Beyond the Ariel

Interesting story about piano. I have a good friend (Dickran Atamian) who is a phenomenal concert pianist (a Nomburg Prize winner at 18). I once convinced him to let me record his playing a high-end electronic piano directly - no mics. When played back over my system even he was amazed - the piano was in the room with us. The only acoustic was that of the room and there were no conflicting cues of reverb from a larger room superimposed on the reverb from the smaller room. It was the most realistic playback of music that I have ever heard. Better than any miced recording that I have heard.
 
It's just a resonant cavity. If not properly damped will ring. An experiment: place cell phone on top of a glass, cup or whatever is adequate.

I agree. That is why I suggested filling it with damping material similar to a TL. It will basically be a 1/2 wave TL for an HF driver. The HF driver will just be something like a vintage 3" cone-tweeter or something like that. Because of the OB's figure-of-8 radiation pattern, most of the backwave will be absorbed by the TL. That way the front wave can be used for the 'spectrally shaped HF content' to flood the ceiling, as previously discussed, with very little of the backwave to smear the timing.
 
Off topic but know the best audio minds are here on this thread. Can someone help me out please. I need to find the answer to this,,,,,,,,,,,
The signal goes from transport to DAC, then to DSP, then to Digital crossover,,, in some instances. WHICH CONVERSION ARE WE LISTENING TO?
Is it senseless to have a high quality DAC before a DSP or Digital crossover or does the first conversion the DAC does stay in the line to the end somehow?
Someone please fill me in here,
Thanks so much,
 
Last dumb question.
Is vital, active bi-amping an ok thing for a tube amp to do?
I'm referring to the amp having different loads on each of the two stereo channels.

I'm sure you're aware that the topography that you describe is just one way to Bi-Amp. You can use one amp for the mids and highs on both channels and a separate amp (SS?) for the bass drivers. I've always heard using a separate stereo amp for each channel as "Vertical Bi-Amping". This method requires identical amplifiers for each channel.

Best Regards,
TerryO
 
Off topic but know the best audio minds are here on this thread. Can someone help me out please. I need to find the answer to this,,,,,,,,,,,
The signal goes from transport to DAC, then to DSP, then to Digital crossover,,, in some instances. WHICH CONVERSION ARE WE LISTENING TO?
Is it senseless to have a high quality DAC before a DSP or Digital crossover or does the first conversion the DAC does stay in the line to the end somehow?
Someone please fill me in here,
Thanks so much,

==============
With all due respect you have gone way OT. I notice you have asked the same question in a number of other threads that have been OT. Please use the search function, wiki or start a new thread.
 
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With all due respect you have gone way OT. I notice you have asked the same question in a number of other threads that have been OT. Please use the search function, wiki or start a new thread.

You're certainly correct. I've also been guilty of doing the same thing, although I try to avoid it.
Thanks for the reminder!

Best Regards,
TerryO
 
I agree. That is why I suggested filling it with damping material similar to a TL. It will basically be a 1/2 wave TL for an HF driver. The HF driver will just be something like a vintage 3" cone-tweeter or something like that. Because of the OB's figure-of-8 radiation pattern, most of the backwave will be absorbed by the TL. That way the front wave can be used for the 'spectrally shaped HF content' to flood the ceiling, as previously discussed, with very little of the backwave to smear the timing.

I am dubious this can be made to work. The #1 task for any loudspeaker is to launch a first-arrival wave with flat spectral response, or looked at a bit differently, a subjectively flat spectral response at the listening position.

After that milestone is accomplished, then we can argue the relative merits of clean time-decay that is free of resonance (good CSD performance), low IM distortion, available headroom, change in directivity index vs frequency, etc. etc. What Duke and I have been discussing over the last several posts is an add-on technique to an existing loudspeaker with good on-axis performance ... adding some of the spatial qualities of an OB without the hassles of equalization and limited headroom at LF.
 
Yes I'm aware of the horizontal configuration and the need for identical amps for vertical, thanks Terryo.
So now this thread is about one of us in stress instead of taking the time,,, "If you have the knowledge" to answer a couple of little questions.
The thread was idle for several days.
I'm so sorry to have caused you such great distress. However I've read this thread from the beginning and active crossing has come up a few times, enough that I know Linn does not and will not tri-amp, but the difference is,,,, Lynn and other elders of this thread would and still may take five minutes and resolve my problems.
This thread has gone so many directions and off topic so many times it's not funny. I put my understanding it was off topic right up front. What does a room splashing tweeter have to do with the Ariel? This kind of selfish behavior is unnecessary and I believe you owe me an apology.
Would it be better the thread go back to sleep???
And yes I will go back onto the net for a few more hours on the chance someone actually slips up and specifically answers my somewhat different questions.
Yes I've asked elsewhere, and no, no one has answered. You didn't.
 

The signal goes from transport to DAC, then to DSP, then to Digital crossover,,, in some instances. WHICH CONVERSION ARE WE LISTENING TO?
Is it senseless to have a high quality DAC before a DSP or Digital crossover or does the first conversion the DAC does stay in the line to the end somehow?
Someone please fill me in here,
Thanks so much,

In a signal chain that includes all of these elements, the ideal scenario would have the only conversion to analog at the end. Gary Pimm's setup is like this. The music is on a PC audio server. The DSP and crossovers are done with software.

In the setup you described, the transport's digital output is converted to analog by the DAC, then the DSP or digital crossover does an A/D conversion at its input and a D/A conversion at its output. In this type of system, the quality of the DAC is indeed critical, because its analog output becomes the new basis for everything that follows.

Hope this helps.

In a way, the question is actually not so far off topic. DSP isn't Lynn's cup of tea, but we heard a very interesting speaker from Germany at RMAF that made extensive use of DSP in the audio server. It sounded surprisingly good. In my Beyond the Ariel speakers, I still have eventual plans for using a QSC DSP-30 with the subs. Lynn and I do feel comfortable with using post-DAC DSP for bass management, but prefer to avoid its use at higher frequencies.

Gary Dahl
 
In a way, the question is actually not so far off topic. DSP isn't Lynn's cup of tea, but we heard a very interesting speaker from Germany at RMAF that made extensive use of DSP in the audio server. It sounded surprisingly good. In my Beyond the Ariel speakers, I still have eventual plans for using a QSC DSP-30 with the subs. Lynn and I do feel comfortable with using post-DAC DSP for bass management, but prefer to avoid its use at higher frequencies.
Gary,

Considering many of the early A to D converters sounded like garbage, I'm not surprised at your surprise at finding a speaker using DSP to sound good.

Although I can no longer claim to have "golden ears", yesterday I spent about five hours engineering at a friend's recording studio while he played live drums replacing digital drum tracks on five songs another musician had recorded at his studio.

The studio uses a great sounding analog console, very good sounding passive Dynaudio speakers, the ubiquitous Yamaha NS7 near field monitors, and Sony MD7506 headphones for monitoring. The control room is acoustically isolated from the various performance rooms, when listening to the studio monitors one hears only what the microphones are picking up. All the recording is digital, the studio owner long ago having switched from a 24 track analog tape recorder to various digital mediums, the current system offering virtually unlimited tracks.

The playback of the digitally recorded drum tracks to my ears was completely indistinguishable from the live drums during tracking, something I can't say about any analog recording medium.

There are still bad A to D converters being made, but the good ones are now at the point where they impart no audible change to the recorded signal.

DSP can correct many loudspeaker problems that are simply not possible to correct in the analog domain, and can now result in recording reproduction more accurate, more "high fidelity" than possible using an all analog path.


Also of interest, digital algorithms have been developed that can emulate the euphonic sound of tube (valve) amplifiers to a degree that even "dyed in the wool" tube fans are converting. Of course, the opposite is also true, even though digital reproduction is now more accurate than analog, there is a resurgence of a new generation embracing the "warm" sound of analog tape decks, vinyl records, and tube amplification.

I still own analog tape decks, vinyl records, and tube amplification, but no longer use them when accurate reproduction of music is the goal.

Art
 
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digital algorithms have been developed that can emulate the euphonic sound of tube (valve) amplifiers

Find me a brick wall - that's NOT what valves in hifi are about. :bomb:

It's about avoiding the signature high order odd harmonics; poor HF control; clipping issues and problems with back EMF/weird loads which are found in the TYPICAL diff-pair/integrator/complementary-AB-output design.
 
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Find me a brick wall - that's NOT what valves in hifi are about. :bomb:

It's about avoiding the signature high order odd harmonics; poor HF control; clipping issues and problems with back EMF/weird loads which are found in the TYPICAL diff-pair/integrator/complementary-AB-output design.
You seem to have missed the point, there are now digital algorithms that can emulate the avoidance of the signature high order odd harmonics; poor HF control; clipping issues and problems with back EMF/weird loads which are not found tubes/valves use in "high Fi" or musician's stage amplification.

I love the sound of a good tube amp, but I have found algorithms that reproduce the "tube effects" quite nicely.

Reducing my tube count considerably has afforded purchase of digital kit that can do more with less heat.

Still, like fireplaces on a winter's night, there is nothing like the smell of dust burning off glowing tubes to bring back memories from those years when I could still hear the flyback tone (15.734 kHz) of a CRT (cathode ray tube) TV set from across the room :).

Art
 
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I think that I'll just stay with the "Prehistoric Sound" that has served several generations so well already.

Do we need a "Don't Ask, Don't Tell Policy" for all op amps used at the recording studio? The Ghosts in the Machine.
 

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