually do NEED adjustments to get an even remotely balanced enough/satisfying sound coming through.
Hence, with such diverse materials at hands, for me, my linearizer a necessity. Although quite a hefty investment upfront, it actually saves me lots of time, and bottom line--money, down the road. Halting the endless churning of cables/equipments as it once was in the past. The few alternatives suggested above are good, and cost wise seems very reasonable. Or if you want to pursue further, as mentioned in post number one a suggestion by GregM could be your ticket.
However, could also envision as some others here, if my library were to consist of mostly only good to great recordings, say 70-80% of total, I would probably not even bother having one too. Agreeing with those stating that a carefully thought out well-balanced system will suffice doing the job just fine--for most of the time at least.
*imho, the higher the fidelity, the higher the resolution ones' system evolves to/gets, the more likely you may need/want one.. That is, if your love for music has no barrier and is unrestricted by the quality of recordings, how ever they are presented in.
I can confirm what Al said. You can run the DSPeaker Dual Core through your tape loop, if you want to have the option of running analog with the signal not touching the Dual Core at all.
AND, it also has the capability of storing four different EQ & DSP profiles, in addition to a Bypass button on the remote that removes an of the DSPeaker settings (leaving you with just the DAC or A/D/A conversion of an analog signal).
So it is possible to have separate settings for low quality recordings that can tone down the aspects that you are not enjoying (and for treble there is a specific "Tilt" setting that begins at the frequency of your choosing), and settings that best suit certain genres of music, volume levels, etc.
If you can't find these locally to audition, a number of online dealers have audition or return possibilities. Tweek Geek was who I used, with both Audition and 30 day return options.
Room correction electronics such as the recommended Antimode Dual Core 2.0 ... would seem to impose their changes to the system all the time, for all recordings .... An equalizer of some sort that can be switched in or out as needed seems the best recommendation I've heard so far.
See my post above, dated 11-23-12. As I understand it the DSPeaker Antimode Dual Core 2.0 can be used purely as a parametric equalizer, albeit a very sophisticated one that operates in the digital domain internally (while providing both analog and digital interfaces), and I see no reason that it couldn't be installed in your system such that it can be completely removed from the signal path, when desired, at the flick of a switch.
HArd to say without hearing, but my gut tells me that there may still some system level tweaking along the lines folks have suggested possible to get the best general performance possible out of you setup.
If it were me, I would exhaust all the possibilities towards that end first to be in the best general position possible before looking to tweak specific recordings or recording types. You could find the need is gone at that point or perhaps greatly reduced at a minimum.
IF there are a lot of recordings of any genre that do not sound good or right, that is usually an indicator that something is still not right in general.
However, the sound of recordings varies widely. If you goal is to merely make them more uniform tweaking the whole system to best effect is not the solution. A flexible or even programmable sound processor of some sort is needed, though I think the extent to which any lesser recording can be made to sound like the best recordings is inherently limited.
I certainly appreciate all the comments, but I must not have communicated the issue properly, based on some of the responses. Room correction electronics such as the recommended Antimode Dual Core 2.0, power cords, interconnect cables, or power conditioners would seem to impose their changes to the system all the time, for all recordings. I'm not opposed to general improvements, but good recordings already sound fabulous on my system as is.
What I was looking for were recommendations on tweaking the sound ONLY on those recordings that suffer from the misguided hand of the recording engineer.
An equalizer of some sort that can be switched in or out as needed seems the best recommendation I've heard so far. Granted, use of such a device will reduce transparency, and modifies the sound from what the engineer (or the artist) created. Unfortunately that might have to be the price to pay to make the material listenable (to me at least)'
You might be a good candidate to try a decent power conditioner on your source gear and/or pre-amp as a first step. Probably can't hurt and only help.
I am not familiar with your digital source gear but assuming that is solid IC changes for a tweak might also help. Try DNM Reson for great top to bottom balance with top notch coherency especially through the upper mids. Or even inexpensive used networked MIT ICs like the Terminator series for a softer sound with more weight in the low end.
I've heard those M-Ls, and they sounded, to me, anything but ragged in the upper-mid/lower-treble range. They were driven by a SS amp and tube-pre, with computer files as a source. FWIW, as my system has evolved, I have gradually lost any edge or excessive brightness in this very sensitive range. I can now listen to those bad recordings without engaging my McIntosh preamp's tone controls, and enjoy them. I went from asking, "what the heck was the recording engineer thinking?!" to "Oh, so that's what they were hearing in the control room when this recording was mastered!" Yet, audiophile recordings still sound wonderful. I actually had a parametric EQ for the same purpose as you want one, but sold it after I realized I wasn't using it for listening any longer. I got there through numerous upgrades of equipment, speakers, cables, etc. I like having the tone control option, but I like not having to use them even more. I think your M-Ls are fully capable of providing enjoyable listening, even to poor quality recordings. You need to rethink your system synergy and try some new things. And don't ignore your room acoustics, either.
Just so you know, the Behringer is not at all intuitive to use. There is quite a learning curve, contrary to what some make describe. For many I think that would be a deal-breaker.
After tinkering around with cables and swapping equipment for years, I've finally settled on a japanese mammoth of an integrated amp with defeatable frequency trim function (aka tone controls). Finally, I feel at peace and content. While listening to my computer system, I've also discovered the usefulness of Amarra-based equalization. No shame in either
Or the folks at AR, CJ and Krell know about hacks who drink deeply of the Kool-Aid and would NEVER use tone controls or an EQ, but futze with cables and IC's to modify the sound. Roll eyes.....
Check out the DSPeaker Dual Core 2.0. A fabulously flexible unit that gives you DSP for under 500Hz, and a full EQ for the entire audio listening band. Balance controls too.
I definitely agree that an equalizer of some sort is probably needed. I like the idea of being able to switch this tone modification out of the loop when playing decently recorded material. The two methods suggested (either via the tape loop or as a separate path from the CD transport) both have merit. The Behringer sounds like a bargain.
Relative to Al's comments - yes, my system still includes the Goertz cables, and I am using the Zobels. You bring up good points relative to the capacitance issue. I had thought that the Pass X250 could handle the load without too much problem, but it's worth further investigation. The counterindication, however, is that the system sounds fabulous with good recordings.
Your system description covers two different configurations, and appears not to have been updated in many years. Are you presently using biwired Goertz MI-2 speaker cables, as shown in one of the descriptions? And if so, are you using them without a Zobel network?
If so, between the ultra-high 950 pf/foot capacitance of those cables, which as seen by the amplifier would be doubled in the biwired configuration, and the very low impedances and highly capacitive phase angles which I presume your M-L Odyssey speakers have at high frequencies, you would be subjecting the amplifier to brutally capacitive load conditions at high frequencies. While normally I would be among the last to suggest that the symptoms you are describing be addressed by focusing on cables, if the answers to my questions above are "yes" it seems very conceivable to me that the effects of all that capacitance on the amplifier could be a significant contributor to the problem, exaggerating the consequences of excess high frequency energy when it is present in the recordings.
As far as digital EQ is concerned, you may also want to consider the DSPeaker Anti-Mode 2.0 Dual Core, at around $1100, which has been getting a lot of good press. (I have no experience with it). Among many other functions, as described in Kal Rubinson's review (scroll down to the middle of the page) it "includes a 16-band, user-configurable parametric equalizer with a center frequency range from 20Hz to 24kHz, each band assignable to the right, left, or both channels."
Based on reviews and comments I have seen its transparency is quite good, but if you go that route I would, as suggested by some of the others above, still configure the system so that it could readily be switched completely out of the signal path when desired.
To correct/tame less than stellar (bad) recordings, tone control of some sort is needed. Whether it be in the form of simple bass/treble ones, digital correction, equalizer, or linearizer. I would avoid using cable as band-aids, as then you'll need loom$ of them for the diverse recordings, not to mention the trouble of swapping. Fine tuning or use as finishing touch to an already well balanced system is more the cable's duty, imo.
"My system sounds wonderful when playing well recorded jazz, classical, or "audiophile approved" material. Unfortunately, mass market pop frequently sounds horrible, with screechy splashy highs. It's obviously recorded with a built in bias to be played on car radios or lo-fi mp3s."
Welcome to the club. Your situation is very common; its one of the negative aspects of having high resolution audio equipment. There are a lot of different things you can do to fix the problem. I believe the best, and most effective fix, is to get an EQ.
"Oh for heaven's sake, he's already listening to abysmally recorded albums and you're gonna worry about the supposedly deleterious effects of an equalizer? Some times I wonder if some of you guys made it thru grade school."
I couldn't agree more. I have an EQ that I use for the same reason. You may want to look at a Behringer Ultracurve PRO DEQ2496. I have one and get great results with it.
Oh yeah, that's a smart way to go about it - not! Interconnects and speaker wires have very subtle effects if any at all. And they would always be in the signal path, thus "tailoring" the sound for all sources. Not to mention the hassle of swapping them in and out and the time and money spent purchasing them only to find out they didn't do what was needed.
And if you're buying interconnects and speaker wires that have multi-decibel effects upon certain portions (frequencies) of the signal, you're buying very poorly made products. EQ is the way to go, plain and simple...
I would suggest rather than use tone controls, do some research and make sure you use proper interconnects and speaker wires to tailor the sound you want. Some products are more forgiving than others.
This problem is exactly the reason I recently bought a Luxman 509U integrated amp, it has by-passable tone controls, makes hard sounding recordings listenable. Wish more equipment makers would add tome controls to their gear.
There is absolutely no valid reason not to use an equalizer, or a set of tone controls, IF you put it in the tape loop of your pre-amp. Judging by the effort you have already made to get quality sound, I think you would be disappointed if you just put it between your amp and pre-amp. It will affect the quality of the sound you have and be a PITA to use all of the time as you went from one type of redcording to another.
Another possible solution - get an inexpensive tubed DAC or CDP - then find tubes that will give you a very warm sound. That helps a lot and you can keep it cheap if you're careful.
But on the whole I would use an equalizer in the tape loop.
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