You just add the sample values. Nothing special needed. If you exceed 1.0, you can just clip when you convert the final result to integer samples, or divide every sample in the final result so that the highest peak is 1.0.
TL;DR Of course this is the simplest algorithm and if you're a newbie, stick with that.
<IF YOU'RE NOT AUDIO FANATIC, DON'T READ THAT CRAZY NERDY STUFF> However, what if signals are out-of-phase? In the worst-case scenario (two inverted signals, as in attached picture) waveforms cancel each other and after down-mixing you'll hear literally NOTHING! It's intended while doing technical stuff, for example in a nulling test - if you want to check whether streams are the same, invert one of them and if you get silence, they're the same. However, if you're doing musical stuff, that effect is unwanted and this is why downmixed-to-mono music sometimes sounds a bit differently than in stereo - some frequencies are out-of-phase and they change their power after downmixing because of interference. If you want to do it really-hyper-super-good quality way, dive into advanced programming and do FFT - change the signal into frequencies creating it. After FFT simply average resulting coefficients and then do Inverse FFT - create a signal from the information about its frequencies. For example: let's see at the picture again. With normal downmixing they cancel each other, yeah? But let's FFT both signals. They both have the same frequency and the same amplitude, so FFT will show: FREQ1 blah-blah-blah, AMPL1 1.0, FREQ2 blah-blah-blah, AMPL2 1.0. FFT doesn't show info about the phase, it's ignored. Now because FREQ1 and FREQ2 are perfectly the same, wa can average the amplitudes. That's simple: average of 1.0 and 1.0 is 1.0 We get FREQ blah-blah-blah, AMPL 1.0. Then simply inverse FFT and voila, you get audibly perfect downmix - no single frequency lost its power.
It's a bit nerdy, isn't it? I told you not to read that if you're newbie
Last post by magicgoose -
If you care about the quality even in the slightest, then clipping is unacceptable. If clipping is unacceptable and you need an universal solution, then there's no other choice than to divide everything by the maximum possible sum of peak levels. (And if they are 1.0 for all streams, then it's simply the number of streams). Because this is the maximum gain which cannot clip under any circumstances. If you don't need an universal solution, then it's up to the user to choose the gain. This is quite simple IMO; you are making it more complicated.
3) If you are summing totally independent sources (e.g. multiple bands of frequencies that do not overlap), then the sqrt(N) rule comes closer to correct.
Nope it doesn't; actually the opposite. It's easily tested by summing 2 pure sine waves with different frequencies. This test shows that "multiple bands of frequencies that do not overlap" is not enough; it's also easy to show that it's not necessary too.
Last post by Fairy -
Thank you. I know placement is important, but I in fact have only one position that is realistic. It is a corner indeed. If I place it somewhere else my main speaker ends up in the corner and I think that is not the best position for a fullrange speaker to go.
The DB4S does have a calibration system. I hope this works good enough to overcome any room problems. Until now I have had little problems with my listening room, apart from a fibrating window above the door that needed some silicone sealant.
Last post by polemon -
OK, just going by the idiocy of the actual article:
It implies, because Lasers, must be more accurate: Well, no. Lasers have the potential to be more accurate for certain processes, but when it comes to cutting, surface and material impurities amplify inconsistencies with laser cutting. So that alone is kindof a red herring. Anyone in the medical field knows this.
So, when the LPs are supposed to be playable on existing turntables and cartridges, the limits of those are already in place, aren't they. To make the signal "30% louder", I'd have to have the pickup to be moved at a 30% higher amplitude, vertically laterally, depending on the channel. In my mind, my brain initially parsed the "HD" part in the article as "They're putting 5.1 channels into a vinyl" for some reason, made me actually curious how...
It seems the article is weirdly fond of it being a Swiss "invention" and the amount of money dropped on the entire endeavor, supposedly.
Also, they seem to very much like the sound of certain words: "3D topographic map", that's really just putting a gerber file into an extruder. Every 3D printer and milling machine does that.
It sounds awfully like a cost-cutting scheme to me. Instead of cutting the master on a lathe and then make a stamper out of that, they simply use a commercially available laser etching machine, and simply upload the gerber file or whatever on to that. Now that the hipsters re-discovering Vinyl from a couple years ago have gotten bored with it, it might be simply too expensive to keep creating vinyls "the old way", and companies are looking for a modern alternative. This article seems to be trying to sell this as an improvement, but I really believe it's just done out of necessity, since it's so much more quicker, easier with modern tools, and industrially easier to automate. It also requires less specialized machinery.
Here's an example: https://www.youtube.com/watch?v=-NmJG5abDV8 Note how little space that machine needs, and how it's not in a machine shop. Technically, you can put it next to a large office laser printer in a copy room, etc. totally wouldn't look out of place. Also, I have no reason to believe, that the laser would not be able to create stampers of superior quality to the conventional method, using a small enough power setting and enough time, and the correct material. Certain machine parts are made to much tighter tolerances these days using laser cutting and laser etching.
Sounds very much like vaporware to me, but what do I know, it might just be the most cost effective way to create stampers today. The narrative catering to no one but the mentally weak is a bit insulting, though. The fact they didn't even say what laser engraving machine they're using etc. speaks volumes about how this is nothing but a marketing bullshit article, selling ordinary things as the new gold.
The accuracy of the stampers, will depend on the quality of the laser engraver and the metal plate they're gonna be using, as well as things like settings for time, etc. Laser etching is an ablative process, the slower it's done, the more accurate it is. I can see how that might be a selector for future vinyl pressings, btw.
Last post by polemon -
Well, the document is authored by Argon Design Ltd, http://www.argondesign.com/about/ , they are in the hardware design business it seems. Unfortunately, they just say "might change significantly" on the spec page, and not really where things might change, and where things are likely to stay as they are.
Last post by jsdyson -
Obviously, in the worst case (as noted above), one needs to reduce the sum of the levels by the count of sources. However, there are some possible modifications of that rule if you have control of the result (and just want to avoid blowing out your ears before tuning the levels later on.) So here are the mathematical rules:
1) in the general case, you need to divide the level by the count of inputs. when doing this math, you need to be careful about underflow (esp if not properly dithered) and overflow during the actual summation.
2) if you are summing multiple sound sources with the same song, mostly uncorrelated -- then an approximation using sqrt(N) instead of N might still overflow, but can come fairly close. This will NOT work in the worst case, but gives you an idea of a starting point for each individual instance.
3) If you are summing totally independent sources (e.g. multiple bands of frequencies that do not overlap), then the sqrt(N) rule comes closer to correct.
I am NOT meaning to confuse the hard rule about dividing by 'N", but rather trying to show the subtle nature of the statistics involved.
AGAIN, the initial answer (divide by N) is perfectly accurate -- if you never can overflow (including during the math), then you need to divide each input by N before doing the summation. If you can 'overflow' during the summation (happen to temporarily be using floating point or larger range number), then you can do the summation and then the division (it is likely to maintain quality better if you can do so and avoid the overflow of the temporary value.) This also applies in the analog situation, where it is better to keep the signals as strong as you can for as long as you can -- however avoiding overflow (or the analog 'clipping') of the circuit in every part of the circuit.
Isn't it amazing about how complicated a simple 'putting together' of multiple signals can be? (It really isn't that complicated, but it is always a good idea to keep aware of what is going on in the circuit or software.) I guess that I just might be making it complicated, but really trying to help with 'thinking' and trying to help new thinkers practice their learning skills!!!
I haven't used it myself but I contacted a person that uses Dolby A and I Think he has contacted you. I really appreciate what you are doing even though I have not used your software since I do not have any Dolby A encoded material. I do look forward to try out your future compressor you talked about earlier. I also love that you software is available to Linux and hopefully your compressor can be a LV2 or/and VST plugin.
FIRSTLY: if there is more interest, I can make the compressor available soon -- just needs to be updated with better code from DolbyA and expander. If you want to hear the results of a relatively mild run of the compressor, listen to SOS-DAfull-finalized.mp3 on the repository: https://spaces.hightail.com/space/z3H68lAgmJ 0 The file SOS-DAfull-finalized.mp3 has been DolbyA decoded, expanded (to remove some of the old compression artifacts), and re-compressed/limited with the modern compressor at a fairly controlled level. If you are really, really interested, I can make a limited version of the compressor/limiter available for Windows64 in about a week -- but it would be command line oriented (lots of options/tuning to get that special sound), and as a side-bonus the expander is built-in also. The bad news is that heavy baggage runs a little slowly, and the compressor & expander cannot run realtime. However, on a relatively fast machine, the compressor running alone usually has NO trouble running in realtime at 96k. This MIGHT change as I move some of the filtering from the expander, so AS YOU CAN SEE, the code is still in flux.
REGARDING THE DOLBYA ENCODED STUFF: One thing -- I truly do not know what you know or even about what you know, so PLEASE forgive me if this might seem rude... Before 1yr ago, I would have also said that I had no DolbyA encoded material, yet a large amount of my music collection WERE DEFINITELY DolbyA encoded... For example, I had my ABBA collection for over 10yrs (really, really, long time), yet knowing what I know now, and listening to it, I kind of 'whack' the side of my own head now realizing that my ABBA collection IS DolbyA encoded. Here is an example of my UNINTENTIONAL DolbyA material collection (likely well over 1/2 of everything I own including): Petula Clark collection of 4 CDs, massive Carpenters collection, a copy of ABBA Gold, 3 Simon&Garfunkel albums, 2 Linda Ronstadt albums, Fleetwood Mac Rumors, Christopher Cross Album, massive Olivia Newton John collection, Herb Alpert Album, Herb Alpert & Brasil66 Album, Burt Bacharach set of two albums, Dionne Warwick Album, Chicago CD, Queen CD, Carly Simon CD, 2 Anne Murray albums, Bangles CD, Bananarama CD, Moody Blues CD, The Cars, Suzanne Vega, Paul McCartney & wings from HDtracks, Carpenters from HDtracks...
My point in listing these -- you (or someone) just might (more likely than not) REALLY have a DolbyA encoded album (not vinyl, but CD or electronic.) This has been a very common thing, because I have almost had to check my sanity when I realized this. I have gone back directly to the CD source and lo&behold a CD that I might have purchased over 5yrs ago or something I downloaded deep in the past has been left DolbyA encoded.
I did NOT have to carefully select the material that I put on my repository, because I have found that more likely than not that a given piece of material that I have TRULY, REALLY benefits from DolbyA decoding. I honestly cannot claim that every item that seems to be DolbyA encoded really is -- but I have some traceably DolbyA encoded material and it really doesn't sound much different in 'compressed' sound and HF emphasis from the material that I have given as examples.
The only reason why the material can be left DolbyA encoded is that the result isn't really all that fatal (just a bit too much HF empahsis, too much of a compressed sound, and more hiss than there should be on older recordings from older tape formulations and tape machines..)
Some of my collection initially sound quite nasty, but then turns into audiophlie grade upon decoding -- REALLY!!!
If you don't have anything that is DolbyA encoded, I understand... But if you have a large electronic or CD sourced collection (or a moderate sized one like mine), then there is at least 1/2 of the selections that benefit from DolbyA decoding. I'd estimate that at least 50% -75% of my collection is almost provably DolbyA encoded, and perhaps 10% is questionable either way (might be sidechain compressed because of excess HF loss but no expansion dynamics problems.) The rest of the 15% is NOT DolbyA encoded.
Wrt my compressor (and possibly expander project) -- my compressor project is essentially on hold for two reasons 1) percieved lack of interest, 2) I have learned a hell of a lot on the expander and psuedo-DolbyA project -- need to feed back info into the compressor project. Note my comment about the compressor project above.
Wrt my general purpose expander project -- it isn't really 'on hold', but has been in development on an ongoing basis during the psuedo-DolbyA project. Initially, the psuedo-DolbyA was part of the expander project, but I found that the pseudo-DolbyA design requirements to be so very specific, that I seperated out the high level portions of the psuedo-DolbyA. However, the vastly improved anti-intermodulation dynamics techniques that I have used to make the psuedo-DolbyA work so beautifully (clean, clear, no roughness) are being fed back primarily into the expander project and also secondarily the compressor project.
I already had pretty darned good dynamics classes (attack/decay/gain-riding/etc) in C++ for numerous purposes when starting the expander & DolbyA projects -- the compressor was the basis for all of the work. There was already A LOT of intermodulation mitigation and well controlled attack/release characteristics,. However, given my own sense of perfection, I have been striving for perfection in the DolbyA project -- FURTHER MORE STRONGLY controlling the mathematically unnecessary intermodulation effects -- thereby producing a DolbyA decoder that changes the character of the sound back to before the DolbyA encoding. The pseudo-DolbyA decoder has few of the typical artifacts of either SW compression/expansion or real HW compression/expansion. I think that the ONLY intermodulation distortions left over on the psuedo-DolbyA is greatly attenuated and the sidebands are limited to below about 50Hz wide on the LF bands, and 450Hz wide in the midrange bands and 1000Hz wide in the HF bands. Even that those very significant limits (probably 10dB-20dB down at those bandwidths), the technically unneeded modulation products are attenuated even for the nearby sidebands. I have successfully controlled the intermodulation without fatally affecting the COMPATIBLE attack/delay times (was quite a feat to do correctly.)
Of course, one cannot do gain control without creating sidebands, so I had to leave the NECESSARY sidebands intact so that gain control would happen (yea, I know that there are tricks to move the sideband power over to one side and to make the distortion less noticeable, but that is too much work.) For example, the compressor is built from a 3band compressor, an 8 band compressor, a 1 band last chance compressor, and a carefully crafted set of limiters (lots of stuff.) The expander has 48 degrees of freedom in its gain control, and the vastly simpler pseudo DolbyA needs to meet really specific design criteria. (BTW -- I am not exaggerating about the 48 degrees of freedom in the expander -- that is why the expander can provide 1:1.6 dB or higher expansion ratio without pumping.)
Oh well, I am really just trying to help, and anyone is willing to email me for access to the psuedo_DolbyA. The SW will work as long as the OS/HW supports the code. I am limiting the time for the offer for my own sanity, however :-).
Last post by Vittorio -
I don't know if this is the right forum to suggest new features...
1) Layout Switch What I would like to see in new foobar versions is a simple switch functionality where I can switch between different layouts. Currently I have the following workaround: - two tabs with different layouts - default layout = "foo_full" (maximized window) 1. select tab with second layout "foo_min" 2. double-click on window title to restore last size and position of the window Now, the problem is No. 2: windows remembers the last state of an application window. But what I'd like to have is a window with fixed size saved within the layout, which can only be changed in Layout Editor
2) Dynamic Artwork Another idea is to set an stub image folder instead of just a single image with Dynamic Artwork for online media and files without cover. Here is the concept: