Thanks GM, I don't think I've used the older crossovers. The later 800 and 500 Hz crossovers just did not sound good.
Bear has summed it up well in his last post. The only exception I take is the woofer. The 416 is one of the few that goes high and low and sounds good doing it. Thus its popularity. Yes it is better crossed at ~700Hz, but it does work an octave above that.
Bear has summed it up well in his last post. The only exception I take is the woofer. The 416 is one of the few that goes high and low and sounds good doing it. Thus its popularity. Yes it is better crossed at ~700Hz, but it does work an octave above that.
Agree 100% with everything GM and Pano say about the 416. It's a wonderful woofer which thinks it's a midwoofer ... one of the very few 15" woofers that can be used that way.
But it's not a subwoofer, and don't try and use it for long-excursion (more than a few mm of travel) applications. The old-school underhung VC is very linear in its range, but all you get is mush once it starts leaving the gap.
Some of the bad rep of Altec deep bass is the result of 416's used in not-very-solid bass-reflex cabinets combined with phonographs that have a substantial VLF peak in the 8~12 Hz range ... this results in the notorious "woofer bounce", which pushes the VC partially out of the gap. Keep the VC in the linear range, and the sound is very crisp and clear, with vivid tonality. Think full-range electrostats with better tone and a lot more physicality to the sound.
As for amps, I'm doubtful the 416 can sound its best with SET amps with 8 watts or less power. My experience is that 20W of PP power ... triode or pentode ... is a reasonable minimum to control the woofer. We are talking about a 70-gram cone here, and it generates substantial back-EMF currents for the amp to contend with.
As far as I know, 15" woofers are good for 20~300 Hz (most of them) or 60~800 Hz (very few). I don't know anything that's good from 20 Hz to 800 Hz. The traditional rule-of-thumb is to use drivers over a decade of frequency range (1:10), and not expect more.
The 416 is unusual in having a pretty well-behaved region from 800 Hz to 1.6 kHz, with just a single gentle broad peak, and smooth rolloff above that. Most 15" woofers go completely nuts in this region, and are not suitable for midrange applications. This is partially for historical reasons; as power requirements kept going up and up over the last couple of decades, prosound 15" drivers started looking more and more like subwoofers. The 416 is nothing like modern long-excursion drivers with 500-watt voice coils.
If you want combine the 416 with an 811, it's easier to just quote Pano. He knows what he's talking about.
But it's not a subwoofer, and don't try and use it for long-excursion (more than a few mm of travel) applications. The old-school underhung VC is very linear in its range, but all you get is mush once it starts leaving the gap.
Some of the bad rep of Altec deep bass is the result of 416's used in not-very-solid bass-reflex cabinets combined with phonographs that have a substantial VLF peak in the 8~12 Hz range ... this results in the notorious "woofer bounce", which pushes the VC partially out of the gap. Keep the VC in the linear range, and the sound is very crisp and clear, with vivid tonality. Think full-range electrostats with better tone and a lot more physicality to the sound.
As for amps, I'm doubtful the 416 can sound its best with SET amps with 8 watts or less power. My experience is that 20W of PP power ... triode or pentode ... is a reasonable minimum to control the woofer. We are talking about a 70-gram cone here, and it generates substantial back-EMF currents for the amp to contend with.
As far as I know, 15" woofers are good for 20~300 Hz (most of them) or 60~800 Hz (very few). I don't know anything that's good from 20 Hz to 800 Hz. The traditional rule-of-thumb is to use drivers over a decade of frequency range (1:10), and not expect more.
The 416 is unusual in having a pretty well-behaved region from 800 Hz to 1.6 kHz, with just a single gentle broad peak, and smooth rolloff above that. Most 15" woofers go completely nuts in this region, and are not suitable for midrange applications. This is partially for historical reasons; as power requirements kept going up and up over the last couple of decades, prosound 15" drivers started looking more and more like subwoofers. The 416 is nothing like modern long-excursion drivers with 500-watt voice coils.
If you want combine the 416 with an 811, it's easier to just quote Pano. He knows what he's talking about.
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Lynn is making a funny...
...this was a solution that I foolishly agreed to put together (lost $$ given the man-hours that ended up going in...).
The image with the horns vertical (they sound best that way...) is where they were residing, the previous owner passed them along to another friend in exchange for players to be named at a later date and a draft pick. The 203s in the rightmost pic are vertical now.
These are unlike any A7 you've ever heard or seen. A bit heretical, but the key is a xover that is acoustically aimed to work at 275Hz. This has a number of effects, including limiting the rise due to the front horn loading on the now non-Altec woofer. Also the LP is somewhat akin to a cliff.
So, you can run the top end with a SE 10 or 45 and the bottom with something bigger, including the Phase Linear 400 in the picture.
How does it sound? Maybe Pablo will invite you over. Wear a flak jacket to protect urself against the impact. 😀 Otoh it does subtle.
_-_-
...this was a solution that I foolishly agreed to put together (lost $$ given the man-hours that ended up going in...).
The image with the horns vertical (they sound best that way...) is where they were residing, the previous owner passed them along to another friend in exchange for players to be named at a later date and a draft pick. The 203s in the rightmost pic are vertical now.
These are unlike any A7 you've ever heard or seen. A bit heretical, but the key is a xover that is acoustically aimed to work at 275Hz. This has a number of effects, including limiting the rise due to the front horn loading on the now non-Altec woofer. Also the LP is somewhat akin to a cliff.
So, you can run the top end with a SE 10 or 45 and the bottom with something bigger, including the Phase Linear 400 in the picture.
How does it sound? Maybe Pablo will invite you over. Wear a flak jacket to protect urself against the impact. 😀 Otoh it does subtle.
_-_-
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+1 to what wesayso says about FIR filters and matches my years of experience with FIR based filtering. If you are into computer audio, then FIR based filtering is the way to go for digital XO, time alignment, driver linearization and (some) room correction. 64 bit FIR filters are completely transparent.
For example, my active 3 way with a 15" in a Cornwall cab, digital XO'd at 500 Hz with mid and high frequency constant directivity horns shows smooth and even frequency response across the listening area. Here is an example of 6 measures covering a 6' x 2' grid at the listening area:
Time alignment is also the same across the listening area. I used a single point measurement as input to design the FIR filters. As wesayso mentions, part of the DSP solution is the sliding correction window.
In my experience, it helps to use constant directivity waveguides or drive units with very similar polar responses. The traditional exponential horn, (or dome tweeter for that matter), in my experience, has rising directivity as frequency increases, to the point that is it so narrow at the listening position moving your head (or mic) 6 inches yields (sometimes) a completely different sound/response. Drove me nuts for years until I came across Earl Geddes work on constant directivity waveguides.
So if you like the A7 bottom, match it with a constant directivity waveguide and use DSP to fine tune the XO point, time align the drive units, correct the frequency response and you will be enjoying dynamic sound without the usual headaches.
Good luck with your project!
Hmm ... so the 275 Hz LPF is acoustically offset by a sharp rise in basshorn output, resulting in a net acoustical lowpass function somewhere between 450~550 Hz? If so, excursion and IM distortion would be reduced in a musically important octave, and the radiation pattern is also a better match to the MF horn. Interesting.
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I would think so. This is similar to what I had to do with my A5. The low pass was almost an octave lower than nominal to make up for the woofer horn gain. A similar approach works for Open Baffle to balance out bass losses.
Once I shoved the speakers into the corners, less crossover gap was needed.
Once I shoved the speakers into the corners, less crossover gap was needed.
I am like a 19th century physician whose mainstays are blood letting and tone controls learning about successful stem cell treatments and other recombinant genetic interventions. If one is accustomed to flying a Cessna and you get the opportunity to be dropped into the Space Shuttle it would be nice to find the manual. I have been scrambling to read about FIR filters and there are plug in modules for the minDSP family of products. Is there another DSP or platform that is out there for hobbyists or is the miniDSP the Tuesday after-school audio club place to start?
Is there any downside to this Philosopher's Stone?
Doing a mental experiment, the manipulation of digital source ahead of the DAC does not mess up the broth the way that filtering analog signals does. I have heard terrible cross-overs and "equalizers" over the years. Is there a downside to manipulating the digital source prior to the DAC? Are there unintended consequences or degradation of content? I get that one can botch the sound by turning up the "bass" to a Spinal Tap "eleven" but if one gets the balances right is there a difference in the final sound after all this pre DAC Sorcerer's Apprentice manipulation?
Is there any downside to this Philosopher's Stone?
Doing a mental experiment, the manipulation of digital source ahead of the DAC does not mess up the broth the way that filtering analog signals does. I have heard terrible cross-overs and "equalizers" over the years. Is there a downside to manipulating the digital source prior to the DAC? Are there unintended consequences or degradation of content? I get that one can botch the sound by turning up the "bass" to a Spinal Tap "eleven" but if one gets the balances right is there a difference in the final sound after all this pre DAC Sorcerer's Apprentice manipulation?
PM sent with FIR filter info. Not to thread jack, but to answer the one question about any signal degradation. The DSP software I use does all calculations with 64 bit double precision floating point. To put context around what 64 bit audio means, from JRiver’s wiki: http://wiki.jriver.com/index.php/Audio_Bitdepth#Bit-Perfect
“The precision offered by Media Center's 64bit audio engine is billions of times greater than the best hardware can utilize. In other words, it is bit-perfect on all known hardware.
To demonstrate the incredible precision of 64bit audio, imagine applying 100 million random volume changes (huge changes from -100 to 100 dB), and then applying those same 100 million volume changes again in the opposite direction.
Amazingly, you will have the exact same signal at 32bit after 200 million huge volume changes as when you started.
In other words, this incredible number of changes results in a bit-perfect output at 32bit, which is the highest hardware output bitdepth (most high-end hardware is 24bit).
This also means one volume change or a series of 100 million volume changes that add up to the same net result is bit-identical.”
In other words, no signal degradation, completely transparent, whether the FIR filter contains room correction or room correction plus digital XO, driver linearization, time alignment, etc.
“The precision offered by Media Center's 64bit audio engine is billions of times greater than the best hardware can utilize. In other words, it is bit-perfect on all known hardware.
To demonstrate the incredible precision of 64bit audio, imagine applying 100 million random volume changes (huge changes from -100 to 100 dB), and then applying those same 100 million volume changes again in the opposite direction.
Amazingly, you will have the exact same signal at 32bit after 200 million huge volume changes as when you started.
In other words, this incredible number of changes results in a bit-perfect output at 32bit, which is the highest hardware output bitdepth (most high-end hardware is 24bit).
This also means one volume change or a series of 100 million volume changes that add up to the same net result is bit-identical.”
In other words, no signal degradation, completely transparent, whether the FIR filter contains room correction or room correction plus digital XO, driver linearization, time alignment, etc.
digital future
Mitchba,
This is all so fascinating to me, but at the same time very foreign, because it has taken me all these years learning how to get analogue just "right" (as far as active crossovers).
My attitude toward digital is admittedly from my "gut instinct". I would have to hear it, to believe it. Although, I must say there seems to be such beauty in control [in that] you can tell it to do, what you want it to do:
Compensate for the room's influences (?)
Compensate for baffle step diffraction (?)
Compensate fro frequency response non linearity (?)
I would wonder what happens to the pace, rhythm and timing aspect ?
Mitchba,
This is all so fascinating to me, but at the same time very foreign, because it has taken me all these years learning how to get analogue just "right" (as far as active crossovers).
My attitude toward digital is admittedly from my "gut instinct". I would have to hear it, to believe it. Although, I must say there seems to be such beauty in control [in that] you can tell it to do, what you want it to do:
Compensate for the room's influences (?)
Compensate for baffle step diffraction (?)
Compensate fro frequency response non linearity (?)
I would wonder what happens to the pace, rhythm and timing aspect ?
Mitchba,
Its like waking up Christmas morning. There is a lot here to digest. I think I have ignored these developments thinking they were for studio types developing content. But it has been increasingly on my radar with the arrival of the miniDSP products. I see you have been very active in this and I had a chance to look over your posts here on diyAudio and on Computeraudiophile. Brave new world.
What are the hardware requirements of a "computer based" system? Where does the slicing and dicing happen? Clearly the amplifiers are external. And content is likely CD or hard drive based. My guess is you employ external DAC's. But having scanned your articles/posts and the software product Acourate over breakfast coffee, I am unsure where the DSP happens. Is this a PC based sound card or some external box with two channels digital in and multiple digital out's?
Its like waking up Christmas morning. There is a lot here to digest. I think I have ignored these developments thinking they were for studio types developing content. But it has been increasingly on my radar with the arrival of the miniDSP products. I see you have been very active in this and I had a chance to look over your posts here on diyAudio and on Computeraudiophile. Brave new world.
What are the hardware requirements of a "computer based" system? Where does the slicing and dicing happen? Clearly the amplifiers are external. And content is likely CD or hard drive based. My guess is you employ external DAC's. But having scanned your articles/posts and the software product Acourate over breakfast coffee, I am unsure where the DSP happens. Is this a PC based sound card or some external box with two channels digital in and multiple digital out's?
Scott L, yes to your questions and more. Wrt pace, rhythm and timing, I refer you to this post on Smooth (Flat) vs Accurate (Hi-Fidelity): http://www.diyaudio.com/forums/multi-way/271119-smooth-flat-vs-accurate-hi-fidelity-4.html#post4258668
The post contains some research on what target frequency responses are perceived as being perceptually flat at the listening position. It also describes the ideal or theoretical target step response (timing) of a speaker where all frequencies arrives at the listeners ear at the same, with a specific decay rate.
Compare those targets to the measures in this post: http://www.diyaudio.com/forums/multi-way/271119-smooth-flat-vs-accurate-hi-fidelity-5.html#post4259757 and: http://www.diyaudio.com/forums/multi-way/271119-smooth-flat-vs-accurate-hi-fidelity-9.html#post4261806. Reasonably close. The only way I know how to achieve this level of accuracy and precision is with DSP software and 64 bit FIR filtering.
Brucegseidner, the hardware requirements of computer based audio are pretty basic. My audio computer is nothing special: http://i1217.photobucket.com/albums/dd381/mitchatola/Win7PC.jpg
One can break down FIR filtering into design time activities and run time execution. The DSP software involves one to have a calibrated measurement microphone, mic preamp, ADC section to take acoustic measurements, analyze those measurements, design the FIR filter, which may involve several incremental iterations of these steps.
Once the filter design is complete, the output of the DSP software packages the Finite Impulse Response (FIR) filter into a wav file. JRiver’s 64 bit audio engine, described in the previous post, contains a Convolution engine that hosts this FIR filter file. This is where the “runtime” slicing and dicing occurs. Here is a screen shot: http://i1217.photobucket.com/albums/dd381/mitchatola/JRiver%20Convolution%20Engine_zpsvqlklkba.jpg
Note that the input is the stereo music file, but JRiver has been configured to output 6 digital channels (it can be configured for any number of channels). The Convolution engine hosts the FIR filter which contains the 3 way stereo XO (which matches up with the 6 channel JRiver config), the time alignment and room correction which is “convolved” in real-time with the music signal at 64 bit resolution.
I have an external AD DA multi-channel converter, but there are also multi-channel sound cards as well. In my case, I run the each of the 6 DAC analog outputs directly to the inputs of the six amps.I use JRiver’s digital volume control to control the level, which at 64 bit resolution is again completely transparent.
I don’t mean to highjack your thread as this is a long post, but barely scratching the surface of the power of computer based software DSP and FIR filtering. Reading those links will go a long ways to getting you there.
The post contains some research on what target frequency responses are perceived as being perceptually flat at the listening position. It also describes the ideal or theoretical target step response (timing) of a speaker where all frequencies arrives at the listeners ear at the same, with a specific decay rate.
Compare those targets to the measures in this post: http://www.diyaudio.com/forums/multi-way/271119-smooth-flat-vs-accurate-hi-fidelity-5.html#post4259757 and: http://www.diyaudio.com/forums/multi-way/271119-smooth-flat-vs-accurate-hi-fidelity-9.html#post4261806. Reasonably close. The only way I know how to achieve this level of accuracy and precision is with DSP software and 64 bit FIR filtering.
Brucegseidner, the hardware requirements of computer based audio are pretty basic. My audio computer is nothing special: http://i1217.photobucket.com/albums/dd381/mitchatola/Win7PC.jpg
One can break down FIR filtering into design time activities and run time execution. The DSP software involves one to have a calibrated measurement microphone, mic preamp, ADC section to take acoustic measurements, analyze those measurements, design the FIR filter, which may involve several incremental iterations of these steps.
Once the filter design is complete, the output of the DSP software packages the Finite Impulse Response (FIR) filter into a wav file. JRiver’s 64 bit audio engine, described in the previous post, contains a Convolution engine that hosts this FIR filter file. This is where the “runtime” slicing and dicing occurs. Here is a screen shot: http://i1217.photobucket.com/albums/dd381/mitchatola/JRiver%20Convolution%20Engine_zpsvqlklkba.jpg
Note that the input is the stereo music file, but JRiver has been configured to output 6 digital channels (it can be configured for any number of channels). The Convolution engine hosts the FIR filter which contains the 3 way stereo XO (which matches up with the 6 channel JRiver config), the time alignment and room correction which is “convolved” in real-time with the music signal at 64 bit resolution.
I have an external AD DA multi-channel converter, but there are also multi-channel sound cards as well. In my case, I run the each of the 6 DAC analog outputs directly to the inputs of the six amps.I use JRiver’s digital volume control to control the level, which at 64 bit resolution is again completely transparent.
I don’t mean to highjack your thread as this is a long post, but barely scratching the surface of the power of computer based software DSP and FIR filtering. Reading those links will go a long ways to getting you there.
mucho gratitude !
Mitchba,
Thanks kindly for taking the time to post !
I will contact my broadcast engineer friend, to help me absorb all this.
Mitchba,
Thanks kindly for taking the time to post !
I will contact my broadcast engineer friend, to help me absorb all this.
Its only hijacking if uninvited and irrelevant. Your generous introduction is neither. It might be old hat to some, but thank you. This has been helpful. In your system I was surprise there is no DSP sound card that I could see. Your computer CPU is doing the work, unless I have missed something. Then the JRiver software creating channels. But this is where it gets murky. What is the hardware I/O? I will be looking at "multi channel sound cards" but do you have some examples? And would the nanoDigi be considered an "external sound card" in this sort of set up?
The ability to create software crossovers, on the fly, is something that I look forward to. As long as it happens prior to the analog I am hopeful that it is "clean" and transparent. I have heard good active crossovers and bad active crossovers. Well, actually I did not hear the good active crossovers. They just did what they were designed to do without coloring, compressing, or mucking up the music.
Computer crossovers can be a royal pain in the backside, but they are getting better, much better. I used dedicated hardware at first because it's what I am used to and thus much easier for me. You can learn a lot that way.
A friend of mine - forum member RA7 - uses all convolution in JRiver for his multiway crossover and is very happy with it. There are plenty of other people here with stable software crossovers. If I were doing it over again, I'd go with a multichannel soundcard and software crossovers, for sure. The learning curve will be steep, but you can simply start with textbook filter functions in software and learn as you go. The software will allow you to be as simple or complex as you want.
A friend of mine - forum member RA7 - uses all convolution in JRiver for his multiway crossover and is very happy with it. There are plenty of other people here with stable software crossovers. If I were doing it over again, I'd go with a multichannel soundcard and software crossovers, for sure. The learning curve will be steep, but you can simply start with textbook filter functions in software and learn as you go. The software will allow you to be as simple or complex as you want.
Brucegseidner, yes, all DSP is executed in software on the computer. Today’s computers are more than capable as this kind of processing which takes maybe 1 or 2% of the CPU tops. The hardware I/O is typically PCIe for an internal sound card or (asynchronous) USB when connecting an external AD DA converter.
There are several manufacturers of multi-channel AD DA converters: http://www.steinberg.net/en/products/audio_interfaces/ur_serie/modelle/ur824.html is popular as is http://exasound.com/e28/Overview.aspx (but needs an ADC). JRiver’s forum is also a good place to look through the threads on multi-channel DAC’s: http://yabb.jriver.com/interact/index.php?board=9.0
One can also join the Acourate forum as many folks there have multi-channel systems: https://groups.yahoo.com/neo/groups/acourate/info
I am partial to pro sound AD DA converters as I spent 10 years as a recording/mixing engineer for a number of studios. http://www.lynxstudio.com/ is probably my favorite manufacturer of top shelf converters without audiophile prices. Especially the Aurora and Hilo.
As far as transparency is concerned, I would agree with all that Bob Katz says here: http://www.audiovero.de/en/testimonials-and-references.php#Ref1
PS agree to what Pano says. For sure there is a learning curve and you can go easy or hard, even with the same DSP software. However, the reward is worth it in my opinion. Good luck!
There are several manufacturers of multi-channel AD DA converters: http://www.steinberg.net/en/products/audio_interfaces/ur_serie/modelle/ur824.html is popular as is http://exasound.com/e28/Overview.aspx (but needs an ADC). JRiver’s forum is also a good place to look through the threads on multi-channel DAC’s: http://yabb.jriver.com/interact/index.php?board=9.0
One can also join the Acourate forum as many folks there have multi-channel systems: https://groups.yahoo.com/neo/groups/acourate/info
I am partial to pro sound AD DA converters as I spent 10 years as a recording/mixing engineer for a number of studios. http://www.lynxstudio.com/ is probably my favorite manufacturer of top shelf converters without audiophile prices. Especially the Aurora and Hilo.
As far as transparency is concerned, I would agree with all that Bob Katz says here: http://www.audiovero.de/en/testimonials-and-references.php#Ref1
PS agree to what Pano says. For sure there is a learning curve and you can go easy or hard, even with the same DSP software. However, the reward is worth it in my opinion. Good luck!
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