Members OneEng Posted December 31, 2010 Members Share Posted December 31, 2010 That is a close call for me at the higher SPLs. Of course this depends on the box and driver. But then again, depending on the sub's design, those can be on the high end as well. I would be concerned that the sub would prove "muddy" compared to the 12" in the 90-120Hz range. Link to comment Share on other sites More sharing options...
Members Mutha Goose Posted December 31, 2010 Members Share Posted December 31, 2010 Really! I am surprised.Considering the vast array of DSP's available today, I would think that it would be a negligible cost (I know, no such thing as negligible in the consumer market) in monitoring the output and there would be a considerable value to doing so (both for diagnostics and control algorithms). In order to sample the output, you must have a scaled down version that is within the range of the DSP and converters. This means putting a small series precision element in the output. Depending on the load variations (in manufacturing transducers will vary widely in many charactoristics) and the dynamic range of your D/A converters (DSPs with integrated DA of usable range drives up the cost, as does external converters, greater the dynamic range, the more the $ in component costs) you may need some real precision in those output elements (i.e. driving up component costs). But those elements are also in some very hi current environments, so they must be able to withstand the dissipation while maintaining tight tolerances, again increasing cost and complexity. I am only scratching the surface, and already the component cost of the pcb assemblies are increasing by more than 50% and Ive introduced failure mechanisms that could render the system inoperable. The devil is always in the details Link to comment Share on other sites More sharing options...
Members OneEng Posted December 31, 2010 Members Share Posted December 31, 2010 In order to sample the output, you must have a scaled down version that is within the range of the DSP and converters. This means putting a small series precision element in the output. Depending on the load variations (in manufacturing transducers will vary widely in many charactoristics) and the dynamic range of your D/A converters (DSPs with integrated DA of usable range drives up the cost, as does external converters, greater the dynamic range, the more the $ in component costs) you may need some real precision in those output elements (i.e. driving up component costs). But those elements are also in some very hi current environments, so they must be able to withstand the dissipation while maintaining tight tolerances, again increasing cost and complexity. I am only scratching the surface, and already the component cost of the pcb assemblies are increasing by more than 50% and Ive introduced failure mechanisms that could render the system inoperable. The devil is always in the details Still not with you. You make a simple circuit consisting of a voltage bridge (2 small resistors) and a diode clamp just before the A/D. You size the resistors such that the inrush current needed by the A/D is satisfied, and such that the full range output of your signal mates up with the full range input of the A/D. The whole circuit costs pennies and takes a few mm^2 on the board. Most DSP chips have ample A/D inputs so there is likely plenty of A/D inputs to do this with. Link to comment Share on other sites More sharing options...
Members Mutha Goose Posted December 31, 2010 Members Share Posted December 31, 2010 Still not with you. You make a simple circuit consisting of a voltage bridge (2 small resistors) and a diode clamp just before the A/D. You size the resistors such that the inrush current needed by the A/D is satisfied, and such that the full range output of your signal mates up with the full range input of the A/D. The whole circuit costs pennies and takes a few mm^2 on the board. Most DSP chips have ample A/D inputs so there is likely plenty of A/D inputs to do this with. You are over simplifying. This all adds cost (more than just pennies) and and it adds complexity. Versus using rudmentary diagnostics; knowing that if they are good, it is safe to assume the outputs are working as designed (as expected). And this is the case the DSP algorythyms are designed to. They don't need to sample the output. And I can have all of this for a couple of traces and a few lines of code with no extra cost nor complexity and yet I still achieve my target. But this is making minor assumptions which would be the design trade off. Again, I'm not saying it isn't sometimes desireable to reduce the assumptions (and therefore improve accuracy), but that is the exception, not the rule. Link to comment Share on other sites More sharing options...
Members agedhorse Posted January 1, 2011 Members Share Posted January 1, 2011 To clear a few things up... DSP for audio applications is a digital emulation of traditional analog transfer functions like crossover filters, eq filters, comperssion/limiting, delay and of course routing and metering. There is no need to monitor the output (other than for a failure) because by the very definition the output has a precisely defined (and caluulated) relationship to the input. There is generally no need to add additional features that would complicate things for the average user. In fact, the elegance of the DSP solution is the complete transparency to the user of what's going on under the hood and in fact most manufacturers do not want the user to have any access to the "secret sauce" because the manufacturer's engineers are MUCH better equipped to generate the necessary algorithems based on their internal propriatary data on the parts (drivers) they use. In a few very high end powered products, there is some access to programming parameters. In some cases there are presets that adjust internal eq and level balances to adjuste for non-linear coupling of cabinets as a line array gets long, and for remote diagnostics (feedback of operational data like temperature, driver impedance, protection status) but not for the powered speaker products that would typically be used here. This subject is something that I have spent a lot of R&D time on over the last 10 years, it's the way of the future once folks stop worrying about how damn cheap they can get something for only to find that by the time it's all assembled it's a lot more money than the "cheap" money they thought it was going to be. The PRX-612 should do just fine crossed over at 90Hz using a LR 4th order filter. 90Hz is the -6dB point for constant power summing. The reason for JBL recommending an external crossover for the system if the PRX sub is not used is because somehow you need to define accurately what is sent to the sub and the top and it needs to sum relatively flat at the crossover point. I would recommend sticking with the PRX sub, together the SYSTEM sounds very good... better than 99.9% of so called experts could program if given a week with the system... right out of the box. Link to comment Share on other sites More sharing options...
Members Rumorsband Posted January 1, 2011 Author Members Share Posted January 1, 2011 DSP - "Digital Signal Processing". Inside the speaker is a small computer which does real time analyzation of the incoming signal as well as monitoring the output signal from the main amplification. Using computing algorithms, the output is determined based on the current reading of the input, the rate of change of the input from the last time sample, and the rate of change of the rate of change (sorry, engineer talk).Essentially, it is like a car "drive by wire" example. In newer vehicles, the throttle doesn't actually go to the throttle, it goes to the computer. The computer then uses a small stepper motor to adjust the throttle for you.I am not sure what you are saying about the cross-over. Does it make the sound better or worse? There is more tonal clarity and definition ...so yes it does sound a bit better, but it sounds pretty good without it also. I was saying I dont seem to get the same output when using the xover as I do when I dont. Do xovers use up power from the speakers? Someone told me once ...anything you add to a system takes up some power...Eq's etc> Oh before I forget ... Thanks to all (I started this thread and I'm glad I did) and Happy, Healthy, Prosperous 2011 Link to comment Share on other sites More sharing options...
Members OneEng Posted January 1, 2011 Members Share Posted January 1, 2011 There is more tonal clarity and definition ...so yes it does sound a bit better, but it sounds pretty good without it also. I was saying I dont seem to get the same output when using the xover as I do when I dont. Do xovers use up power from the speakers? Someone told me once ...anything you add to a system takes up some power...Eq's etc> Oh before I forget ... Thanks to all (I started this thread and I'm glad I did) and Happy, Healthy, Prosperous 2011 Without the xover, your going to get a clash between the top and bottom where the frequencies are both being produced. I have never done this intentionnaly, so I don't know what it sounds like (perhaps agedhorse or someone else could enlighten us ), but it is supposedly not good. The xover doesn't take power away from the speaker unless you set it up to do so. You will need to set-up your gain structure to get your system ballanced as follows (my method anyway): With this setup, you achieve maximum pink noise power at your speakers at the same point where your mixer output is at its maximum. With this setup, you will never be clipping your speakers or any part of your signal chain; however, you may want to raise your speaker gain a little higher since this setup leaves a little headroom on the table by using "pink" noise (which is equal amounts of all frequencies) .... ie an unreal situation when you are playing live music. Could you provide the make and model of your cross-over? Link to comment Share on other sites More sharing options...
Members OneEng Posted January 1, 2011 Members Share Posted January 1, 2011 I would recommend sticking with the PRX sub, together the SYSTEM sounds very good... better than 99.9% of so called experts could program if given a week with the system... right out of the box. Can't see any way to dispute that; however, if Rumosrband wants to keep his existing subs and save some money, a little tweaking will be necessary. I don't see it as a real big deal to have a 1U xover in the instrument rack. It might be a little less cables to simply have only left and right out vs left and right top and left and right bottom, but that seems like a fairly trivial issue to me. Link to comment Share on other sites More sharing options...
Members Mutha Goose Posted January 1, 2011 Members Share Posted January 1, 2011 To clear a few things up...DSP for audio applications is a digital emulation of traditional analog transfer functions like crossover filters, eq filters, comperssion/limiting, delay and of course routing and metering. There is no need to monitor the output (other than for a failure) because by the very definition the output has a precisely defined (and caluulated) relationship to the input.There is generally no need to add additional features that would complicate things for the average user. In fact, the elegance of the DSP solution is the complete transparency to the user of what's going on under the hood and in fact most manufacturers do not want the user to have any access to the "secret sauce" because the manufacturer's engineers are MUCH better equipped to generate the necessary algorithems based on their internal propriatary data on the parts (drivers) they use. In a few very high end powered products, there is some access to programming parameters. In some cases there are presets that adjust internal eq and level balances to adjuste for non-linear coupling of cabinets as a line array gets long, and for remote diagnostics (feedback of operational data like temperature, driver impedance, protection status) but not for the powered speaker products that would typically be used here.This subject is something that I have spent a lot of R&D time on over the last 10 years, it's the way of the future once folks stop worrying about how damn cheap they can get something for only to find that by the time it's all assembled it's a lot more money than the "cheap" money they thought it was going to be.The PRX-612 should do just fine crossed over at 90Hz using a LR 4th order filter. 90Hz is the -6dB point for constant power summing.The reason for JBL recommending an external crossover for the system if the PRX sub is not used is because somehow you need to define accurately what is sent to the sub and the top and it needs to sum relatively flat at the crossover point. I would recommend sticking with the PRX sub, together the SYSTEM sounds very good... better than 99.9% of so called experts could program if given a week with the system... right out of the box. That's what I said. :-) And FWIW: this is what I do for a day job (and have for the past 17 years). I design DSP-based amplifiers for the automotive industry and have assited in the design of a few speciallized DSPs used as the core in RF multi-path reduction (patents awarded to the upper-level design team). Link to comment Share on other sites More sharing options...
Members Rumorsband Posted January 2, 2011 Author Members Share Posted January 2, 2011 In a temperature controlled lab under on set of conditions (ie beginning of life for the components) that may be true; however, even in this instance, you are a slave to manufacturing inconsistencies. The specific component in any given unit may vary considerably from the "lab" test unit in its properties. This is getting to be more and more of an issue since everyone wants cheaper and cheaper components. Quality and consistency costs money to hold the manufacturing process and materials to higher consistency levels. I work in automotive, so I know all about manufacturing costs and how to keep them low by using adaptive controls to compensate for cheap inconsistent components. It is all about cost and this is a very good way to control costs. Perhaps things are different in speaker production? Then there is environmental condition changes to take into account. You can either monitor all the environmental conditions, or monitor their effect on the output. As it turns out, it is cheaper to just monitor the output and adapt your algorithm to adjust for any changes. In vehicles, these adjustments happen over sensor life, and differing atmospheric conditions all the time. Limits are set for how far adjustments can be made before sensors need to be replaced. This is where that "check engine" light comes in which lets the vehicle operator know that the vehicle needs to be taken in for fixes Having watched the automotive industry progress from vacuum carbs to sophisticated adaptive controls, I can see speakers being a good fit for the same type of evolution. Can't see any way to dispute that; however, if Rumosrband wants to keep his existing subs and save some money, a little tweaking will be necessary. I don't see it as a real big deal to have a 1U xover in the instrument rack. It might be a little less cables to simply have only left and right out vs left and right top and left and right bottom, but that seems like a fairly trivial issue to me. I intend to buy the prx sub...it is on back order, and I dont want to buy a xover for a couple of weeks. Also I'd like to sell my york system if i could. Had all these people interested but now that I'm ready nobody has any money. Link to comment Share on other sites More sharing options...
Members OneEng Posted January 2, 2011 Members Share Posted January 2, 2011 I intend to buy the prx sub...it is on back order, and I dont want to buy a xover for a couple of weeks. Also I'd like to sell my york system if i could. Had all these people interested but now that I'm ready nobody has any money. It always seems to be that way Link to comment Share on other sites More sharing options...
Members agedhorse Posted January 2, 2011 Members Share Posted January 2, 2011 In a temperature controlled lab under on set of conditions (ie beginning of life for the components) that may be true; however, even in this instance, you are a slave to manufacturing inconsistencies. The specific component in any given unit may vary considerably from the "lab" test unit in its properties. This is getting to be more and more of an issue since everyone wants cheaper and cheaper components. Quality and consistency costs money to hold the manufacturing process and materials to higher consistency levels. I work in automotive, so I know all about manufacturing costs and how to keep them low by using adaptive controls to compensate for cheap inconsistent components. It is all about cost and this is a very good way to control costs. Perhaps things are different in speaker production? Then there is environmental condition changes to take into account. You can either monitor all the environmental conditions, or monitor their effect on the output. As it turns out, it is cheaper to just monitor the output and adapt your algorithm to adjust for any changes. In vehicles, these adjustments happen over sensor life, and differing atmospheric conditions all the time. Limits are set for how far adjustments can be made before sensors need to be replaced. This is where that "check engine" light comes in which lets the vehicle operator know that the vehicle needs to be taken in for fixes Having watched the automotive industry progress from vacuum carbs to sophisticated adaptive controls, I can see speakers being a good fit for the same type of evolution. Completely different application, a DSP based audio processor is an open loop processing device. There are no sensors, no need for sensors, no feedback, nothing like that. It's purely (for this discussion) an analog input, an A-D conversion with whatever scaling is necessary, a series of computations on the digital data to accomplish the transfer functions, rescaling and a D-A conversion to an analog output. Metering is (more or less) a basic subroutine function, scanning and displaying the various metering points and levels (like multiplexing) in the digital domain and then displaying it in analog format. The only practical deviation from this is servo feedback for artificial woofer damping, but this is becoming rare as the benefit is minimal but when things go wrong they go VERY wrong. Conceptually it's very simple though what is happening is complex in detail. Automotive (and aero.) applications take each subsystem, perform calculations to data based on an sensor data value and a control data value, perform a calculation then output that data to an actuator which causes a corresponding change to the sensor data which will recalculate with the data value and output a new sensor data value etc, as the system "settles" to a steady state but as soon as either the sensor or the control data changes then new calculations result. Part of the reson for this feedback is for calibration (intolerance to variations), part for dynamic considerations (integration or differention in the transfer function) and part is for generating a confidence/no confidence tally so that failures in the control system are alarmed or flagged. My engineering specialty was analog industrial power systems and controls. I am horribly rusty with the rather involved math behind such systems these days but I am very familiar with how the basic processes work and why. One of the three companies owned by our parent company was an aerospace company, and a couple of the engineers split their time between the two industries. They were very comfortable with my product developement appraoch based on a thorough understanding of system control theory and especially the stability part. He who does not make peace with Nyquist's stability theories is doomed to failure before he starts a project. Link to comment Share on other sites More sharing options...
Members OneEng Posted January 2, 2011 Members Share Posted January 2, 2011 He who does not make peace with Nyquist's stability theories is doomed to failure before he starts a project. LOL. Engineer humor I realize there are no sensors in a speaker, although that might be a neat sandbox project to do sometime. In an ideal situation, you could compare the input signal compared to an RTA signal in real time to determine if you were achieving your sound reinforcement design goals while making real time changes to ensure that you were. Since this is not possible (within a reasonable cost budget and without an external condenser mic positioned out in front of the speaker system), the next best thing is to measure the output voltage from the amplifier. The fundamental assumption in this type of measurement is that one could assume a certian output from the speaker given a certain output from the amplifier. You are going back one step further and saying that you need only monitor the input signal in order to determine what your transfer function is going to do to the output of the speaker. I simply point out that your assumption is filled with more inaccuracies than mine. I further contend "why not"? I see no reason not to measure the output of the amplifier and use it as feedback. In any control system, it is always a good idea to measure the output if you can. In the case of a speaker, you can .... so why wouldn't you? I don't even think it would cost more than few cents of cost to do so since it is very likely that you have open A/D 's just sitting around on your micro anyway. Oh well, at the end of the day, I can't spend enough time and money to design something for real on a speaker to prove my theory one way or another while you do this for a living I do believe that powered speakers are going to become more intelligent in the future. DSP is going to become very important. In theory, one could DSP a pretty crappy set of drivers into sounding just as good as really good drivers without a DSP. This theory assumes that between the amplifier and the drivers, the capability exists to create the output frequencies desired. Everything else can be done in the DSP algorithm. All you would need for the most ultimate feedback would be a RTA mic input on the speaker system and the thing could eq the room for you too. While it is at it, it could keep you from clipping by auto-reducing the gain if you reach the limits. I see lots of firmware in your future. You can get some pretty integrated SOC's these days by using a PPC core and putting your own FPGA DSP logic blocks around it. ASIC's have become pretty easy to integrate into designs in a cost effective manner. Do you use ASIC's or off-the-shelf micros? Link to comment Share on other sites More sharing options...
Members madjack Posted January 2, 2011 Members Share Posted January 2, 2011 All you would need for the most ultimate feedback would be a RTA mic input on the speaker system and the thing could eq the room for you too. While it is at it, it could keep you from clipping by auto-reducing the gain if you reach the limits. I'll be the first to admit that I don't have a clue as to any of this, but this sounds to me like what a Driverack purports to do. Link to comment Share on other sites More sharing options...
Members Bobby1Note Posted January 2, 2011 Members Share Posted January 2, 2011 Also I'd like to sell my york system if i could. Had all these people interested but now that I'm ready nobody has any money. I just dug through the lint in my pocket, and found a wee bit. Where you located? Link to comment Share on other sites More sharing options...
Members OneEng Posted January 2, 2011 Members Share Posted January 2, 2011 I'll be the first to admit that I don't have a clue as to any of this, but this sounds to me like what a Driverack purports to do. Indeed it does. What I am saying is that in the future, a powered speaker will have the same ability built into it. In fact, the powered speakers we have been discussing already do much of what the driverack does, it just isn't user adjustable. Cross-over - hard wired in the speaker for 2 way and the crossover point is not flexible Equalizer - in the speaker there are 2 settings only today, I contend that there will be a much more flexible setup in the future. Limiter - Fixed in the speaker Compression - don't know if the DSP's put any compression on the signal or not Time alignment from driver to driver Things that speaker management systems do that are not currently (as far as I know) in powered speaker DSP's RTA and auto-eq Time delay/alignment from sub to tops Time delay from speaker to speaker (for more than 2 FOH speakers like in a stadium) Feedback elimination Subharmonic maximizers I am simply predicting that the features that were needed in a speaker management system are needed in powered speakers for the same reason. It only makes sense that we will be seeing these features in powered speakers in the future Link to comment Share on other sites More sharing options...
Members DaBender Posted January 2, 2011 Members Share Posted January 2, 2011 I think they'll add reverb. They've basically taken their cue from guitar combo amps. Link to comment Share on other sites More sharing options...
Members OneEng Posted January 2, 2011 Members Share Posted January 2, 2011 I think they'll add reverb. They've basically taken their cue from guitar combo amps. Don't know why I didn't think of that too I am sure you are correct. Link to comment Share on other sites More sharing options...
Members agedhorse Posted January 3, 2011 Members Share Posted January 3, 2011 Reverb, surely not. I don't want verb on the entire program whatever it may be. The reason why feedback is not important on a speaker is because the transfer functions are so well known and behaved, and the amplifiers so close to a perfect voltage controled voltage amp (VCVA) that virtually every other variable has more influence on the sound. The room is one example. The reason why a mic and feedback for realtime auto eq, feedback reduction or delay of the speaker is not really all that useful is because as you add a second speaker source, you end up with a second output signal that will be picked up by the mic that is not controlled by the first contro, system leading to a stability quandry. (ie. hunting, oscillation, inaccuracies use to cross-control issues). There are DSP controlled and networked higher end line array boxes out there but the adjustments are not made in real time via feedback control. And, the reason why most DSP control of powered speakers is transparent and non-adjustable is to facilitate better plug and play performance to the average user... and even the above average user frankly. Bridge switches on power amps should also be "restricted use" controls IMO. Link to comment Share on other sites More sharing options...
Members OneEng Posted January 3, 2011 Members Share Posted January 3, 2011 Reverb, surely not. I don't want verb on the entire program whatever it may be. The reason why feedback is not important on a speaker is because the transfer functions are so well known and behaved, and the amplifiers so close to a perfect voltage controled voltage amp (VCVA) that virtually every other variable has more influence on the sound. The room is one example. The reason why a mic and feedback for realtime auto eq, feedback reduction or delay of the speaker is not really all that useful is because as you add a second speaker source, you end up with a second output signal that will be picked up by the mic that is not controlled by the first contro, system leading to a stability quandry. (ie. hunting, oscillation, inaccuracies use to cross-control issues). There are DSP controlled and networked higher end line array boxes out there but the adjustments are not made in real time via feedback control. And, the reason why most DSP control of powered speakers is transparent and non-adjustable is to facilitate better plug and play performance to the average user... and even the above average user frankly. Bridge switches on power amps should also be "restricted use" controls IMO. I agree, there are few times when the entire mix would be served by reverb Feedback though? While the speaker physics may be perfectly know (I disagree even with this since things change in different atmospheric conditions and degradation of components over time), the room and the microphone placement is not know... or even consistent through a single gig. I am sure as a soundman, you have gritted your teeth as the lead singer decides to "go out" to sing on the dance floor ..... directly in front of one of the mains ...... with his mic pointed right at the high horn. (my personal opinion is that lead singers should have shock collars to help prevent bad mic usage ). As far as the linking of speakers, I agree. In order to do much of what I have said could be done in the future, the speakers would need a method of communicating with each other. Things like overall eq and feedback suppression could be controlled via point to point communications like wireless bluetooth for very little cost in hardware (the chips cost ~$2-5 with all components). I guess in retrospect, it is never going to be convenient to walk to the speaker to change things that are traditionally changed on a mixer; however, things that are traditionally "set and forget" like speaker management will likely be moved into the powered speaker electronics IMHO. Link to comment Share on other sites More sharing options...
Members OneEng Posted January 3, 2011 Members Share Posted January 3, 2011 Ok, I changed my mind The speaker will contain every control that is in both the speaker management system as well as in the mixer (and efx rack). All central controls and box to box communication will be done through wireless. Once this is done, you can use either a laptop/desktop for the control center and/or a mobile phone. This is the speaker/system of the future Once everything is included in the speaker, hookup and control are very simple. P.S. I suspect it is going to take ~10 years for something like this to make it to market. For the most part, I suspect that speaker companies see themselves as mostly a hardware company. The shift to being a technology driven by software will take a while, but it will come The hardware is the simple part of my proposal. Link to comment Share on other sites More sharing options...
Members agedhorse Posted January 3, 2011 Members Share Posted January 3, 2011 Ok, I changed my mind The speaker will contain every control that is in both the speaker management system as well as in the mixer (and efx rack). All central controls and box to box communication will be done through wireless. Once this is done, you can use either a laptop/desktop for the control center and/or a mobile phone. This is the speaker/system of the future Once everything is included in the speaker, hookup and control are very simple. P.S. I suspect it is going to take ~10 years for something like this to make it to market. For the most part, I suspect that speaker companies see themselves as mostly a hardware company. The shift to being a technology driven by software will take a while, but it will come The hardware is the simple part of my proposal. Here's a good example of not grasping the big picture of what the average user hopes for. The average user NEEDS a plug and play solution, something that takes all of the guessing, hoping, connecting and manipulating of technology away from inquiring minds. Hell, the AVERAGE musician can barely hook up and use a powered mixer to full potential. The greatest number of variables must be removed from the equation, features that would only be used by a very few get in the way of the rest of the users that just want the product to do the basic stuff but to do it well, reliably, repeatadly and easily. Speaking of this, my cell phone company still can not deliver what it promises. The features do not work reliably, they get in the way of the sole reason I bought my cell phone... to make and receive phone calls. If I can't receive phone calls reliably, and the features get in the way of the operations, everything else sucks rocks no matter what the promises. Engineers really need to learn when to stop {censored}ing with the product concepts and focus on making solid, reliable, easy to use, productive products. I learned from some pretty old school engineers and boy do they have some critical thoughts about what's coming out on the market these days. Far worse ranting than I do. Link to comment Share on other sites More sharing options...
Members OneEng Posted January 3, 2011 Members Share Posted January 3, 2011 LOL. I figured that would be your opinion of such a gimik-ridden speaker Yes, your cell phone example is very adept. Just like cell phones, speakers will evolve to include all the bells and whistles and they will do it for the same reason..... it sells. Once a microcontroller gets put into a device, the device is never the same Link to comment Share on other sites More sharing options...
Members Mutha Goose Posted January 3, 2011 Members Share Posted January 3, 2011 Engineers really need to learn when to stop {censored}ing with the product concepts and focus on making solid, reliable, easy to use, productive products. I learned from some pretty old school engineers and boy do they have some critical thoughts about what's coming out on the market these days. Far worse ranting than I do. I can tell you from my own personal experience in design; when I got my start, the engineers designed a product & stated the parametric performance, and the marketing weenies sold it. Now, those same weenies are not only defining the design targets regardless of physics, and are also the ones creating the spec sheets. "Can't hit power requirements at 0.1%THD? No prolem, state it at 10% and no one will know the difference." "Max out? Who cares if you can only reach that for 5 seconds before thermal shut down? You hit it for a moment, right? Print it" It's funny, as an engineer, I am judged on my technical integrity. Why, when marketing controls the message anyway?!?!" Link to comment Share on other sites More sharing options...
Members Rumorsband Posted January 3, 2011 Author Members Share Posted January 3, 2011 I just dug through the lint in my pocket, and found a wee bit. Where you located? I am in Rhode Island Link to comment Share on other sites More sharing options...
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