I don't believe it. I need more power?

TLS Guy

TLS Guy

Seriously, I have no life.
maybe it produces the same sound above 80hz, and better sound below 80hz.

Holistically, that would be "better sound"
I can believe that it does sound better.

I would use my cap in the lead solution. It will work fine, add no distortion or noise and can't clip. That actually is the ideal solution for what you are trying to do anyway. It is simpler and actually better than any other solution that can be devised for you.

If you don't know how to construct this, then just send me the termination and length specification and I will make the interconnects for you.
 
Irvrobinson

Irvrobinson

Audioholic Spartan
maybe it produces the same sound above 80hz, and better sound below 80hz.

Holistically, that would be "better sound"
Regardless of the specs, the measurements show differences up to just over 100Hz.
 
Irvrobinson

Irvrobinson

Audioholic Spartan
I can believe that it does sound better.

I would use my cap in the lead solution. It will work fine, add no distortion or noise and can't clip. That actually is the ideal solution for what you are trying to do anyway. It is simpler and actually better than any other solution that can be devised for you.

If you don't know how to construct this, then just send me the termination and length specification and I will make the interconnects for you.
Thanks, but I really want Velodyne to fix this. The DD18 Plus is expensive, and it shouldn't overload like this.
 
TLS Guy

TLS Guy

Seriously, I have no life.
Thanks, but I really want Velodyne to fix this. The DD18 Plus is expensive, and it shouldn't overload like this.
Well an awful lot of gear does do this, including mix desks costing millions.

It is a very hard problem to fix and frequently encountered.

I posted this last night, on another thread, with the usual claim, LP is better than digital. Well an LP won't drive any voltage amp to clipping unless designed by a total idiot. We need to move to the digital domain form mic capsule to speaker. If we don't then we will continue with issues like yours and worse.

We need a total change of practice. Here is what I posted and everyone needs to embrace this huge change of gear from professionals to consumers.

We get these sort of claims again and again everywhere. If people are really hearing this, then their systems are really screwed up and not of a standard for modern digital media. That could be, as I surveyed what was on offer in the Best Buy Magnolia section on Monday at Eagan Minnesota. I went through both rooms and every system was totally worthless. Just plain awful. So I think may be the front end of modern digital media are too hot for most systems.

The fact is that for most music for CD does not need compression unless the producer uses it. CD is loss less, and will NOT degrade the ambiance in the original recording. There is no information thrown away, as it is in codecs like mp3, which I agree does degrade the ambient envelope and often severely.

CD has a dynamic range of 93 db, in practice with over sampling it is probably at least 10 db better than that.

LP on the other hand has a dynamic range in the 60 to 70 db range and for most sources requires around a 2/1 dynamic range compression to cut the disc.

SACD and BD have a dynamic range beyond the capability of current analog microphones. The absolute maximal theoretical dynamic range required for any music is 144 db, and most advanced digital media can achieve that with headroom. Analog microphones can not get there and never will, the best current digital microphones can achieve 140 db, and Neumann expect to push the envelope to 144 db shortly.

Right now I'm listening to a 1964 recording of Elgar's symphonic study Falstaff, with the Halle orchestra under the baton of "Glorious John," as Vaughn Williams called him. Now I have that LP, but I'm listening to the CD. The LP is quite incapable of rendering that recording with the dynamic, realism and low distortion of the CD. I can be certain the CD is the same as listening to the original master tape. The LP is short of that by a wide margin.

So where do all these claims like yours come from.

Two weeks ago we had an AES meeting at MPR studios, that was very interesting. Engineers were there from Sennheiser_ Neumann, including an experienced senior engineer from England. It was an interesting presentation on their new range of digital microphones.

This topic you are discussing came up. The overwhelming consensus was that most current domestic and even a lot of studio equipment is not yet adequate for the digital era.

These new microphones convert to digital right at the output of the capsule. An new AES standard has been developed to cope with this. The current AES/EBU of which the domestic version is SPDIF has been deemed inadequate. The new AES 42 standard is now released, but with few early adopters.

AES 42 envisages only 2 conversions. Analog to digital right after the microphone, and digital to analog right in the speaker, with EVERTTHING in the digital domain from microphone capsule to the speakers, with passive crossovers not allowed for. The process will require DSP crossovers and switching class D amps for every pass band in the speaker.

If we are to overcome your perceptions, it seems this is what is required. To get the results required with current approaches is too costly and cumbersome.

This will require a huge change in both professional and domestic practice. So don't bank on this system being your last.

At last I can see the sun starting to set on those awful receivers.

In the next generation most audio equipment in use will seem like the 78 era does to us now. The potential for greatly improved performance is right round the corner now.
 
TLS Guy

TLS Guy

Seriously, I have no life.
Thanks, but I really want Velodyne to fix this. The DD18 Plus is expensive, and it shouldn't overload like this.
Well an awful lot of gear does do this, including mix desks costing millions.

It is a very hard problem to fix and frequently encountered.

I posted this last night, on another thread, with the usual claim, LP is better than digital. Well an LP won't drive any voltage amp to clipping unless designed by a total idiot. We need to move to the digital domain form mic capsule to speaker. If we don't then we will continue with issues like yours and worse.

We need a total change of practice. Here is what I posted and everyone needs to embrace this huge change of gear from professionals to consumers.

We get these sort of claims again and again everywhere. If people are really hearing this, then their systems are really screwed up and not of a standard for modern digital media. That could be, as I surveyed what was on offer in the Best Buy Magnolia section on Monday at Eagan Minnesota. I went through both rooms and every system was totally worthless. Just plain awful. So I think may be the front end of modern digital media are too hot for most systems.

The fact is that for most music for CD does not need compression unless the producer uses it. CD is loss less, and will NOT degrade the ambiance in the original recording. There is no information thrown away, as it is in codecs like mp3, which I agree does degrade the ambient envelope and often severely.

CD has a dynamic range of 93 db, in practice with over sampling it is probably at least 10 db better than that.

LP on the other hand has a dynamic range in the 60 to 70 db range and for most sources requires around a 2/1 dynamic range compression to cut the disc.

SACD and BD have a dynamic range beyond the capability of current analog microphones. The absolute maximal theoretical dynamic range required for any music is 144 db, and most advanced digital media can achieve that with headroom. Analog microphones can not get there and never will, the best current digital microphones can achieve 140 db, and Neumann expect to push the envelope to 144 db shortly.

Right now I'm listening to a 1964 recording of Elgar's symphonic study Falstaff, with the Halle orchestra under the baton of "Glorious John," as Vaughn Williams called him. Now I have that LP, but I'm listening to the CD. The LP is quite incapable of rendering that recording with the dynamic, realism and low distortion of the CD. I can be certain the CD is the same as listening to the original master tape. The LP is short of that by a wide margin.

So where do all these claims like yours come from.

Two weeks ago we had an AES meeting at MPR studios, that was very interesting. Engineers were there from Sennheiser- Neumann, including an experienced senior engineer from England. It was an interesting presentation on their new range of digital microphones.

This topic you are discussing came up. The overwhelming consensus was that most current domestic and even a lot of studio equipment is not yet adequate for the digital era.

These new microphones convert to digital right at the output of the capsule. An new AES standard has been developed to cope with this. The current AES/EBU of which the domestic version is SPDIF has been deemed inadequate. The new AES 42 standard is now released, but with few early adopters.

AES 42 envisages only 2 conversions. Analog to digital right after the microphone, and digital to analog right in the speaker, with EVERTTHING in the digital domain from microphone capsule to the speakers, with passive crossovers not allowed for. The process will require DSP crossovers and switching class D amps for every pass band in the speaker.

If we are to overcome your perceptions, it seems this is what is required. To get the results required with current approaches is too costly and cumbersome.

This will require a huge change in both professional and domestic practice. So don't bank on this system being your last.

At last I can see the sun starting to set on those awful receivers.

In the next generation most audio equipment in use will seem like the 78 era does to us now. The potential for greatly improved performance is right round the corner now.
 
Irvrobinson

Irvrobinson

Audioholic Spartan
Latest update: I talked to Velodyne and they're looking into the input overload problem. While I'm waiting, I decided to spend a good bit of the last weekend measuring with the OmniMic and optimizing the speakers. Running the mains full-range without the sub initially showed the same old 55-65Hz room mode peak, which was so nicely subdued by the low-pass / high-pass filter combination I'm using. (Part of the problem here *might* be my 22.5x16.5 room... those dimensions are in the vicinity of 50-65Hz wavelengths.) Same sub-40Hz suck-out too.

So I started micro-positioning the mains and tamed that 60Hz spike by 5db out of ~10db. Believe it or not it only involved moving the mains three inches closer to the back wall and one inch closer together. I can't discern any difference in imaging. I'd love to see those sound waves. Now the suck-out starts 5Hz higher at 45Hz and is deeper. Adding the sub, I adjusted the equalization to flat, with a 40Hz low-pass filter knee frequency and a 36db/octave slope. I spent about 90min micro-positioning the sub, and I tried 12db and 24db filter slopes too. What a PITA. Now the sub's position is slightly forward of the mains, and slightly offset to the right rather than the left.

The OmniMic now says I'm roughly back to where to I started, smoothness-wise, to within +/- 5db at 1/12th octave smoothing across any two octaves, with the 20Hz-125Hz range running about 7db hot compared to the 10KHz-20KHz octave.

The lowest octaves seem to sound smoother, which doesn't surprise me a whole lot. The Salon 2s are rather potent in the 20Hz-40Hz octave, so running them full-range with the DD18+ is a bit like having three subs. The improved bass smoothness is especially noticeable in any listening position beyond the sweet spot I optimize for.

The ATI peak indicator light doesn't light up at all now, for any volume I can stay in the room for. I'm not bi-amping any longer, so the AT3005 is being used as a 3002. Whatever.

My learning from all of this that if you really want your system to sound great you've got to put a lot of effort into speaker positioning and invest in good measuring equipment. A couple of inches one way or another makes a measurable difference. You can compensate with other strategies, which is pretty much what I was doing with a high-pass filter and equalization, but now that it's done I'm glad I figured out the problems rather than put bandages on them.

It'll still be interesting to see what Velodyne eventually concludes.
 
GranteedEV

GranteedEV

Audioholic Ninja
Now for our bored pleasure, tell us if plugging the port on the salon 2s makes any audible/measurable difference. ;) :D

And get a DEQX
 
Irvrobinson

Irvrobinson

Audioholic Spartan
Now for our bored pleasure, tell us if plugging the port on the salon 2s makes any audible/measurable difference. ;) :D

And get a DEQX
Now that's an interesting test... how dense does a foam plug need to be to work properly?

I'm still thinking about the DEQX.
 
GranteedEV

GranteedEV

Audioholic Ninja
Now that's an interesting test... how dense does a foam plug need to be to work properly?
*shrug*

I just ripped off the foam from the cushions of an old sofa, and then shoved in the port of my speakers. It was effective enough to affect the frequency response as predicted.
 
A

awdio

Audioholic Intern
Why are you surprised?

Lets take a look and see where the problems come in a four way speaker like that with passive crossovers.

I can easily run my friends 400 watt per channel Macs out of gas driving his B &W 800 Ds.

Let us just see why you actually need 3000 to 4000 watts, but that if you provided it your crossovers would smoke.

Now you have a speaker that has a sensitivity of 86.4 db 2.83 volts I meter.

At 90 Hz the impedance of your speaker is 3.7 ohms and it does not go significantly above four ohms until 600 Hz. So it is a low impedance speaker where the majority of the power is.

So your speakers will draw 2.16 watts at 90 Hz from the amp to produce 86.4 db at I meter, versus 1 watt if it were 8 ohms. So the low impedance in the range where the power is has doubled the power requirement for your speakers versus 8 ohm ones.

Now it gets worse, your speakers have fourth order crossovers, and at a low frequency. Your insertion loss is1 db per order under the best of circumstances, assuming the use of the highest quality inductors. Since the crossover is far lower than can be recommended in a passive design I suspect you power loss is around 5 db and may well be as high as 6 db.

So, slightly over half your amp power is heating up the crossover.

Now lets take a look at an active situation. Lets take my speakers.

The bass units are each 8 ohm, since there are two drivers of 8 ohms, each directly coupled to an amp sensitivity is 93 db 2.83 volts 1 meter and that will take one watt of power. You will have to provide at least 8.4 watts of power to achieve the same spl, and more likely 10 watts.

Now even with that situation I still need to provide 750 watts per speaker, to keep things relaxed at concert levels.

That is why good monitor companies like ATC and PMC provide about 3 to 4 KW per speaker and that is with active crossovers.

Billy woodman showed me that most amps driving a passive three way speaker of average sensitivity are in almost continuous recovery mode.

What is required is active speakers with wide bandwidth drivers of good sensitivity.

I no longer believe it is possible to power a three way speaker to concert levels without horn drivers.

The time is now to ditch passive crossovers.
You don't have a clue of what you are speaking.
 
Irvrobinson

Irvrobinson

Audioholic Spartan
While awdio is just being argumentative, I did enjoy re-reading this nearly 7.5 year-old thread. I still have the same amp, speakers, and subwoofer, but the DAC-preamp is now the Benchmark DAC3 (after a misstep with an Oppo),but I've moved and I'm in a completely different room which is much more advantageous for an audio system. The set-up and adjustment of the system is also quite different.

In the house I was living in when I posted the thread I used a room that was very problematic for smooth frequency response at the listening seat. When I was shopping for speakers in 2009, I listened to so many alternatives. I was using a pair of Legacy Audio Focus at the time, and they sounded good, but by 2009 they were not good enough compared to the competition. (The Focus was also not voiced for accuracy, and had a classic saddle-shaped frequency response. Accuracy became paramount to me by then.) Then I heard the Salon2s at a dealer with what must have been a nearly optimal set-up, and I fell absolutely in love. I spent hours listening to many kinds of music on them, I was enthralled, and ended up buying a pair.

Unfortunately my room at the time was nowhere near as good as the dealer's, especially below about 100Hz, and I spent the next two years fiddling around with placement to no avail. My room just sucked. Later in 2009 I decided to take matters into my own hands, and I came up with a staggering brain fart of an idea to use a pair of passive subs with equalization I would tune myself. In preparation I swapped out the amps I had been using and ordered an ATI AT3004 to power the subs and the Salon2s. The AT3004 was turned into a 3005 for no extra charge by ATI when I ordered the amp, because they ran out of AT3004 back panels. How could I say no?

The subs I was considering were a custom-made pair using the Linkwitz Thor design, and I was looking into various crossover / EQ electronics options, but my high intensity job postponed action. A speaker builder friend of mine back then convinced me my concept was overly complex and expensive. He made the case that I should instead get a pair of the latest powered subs with built-in PEQs, and then I'd have many tuning options. He also recommended getting one sub at a time, because one might be enough, depending on the room. He was very impressed with the new 2010 Velodynes, and I was impressed in the local dealer demo, so I ordered a then brand-new DD18 Plus. I wasn't in the mood to fiddle with ID vendors back then either, and the sub market wasn't as advanced as it is today. I also ordered a Parts Express OmniMic, and a lot of my free time was spent fiddling with the system.

The DD18 Plus was very impressive in my room in test tones, but the combination of Salon2 placement, sub placement, crossover frequency, and PEQs was pretty overwhelming. I finally got the Salon2s to sound okay by running them full-range driven by one ATI channel each, and I used the Velodyne and it PEQs with a 100Hz low-pass filter to just fill-in the suck-outs in the Salon2s in-room response at my listening seat. Yes, three ATI amp channels sat idle. "Okay" in this case meant very good by most people's standards, but the Salon2s still didn't absolutely blow me away like they did at the dealer. My room really did suck, and apparently no amount of tuning was going to fix that. Since I wasn't completely thrilled I wasn't ordering another DD18 Plus either, because I figured the Salon2s weren't a permanent solution.

The point of this thread, the clipping of the ATI amp on a rock CD, did indeed turn out to be a Velodyne design flaw that applied only to the XLR inputs. The single-ended inputs had gain controls on them, but the XLR inputs didn't and the low gain of the ATI amp was requiring output settings on the Benchmark pre-amp that overloaded the XLR inputs on the Velodyne amp. Velodyne had clearly assumed higher-gain mains amplifiers for the DD Plus designs. When I noticed this I was experimenting with using a high-pass filter for the Salon2s which was built into the subwoofer, so the distortion from the sub was passed on to the ATI channels, which use a comparator circuit to recognize clipping. Hence the clipping light.

The clipping problem was solved two ways. First, as I mentioned, I decided to run the Salon2s full range, which eliminated the circuitous signal path through the sub. Second, since the sub inputs would still overload, I inserted 10db XLR attenuators into the subwoofer input cables, which allowed the ATI to get the volts it needed, and allowed the Velodyne to operate in a very comfortable gain range. Velodyne blew me off when I discussed the flaw with them, even though their support guy agreed with me that it was a design flaw. Whatever. The DD18 Plus operates flawlessly otherwise to this day, and Velodyne appears to be far less serious about the subwoofer business anyway, as owner David Hall focuses on Velodyne LIDAR. In other words, this is almost certainly my last Velodyne product.

The ultimate solution was really a different house with a better audio room, which happened in 2015. Essentially the same set-up, now tuned much differently, with the Salon2s running full range and a single DD18 Plus fed with a full-range signal using a 100Hz low-pass filter and five PEQs (of the eight total) adjusted for fill-in of the Salon2 in-room response. Numerous hours of measurement, tuning, and placement experimentation resulted in very smooth and tight bass down to 20Hz at my listening seat. The resulting sound at least as good as I remember at the dealer's years ago (using many of the same recordings). Imaging and mid-high smoothness and realism were now exemplary. I flirted with a few alternative speakers, mostly electrostatics and the KEF Blades, but nothing seemed anywhere near worth the huge expense, so I've stuck with the Salon2s. The DD18 Plus has been out-classed by several more recent subs, but the truth is that with the Salon2s running full-range I have the equivalent of three subs, and the Velodyne doing fill-in only isn't breathing very hard, even when I'm getting ambitious with the volume levels. The system is music only, so HT/action movie requirements don't apply.

I will confess that I do currently passively bi-amp the Salon2s. With the Legacy Focus passive bi-amping did make an audible difference, IMO, because of its idiosyncratic design. The Salon2s are an easy load by comparison, so for two years I was running the Salon2s single-amped. But a while back, in a thread on bi-amping, Alex2507 dared me to try bi-amping again with the Revels, and on a boring weekend day I did, since I had all of the cables just sitting in a box in a closet, and I had two spare amp channels just idling. There was no audible difference I could discern at all with any material, including a well-made recording of Respighi's The Pines of Rome. Because the cabling changes are actually a pain in the butt, or the knees in my case, I left the bi-amp configuration in place for a time, since it was doing no harm. While I still had the Salon2s bi-amped, a friend came over, an electrical engineer I worked with, and he gave me a precious lecture on how stupid passive bi-amping was. I so enjoyed the lecture I left the configuration bi-amped, and I later enjoyed two other lectures, one from a high-energy physicist, so I made the passive bi-amping permanent, and even purchased fancy Blue Jeans speaker cables to make it all pretty. I haven't had one of these lectures recently, so I'll have to invite a physicist or two over to tell me how stupid I'm being. (Yes, my behavior is probably indicative of some personality flaw, but psychotherapy costs money and takes time, so the hell with it.)
 
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Kvn_Walker

Kvn_Walker

Audioholic Field Marshall
Love your reverse psychology! Next time you get lectured, you should bi-wire from your bi-amp setup! 4 pairs of speaker wires to each speaker should be enough to make their heads explode!
 
RichB

RichB

Audioholic Field Marshall
I will confess that I do currently passively bi-amp the Salon2s. With the Legacy Focus passive bi-amping did make an audible difference, IMO, because of its idiosyncratic design. The Salon2s are an easy load by comparison, so for two years I was running the Salon2s single-amped. But a while back, in a thread on bi-amping, Alex2507 dared me to try bi-amping again with the Revels, and on a boring weekend day I did, since I had all of the cables just sitting in a box in a closet, and I had two spare amp channels just idling. There was no audible difference I could discern at all with any material, including a well-made recording of Respighi's The Pines of Rome. Because the cabling changes are actually a pain in the butt, or the knees in my case, I left the bi-amp configuration in place for a time, since it was doing no harm. While I still had the Salon2s bi-amped, a friend came over, an electrical engineer I worked with, and he gave me a precious lecture on how stupid passive bi-amping was. I so enjoyed the lecture I left the configuration bi-amped, and I later enjoyed two other lectures, one from a high-energy physicist, so I made the passive bi-amping permanent, and even purchased fancy Blue Jeans speaker cables to make it all pretty. I haven't had one of these lectures recently, so I'll have to invite a physicist or two over to tell me how stupid I'm being. (Yes, my behavior is probably indicative of some personality flaw, but psychotherapy costs money and takes time, so the hell with it.)
I'd rather confess to bi-amping than my global warming sins :p
This is the method that I find works best to compare bi-amped to single amped:

- Listen to a single speaker (the Harmon Method)
- Split the channels using a Y connector (This avoids DSPs involvement on some processors)
- Use stacking banana plugs on one of the pairs
- Create a small (6 inch) cable the patch from the stacking bananas
- Have a friend change them (you should be able to SBT) and change in a few seconds

I'm still holding out hope for you :p
My hypothesis for bi-amping gains reduces distortion by removing the interaction between the crossovers.

Recently, I decided to figure out how much power I actually use, primarily because the AHB2s are only 100 WPC into 8 Ohms and 180 WPC into 4 Ohms and I have a choice of bi-amping or bridging to about 380/500 WPC into 8/4 Ohms.

I measured 2.83 volts playing stereo 400 Hz and 1kHz tones at my 11 foot listing position.
The Salon2's are basically 4 Ohm speakers so this SPL requires 2 watts.

Here are the results I obtains. I am using much less power than predicted by on-line power calculators.
Measured SPL at Listening Position.jpg


I never illuminate the clip indicators on either channel (bi-amped if course) with music or movies and now that I see this chart I understand why.
With Aquaman at -16 I could not clip this amp and the Salon2 woofers were really moving. It was way too loud for me. I estimate that -20 to be equivalent the experience in a loud cinema.

Some may take this to mean that an AVR amp is all that is required for this system but that is not the case.
These amps double down and have infinitesimal distortion while doing it.
To me, this emphasizes the critical importance of the first watt.
Many amps distortion specifications are at maximum rated power, have far worse THD+N at low power.
If you have looked at a Sound and Vision amplifier THD+N chart, basically you are likely listening to the distortion and noise that is shown by that red line that crawls up the wall. :p

- Rich
 
Irvrobinson

Irvrobinson

Audioholic Spartan
If you have looked at a Sound and Vision amplifier THD+N chart, basically you are likely listening to the distortion and noise that is shown by that red line that crawls up the wall. :p

- Rich
If I understand what you're saying correctly, on many THD graphs the curve rises as you get down to the milliwatt range because noise is a fixed voltage level, and in a low number of milliwatts noise swamps distortion, so the THD curve rises. Is this what you're referring to?
 
Irvrobinson

Irvrobinson

Audioholic Spartan
I'm still holding out hope for you :p
Unlikely with the Salon2s. I chose the Respighi piece because strings and the rest of the orchestra play at the same time as ~20Hz pipe organ fundamentals. This combination should maximally stress a single amp configuration and maximally de-stress the mid-high section when bi-amped. In my pretty large room at high volumes, no difference I could discern. Of course, the ATI channels are ~450 watts into 4 ohms. Perhaps on a lower power amp there *might* be a difference, but I'm still a skeptic.
 
RichB

RichB

Audioholic Field Marshall
If I understand what you're saying correctly, on many THD graphs the curve rises as you get down to the milliwatt range because noise is a fixed voltage level, and in a low number of milliwatts noise swamps distortion, so the THD curve rises. Is this what you're referring to?
It could well be noise or other distortion at or below 1 watt.
- Rich
 
RichB

RichB

Audioholic Field Marshall
Unlikely with the Salon2s. I chose the Respighi piece because strings and the rest of the orchestra play at the same time as ~20Hz pipe organ fundamentals. This combination should maximally stress a single amp configuration and maximally de-stress the mid-high section when bi-amped. In my pretty large room at high volumes, no difference I could discern. Of course, the ATI channels are ~450 watts into 4 ohms. Perhaps on a lower power amp there *might* be a difference, but I'm still a skeptic.
I listen at 80 dB or lower with content with natural instruments and female vocals.

Edit: For example, I have used Celtic Woman Songs from the Heart in SBT's.
https://www.amazon.com/Songs-Heart-Celtic-Woman/dp/B002UZXJA6/ref=sr_1_7?keywords=celtic+woman+cd&qid=1568981819&s=gateway&sr=8-7

- Rich
 
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