AudioCircle
Industry Circles => GR Research => Topic started by: HAL on 31 May 2015, 07:48 pm
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I have been asked by folks if a digital crossover for the LS-9 and LS-6 speakers would be possible. The answer is yes, but would require some rework of the existing speaker to accept direct connection to the drivers for biamping the speakers. Each amp would be driven by a separate channel of the dspMusik digital crossover.
Danny has given me an LS-9 crossover to measure to get the parameters to design a dspMusik crossover version. I have CLIO here to do the electrical measurements to make a digital version. If there is interest, an LS-6 version would be done the same way. If LS-6's are of interest, I can see if Danny has a crossover board to measure as well.
I can now make the dspMusik a stand alone device that has a PC USB driver that is ASIO compliant so that it will also work with a MAC. This has been tested with two different MAC Mini systems and has worked very well. The MS-3 PC Music Server has worked very well from the other side.
This thread is to gauge interest in this as a project. If people are interested, please let me know in this thread. I have production dspMusik digital crossovers available and can start the project with the LS-9 crossover in house.
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I might have been interested, but I've already had the Ninja build me a pair of external crossovers:
(http://www.audiocircle.com/image.php?id=87581)
(http://www.audiocircle.com/image.php?id=87580)
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skiman,
I have heard the dspMusik in comparison to Danny's design of the passive crossover on the latest OB three way planar speaker in his system, and to me the dspMusik is another step up in sound quality. He did the comparison with his MAC Mini and his music and
It has to be used in a biamp setup with each amp driving each array of drivers directly in order to accomplish that goal. You already have that as does S Clark with his LS-9's since you both have external crossovers.
S Clark was one person asking about it, so making it available to him as a test case is probably a good starting point if he is interested.
I also have Ninja Nobel crossovers for my N2X speakers.
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Hi Rich, I was wondering if you were going to try to do something with the LS's. I still use Danny's crossover in my 9's, but I upgraded those with Sonicaps. I was be interested to hear about the results.
Mark
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Mark,
Good to hear from you!
Will keep everyone posted.
I do not have either of the LS-9's or LS-6's, so will have to work with someone that does and wants to mod them or are already moded.
Rich
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Hmmmmm...........
I'm interested from a theoretical perspective at this point. Could be interesting. I always thought bi-amping with SS bottom and tube top was the way to go. This makes that possible.
Put me down as "following with interest" at this point - I still have to figure out if the LS 6's will even work in the new house - might know in a couple of months.
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When we tried the dspMusik on Danny's new OB planar line array's, that is exactly what we did. Gary's 30 watt tube monoblocks on the NEO3's and Class D amp on the NEO10's with the servo subs. :thumb:
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I have LS-9s. First I've heard about a Ninja Nobel crossover. How can I find out more?
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Skiing Ninja made them.
Sadly, they are no longer in business.
http://www.skiingninja.com/ (http://www.skiingninja.com/)
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Thanks, HAL. Sad indeed. I wish them well.
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I'll be having some guys over this summer (or maybe fall) to compare my LS-6 speakers with the following:
1. GR-Research crossover with Sonicap Platinum bypass caps
2. Response Audio crossover with Clarity Caps
3. Digital crossover using JRiver Media Center
The only difference in the playback chain for digital will be the removal of the passive crossovers and the addition of another pair of Digital Amp Company amps.
With the digital crossover I'll also be comparing a two way system to a three way system by integrating a separate infinite baffle subwoofer for the left and right speakers.
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What two stereo DAC's are you planning to use with the JRiver crossover?
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Only SoCalWJS is interested in feedback?
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What two stereo DAC's are you planning to use with the JRiver crossover?
I've tried multiple stereo DACs. They work fine if using a master clock like an AES card or daisy chaining with S/PDIF, MADI, Firewire, Thunderbolt, ADAT Lightpipe, or Ethernet. When daisy chaining, the first device still needs to be connected to the PC with a multi-channel format (USB in addition to previously mentioned methods) so you can't use SPDIF. You can also connect work clocks like Mytek does with multiple Stereo 192-DSD DACs. You won't get any drift between channels with any of these methods.
However, I prefer to use multi-channel DACs. I feel the best for this purpose is a MOTU 1248, which I have. I like it for the following reasons:
- +20 dBu line output max level - It has enough voltage on the output to account for headroom lost with digital crossovers. You can also use with low gain amps for the highest SNR possible.
- Dual 8-channel DAC chips - one can be used for the left speaker and one for the right for no chance of crosstalk.
- It is fully balanced - There are 12 balanced outputs and you can route up to 64 channels out through ASIO and can assign any ASIO outputs to any of these 12 physical outputs.
- Output Trim range of 24 dBu (+20 dBu down to -4 dBu) - Even though the maximum output is 20 dBu, you can lower the maximum output level on each channel using 32-bit digital trim levels. This means you can set the tweeter levels and never risk damage. You can also use it to make sure you don't ever overload the input on a small amp.
- 32-bit output for highest quality volume control
- No noise on sample rate changes - Sample rate changes are completely silent! No slight tick, pop, or anything else. This is perfect for home use when listing to a playlist of mixed sample rates.
- No noise during on/off when amps are on - You can turn the Motu 1248 on/off and never get any noise out of the amps - no tick, pop, click!
- DC Coupled Output - With no capacitors in the signal chain, there is no rolloff or phase shift of the low frequencies.
- Microphone input and line-in use same master clock as output - Measurement software can know the exact delay from output to input so driver distances can be calculated to the 1/100th of a millisecond
- Routing - Mixer software allows one to route output from measuring software through JRiver for crossover design
- Web Interface - Mixer software is controlled via a web interface on the device itself. This means routing, levels, trims, etc. can be control with a tablet or phone
- 8 analog inputs - other sources can also be routed through JRiver to use the crossovers
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Here are pictures of the dspMusik stand along digital crossover. Has USB2 asynchronous, S/PDIF and balanced stereo ADC inputs and 8 programmable balanced DAC outputs all running at 24bit/192KHz processing. No linear phase processing in the signal path.
(http://www.audiocircle.com/image.php?id=122170)
(http://www.audiocircle.com/image.php?id=122171)
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That looks really nice.
I just saw you are in Callaway, MD. My sister and her family are about 3 miles away in Lexington Park. They moved from Nebraska a couple years ago and I still haven't been there. We were going to visit this summer, but now have pushed it off to next year.
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I'm interested. I have the LS-6, bought some upgraded crossover parts from Danny but have not installed them yet.
Do you have a ballpark estimate of the cost? More than $500, less than $1k, etc? Would this only work with PCs/Macs
as a source?
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The dspMusik is $1500 custom programmed for the crossovers for the LS-9 and LS-6.
It has a USB driver to install for Windows that is ASIO compliant for PC's and will work natively for MAC OSX via the USB.
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That looks really nice.
I just saw you are in Callaway, MD. My sister and her family are about 3 miles away in Lexington Park. They moved from Nebraska a couple years ago and I still haven't been there. We were going to visit this summer, but now have pushed it off to next year.
mojave,
If you do get to the area, a friend has a dspMusik crossover running his VMPS RM50 speakers that you can demo.
Just let me know in advance, and can set something up.
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The idea behind the dspMusik digital crossover was to make a system to beat the sound quality of the DCX2496 and DEQX 2.6P that I had. It also has 8 outputs to add things like rear channel subs with time delays.
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guess I'm a bit daft so how does this work?
Digital source only, built-in dac, output 4 channels analog to amps, separate tweeter/woofer amping?
Volume control handled by?
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Takes a stereo signal feed from either an analog source like a preamp, S/PDIF coaxial digital or USB2 asynchronous audio data from a PC or Mac.
There is a floating point DSP chip doing the signal processing for up to a 4 way crossover with time delay if needed. Also other processing is available like parametric EQ. Think of it as a Behringer DCX2496 digital crossover on steroids. All 8 output channels are independently programmable.
Has four stereo 24bit/192KHz DAC's for 8 analog outputs that is balanced.
The analog outputs feed the power amps directly. The amps then directly tie to the speaker drivers. All the amplifier damping factor is now controlling the speaker motion.
Does that help?
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Add me to the list of interested folks HAL. I'd like to hear some folks impressions of these verses their normal crossovers.
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Will do.
Might take awhile since volunteers with LS-9's and LS-6's with external crossover connections are needed for this one.
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I may be able to help with this. :)
Will do.
Might take awhile since volunteers with LS-9's and LS-6's with external crossover connections are needed for this one.
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Hugh,
If it is the LS-9's, I can start the process.
You just need to be able to biamp directly to the driver wiring inputs that would normally go to the crossover outputs.
Setting levels between tweeter and woofer amps, if they are different, should be doable with REW and a UMM-6 mic that I can send with the MS-2 demo computer.
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Sorry for joining the party so late, but I'm very interested. I have LS9s with external crossovers and I'm biamping them with 4 * Dodd 120 monoblocks - so consider me the perfect patient.
My initial plan when building these was to use a DEQX to crossover and EQ the bass arrays and use the stock passive crossover on the tweet arrays. Danny talked me out of this as he was completely dead set against DEQ, but said that he would be open to it if someone could make one that was any good. Danny built me a custom passive line level crossovers (really just one cap in the signal) to roll off the deep bass before going to the bass amps, this helped the overly exuberant bass that the LS9s were pumping out in my room.
Have you considered designing this for the bass section only? Isn't that where the real room problems happen, and where less harm is done by a ADC/DAC conversion? Although I suspect the issue could be creating a time delay if it is used for the bass only.
PS I've tried biamping with a pair of 200 CI Audio class D amps I have kicking around, and the 120 monoblocks sounded substantially better in my system.
Thanks,
Fraser
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I have done one install for bass only and it worked really well.
Danny has heard the dspMusik in his system with the latest BG driver two way with servo subs. He was very impressed from what I discussed with him. I had both a DCX2496 and DEQX and this is in a different level than either system.
The system can do time delay as well if needed. Did that at Danny's with excellent results.
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Thanks. I suppose it can only create delay - it can't move time forward. :scratch:
So if it's only hooked up to the bass drivers, and it has any inherent processing delay, it could lag the bass drivers behind the treble.
I've often worried about this issue using the DEQX - the extent to which processing through the unit creates delay, such that it doesn't work well unless it's hooked up to the tweeters as well.
I'm happy to be the guinea pig on this.
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That is why if it is used as the entire crossover it is not an issue. The processing time is very low for delay.
Will keep everyone posted
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If I still had my LS-9's I would be considering the Lyngdorf TDAI 2170 to drive the NEO 8's and the SDA 2400 2 channel amp to drive the woofers. You get "room perfect" digital correction, Digital Crossover, time delay and tons of clean, cool running power. With built in state of the art on board DAC including USB and HDMI inputs. It seems like a dream come true.
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Considering the Lyngdorf system (processor and amp) you mention is $6499 for the two units and the dspMusik is $1500, would be an interesting comparison.
I am still a proponent of correcting the room first with minimal digital correction. A few PEQ's here do very well. I put REW on the MS-3 PC Music Server and use a UMM-6 USB mic for the room measurements. I make the measurements using the dspMusik as the output DAC for REW.
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If I still had my LS-9's I would be considering the Lyngdorf TDAI 2170 to drive the NEO 8's and the SDA 2400 2 channel amp to drive the woofers. You get "room perfect" digital correction, Digital Crossover, time delay and tons of clean, cool running power. With built in state of the art on board DAC including USB and HDMI inputs. It seems like a dream come true.
Keep in mind that the digital "room correction" or "room perfect" correction is a bit misleading. I really see it more as a marketing gimmick than a real product. It is basically just an amplitude EQ. It doesn't correct room related reflections in any way. The reflections are all still there disrupting the sound stage and imaging. They have just been tuned down in the heavy areas.
Correction in this manor or in any regions above 200Hz or so is placement sensitive. In other words you can correct it for one point in space, but if you move over a foot in any direction it could be just as bad as it was before.
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I've always prefered to first get my room as flat as possible with corner bass traps, first reflection absorbers, etc. and then to use DEQ for bass only. That way it keeps all that processing out of treble regions that are more sensitive to digital corruption (ie. ADC/DAC).
But I'm curious as to how bad a delay is created by using a device like this on the woofer arrays (and subs) only. That's why I stayed away from it last time. It would be good if we could measure the delay that it creates (when the delay function is not used, obviously). I recall your telling me that any small delay created when dealing with very low frequencies, like the sub, is unlikely to be a big deal because those waves are bigger - but it might be a big deal when we're up around the LS9 crossover point if the DEQ is delaying the woofers.
I couldn't find any figures for the DEQX, Berringer, etc. since all of the manufacturers contemplate the devices being used on every driver.
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Let me just give you guys my feedback on Rich's product and what I learned from the listening comparatives we made.
First off, when comparing all of the digital crossovers out there to one another it is clear to me that what Rich has is a cut above all of the rest. There is no question about it. The quality of the DAC alone in Rich's unit is better than any of the others and compared to the lower cost units out there the difference is very significant. Rich's unit is the first and only digital crossover out there using a top level DAC.
The listening session we had also confirmed to me that passive crossovers are not the detriment many people would like to believe. Using Rich's DAC just as a passive DAC (like any other DAC) driving the speakers through the passive crossovers and then comparing it to Rich's DAC and digital crossover revealed very little differences.
The digital crossover had a slight disadvantage of having to use more amps and more interconnects and more speaker cable verses the passive crossover and single amps. Still I thought there were a few times that some things were a little farther outside the sound stage using the digital crossover, but not by much. Since the digital crossover allowed a finer adjustment of the phase between the main speakers and the servo subs towers there was better integration of the servo subs.
We used the digital crossovers to match the passive crossovers slopes and made them identical for the comparisons. However, the digital crossover could have been used to create different slopes and lower crossover points without any negative effects. Passive crossovers don't allow that kind of flexibility. So there is a lot of possibilities with the digital crossovers that can allow it to excel over a passive crossover in a lot of ways.
Again it all came back to the quality of the DAC much more than whether the crossover was passive or digital. We had two other high level and really similar DAC's to compare to. My personal DAC shared some of the same guts as Rich's but there are a few things done to mine that took it up a notch. So for me the best overall sound was still with my DAC and a passive crossover. But the quality of the DAC in Rich's unit alone will make it a big improvement over 95% of everything out there even if you aren't using the digital crossover.
My recommendation for anyone with an LS-6 or LS-9 wanting to try Rich's unit is to get one from him and just try it as a stand alone DAC first. That's real easy to do. And chances are really high that it will be an improvement over your current DAC. If you like what you're hearing then you'll also like it as a crossover. Using it as a crossover will give you some slight improvements in a few areas and some powerful tools to correct room loading below 200Hz as well. But you will need more interconnects, more speaker cables, and more amplifiers.
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Thanks Danny for posting your observations!
Syrah,
The time delay through the dspMusik is 0.000333 seconds. It is a fixed delay time for all channels the way it works.
At the speed of sound in air at 1130ft/sec, that is 4.5" of distance offset.
The system that uses the dspMusik for bass crossover are VMPS RM40 speakers and four subs. The dspMusik runs the two woofers and four subs, and an active analog crossover runs the mids and tweeters. At the crossover point of 309Hz in use, it is only a few degrees of phase difference. No issues in that setup.
At 800Hz and above, the phase offset starts coming into play. That is around the frequency of the LS-9 crossover from woofers to tweeters. That is why I recommend using it as the full crossover and not a bass management system. That is well into the midrange.
The other way to do it is to use two channels of the dspMusik as just a DAC and run that to the amps driving the tweeter crossovers. That way the time delay is maintained correctly to the original passive crossover to the tweeter if that is the preference. You can still use all the features on the woofers like adding PEQ for mild room tuning without affecting the tweeter response.
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Keep in mind that the digital "room correction" or "room perfect" correction is a bit misleading. I really see it more as a marketing gimmick than a real product. It is basically just an amplitude EQ. It doesn't correct room related reflections in any way. The reflections are all still there disrupting the sound stage and imaging. They have just been tuned down in the heavy areas.
Correction in this manor or in any regions above 200Hz or so is placement sensitive. In other words you can correct it for one point in space, but if you move over a foot in any direction it could be just as bad as it was before.
I agree with Danny that EQ dos not correct room all room related reflections. However, I have heard it do amazing things in my room. I had a builder make a pair of 3 way loudspeakers with Digamoda amplifiers a couple of years ago. being able to dial in the bass with slope, Time delay and EQ all in the digital domain changed my mind quite a bit on DSP and ICE powered amps. Done right, it is quite amazing to hear what is possible with that technology. The user interface is a bit advanced and not really DYI friendly for the average builder. Best left to an experienced engineer IMOP. I had quite a bit of help in that area.
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Okay, I'll admit that a lot of what is being said on this subject is over my head. But let me speak up for guys like me.....guys who have gone to great lengths to keep their music reproduction as "organic" (tubes yes, op-amps no) as possible. And we are now supposed to put a digital crossover in our music system? Ain't gonna happen. Yes, digital has improved, and perhaps when it gets above a certain bit rate/sampling frequency/whatever it is transparent. I send one of my tube pre-amp's outputs to my tube power amp which powers my loudspeakers from 180Hz up. The other pre-amp output goes to a DSPeaker 2.0 Dual Core (a digital room mode correction device) and on to Rythmik A370 amps which power GR Research 2-12 OB dipole subs from 180Hz down. The DSP 2.0 Dual Core is confined to frequencies of 180Hz and below, for which digital is fine. But a digital crossover through which frequencies above 180Hz pass? Surely you jest!
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No jest involved.
Until you hear a fully implemented no linear phase digital processing system including DAC's, you have not heard this capability.
The only linear phase part are the A/D converters. That is not used with the full digital system from Music Server to DAC outputs, just the analog inputs.
At this point I use the MS-3 to dspMusik to PA's directly driving OB BG NEO3's, NEO10's drivers and GR-Research/Rythmik servo subs. Sounds better than all the speakers I have tried in my setup and there have been many.
In Danny's setup we used his small Dodd tube monoblocks on the tweeters and had Class D amp on the midranges and it sounded excellent with his new line arrays.
So far it has been used with VMPS RM40, RM50, Sanders 10D, GR-Research N2X and moded V1's speakers with reported improved sonic results.
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Setting up my CLIO measurement setup today and have the LS-9 crossover on the bench.
Will get the setup working and see what I can do.
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I am excited about this, but I do share bdp24's concerns. In a perfect world, I'd like either a unit like this that has negligable time delay so that it can be confined to the bass section.
I have a vinyl rig and a digital server with DSD capability and a DSD DAC. It seems strange to take those analog signals and then put them through a round of ADC/DAC once more.
Maybe if I had a high end digital phono stage that did ADC and the whole RIAA in the digital domain, and a server that went straight to your DAC keeping it in the digital domain right to the end with only one digital to analog conversion, that might make me more comfortable with the technology.
In terms of vinyl, most reviews that I've read still give the nod to traditional phono stages over those that convert the signal to digital and then do the RIAA curve in the digital domain - although I suspect it's a matter of time before a digital phono stage comes along that's a giant killer.
Any chance this could be DSD capable? I'm using a Chord Qute EX now with a JRiver server and a decent sized DSD collection of music. If this DAC is as good as Danny says (and is better than my Chord) and could do DSD, it would be a real contender.
Thanks!
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Syrah,
Sorry, but the dspMusik is not a DSD DAC and will not be due to the signal processing required. I use it with my DSD DFF files by downsampling to 24bit/176.4KHz PCM in my player. That works very well.
It was compared as a PCM DAC to a EMM Labs DAC2X. The customer that has the dspMusik liked the dspMusik in comparison to his DAC2X. He is the person with the hybrid setup using the dspMusik as the bass management section below 300Hz. woodsyi is a big vinyl person and that setup was an improvement over his original setup.
Does not sound like it is for your system as the crossover point is to high to use it as a hybrid due to the time delay. Sorry it will not work out for you.
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HAL,
You're going to have to attach the drivers to the crossovers to measure/capture the exact electrical response. You are aware of this?
Dave.
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For the NEO8, I can use a resistor as it is a planar.
The woofer is going to be another resistor, but may need a woofer to see the difference in loading.
Starting simply to test the setup.
Would be more fun if I had a pair of LS-9's to measure directly, but this will be a good start.
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First set of crossover measurements completed. Will see what I can do.
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For the NEO8, I can use a resistor as it is a planar.
The woofer is going to be another resistor, but may need a woofer to see the difference in loading.
Starting simply to test the setup.
Would be more fun if I had a pair of LS-9's to measure directly, but this will be a good start.
Can't you trace out the coils and capacitors and draw a circuit diagram to figure out the transfer function?
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It is easier to measure the impulse response of the crossover with loads and work from that.
CLIO does the circuit impulse measurement and they are faster to do than building a Spice circuit model with a speaker load.
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Hello From Denmark
follow this treat with great interest ..because i have used something like this.on my LS9
my first digital crossover was Groundsound 2 ways,,,where i used a Groundsound amp for the bass and my Gryphon Encore amp for the top...it sound really good,,,only problem was ,that my Gryphon amp sounded better than than the Groundsound in all aspect,,,so i could not get the best sound out off the speaker..
next step was to change to passive again,,and upgrade some off the component in the filter,,,
for the bass i use a Jantzhen c-coil with very low resistance ---give me a very dynamic bass
for the neo i change to mundorf supreme caps,,and cross coil all over----all parts with the same value as in the original
Then i use Jriver on my pc ,,and go from usb to my DDdac pcm1794..where i can play hi.res music
the Jriver prog is very nice ,,it have also the same functionality as my former Groundsound digital filter,,,,like room correction and so on
The biggest problem .in my opinion ,is ,the need for to amp in the same quality ,,,remember that the LS9 crossover in a place, where it is important, that the amp is equal good
another important things in digital filter and dacs ,,is the output stage,,,no Opamp is good enough,,,compare to a cap, or trafo, or a good tube output
for now i must say, the sound in LS9 is very very good..... :)
Best Regards Bjarne
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Yes, in biamp situations all the amps need to be excellent sound quality to show the improvements.
The link in my opinion that caused the issue with the Groundsound digital XO is that it uses linear phase reconstruction filter DAC's just like the DDAC. From the units I have researched, no one uses this technique to keep the linear phase process out of the crossover.
It's processing rate is 1/2 the dspMusik and has more time delay through the unit from the spec sheet.
The dspMusik has an opamp based output stage that is DC coupled with very low DC offset. This gives the best bass response of any of the digital filters I have used. Sound quality to me is very good indeed.
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next step was to change to passive again,,and upgrade some off the component in the filter,,,
for the bass i use a Jantzhen c-coil with very low resistance ---give me a very dynamic bass
for the neo i change to mundorf supreme caps,,and cross coil all over----all parts with the same value as in the original
The Jantzen C-coil is a toroidal style coil wound on a laminated iron core. So you lowered the resistance and picked up some low end output but it will also smear the mid-range. Their coils are also not as advertised. The Copper wire comes from China and is recycled. It averages 94 to 96% Copper. The coil that was on the crossover to begin with was a high quality four 9's pure Copper that was an air core. It was much higher quality than the coil you went to.
Also, the stock crossover used Sonicaps. They are high quality and better sounding than the Mundorf caps you swapped out for.
If you really want some improvements go back to the original parts and add some Sonicap Platinum by-pass caps to the tweeter circuit. This will improve clarity and low level detail by quite a bit.
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HAL,
I think you're confusing folks with your assertions that linear-phase filtering is not desirable....either in the crossover or the DAC portion of these DSP gadgets. Elaboration and some explanation of your point of view would probably be appreciated.
Most users who are in search of information regarding the potential phase characteristics of DSP crossovers, are probably under the impression that linear-phase operation is desirable to minimum-phase operation in all instances.....all other things being equal.
For reference: All DAC's are linear-phase....in the passband (the audio frequency range.) If not, they're not very good DAC's. :)
"Linear-phase" recon filters remove the phase shift above the limit determined by the sampling rate, but it's questionable how much subjective improvement is gained from that.
Cheers,
Dave.
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Dave,
In the other thread I started for the MS-3 and dspMusik I posted white papers on the subject of why I am not using linear phase filtering in the processing and output DAC used in the dspMusik. Here is that thread:
http://www.audiocircle.com/index.php?topic=132170.0 (http://www.audiocircle.com/index.php?topic=132170.0)
The DAC's are Wolfson Micro WM8742's for anyone interested. I have selected the minimum phase reconstruction filters for the DAC's in the system, since those sound better to me. It is also using the Apodising filter for the A/D section to minimize the effects of it's linear phase filtering for reconstruction. Those DAC's were developed based on the research in the white papers. The specific part is the pre-ringing in linear phase filters. Minimum phase does not have pre-ringing and from the psycho acoustic testing in the papers the one test subjects preferred.
Those same DAC's have been used in prototypes used at RMAF in a room I know of. Usually reviewers have stated they are state of the art sound quality. Those did not use linear phase reconstruction filters.
In my filter design class in college, filters were discussed as to how the alignment is done. Hence I used the term no linear phase filters. Minimum phase is the alignment used for the DAC filters in this case.
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Danny: May I ask what value of Soniccraft platinum bypass caps you recommend for he LS-9 crossover?
Thanks,
Chuck
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Danny: May I ask what value of Soniccraft platinum bypass caps you recommend for he LS-9 crossover?
Thanks,
Chuck
Technically anything below a .1uF works well. I most often use a .022uF value. It hits a good medium of performance verses price since the smaller values are less expensive.
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Thanks, Danny. I'll order some up.
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Anyone else interested in demoing the dspMusik with their LS-9's? I have one system that can be sent for trials.
I need at least one person in the Continental US that wants to demo it so I can verify the digital crossover is correct. I have no way to test it here with speakers. If you are in the MD, VA or DC area, I can setup a demo with your system.
This will be the full HAL MS-3 and dspMusik demo system with TRS to XLR cables, as changes may need to be made and much easier using the Audio Weaver development system on the PC to change things if needed. I just email you a file to load as needed.
So far each demo has been done that way in every system. It is very easy to use and comes ready to play your files with Foobar2000. Just hook up a display, USB drive, keyboard and mouse to the front USB ports and the dspMusik to the back USB port. Connect the XLR connections to source and amps. Then you are ready to play music. If you need single ended connections, I can send TRS to RCA adaptors to use your RCA interconnects.
You can demo the system for 2 weeks to give you time to evaluate the sound quality. Just ship it back to me after the demo time.
In this case it has to be used as the full crossover for the system, not as bass management as discussed above.
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First person to try using the MS-3 and dspMusik with his LS-9's is happening.
First he is moding his LS-9's to put the passive crossover outside the chassis. Will be sending CLIO to him to make the crossover measurements and then sending the block diagram for the LS-9's.
In the mean time they will be comparing their MS-1 running Windows 7 to the MS-3 running Windows 10 using the dspMusik as the USB DAC.
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Rich, What DAC will the dspMusik be compared to in his system?
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I look forward to the results. Such a shame I'm not in the US. I have all external Xovers with LS9s powered by 4 * Dodd 120w. I would have been a good guinea pig for you!
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007KLH,
He has both an April Music and a Gary Dodd DAC.
Syrah,
Will keep everyone posted on results. The system shipped out today for trials.
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The MS-3 and dspMusik have arrived at Forest Dweller's place for LS-9 crossover trials. Modding his LS-9's to move the passive crossover externally for measurements, then trials with the dspMusik crossover running his Bryston 7BST's.
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For this digital crossover, will be trying the high performance biquad filters in the Audio Weaver software for the dspMusik.
The PC laptop running Windows XP to run CLIO 8 system is setup and calibrated to send for the passive crossover measurements for the LS-9's. Will be packing that system up and shipping it next week. Might have to include my Parasound Z-Amp v3 to make the measurements, as the customers amps have H-Bridge outputs and CLIO has a single ended input system.
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After discussions, will be sending the new Pocket CLIO measurement system and Parasound Z-Amp v3 for the LS-9 crossover measurements.
The passive crossovers are now external to the LS-9's to attach the CLIO input leads for the measurements.
Will be doing a short document to show how the measurements are made and shipping the system to the customer to test his speakers.
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The complete Audiomatica Pocket CLIO system and MS-1 computer is with the customer with the Angel City Audio and Skiing Ninja built LS-9's at this time. He is unpacking and setting it up for passive crossover measurements.
His normal amps have balanced outputs, so sent a Dayton Audio APA-100 stereo amp to use that has grounded outputs needed to work with the Pocket CLIO for measurements.
Once the measurements are complete, the data will be sent to use to create the digital crossover design.
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The first trials of the LS-9 with HAL MS-3 and dspMusik setup occurred today with a prototype digital crossover.
First listening was 4-5 hours and so far he is very happy. Next up is the full set of measurements, full crossover and speaker correction.
Each array has it's own level set for exact matching with measured SPL from the UMM-6 mic and REW.
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Would this mean that I could get rid of all the room treatments that have transformed our living room into something similar to your laboratory? The LS 9's are great speakers, but not the easiest to get right. Something like that could make my wife VERY happy.
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Scott,
Forest Dweller has lots of GIK room treatments with his setup. I would empathically say no to getting rid of them. I am not getting rid of mine either.
What this should do is the next level of LS-9 integration with the room to your listening position.
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Would this mean that I could get rid of all the room treatments that have transformed our living room into something similar to your laboratory? The LS 9's are great speakers, but not the easiest to get right. Something like that could make my wife VERY happy.
:lol:
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:lol:
Well, I could always hope. My wife is a saint... putting up with LS9's, massive room treatments, mystery UPS packages (often from HAL), eccentric husband...
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Well, I could always hope. My wife is a saint... putting up with LS9's, massive room treatments, mystery UPS packages (often from HAL), eccentric husband...
Scott,
I wasn't amused by you wanting to get rid of the treatment (we'd all love to be able to do that right ? ), kind of found it funny at the reference of turning your living room into Rich's lab :thumb:
We often hear people complain about WAF, seldom are the women that put up with our "sickness" mentioned. Sounds like you're one of the fortunate ones :beer:
jay
ja
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Jay,
My wife and I came to an agreement. I have the basement and she has the top floor.
I put a door on the basement and close it to carry on the audio experiments in the Lab. My listening room is basically sound proof with it's door closed. :D
More feedback is that he was up most of the night listening and could not stop. I think we are making progress!
Scott,
Mystery UPS packages are fun! :D
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As a point of reference, the LS-9's are one of my favorite speaker designs of all time. Ranks up there with the Infinity IRS Reference V's.
Danny's passive crossovers are some of the best I have ever heard. If you do not start with a reference design like the LS-9's, you really never know if you are making sonic progress. There are a lot of crappy crossover designs out there, so anything is sonic progress in those cases.
The test crossover being demo'd is a 6 pole Butterworth design for high pass and low pass at the original crossover frequency of the LS-9's. No bass EQ at this point, just flat. Nothing fancy, just very accurate for both channels.
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Just curious, HAL. I have always heard that, if possible, a first-order x-over was the way to go. Did you play with a first-order (6dB/octave) x-over before you settled on a sixth-order? I called BG several years ago and was told that the NEO-8's, especially in multiple arrays, could take a shallower high-pass slope without a problem. Looking at the curves of the cone midranges, I conclude that the cones could be operated higher, such that a first-order x-over could be used as a low-pass filter as well. I'm already running my cones without attenuation at the low end.
So, it seems that a first-order x-over at a higher x-over point is at least feasible, and - - theoretically, at least - - desirable. I'm chicken to try it, but I don't mind your blowing up your own speakers or lab in the interests of science.
Not that I am at all dissatisfied with what I have. My Danny Ritchie 9's are the best speakers I have ever had (many, many really good ones, including KLH-9s and Maggie 20.1s) or heard (a veteran of many shows), but I can't help wanting to mess around with them. "They're coming to take me away, away . . ."
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THROWBACK,
This was the first idea to make an acoustic 8 pole filter roll off to keep the same polarity drive to both drivers.
Since single pole filter roll off adds to the existing driver roll off, it would be about a 3 pole filter at the crossover point. That would put the drivers phases off and not work well. Possibly a 2 pole filter for a 4 pole acoustic crossover that would be in phase might work.
The dspMusik's processing is very clean. I doubt that going to lower number of poles will increase sonic performance, and the higher number of poles will increase power handling in the speakers. Not that the LS-9's really need that.
It really is all about the measurements to be done.
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Just curious, HAL. I have always heard that, if possible, a first-order x-over was the way to go. Did you play with a first-order (6dB/octave) x-over before you settled on a sixth-order? I called BG several years ago and was told that the NEO-8's, especially in multiple arrays, could take a shallower high-pass slope without a problem. Looking at the curves of the cone midranges, I conclude that the cones could be operated higher, such that a first-order x-over could be used as a low-pass filter as well. I'm already running my cones without attenuation at the low end.
So, it seems that a first-order x-over at a higher x-over point is at least feasible, and - - theoretically, at least - - desirable. I'm chicken to try it, but I don't mind your blowing up your own speakers or lab in the interests of science.
Not that I am at all dissatisfied with what I have. My Danny Ritchie 9's are the best speakers I have ever had (many, many really good ones, including KLH-9s and Maggie 20.1s) or heard (a veteran of many shows), but I can't help wanting to mess around with them. "They're coming to take me away, away . . ."
First order filters are not really possible with line sources. As frequency decreases you get coupling, but as frequency increases you do not. So the passive parts first and foremost have to adjust for that. They it takes a couple of additional parts to create the crossing over slope and adjust for phase.
And when BG says the tweeters will handle it, what they really means is that they will handle it power wise. It doesn't mean that the response curve will look like anything.
Plus, these really shine with lower crossover points. The lower crossover point minimizes cancellation in the horizontal off axis that you would have with typical line sources that have higher crossover points. At longer wavelengths the off axis phase rotation is just a few degrees. While at higher crossover points phase rotation in the off axis will be extreme and cause cancellation.
And the differences in the low crossover points are easily heard. The LS-6's cross at 1kHz and the LS-9's cross at 850Hz. This isn't a chosen crossover point either. It is where the line natural begins to roll off. The longer line of tweeters have more low frequency coupling. And the LS-9 has a little better resolution and detail than the LS-6 as the tweeters pick up the response a little sooner.
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Updated LS-9 crossover running and good feedback on sound so far. This test has flat response low pass filter for the woofers, no EQ.
Will be interesting to read the write-up in work with pictures.
Last step is trying MathAudio Room EQ with them as well. The Dayton Audio UMM-6 measurement mic is with the kit of demo gear to try.
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HAL, Dannie. Thanks for the response. I'll give up on the first-order x-over idea. I'd write more, but I just received a marvelous rendition of the Bruckner 8th (Karajan/EMI original) and I have listened only to the adagio movement. Sublime.
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First order filters are not really possible with line sources. As frequency decreases you get coupling, but as frequency increases you do not. So the passive parts first and foremost have to adjust for that. They it takes a couple of additional parts to create the crossing over slope and adjust for phase.
And when BG says the tweeters will handle it, what they really means is that they will handle it power wise. It doesn't mean that the response curve will look like anything.
Plus, these really shine with lower crossover points. The lower crossover point minimizes cancellation in the horizontal off axis that you would have with typical line sources that have higher crossover points. At longer wavelengths the off axis phase rotation is just a few degrees. While at higher crossover points phase rotation in the off axis will be extreme and cause cancellation.
And the differences in the low crossover points are easily heard. The LS-6's cross at 1kHz and the LS-9's cross at 850Hz. This isn't a chosen crossover point either. It is where the line natural begins to roll off. The longer line of tweeters have more low frequency coupling. And the LS-9 has a little better resolution and detail than the LS-6 as the tweeters pick up the response a little sooner.
Danny, there are some guys over at the Planar Speaker Asylum who have substituted six NEO 8's aligned vertically as a line source in place of the stock midrange driver in the old Magneplanar Tympani T-IV speaker, wiring them so as to achieve a 5-6 ohm load. They are using symmetrical 1st order filters, low pass for the T-IV woofer panels and high pass for the NEO's at around 250Hz, and the same arrangement at around 6-8kHz, low pass for the NEO's and high pass for the T-IV ribbon tweeter. Do you feel the simple symmetrical 1st order filters alone are inadequate for what the six NEO 8 drivers require in this specific line source application?
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Danny, there are some guys over at the Planar Speaker Asylum who have substituted six NEO 8's aligned vertically as a line source in place of the stock midrange driver in the old Magneplanar Tympani T-IV speaker, wiring them so as to achieve a 5-6 ohm load. They are using symmetrical 1st order filters, low pass for the T-IV woofer panels and high pass for the NEO's at around 250Hz, and the same arrangement at around 6-8kHz, low pass for the NEO's and high pass for the T-IV ribbon tweeter. Do you feel the simple symmetrical 1st order filters alone are inadequate for what the six NEO 8 drivers require in this specific line source application?
Yeah, that won't work for a number of reasons.
And low order crossovers can work great for low crossover points, but the Neo 8's really won't play that low. And the crossover becomes more of a correction filter for the coupling down low verses what they do up high. So it requires a lot more than one part. A whole line of the much larger Neo 10's will barely play to just below 200Hz. No way on the Neo 8's.
And the Neo 8's actually play to nearly 40kHz. So why cross them to something else? And crossing them at 6-8kHz using a first order crossover will cause out of phase cancellations just above that point. Plus very slight movements to either side will cause out of phase cancellation in multiple places and they will be 15db suck outs. No if ands or buts about it. That will cause cancellations in the horizontal plane all over the place.
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Glad I asked! I saw a frequency response graph for the NEO 8, and it looked like it nose-dives above about 10k. Are you sure it's not the NEO 3 that goes out to 40k?
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Glad I asked! I saw a frequency response graph for the NEO 8, and it looked like it nose-dives above about 10k. Are you sure it's not the NEO 3 that goes out to 40k?
It drops off a little from 13kHz to 20kHz then levels out a little and extends on out. If you notch out the peak you can get a pretty flat response across the top end.
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Ah so. So when you used the NEO 3 it was for reasons other than fr?
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Danny discussed that in a thread back in 2010. Here is one of his other designs with Neo3's for the line of tweeters.
(http://www.audiocircle.com/image.php?id=144009)
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I still have a pair of those cabinets for the 20/21's.
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:o
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Ah so. So when you used the NEO 3 it was for reasons other than fr?
They were faster and had a little better detail level. And they didn't require any notch filter for the top end. It cost more to use them in that application though.
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Interesting. Looks like Epiphany Audio still have a website.
http://www.hellotrade.com/epiphany-audio/20-21-system.html (http://www.hellotrade.com/epiphany-audio/20-21-system.html)
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Forest Dweller's LS-9 measurements with the Pocket CLIO system are complete and look very good.
Will be working on the dspMusik digital version and send him the version for listening trials.
The challenge will be the Bass EQ system, so that will be in work for trials.
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New digital crossover version complete and sent to Forest Dweller for his listening trials. :D
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The dspMusik LS-9 digital crossover update is working and making music (at least what I could hear over the phone). Now waiting on listening observations.
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Listening observations are very good, so this will be the standard digital crossover for anyone interested in the dspMusik for the LS-9's.
Forest Dweller is doing a write-up and will post it here when he has time. I think he likes it! :thumb:
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Listening observations are very good, so this will be the standard digital crossover for anyone interested in the dspMusik for the LS-9's.
Forest Dweller is doing a write-up and will post it here when he has time. I think he likes it! :thumb:
:thumb:
How difficult will it be to modify for the LS6?
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Is the crossover internal to the LS-6's or external? If external it is very easy.
Skiing Ninja did Forest Dwellers change for him, but that might be his last one. He and the family moved to WA state.
The PC board or handwire board are screwed to the bottom of the LS-6 cabinets. It has to be removed via the cutout and a pair of 5-way binding posts added to the existing plate which means drilling new holes. Then the 4 wires to the drive arrays have to be soldered directly to the 4, 5-way binding posts.
The passive crossovers will still work, and now that it is outside the speaker, no longer subject to vibration. There maybe enough exiting wire to keep wire on both the PC board and still wire to the binding posts.
It also means you now need to biamp the speakers. That has to be factored in. There are lots of very nice multichannel amps that have more than 4 channels to use, or just dual stereo amps.
The bass EQ is now done with a control in the Audio Weaver software with 1dB steps, so that function is available along with delay capability if subs are used. The dspMusik can drive the line level inputs on powered subs directly and offset delays added for the mains for the offset distance from the listening position to the subs.
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:thumb:
How difficult will it be to modify for the LS6?
It's doable BUT it's also time consuming.
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Is the crossover internal to the LS-6's or external? If external it is very easy.
Skiing Ninja did Forest Dwellers change for him, but that might be his last one. He and the family moved to WA state.
The PC board or handwire board are screwed to the bottom of the LS-6 cabinets. It has to be removed via the cutout and a pair of 5-way binding posts added to the existing plate which means drilling new holes. Then the 4 wires to the drive arrays have to be soldered directly to the 4, 5-way binding posts.
The passive crossovers will still work, and now that it is outside the speaker, no longer subject to vibration. There maybe enough exiting wire to keep wire on both the PC board and still wire to the binding posts.
It also means you now need to biamp the speakers. That has to be factored in. There are lots of very nice multichannel amps that have more than 4 channels to use, or just dual stereo amps.
The bass EQ is now done with a control in the Audio Weaver software with 1dB steps, so that function is available along with delay capability if subs are used. The dspMusik can drive the line level inputs on powered subs directly and offset delays added for the mains for the offset distance from the listening position to the subs.
It's doable BUT it's also time consuming.
Sounds like it.
:o
I've dreamed about biamping the LS6's forever though....
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SoCalWJS,
It took Skiing Ninja about 4 hours with the background to make the mods to both LS-9 speakers. The LS-6's are very similar in that area.
If nothing else, the passive XO's outside and biamping would be a nice advance. Always did that with my MGIIB's to sound quality improvements.
I know sandbagger has been thinking about something like this for his LS-6's, so maybe a collaboration. He was looking for a new audio project last I talked to him.
Just a thought.
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Greetings All !
Yes, it’s true; my beloved LS-9’s are making music with the fully-integrated HAL MS-3 / dspMusik system. And I’ll just cut-to-the-chase right up-front: There is no going back! :thumb:
This latest project with Rich was a bit out-of-the-blue for me. Long day-gig hours and the beginning of fire season here on 24 acres of forestland was (and still is) keeping me extremely busy. So, apologies in advance if I can’t respond quickly to questions or post as much as I’d like right away. I’ll defer technical questions to Rich; and concentrate on my perception of where I was / have been before, and where I am now in terms of my personal listening experience.
My LS-9’s were born thanks to Hugh Nguyen of Angel City Audio, and Sean Ries of Skiing Ninja. And of course to Danny Richie for a killer design! Hugh provided the gorgeous piano black cabinets and drivers. Sean provided his custom LS-9 XO’s, plus got everything wired-up. I’ve had the LS-9’s in my main system since March, 2011. I’ve had a lot of different equipment in-line since then!
The system currently consists of:
HAL MS-3 Music Server
HAL /Danville Signal dspMusik 2/8 used as the digital XO
(4) Bryston 7BST monoblocks (with factory upgrades)
Audible Illusions L-1 tube preamp (waiting for my new L-3 to show-up)
April Music CDT 100 CD Transport (connected to the digital input on the dspMusic – haven’t tried this out yet)
Audiolab 8000T tuner
Studer A-810 ½ track master R2R
GR Research LS-9 Speakers (bi-amped) (no passive xo)
Electra Cable and JPS Labs Interconnects
Mogami TRS to XLR-M cables for connection from the dspMusik to the Brystons
TRS to dual RCA adapters for analog to digital input from L-1 to dspMusik
Virtue Audio USB2 cable for connection from MS-3 to dspMusik
Analysis Plus “Black Oval 9” speaker cables
GIK and P.I. Audio Group room treatments
The room is 24’ wide x 26’ deep. Speakers are 12’ center-to-center; and 12’ to listening position.
Listening:
Growing-up in the San Fernando Valley, CA. in the late 60’s – 80’s, playing drums and percussion and combined with recording and mastering engineers who were family friends afforded me an incredibly lucky and humbling end-to-end experience, at some of the most famous studios in existence at the time. From playing on tracks (or listening to someone else playing) to following them all the way through mixdown to master, was what brought me to wanting to reproduce those sessions, like all of us; with the best possible quality that we could actually afford, into our own listening space. So I was “assimilated” as an audioholic at a very young age. Resistance (not that there was any to begin with) was indeed futile.
When digital recording and playback entered the picture, it was certainly a brave new world. Hence the reason for the Studer A-810 in the system. My entire LP collection was destroyed in the Northridge, CA. earthquake. Digital has certainly come of age, in a world of “desiring all that is wonderful with analog sound reproduction”. Tough task. But, it’s finally getting really good, and really affordable.
I listen to everything from Big-Band, Fusion, Bebop, Latin, Classical, Rock, R&B, “Classic Country” (“modern pop and country” are usually a bit “too templated” for me) to even a little Rap -again, mostly the “old school stuff”, which was more fun than violent in nature. I listen to music from all over the world. My “farthest-back memorable musical moment” was listening to “Antonio Carlos Jobim / The Composer Plays”, in 1964 on a mono system at home. I was 4 yrs. old.
I’ve had the opportunity to play with quite a few DACs, starting with the Audio Alchemy DDE 3.0, shortly after it was first released. Most recent DACs have been a Dodd Audio DC powered DAC, and an April Music DP1. Each is different but wonderful sounding to me. I’ve tried, so far unsuccessfully, to get my hands on a Sabre DAC for comparison, but I’ve read some great comparisons between the Sabre DACs and the (4) Wolfson 8742 DACs in the dspMusik.
Rich and I have been playing around with components and great projects for a long time. I’d been experimenting with the Wolfson 874x series DACs for quite a while, just playing back tracks at their native rates, as Rich recommended. After years of up-sample and over-sample everything to the max, this was a new paradigm for me. Now that the industry is re-mastering tracks at very high resolutions, we can enjoy playing them back at their native rates. I think this is an important point, in that, I’m listening to music at Red Book resolution, all the way to 24/192kHz through the HAL MS-3 /dspMusik system, and (so far) even the lowest resolutions sound better than I’ve ever heard in my system.
I have heard, and thanks to Bruce Brown at Puget Sound Studios, actually created a DSD file from a Studer Mastering R2R, late one night at RMAF a couple years ago. He wanted me to see the process, and hear the difference. Pretty astounding! So while the dspMusik doesn’t (yet) do DSD; I’m thrilled about what it does do. I know Rich plays his DSD DFF files, using PCM Conversion, and is very happy with the results. On another note; I haven’t used any Behringer components or outboard XO’s in the past; so I can’t comment about that difference.
Rich introduced me to Minimum Phase Apodizing Filters used in this Wolfson DAC. They are slow roll-off filters. "Minimum phase apodizing" signifies that pre-ringing is eliminated, post-ringing reduced and aliasing distortion eliminated. When I said, “what??”, he said, “listen to it now and read about it later”. :lol:
This project had these milestones:
Re-wire the LS-9’s to accept outboard XO’s.
Connect the MS-3 and dspMusik to my system.
Measure the Skiing Ninja XO (based on Danny’s design) using CLIO, and send the resulting measurement files to Rich. He’d return the necessary XO file for the dspMusik.
Copy the .awd (Audio Weaver) file derived from the passive XO measurements to the MS-3.
Start Audio Weaver, and select that XO file for use with the dspMusik.
Audio Weaver will build the file and the dspMusik will see it and use it.
Make sure I’m running the ASIO driver from within the Foobar player.
And then…
The Moment I’d been waiting for: MUSIC !
(probably far past the moment of “just get to the results!” that you’ve been waiting for; reading through all of this. :roll: ).
The result was immediate; no break-in time required. And now I’m going to throw-around the “usual descriptions”, but ones that are hopefully useful to you, my fellow audioholics, in relating as clearly as possible “what I heard”.
The very first things, involved Resolution, Musicality, Smoothness, Dynamics (macro and micro), and Sound Stage.
I started pretty, “middle of the road” with 24/96 .wav and .flac files. I played following tracks from different genre’s and at varying volume levels and resolutions. I could put together some sort of track list; but I know all the tracks I listened to very, very well. And you know most of them as well, as “fairly commonly used reference tracks”. Very quiet, to very dynamic.
The most important thing was: was I listening to ‘a system’, or was I listening to ‘music’?
Each of the five points of “what I heard” listed above were enhanced to my personal liking, and not in a small way. I knew within 30 minutes of hearing the first track and very differing following tracks, there was no-going-back. :nono: :D
From top end, through mid-range, to very low (but properly tight ‘n punchy) extended bass which the LS-9’s kick serious butt at. I remember the first time I tried NOS Amperex 7308 Gold Pins in my Audible Illusions line stage preamp. Wow. Liquid bass, very smooth mids, and very clean and smooth highs. This was like that moment; but exceeded it to an entirely different level of Wow. It's the combination of the fully programable dspMusik, hand-in-hand with the LS-9's. I really wasn’t expecting sound this good; this fast. And it’s digital. Really good, affordable digital! I do know that would be an oxymoron to some - even me not all that long ago; but as we have come to find; it's an ever-evolving adventure, with good stuff goin' on in both analog and digital realms! I'm not giving-up feeding my Studer 8)
Not sure how to explain this as much as it hit me; but the soundstage both “opened-up” while becoming more definitive and still silky smooth while retaining the LS-9 Punch where I expected it. Not digitally, clinically, or overly so. So far, it’s the closest this system has gotten to the analog repro of my Studer to date. I’ve literally just gotten this all on-line, have hardly slept (can’t stop listening!) and have, as you would imagine, a lot more listening to do with different material and vs. different sources.
I’ve been fortunate to spend some time, listening with Danny (and many others, of course). Those of you who know him, know he is very intense on what he’s listening for and how to listen. I learn more every time I get to spend some time listening with him. I agree with his previous post (#34) in this thread. I’m not sure Rich will build stand-alone, Wolfson-based DACs in the future or has one to recommend, but outside of very, very expensive DACs you really can’t go wrong with the Wolfson’s. I agree that would be a good place to start, and then see if you want to try out the HAL MS-1, and then maybe the entire dspMusik system. I’d say, “Go for it”, but it does take a few extra things to make this all work. It’s so far, been more fun and exhilarating than I even expected.
Speaking of that, gotta give my sincerest thanks to The Ninja himself, as well as Mrs. Ninja for coming a long way off-course on their trek from Colorado to Washington to visit me, and re-wire the very speakers he originally wired, to accept outboard XO’s. We did a lot of listening with his passive XO's still in the system. Great time. I’ll have that story + pics soon as I can.
My thanks to all who have contributed (and continue to do so!) to my ever-surprising musical reproduction journey. Greatest thanks to Rich, Sean and Danny on this one. :beer:
Now for some pics!
The Preamble.
Note: HAL MS-1 with custom fan and wood top! I'll be using that once again 8)
(http://www.audiocircle.com/image.php?id=146338)
CLIO Measurement Workspace behind Rt. LS-9
(http://www.audiocircle.com/image.php?id=146339)
CLIO Test Measurement on passive XO
(http://www.audiocircle.com/image.php?id=146340)
Audio Weaver LS-9 Schematic on MS-3. Generated from CLIO Measurements.
(http://www.audiocircle.com/image.php?id=146341)
(http://www.audiocircle.com/image.php?id=146342)
(http://www.audiocircle.com/image.php?id=146343)
(http://www.audiocircle.com/image.php?id=146344)
Cheers!
Chris
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Chris, I've been waiting for this review. With the same speakers, and some overlapping equipment, your results are especially meaningful to me.
But if I were to go down this rabbit hole, would my crossover be valid only for digital. How does vinyl enter into the equation?
Scott
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Chris, I've been waiting for this review. With the same speakers, and some overlapping equipment, your results are especially meaningful to me.
But if I were to go down this rabbit hole, would my crossover be valid only for digital. How does vinyl enter into the equation?
Scott
Hi Scott,
Good Question! And I've thought about that as well. I've played some tape through the ADC input; and it sounded very good. I know Danville Signal is working on a follow-on board, with greater emphasis towards the ADC as well as digital. No idea when that will be available though. I'll keep running my Studer through the dspMusik and report my findings on the analog side.
Hope to see you at RMAF this year. I'm all set to go. Have peppers; will travel. :green:
Not sure where all my previous posts disappeared to. Plus my gallery pics. Guess it's been a while. Lots to catch-up on.
We'll see what Rich, and others have to say in response to your very significant question.
Chris
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Scott,
Chris is using the dspMusik A/D with his preamp for his 1/2" tape system. Should be similar to your vinyl setup. It can run balanced or single ended input depending on the source. Since the system is running 24bit/192KHz A/D's, you have 96KHz worth of audio bandwidth.
I have a full MS-3 and dspMusik demo system you can try and see if it fits your needs. It will be going to CAF2016 this week, and can be shipped to you after that. Will load the LS-9 crossover like Chris. You will just need to biamp the setup and directly connect the amps to the driver terminals since you have external passive crossovers. I can supply adaptors needed for either RCA or XLR inputs to the amps.
If subs are involved, can also setup the time delay capability for the extra channels. Lots of options with 8 outputs.
Just let me know.
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As Chris has stated, there is a dspMusik RevD system under development. It will have minimum phase A/D conversion and new D/A conversion since Wolfson Micro was purchased by Cirrus Logic and no new production is available and existing stock is going up in price.
The question will be how it sounds as a system. That is unknown since no software is running for testing. Time will tell and the length is unknown.
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On the question about the A/D converters for analog inputs, I know four installed systems using it for phono, dpd, woodsyi and Avoosl and mine. These all have very good phono stages and table setups. There are various opinions on this from all four.
I took all those comments to Danville Signal and that led to the RevD's new ADC module development for analog. As with any new development the length of time is indeterminate as to when it will be complete. I for one am not holding my breath and use the RevC every day.
Hope that gives you some insight to the status. Luckily we have the RevC units until that is done and tested.
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Chris (Forest Dweller):
Great room, thanks for the impressions and pics.
Congrats, it must sound amazing.
Rich (HAL):
When (or will?) DSD/DFF native playback capability be available on Music Server systems/DACs?
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The MS-3 running Foobar2000 will playback Native DSD files with a DSD DAC. Have done it with an EMM Labs DAC2X. Amanero USB board and I2S Bus DSD DAC and Gustard X20u with their USB drivers.
You cannot signal process DSD signals in a DSP processor for a digital crossover, so it has to be converted to PCM first. That is what is done with the MS-3 and dspMusik. So the answer is the dspMusik will never do Native DSD playback. It sounds excellent with Foobar2000 downsampling it to PCM to play.
Rich
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Thanks Rich (HAL).
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THROWBACK stopped by the CAF2016 room today for a listen to the HAL MS-3 and dspMusik setup.
Showed me a picture of his LS-9's and 3x12 servo sub arrays with his system. Very nice system setup. Good to chat with him for awhile.
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Well, it is good to see Chris (forest dweller) getting to do something other than work. Seems like he is always working. And he has done a great job on his room and system. I am sure Rich's new gizmos are well worth the effort. Maybe one day I will work up the courage to try that system on my LS-9's too. Chris called me last month and I think we talked about 2 hours. He was giving me the details about the cross-over, etc. He also convinced me to go to RMAF this year, so I have my ticket and hotel room. It will be good to see him, Rich, and Danny again.
Until then, take care,
Mark
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Mark,
Good to hear.
Have my plans set to be there as well, so see you both in October.
Rich
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I did assume that you were going Rich. Now all we have to do is convince Chris to bring his LS-9's and set them up in his hotel room so I can hear the new crossovers.
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The Pocket CLIO measurement system is back from Forest Dweller and he should have his computer back to run the dspMusik by Monday.
He has more listening comments to post once he has more time to write them up.
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I live only 50 miles from RMAF. For the last couple of years Danny has made a stop here on his way from Texas to the Show to scarf a Chick-fil-A (health nut). You LS-9 guys are welcome to do the same. Worse-to-worst we could dip down here for a listen.
http://www.audiocircle.com/thumbnails/user/57012/19949/139850_thumbnail.jpeg
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THROWBACK,
Sounds like a fun time to me.
If interested, I can always bring an MS-3 and dspMusik to demo like I did at CAF2016 that you heard, with your LS-9's.
We can even try the time delays between the mains and servo subs with the setup if interested.
Just a thought.
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THROWBACK,
Sounds like a fun time to me.
If interested, I can always bring an MS-3 and dspMusik to demo like I did at CAF2016 that you heard, with your LS-9's.
We can even try the time delays between the mains and servo subs with the setup if interested.
Just a thought.
Now that DOES sound like fun, Rich. I could get some of the guys together, etc. Why don't you PM me your schedule and let's see what we could work out.
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Will do.
Best,
Rich
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Ok, planning to take a MS-3 and dspMusik demo system to CO with the LS-9 digital crossover for triamping with the 3x12 OB servo subs.
Just updated the MS-3 to the latest build of Windows 10, as the MS-2 update went well.
This should be interesting!
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Well after dislocating, my left leg on the way to the airport for RMAF2016, did not make the show or demo at THROWBACK's place with the setup.
However, the latest dspMusik RevD system is at Forest Dweller's for trials with his LS-9's. This is the unit that was demo'd at RMAF2016 with the MS-3. He already has a RevC system running his speakers, so this will be interesting for feedback.
Hopefully he gets some time off during the holidays to give it a listen.
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Forest Dweller has the new dspMusik RevD digital crossover and HiRez DAC in his system with the ACA / Skiing Ninja rebuilt LS-9's.
Had a nice chat last night with him and he is now the owner of the new system.
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Sounds like a winner
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Um . . . "Rebuilt LS-9s?" Just the crossover?
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The passive crossovers were moved outside and extra speaker terminals added to allow the comparison to the dspMusik digital crossover with the 4 monoblock Brystons.