@ 23:20 interesting how due to the phaseshift there is a 2nd order distortion to the laser measured output at the begining cycles. In this tone output it settles down to low distortion in a few cycles I see but with music signal that is not a constant tone it would be more or less all the time distortion and imd. Amplifier damping factor would influence this to an extent I think, high damping = quick recovery but larger amplitude distortion, low damping = slow recovery but less distortion of amplitude. Very interesting topic, thank you so much for making this higly educational video and thank Filipo @ B&C.
Hi, how wuld you value adaptive dereverberation providing about 10...15 dB of spk->mic path attenuation? provided Spk & Mic position fixed, lips are > 15 cm from the mic (otherwise back path varies a lot).
This is normal since the light is faster than sound ;) Assuming the electronics measuring this do not add any/same phase shift or time delay. Well done!
Audio engineers know that acoustic instruments take time to create sound as the resonant volumes are not capable of immediate exitation. This is post is academic at best. Our ears (face, nose, head shape) also alter phase dramatically. Electrical integrity does not necessarily mean better sound. Add a port or passive radiator and things get much uglier. Thank you for the easy to understand description of what is happening. Now the real work starts to quantify audibility.
Please tell me how to vibrate specific body parts/organs using directional audio. Please tell me how to make someone's left ear only, hear a high pitched tone directionally.
sorry to be negative here but that grafic at 1:56 is an abomination. I came into the video interested because I like science about sound, but seeing that I couldn't help but think "I kinda doubt this guy knows his shit"... :/
To explain if somebody doesn't understand, if you know anything about how high pass filters (which is what an AC coupling circuit is) work, you know that what they do is they "pull down" the voltage if it stays at a fixed value for too long, effectively preventing slow oscillations or constant voltages (DC) from happening. There are two things very wrong with the grafic: First, the AC coupling pulls down the DC voltage faster than the AC oscillation before is swinging. If the filter was that aggressive, it would have an impact on that AC voltage as well. Kinda nitpicky, and yes it's only for demonstration, but if you do it you might as well do it right. And second, and that makes absolutely no sense, when "the DC is over" the voltage just goes back on high, as if the filter knew there was an AC coming now. How is that supposed to happen? The filter can't add voltage to the signal (unless it would be resonating but you don't do that with a coupler). What would happen is the output signal just stays low until there is a change again in the incoming signal, meaning until it goes below 0V. Again, all the coupler does is prevent a voltage (besides 0V) from being constant too long. TLDR: The grafic looks as if someone just went "yeah AC coupling takes the DC and removes it" which, yeah is kinda what it does but just not at all in the way it is shown in the grafic.
This is an interesting philosophy, and well explained, thanks! Could you replicate the same results by tuning with software, then apply your standard EQ curve to the result get your prefered tuning precisely?
I was interested in this topic as a young man in the 90s and went to libraries reading all books about speaker building. In english (I am from austria) You explained everything there is to know in one video.. Simply amazing!
Hi very interesting video for the most part above my ability to understand Anyway i have a question I wonder what would happen with a diode placed in series with a woofer What would the sound be like ? This should prevent the woofer cone to move backward ... only from the zero point forward (generating a positive pressure) I ask this because if we measure the sound pressure during a concert at the listening spot the pressure will vary only above zero Never negative i guess Instead a cone when moves backwards will cause a negative pressure towards the listening spot This could be unnatural ? The goal is always to reproduce the reality
19:15... ".... at higher and higher frequencies. It doesn’t seem to be a set amount of time, << otherwise we’d see a continued downward phase response on the graph with higher frequencies>>, but it levels out at around the minus 180 degree mark or so. Basically, above 200Hz, frequencies are going to be about a half-cycle behind in their full steady-state formation." >> To keep your phase plot 'on screen' - your oscilloscope software adds 360 degrees to the (indicated) phase data. This will always be the case no matter what phenomenon you are probing. The phase in your graph is in fact continuous laging well past 180 degrees >> it does not level out. Every apparent vertical jump in the plot indicates a 360 degree instrument scale change. Check out 600 Hz - your instrument is indicating a 360 degree phase shift. Good luck with your endeavors..... If something doesn't look as expected - you have either discovered something new or some peer review might help you to learn something new. >> Keep up your inquisitive nature.
very nice video, but I have some corrections and contributions 1) 6:20, while it is true DC resistance of voice coils, do vary with frequency, it is a very little change in comparison with reactance, which varies greatly. The text should say "impedance varies with frequency", which is not the same of resistance. But all of these varies with frequency, resistance, reactance and impedance overall. (ideally resistors have a flat frequency response, but complex AC circuits like speakers are very complicated and do exhibit non flat resistance component) 2) 3:34, I think what you tried to explain here is right, but not correctly explained. The dB scale for power is calculated as P (dB) = 10 * log10(P/Pref) While dB scale for pressure and voltage is calculated as V (dB) = 20 * log10(V/Vref) These formulae means that doubling the power is to add +3dB of power, but doubling the current, voltage or presure is adding +6dB of the quantity. That said, what really happens when you connect a second speaker in parallel with the first, while remaining a constant signal or voltage output, it is true the amplifier delivers +3dB of power, because it is doubling its power output, its impedance load is divided by two, thus power is multiplied by two, because of P = V²/Z . Where V remains the same and Z gets divided by two. Doubling the power is adding +3dB of power. Say for example if your first spkr is developing 1 Watt, the second spkr you connect will develop the same 1 Watt, while the power of the first one remains unaltered. Being the total power delivered by the amp just the double of just one speaker connected. But with SPL pressure is a different story. While signal/voltage V remains the same, the power of the first spkr will remain the same, and the second will develop the same power of the first. You have +6dB of SPL pressure output, because the pressure is multiplied by two , these pressures of every spkr is just added. and the dB scale for pressure is different that the scale used for power. Double the sound pressure (say in Pa units) will result in adding +6dB to the SPL. So why you have +3dB power in the amplifier and +6dB pressure at the spkrs?, why this 3dB difference? Because doubling the amount of speakers raise +3dB of the sensitivity of the system. Say for example one spkr has a sensitivity of 95 dB 1Watt/1meter, two speakers have 98dB 1Watt/1Meter. But this is far more complex because there are issues like vector sum, phase coherence, wavefront shape, polar response and many things that make this very complicated in practice.
Nice ok so: adding a second speaker in parallel but sharing 1W of power raises the total sensitivity by 3dB as measured by the microphone. So then, what if we also double the total power to 2W? The mic measurement goes up another 3dB, for a total of +6dB. If we double the power going to only one speaker, the mic reading only goes up 3dB. So what is the mic measuring?
@@devinlsheets_alphasound "adding a second speaker in parallel but sharing 1W of power raises the total sensitivity by 3dB as measured by the microphone. So then, what if we also double the total power to 2W? The mic measurement goes up another 3dB, for a total of +6dB." Yes, correct. " If we double the power going to only one speaker, the mic reading only goes up 3dB. So what is the mic measuring?" Assuming you disconnected the second spkr, and have only one, if you double the power (+3dB of power), you get +3dB of pressure. while doing this you multiply speaker voltage by 1.4142 (square root of two) the mic measures pressure, voltage reading of microphone terminals are proportional to pressure waves at the mic capsule, say for example a calibrated measurement mic could say its sensitivity is 1 Pa = 94dB SPL = 8.15 mV RMS. Pressure reference for the dB SPL scale is 20 uPa (twenty micro pascals) as in formula dB SPL = 20*log(P/Pref) Because power is proportional to the square of either voltage or current, the 10*log formula was chosen for power, and 20*log for voltage, current, and pressure that is proportional to spkr voltage. That way when you add +3 dB or +whatever dB to power, you get equal +dB to voltage or pressure. that means, adding +3dB of power (doubling power), you get +3dB at SPL pressure (pressure/voltage is multiplied by 1.4142 square root of two) Adding +6dB of power (multiplying power by four), you get +6dB at SPL pressure (pressure/voltage is mutiplied by two) Or in other words, if you multiply voltage by two, power will be multiplied by four, because power is proportional to square of voltage.
You should do a laplace transform of the pulse response of the speaker input to laser output. This will give you the transfer function, and you have all the ansswers you need.
hi, not related to the subject , but i had a question and you might know much better , would a digital to analog conversion be better before , or after amplification ?
