Kvalsvoll

Kodi uses alsa on my machine, and this dsp can use alsa. But it will not work for movies, because audio is sent to hdmi directly as passthrough, for external decode in the av processor. It would be possible to configure to allow software processing if the movie audio is re-encoded to pcm, but then you still would need to extract, decode, encode and re-mux. Options here are severely limited due to the proprietary formats of audio in movies. Which leaves processing post-bm in the bass-system dsp is the best option for quick and reasonable quality bass-eq. This also eliminates problems with clipping.

There is no dsp available on the computer. But that is possible to fix, this is a linux machine running ubuntu and kodi, it would be difficult to implement at the time it was built, but there may be options available now. If filtering is done real-time during playback there is no need to do any processing on the soundtrack. This does however require some sort of analysis to ensure no clipping.

@SME, the graph files would be used by those who do not have the .mkv file, and thus have no intention of doing eq on individual channels. Or by someone like me, when I just want to watch a movie using the bass-system dsp to do a simpler bass-eq, not wanting to spend hours to do the remaster process. For this, the bass-managed graph will do just fine, and when applying proper eq in dsp on the bass-system there will not be problems with internal clipping. @3ll3d00d, ffmpeg can do the necessary signal processing, filters are available. I just took a brief look into what is in there, and I believe ffmpeg also can handle the remux. The process of applying the filters to the .mkv should be automated, so that you run this as a command after creating all filters and adjustments. This will take some time, because all channels must be extracted, processed and then put together into one new track, at full sample rate, which is then at last remuxed into the original .mkv so that you end up with an additional bass-eq track. Using flac encoding is a good idea here, saves space and bandwidth. This can be done.

@3ll3d00d, signal processing and remux would require some effort. First, all tracks must be extracted at original sample rate, then apply signal processing for the filters, and finish by muxing the new beq remaster track together with the original .mkv. Extract and remux is not difficult, you already use ffmpeg to extract, perhaps look at mkvtoolnix for the remux. Signal processing I am not sure of, would start to look at ffmpeg to see what functionality is available, also search for open-source projects. Graph files of movies would be very beneficial for most users, as they will not have access to the movie as a .mkv file, but could still use the beqdesigner if graph files are available. This also prevent problems with copyright issues, which would be an issue if it was necessary to make the sound track file available. Lots of movies have already been analyzed, and if the graphs were accessible in some kind of library, people could find them there, and use beqdesigner with no need to extract audio files.

One advice on bass-eq: Bass-eq is not about correcting the frequency response of a movie sound track so that the spectral distribution becomes flat. Bass-eq is about lifting up the filtered ulf so that the level is restored back to what it was before filtering. This can only be done by guessing how the spectrum should look like, and apply appropriate filters. There will be peaks and dips and deviations from a flat curve, depends on what sounds are present in the sound track, and how they are made. So a bass-eq correction should not look like the eq one could do on a bass-system, where peaks are cut down and frequency ranges with too high or too low level is corrected. Though on some movies there can be improvements by cutting a few dB in the 30-40hz range, and to decide what works best, you have to listen to it and correct accordingly. ( @SME also speaks about this in his post above.)

@SME, the interest for bass-eq is very, very low considering the huge impact on sound experience. One reason for that is that very few people actually has a sound system capable of reproducing full range with decent capacity and quality. One purpose of this thread was to create awareness, so that the producers deliver better, unfiltered sound in their movies. No bass-eq necessary. Only a few movies deliver that, even today it is rare to see a new release with full frequency range intact. The reason why is that they are clueless - they have no idea that anything is missing in their sound track. No producer would allow a filtered movie to be released if they knew their product could deliver a much better and more involving experience. Most people play the bd and have no option to change the sound track, the only option is to use eq in a dsp somewhere in the chain, and since all decent bass systems has a dsp, it is convenient to implement bass-eq on the bass-system. The drawback is that you can not do individual filtering of the channels. But compared to no bass-eq, this is a good and very useable alternative. I prefer to do individual corrections, because it is better, @maxmercy is very clear on this, he is the true bass-eq purist among us.

Unchecked the mono-button, and it works, great. Still unable to Add signal, but that is not really necessary now. Some cosmetics - graph display gets corrupted by legend, too few colors on graph signals. Workflow - how you use the app - is very important, so that users easily understand how to use it. Now you must know that you start with Tools - Extract. When you know that, the rest is quite intuitive. Functionality - nest step could be to implement signal processing to apply the filters to the sound track. Then re-mux the beq filtered back into the .mkv. Most people will use hardware dsp on the bass system, so this would be of no use. And most people do not have access to the movie as a file, which effectively prevents them from using this app in the first place. This is something that this app obviously can not solve.

Seems to work now, but I do not see the individual 8 signals. Extract Audio, then Create Signals, and I get a graph with peak and avg. I can then create filters, and the graph shows filtered response. As it is now, this is a great tool for creating beq for dsp. And it helps creating a remastered sound track because it is much faster and easier to find the right filters.

Does not run on my system, so I test it on windows instead, no problem. Focus on improving functionality of the beqdesigner instead of trying to support running on a system that very few actually has. It is not that the app can not run on ubuntu linux, it has to do with the effort that is needed to make it work. Now, there is a problem with my windows app - add signal does not work, the graph does not show all signals, and the last release does not create the xys_fl, .., signals. Extract audio works and creates a .wav, but this wav can not be loaded afterwards.

Excellent, @3ll3d00d, will try this now.

Does not run on linux - on my system, for now. Which means there is some work to get it up and running, perhaps I will look into it later. Because the idea of this app is great, it can actually end up as a completely automated bass-eq for any movie at click of a button.

This is excellent, of course I need to test it. But before I waste my time - does it work on linux, should work if ffmpeg is the engine for signal processing, and the ui is python/qt. The media computer is linux, it is just more convenient to use that one.

I agree about both the sound and the movie.

Alien - Covenant: Anyone measured this? Could not find any graph. Just watched it, sounded quite good to me, did not feel like much was missing in the low end. In my opinion, it does not need BEQ. Sound aside, it is note a very optimistic film. Would not recommend it as a good way to pass time while on your first journey to March. If you do, since such a trip takes quite some time, you can watch the whole Alien series, then top it off with Prometheus and Covenant.

It is the same. Most sci-fi films use some kind of alternative set of laws for physics. Allows for sound in space, fast travel - without all this, it would not be entertaining. We accept that, and enjoy the adventure in a fiction world.

Here on earth, and elsewhere in real life, physics do actually follow those generally accepted rules. Big objects no longer supported fall down, and in the process makes a lot of sound, and the bigger the object, the more low frequency energy it creates. Because there is gravity, and there is air. Like when this glacier brakes loose a quite significant piece of ice (see below for appropriate BEQ): This clip has decent sound, only 2 channels obviously, and it rolls of around 30hz, most likely due to the mic+recorder. A BEQ something like this restores enough to get an idea of what this was: sfm 22hz q=1.4 gain=+12dB sfm 14hz q=1.4 gain=+8dB If you expand to 5.1 you get more headroom and can retrieve even more below 10hz, by moving the low frequencies to LFE.

Now I have watched half of it, and the impression of the sound is even better than the first brief screenings. ULF is well done, and it sounds much cleaner and less distorted than other movies with clipping and heavy limiter use. I used to believe that a spacecraft that runs out of fuel will just continue at same, constant speed in same direction, but you always learn something new, from watching this movie it is obvious they loose speed and eventually go to a halt.

Testing new configuration in Room2. V6030 compact horns placed in the middle of the room, on sides. Quite close to the lp, nearfield-horn?

Same as the Marantz and Denon I tested, it is very likely they share the same processing. Note that it is the DIFFERENCE (17.5ms/20ft) that is interesting, as this is the number that defines how much delay is possible on the closer speaker to make it match the farther.

Maximum speaker delay in processors/receivers - a critical property, which is usually not sufficiently described in the manual or product presentation. Anyone know the limits for different types, brands? I seem to remember this issue has been up before, but oooohhh.. using the rest of the day searching will not happen, and new information and new models may be available. The problem: Getting the timing correct is crucial for high performance sound quality. For systems with front main speakers and separate bass system that means to delay the main speakers so that they sum correctly with the bass system in the frequency response AND IN TIME/PHASE. On most processors this is done by setting a distance on the different speakers. Typically, you set the front to the measured physical distance, and end up adding several meters for the bass system (subwoofer) - THIS WILL DELAY THE FRONT SPEAKERS. If you know what you are doing, you set the delay using measurements, so that timing and phase gets as good as possible to achieve. If you read on-line guides and audiophile magazines and pay no attention to how things really work, you set the distance for the bass-system equal to the physical measured distance, and conclude that subwoofers always sound kind of sluggish and is best switched off for music. Since you are smart, you want to do it the way that actually works to get better sound, and end up seeing that the distance entered can be quite huge. And in some cases it may be possible to reach limitations of the processor in use. Obviously this is a no-go limitation for a processor, so if you have an installation that you know will require large delays, you want to choose a processor that satisfies this requirement. You want to see a specification for this number. But this number is not in the brocheur or manual, it is not in any "test" performed by on-line or paper magazines - because the don't even understand why this number is important - so the only way to know is if someone have found the data. My contribution: Denon/Marantz processors, AVR: Max distance difference 6m / equal max delay 18ms. Devialet amplifiers: 20ms. Hypex DLCP and my SA-700 amplifiers: 15ms. (Though not relevant on the sub amp, becuase it is the mains that need delay.) Onkyo processors, AVR: ??? Some readers now realize I need those numbers for the Onkyo.

Speaker tuning with radiation pattern, on-axis response, power response and then add in room acoustics - which will be more or less an unknown parameter for a speaker designer. This is difficult and complex, and has huge impact on perceived sound - in contrast to amplifiers, dac's, all the nonsense products. I also believe that this field has not yet been fully discovered, there is still more to learn and find out. Experiments focusing on how perception of sound relates to the technical parameters are key factors for improvement. Horn speakers sound different from trad-hifi partly due to fundamental differences in radiation pattern. It is impossible to make them sound equal, because tonal balance depends on what sound is being reproduced. If you tune for flat and equal steady-state, the transients will sound different because decay profiles are different.

@SME, you are aware that loudspeaker manufacturers tune their speakers ("voicing") according to power response and estimated typical room acoustic properties, and have done so for decades? If you look at on-axis measurements of typical hifi-speakers, they often have a response that deviates considerably from flat - look up some of the well regarded speakers on stereophile. This is because the radiation pattern changes with frequency, and by adjusting the response the "sound" can be changed into more balanced and neutral, but the measured charts seemingly show a "defect" speaker. Since most speakers have a pattern omni at low f and then narrowing towards higher f, they will radiate more power at lower frequencies. This causes some problems if you design a speaker with radically different pattern - a speaker with flat fr and flat power will sound different from these typical once, and since music is mastered for use with the typical speaker and room, it is necessary to alter the reposne of the flat speaker into something that gives a perceived tonal balance more similar to those typical ones.

Compact Horn, basically a rear loaded horn with resonance chamber for extended usable range.

Testing midbass horns in Room2. Calibration is complicated, and benefits in this system and room are questionable. This is just to ensure they work, and to learn how to set up and calibrate the system, so that the customer can receive some useful guidelines. Usable range around 45hz up to 200-300hz, but for cf above 150hz you should absolutely use stereo processing, so for a typical av-processor system with bass-management it can be used up to 150hz. Capacity around 120dB+, depends on how much power you have.

Huge improvement (Star Wars - The Last Jedi). It already sounds quite good as-is, due to nice dynamics and the fact that levels are not pushed beyond clipping. But Bass-EQ improves this in a dramatic way. And you can increase mv a bit, I tried +3dB now, and that still works very well. Effects now have full frequency range, and anyone who experiences this difference will not need any abx-blind-testing to hear and FEEL this improvement. This is not about shaking the house more, it is about lifting the experience with improved sound quality. Mid-bass level is quite moderate, so a little lift from 50hz and up may give more impact on transients. This is exactly what sound quality is about - excitement without fatigue.