Earl,
I am so curious how the B&C driver can compare and you have been quoted as saying that this driver is comparable to a TAD beryllium 2001 driver? I just have a hard time with the diaphragm material and the comparison of the Young's modulus being so different it is hard to get my head around this driver having similar characteristics. The stiffness and damping of the material would be so different, and the mass must be much higher with a polyimide material. What comparisons did you do to come to the conclusions about this driver that you have? Do you have a paper on the comparisons, or anything technical that you can divulge here? PM me if you don't want to discuss this here.
I am so curious how the B&C driver can compare and you have been quoted as saying that this driver is comparable to a TAD beryllium 2001 driver? I just have a hard time with the diaphragm material and the comparison of the Young's modulus being so different it is hard to get my head around this driver having similar characteristics. The stiffness and damping of the material would be so different, and the mass must be much higher with a polyimide material. What comparisons did you do to come to the conclusions about this driver that you have? Do you have a paper on the comparisons, or anything technical that you can divulge here? PM me if you don't want to discuss this here.
Hi Earl,
Hi Earl,
For the purpose of phase equalization in multi-channel system, I prefer to equalize all channels to 0deg exactly – in absolute terms. Therefore, there is no inter-channel phase difference between any of the channels. This is supposed to be good for imaging.
If I referenced phase equalization to woofer driver in each channel, the accuracy of inter-channel phase equalization would depend on the accuracy of woofer’s phase repeatability – a parameter beyond my control.
Best Regards,
Bohdan
Bohdan,
Phase response can be looked at from any offset of the impulse and be equally "correct", and one can choose any "view" (offset and polarity of the impulse) of the phase that fits its needs for a particular analyze or correction.
Minimum phase (as in "obtained from an hilbert transform", as you imply here) is just one way of looking at it, and not always the most useful one: I see absolutely no interest in showing the tweeter (and system, possibly including linear phase antialiasing filters in DAC/ADC that will screw that up!...) LP phase shift here.
Furthermore why would you consider the complete speaker here minimum phase to boot, with its IIR crossover and the allpass it implies?...
By the way I don't think that compression driver follows a minimum phase behavior up high. Not because of the waveguide, but simply because of the heavy (albeit probably well dampened thanks to mylar) breakup modes the diaphragm encounters up there.
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I do have the evidence. But first here is the way I see it.
Sure there are differences in the two drivers, but is this difference audible or is it just marketing?
The way to resolve this situiation is to build two identical designs, but use a different driver set in the two and then via a blind study do a listening test (several subjects) and compare.
I did just that. I built one Summa with all TAD drivers and another with all B&C drivers. Each system was optimized for the driver set. In a blind listening test the data showed a not significant difference between the two sets, although the B&C were slightly prefered.
These results have been posted on my web site since they were first done with 16 listeners some eight years ago.
As far as I am concerned that test put the issue to rest. Drivers, as I see it, are pretty much a commodity. Get the "system" right and any decent drivers work just fine.
I once used a $16 Chinese DE250 knock-off. The sound was not so different that I would conclude it was noteworthy. The TADs cost about fourty times as much as that knock-off.
Sounds extreme and hard to support technically or with psychoacoustics. But if its what you want to do then go for it.
I don't bother with phase and I don't find anything else that is any better.
Hi Pos,
I am sorry if I created unintended misunderstanding – of course, complete loudspeaker system is NOT a “minimum-phase” device. Loudspeaker driver typically is.
I am not an expert on break-up modes in waveguides, so I can only go by what I have measured in a typical driver, calculated and visualized in the paper http://www.bodziosoftware.com.au/Acoustic Center Evaluation.zip
As you can see from this paper, including or excluding driver’s break-up region does not really influence guiding principles for calculating minimum-phase response.
If I understand correctly Earl's post #5796, the waveguide can be considered MP device?.
Finally, I am happy for you to disagree with what I presented so far. You continue to disregard the tweeter as a sound radiating device – that’s your choice.
Best Regards,
Bohdan
Hi Earl,
That's OK.
Comments below, have probably been circulated many times before, so I apologize, if I make it boring. Boston Audio Society - The BAS Speaker
Boston Audio Society has an interesting view on time-corrected loudspeakers.
“….If the stereo loudspeakers differ in their time-shift behaviour by more than about thirty millionths of a second (or a finer tolerance, perhaps, for critical listeners), the stereo image will be perceptibly smeared. The two speakers must "speak" together at all frequencies if the subtlest details in the stereo field are to be preserved.
This, quite simply, may be the principal advantage to be gained from "linear-phase" or time-corrected" loudspeakers. The manufacturers who are striving to reduce the time dispersion of loudspeakers to zero may also be ensuring that there will be no significant differences in signal propagation timing between the two speakers in a stereo pair. The delicate timing information in a stereo recording is thus accurately retained and is transmitted to the listener unaltered…”
They also point to some of the advantages of such loudspeakers:
1. Depth.
This may surprise some listeners when they first hear it, since many speakers (and records) elicit only a general left-to-right spread. But "stereo", as originally conceived, implied a three-dimensional sound in which voices or instruments could be localized at different apparent distances from the listener as well as at various lateral positions. Listeners to time-aligned speakers consistently report hearing a stereo image with unusual depth.
2. Resolution.
The stereo image is reproduced precisely, each voice or instrument having its proper place and width. In complex sound sources such as symphony orchestra, individual instruments can be resolved with unexpected clarity. In the old cliche, "I hear details I never knew were in the recording. " Some listeners have incorrectly attributed the improved resolution of detail to more accurate transient response, but the better definition of details is simply the result of the reduction of blending in the stereo image.
3.Separation of ambience.
With loudspeakers whose stereo image is slightly blended because of time-smear, any hall ambience or reverberation in the recording tends to become slightly mixed with the instrumental sounds, causing coloration of those sounds. Consequently, with such speakers closely-microphoned recordings tend to sound better because of their distinctly defined sound. But with time-corrected loudspeakers, the ambience is resolved as a separate sound, and larger amounts of hall ambience in recordings can be enjoyed…….”
Best Regards,
Bohdan
It's hard to convince someone who cannot hear the difference to believe in something technically more correct. Bear in mind the all evolution of technoly sprout from reasonable belief, and inmost cases experimented to satisfaction of th people exploring the technology before mathematical means to predict results within reasonable tolerance were developed. The more precision one desires drive the means to achieve such. It's like you cannot rely purely on Newtons Laws when you are dealing with ballistic missiles, not even when you you want to predict how a golf ball is going to fly.
I really much appreciate Bohdans work, it allowed me to explore more possibilities and understand the audible effects of various flaws in audio design. All this is really hard to convey to others until they have similar convictions and experiences.
I agree that designs that provide better directivity over as much spectrum as possible should provide better audio playback experience, but the means do how this is accomplished is equally important. The same goes with obtaining a linear phase. We must understand that whenever we take measurements as discussed here is SISO, where in fact the data taken already contains a variety of sources, this can be verified by doing measurements starting at some 5mm from a vibrating diaphragm going out at 1cm increments. People can argue that this is not audible, and we must agree that such audibility will vary among people. The thing is then person that makes the decision is either the designer or the user. Sure it is reasonable for people who don't hear it to say that it is just the imagination of others, same as people whom believe or not believe in religion.
One thing I like about a well designed wave guide is the way it the SPL drops. If you compensate for that in the right way, you also get it closer to a linear phase.
I really much appreciate Bohdans work, it allowed me to explore more possibilities and understand the audible effects of various flaws in audio design. All this is really hard to convey to others until they have similar convictions and experiences.
I agree that designs that provide better directivity over as much spectrum as possible should provide better audio playback experience, but the means do how this is accomplished is equally important. The same goes with obtaining a linear phase. We must understand that whenever we take measurements as discussed here is SISO, where in fact the data taken already contains a variety of sources, this can be verified by doing measurements starting at some 5mm from a vibrating diaphragm going out at 1cm increments. People can argue that this is not audible, and we must agree that such audibility will vary among people. The thing is then person that makes the decision is either the designer or the user. Sure it is reasonable for people who don't hear it to say that it is just the imagination of others, same as people whom believe or not believe in religion.
One thing I like about a well designed wave guide is the way it the SPL drops. If you compensate for that in the right way, you also get it closer to a linear phase.
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Hi Bohdan
It is not the waveguide I am questioning here, but the compression driver diaphragm. Given the material and dimensions I doubt it can be considered minimum phase. How far will it deviate from that is another question and you might consider it "close enough" for your needs, and that is fine. I don't think it matters in real life, but as you want to be fully theoretically correct in your analysis then you should consider that this device will not be minimum phase up high.
And while we are at it, can the effect of reticulated foam be considered minimum phase?
But where exactly do you read I "disregard the tweeter" ?
In the curve I posted I disregard the global LP function of the system. That includes tweeter LP, amplifier and other analog devices LP, measurement microphone LP, and ADC/DAC LP (possibly linear phase). It is my opinion that most of the time those LPs have absolutely no interest and will do nothing but confuse the analysis. Unless that is precisely the part you want to analysis of course, but that was certainly not the case here, and most LP factors were unknown to begin with! (and part of it comes from the measurement ring...)
As it is not going to be crossed over to another unit up high it is not a crossover but just a filter. I would add that considering the tiny (to say the least) group delays phase shifts imply at those frequencies I see no interest in dealing with them (contrary to the global HP of the system). The only thing you would gain by correcting the phase of that LP would be pre ringing...
I did not "disregard" the important parts of it (for me, considering the problem at hand), namely the HP of the tweeter, and any phase variation it might have in its passband.
Measurement is *always* a about choices, isolating problems, and focusing on what you want to analysis, nothing new here.
It is not the waveguide I am questioning here, but the compression driver diaphragm. Given the material and dimensions I doubt it can be considered minimum phase. How far will it deviate from that is another question and you might consider it "close enough" for your needs, and that is fine. I don't think it matters in real life, but as you want to be fully theoretically correct in your analysis then you should consider that this device will not be minimum phase up high.
And while we are at it, can the effect of reticulated foam be considered minimum phase?
If that makes you happy...
But where exactly do you read I "disregard the tweeter" ?
In the curve I posted I disregard the global LP function of the system. That includes tweeter LP, amplifier and other analog devices LP, measurement microphone LP, and ADC/DAC LP (possibly linear phase). It is my opinion that most of the time those LPs have absolutely no interest and will do nothing but confuse the analysis. Unless that is precisely the part you want to analysis of course, but that was certainly not the case here, and most LP factors were unknown to begin with! (and part of it comes from the measurement ring...)
As it is not going to be crossed over to another unit up high it is not a crossover but just a filter. I would add that considering the tiny (to say the least) group delays phase shifts imply at those frequencies I see no interest in dealing with them (contrary to the global HP of the system). The only thing you would gain by correcting the phase of that LP would be pre ringing...
I did not "disregard" the important parts of it (for me, considering the problem at hand), namely the HP of the tweeter, and any phase variation it might have in its passband.
Measurement is *always* a about choices, isolating problems, and focusing on what you want to analysis, nothing new here.
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OK, so this is not about removing all delay at the same time then. Anyway, this is about a half wavelength at 20k isn't it, so which effect is the one of concern?
I would think so, but then again MP has no meaning to an acoustic three dimensional system, so its hard to know for sure. Clearly HOM are not MP. I have not given it much thought, again, since I don't find phase all that interesting.
Bohdan
There is no psychoacoustics to support that BAS claim. It seems completely made-up and typical audiophile technospeak. If it were true, then just turning ones head or a slight shift sideways would completely change the "image" - which does not happen.
I am very surprised that you would quote something like that - especially to me. I am disappointed. Your personal discussions of phase, which are much more scientific, are far more interesting.
There is no psychoacoustics to support that BAS claim. It seems completely made-up and typical audiophile technospeak. If it were true, then just turning ones head or a slight shift sideways would completely change the "image" - which does not happen.
I am very surprised that you would quote something like that - especially to me. I am disappointed. Your personal discussions of phase, which are much more scientific, are far more interesting.
In the process of audio technology development, there needs to be a link between technology implementation, expected improvement, and perceived achievement from target group of users. I have had projects done by achademic institutes which have generated no useful results. Very few understand the link with real improvement goals because they are used to solving textbook problems which were just designed to practice the mathematical process. Even when you go through most of the discussion hear, how many can explain what perceived improvement goal is expected?
What Bohdan has referenced is one example of what users perceived as an improvement due to linear phase. That perceived improvement is similar to what I had experienced as well. Additionally further understanding of the phase characteristics of equalized wave guides my shed more light on the sonic characteristics of the design. I personally look forward to any further findings from Bohdan. It will be a while till I get back onto wave guides, but I might just go back and review my previous work which was before I experienced a linear phase system.
What Bohdan has referenced is one example of what users perceived as an improvement due to linear phase. That perceived improvement is similar to what I had experienced as well. Additionally further understanding of the phase characteristics of equalized wave guides my shed more light on the sonic characteristics of the design. I personally look forward to any further findings from Bohdan. It will be a while till I get back onto wave guides, but I might just go back and review my previous work which was before I experienced a linear phase system.
Unfortunately there is a vast difference in perceptions of tests done blind and un-blind. (The audio world would be a much easier place if that were not true, but it is, and it has to be dealt with.) Unless the test is blind, it is not a reliable test. To my knowledge, no blind test has ever reported phase as an audible quantity. And I know blind tests that have been done which have reported that phase was not audible.
Earl, unfortunately the reports you may have read do not provide enough data that may exclude the possibility that systems used did not have other problems that could have masked the effects of phase variation, meaning they may have used poor performing systems in the first place. Additionally, if earphones were used like the ones you used, the Etymotic ones, I do believe that phase variation cannot be detected. I have personally sampled those.
The phase tests were well done - you can't just wave away the results by claiming that they weren't.
If you still believe that you are right then do a well designed blind test and Wow the whole rest of the world if it comes out as you say. But for now, the data that is available, perfect or not, does not support your contention.
If you still believe that you are right then do a well designed blind test and Wow the whole rest of the world if it comes out as you say. But for now, the data that is available, perfect or not, does not support your contention.
If I understand your post correctly, you are doing the following:
1) You're using a shallow high-pass filter, possible as shallow as 6db/octave, and you're setting the xover point at a very high frequency. Perhaps as high as 12khz.
2) Technically, some might consider this a 12khz xover point. But that's not really true, as the high pass filter complements the compression drivers response shape. Basically, as you get closer and closer to 1khz, the output of the compression driver is rising. So the highpass filter isn't really 'highpassing' the compression driver. It's flattening it.
3) The net effect is that the output level of the compression driver is lower, but it's flatter and it's more extended.
IMHO, this is a really effective way to get 'the best of both worlds.' In home audio, there's really no use for a tweeter with an efficiency of 108dB. So we sacrifice sensitivity, but we end up with a driver that's flatter, more extended, with very high power handling.
Also, it might not be immediately obvious, but sacrificing sensitivity doesn't necessarily sacrifice output. This is because a compression driver that's playing to 900hz, as in the Summa, is basically going to be limited by displacement, not thermal power handling. So even though the lowpass filter at 12khz is reducing output levels at one watt, the maximum output is basically the same. (This assumes that one has an amplifier large enough to *reach* that output level.)
Also, my graph of the waveguide and compression driver isn't from a Summa, it's from another waveguide project. Simply used the first pic I could find on Google, but the response shape of an unfiltered compression driver on a waveguide is relatively similar.
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Hi John
Yea that's basically correct. And you are right about the power handling. The way I do the crossover damage is simply not an issue in home use. Lots of companies have learned the benefits of using a "waveguide", but most just go for a wider polar response to match a smaller woofer. I opted the other way around - a narrow waveguide and the much larger woofer required to match the polar response. They both get the same basic polar response - to wit the Behringer versus the Abbey - but the larger driver approach has some 20 dB more headroom. That's not always necessary, of coarse, but there are some audible advantages of that kind of headroom.
Yea that's basically correct. And you are right about the power handling. The way I do the crossover damage is simply not an issue in home use. Lots of companies have learned the benefits of using a "waveguide", but most just go for a wider polar response to match a smaller woofer. I opted the other way around - a narrow waveguide and the much larger woofer required to match the polar response. They both get the same basic polar response - to wit the Behringer versus the Abbey - but the larger driver approach has some 20 dB more headroom. That's not always necessary, of coarse, but there are some audible advantages of that kind of headroom.
Hi Earl,
There are papers published on the internet, that point to the importance of phase in transients, as opposed to steady-state signals, where the phase distortion is inaudible (agree on this point).
Source: http://www.cirrus.com/en/pubs/whitePaper/DS668WP1.pd
7. Inter-channel accuracy of sound reproduction.
“…….5. Audibility of Phase Distortion
One of the confusing issues regarding the audibility of phase is that the discussion is generally considered to be a single topic when in reality should be discussed as two distinct situations. The audibility of phase distortion must be evaluated as follows:
1) Inter-channel phase distortion. Characterized as differences in phase response between two or more channels.
2) Intra-channel phase distortion. Characterized by non-linear phase response within a channel with the stipulation that the phase response is matched between all channels within the system (i.e. inter-channel phase distortion is equal to 0 msec)
6. Inter-Channel Phase Distortion
We use the amplitude and phase relationship between the sounds received by our ears to localize the source of the sound. Modern audio systems use this attribute to create what is known as imaging, or the perception that an instrument or vocal is coming from a location that is different than the actual speaker location. The audible effects of inter-channel phase distortion can be easily demonstrated by simply reversing the speaker connections on one channel of an otherwise properly configured stereo system. The loss of imaging is immediately noticeable even to those without a trained ear. Granted this test is rather dramatic and 180 degrees of inter-channel phase distortion is not indicative of standard operation but it does demonstrate the potential effects. As a result of this test, you would be hard pressed to find someone that would argue that 180 degrees of inter-channel phase distortion is acceptable, but where between the two extremes is the threshold of audibility? Tom Holman reports [10] that in his laboratory environment at the University of Southern California that is dominated by direct sound, a channel-to-channel time offset equal to one sample period at 48 kHz is audible. This equates to 20 μsec of inter-channel phase distortion across the entire audio band. Holman [10] also mentions, “one just noticeable difference in image shift between left and right ear inputs is 10 μsec”.
7. Intra-Channel Phase Distortion
Recall that we use the differences in signal amplitude and phase to localize or determine the source of sound and relatively small amounts of inter-channel phase distortion can be audible. But how does our hearing react when each channel in a multi-channel system is subjected to non-linear phase response but the phase response is matched between all channels? Douglas Preis [11] did an extensive survey of existing literature and Tom Holman's [10] experiences and research through his work at USC gives us an interesting insight into this phenomenon. Both report that the threshold of audibility is frequency dependent, which correlates with all other audibility thresholds. In laboratory environments when using test tones and headphones, research has shown that the human ear is sensitive to intra-channel phase differences of 0.25 msec [8] or +/-0.5 msec [9] in the mid-range with the threshold increasing at higher and lower frequencies. Preis states “the tolerances shown.... are not directly applicable to speech or music signals irradiated by loudspeakers in a reverberant environment. Most likely, the perceptual thresholds for these conditions would be at more than twice those shown”. Essentially, the data suggests that for high quality music or speech reproduction in a reverberant environment intra-channel phase distortion of 1 msec is inaudible to a trained listener. Notice that this threshold is a relatively conservative statement and is still two orders of magnitude greater than that for inter-channel phase distortion.....”
The above is just an example. There is more.
Best Regards,
Bohdan