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Ukko Kari

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Everything posted by Ukko Kari

  1. Faital 15PR400, BMS 4594 MTM

    For the moment I will just replace it with another. I have a bunch of filters loaded in a DCX 2496, but work has been crazy, lots of overtime and not enough free time to swap it in with a QSC. Fast and easy to flash another Nu3000DSP. Crown should have taken notes from Berhinger on the user friendliness of the DSP.
  2. Faital 15PR400, BMS 4594 MTM

    Vertically pair symmetric 15” woofers with BMS Co-axial M/T, a practical approach. As with most design decisions, rubber meets the road, eventually. This project has culminated a few years of planning and gone through at least a half dozen design decisions prior to execution. Life gets in the way sometimes, and it's not just analysis paralysis. What is practicality with regards to loudspeakers? In short, the ability to move from location to location without problems, and easy integration into different rooms and situations with similar performance characteristics. Fully horn loaded loudspeakers can exhibit polar performance that is boundary independent, however their sheer physical size can limit placement, movement and practicality. Small direct radiator loudspeakers of the domestic type exhibit poor distortion performance, a trade-off of small size, yet are easily moved and installed in various locations and room sizes. http://www.prosoundtraining.com/site/synaudcon-library/how-boundaries-affect-loudspeakers/ Larger, professional speakers intended for sound reinforcement work exhibit better distortion performance than small domestic bass limited loudspeakers, albeit some trade off smoother polar response for sheer output levels. Lately, the trend in the DIY world has been gravitating to larger loudspeakers with horns or 'waveguides' that exhibit smoother frequency response, well defined polars, with less trade-offs than a typical small domestic type loudspeaker. Beam steering: Typical TMM layouts have lobe steering effects that exhibit frequency dependent polar tilt. On the other hand, the woofers will exhibit better coupling at the high end of their response. In this MTM application, the polar radiation will start out omni-directional at low frequencies, smoothly transitioning to a 60 degree wide by 50 degree vertical pattern. Trade-offs are with the 21” deep +-boxes, a sheet of rockwool will be installed to fill in the space between the speaker baffle and the room corner, without, there will be a null in the 60-80 hz area from a 1/4 wave reflection. The low frequency drivers are neodymium motor, 3” AL coil Faital Pro 15PR400's, in 4 ohm configuration, wired in parallel for 2 ohm nominal load, with an EBP of 106.1. DC Re is 3.3 ohms, and Bl^2/Re falls at 56.7. No inductance countermeasures, however overall inductance of 0.6 mh is low, a result of the short coil winding depth and thickness. Le/Re from the manufacturers reported numbers is 0.182 milliseconds. Not a lot of steel in this lightweight 8 pound woofer. The low frequency drivers will cover approximately 4 octaves of sound. http://www.faitalpro.com/en/products/LF_Loudspeakers/product_details/index.php?id=101060101 Simulation of a 9.5 cu ft vented box shows that in the range of interest, between 40 hz and 160 hz, the minimum impedance result would be 4 ohms, 2 ohms with woofers in parallel. This drops to 1.65 ohms in parallel from 160 hz up to the crossover point. Impedance maxima occurs at 66.67 hz with a 40 hz tune and as a result, with parallel woofers, 16.589 ohms. At a drive level of 300 watts, we can see that in band between 160hz and crossover point, this results in 6.742 amps through each voice coil. At the impedance maxima of 66.7 hz, 3 amps of current. Lower the drive level to 50 watts, 160 hz + would result in 2.75 amps through each coil. 40-160hz average 2.5 amps, with only 0.86 amps through each coil at the impedance maxima. Horn is an 18 Sound XT1464, and the 1.4” throat co-axial compression driver is the BMS 4594, with 8 ohm diaphragms. The elliptical horn mouth circumference as measured by string is 43”, and depth to the bug screen on the driver is 10”. 18 sound recommends an 800 hz crossover, albeit from their graphs it is mostly "usable" to 500 hz. In the range of transition from horn to woofers, there are 3 distinct radiating elements, with center to center distances of 13.5 inches. 600 hz has a wavelength of 20.86 inches, and this inter-driver spacing is less than 2/3rds of a wavelength. If the crossover is pushed down to 500 hz, acoustically the drivers are even closer together. A High Power Satellite Speaker, Joeseph D'Appolito, Speaker Builder, April 1984. http://www.xlrtechs.com/dbkeele.com/PDF/Keele%20(2007-09%20AES%20Preprint)-%20Linear%20Phase%20Digital%20Crossover%20Flters%20Part%202.pdf http://www.eighteensound.com/Products/Articles/Detail/catid/4063/eid/3347/xt1464 http://www.bmsspeakers.com/index.php?id=bms_4594nd In dissecting the work of Electrovoice, D'Appolito, and Keele, for a c-c distance of 13.5 inches and a crossover frequency of 600 hz, this will result in a vertical beam width of 80 degrees widening below 600 hz towards omni, and tighter beam width above, likely resulting in a mostly smooth transition from omni to 50 degree beamwidth over approximately an octave. http://www.electrovoice.com/downloadfile.php?i=1270 Bracing of some sort is required to keep the large panel surfaces from radiating sound on their own. 3” deep Roxul Safe N' Sound will reside in the cavities formed by the bracing, along with a sheet in the middle of the enclosure behind each woofer. Care has been taken to ensure a clear path from top to bottom around the rear brace windows. External size has been set in stone at 60” tall, 21 inches deep and 17.75” wide, resulting in a gross internal volume of 10.727 ft3. Bracing is extensive, with small spans. The front baffle, as the rest of the enclosure is single layer, each brace is 3” deep. The rear and front are connected together, as are the sides. In short, every side is connected to the opposite and at least one adjacent side. The prototype has been constructed with butt joints and PL Premium adhesive, utilizing many pocket screws for both mechanical strength and clamping during assembly. 3/4” birch B3 plywood was used. Assuming the prototype measures well, it will become the center channel in an L/C/R triplet, with only the center residing behind an AT screen. Why MTM? I chose MTM to minimize ceiling and floor reflections, and minimize the effect of lobe steering at different frequencies compared to a TMM layout. I have had experience with both the Yorkville Unity U215, and JTR's Noesis 215HT, both MTM configuration. http://yorkville.com/loudspeakers/unity/product/u215/ http://jtrspeakers.com/home-audio/noesis-215rt/ A visual representation by Patrick Bateman of the MTM alignment is visible here: http://www.diyaudio.com/forums/multi-way/301131-illustrated-guide-waveguide-array-radiation-2.html#post4927141 One higher profile MTM build with the 18 sound XT1464 horn for the Atlanta DIY meet, Paul W's “Raptor”: http://www.htguide.com/forum/showthread.php?35812-The-Raptor-a-10-quot-MTM&highlight=xt1464 Why vented? Some would argue there is no need for ports in a main loudspeaker, especially with this much radiation area. I disagree. Many prefer a limited bandwidth sealed box, and it can ease placement woes. The 12 db slope from an 80 hz f3 will match up well with the slope built into AVR's, and give you 24 db total slope. With a decent vented design, you can have your cake and eat it too. Of course, this results in other trade-offs. Keeping the baffle surface as narrow as possible, and using the largest ports that are practical results in a slightly deeper depth cabinet. If I want to emulate a sealed 80 hz f3, I can do that actively. I model for worst case Ontario ( Trailer Park Boys ) and adjust accordingly. Even a modest average drive level will increase the motor temperature above ambient, and as the motor strength drops, the woofers 'like' a larger box. Standing waves calculated from internal enclosure size Top to bottom 116 hz ~ 58.5 inches Front to back 348 hz ~ 19.5 inches Side to side 417 hz ~ 16.25 inches The ports consume a large amount of space inside the enclosure, with an I.D. of 7.625”, 2 per enclosure. Port cross sectional area >91 in2. Group delay approaches 18 ms @ 40 hz, below 1 cycle, with an appropriate Butterworth high pass filter in place. At 300w PIN, the velocity in the ports will be 7.061 meters per second, or 23.166 ft/s, comparatively low with regards to commercial designs. Tuning is for the moment, going to be a variable. For the moment, they will be removable through the lower woofer opening. Once tuning is confirmed, they can be glued in place. Port self resonance should fall within a range of 520-660 hz, depending of course on how short they must be trimmed for tuning to be close to 40 hz. BMS C8-8 passive crossovers will handle the midrange to tweeter crossover, with the aid of a UT3636 Autoformer to attenuate the level and reduce hiss from the BMS co-axial compression driver, and better make use of the gain in the system. Each loudspeaker enclosure will have it's own dedicated QSC PLXII 2502 amplifer, with both Faital Pro woofers in parallel on one channel, and the other channel reserved for the BMS co-axial compression driver. Active crossover duties from woofer to compression driver will be handled by a Behringer DCX 24/96 unit. Measurement will consist of a Behringer ECM 8000 microphone, phantom power supply and sound card. Horn data will be derived from 1m measurement, woofers from 4 meters, adjusting with offset in RoomEQ wizard. Acoustic crossover will be 3rd order. The use of 2 15” direct radiators in a 9.5 cu ft 40 hz tuned cabinet affords low diaphragm displacement for relatively high sound pressure levels. Achieving low FM and AM distortion should not be a problem. Kellogg in 1931 proposed 0.1 cm as a maximum amplitude of motion. At a sound pressure level of 110 db / 1m on axis, the diaphragm motion is approximately 1 mm as simulated. A moving diaphragm alters the pitch of a high frequency sound, modulating it up or down in frequency. The frequency deviation by of itself is not that deleterious, however if the diaphragm is reproducing simultaneous 50 hz and 440 hz signals, the result will be 440 hz [+- 0.4 hz], plus additional side bands of 440 hz [+-50 hz]. In other words, the transducer will be producing 390 hz, 440 hz, ( +-0.4 hz modulation ) and 490 hz output. This active loudspeaker project shall be known as Väinämöinen.
  3. Faital 15PR400, BMS 4594 MTM

    Tonight listening to music fairly loud, hitting the peak limiters in the iNuke 3000 DSP, suddenly I had red rings of death. I did have a saved preset file, and also some screen shots of the DSP settings. The amp made 6 years before it failed.
  4. Bulding the Room2 listening room

    I like your format. Even though the room will dominate in the actual measured response, outdoor GP is the best measure, since all devices are on an equal playing field. A 'FAQ' or frequently asked questions section on your website can answer a lot of the same repetitive questions that can tie up your time replying to inquiries. Well informed customers can then email you for additional information on your products.
  5. Eventually ( after the new year ) I will be looking at a new sound card and a measurement mic. Currently I have a Behringer ECM8000, and UCA-202 sound card. From this post on the HT Shack, it appears the self noise of the ECM8000 is higher than I would like to see: http://www.hometheatershack.com/forums/spl-meters-mic-s-calibration-sound-cards/10224-ecm8000-microphone-measuring-techniques-usage-discussion-10.html#post175019 I can pick up the Focusrite Scarlett 2i2 without breaking the budget, and from measurements that notnyt posted, looks like a great interface. http://www.avsforum.com/forum/155-diy-speakers-subs/1780914-measuring-amplifiers.html Any solid recommendations for mics that aren't stupid expensive, yet have decent response to at least 20khz? Low self noise and the ability to use common 48v phantom power would be great.
  6. Sound card and measurement mic upgrade

    Just an update, I was gifted a Behringer UMC204HD at xmas. I could not get it to play nice with Windows 10, even with the latest drivers, not sure if it was defective or not, but searching the internet other people had similar experiences, swirling garbled audio, regardless of connection. It went back to the vendor, and there is a credit to use up. Still thinking of the Scarlett 2i2 2nd gen, any other suggestions?
  7. Faital 15PR400, BMS 4594 MTM

    Now that I have spent time with voicing the speaker, I haven't worked up the ambition to dial in a crossover on the Behringer DCX 2496, and move to the QSC power amps. Not to mention the other projects on the go! Cold weather puts a damper on working outside.
  8. X-curve compensation re-EQ

    You just need the right guy to move and install those wooden refrigerators. https://en.wikipedia.org/wiki/Magnús_Ver_Magnússon
  9. Sound card and measurement mic upgrade

    From Ethan Winer: http://ethanwiner.com/myths.html Myth: Absolute microphone or speaker polarity makes an audible difference. Fact: While nobody would seriously argue that it is okay to reverse the polarity of one signal in a stereo pair, I've never been able to determine that reversing the polarity of one signal - or both if stereo - ever makes an audible difference. Admittedly, it would seem that absolute polarity might make a difference in some cases, for example, when listening to a bass drum. But in practice, changing the absolute polarity has never been audible to me. You can test this for yourself easily enough: If your console offers a polarity-reverse switch, listen to a steadily repeating bass drum hit and then flip the switch. It is not sufficient to have a drummer go into the studio and hit the drum while you listen in the control room, because every drum hit is slightly different. The only truly scientific way to compare absolute polarity is to audition a looped recording or drum sample, to guarantee that every hit is identical. Important Update: Mike Rivers from Recording magazine sent me a test Wave file that shows absolute polarity can be audible in some circumstances. The polarity.wav file (87k) is a 20 Hz sawtooth waveform that reverses polarity in the middle. Although you can indeed hear a slight increase in the low end fullness after the transition point, I'm still not 100 percent certain what this proves. I suspect what's really being shown is a nonlinearity in the playback speaker, because with a 50 Hz sawtooth waveform there is no change in timbre. However, as Mike explained to me, it really doesn't matter why the tone changes, just that it does. And I cannot disagree with that. More Update Info: After discussing this further with Mike in the rec.audio.pro newsgroup I created two test files you can download and audition yourself. The Kick Drum Wave file (324 KB) contains a kick drum pattern twice, with the second reversed. Play it in SoundForge or any audio editor that has a Loop mode, so you can play it continually to see if you hear a difference. The Voice Wave file (301 KB) is the same but with me speaking, because Mike says reversing polarity on a voice is surely audible. I don't hear any difference at all. However, I have very good loudspeakers in a room with proper acoustic treatment. As explained above, if your loudspeakers can't handle low frequencies properly that could account for any difference you might hear.
  10. Sound card and measurement mic upgrade

    Regarding reversed polarity, consider the example of simply using the device for playback: Instead of a forward motion on the speaker diaphragm, you will have a rearward movement. This will change our perception of the sound of a completed system if this is not corrected, especially in the bass, not so much in the region where wavelengths are short, and small head movements will cause many rotations of phase.
  11. Pushing a horn: how to measure (with an UMIK1).

    I concur, a boom mic stand is a solid investment when it comes to measurement gear. Here is an example from Amazon: https://www.amazon.com/AmazonBasics-Tripod-Boom-Microphone-Stand/dp/B019NY2PKG/ref=sr_1_3?s=musical-instruments&ie=UTF8&qid=1513464419&sr=1-3&keywords=boom+mic+stand I am sure you should be able to source one locally similar to the example. It does not need to be a heavy duty model, and the slimmer the better. ( you do not want reflections from the mic stand to reach the mic capsule )
  12. Pushing a horn: how to measure (with an UMIK1).

    Also of note, you must ensure that the settings in Windows ( I assume you are using Windows, not Mac or Linux ) match those set in REW ( 2 channel, 44.1 khz ) as these can be set differently, and your results can be wonky. Every time I use REW, I have to change the settings for Windows under the sound device.
  13. Pushing a horn: how to measure (with an UMIK1).

    Good question, with my duplex sound card, I use a loopback cable, in essence, returning the output of the sound card to the input. The software calculates the round trip latency and deducts that from the signal arriving from the microphone input, thus giving you your delay measurement. On one channel, the microphone is connected, on the other, the loopback cable. Alternately, REW can use a high frequency device to calculate delay. This requires the sweep to be set full range, and it will listen for the high frequency chirp before the measurement. ( Typically only used for full range loudspeakers ) Yes your Marantz has time delay functionality. I am not familiar with Audessey products, and if utilizing it will give you an appropriate delay. Someone more familiar may be able to help you out in this regard. Since the UMIK-1 is a USB device with no other input, you could add a second sound card with just a loop back cable in order to have REW calculate the time delay. In the drop down menus in REW, you have the option of telling the software what inputs and outputs you would like to use. Hope that helps you.
  14. Sound card and measurement mic upgrade

    I have done some more research on sound cards, and came up with some interesting information. The Behringer UCA-202 has reversed polarity on both the input and outputs, the only one in the test to exhibit this phenomenon, according to this website: http://www.daqarta.com/dw_gguu.htm Polarity: The Polarity tests were done using a separate scope. The Generator was set to produce a biphasic Pulse waveform, such that the positive phase preceded the negative phase, and there was a dwell time at zero before the next pulse. The output had normal polarity if the positive phase appeared first on the scope. A loopback cable was then used to feed the output back to the input, and if the Input waveform seen by Daqarta also showed the positive phase first, the input likewise had normal polarity. This was the case with all devices except the Behringer UCA202, which inverted both the output and input (which means that it could not have been detected with a loopback alone). Read this thread on DIY audio, might have second thoughts on the Focusrite Scarlett 2i2 sound card. http://www.diyaudio.com/forums/equipment-and-tools/301166-focusrite-scarlett-2i2-2nd-gen-measurements-whats.html This thread has me looking at another Behringer: http://www.diyaudio.com/forums/equipment-and-tools/312454-usb-audio-interface-measurement.html
  15. Pushing a horn: how to measure (with an UMIK1).

    Typically, horns with short path lengths are used in multiples, to extend the low frequency capability, when tightly packed. This appears to be a front loaded horn, which below the corner where the horn is effective, is still an 18" woofer in a small sealed box, so you can apply a modest bit of eq to boost the bottom end, but will have to keep in mind the short throw of the driver. I would only recommend doing so in a domestic environment like yours, not in a dance club or other pro use. With REW, you can play a sweep through the subwoofer from say 25 hz to 200 hz, and look at the response. I am not familiar with the UMIK-1 mic and interface, other than having read of the possibility of clipping the signal with high sound pressure level. This is dependent on the dip switch settings inside the mic on the preamp board. http://www.avsforum.com/forum/155-diy-speakers-subs/1797489-massively-clipped-umik-1-a.html People have received the UMIK-1 with different dip switch settings, both 12 db and 18 db. If you do not have a sound pressure level meter, you could likely use an application with a smartphone to set the level to calibrate the level with pink noise in REW before making a sweep. Start by measuring the time delay of the subwoofer, you will need that to tell the Marantz to delay all other channels with respect to the bass horn.
  16. Bulding the Room2 listening room

    In a low volume setting, I would agree. Once the levels get above where the typical domes run out of excursion, there may be some that can tell the difference.
  17. Sound card and measurement mic upgrade

    I can certainly wait a bit longer to grab something in that price range. Have a link?
  18. Faital 15PR400, BMS 4594 MTM

    Life has been busy, second and third are partially finished, but not complete.
  19. Replacement AVR / processor

    Not sure why, but playing audio from computer > UCA-202 sound card > optical input of Denon DN-700 AVP results in a small 'thump' every time there is no sound output, IE: between songs, or between videos on YouTube. There were no issues previously with the same sound card to an old Marantz. Zero thump / pop noise. Using XLR from the BD player, it is dead silent between songs, no thump or pop noise at all. If that is the case, I may have to look into a new sound card as well, something with a lot better cross-talk performance between channels. I have been looking at acquiring a Focusrite Scarlet 2i2, based on what I have read. Any other suggestions?
  20. Replacement AVR / processor

    My ~16 year old AV Receiver is starting to get tired, I had to disassemble it to do some soldering this spring. I am not inclined to pay $ 3K for features I will never use, at the moment Atmos is not on my radar, nor will it for the next 5 years or so. http://denonpro.com/products/view/dn-700av#.WMyEHn_QBco Any experience or thoughts on this unit? Checks off all of the boxes with balanced outputs. Also looking to get a new BD player: http://denonpro.com/products/view3/dn-500bdmkii
  21. Replacement AVR / processor

    The lighting around the knobs is too bright for my taste, and too bright to use on the front wall with a projection screen. I may have to install some O-rings on the knobs like others have done on the iNukes to limit the direct viewing of the white ! LED's.
  22. Replacement AVR / processor

    Just inserted into the chain, and powered it on for the first time. The knobs are lit around the outside, with recessed lighting rings that make the knobs appear to float. They are bright, I haven't investigated whether or not they can be turned off yet.
  23. Bass system integration

    I concur that the time portion of the system response is what we must strive for correctness. Time of flight, processing delay needs to be accounted for. What I have done: in a multi-sub system, measure time of flight ( with no DSP engaged ) from each device in it's intended location to the main listening position, and compare that to the main L/C/R channels. I helped set up a system with subs that were radically different in distance, with a nearfield sub behind the MLP area. In that particular case, the closest subwoofer was set for level, and run at a bandwidth of <40 hz. Delay was set so it's time of flight was identical to the average of the main L/R speakers. The other 3 subwoofers in this large room were set so their time of flight was identical. The pair closest to the L/R were run <120hz ( main speakers were not on the level of some enthusiasts here ) The third subwoofer was run <60 hz. Subwoofers were not gain matched, the one behind the MLP was reduced considerably in level, it's 4kw amp was just idling barely with respect for the required levels. Processing added 1-1.5 msec of delay, so the main and subs required a bit of fine tuning. Once all 4 were playing 'nice' with respect to each other and the main L/R, Anthem room correction was run. Every situation is unique, and requires a customized solution. You may get better smoothing in the upper bass running all subs as full bandwidth, but may result in a softening of the attack across some of the listening positions, if there are many, due to the various complex phase relationships in different physical locations. Longer wavelengths ensure that you will have better success at integration. It's not hard to get sources within 1/4 wavelength at 20 hz, or even at 40.
  24. Replacement AVR / processor

    Just received notification that the DN-700AVP has shipped.
  25. New SVS Ultra 16"

    The product page says edgewound coil: https://www.svsound.com/products/pb16-ultra Taking a stab at the finished enclosure dimensions after bracing and ports of ~ 7.5 cubes and plugging in the numbers in WinISD, letting it autocalculate the unknowns, the actual power across the coil looks surprisingly low, with a 47 ohm impedance peak in the 40 hz range.