Hello Jean-Michel,
How is the measurement done, what is the quality of the anechoic room? For me it looks like the data is contamined with room reflections, and thus the ripple in phase and therein in group delay. Then, considering this what we actually see from this picture?
Another thing I'd like to know is what is the order of the high pass transfer function of these horns? Because that will naturally define the group delay of the system at low freqs.
- Elias
How is the measurement done, what is the quality of the anechoic room? For me it looks like the data is contamined with room reflections, and thus the ripple in phase and therein in group delay. Then, considering this what we actually see from this picture?
Another thing I'd like to know is what is the order of the high pass transfer function of these horns? Because that will naturally define the group delay of the system at low freqs.
- Elias
Jmmlc said:
Hello Elias,
I guess you are joking ;-)
How would you that we move those bass horns inside an anechoic room
http://nsm01.casimages.com/img/2009/05/17//090517055721545253677910.jpg
as they are conceived for the ground and lateral walls of the auditorium to be their natural extension.
A 130Hz high Q annulation in the response could eventually correspond to the interference between 2 signals having a diffference of path of 3,9meters which value don't correspond to anything precise in the auditorium. Eventually it can be due to the large rear chamber.
You'll find the wave (in TXT format) attached to that message.
I guess to derive the transfer function of such a bass horn will be a difficult task (the rear close chamber cancels partly the throat reactance of the horn), better to use Hornresp or Akabak to study their behaviour. (While Hornresp mouth area is limited...).
Remember also that those bass horns have very low cut-off frequency (lesser than 30Hz) and that the acoustic impedance becomes rapidly purely resistive above that cut-off .
Best regards from Paris, France
Jean-Michel Le Cléac'h
I guess you are joking ;-)
How would you that we move those bass horns inside an anechoic room
http://nsm01.casimages.com/img/2009/05/17//090517055721545253677910.jpg
as they are conceived for the ground and lateral walls of the auditorium to be their natural extension.
A 130Hz high Q annulation in the response could eventually correspond to the interference between 2 signals having a diffference of path of 3,9meters which value don't correspond to anything precise in the auditorium. Eventually it can be due to the large rear chamber.
You'll find the wave (in TXT format) attached to that message.
I guess to derive the transfer function of such a bass horn will be a difficult task (the rear close chamber cancels partly the throat reactance of the horn), better to use Hornresp or Akabak to study their behaviour. (While Hornresp mouth area is limited...).
Remember also that those bass horns have very low cut-off frequency (lesser than 30Hz) and that the acoustic impedance becomes rapidly purely resistive above that cut-off .
Best regards from Paris, France
Jean-Michel Le Cléac'h
This is a perceptual difference. Above a certain critical frequency (I'm guessing 300~500 Hz) the ear starts to distinguish between room-sound and the direct arrival. Above this frequency, time-decay performance (stored energy) of the loudspeaker affects the timbre and coloration of the music, and sensitivity to IM distortion increases, reaching a peak in the 1~3 kHz region.
In the 100~300 Hz region, loudspeaker and room sound start to merge together, partly because the wavelengths are getting long enough that room standing-wave modes are starting to dominate both perception and measurement (which is why it is industry practice to "splice" curves at 300 Hz, using nearfield measurements below that).
Loudspeaker coloration is a single, unitary perception, and what "sounds" fast might have nothing at all to do with transient response. It could any of several things - lack of stored energy in the loudspeaker cabinet, avoiding room modes, or low IM distortion. They could all sound alike, or subtly different.
This is why measurements are necessary, because correlation between what you "think" you heard and what's actually there may have a rather complex and non-obvious relationship. I've worked enough with amplifiers and speakers to confuse the sound of each - low distortion can sound "fast", and good impulse response can sound like good spatial resolution and a realistic ambient impression, but not "fast" in a subjective sense.
I think people are somewhat misled by harmonic distortion. 2nd harmonic, as generated by a test circuit, is really pretty hard to hear, even in the midband. 3rd sounds like a more "edgy", sharp sound, but again, that's in the midband where it has peak audibility. Audibility of THD in the bass region pretty much follows the Fletcher-Munson curve - very low below 100 Hz.
But IM distortion is another story. Even though the mechanism creating THD and IM are the same, the distortion itself has a quite different audibility, because it "thickens" the spectrum with harmonically unrelated tones. The denser the spectrum, the more sum-and-difference tones are generated. The BBC did a study a few years back and calculated with as few as three tones, the amount of IM distortion exceeds THD terms, and with more tones, it goes up geometrically. With a complex spectrum from music, IM distortion dominates by far, and the more complex the music, the more IM distortion is present. In the simplest model, it acts as a dynamic noise floor.
IM distortion is particularly relevant for loudspeakers, because below-passband stimuli, although not directly audible by themselves, crossmodulate up into the audible band, and create unrelated sum-and-difference spectral artifacts. One of the most important functions of the tweeter crossover is to protect against below-band energy creating IM distortion in the passband - and the passband of the tweeter typically falls in the region of peak distortion audibility.
Think of 10 Hz and 1 kHz going through a single vented-box midbass driver; the 10 Hz isn't audible at all, since it is well outside the passband of the VB system, as well as falling off the far edge of the Fletcher-Munson curve (10 Hz has to be very loud to be heard at all). But the 990 and 1010 Hz distortion sidebands are really there, and do no favors to the midrange. Now add 10 Hz, 20 Hz, 1 kHz, 2 kHz, and 3 kHz - we're starting to get a real thicket of sum-and-difference terms that are not present in the original musical material.
In the 100~300 Hz region, loudspeaker and room sound start to merge together, partly because the wavelengths are getting long enough that room standing-wave modes are starting to dominate both perception and measurement (which is why it is industry practice to "splice" curves at 300 Hz, using nearfield measurements below that).
Loudspeaker coloration is a single, unitary perception, and what "sounds" fast might have nothing at all to do with transient response. It could any of several things - lack of stored energy in the loudspeaker cabinet, avoiding room modes, or low IM distortion. They could all sound alike, or subtly different.
This is why measurements are necessary, because correlation between what you "think" you heard and what's actually there may have a rather complex and non-obvious relationship. I've worked enough with amplifiers and speakers to confuse the sound of each - low distortion can sound "fast", and good impulse response can sound like good spatial resolution and a realistic ambient impression, but not "fast" in a subjective sense.
I think people are somewhat misled by harmonic distortion. 2nd harmonic, as generated by a test circuit, is really pretty hard to hear, even in the midband. 3rd sounds like a more "edgy", sharp sound, but again, that's in the midband where it has peak audibility. Audibility of THD in the bass region pretty much follows the Fletcher-Munson curve - very low below 100 Hz.
But IM distortion is another story. Even though the mechanism creating THD and IM are the same, the distortion itself has a quite different audibility, because it "thickens" the spectrum with harmonically unrelated tones. The denser the spectrum, the more sum-and-difference tones are generated. The BBC did a study a few years back and calculated with as few as three tones, the amount of IM distortion exceeds THD terms, and with more tones, it goes up geometrically. With a complex spectrum from music, IM distortion dominates by far, and the more complex the music, the more IM distortion is present. In the simplest model, it acts as a dynamic noise floor.
IM distortion is particularly relevant for loudspeakers, because below-passband stimuli, although not directly audible by themselves, crossmodulate up into the audible band, and create unrelated sum-and-difference spectral artifacts. One of the most important functions of the tweeter crossover is to protect against below-band energy creating IM distortion in the passband - and the passband of the tweeter typically falls in the region of peak distortion audibility.
Think of 10 Hz and 1 kHz going through a single vented-box midbass driver; the 10 Hz isn't audible at all, since it is well outside the passband of the VB system, as well as falling off the far edge of the Fletcher-Munson curve (10 Hz has to be very loud to be heard at all). But the 990 and 1010 Hz distortion sidebands are really there, and do no favors to the midrange. Now add 10 Hz, 20 Hz, 1 kHz, 2 kHz, and 3 kHz - we're starting to get a real thicket of sum-and-difference terms that are not present in the original musical material.
There is also a third distinction between Open Baffle, Closed-Box, Vented-Box, and Basshorn: CB and VB are very close to omnidirectional below 300~500 Hz (depending on size), while OB is a dipole down to the lowest working frequency, and horns are directional down to the cutoff frequency (the pattern spreads out below cutoff).
This means that there are three main groups of directivity patterns (omni, dipole, and quasi-conical), and they excite room modes in completely different ways, making comparisons between types in different rooms very difficult, since the room/loudspeaker interaction is going to be unique to each combination.
So there are three distinctions between the four main types of bass reproducer: group delay variation, IM distortion magnitude and spectral distribution, and LF dispersion pattern. No wonder they sound different - they are different in significant ways, and will interact with different rooms in different ways.
The loudspeaker group delay variation merges with the room modes at the lowest frequencies; as a result, the LF dispersion pattern has a strong influence on the overall loudspeaker/room group delay performance. Sure, you can take an OB, CB, VB, or basshorn outdoors and measure it, but do you plan to do all of your listening outdoors as well? Once you put it back in a room, the LF dispersion pattern matters a great deal to how it sounds.
Note that I haven't mentioned drivers - the differences between drivers (distortion spectra, et al) are much smaller than the essential differences between the four types of bass reproducer.
This means that there are three main groups of directivity patterns (omni, dipole, and quasi-conical), and they excite room modes in completely different ways, making comparisons between types in different rooms very difficult, since the room/loudspeaker interaction is going to be unique to each combination.
So there are three distinctions between the four main types of bass reproducer: group delay variation, IM distortion magnitude and spectral distribution, and LF dispersion pattern. No wonder they sound different - they are different in significant ways, and will interact with different rooms in different ways.
The loudspeaker group delay variation merges with the room modes at the lowest frequencies; as a result, the LF dispersion pattern has a strong influence on the overall loudspeaker/room group delay performance. Sure, you can take an OB, CB, VB, or basshorn outdoors and measure it, but do you plan to do all of your listening outdoors as well? Once you put it back in a room, the LF dispersion pattern matters a great deal to how it sounds.
Note that I haven't mentioned drivers - the differences between drivers (distortion spectra, et al) are much smaller than the essential differences between the four types of bass reproducer.
Hello
Michael (mige0) asked me some time ago to help him to design a horn in order to load a ribbon tweeter AMT both at its front and at its rear ("dual horn").
He wanted that the horn could be inserted in an existing open baffle. Only a rectangular place of 30centimeters (width) x 15centimeter (height) was usable on the OB.
I calculated fro him a "Le Cléac'h quasi cylindrical wavefronts horn".
You can see a beautiful 3D view of that dual horn at:
http://www.diyaudio.com/forums/showthread.php?postid=1839560#post1839560
And read the message:
http://www.diyaudio.com/forums/showthread.php?postid=1837992#post1837992
The response curves at 0°(equalized ) , 10°, 20°, 30° and 40° are visible at:
http://members.aon.at/kinotechnik/diyaudio/diy_audio/dual-horn/dual-horn_0-10-20-30-40-deg.gif
Additionally the response of the original tweeter taken as a 0dB reference Michael calcualted the gain of that dual horn (gray cruve) compared to open baffle (yellow curve).
As you can see that the dual horn provide an intersting gain in the frequency range 700 - 3000Hz. This is due to a better acoustical loading of the ribbon. It lead to less ribbon displacement for the same SPL and thus to less distortion.
http://members.aon.at/kinotechnik/d...n/monopole-ref_dipole-gain_dual-horn-gain.gif
Best regards rfom Paris, France
Jean-Michel Le Cléac'h
Michael (mige0) asked me some time ago to help him to design a horn in order to load a ribbon tweeter AMT both at its front and at its rear ("dual horn").
He wanted that the horn could be inserted in an existing open baffle. Only a rectangular place of 30centimeters (width) x 15centimeter (height) was usable on the OB.
I calculated fro him a "Le Cléac'h quasi cylindrical wavefronts horn".
You can see a beautiful 3D view of that dual horn at:
http://www.diyaudio.com/forums/showthread.php?postid=1839560#post1839560
And read the message:
http://www.diyaudio.com/forums/showthread.php?postid=1837992#post1837992
The response curves at 0°(equalized ) , 10°, 20°, 30° and 40° are visible at:
http://members.aon.at/kinotechnik/diyaudio/diy_audio/dual-horn/dual-horn_0-10-20-30-40-deg.gif
Additionally the response of the original tweeter taken as a 0dB reference Michael calcualted the gain of that dual horn (gray cruve) compared to open baffle (yellow curve).
As you can see that the dual horn provide an intersting gain in the frequency range 700 - 3000Hz. This is due to a better acoustical loading of the ribbon. It lead to less ribbon displacement for the same SPL and thus to less distortion.
http://members.aon.at/kinotechnik/d...n/monopole-ref_dipole-gain_dual-horn-gain.gif
Best regards rfom Paris, France
Jean-Michel Le Cléac'h
Hmm ... could the "Le Cléac'h quasi-cylindrical wavefront horn" be scaled up to match an Altec 414 12" driver? I'm thinking of something like the short horn built into the Altec A5 or A7 cabinets that would cover the range from 150 to 800 Hz. I know it would end up being about the width of the A5/A7 in order to have enough mouth area, but I'm sure a Le Cléac'h profile would have much less diffraction than the Altec original, while retaining aspects of the Altec sound.
Jean-Michel
Yes - what was intended as an experimet only with rather modest expectations from my side turned out very very well.
Big thanks ! and my compliment to your contour - as I think it makes all the difference between "only" good loading and directivety control and a wonderful sound.
I have played around with your spreadsheet but - being only a monkey pressing buttons when it comes to horn contour design - havent found the right ones.
Maybe you could do an english text for the parameters in question as well and underlay the fields which have to be individually adjusted in one color and the results of calculation (coordinates) in an other color?
Is there a possibility given in your spreadsheet to calculate the coordinates of the wave front as this is somthing to play with for cylindrical horns especially
the sexy kissing lips I derivied from:
http://www.diyaudio.com/forums/showthread.php?postid=1790568#post1790568
Michael
Yes - what was intended as an experimet only with rather modest expectations from my side turned out very very well.
Big thanks ! and my compliment to your contour - as I think it makes all the difference between "only" good loading and directivety control and a wonderful sound.
I have played around with your spreadsheet but - being only a monkey pressing buttons when it comes to horn contour design - havent found the right ones.
Maybe you could do an english text for the parameters in question as well and underlay the fields which have to be individually adjusted in one color and the results of calculation (coordinates) in an other color?
Is there a possibility given in your spreadsheet to calculate the coordinates of the wave front as this is somthing to play with for cylindrical horns especially
An externally hosted image should be here but it was not working when we last tested it.
the sexy kissing lips I derivied from:
http://www.diyaudio.com/forums/showthread.php?postid=1790568#post1790568
Michael
Lynn Olson said:
Do you mean something with a flat top and bottom like Michaels's ribbon horn, or the "under the floor" horns in France? If so, that would look a bit different from the A7 type horn, yeah?
The mouth termination of the Altec isn't great, but it's helped some with the side wings. Would be interesting to do a better horn.
Hello Jean-Michel,
I tried your Excel-sheets for the horn calculation too. Thank you for providing this tool.
The "cylindrical one" is a bit older. Is that the reason, that the calculated contour is not very smooth? Lowering the resolution (step size along the contour, which is 1/4000 in the original) ends up with growing undulation of the contour itself. It looks much better with a 25-member moving average, but this is not very elegant.
The WG should be used for frequencies from approximately 800Hz up. This frequency results in relatively small dimensions but the WG may and should be 360mm wide. So I tried 286Hz and did some scaling, in x and y directions with slightly different factors. I wondered that calculations for different frequencies (here 375Hz) got _very_ similar results to the stretched one, when scaled proportionally, shifted and rotated a bit (less than mm deviations).
Cheers, Timo
I tried your Excel-sheets for the horn calculation too. Thank you for providing this tool.
The "cylindrical one" is a bit older. Is that the reason, that the calculated contour is not very smooth? Lowering the resolution (step size along the contour, which is 1/4000 in the original) ends up with growing undulation of the contour itself. It looks much better with a 25-member moving average, but this is not very elegant.
The WG should be used for frequencies from approximately 800Hz up. This frequency results in relatively small dimensions but the WG may and should be 360mm wide. So I tried 286Hz and did some scaling, in x and y directions with slightly different factors. I wondered that calculations for different frequencies (here 375Hz) got _very_ similar results to the stretched one, when scaled proportionally, shifted and rotated a bit (less than mm deviations).
Cheers, Timo
panomaniac said:
Gary Dahl used to own Altec A7-828's (the late model), and the top part was flat (horizontal), the horn flares were on the sides, and there was a modest straight flare on the bottom. In effect, the vertical dimension was a slight conical expansion that pointed downwards (I guess this made sense for a small theater speaker), and the horizontal dimension was a short exponential. The rest of the cabinet was an early-generation bass reflex. They were used in small movie theaters with 1-foot wings on either side, and power amplifiers with a Zout of about an ohm or two (damping factor of 4 to 8).
The side wings and the high-Z source "filled-in" the hole between the bass-reflex bump and the horn cutoff around 200 Hz - at least that's my understanding of how the A7 was intended to work. In the A2 and A4, the horn is also short, but oriented vertically, and the side wings are much larger.
Here's a photo from the 1975 Altec catalog (source: Lansing Heritage website) that shows the A2, A4, A5X, and A7 series loudspeakers. The setting of a Hollywood backlot is entirely appropriate, since the A-series were the standard of the industry from 1945 onwards, and were behind the screens of Cinerama, VistaVision, Todd-AO 70, and Super Panavision 70 movie theaters.
Anyway, I'm interested in a LeCleac'h version of the Altec A-series short horn, minus the bass-reflex enclosure. It seems like a scaled-up version of mige0's tweeter horn might do the job.
Attachments
Nice article by J.K. Hilliard on the installation of Stereophonic Sound in the Theater, presented at the S.M.P.T.E. convention in Los Angeles on May 1, 1953. Interesting read. Note the power levels used for 1600-seat theaters. I've seen many movies at Todd-AO 70 and Super Panavision 70 widescreen theaters, and I can assure you not only was it plenty loud, but the sound quality was stunning - for example, here's the Introduction to Ben-Hur. Not bad for 1959, is it? Now imagine an IMAX-quality picture on a seventy-foot curved screen.
Hello tiki,
As you probably know I don't possess myself any website, so I rely on my friends to house the documents I want to share. Some of those documents are old.
Also as you may know the main goal of that spreadsheet is the calculation of very large bass horns not horns for tweeters, this explains why at the time it was published I didn't pay too much attention to that micro roughness.
But you are right the quasicylindrical wavefronts version of the spreadsheet on Nicolas Davidenko's website is quite old. I use to put on the files in the [son-qc] forum on Yahoo a more recent version but for conveniency I can provide to whom requests the more recent version which is free of that infra-millimeter roughness. (2Mo)
Best regards from Paris, France
Jean-Michel Le Cléac'h
As you probably know I don't possess myself any website, so I rely on my friends to house the documents I want to share. Some of those documents are old.
Also as you may know the main goal of that spreadsheet is the calculation of very large bass horns not horns for tweeters, this explains why at the time it was published I didn't pay too much attention to that micro roughness.
But you are right the quasicylindrical wavefronts version of the spreadsheet on Nicolas Davidenko's website is quite old. I use to put on the files in the [son-qc] forum on Yahoo a more recent version but for conveniency I can provide to whom requests the more recent version which is free of that infra-millimeter roughness. (2Mo)
Best regards from Paris, France
Jean-Michel Le Cléac'h
tiki said:
Hello everyone,
How one can define phase at low freqs in a room anyway? Consider a simple illustrative example of a direct wave and a later arriving reflection. The summed signal will exhibit temporal phase shift at the time the reflection hits in. Then imagine a room with multiple reflections - at a given frequency and at a given time the signal will have a random phase! How one can have phase linearity and constant group delay in a room at low freqs? This leads to the situation where phase relations of the harmonics of a musical instrument when reproduced in a room will be all messed up.
How to measure phase at low freqs? Since the phase will have temporal randomness the measurement result depends on the time window samples are taken. What is the correct time window? Since in a practical sized living rooms multiple reflections arrive already before one full cycle of direct wave has passed, there is no change to have uncorrupted wave to measure the phase in the first place.
It is very true that different types of speakers do better than others in this respect. The one with the highest directivity wins of course. The most practical high directivity source at low freqs is a dipole line array. A second order gradient source would be even better, but it's very rarely seen in practise even with diyers.
I think it's also very true that it does not make sense to do measurements in an anechoic chamber. As we are doing high quality systems they must be designed for the room they will be based in.
- Elias
How one can define phase at low freqs in a room anyway? Consider a simple illustrative example of a direct wave and a later arriving reflection. The summed signal will exhibit temporal phase shift at the time the reflection hits in. Then imagine a room with multiple reflections - at a given frequency and at a given time the signal will have a random phase! How one can have phase linearity and constant group delay in a room at low freqs? This leads to the situation where phase relations of the harmonics of a musical instrument when reproduced in a room will be all messed up.
How to measure phase at low freqs? Since the phase will have temporal randomness the measurement result depends on the time window samples are taken. What is the correct time window? Since in a practical sized living rooms multiple reflections arrive already before one full cycle of direct wave has passed, there is no change to have uncorrupted wave to measure the phase in the first place.
It is very true that different types of speakers do better than others in this respect. The one with the highest directivity wins of course. The most practical high directivity source at low freqs is a dipole line array. A second order gradient source would be even better, but it's very rarely seen in practise even with diyers.
I think it's also very true that it does not make sense to do measurements in an anechoic chamber. As we are doing high quality systems they must be designed for the room they will be based in.
- Elias
Jmmlc said:
Lynn Olson said:
Attachments
Hi Lynn,Lynn Olson said:
In his latest AES-paper, Angelo Farina describes an interesting new measurement technique for this "dynamic noise floor" (not only from IM alone) :
Silence Sweep: a novel method for measuring electro-acoustical devices
BTW, thanks @Janneman
BTW2, Farina's ESS is spinning my mind big time these days...
BTW3, maybe the silence sweep is the method in charge to finally nail down what enABL does or is suspected to do....
BTW4... (I know it gets boring)... reading the references in the document I see MESS (sic!) is already invented, but it's unclear if it was used for 1:1 two-tone IM test so far....
- Klaus
Have a look at my recently finished 250Hz Kugelwellen horns here:
http://www.diyaudio.com/forums/showthread.php?threadid=138582
http://www.diyaudio.com/forums/showthread.php?threadid=138582
Hello Elias,
When it comes to propagating waves considering values of phase in an absolute way is pretty much useless.
IMHO what is more important is how energy is liberated inside a frequency / time plane. That's why I give some importance to CSD (...sonograms, waterfall, spectrograms....)
We are so accustomed since decades to read that correct phase reproducing of low frequency signals (e.g. : f < 200Hz) is useless because the room destroys any phase relationship at those frequencies... it beacame very rare someone questions the veracity of such postulate.
In fact, audiophiles have very little or no experience on low frequency sound respecting relative phase of the harmonically related components. Low frequency loudspeakers are the worst when it comes to variation of group delay.
So when I could listen to a large "Le Cléac'h" basshorn with a very constant group delay above 30Hz, I was really puzzled on how natural was the sound of the acoustic bass played on the bass horn... (and this is not only a question of very low distortion).
Best regards from Paris, France
Jean-Michel Le Cléac'h
When it comes to propagating waves considering values of phase in an absolute way is pretty much useless.
IMHO what is more important is how energy is liberated inside a frequency / time plane. That's why I give some importance to CSD (...sonograms, waterfall, spectrograms....)
We are so accustomed since decades to read that correct phase reproducing of low frequency signals (e.g. : f < 200Hz) is useless because the room destroys any phase relationship at those frequencies... it beacame very rare someone questions the veracity of such postulate.
In fact, audiophiles have very little or no experience on low frequency sound respecting relative phase of the harmonically related components. Low frequency loudspeakers are the worst when it comes to variation of group delay.
So when I could listen to a large "Le Cléac'h" basshorn with a very constant group delay above 30Hz, I was really puzzled on how natural was the sound of the acoustic bass played on the bass horn... (and this is not only a question of very low distortion).
Best regards from Paris, France
Jean-Michel Le Cléac'h
Elias said: