- Head related transfer function individualization for hearing device
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- US20110103590A1 - Audio system phase equalization - Google Patents
From four feet away? How about experiencing a voice chorale, on stage from 20 feet? Across weeks of rehearsals? I know this is what a speaker designer needs to do. Those are not common experiences for anyone, let alone to have spent hundreds of hours in studios learning what the mic hears and how the studio must alter its sound, so that we think it is a pleasing, and b realistic. Golix, You say "Also a speaker which has the drivers on a vertical axis can only be phase coherent at one point in space. This of course is a purely geometrical problem independent of any electrical feature.
And in a two-way speaker, that point can be aligned to anywhere via tilting the speaker and with a small change in your ear height for the final touch. For a three way, one has to move the top two drivers relative to the woofer. That adjustability has become a standard feature in all our three-way designs, and for several new models coming into production over the next many weeks.
We call this "adjusting the Soundfield Convergence tm ". Golix, please do reconsider your notion that "Basically a phase coherent speaker is one that is not only [coherent] in time but also in phase; a time coherent speaker is one that's [coherent] in time but not in phase.
Head related transfer function individualization for hearing device
All the math, all the physics supports that. And we speaker designers get to screw that up! Nowhere in the recording chain, nor in the playback chain is the timing split between highs and lows, or the polarity, or both. One can learn to recognize that, the way an amplifier designer can hear if someone's amplifier needs a bigger transformer- it's a unique sound distortion. Thus, Golix, please understand that while those Tannoys are indeed smoothly phase coherent, they are not time coherent. A step function would show this: Perfection in a step function looks like the plus-half of a single square wave, rising up quickly, leveling off and then going on forever- like a single stair step.
There would be no ringing or rounding over at the initial corner, and the top would stay level forever, never returning to zero. In that Tannoy, the first energy to arrive from that step-input is upwards-going, as it should be. A moment later, the late-arriving, inverted-polarity tweeter shoves sucks that initial positive air-pressure-increase down into the negative-air-pressure portion of the graph. The tweeter's dome then returns to rest from its full "-" excursion, because the crossover cannot pass the "DC" to tell it to "hold your position, albeit sucked in".
The air pressure then returns to the positive from the midrange tones' positive-pressure continuing to arrive.
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Finally, no speaker ever then "levels off" and holds that air-pressure "positive" interminably, because the room leaks that pressure away. So the step droops back to zero, even though you see the woofer still shoved "out". With regard to our measurements- we have those measurements supplied by the driver manufacturers, such as MorelUSA, taken in their chambers. But those measurements are usually taken in a half-anechoic chamber.
Can anyone say that is a realistic test of the direct sound from a tweeter? You will see a picture come up of a woofer mounted in this fully-reflective wall of that test chamber- their tweeters are tested in that same position. For our own testing, we are still in the analog days here, not for want of trying to go digital, so I cannot show you hard copy.
I will be working to present this information on our website as we continue to grow. Testing indoors, in an average room, is fine for looking what happens from Hz on up.
There are also certain ways to combine very nearfield measurements, that I must decline to describe, which obviate the need for a chamber. I do know that it is easier to perform many more misleading tests in the digital domain. One has to scrutinize for many problems, with very specific measurements, either by analog or digital means. There is no one, or two, or three measurements, or even "dozens", that tell anyone how a speaker will actually sound. In particular, we look at the moment of first arrival of a burst of cycles of a single sine-wave tone, taken all the way up the frequency scale.
The Tannoys would show a left-to-right motion of that starting point which is the time delay creeping in in the crossover range, and then the tone burst smoothly flips upside down in the tweeter range. Ours stand still from Hz to 8kHz, and always have the same polarity. I can hear when the focus becomes as sharp in that crossover range as it is away from that range.
So has every person for whom I have demo'd this. The audible change from moving that mid back and forth, even an eighth of an inch, cannot be explained by wave-cancellation math, nor can it be explained by any change in the cabinet-face or wall-surface reflections in my designs. We hear the difference as a loss of sharpness, or definition, of a sound's location from front-to-rear.
Depth is time delay, and the sharpness of the image begins at its front-most element the singer's mouth. If that initial location is smeared from front-to-rear, then the depth "behind" that voice is also smeared over by that initial information, and the depth itself is also smeared in time. This is all information audible by WHEN it arrives. If that initial location is smeared, we also hear a loss of attack, which is a leading-edge phenomenon- another time-domain aberration. There exist many more ways the ears can guide time-domain measurements, and vice versa.
It can be heard however, as an overall clarity of the top end, because there are a lot of frequencies nearby that 8kHz- notably the ones all the way down to 4kHz- only one octave, one "undertone" away. They should have begun after the stick hit and then was removed from the metal body, right? Yet, the timing can be warped just enough so that one hears the stick-hit occurring AFTER the tones start. Now that is an un-natural sound anyone can identify! And the time-delay from this small offset of that supertweeter?
Millionths of a second. The same thing happens when judging the firmness of the felt on a mallet on a tympani or vibraphone.
Or no felt at all- just the sound of hard wood, or a large-diameter mallet head or a small one. They each make their own sound, which a time-coherent speaker easily reveals, even in the midst of an entire orchestra reaching its crescendo around them. In the usual mid-to-tweeter crossover range, achieving precise focus lets us hear exactly when the singer's tongue leaves the roof of her mouth- important to her shaping that note.
Or to the definition of any other instrument that requires half-mid and half-tweeter, such as tambourine, trumpet, guitar, piano Then include the distinctive sound of each one's ambience directly behind those events- there is much to listen for, that leads to more musicianship being heard. Also, it is possible for nearfield, tweeter-only measurements to have a standing wave build up between the microphone and the tweeter's dome, on sinewave tones, which totally fouls up anything we are trying to measure. This is somewhat related to the how the notion of first-order speakers having comb-filtering effects comes about- from applying the math, and measuring, with specific single tones.
Which do not occur in music, especially if that particular frequency lies between the tones of the musical scale. I do agree with all of what Karls goes on to say in his post right above, including his analysis of lobing. However, I find lobing is exaggerated when the cabinet-face, or even the area right around the tweeter, is contributing many reflections. Comb filtering, from simple, "fewest possible drivers", first-order speakers, is not apparent to me, or at least objectionable on music.
We have done that, and found no significant differences that we can say were from the comb-filtering effects that must indeed arise from having two drivers producing the same range. Multiple drivers in the same tone range present a lot of different frequencies to cancel out, because those six tweeters, for example, each arrive later than the one nearest your ear. That leads directly to lobing, which is a frequency-dense form of comb filtering. What you want to call it depends on how you measure it. Stereophile thoroughly tested our original Diamante model in April , and showed how its step response aligned quite well between mid and tweeter as the microphone was moved down to their time-coherent axis.
JA was really nice to us by also showing how the corresponding step response also changed for the better as that time-coherence was achieved. He then showed how the overall phase response measured, which was pretty good. Ten years later, our deviation from zero is far less. Also, the Diamante tweeter's tone balance on that "best" axis was not flat for him, because at 50" away, the tweeter was well above the mic the mic was far off the tweeter's axis.
Not even close to being a step at all The Diamante review is not archived on-line, unfortunately. Maybe those measurements are- I have not searched for those in JA's database he graciously offers. Returning for a moment to the use of tone-bursts: One can also look at the envelope shape of each burst, from which many things can be seen, such as cones and cabinets flexing. That is something that can be seen at the beginning of the envelope, as the output failed to reach full height on the very first cycles.
The cone flexing absorbed that energy, only to give it back later. Of course the cone could be highly damped, then it never gives it back as audible sound, but just leaves the initial dynamic-rise blunted. Too "laid back" you would hear. Think about the dynamic response heard from soft plastic cones One can see a returning echo from inside the cabinet, after the end of the pulse, which can be fixed.
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A flex in a cabinet wall can be detected, and that can be stiffened. A reflection off the cabinet-face can be seen, and that can be absorbed or avoided. One does not need an anechoic chamber to perform those tests. There is digital hope for us: This Summer, I look forward to working with Agilent Technologies in developing a system that will do what we need. A few years ago, the computing power was also not available for certain tests I have always wanted to make in the digital domain. Now it looks like it is. It is therefore true to say that its properties are closer to random signals.
It follows that the distortion of a complex audio signal creates a complex noise level, which masks and disturbs the low levels of the original signal. Practical tests have proven that we can hear a 1Khz sine wave tone with 0dB level with white noise at levels of to dB. What this demonstrates is that a high distortion system will completely mask low level signals, i. The color of sound All-specific designed KV2 Audio transducers and components exhibit extremely low distortion, below 0.
As previously stated, live music has the capability of producing a dynamic range in excess of dB. To reproduce this through an audio system with a suitable degree of headroom, a dynamic range capability of around dB is required. Secondly, while a 96kHz sampling rate has been deemed adequate when professionally converting an audio signal consisting solely of harmonic signal components, analog audio signals have complex harmonics and overtones and therefore should be regarded as random signals. The spectrum of random signals is infinitely wide, so when converting analog signals to digital, the sampling rate must be as high as possible in order to maintain quality of the transferred signal in full resolution.
At KV2 we undertook a different approach to digital to overcome the inherent problems in existing systems.
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The amplitude of the analog waveform is represented by the density of pulses and is called Pulse Density Modulation PDM. The resulting digital bit stream is encoded at an enormous 2,, samples per second! A special step compander circuit adds a further 20dB of dynamic range to utilise the maximum range of the converter at low levels. This best of both worlds approach provides unmatched dynamic range and audio reproduction. To maintain a high-resolution audio signal, it is also important for the system to maintain the shortest possible impulse response time. Impulse response time is affected by the settling time and circuit design in analog electronics.
The distortion, created by slow settling times is not commonly discussed by many manufacturers as they fail to understand its significance, often overlooking it in providing the technical specifications of products. Sampling frequency is the major determining factor of impulse response in the digital domain. In Figure B below it is evident that commonly used commercial systems, particularly digital, cannot pass the full resolution of the original signal.
Impulse response time is affected in the digital domain by the sampling rate and in the analog signal path by the speed of the electronics settling time and control of the acoustic component motion speaker movement. The change in the original signal caused through poor impulse response creates distortion.
Systems with a long impulse response time are unable to transfer high dynamics and high definition signals. KV2 Audio design amplifiers from the ground up for specific applications. This approach allows us to employ and refine the perfect types of power required for accurately reproducing highs, mids and bass frequencies. Low frequency devices have a unique set of requirements.
Woofers are large, heavy and difficult to keep under control. Simply put, phase shift is when current does not follow voltage as power flows through a voice coil. If you are sending 1,Circuitry2 watts volts and 10 amps coming out of the amplifier under phase shift conditions, you may be required to produce double the amps at half the voltage in order to keep the woofer under control. The design features a switching voltage power supply that keeps the voltage across the output devices low, but capable of providing much higher current and better damping characteristics than standard Class H designs.
For sound quality reasons in mid range and high frequency reproduction we use amplifier topologies based on Class A or Class AB. The warmth and clarity provided by this type of amplifier is ideal. Our design uses Mosfet output devices in a push-pull, transformer balanced amplifier featuring a fast recovery time. One of the most important parameters in transducer design for Super Live Audio Systems, is the removal of unwanted resonances. These resonances are usually caused by the mechanical design of the speaker and its failure to control the diaphragm motions.
Resonances reduce overall definition by masking smaller signals and producing tones not related to the original signal. Figure C below shows an original sine signal red, top with its sharply defined end and the same reproduced signal blue, bottom , still oscillating after the signal stops due to poor control of speaker mass. Poor pulse response has a very negative effect on the ability of a speaker to reject feedback. Every loudspeaker used in a KV2 Audio system is specifically designed. This leads to the development of components that become the ultimate solution for their given application, not just an off the shelf driver.
One of the most challenging projects undertaken by the team was the development of our new NVPD range of compression drivers. The idea came during an Italian lunch, where we discussed a new nitrate coating used in Formula One racing, offering extreme strength and rigidity. Extremely light, it is great for cars but had never been tried in pro audio. By adding some of the largest Neodymium motors available today and our advanced phase plug design the result was a range of world beating high frequency units that produce distortion of less than 0.
SLA systems feature exceptional feedback rejection and this in part is due to their excellent pulse response. Additionally, control over the speaker mass can be very positively impacted by using an active impedance control, trans-coil speaker system. This system utilizes a secondary stationary coil, which reduces inductance close to zero and dramatically improves pulse response. Inductance is the main reason for odd harmonic distortion. Odd harmonic distortion is far more audible than even harmonic distortion.
Figure D below shows the effects of AIC. The Active Impedance Control or AIC is an additional fixed, multi turn coil positioned in the loudspeaker magnetic circuit gap. This coil is almost as long as the gap height and is wound around the pole piece to be very close to the primary voice coil. A current flowing into this coil generates a magnetic field that is in opposition to the field generated by the moving coil. This cancels out most of the voice coil inductance and reduces the flux modulation and inductance modulation. The two AIC terminals allow driving the additional coil in many different ways according to specific application needs.
There are two main types of sound system designs that have been prominent in the market, consisting of single point source or multiple point source concepts. Multi point source arose from the requirements for very high output power. The idea satisfied that criteria, but with the increasing number of sound sources came an overall reduction in the quality of the sound.
The two big disadvantages of multipoint source systems were the suppression of the high frequency output and the physically time-shifted outputs from the individual speakers. Adding a number of time-shifted outputs from individual speakers together causes poor system impulse response.
US20110103590A1 - Audio system phase equalization - Google Patents
The first types of multipoint sources were simply a large pile of cabinets, stacked together like building blocks and intended to array on all axis. A major improvement in the next generation of systems was the introduction of multipoint, one-axis systems that provided better frequency response and increased definition than previous multi axes systems. Unfortunately, whilst a step forward, the frequency response and impulse responses were still not ideal and the coverage was often inconsistent.
A typical representation of the one axis multipoint source sound system used commonly today is a line array system. Line array does reduce the effect of multipoint sources interfering with each other like the systems of twenty-five years ago, but it is still a long way from the superior results achievable with single point sources. A single point source sound system offers the highest possible definition and dynamic range available today.