In this thread I will describe my endeavors with designing my own three band audio processor aimed at AM hobbyists. This is a followup from my previous work on a very simple FM audio processor that was essentially a dual band limiter, low pass filter, and unfiltered clipper output utilizing high frequency pre-emphasis correction. I've learned a heck of a lot on this topic from 2012 when I designed that and I plan to share everything I know here to fellow pirates and part 15 operators along with anyone else who is curious. I will start this with a primer on audio processing in general and move on to the circuitry involved in solving these problems, and eventually my finished circuit which has come along nicely shared to the general public free to copy and use. This will be a multiple part thread and eventually converted to PDF. I do not claim to be a writer or professional engineer. This is purely out of the hobbyist spirit to guide others along with myself for learning and entertainment purposes only.
If you're only interested in the schematic please give me some time. I am writing this like a blog and need to start with the basics first so we all are on the same page. My project isn't even complete yet, but I have enough of it designed now that I feel comfortable giving this post a start. First will be some background. The second major part will be completion of the schematic with scope/spectrum images and possibly sound samples. Third major block of text and info will be of the final design before I put it in use at my own station
April 10, 2019
A primer on audio processing as Kage knows it from the analog realm and experience with modern digital processing:
I have spent a great deal of time in the not so distant past studying how FM broadcast audio processing chains work along with the mostly all in one solutions that many stations still use. There were a few things that stuck out to me with my previous FM design that I really want to tackle but for AM instead for two reasons..
-AM transmission is relatively simple and requires far less circuitry, we can design monophonic processing whereas stereo transmission often used on FM requires duplicate stages of processing and sound stage platform sharing to reduce "ping-pong" aural tilt.
-With AM we don't need to focus on protecting a stereo pilot, and pre-emphasis can be designed simpler given that our audio cutoff is far less than 15kHz for stereophonic broadcasts on FM.
One of the main things I wished to focus on after other people suggested the idea was multiband audio compression/limiting. This is especially important with AM since loudness is king given all the various things that interfere with the mode of modulation over FM. I've always been a fan of keeping FM audio clean and mostly unprocessed outside of basic brick wall limiting on the stereo sound stage to keep dynamics since this is where FM rivals in sound reproduction, whereas we know AM and especially the lower frequencies used for broadcast need to fight hard with modern electronic noise and lack of "capture effect" rewarded by FM.
There's one huge issue with multiband compression though, it sucks dynamics out and creates listener fatigue when overused. This means there must be even another stage before compression to cut back the pull-up of soft passages that contain tape hiss, record pops, 60hz hum, or other background noise in a studio. This is where downward audio expansion comes in handy which is the exact opposite of compression, often simplified in the cheapest processors as audio gates that work wide band on audio. I chose to go further with my design by expanding on the three audio bands independently much like the well known Dorrough DAP 310 or Texar Audio Prism which I was highly influenced by. The balance between these two major stages, or rather 6 (3 audio bands of operation for expansion to compression) make my project compare to the higher end units of other manufactures while remaining on a hobbyist price tag using common op amps circuits.
It gets worse.. What happens when you feed this multiband compander (expansion/compression) with random volume levels from the studio at the mixer? This causes large variances at how the circuitry will create overall sound presentation since both the compressor and expansion stages being split into various bands will suck up or back down frequency selective ranges causing great variance in program sound stage given nothing more than how far the DJ cranks up the fader so we need to introduce yet another stage!
This is where the *slow* automatic gain control (AGC) comes into play, and is the first stage of an audio processor, so that it keeps the overall program material at a steady level with molasses level changes so the rest of the compression/expansion circuitry can do its thing (while the DJ is drunk and twiddling studio mixer controls) as the later processing stages expect a semi-standard level of input, preferably preceding an adjustable manual "processing" control level which feeds the seemingly constant AGC output to the later stages where loudness can be contoured properly outside of what drunken party happens in the studio.
But wait there's more...
So far we have our studio mixer board running out to our hypothetical rack of AGC -> audio crossover (split audio bands for multiband) -> downward expansion circuits -> compression circuits.
At this point it's safe to combine the compressor outputs now that the bands were split and worked individually upon to increase overall loudness across music/voice spectrum. In my project I split the audio into three ranges, low (25hz - 250hz), medium (250hz - 3.5kHz), and high (3.5kHz - 10+kHz). These crossover points seem to be a good standard for AM broadcasting as it works on music bass, vocal range, and high pitches individually. I found 3dB crossover points were good enough much like the DAP 310 from the 1970s, it's enough to bring classic rock up to volume without audible pumping.
This however is still not enough to be useful for broadcast and clean loudness. Anyone who has worked with broadcast processing knows what comes next is even equally complex and there's a reason why this is done in the digital realm now using software like Stereo Tool.. We now have to add pre-emphasis to brighten up the AM broadcast because modern AM radios have built in de-emphasis. This is a steady rise in frequency from 1kHz to 10kHz, rising 10dB at 10kHz. It's even worse with FM rising to 17dB at 15kHz.
This leads to a serious audio engineering issue V.S. modulation maxima, we have to pass the huge increase of high frequency audio to the transmitter without over modulation and that shrill noise, but how? If we sweep the audio processor at this point with a sine wave from bass frequencies to the highest pitch frequencies we notice modulation climbs from acceptable to grossly over modulated at the highest frequencies to accommodate for 75uS pre-emphasis (50uS in Europe). Welcome to the next stage in the air chain, the audio clipper.
This is often done with hard diode clipping or a combination of soft/hard diode clipping, sometimes soft clipping early on before combining the compressor outputs. This simply sheers off the loud over modulation of increasing the equalization curve with pre-emphasis which works because most music and voice doesn't drive those high frequencies up too high anyways unless you broadcast electronic music, but we need to make sure not to go beyond <0% modulation where the transmitter carrier cuts off, and some known value of positive modulation. Oh and about that positive modulation, we can use higher positive modulation than negative on AM within reason to increase loudness. Legal licensed radio stations can use up to %125 positive modulation, and some %95 negative modulation before risking carrier pinch off which causes interference to stations next to ours because of channel splatter from the rapid cutoff of carrier. The clipper stage is where this is usually accomplished by DC unbalancing the audio before heading into diode clipping so that one side of the audio waveform is sheered off sooner than the other on peaks. To follow nice symmetry this all can be preceded by all-pass-filters to average out asymmetrical peeks before clipping. A topic for another time.
Sounds good so far, but we now just created terrible audio harmonics by using diode clipping. Any clipping of audio causes nasty square wave harmonics, even if we don't hear it because it's high frequency content, it now exists because the audio waveform is no longer pure. Running this audio at this point into the AM transmitters modulator will sound fine, no one will notice there's an issue and you'd think we now have what is needed to process our music broadcasting out to the public, it will sound loud and proud and this is exactly what we want, you're now doing everything the big guys do but.... unless you're the listener of a radio station adjacent to yours...
We're not done yet
What we've done with the clipper is widen up our audio bandwidth substantially. We cured one issue with pre-emphasis overshoot but caused widening of broadcast bandwidth far outside our channel causing splatter on stations near us. So... we can filter the clippers output right so we still remain <10kHz audio? Yes we can.. sort of. We need to filter it, but by doing so we cause what is called audio filter overshoot or "ringing". This will cause the loudest peaks of our clipped pre-emphasized audio to be a few dB louder to the modulator following it compared to our seemingly leveled clipped bricked walled audio. So we just reintroduced the issue again but from a different process because of the nature of audio low pass filtering. You can see this process happen by taking a simple stereo equalizer and lowering the high frequencies and watch as you drive a square wave test tone into it and sweep it from low to high compared to a sine sweep, it will show peaks at high frequencies where you thought you reduced them because of the nature of Fourier transform and natural audio filter ringing.
Finally the last fix, and the last stage of the audio processor, and it's an important one...
Overshoot-compensation..
This is simply one of the most hairy stages of the entire broadcast audio processor. What we need to do is somehow bring our clipped/filtered overshoots back to a nice brick walled window of audio for our modulator even after all of that limiting and AGC and everything else, you'd think at this point you'd get 100% modulation without issue right? You'd be wrong sadly. The simplest way to do this is to phase correct overshoots because we know that overshoot is caused by a phase difference in audio filtering (phase shifts in audio filters as frequency increases), so we can take something like an OP Amp stage and phase correct audio by subtracting overshoot by analog prediction. This alone gets into territory of patent protections and the bread and butter of high end designs. I used the simplest idea I could find for this and not sure who owns the idea behind the invention so I won't get too far into this topic other than to say it's an art, and there's a reason why Orban, CRL, Dorrough, and other major broadcast industry companies own the secrets to this stage of their equipment. Even I am not absolute sure I understand the underpinnings but luckily for hobbyists we can look at the patents and recreate our own as long as we don't resell the idea. I spent the last month trying to find a workable circuit that could remove this filter overshoot to compensate for ringing of high frequencies from the final pre-emphasis stage and what I can say is that I don't know how I made it work, but it does, loosely based off of reading a ton of patents.
A tiny teaser of the work I have been soldering up after rigorous design work for the last few months...
For clarification, that's from top left to bottom right AGC, three band active crossover circuit, three band expansion, 12db user EQ, 3 band compressor/limiter/soft clipping, (moving onto breadboard in reverse) asymmetric clipper with POT to control between 100/100% modulation to +-125%/-+85% modulation, 9 pole 10kHz filter, overshoot compensation circuit.
May not look like progress but after almost a month I figured out how to overcome filter overshoot for tight modulation, and top breadboard showing a split virtual ground supply to take 30vdc and give me +15 and -15 because previous circuit wasn't cutting it. Lot of chips to power!
The circuitry for the filter overshoot compensator has been an absolute NIGHTMARE (Bottom breadboard, quad OP Amps). I don't even know how to describe the time this has consumed out of my life to study and understand why low pass audio filters overshoot when presented square waves and how to solve that issue.
The thing is with a broadcast audio processor the last stage is a clipper. It's a hard limiter basically, usually designed with two simple diodes so that if the compressor(s) overshoot upon transients which happens with every analog audio compressor circuit the diodes will "hard limit" those overshoots. Rarely audible and cures the issue. It's a necessity to achieve right up to 100% modulation without going 1% over. Brick wall modulation.
Most would think after the diodes you can just drive the transmitter audio input, and some el-cheapo circuits do just that, but problem is audio bandwidth. With mediumwave we want to limit our audio to 10kHz, or shortwave to 5kHz (legal definitions here, pirates play by their own game), that means the audio processors output must be audio bandwidth limited with a sharp cutoff filter. Even us pirates should obey this so not to splash over on adjacent channels. Studies I read suggest audio bandwidth outside of channel limits can be more detrimental than carrier cutoff from over-modulation to a point.
If we take the audio at this point from an oversimplified audio processor with nothing more than a hypothetical compressor, hard diode limiter right to the transmitter we will not over-modulate, but we will drive the diodes to clipping which will cause audio harmonics well above the audio range, some reaching out to 20kHz or more, and thus splatter on our neighboring stations a channel away!
Filters cause overshoot, so we clip audio and this gets rather harmonically nasty with pre-emphasis at high frequencies (AM broadcasts in the US use a 75us curve, 10db up at 10kHz then sharp cutoff), only way to cure that is to filter, but filtering squared off audio which clipping does causes modulation overshoot. The audio that is hard limited suddenly becomes peaky at high frequencies and not within our nice tubular limited envelope. The solution?
Sit back, this is going to be long..
There are multiple ways to solve filter overshoot. The easiest way is to rip hair out and stomp around or not give a shit and splatter on channels next to yours. You can build a precise slow roll off filter (bessel) after the main clipping filter. I've seen ham circuits that do this... (2), another way to do it is to use an OP Amp stage to phase shift and introduce the phase "clipped" audio into the negative side to subtract them (3), but this method doesn't work well for brick wall filters as the phase shift doesn't change in time fast enough with rising filter group delay time. Ugh this is not meant to be complex! Bare with me..
The simplest way to do this without getting into theory as to how Orban and other large companies figured it out is to simply say.. Capture the clippings from the overshooted waveform. Subtract those clipped overshoots from the output waveform. To sum this up I present you this incredible invention patent that I found on some search engine..
(1) THIS RIGHT HERE is the exact circuit I tested and am now using in the patent. It works, it is clean, and almost flawless. Oddly unlike other circuits, this puts the overshoot correction BEFORE the filter to "pre-condition" the signal to the filter.
The idea is to clip audio twice, but first time is your normal audio clipper to catch transients or your pre-emphasis limited overshoots, then phase delay the whole audio signal so to position high frequencies at a different time than low frequencies (all-pass filter circuit) clip again and combine with the subtracted clippings from that to arrive at a signal that is overshoot free. Clipped overshoots now subtract from the source causing once again that nice flat wall of audio worked so hard for by previous circuits.
I hope I got that right, my head is spinning and I am building this! Thing is once I get done with this design I will release it to everyone freely so maybe some of you can replicate it or someone can make a kit using similar circuits that way us pirates or part 15 ops can have a nice clean audio processor with real-time processing unlike Stereo Tool, and improve on it without the high cost of the high end stuff but performing equally.
100% modulation CLEAN and LOUD is the goal folks. Sure you can use asymmetric modulation to get out to 125% modulation but you gain *nothing* sonically. Precede your audio processing with phase scrambling (all-pass filtering )
and you will achieve the same *loudness* with less stress on modern transmitters that can't throw their modulator above symmetric carrier level.
Bob Orban said it ages ago and no one listened. You can push transmitters to higher positive over negative modulation but the original reason to do so was missed. The soul purpose was to pass asymmetric audio often from the radio host with a funny voice that peaked asymmetrically negative, but later on it was shown that phase scrambling the input to a transmitter with all-pass filtering did the SAME EXACT THING if not BETTER. Instead of pushing the transmitter to go more positive, we push the audio to smooth out asymmetric waves thus allowing us to continue less distorted 100% modulation with EQUAL loudness.
Building this audio processor I was so dead set on making the final clipper design capable of DC imbalance so to push positive modulation. After listening and seeing it on my oscilloscope myself I can say that the ONLY thing that was peaking up to %125 modulation was transients that weren't even audible. It did not increase loudness and anyone who tells you that going above %100 modulation is a benefit hasn't studied audio processing and broadcast transmission. It only benefits where asymmetric audio can, often only peaks of it, going one step further and using a phase scrambler before any processing can do the same and far better while preserving carrier level and quietness in carrier blackout upon silence, and loudness where needed.
NRSC-1 compliant audio filter with overshoot compensation circuit soldered up (the two quad OP Amp chips utilizing the US4737725A patent schematic linked above). It is completely flat out to 9kHz and follows the standard NRSC mask, down to >50dB at 15kHz. Testing it through my AM transmitter into a leaky dummy load I was not able to hear any adjacent channel interference even with music containing a lot of high frequencies like cymbal crashes or electronic sounds. Bypassing the filter definitely showed adjacent interference, even two channels away when playing CD content!
Next up is possibly building another filter with a switch to select either the NRSC compliant one or something different, not yet decided if I want the optional filter to be 5kHz for shortwave use or maybe wideband out to 12kHz+ for hifi AM broadcasting. After that I only need to build a simple one stage OP Amp to mix stereo input to mono, an output buffer stage to get everything up to correct line level for 600ohm transmitter input, and then some bar graph VU metering for each of the 3 band compressor stages for that cool psychedelic light show
Post by ogrevorbis on Jun 19, 2019 13:06:53 GMT -6
Nice design! I wonder what your take is on DSP.
I don't know as much as you about electronics, but I had much success with the ADAU1701 DSP from analog. The sigmastudio software is really fun to play around with. I'd recommend you try it. On aliexpress there is a board from MediaWorks shenzhen that contains the ADAU1701 for much cheaper than the official dev board and it works just fine with the official analog sigmastudio.
I don't know as much as you about electronics, but I had much success with the ADAU1701 DSP from analog. The sigmastudio software is really fun to play around with. I'd recommend you try it. On aliexpress there is a board from MediaWorks shenzhen that contains the ADAU1701 for much cheaper than the official dev board and it works just fine with the official analog sigmastudio.
I've never toyed with DSP chips yet. Of course the current trend in audio processing is digital, and if it can be implemented in single board programmable form that gives best results. This is exactly what the major broadcast audio processor companies do now, they have all moved away from analog at the heart of their units which is fine and accomplishes the goal while adding in tons of future possibilities. I prefer to go the old way with OP Amps because it's fun and there is no boot wait time or audio delay and is bare bones yet accomplishes the same goal. When something lands in the shop that is analog from input to output it's fairly easy to diagnose and repair, with digital gear you might not be so lucky.
There's a bit to be said about analog vs. digital audio processing too, besides the obvious audiophile cavetes with digital>analog and analog>digital conversion and bitrate and other distortions, there is also the imperfection of analog only audio filtering and compression and other things. Analog tends to give a lot more wobble room so to speak.
I've read text from broadcast engineers that swear digital has the bad drawback of sounding almost clinical. This makes sense because software can manipulate audio down to a precision, but it definitely loses something musically when doing so. I've heard big AM stations around here use those DSP Omnia and Orban processors and they sound very loud, and boring. When I listen to broadcasts that still chuck in the old analog processors they retain that pumped 80s gear sound and though it may not be perfect it certainly has a distinct sound to it that I've always preferred.
Of course there is Stereo Tool too, and I have played with it plenty, in fact it was that software that made me want to design a real AM analog audio processor, which is funny because the author of that software I believe got into programming that because he wanted a digital version of the expensive analog stuff. The cycle/spiral of audio gear lol.
Quick update too: Just got the input HPF built, simple 6 pole with a -3dB starting at 30Hz, necessary for when playing CDs and especially records that might be warped or something. Those subsonic frequencies can really screw with audio compressor circuits as inaudible low frequencies can cause them to gain ride useless parts of the waveform. A lot of transmitters can't handle subsonic stuff well anyways if they are built with series modulation. Makes me wonder what FM subsonics can handle without signal dropout but I digress.
This project is coming along nice, next update will probably be one with audio test clips
If you post a schematic, I'd be willing to design a custom PCB for you if you want. Maybe others would want it also. I don't know if you can design PCBs or not. . .
If you post a schematic, I'd be willing to design a custom PCB for you if you want. Maybe others would want it also. I don't know if you can design PCBs or not. . .
I will place the full schematic here when all is done which should be soonish and I'd greatly appreciate the help.
This project has been slow coming since most of the time and effort was put into research and bench testing various ideas while staring at my dual chan. oscilloscope for eye straining hours. The complete schematic will be quite large.
Since it's mostly designed on building blocks it is possible to slim down the design by removing features if someone wants something more basic, like removing the three band compressor to use only a single compressor, or not using the expander circuits if someone doesn't need the background noise removal from playing tape or vinyl. VU meters will of course be optional too and it would make sense to add PCB output points for metering as an option. I just ordered a handful of LM3914 ICs and a bag of LEDs just for this purpose so I can visually see the low/med/high bands gain reduction but I will release a schematic before I add all that in.
I'd love for someone to whip up a PCB design. That is one thing I have never tried doing yet. Looked around for free software to do this before and found some tools for Linux but that's another level of effort for an already huge project. It certainly would be great though because I would love to build a stereo pair eventually for FM use with different low pass audio filter values but there is no way I'd ever do all this again on protoboard because of the crazy complexity.
Edit: I should also mention the SSM2164 IC might be hard for some people to find, not sure if Analog Devices still sells them (chip with the cool play button logo on it in my pics) but there is an identical alternative made by THAT Corp. Only issue is the IC pinout is slightly different which would require a PCB IC socket layout difference. Kind of a pricey chip and it's doing the AGC and triband expansion in my circuit before it hits the OP Amp compressors, it can be omitted if a person only wants the multiband compressor/peak limiter guts. Otherwise everything else is generic OP Amps and nothing hard to find online. Tried to keep simplicity in mind while using all cheap easy to find components. Betting that if quad OP Amps are used the whole thing can be made far more compact than my prototype.
Input/output stage completed with active stereo to mono input converter, besides the RFI input filtering which will be simple to add. At this point it's a complete working AM audio processor. Controls so far left to right are Input Level, AGC In/Out, Bass Enhance, Treble Enhance, Processing Level, Pre-emph In/Out, Output Level, NRSC Filter In/Out. I will probably add a Asymmetrical In/Out toggle too since I did decided to include the variable 100-125% positive/negative modulation peak internal adjustment. Other than that, it's a working unit! Next step is drawing out the schematic using some software that hopefully produces files others can easily load into their own software. I use Linux so going to have to find something that is compatible cross-platform for others on Windows, and of course I'll save them in JPG image format too.
Only other things to add now are the power supply circuit which takes my 24VDC wallwart and converts it to +12 and -12 VDC using a virtual ground, and VU metering which I am still waiting on my LM3914 meter ICs to arrive in the mail so I can rig them up for all three multiband channels and output display. After that I have essentially recreated the Dorrough DAP 310 broadcast audio processor (also similar to the Texar Audio Prism minus the vactrols, popular AM/FM broadcast processor for top stations of the time) using some modern components to simplify a few things along with an included NRSC overshoot corrected filter and input rumble 30Hz high pass filter!
Here is the project case I plan to use that I modded with the circuit laying in before drill cuts, an old DVD player box that needs more work yet before becoming the housing..
Looks like shit right now but lots of leftover space in there to add an internal power supply if I want, but the right side will also contain the 10 LED x 4 circuit VU metering which is yet to be built, 3 VUs for each of the tri-band companders and one for final output VU possibly. As most of my projects start out the project box often looks awful at first but the end product ends up looking pro. Trying to keep the price down requires alternative project box ideas since a professional box would cost more than this entire project so far.
Yeah I'm showing off lol. Proud of my work so far and can't wait until the finished prototype is fully designed so that I can share the entire circuit to the public. I think I deserve to show off my skills a bit and gloat when it's free info for the pirate community
I promise next thing I will post will be the schematic, at least hand drawn at first to give an idea. My notes are so scattered right now and I left lots of provisions along the way for added accessories like alternative low pass filtering for other countries and VU connection points, modification areas, and so on.
When you produce the schematics, I will start a PCB file for you. I'll try my best to keep it error free the first time, but that doesn't always happen. I want one myself, so I'll pay for the PCBs (usually 5 or 10 of them). Let me know if you're interested.
Wish I had the opportunity to try it but by looking at the specifications it's not far off from what my own processor does. It doesn't have a triband downward expander like mine that I can tell but does have a noise gate which almost accomplishes the same task for removing background noise or tape hiss before hitting the compressors.
Only thing I don't like about the specs is the output filter starts rolling off right around 6kHz on the high setting, and on the low setting to make it NRSC compatible it rolls off around 4kHz which is far too low in my opinion for high fidelity audio. Would be fine for shortwave use though or even SSB. Guessing they didn't use a steep filter because that would complicate things by needing an overshoot correction circuit.
Just got my LM3914 ICs today in the mail, talk about perfect timing so now I can get some simple LED VU meters built. Should get to writing up the schematic in the next few days if I get the time.
Looks like I might give this open source free software a try to draw up the schematic.. www.kicad-pcb.org Curious if anyone here has tried it. Sounds like it can take hand drawn schematics and convert it too. I don't know much about the PCB design side of things but this should make my life easier to draw the circuit up later when time permits.
I took a look at that years ago and it seemed OK, but I decided to go with Sprint Layout instead because the learning curve with it is basically non-existent. I was making boards in a day or two. It's not free or for Linux though.
I think that is your best option for Linux unless you try to run Sprint in Wine.
On a side note, I used to be a Linux user also (Linux Mint), but I eventually went back to Windows 7 because I got tired of trying to compile stuff and dependency issues. I know that a lot of stuff comes pre-compiled, but I still found myself often needing to build stuff myself. I will probably stick with 7 until it gets old enough that it doesn't do what I need and then go back to Linux again, but I'm never going to Windows 10.
It is an online tool, but it's actually pretty full-featured. I've used that for schematics before. I don't use it to make PCBs, but that is also an option.
It is an online tool, but it's actually pretty full-featured. I've used that for schematics before. I don't use it to make PCBs, but that is also an option.
Decided to go with gEDA. It's simple and contains a huge list of components. Just tried easyeda and I don't like how it selects components through a buy part list or whatever, it confused me and seemed overly complex for the task, and KiCad ended up being overkill. Drawing up a few simple things tonight in gEDA to get the feel of it, real easy learning curve so far and solid software overall. Not sure if the .SCH files are cross compatible but I'll find out. There is PCB design software for gEDA too but not sure how good it is. I'll just be happy to draw up the schematic at this point given the sheer size of it.
As far as linux goes if I were you I'd give it a try again on a spare computer, Mint is up to version 19 after all and a solid OS distro now. I haven't had to compile anything in years now as most everything is in the software manager and up to date in most distributions. The last 5 years has seen some massive changes in the desktop area of linux for getting simplicity out of the box for the average user, but then I am a little biased since I haven't used Windows since XP and later versions of Windows feels foreign to me compared to almost any distribution of linux or BSD. Even pulseaudio has fixed most of the glitches making audio seamless with the software mixer and JackD audio for virtual audio connections. Makes for a great platform for radio automation now.
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