One can work fine even if two are better; I used one first, then 2.
31 responses Add your response
Thanks for the responses. I have somehow expected the answers. I need to justify the cost of adding another sub as two subs would cost more than the main speaker itself. That will come later. I'm currently using one sub with a bookshelf speaker with great success. Integration was pretty much seamless. Going to consider some higher quality monitors and add another sub later when opportunity beckons. Need to figure out how to add another sub into the system though as the preamp has run out of inputs.
IME, two subs typically yield better results, because two subs usually provide smoother in-room frequency response.
Having said that, I use a single sub for 2 channel listening, and it sounds great. But part of the reason is that I am able to (1) EQ the sub and (2) control the delay on the sub and the mains, so that the two are time aligned to within about a millisecond.
Regardless of whether you use one sub or two, I believe that it is essential to SQ that the sub(s) are time aligned with the mains, either through positioning, or through the use of delay. This is my opinion, and it is a somewhat controversial one, as you can read on another thread. If you cannot position/delay BOTH subs so that they are time aligned, but you can time align a SINGLE sub, then personally I would go with a single sub.
My local Hi Fi store carries the Focal Utopia line and my absolute favorite listening experience is the Diablo speakers with a JL Audio subwoofer. I think it was the F113.
I know that had one customer that matched the Diablo's with the Gotham G213 but I don't think I've actually heard that combination.
I have heard the G213 with the Grande speakers and that was pretty great, but not worth the difference in price since I actually like the Diablo sound better at any price.
Ryder, I have a two channel system with one sub. I agree with Byron that it is important to blend the sub with the mains with respect to frequency cut-off, phase and loudness. My sub is a Paradigm Signature Servo, which is self powered and has separate controls for frequency cut-off, phase and loudness.
I don't know how to optimize time delay (phase) as close as Byron (within one miilisecond) other than just by using my ears to minimize phase cancellation at the cross over point. As regards FR, my mains start to roll-off at 40-50Hz, so I set my sub cut-off frequency at 34-40Hz. So . . . after phase response/time delay and frequency cut-off is adjusted as best as I can, the only adjustment remaining is loudness. I generally don't fool with loudness very often, but may based on source material.
FWIW, using one sub in a two channel system presents two issues in my rig. The first issue relates to summing the stereo signal from the pre into mono for the sub. That topic has generated a lot of discussion is several threads that I participated in, e.g., impedance matching and asymetrical loading of the pre's balanced and SE outputs.
The other issue relates to the placement of the sub. I don't have a lot of placement options in my basement. However, I am able to place the sub in line with the fronts of the Mains and I aim the sub at my left ear. As a result, it's not to difficult to adjust phase. Even though it's hard to hear the directionality and stereo effect of very low frequency sound, I can actually feel (though not hear) high SPL in my left ear. A cheap SPL meter I bought from Radio Shack confirms this. There is another thread on A'gon which discusses the concerns of listening to music at sustained high SPLs.
Anyway that's my experience with a single sub in a two channel system. Hope it helps. Good luck.
I'm using a single Rel B3 with my Dyn C1's. The time alignment/phasing is the trickiest part. Took me about a year to get it right. I had thought of adding a second sub but now I personally don't feel the need for it especially now the timing is right. I have my sub setup about 5ft to the right of my right speaker. I could never get the sub to sound right when I tried placing it between the speakers. Nearfield listening is why I guess it just didn't sound right to me.
Consider the trade off; a smallish bookshelf with a set of subs may cost about the same as a larger full range, and the full range, while taking up less room, perhaps, will likely be more coeherent across the entire spectrum.
I tried sub/monitor combos several times over the years and always returned to full range. I found that usually the midrange suffers the most due to the cabinet considerations and size of drivers. However, an exception was the much larger Wharfedale Opus 2-M2 monitor and matching subs. It is a much larger than usual "bookshelf" speaker, which was critical in competing well with its floor standing sibling. You can read it here:
As Jim Smith (Get Better Sound) would say, your budget should be spread over two, not one. Having said that, I have two, it is very difficult to set up two, alone one. Expect to work on it for weeks, maybe months. Both should be driven by a single mono signal. A TRA when used properly will be of immense help. The RatShack meter is useless in setting my subs properly.
I too had this question and found the solution with 2 subs. I use Merlin TSM-mme speakers. The sound is amazing, but pipe organ pedal notes and very low bass were just not going to happen.
I purchased one subwoofer and carefully dialed it in. The payoff was better sound, but the system became somewhat directional. The Merlins did not disappear as well before the subwoofer.
I bought a second sub. I placed one sub next to each speaker stand and carefully dialed them in. Now I get the Merlin magic and low bass. My experience is that 2 subs are better than one. Make sure you use the same model subs and they are very high quality.
The Sumiko subwoofer setup instructions can be downloaded here. According to Sumiko...
The optimal position for a REL is in one of the corners behind the main speakers...
No doubt the folks at Sumiko are knowledgeable, but I respectfully disagree with their advice. IMO, placing sub(s) BEHIND the mains may result in good *frequency* response, but it rarely results in good *transient* response.
The reason is because, by placing the sub(s) behind the mains, as Sumiko recommends, the sound from the sub(s) will arrive at the listening position several milliseconds AFTER the sound from the mains. Hence the sub(s) and the mains will not be time aligned.
Some people say that a few milliseconds of time misalignment isn't audible at low frequencies. My experience, and some scientific research, suggest otherwise.
Adjusting the sub's phase will not solve the time misalignment created by placing the sub(s) behind the mains, since the signal that needs to be delayed is NOT the signal for the sub, but rather the signal for the mains. Adjusting sub's phase will do nothing for that.
The time misalignment created by placing the sub(s) behind the mains CAN be solved by digitally delaying the mains. But judging from the systems on A'gon, very few people have that capability. So the most practical way to time align the sub(s) with the mains is...
On the z axis, place the sub(s) roughly coplanar with the mains.
By placing the sub(s) coplanar with the mains, the sound from the sub(s) and the sound from the mains will arrive at the listening position at close to the same time. Hence the sub(s) and the mains will be time aligned, resulting in better transient response. This is audible as better coherence and PRaT.
Keep in mind that subs typically introduce latency into the low-passed signal. Therefore, if you cannot digitally delay the mains, the sub position that results in optimal time alignment will often be slightly IN FRONT of the mains. This is, of course, the *opposite* of what Sumiko recommends.
Another benefit to placing the sub(s) coplanar with the mains is that, in addition to giving better transient response, it will often result in better *frequency* response, for the following reason: Coplanar sub placement minimizes the destructive interference around the crossover frequency. An explanation for why that is so can be read here.
That brings me to Roscoeiii's question about how to time align the sub(s) with the mains. The easiest method is the following
1. Flip the polarity of either the sub or the mains (but not both).
2. Play a test tone at the crossover frequency.
3. Use an SPL meter to measure the output level.
4. Adjust the sub position (or digital delay, if you have that capability) until you MINIMIZE the SPL at the listening position.
5. When the sub is in place, flip the polarity back so that the sub and the mains are the same polarity.
In steps 1-4, you are essentially maximizing the *destructive* interference between the sub(s) and the mains. In step 5, you are restoring the sub to the correct polarity, which now maximizes the *constructive* interfere between the sub(s) and the mains.
Following this method, the sub(s) should be time aligned with the mains to within about a millisecond or two. You can further improve the time alignment, by adjusting the position of the sub(s) in 1-3 inch increments, searching for the position that results in the best coherence and PRaT. This last step has to be done by ear, so it requires some experience and patience. But it is well worth it, IMO.
You can read more discussion/debate of sub time alignment, transient response, and the limits of human temporal resolution, on another thread.
Hope that helps.
For me, I'd opt for a higher quality sub versus two less costly ones, but that's just my opinion. For example, I'm a Velodyne dealer and to have two Optimum series subs or one DD Plus series sub would be an easy choice for me. I'd go with the Velodyne DD Plus single sub in a heartbeat. The Optimum series is good, but the Digital Drive Plus is in a whole other league. I use a DD-10 Plus and, honestly, two Optimum 10 would not play as deep and the sound quality are in different leagues as you can guess by the price difference. Two Optimum 10 subs would also cost more than a single DD-10 Plus. The original poster is looking at using very high quality standmounted speakers, so in that situation, two lesser quality subs may stand out more than a higher quality sub which would match the quality of the speakers better. Once again, just my 2 cents.
Hi Bryon, great post! I have a practical question. As I mentioned above, I don't have a lot of placement options for my sub. So, my sub is aimed directly at my seating position and sits about 4-5 inches behind the mains.
In addition, I summed the channels into mono for the sub, so I don't think I can reverse polarity as you suggested, or perhaps I still don't understand how the method works. Having said that, my sub has a built in phase adjustment capability which I adjust to achieve the loudest bass when playing music that puts out a lot of bass. The thought is that I am trying to reduce phase cancellation at the crossover point.
Anything else I can do to improve the bass with my sub?? Thanks.
Bifwynne - Glad my post was helpful to you. If you can manage it, it would be worth the experiment to move the sub roughly coplanar with the mains and see if it improves the system's transient response. It will be audible as better coherence and PRaT.
The fact that you sum the channels into a mono signal for the sub doesn't affect anything in the procedure I described in my last post. If your sub has a polarity switch, use it. If it doesn't, just reverse the positive and negative leads on your speaker wire for *both* of the mains. Then the mains and the sub will have opposite polarity, assuming that your sub doesn't invert polarity.
Adjusting the sub's phase can be helpful, but not nearly as helpful, IME, as time aligning the sub. As I mentioned in my last post, if the sub is behind the mains, the signal that needs to be delayed is *the mains*. Adjusting the sub's phase will do nothing for that. If you cannot digitally delay the mains, you can time align the system by placing the sub coplanar with the mains, or possibly a little in front of them (to compensate for the potential latency - i.e. delay - introduced by the sub itself).
To sum up, if you can do it, move the sub so that it's roughly coplanar with one of the mains, then follow the procedure in my last post. That should get you pretty close to time aligned. IME, the results of sub time alignment are dramatic, especially when the sub is "fast" enough to keep up with the mains.
It's been decades since I did research in binaural processing, but I wondered why someone from a university in South Carolina would publish such an article in other than JASA. Some of the contentions about levels of neural processing of temporal information seem astonishing, so I wonder if peer review may have been a problem. I'm also skeptical about the use of non-standard equipment; it makes replication of results difficult. I'm skeptical about your arguments and your source of "scientific' support. As I said, though, I haven't followed psychoacoustic literature for decades.
The link I provided to the paper by Milind Kunchur was published in the Journal of the European Acoustics Association, which is a peer-reviewed journal, according to this page on their website.
I'm skeptical about your arguments and your source of "scientific' support.
Whether Kunchur's science is valid, I cannot say. I am no expert on auditory perception, but even if the limit of human temporal resolution is twenty times higher than Kunchur claims, that would still place it at only 1ms, in which case placement differences of approximately a foot or more would result in potentially audible time misalignment.
What I can say is that, IME, differences in z-axis sub placement are audible down to about 4 inches. That would correspond to about 25µs.
Regarding the science, you are certainly entitled to your skepticism. Regarding my statements about subwoofer time alignment, I would humbly suggest that you experiment with it before concluding that they are incorrect.
Correction to my last post. It should have read...
...even if the limit of human temporal resolution is *two hundred* times higher than Kunchur claims, that would still place it at only 1ms, in which case placement differences of approximately a foot or more would result in potentially audible time misalignment.
What I can say is that, IME, differences in z-axis sub placement are audible down to about 4 inches. That would correspond to about *300µs*.
I was off by an order of magnitude. D'oh! I should not do math in my head at midnight.
In any case, my error actually *underestimated* the degree to which Kunchur's estimate could be off while still making time misalignments of about a foot potentially audible, leaving my argument for the importance of subwoofer time alignment unaffected.
It's true that Kunchur's article wasn't published in JASA, but here is a list of articles that WERE, all on the subject of human temporal resolution:
--B. Leshowitz, Measurement of the two-click threshold, J. Acoust. Soc. Am. 49, 462466 (1971).
---Ronken, D. (1970). Monaural detection of a phase difference between clicks, J. Acoust. Soc. Am. 70, 10911099.
--Henning, G. B., and Gaskell, H. (1981). Monaural phase
sensitivity with Ronkens paradigm, J. Acoust. Soc. Am.
--Plomp, R. (1964). Rate of decay of auditory sensation, J. Acoust. Soc. Am. 36, 277282.
--Penner, M. J. (1977). Detection of temporal gaps in noise as a measure of the decay of auditory sensation, J. Acoust. Soc. Am. 61, 552557.
--Eddins, D. A., Hall, J. W., and Grose, J. H. (1992). Detec- tion of temporal gaps as a function of frequency region and absolute bandwidth, J. Acoust. Soc. Am. 91, 10691077.
And here is Kunchur's synopsis of the conclusions of those JASA articles...
In one experiment by Leshowitz (1971), listeners were presented with a single pulse or two narrower pulses (with the same total energy) separated by an interval Δt. The click and click-pair could be distinguished down to Δt ≈ 10 μs. In this case, the two stimuli have differences in their amplitude spectra and their discernment was explained on this basis. Isospectral variants of this experiment were carried out by Ronken (1970) and later by Henning and Gaskell (1981) where one stimulus consisted of a short pulse followed by a taller one separated by an interval Δt. The second stimulus was a similar pair with the time order reversed and hence had the same amplitude spectrum. The shortest Δt for which these stimuli could be distinguished was about 200 μs. Another type of constant-amplitude-spectrum experiment involves the detection of gaps in noise (Plomp, 1964; Penner, 1977; Eddins et al., 1992). In these the threshold for gap detection was of the order of 2 ms.
As you can see, those estimates of the limits of human temporal resolution range from 2ms to 200μs all the way down to 10 μs. If we average those estimates, we get 736μs, which corresponds to differences of less than a foot.
Hence there is reason to believe that the human limits of temporal resolution may be sensitive enough to make time misalignments of a foot or less potentially audible.
I quit following the literature around 1970. I was an admirer of the Jeffries model of binaural processing, but a study I did using computer generated signals and narrow bands of noise was inconclusive about the phase relationships of the signal and noise even though the detectability of the samples differed markedly, and the models did not explain the results. I reported the results at a special JASA session devoted to Lloyd Jeffries, but never published them. After that, I lost interest in psychoacoustics.
I can believe moving a sub a foot or so can make a detectable difference, but I'm not sure I'd attribute it just to time alignment, given the potential interaction with room boundaries.
I make it a point not to hijack threads. But that is exactly what I've done to this one. My apologies to Ryder (the OP) and to everyone else following along.
Dbphd - Your last observation is a valid one, but I have experiences that are inconsistent with attributing the audible variations resulting from small differences in sub room placement to frequency response. You can read about my experiences on a recent thread I initiated.
I'd be happy to discuss this further, as it is a topic that interests me a lot. But, in the interest of courtesy to other folks, we should move our conversation to the thread I linked above.
One of the most reliable phenomena in psychoacoustics is what Is know as the masking level difference (MLD). Present a mid-freuquncy sinusoid in correlated noise to both ears and adjust the level until it becomes inaudible; flip the phase of the sinusoid in one ear, and the signal pops up as much as 15 dB, depending on frequency. At the time, the data suggested the auditory system doesn't preserve timing as such up the neural chain, but may convey such information by the more central areas excited. Thus, the vector model of Lloyd Jeffries.
I couldn't afford a pair of the subs I wanted...so I went a little upline and bought a single sub.
That's OK, though. My room has enough walls to bounce the sound around and leave no 'hot spots' for bass. That would be 9 walls and a vaulted ceiling. Very asymmetric.
That nasty room and a very low crossover ensures the sub is both invisible and seamless.
Depending on how messed up your room is, that may also work for you.
The manufacturer of my sub recommends a near-field approach to sub use.
BryonCunningham -not a problem at all. I've briefly gone through the thread that you linked over here and can see that you are very knowledgeable in this area in setting up a sub at an advanced level. I'll do some in-depth study on this subject once I managed to digest all the technical stuff that have been elaborated here. Keep up the good work. I'm sure we all appreciate the time and effort you've taken in contributing your thoughts and beliefs on this forum.
To All --reporting back on results of using impedance buffer. Ok, I hooked up Tom Tutay's buffer. As a threshold matter, let me say a couple of things about Tom. He is very professional and knowledgeable -- a real gentleman. He turned around my order in lightening time. The shell of the buffer device is made wholly of metal, and is about 9" x 5" x 2". By all appearances from the outside, the device seems to be very sturdy and well made.
Now . . . as to how it works. As you can see from the posts above, the device was designed to: (a) load my Ref 3 symetrically (i.e., Main 1 and Main 2 in balanced mode), (b) sum the left and right channels without shorting the Mains since I am using only one sub, and (c) raise the impedance level of the two Mains to a level that better matches ARC's design specs.
As to tube life, I take it on faith based on conversations with ARC that by reducing the load on my Ref 3's outputs, tube life will be extended. I will have to wait and see.
As to sound quality, I can say with 100% certainty, my rig does NOT sound worse. Does it sound better?? Yeah, I think so. I think the imaging, detail and sound stage are a little bit improved. The highs seem a little smoother and better refined. But quite honestly, I don't think the change, if any, is day and nighgt. It's just subtle, kinda like a tweek, maybe two or three tweeks.
Now the impact on bass is more noticeable. Although I removed one active artifact from the system and added another, I think the X-30 crossover/controller did more than the buffer because the X-30 could adjust loudness, phase, cut-off and summed the channels. By contrast, the buffer has an active unitary op amp that raises the input impedance and just sums the channels. That's it.
When I first inserted the buffer, I initially thought something happened to the bass output from the sub. On Tom's suggestion, I tried using the old hook-up again by running an SE I/C from one of the SE Mains off the Ref 3 directly to the sub (sans X-30). I then ran the Ref 3 in mono mode to sum the channels. Interestingly, the bass seemed equally lame this way too.
Then I realized what was going on. I had set the cutoff frequency on the sub at 35 Hz to blend into my fronts. What I realized is that there simply wasn't very much going on below 35Hz in my source material. By contrast, when I raised the cutoff frequency -- bass galore.
So realizing what was going on, I then critically listened to some bass heavy source material, e.g., Norah Jones CD and Solti conducting Chicago Symph Orchest., Beethoven 9th 1st and 4th movements LP. Tons of bass in these source materials. But when I raised my sub's cutoff to about 50 Hz I had more than enough tight and smooth bass -- NOT boomy.
In summary, Tom Tutay of Transition Audio Design is a great resource for audiophiles. He is not limited to just buffer devices, but can custom design many different types of devices. Keep him in mind. Tom's phone number is: (850) 244-3041.