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Showing content with the highest reputation since 09/19/2018 in all areas

  1. 1 point
    I'm sorry guys, I'm doing a terrible job of updating my thread. This year has just been crazy for me. I'll eventually update! I wanna at least put up a measurement of my speaker, even if it is an older measurement, and the current speaker performs better. There is no smoothing at all in this measurement. I want to repeat this again, because accurate measurements should never be smoothed. The measurements are in 20 degree intervals, 0, 20, 40, 60, 80, 100 degrees. I have measurements in 10 degree intervals, but not for outdoor ground plane. I have limited battery power when measuring outdoors, so I did the outdoor measurements in 20 degree intervals. <900Hz it is ground plane measurements. Above 900Hz it is quasi anechoic with 10.5ms of gating. The speaker is even better now, since the minor crossover dip is now moved to ~3100Hz (the most sensitive part of our hearing, and also the region where the ear will be least noticeable to dips), and steered so the deepest dip happens at around 40 degrees instead of 20. The midrange is brought up to be flat with the treble as well. The graph has less wiggles due to further removal of sources of measurement reflection. But I don't have a complete set of measurement yet. Measurements is really hard, much, much harder and more complex than people think. I've literally done thousands of measurements on my speaker and I still don't have measurements that I would consider to be accurate. I do think I have the methodology down now, but just don't have the time. To give an idea of what's required: For each axial measurement, it takes 3 splices - Ground plane measurement for <1000Hz - Quasi anechoic measurement for 1000-10000Hz in a large space like gym or auditorium to achieve ~10ms gating. The typical 4-5ms reflection free time achieved in a typical room just doesn't have enough resolution - Flush mounted microphone for >10000Hz measurements to avoid reflections from the mic clip and mic stand. Then the ground plane measurement has to be compensated because it shows the result of a double height baffle since the ground is a reflector. A compensation factor of the difference between the double height baffle and the normal height baffle needs to be applied. Then for polar measurements, things get real tricky. The standard way of rotating a loudspeaker on a turntable does not give the true angle and distance because the centre of rotation is at the centre of the speaker instead of at the centre of the baffle. Therefore, there needs to be angle and distance compensation applied. The angle compensation needs to be done physically by moving the mic. Distance compensation can be done after measurement. So as you can see, for polar measurements, which I do in 10 degree intervals, requires 36 measurements for horizontal, 36 measurements for vertical, so 72 measurements. Multiply 72 by 3 for 3 different drivers, and you get 216 measurements. Then another 72 measurements for the final response for a total of 288 measurements. But wait, I have 3 measurement conditions, ground plane, quasi anechoic, and flush mounted mic! So that's 864 measurements! Therefore, I have to do 864 measurements to get a complete polar response for this speaker, AND I have to do the baffle, angle, distance, and splicing compensations for all of those measurements! Yes, there are some shortcuts that'll cut the work by half, but that's still a crazy amount of work. One day I'll actually do all that, but not now. And yes, every one of those complications is important to do in order to get a truly accurate measurement, and it makes a HUGE difference.
  2. 1 point
    Check the Othorn files again. I put the original print in there. The dimensioning is clearer.
  3. 1 point
    Hey everyone, first I would like to thank you all members of this forum, you have inspired me to build myself some speakers. I also like to say, thank you Ricci for sharing the Othorn plan and giving all your support to me as a diy. As you probably know by now I will start to build Othorns to make my soundsystem complete. We are right in the beginning of this project, the speakers arrived today and the building start is not yet set. My goal with this build is to make the Othorns to look like the rest of the sound system. Please feel free to guide me through this project, ask for pictures if you like. I will gladly share this building project with you guys, you have giving me so much and I'll like to give something back in return. //Mattias
  4. 1 point
    Keep us updated!! JSS
  5. 1 point
    ok problem solved, adding an aformat step (to convert to dbl and then back to s32) around the biquads fixes the issue.
  6. 1 point
    If it were me, I would probably study the source code more carefully to try to figure out what it will do, but a quick test would be to try a pair of filters that cancel each other, for example: Low Shelf gain=+X; Low Shelf gain=-X at a common frequency, where the X is some relatively big number that will cause clipping depending on how much internal headroom the processor has. After running a track through the pair of filters, the input and output should be very nearly identical unless clipping or severe precision loss occurred. If the internal processing is floating point, then internal headroom and precision should be very high.
  7. 1 point
    Just a small update: My horn sub project in the form of @lilmike's MicroWrecker tapped horn has been initiated. I've ordered two B&C 15TBX100 15" drivers for a pair of Micro's, so plenty of headroom should be available. They'll be build in 11-layer Russian birch ply by a cabinet maker I know. Amp and DSP solution as of yet not determined.. /Mikael
  8. 1 point
    There seems to be some confusion about what the app actually is at this point in time so let me clarify. It's an interactive *per channel* minimum phase filter designer with the tools required to quickly and easily work with either mono bass managed tracks or multichannel tracks, i.e. designing pre or post BM BEQ filters. Interactive means it must be quick hence the filter view is based on the transfer functions. Obviously this wouldn't work if we were trying to combine channels but we're not (except when extracting the source track which is pre filter) so this is fine. A post filter clipping check is something I am aware of and had logged it at https://github.com/3ll3d00d/beqdesigner/issues/19 a while ago. This isn't hard to implement (both sox and ffmpeg can apply biquads) so it's just a question of time and desire to implement the feature. One could also implement this in python using scipy or there are other python libs (with an underlying C impl, e.g. http://ajaxsoundstudio.com/pyodoc/) that could also be used if scipy is too slow. Having said that I would have thought that would be something that happens relatively infrequently as a final check so working with existing cli tools seems fine to me and would be quick and easy to implement. @Kvalsvoll the bit I don't get is why you want to remux that back into the original track. If you're playing an mkv then you're already on a computer that can do the filtering in real time so why would you want to alter the source itself?
  9. 1 point
    if you uncheck the "mix to mono" checkbox and click extract then once it finishes, the button should change to "Create Signals" and the field at the bottom will be enabled. Put something in here and click "create signals". https://imgur.com/a/uoOh3Hr It should then automatically close the dialog and add each channel as a separate signal using the channel names (taken from https://trac.ffmpeg.org/wiki/AudioChannelManipulation#Layouts) https://imgur.com/a/y7GGUU2 If this doesn't happen then feel free to log something with appropriate steps to reproduce/pics/supporting files over at https://github.com/3ll3d00d/beqdesigner/issues similarly if you think of any interesting features then do also feel free to suggest them, I'll probably work on this for a little while longer at least.
  10. 1 point
    So I see that BEQ has suddenly taken off on AVSForum, which appears to have partly inspired @3ll3d00d's designer software. These are very positive developments. However, I'm a bit disappointed to see that the BEQs posted to AVSForum are intended to be applied to all channels, and there appears to be little if any post-BEQ QC performed. Instead, most of these seem to involve EQing the PvA to flat and calling it enough. I even see people arguing that the quality of a BEQ should be judged "objectively" by how smooth or flat the resulting PvA is. Ugh! I fear many of these BEQs may be doing more harm than good to the track. As we know very well here, a PvA is not a reliable predictor of perceived tonal balance on a track. It's certainly informative, but is nowhere near definitive. There's really no way to know how something sounds without listening to it on a good "reference" system and making a subjective judgment. Undoubtedly, this is complicated by the facts that personal preferences vary and that it is not yet known how to calibrate different bass systems to sound exactly the same, but I don't know of a better way to deal with the problem. It's probably perfectly OK if there are multiple BEQs out there. Different people will have different insight and of course will hear different things. FWIW, I have a pretty aggressive house curve on my system which arises from my novel calibration approach based on the concept of apparent power. Curiously however, my approach leads me to a curve that tops out around 20 Hz and is somewhat diminished (by a few dB) below there. Furthermore, I've noticed that soundtracks I like also frequently have a bit less ULF than 20-40 Hz bass. They don't have a steep shelf or HPF but often they don't push levels below 20-40 Hz that much. I think @maxmercy gets this right by looking at each channel and trying to ascertain what filter/filters were used rather than just making the PvA look pretty and listening to the final result. I believe it makes all the difference. All the same, it might not matter much for most people, especially those using Crowsons. I doubt very many bass systems out there are particularly balanced, including those with very high output capability. If one's sound already leans heavily in certain directions (such as ULF over mid-bass or vibration over acoustic) then the nuances of better quality BEQ quality may be mostly missed. The ability to send a lot more content to the Crowsons may be "good enough" for most people. P.S. I expect to evaluate @maxmercy's "Ready Player One" BEQ hopefully this weekend. It looks like one I will enjoy. :)
  11. 1 point
    git clone git@github.com:3ll3d00d/beqdesigner.git cd beqdesigner git checkout 0.0.2-beta.2 python3 -m venv beq . beq/bin/activate pip install numpy colorcet scipy qtpy qtawesome pyqt5 matplotlib ffmpeg-python soundfile resampy cd src/main/python # you also need ffmpeg and libsndfile1 installed, e.g. sudo apt install ffmpeg libsndfile1 then open mpl.py in a text editor and change the following change for k, v in cc.cm_n.items(): to for k, v in cc.cm.items(): (this is due to some lib having an older version in pypi vs conda, it is fixed in next release) then python3 app.py and you should find it fires up I'll get round to packaging it properly for linux soon enough next release will properly support pre BM BEQ btw, will publish that tomorrow hopefully I run Debian Testing here and it seems ok, haven't tested it extensively though so let me know if problems, will get round to testing it properly on linux at some point. The same approach has also been used on the mac too btw.
  12. 1 point
    Room response matters far more than sub response. We're not listening to the cabinets out in a field or in an anechoic chamber, we're listening in a room. I know very little about this aspect of acoustics, but I am working on learning about this. There are lots of ways to make bass, none are totally right, but none are really wrong, they're all just varied in their types and degrees of compromise. Certainly, multiple cabinets bring benefits, no matter the alignment selected. Last thing I am trying to do is oversell tapped horns, specifically my designs. I know the guy that designed them, and I know the grades he got when he was studying engineering.
  13. 1 point
    I get away with 125dB of bass on a nightly basis Bosso and Paul will attest, it is perfectly quiet outside my house at those levels so long as all the doors are shut.
  14. 1 point
    Thanks for chiming in, Brandon. From my experiences setting up systems for other people (read; calibrating flat) over the years, then popping back in to say 'hey' months later and seeing the subs bumped to +10dB hot, I truly believe that most enthusiasts would prefer the "peaky mess" you describe. It cracks me up how the AVS fora are clogged with experts comparing a 0.5dB increase in output with 3% less THD to 'prove' which sub is the mostest bestest and then post a (+/-)10dB FR with the sub at +10dB hot. That's simply comparing inaudible distortion vs output in dBSPL differences at ground plane outdoors to utterly gross, ridiculously audible distortion at the seats at home. Yes, it's a fine dance to get 8 discrete channels of 3 Hz-20k Hz digital audio properly calibrated and integrated with multiple seating positions in a relatively very small venue to a (+/-) 3dB response. I always thought that how to do that would be the crux of the discussions vs " my 'x' sub hits 3dB harder than your 'y' sub at 2M GP". It's cool to blast away with whatever source and clock it for dB drag stats and talk about how it cracked the ceramic in your neighbor's shower stall, really, it is. But, everyone should first be able to brag 3-120 Hz at reference level with (+/-) 3dB response as their reference system before engaging in those grossly distorted exercises. I keep going back to the revelation I felt when seeing Keith Yates' Way Down Deep series in which he included digital spec lab caps vs mic'd spec lab caps using the actual scenes of actual soundtrack that we all were trying to reproduce in our HTs. Of course, they showed no room gain because he measured the mic'd version outdoors, but what a metric! I still think it's the best metric and certainly one that should have been expanded upon in the 11 years since.
  15. 1 point
    It can be done right though. With my nearfield, it is not level matched to the front, rather, I run it a good 4-5dB lower than the "main subs" so the response does not suffer. If run level, the response gets boosted from around 40hz through 80-100hz and you no longer have a nice smooth FR to the single digits, instead a peaky mess that basically rolls off at 6dB/oct down to where the nearfield's response is low enough that you get back to your native 3-4m response from your "main subs." This is no good, obviously. Two ways to combat this is either run the nearfield low enough that it just simply adds to the experience, but doesn't affect the nice FR that you had previously with traditionally placed subs, or Cross the nearfield low, like, 30-40hz low, and use it simply as an air pump to increase your tactile sensation. I have to say, done right, it has really added the sensation I was looking for, whilst not affecting the FR all too much, it certainly is a game of finesse to get it all right though.... You talking about Pop's SI's? Oh they definitely added to the experience. Of course, his little alcove where he had his setup in his open basement basically made it like a 6th order bandpass box....and you were sitting in the back chamber!!!!
  16. 1 point
    The A-Team (5.1 DTS-HD MA) Level - 2 Stars (106.9dB composite) Extension - 5 Stars (1Hz) Dynamics - 4 Stars (26.04dB) Execution - 3 Stars (by poll) Overall - 3.5 Stars Recommendation - Buy (by poll) PvA: