On Mon, 18 Aug 2014, Fons Adriaensen wrote: I thought this was quite exciting, the idea of being able to use a linear input and map it directly to DB. However, Mixers don't look like that, the area from plus 10 to -10 takes up a whole lot more room on the fader travel than anything lower than that. Here is a picture that shows that: http://www.allen-heath.com/media/GLD-Faders_2800.jpg As you say 0 is a special case for off. everything below -10 is the same amount of travel for 10db. -10 to -5 and +5 to +10 are the same travel for 5db that the lower parts are for 10db. -5 to +5 uses more travel again. I think they use a lookup table rather than use any formula. I thought this next (analog fader) picture was odd: http://www.allen-heath.com/media/GL2400-4-24_Master-Section_2800.jpg -20 to -30 uses a lot more travel than -10 to -20... -30 to off is much compressed. I guess that is just the taper on the fader. It appears that faders do not have linear or log taper, but rather some custom taper that expands the -10 to +10 area for better usability. My Mackie CR 1604 (yes one of the first ones) with 60mm faders dodges the whole thing and puts unity in the middle with +20 on top and off on the bottom. The faders are probably a log faders. So I was wondering if there would be a difference in faders built for FOH and recording, this image of a Classic Neve 8068 Console from the 70s (or earlier?) shows the same kind of mapping though: http://lghttp.17114.nexcesscdn.net/808784/vking/media/catalog/product/u/s/u… Yes, I liked the idea of that: real easy to build. Probably fine for static mixing too. While many people "draw" the fade on the DAW waveform (or let the DAW do that for them), what about those who like to do that manually? Would that mapping make it harder to fade? Or to put it another way, The manual fade would end up being different than with an analog recording desk. If the artist is "drawing" the fade, why use faders at all? Rotary encoders would work fine and could be set to arbitray resolution on the fly with modifier buttons/keys (The control surfaces I looked at do this). In fact, rotary encoders could be better for mixing with a more positive feel for one "tick" if they are detented. (maybe use bigger knobs like in days of old) Does anyone have any ideas on what wrist movement is more natural, less strain? Would it be better to use an up and a down key? (maybe with up 10 and down 10 keys as well) I guess what I am saying is that todays control surfaces are based on operating methods from the 60s and 70s. Have operating methods changed in such a way that control surfaces should be different? Should operating methods change because of todays tools? Can a manual fade be done in a more musical way than a drawn fade? (thinking about the ear to brain feedback vs. repeat and tweak) -- Len Ovens www.ovenwerks.net

On Wed, 2014-08-20 at 07:37 -0700, Len Ovens wrote: "Audio taper The most used non-linear taper is the logarithmic (log) or audio taper. This is mainly used for audio volume control, to obtain a more natural ‘linear’ perception in sound intensity change when you adjust the volume. Because the human ear is sensitive to sound intensity in a logarithmic fashion, at low sound intensities a small change in intensity is perceived as a big change in loudness, while at high intensities a large change is required for the same change in perceived loudness. To compensate for the ears logarithmic behavior, audio taper pots were developed. While it is called logarithmic, it is actually an exponential curve (the opposite of the logarithmic behavior of the human ear). Sometimes inverse logarithmic (anti-log) pots are used, for example in audio controls which turn counterclockwise, but also in some other specialized applications. The dashed lines in the graph below show the ‘real’ logarithmic curves. In practice logarithmic types which are used for audio applications do not really behave in a correct exponential way, but follow the curve stepwise."- http://www.resistorguide.com/potentiometer-taper/ "P&G [snip] have a real Logarithmic law.... (Alps is a “Audio Taper”)" - http://www.gearslutz.com/board/3027091-post6.html

On Thu, 2014-08-21 at 09:14 +0000, Fons Adriaensen wrote: That is a topic for me on several mailing lists since over a week now. Software mixers provided for Linux user space often are bad designed, so beginners who never have used a real analog mixing console hardly have a chance even to learn by trail and error. IMO each channel by default should provide your (Fons') parametric EQ and post fader aux sends. If people start mixer automation for the fist time, a notification should pop up and mention, that: Flying faders in professional music studios usually are used to restore sessions and to make funny videos for YouTube, but during audio productions there seldom is the need to fade. Regards, Ralf PS: It's not completely OT ;), even a remote control for a virtual mixer is used in the same way as old analog gear. For radio the fade ins and outs could be done by each kind of sane taper/mapping. Fons already pointed out, it's like driving different cars.

On Thu, 21 Aug 2014, John Rigg wrote: Not having anyting other than level certainly ecourages one to record the right sound. It has with me. Ok, I have to ask this somewhere, it may as well be here :) Which log? I would think .5 db for the same amount of movement the entire fader length would be logarithmic (log 10). A linear fader would be +6 at top, 0 in the middle and infinite at the bottom (assuming a 0 to 2 range as Ardour uses). The standard fader seems to be half way in the middle... logarithmic with a linear component. It is easy to see why we do not use linear, and a log 10 would never get to ininity, it would need a special step at -100 or so to jump there. In fact a log pot would have to have an extra resistor at the bottom to keep it from ever turning completely off. The audio taper seems to be made to give good mixing resolution in the mixing range while still fading to 0 during a fade. -- Len Ovens www.ovenwerks.net

On Thu, Aug 21, 2014 at 09:01:03AM -0700, Len Ovens wrote: The P+G faders (generally regarded as the best) are close to log10 characteristic, ie. -20dB at about half travel. There's a PDF data sheet here: http://www.pennyandgiles.com/Products/Audio-Faders-Video-Controllers/Linear… The Panel Graduations/Slots diagrams on page 3 show the taper characteristic. John

On Thu, 2014-08-21 at 20:06 +0100, John Rigg wrote: Actually you get fader units for Studer with P&G, but also with Alps faders. Such a module usually costs more than a complete home recording mixer. Small semi-pro studios more often use something like a Soundcraft Delta with Alps faders. The long and the short of it, more people, in particular the amateurs and semi-pros tend to use gear with Alps faders. P+G are three times more expensive than Alps. Alps audio taper is very good. The only bad Alps faders I ever used (and still use) are seemingly substandard goods used by a Behringer mixer, this mixer cost around the price some Studer fader units do cost, but even those bad Alps faders just feel not good when fading, the Audio taper is ok. IOW regarding to the taper it doesn't matter if you're using P&G or Alps. All those faders use a similar kind of log.

On Thu, Aug 21, 2014 at 08:06:13PM +0100, John Rigg wrote: The taper I'm using in some apps is very similar: <http://kokkinizita.linuxaudio.org/linuxaudio/faders.png> A position vs dB plot of this consists of two parts: the top one (down to -20 dB in this case) is linear, the bottom half is parabolic. The two curves are tangent at the intersection point, so there is no 'jump' in the dB/distance ratio. Mapping functions in both directions are very simple. Ciao, -- FA A world of exhaustive, reliable metadata would be an utopia. It's also a pipe-dream, founded on self-delusion, nerd hubris and hysterically inflated market opportunities. (Cory Doctorow)

On Thu, 2014-08-21 at 23:20 +0200, Ralf Mardorf wrote: ^^^^^ reverb, this should be reverb and not dely :D

On Fri, 22 Aug 2014 09:29:30 +0200, John Rigg &lt;ladev8(a)jrigg.co.uk&gt; wrote: Hi John, you claimed that you know how to record right in the first place, so that you don't need EQs for the mixing, that's why you claimed that EQs by default for each mixer channel could be a nuisance. You might have the gift to know how the frequencies will interact for the mix, already when doing the mix, but that is a very unusual gift. It's common for good audio engineering that EQs and faders are the basic tools for mixing. I confirmed Fons' piece of information that fades are rare when mixing music and the fades you are doing are those rare fades, fates that btw. I'm doing too, at least I do similar fades. The quality of mixes nowadays often suffer from bad engineering, because as Fons' pointed out, people tend to add too much effects to a channel, I added that one reasons is, that using EQs as a basic tool for a transparent mix and to control the dynamic range of a song, nowadays often is replaced by using dynamic effects and much usage of fader automation. I'm not dictating anything, the nature of the interaction of audio signals from several channels, when mixing them to a mono or stereo sum dictates us the way we most of the times should do a mix, to get a transparent acoustic pattern, with an acceptable dynamic range. Dynamic effects, such as a compressor are ok, they are needed, but to use them, EQs are needed too and sometimes EQs can replace usage of a compressor, at least for the individual channels. You get me wrong, I'm not implying idiocy. My intention is to point out, that young people don't have a chance to learn by trail and error, regarding to the odd design of many Linux DAW mixers. Qtractor is one piece of software I very often use. Many young people using Linux are using it too. It's mixer e.g. doesn't provide post fader aux sends for audio, resp. for MIDI the faders are post aux sends, but if you insert an effect, it's also post fader and will add noise, even if you use the channel at a low volume, IOW the odd design of this mixer and some other mixers too, leads to learning how to mix in a wrong way. Learning by trail and error only works, when the mixer has got a sane design, as the mixers had, when you and I learned audio engineering > 30 years ago. Regards, Ralf

On Fri, Aug 22, 2014 at 12:32:19PM +0200, Ralf Mardorf wrote: I'd agree with that (while not commenting on the rest). Regarding EQ pre or post 'tape', that is for a part a matter or personal preference, but also related to history. At the start of the multitrack era, many engineers were perfectly capable of mixing the type of music they usually handled 'live'. They didn't need a hundred trials to arrive at a good mix. And with only 16 or 24 tracks available, you often had to pre-mix things before tape. Using EQ at that point was a natural thing to do. The typical structure of the mixers used at the time also plays a role. Desks used for multitrack production were in fact two mixers in one box. In the first generation, you had a 'main' mixer with everything on it, EQ, inserts, large faders, etc. While recording tracks, this was used to mix the signals going to tape, on 16 or 24 mixing busses. A separate 'monitoring' mixer, usually a lot simpler, no EQ or just a very simple one, was used to create a control room monitor mix of the tape signals. Aux busses were shared between the two, so you could create a monitor mix for a musician using both the mic signals and those already on tape. For mixing you would switch the main mixer inputs to the tape outputs and mix them on a stereo bus, again using all facilities provided. These desks tended to be quite large, since the monitor mixer was a separate section, usually to the right of the main one. Using EQ while recording was a natural thing to do when using such a mixer, simply because it was available, and for the reasons already mentioned above. Then came in-line mixers. In-line means that one channel from the main mixer and one from the monitoring one were combined into a single strip. So each strip had two faders, the second normally being a small one. One advantage was much reduced size. At the same time the signal routing was made more flexible, to the point that the two signal paths within a strip could swap roles. In other words, even while recording tracks you could opt to use the 'simple' mixer to do that and the 'main' one for control room monitoring. The result was that at the end of a session you could have the main mixer completely set up for the mix, including post-tape EQ and any effects, without having to change anything. And on some mixers (more as time went on) you could save that setup and recall it later, so this was an interesting way to work. All the large mixers I ever used (Neve, Harrison, SSL) were in-line, and I usually preferred to use them as explained above. It was a personal thing, some other engineers at the place where I worked did the same, and some others never did unless they had to provide a live mix for broadcasting while recording multitrack at the same time. A DAW such as Ardour has remnants of the in-line structure. Logically there are still two signal paths in each track strip. But the 'pre-tape' part, instead of being a mixer, has been reduced to the mininum: no gain controls and fixed one-to-one connections from inputs to tracks. The only thing that remains is that you can measure the input signal, and use it as input to the main path while transport is stopped or the track is recording. With unlimited tracks available, this makes sense of course, though it encourages postponing everything to the mixing stage. I'm not convinced that is always a good idea, you can easily end up with 'too much to handle' there. Ciao, -- FA A world of exhaustive, reliable metadata would be an utopia. It's also a pipe-dream, founded on self-delusion, nerd hubris and hysterically inflated market opportunities. (Cory Doctorow)

On Fri, 2014-08-22 at 12:05 +0000, Fons Adriaensen wrote: PS: I'm especially thinking about the band in the box approach, For synth plugins I mix already before recording and for sure, others who record a band could use EQs before the recording too. At what time you need the EQs doesn't matter, you need EQs for mixing music. Is there really the need to include EQs only when they are needed and to exclude them, when they aren't used? What exactly is the nuisance, if EQs are always included/available, even when they aren't needed? Horsepower of the computer? From cheap to expensive analog mixers, all provide EQs, you can't get rid of them, when you don't need them. There is no reason to get rid of them. Horsepower of a DAW computer vs searching for a good working EQ, learn that EQs usually are needed etc. ... to make EQs optional comes with more drawbacks than advantages. There's nothing wrong with learning by trail and error, but newbies don't become a chance to do this, when the mixer design isn't good, when even basic tools aren't defaults.

On Fri, 22 Aug 2014, Fons Adriaensen wrote: Partly also for historical reasons, I think. In many ways digital recording started as the "poor man's tape". Direct to disk recording with no effects was at first all that could be handled and most peole using it were replacing 8 track tape with it. They already had a mixer. As the DAW developed, mix down on the computer has been next. But for many people the recording part of the strip has been outside of the DAW, on an analog mixer. This is changing as a new batch of people are going mic-> interface. Their input strip is whatever the interface provides... often only trim (either as a pot on the pre or in ALSA). So digital recording is also going through a two mixer to inline transition. From hybrid to digital only. The trim controls are there, where they should be, as close to incoming signal as possible. I don't suppose it would be too hard to add alsa trim for a card like the d1010 to ardour, but many USB IFs (even PCIe) have no controls in alsa. It is a physical pot somewhere. So rather than being in front of the engineer, it is hidden and easily missed by the newby... or even not so new. So much is done digitally, that the remaining analog items are forgotten. This is a real problem with a two input IF, The trim needs to be set every time and the variety of signals through one channel is huge. Everything from a ribbon to line level. Having a set of good pre amps could be worth while, this is probably the biggest hole in the hobby studio. I have two, tube and solid state. (plus line) unlimited? CPU speed does that even if sw doesn't. As in days of old, buget determines tracks and effects available, though the price per track has dropped dramatically. Mixbus (Ardour with input strips) is known as cpu heavy, because of where the average home/small studio budget is. I have a 4 core i5... about $500. The next step up is a xeon 8 core... two of them on the same board with separate memory. multi hard drives.... 6x the cost pretty easy, but not bad in a paying studio. The reality is, a better interface comes first... and mics... the dual xeon starts to look cheap by comparison. grouping and subs. -- Len Ovens www.ovenwerks.net

On Thu, 21 Aug 2014, Fons Adriaensen wrote: On a 64bit machine, 16 memory locations (128 bytes) would hold a tick by tick map of values. A char[128] lookup table would be very fast. However, I am wondering how accurate the marks next to a fader are meant to be and if such a non-linear map makes sense anyway. If our steps are .5db in the end anyway, then then a "linear" (well straight log) surface from top to bottom seems reasonable in that the same travel always gives the same change. In any case, that is where I am going to start on the hardware side is a linear .5db per 1/128 movement. However, because the middleware is effectively part of my project, I will have to do a conversion to 10bit at some point because I am trying to emulate the Mackie control surface. That is I want to do something that uses some kind of standard rather creating a new something. I am going to suggest that the markings on a surface are guides or aproximations and the user should be using their ears anyway. Because I am building my project around Ardour (it's what I use and have for easy reference), I have already said that my control movement will not mirror the GUI, but if I want it to be true to the .5db per tick, I do need to use the same calc they use for the actual gain. There are no markings on the GUI fader, just the db at the top. Center of the fader is -10.5 (pitchbend 0) and each step is 16 because pitchbend is 14 bit and mackie control is 10 bit. Top of fade is +6 and bottom is off. Hmm, interesting, one step up is -109.1db... going back down one step gives -182.5db even though the pitchbend out is the same number.. but the surface going down one does go off. Not that either one matters as the first value up will be higher at -58db, so 45 ticks up from there (2d) is the same as 1 for my surface. My 2 is translated to 2f for -57.5. Starting from the top at +6, I have to go down 3 mackie ticks just to get to +5.9. To get to my first value of +5.5 is 20 ticks, another 20 to +5.0... It seems a lot of the 1024 available values (and bandwidth) are wasted in the mackie control surface. I can't see anyone being able to hear the difference from 6.0 to 6.96, a part of the fader no one uses at that. My middle range (value 64) is -26db, in mackie speak it is 4016 (looking at the whole 14 bits) or 1/4 range. At this point I have to do some thinking because this matches an analog fader too. Why did the analog fader have such fine control in the -10 to +10 window? I am thinking that at least part of the reason has to do with mechanical play/bend/lag. By the time the user has moved the fader the smallest amount, the fader knob post has bent slightly and the wiper has "jumped" after it has overcome any stickyness rather than making any .1db movement. Good I must be doing things right. The only fades I have used are fade out at end of song and I have drawn that in. Often, to make it sound right, I have to fade the lead (louder) tracks faster than the backing tracks anyway. (they may be 0 as much as 2 sec before the end) (or less) fade would serve the purpose better than a fader. Dead air was a dreaded thing so the next part of the program was already there before the fade started and the fade was mostly not to have a harsh cut. adjustments rather than a long fade. My only question becomes: will a jump from -58 to off be noticable? This would be 64 db down on a digital signal that has a dynamic range of ~109db depending on the analog HW going out. Background noise in the listening area will bring this lower. I think it depends on what the main purpose is. If it is setting levels in a mix, then supporting long slow fades does not make sense... or to put it another way, using the same control for both mixing and fading is maybe not the best thing. The standard fader mapping is because of this joint use. I think the +6 to -58 db range is probably great for mixing and the -58 to off will be smoothed by SW anyway even in a fade use. I got the idea, the use of plugins was because (at the time) the HW could not handle a full input strip like what Mixbus has for example. SO the user could just add what was needed for a channel. Certainly adding individual reverb per track does not make sense as having the whole "band" in one sound space would be the normal target. I am thinking that with the more modern HW an expanded default channel strip with eq and minimal sends and trim might again make sense. -- Len Ovens www.ovenwerks.net

On Fri, Aug 22, 2014 at 08:11:39AM -0700, Len Ovens wrote: You have two mappings: * Position (mm or degrees) to MIDI (0..127) by the control surface, * MIDI (0..127) to gain (dB), or to whatever is being controlled, by the DAW or instrument. Position to gain (or anything else), in other words what is experienced bu the user, results from the combination of those two. This also means that the controller can't have any calibration marks unless it knowns the mapping used by the receiver. This is always true, regardless of the actual mappings used. Having a linear mapping from postion or angle to 0..127 is in almost all cases the only sensible thing to do. The accuracy with which a user can set or move a fader or pot does not depend on its position. Hence the resolution should not depend on it. Such a linear mapping is very probably also what is assumed by the receiver, which will have its mapping defined such that in combination with the one from the controller you get the desired result, e.g. a good audio taper for a fader. For a fader you want small steps (0.5 dB or less) in the upper region. At low gains larger ones (1 dB or so down to -60 and even bigger ones below that) are perfectly OK, and allow to extend the range and/or to have smaller steps in the upper region. What would feel very strange to a user is to have bigger _physical_ (mm or degrees) steps in any part of the range. Hence the linear mapping to 0..127. There may be exceptions for some very specific controllers (e.g. thrust levers in an airplane), but an audio fader is not one of them. Ciao, -- FA A world of exhaustive, reliable metadata would be an utopia. It's also a pipe-dream, founded on self-delusion, nerd hubris and hysterically inflated market opportunities. (Cory Doctorow)