KeithL
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Post by KeithL on Apr 29, 2014 9:06:33 GMT -5
First off, I certainly agree that multiple subs give you more power to eliminate things like room modes - and, if properly configured, should allow you to configure the biggest possible sweet spot. I can think of a few reasons why I would avoid "high distortion subs" - and localization issues are just one of them The "stacking of filters" shouldn't be an issue - as long as one of those filters (usually the one on the sub) can be set high enough to not interact heavily with the other. If the filter on the sub can't be set to "out" or "flat", then set it all the way up. As long as one filter is set to at least twice the frequency of the other, there should be minimal interference between them. Also remember that the room compensation is going to be done after these settings are made (so any slight anomalies caused by interaction between the filters will be "nulled out" by the room correction). With most equipment, even running a 4-way Y-splitter shouldn't be a problem... although it might be in some specific situations. Generally, with solid state equipment, even a 4:1 fan-out is OK. Another interesting thought there is that, if you wanted to use multiple "simple" subs, one of those little "headphone distribution boxes" (with three or four outputs - each with its own volume control) could offer a handy way to set the relative levels of each sub - and would also buffer the signal. I do agree that it would be optimal to be able to set each sub separately for all its parameters, but the viewpoint of many folks seems to be that "multiple subs are intended to remain separate". As for "having to manually tune delay on multiple subs"... I'm honestly slightly dubious about any automatic system being able to do that well anyway. It seems to me you're starting to reach a point where there are just too many decisions and possible variations available to trust an automatic system to get it right. I've always had an aesthetic sense that symmetry is good - and so stereo subs "feels right" to me - even though I can't necessarily hear a difference. Something that people should bear in mind, though, is that using multiple subs also CREATES problems. One of the benefits of having all low frequencies come from a single sub is that you only need to find one good spot to locate it, and you avoid the possibility of multiple speakers causing interference and cancellation with each other. By moving up to multiple subs, you sort of "give up" that benefit. (Basically, what I'm saying there is that using multiple subs isn't going to automatically make things better. In fact, it will probably be more difficult to set up multiple subs, even though you may hope for a better result if things work out and you get it right.) With my Rhythmik subs, the delay knob is disabled when you use the LFE input to bypass the built-in low-pass filter. Using a Y-adapter would mean having to stack the low-pass filters of the subs and the preamp, and manually tuning delay on 4 subs. Audyssey XT32 makes this much more convenient by auto-setting delay independently for my front and back subs. I do use Y-adapters for the front and back pairs, but they are equidistant to the MLP, so that's easy. Using Dirac Live on the PC, I use my pre-amp to initially set delay and levels for all channels, then Dirac doesn't have to worry about multiple subs. It would be possible to do a 4-way y-adapter split, although that's starting to push the impedance. In addition, manually tuning delay and stacking low-pass filters means more compromises. The benefits from using multiple subs to cancel out room modes and get a larger sweet spot are huge. On the other hand, since low distortion subs are non-directional below 80hz, the benefits of stereo subs are nil. (High distortion subs are easier to localize, though.)
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KeithL
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Post by KeithL on Apr 29, 2014 10:36:02 GMT -5
Actually, that sort of depends. With multiple subs, you're adjusting for a combination of room modes and the interactions between both subs and the room - which works out to be rather more complicated than a simple distance compensation. Because of the long wavelengths involved at very low frequencies, the "phase" adjustment provided on many subs will give you a similar - but not identical - adjustment capability. (At 20 Hz, one wavelength is about 50 feet, so a 90 degree phase shift at 20 Hz is equivalent to a 12 foot shift.) Since the "phase" adjustment is implemented differently on different subs - your results may vary - a lot. Yes, the XMC-1 does EQ the two subs individually..... But, here's a devious suggestion : If you really want your two subs to be EQed TOGETHER (like if you have already adjusted their relative levels and responses the way you like them using their own controls, or some other little EQ black box), then just connect them both to ONE of the sub outputs on the XMC-1 using a Y-adapter. It'll work just fine, and the XMC-1 will treat them as one sub - just like you want it to. Yes but. What if the subs are at different locations (you may have one in the far corner and one as a nearfield)? I think the suggestion is that the distances/trim should be independent - which you will get by using both sub outputs - but that the EQ should *always* be based on the combined response. It doesn't seem to me that this is going to be possible unless you can fake it with some two-step process to get the settings you need.
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Post by rcohen on Apr 29, 2014 11:12:28 GMT -5
Thanks for the info. I have generally just trusted the auto-delay in my pre-amp, rather than verifying it, since it gives such consistent results. It sounds like that would be worth trying. When I compare matching the delay from the front & back subs, vs. using the auto-time-aligned delay, it certainly sounds a lot better time-aligned. But maybe I am leaving something on the table, by not fine-tuning it by hand.
For my large closed room, going from 1 to 2 subs was a crazy huge improvement in sound quality, since I had terrible problems with deep nulls using a single sub, regardless of placement. Going from 2 to 4 was a more modest improvement. More power and more seat-to-seat consistency (bigger sweet spot). Placement was still very important for 2 & 4 subs, but at least the nulls could be fixed.
Anyway, individually EQ'ing subs would be disastrous for me, since it's impossible to get reasonable FR from an individual sub in my room, regardless of placement.
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Post by sme on May 4, 2014 19:58:04 GMT -5
I have integrated 4 subs (actually, 2 Hsu Research "true subs" and 2 MBMs) with a MiniDSP 2x4. It allows me to set delays on each sub independently. Unfortunately, the delay is limited to 7.5 ms, which is only sufficient for integrating subs when the maximum difference in distances is around 8.5 ft. Thankfully, that's enough for me.
Another caveat with the MiniDSP is that is clips its input at 2 V (or 0.9V, depending on which option you choose) and clips its output at 0.9 V. My Denon AVR doesn't clip its line-level outputs until >4 V, and the trim adjustment doesn't go low enough for me to ensure enough headroom for the MiniDSP. I addressed this problem by installing a "6 dB" line attenuator (one that measured closer to 8 dB, despite its specs) between the AVR and MiniDSP. I also had to crank the gains on my sub amps, and I'm very thankful they provide as much gain adjustment as they do. Now, with my AVR sub trim set near the minimum, I can handle sub bass peaks beyond 120 dB cleanly! SPL of 120 dB may sound ridiculous, but a handful of films played at reference level get close. The surprising thing is that it doesn't sound anywhere near as loud as you'd think when it's clean and tight.
I have to agree with other posters that it's almost always best to first time-align a pair of subs and then EQ them as a whole. That way the direct sound is coherent, and this coherence will give the room EQ "more control" over the room. One thing you definitely don't want to do is EQ the subs *independently*, whether or not they are time-aligned! Each EQ will alter the phase response of each sub and the combined response will exhibit interference that can't be corrected with delay. Theoretically, it is possible to use different EQs on two subs, but the algorithm must take into account the combined response in the process. I'm not aware of any room correction algorithms that automate this as it's a fairly complicated problem to solve. I doubt Dirac offers this, but if it does, I'd be skeptical about the performance vs. a combined time-aligned EQ.
Similarly, I think stereo bass is usually a bad idea. That's not to say I haven't given it a lot of thought. In fact, I personally am unusually sensitive to the phenomenon of "directional bass" and have heard directionality even when using 80 Hz or even 60 Hz crossovers. I notice it most by far with pure tones, which I encounter in a lot of the electronic music I listen to. With experience, I have discovered that this apparent directionality occurs due to room nulls. Our ears are sensitive to pressure differences between them regardless of frequency. For bass sounding in an open space, the pressure difference between ears will be nearly zero unless one has his/her head right next to the driver because the sound easily diffracts around the head. On the other hand, in a closed space, standing waves can create a significant and audible pressure difference between the ears. This same phenomenon can be recreated with headphones by playing a pure bass tone in only one channel. (I must say, the sensation is rather uncomfortable.)
So the best way to avoid directional bass is to improve the overall in-room response. And because one can typically (note, I don't say "always") achieve a better in-room response with multiple coherent bass sources, it's better to not use stereo bass management. If we are to consider large spaces like concert halls and (maybe) movie theaters where room effects are reduced, stereo bass may have some merit, but for residential rooms, I wouldn't recommend it. With that said, 2 channel owners are right to point out that they already play bass in stereo and that they prefer it to the bass they get from their sub. Notwithstanding the fact that many subs are marketed for home theater use and have relatively poor mid/upper-bass transient response and headroom, a pair of stereo speakers often provide better bass than a single sub for the same reason that two subs generally sound better than one. Much if not most 2 channel music has the bass mixed dead center anyway, so 2 channel users are in some sense getting the "2 subs sound" for free. I'd be willing to bet however that when playing sources with bass panned far off center, a pseudo-bass management system that extracts the bass from each channel, mixes it down to a single signal, and sends that signal equally to the left and right speaker will improve the bass. The listener will benefit both from the better room response by using two coherent sources and the increased headroom by sharing the bass duties equally between two drivers and amp channels.
I also want to make a separate comment about Audyssey in particular. Audyssey's multiple sub EQ time-aligns and level-matches the sub pair first and then EQs their combined response. I've already argued in favor of time-aligning multiple subs, but I think Audyssey's decision to level match the subs is not good at all. The reason they do this is almost certainly because they cannot provide an automated method to *gain match* the subs instead. It is almost always better to gain match than level match subs because otherwise the system headroom will be limited by the sub with the highest level. Another issue I see with Audyssey (MultEQ XT in my case) is that it doesn't always correctly measure the sub distance relative to the mains. It's not too hard to correct this error manually because the sub EQ correction is completely independent of the distance setting. However, if this problem also impacts distance measurement of multiple subs relative to each other, then there will be no way to correct this after the fact because the EQ of the combined response *does* depend on the relative sub distances. Unless Audyssey let you tweak the distances before running the combined correction (it doesn't), you'd end up with a substantially sub-optimal result.
Just one more note about EQing sources separately. For multiple subs, this is generally a bad idea, but if you think about it, this sort of thing happens when we EQ the mains and sub separately and then combine them with crossovers. The mains and sub play together in the crossover region and the separate EQ corrections can cause them to interfere even if they are time-aligned. Ideally, the auto-setup program should EQ each speaker in combination with the sub(s) using the same crossover settings that will be used for playback, but that's not how the consumer version of Audyssey works. When I first ran Audyssey Pro (using the separate installer kit), I was shocked by how much the bass improved from previous calibrations. At first, I thought Pro had enabled Audyssey MultiEQ XT32 on my AVR that only supports XT, but after some reading and careful measurements, I determined that the improvement came from the optimized crossover filters that Audyssey Pro enables. It turns out my mains were putting out enough sound at 40-80Hz, even with the standard THX 12 dB/octave LR crossover, that they were interfering with the subs. It just so happens that the THX standard crossovers that every AVR manufacturer implements were designed to be used with sealed-box speakers that have a natural 12 dB/octave roll off at 80 Hz. My mains are ported and are essentially flat to 60 Hz. What Audyssey Pro did was implement a filter that ensured that my mains rolled off at a full 24 dB/octave with the standard THX crossover enabled. This approach is not as good as EQing the combined output of each speaker with the subs would be, but it still makes a huge improvement. I don't know if Dirac on the XMC-1 will do this kind of thing, but I would hope so.
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Post by rcohen on May 4, 2014 21:59:16 GMT -5
I haven't been much of a fan of the Audyssey XT32/Pro auto crossover selection. It typically suggests 40hz for me, due to room gain. 80hz sounds a lot better and doesn't hurt localization. In my system with 4 subs, the higher the crossover frequency goes, the flatter the pre-EQ response. That's because the subs are placed to cancel out room modes. The mains are placed where mains have to go. Going above 80hz gives me even better pre-EQ response, but above 80hz, I can start to localize the subs. So, 80hz is the ideal compromise for me. My Dirac Live trial ran out today. Not sure what I'm going to do.
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Post by Deleted on May 13, 2014 16:36:39 GMT -5
I've tried Dirac live trial and I am not happy with that. The overall output level from Dirac processor is too low, maybe 6-7 db. I prefer to use REW. Sure it is more complicated than Dirac but the results are much more satisfactory. This old guy tried it and really liked it, lol www.stereophile.com/content/music-round-66 (scroll down)
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Post by Deleted on May 13, 2014 17:06:08 GMT -5
I have been experimenting with a Dspeaker Anti-Mode Dual Core 2.0 for about 2 weeks now. My goal in buying it was to only EQ below the Schroeder Frequency in 2ch audio. However, I have been disappointed in the transparency of this unit. Even in "By Pass" mode audio transparency is affected. Those of you who have tried Dirac Live for PC's have you noticed any affect on your systems transparency? I know I can download the free trial program, but I prefer to ask that question here first.
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Post by rcohen on May 15, 2014 23:06:24 GMT -5
I have been experimenting with a Dspeaker Anti-Mode Dual Core 2.0 for about 2 weeks now. My goal in buying it was to only EQ below the Schroeder Frequency in 2ch audio. However, I have been disappointed in the transparency of this unit. Even in "By Pass" mode audio transparency is affected. Those of you who have tried Dirac Live for PC's have you noticed any affect on your systems transparency? I know I can download the free trial program, but I prefer to ask that question here first. I found that Dirac Live makes the sound more transparent. In my room, it was a little like the difference between listening to sound in a tunnel and listening to sound in an open field. Another way to say it: It's not like listening to a band in your room. It's more like listening to a band on headphones (except the sound isn't coming from the middle of your head.) Also, the ability to quickly experiment with the target curve is awesome.
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Post by Deleted on May 16, 2014 8:46:21 GMT -5
Ref: www.audiostream.com/content/dirac-live-room-correction-suiteQuote by Michael Lavorgna talking about Dirac Live PS - "On the down side, music seemed to lose some dynamic impact. The overall presentation sounded a tad compressed as if some of the vibrancy was gone. Switching the Dirac filter in and out is as simple as a mouse click so I clicked it in and out many times over many songs. This compression was more pronounced on some recordings than others but I found I preferred the sound without the Dirac filter for the majority of recordings especially in terms of tonal balance and dynamic impact. Overall while there was a clear improvement in terms of spatial information and bass response, dynamics and tone seemed to suffer." Later, after talking to Dirac, Michael ran addition positions from which to EQ. He said this made an improvement, but I don't see how this could affect dynamics, and especially tone. I wonder if he really heard an improvement or just conceded? A lot of what you hear is also synergy between components. Had I been using the USB of my W4S Dac 2 I am not sure that I would have noticed a difference using it with DC 2.0, but the FAT DAC is definitely revealing in that area, and there the transparency difference was revealed.
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Post by flak on May 16, 2014 11:03:04 GMT -5
Hello islandman, the reviewer did not concede, I think there are at least two different reasons for his initial sensations... the first is that too small a distance between the measurements points may generate overcorrection. This has been later more clearly explained in a revision of the manual that says the following: "Avoid making measurements in a too small space. Even for the “Chair” listening environment, it is important to spread out the microphone positions in a sphere of atleast 1 meter of diameter. A too small space will result in over-compensation that will sound very dry and dull" the second is that most listeners are accustomed to a specific tonal balance... if you read the review he says: "The frequency response changes, not so much since here it seemed as if some of the life and energy was zapped from my music. Looking at my room's frequency response pre-correction, you can see that it is elevated by nearly as much as 10db around 500kHz which clearly helps account for this meaty and rich presentation" The before correction frequency response was the following: Forgive me now for the following "self quote": "if we look at the frequency response everybody is looking for a "neutral balance" but a perceived neutral balance can be different from a linear frequency response, several listeners for example very much like the sound of the LS3/5A which feature a willingly modified frequency response (you may google about the BBC dip) and in general many speaker manufacturers have applied some sort of "voicing" which gives their unique character to their products. Listeners have often spent many years in selecting the speakers of their liking so it is no coincidence that they prefer that specific tonal balance in their room with their recordings" So I think that the second reviewer's opinion after some time spent in tweaking is the accurate one: "Gone was that obvious sense of dynamic shrinkage, the shut-in off-axis response, gone also was that loss of virility in the tone department but, happily, all of the pluses I enjoyed in the "Chair" measurements remained—improved spatial specificity, improved bass response, and now I was even hearing into the recording in a very pleasant and musically engaging way. It was, in a word, all good" b.t.w. I'm saying the above also to explain that you may prefer a target curve that is different from the default proposed one and that there is nothing wrong with that... that's the reason why the target curve can and must be easily modified. Ciao, Flavio
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Post by sme on May 17, 2014 1:46:58 GMT -5
Yes. I would argue proper target curve selection depends a lot on the room and speakers.
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Post by Deleted on May 17, 2014 10:51:12 GMT -5
Thanks for your time to post Flavio. I will certainly give Dirac a listen. It may somehow be different than the other DSP room control systems....I hope so. So far I have experience with (Multi-EQ, Trinnov, Dual Core 2.0) and they all affect transparency at least to some degree although they are rated "Class A" by most. My experience, like with many, has always been with RC's transparency issues, not it's ability to eq, and compensate for room modes, speaker balancing, etc. E.g., with my W4S DAC 2 there seemed to be an immediate improvement in eq, and in the spatial dimension between instruments. At first my impression was WOW! I now see what all the fuss is about on these W4S Dacs! (And after all, this DAC was given the highest of marks by many reviewers!) But in subsequent listening something that was not quite so obvious at first slowly crept up on me. At first, my overall listening perceptions were distracted by the "new and improved" sound....or at least this is what I thought I was experiencing. But after the "dust settled" after a few days I noticed that even with all these other improvements by this DAC there was a slight flattening, a slight hint of gray overshadowing the sound. This lack of transparency was present with this DAC regardless of me using it tying it with Trinnov (Sherwood) or Harman's own RC (HK AVR 354). Later I was disappointed to learn that owners of the W4S DAC 2 said the best way to listen to the W4S via USB was to purchase an Audiophilleo 1 (600.00)....and that this is necessary to get the best sound and remove this transparency issue of "digital gray" from this DAC.
In a close parallel to my experience with the W4S DAC 2 is the transparency issue with RC in general. Just a quick Googling of this subject will bring up volumes about it. Trinnov with it set to the Auidophile 1 (eq's below the Schroeder frequency only) has been the most transparent sound in my system.
RC control is of course a individual thing, and it's affect on transparency, and the awareness of it has a lot to do with the audio components in the chain IMHO. In 2ch audio especially, it is easy to test the transparency of any RC system by an A/B test of switching it on/off. Not eq, or mode control, or spatial effects (sound stage, speaker balance, etc.) but overall clarity and transparency is what I am listening for.
I think I will try downloading the free Dirac Live Stereo PC version and see how that goes. Hopefully, it is unique in passing the transparency issues.
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klinemj
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Post by klinemj on May 17, 2014 11:08:01 GMT -5
If you do, please report on it here!
Mark
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Post by Deleted on May 17, 2014 11:12:46 GMT -5
Dirac tarath! Dirac tarath!
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Post by rcohen on May 17, 2014 21:21:03 GMT -5
Thanks for your time to post Flavio. I will certainly give Dirac a listen. It may somehow be different than the other DSP room control systems....I hope so. So far I have experience with (Multi-EQ, Trinnov, Dual Core 2.0) and they all affect transparency at least to some degree although they are rated "Class A" by most. My experience, like with many, has always been with RC's transparency issues, not it's ability to eq, and compensate for room modes, speaker balancing, etc. E.g., with my W4S DAC 2 there seemed to be an immediate improvement in eq, and in the spatial dimension between instruments. At first my impression was WOW! I now see what all the fuss is about on these W4S Dacs! (And after all, this DAC was given the highest of marks by many reviewers!) But in subsequent listening something that was not quite so obvious at first slowly crept up on me. At first, my overall listening perceptions were distracted by the "new and improved" sound....or at least this is what I thought I was experiencing. But after the "dust settled" after a few days I noticed that even with all these other improvements by this DAC there was a slight flattening, a slight hint of gray overshadowing the sound. This lack of transparency was present with this DAC regardless of me using it tying it with Trinnov (Sherwood) or Harman's own RC (HK AVR 354). Later I was disappointed to learn that owners of the W4S DAC 2 said the best way to listen to the W4S via USB was to purchase an Audiophilleo 1 (600.00)....and that this is necessary to get the best sound and remove this transparency issue of "digital gray" from this DAC. In a close parallel to my experience with the W4S DAC 2 is the transparency issue with RC in general. Just a quick Googling of this subject will bring up volumes about it. Trinnov with it set to the Auidophile 1 (eq's below the Schroeder frequency only) has been the most transparent sound in my system. RC control is of course a individual thing, and it's affect on transparency, and the awareness of it has a lot to do with the audio components in the chain IMHO. In 2ch audio especially, it is easy to test the transparency of any RC system by an A/B test of switching it on/off. Not eq, or mode control, or spatial effects (sound stage, speaker balance, etc.) but overall clarity and transparency is what I am listening for. I think I will try downloading the free Dirac Live Stereo PC version and see how that goes. Hopefully, it is unique in passing the transparency issues. Some suggestions to get a more "transparent" sound: 1) Start with a target curve close to your speakers' natural curve and tweak from there, rather than starting with the Dirac target curve. 2) Limit the correction window to within the limitations of your speakers' frequency response. When I tried to flatten (boost) the very top and bottom end to push my speakers, I didn't like the sound, and preferred to leave that uncorrected. 3) You can use the UI to zoom in to make fine tweaks to the curve. It can be very sensitive. 4) Keep your favorite variants and do lots of A/B testing with some test songs that you know well.
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Post by sme on May 18, 2014 17:26:15 GMT -5
If it is working correctly, room correction should improve transparency. Unless a room is extensively treated, it does more harm to the sound than most speakers or electronic equipment. Most people have never seen the true frequency response (without smoothing) at their listening positions and have no idea how severely the room affects the response. Many of us are conditioned to the sounds of our own rooms, and it may take a long time for us to break that conditioning.
The trouble is, automated room correction is still a fairly new technology. Both the filtering capabilities and automation logic differ in quality between implementations (i.e., Audyssey, Dirac, Trinnov, etc.). The performance of all of these depends on the measurement techniques used. Any problems with the measurements will adversely impact the result. Any room correction system that attempts to auto-correct on the basis of one measurement is likely to perform poorly, and when multiple measurements are required, the results depend a lot on the choice of measurement positions. Which positions and target curve are best depends on the room, speakers, and end goal. Lastly, there are some room problems that a correction system simply cannot fix.
Over the years, I have obtained very satisfactory results using Audyssey MultEQ, but I have also had it perform very poorly in some instances. Most of the time, this was due to poor technique on my part. Some examples of good practices for Audyssey and probably most other room correction systems are:
* Use a tripod for the measurement mic. * Don't measure too close to any walls. (Try for at least 24", ideally.) * Run measurements in as many different positions as possible. * Concentrate measurements in the area where you typically listen. * Know your speakers. * Don't measure too far off-axis. * Use the target curve to fine-tune the response after checking results at multiple listening positions.
The part about knowing your speakers is important. My speakers have significantly reduced treble output far off-axis, and if I measure there, the resulting sound will be too bright. Similarly, if my target curve is chosen to equalize to a flat steady response, the sound is too bright because in the direct sound, the treble is hot relative to the mid-range. The high frequency roll-off and "mid-range compensation" in the default "Audyssey" target curve may work well for most speakers, but they don't work well with mine. I use a custom target curve in Audyssey Pro and normally run in "Audyssey Flat" mode. In Audyssey Pro, I choose the most aggressive "SMPTE" roll-off for the regular "Audyssey" mode, which I only use when playing movie releases with older theatrical tracks mixed using the X-curve that are otherwise extremely bright (very rare).
After running a set of measurements, I usually have to make minor tweaks to my target curve because I don't always measure in exactly the same places. My primary goal is to avoid fatiguing sound. I use both the steady frequency response and a short-time windowed frequency response to tune my speaker target curves individually. I aim to ensure that the treble (> 4 kHz) is at or below the mid-range in the short-time response at all listening locations. I also aim to keep the upper mid-range (2-4 kHz) at or below everything else in both the short-time and steady responses at all listening locations. I tend to allow a gradual rise from the upper bass to the mid bass (160 or so down to 50 Hz) at the main listening position to make up for the diminished bass outside the sweet spot and especially outside the main listening area. The resulting curve is hardly flat but sounds good (in my opinion) and meets my goals for the room.
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Post by pipeman on Jul 14, 2014 7:40:48 GMT -5
I find I enjoy 2 channel most without any correction and use my sub LF set to 50. When watching movies I turn the YPAO room correction back on. Pathetic I know but my speakers play nicely into the upper 30's so it works for me.
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