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Simple: Computational efficiency. Phase linearity isn’t a design requirement, so IIR is much more CPU efficient.
///Leif
Precisely. Mine is more accurate in the bass region, the region where it’s hard to get good accuracy 😉.
///Leif
Darwin, it was Process Explorer. Process Explorer is the latest incarnation of Regmon. Upon startup, it silently loads a kernel mode driver, and when you close the program, this driver stays loaded, with no way to remove it except rebooting.
///Leif
No, I’m writing the top one 🙂.
I didn’t write the bottom one, TrueRTA has been out for a long time.
An RTA, or "Real-time analyzer", analyzes audio and shows you the energy in each frequency band. It can be used to accurately calibrate speakers, listening environments, car stereos etc.
An RTA is actually very different from a spectrum analyzer:
A spectrum analyzer generally shows every frequency in a linear fashion, that is, an equal amount of bands for an equal amount of hz.
For example, if the bands are: 0-10hz, 10-20hz, 20-30hz, 30-40hz, 40-50hz etc… up to 19980-19990, 19990-20000, 20000-20010 etc.. That would be a regular linear spectrum analyzer.
The problem, as you can see from the numbers, is too high resolution for treble, and too low resolution for bass.
10-20hz is a whole octave in one band, whereas 19990-20000 is an extremely narrow slice of spectrum.
An RTA, on the other hand, has a fixed number of bands per octave. Mine runs as 1/3 or 1/6 octave (that is, 3 or 6 bands per octave). Each octave constitutes a doubling of frequency. For example, a 1/6 octave RTA would have the spaced like this:
20.0, 22.4, 25.0, 28.0, 31.5, 35.5, 40.0, 45.0, 50.0, 56.0, 63.0, 71.0, 80.0, 90.0 etc…If you count the number of bands in the 20 – 40 octave, you’ll see that there’s 6 (including 20). Count the number of bands in the 40-80 octave, you’ll see that there’s 6 again (including 40). The even spacing goes on this way all the way to 25000hz (12500, 14000, 16000, 18000, 20000, 22400, 25000).
Anyway. TrueRTA, like other FFT-based analyzers, START with a linear FFT, and then convert the values to RTA. It’s an efficient / convenient way to do it, but it means having very poor resolution for bass frequencies, as you can see in the screenshots.
///Leif
Hey Camclone,
The reason why I’m recommending not} to use the EQ in this manner, is that it will make you lose peak control, thus removing any benefit from clipping in the first place (while leaving any distortion in place).
However, I have a couple of different ideas.
First of all, I think the Dutch have it right in regards to the ITU-R SM.1268-1 Stokkemasker. They’re (as far as I know) the only country in the world that uses this standard of modulation measurement, but it’s also the one that makes the most sense to me — and the one that seems most accurate.
The spread of the FM carrier is not just a function the deviation (in kHz) but also of the signal causing the deviation.
For example:
All three images depict an FM carrier being modulated at full swing (+/- 75 kHz). The top one is being modulated by 30hz, the middle one by 15000hz, and the bottom one by 38000hz.
I’m no rocket scientist, but the top one (bass) looks a lot easier for a receiver with a narrow if-filter to lock onto, than the bottom one (stereo subcarrier).
The ITU-R SM.1268-1 Stokkemasker standard recognizes this, and requires limiting stereo separation so that an FM stereo station does not have a significantly wider footprint than an FM mono station. I believe this helps reception greatly, more so than limiting bass would.
So, if you’re worried about receiver lock, I recommend turning the ITU-R SM.1268-1 limiter ON. This will ensure a narrower, more controlled FM carrier footprint, and should ensure that your signal is easier to lock onto.
The other thing I recommend is: LIGHTER PROCESSING!
No matter what the modulation level is, it’s still easier for radio circuitry to follow along with a sinewave than with a squarewave. Also, if there’s still some dynamics in the audio, then the listeners ears won’t have to focus hard to try to make sense of the onslaught of brickwalled audio, and thus are less likely to notice the natural flaws of FM such as multipath dropouts, since they’ll be more relaxed while listening. (Disclaimer: No scientific or statistical process or procedure was used or referenced during the composition of the previous paragraph. All data comes directly from a friend of mine, statistician Marge Innovera.)
Try Reference Settings with all sliders in the middle. Yes, it will sound weak at first compared to Plutonium, but listen to it for a while! Don’t tweak. Turn up the volume control on your radio as opposed to the final clip drive. It’s an incredibly clean sound — basically CD quality on FM — and yet only 2-3dB quieter than the loudest, most distorted stations. I believe this light level of processing could work for almost any station, and since you won’t be pushing it hard, bass will not be an issue — any receivers (and any listener!) will be able to lock on and stay locked.
Best regards,
///LeifHi Camclone!
It possible to do the 67 kHz subcarrier with a 192 kHz sound card. However, the 92 kHz subcarrier is too high frequency for 192 kHz sampled audio.
I could possibly add support in a future advanced version 🙂. Would be neat indeed.
It’s not currently possible to run SCAs through the RDS input though — the input is *very* sharply filtered, and only allows 54 through 60 kHz to pass (plus the pilot).
It is however be possible to run Breakaway Broadcast into an external stereo generator software and yield good results — but you’d have to have another program generate the SCAs as well.
Best,
///LeifIt’s a nice looking card, but I would have no idea without testing it.
Asus Xonar DX works fine though — I’ve tested that one, so I can answer for that one, but only that one 😉.
///Leif
Sparky, there’s actually one major advantage you can get from composite clipping that you can’t get from L/R clippping:
2.1dB extra headroom for frequencies 5 kHz and above!
This happens to go come in VERY handy when you have a pre-emphasis curve working against you.
My composite clipping back-end does basically ALL the clipping in the composite domain (and not in the L/R domain), so the combined composite signal (minus the pilot) easily gets 12dB (!!!) of clipping.
However, the 38 kHz subcarrier itself is suppressed in the FM stereo system, so the answer to your question is "NO clipping".. If it’s not there, it cannot be clipped.
RDS/SCA protection is as good as my L/R clipper — there’s a brick wall at 54 kHz and nothing goes beyond.
Since I’m not particularly afraid of putting my money where my mouth is, here’s proof:
Check those numbers… MPX 101.4% (okay, the file is just a hair hot), Left channel 161%, Right channel 176.0%, Pilot Modulation 0.1% (within the margin of error due to the pilot band-pass filter in mpxtool being less than infinitely sharp), RDS injection 0.0%. Check the Output VU meters too — +6dB peaks (???).
Without composite clipping, Left and Right channel maximum is 92% (100% minus 8% pilot). 92% reads as exactly 0dB on the VU meters.
Check the density of that waveform.. Watch out, don’t touch the edge, it’ll split your finger right open 😉. Does it sound good? Can it sound good when it looks like that? Yes, actually. That’s the scary part. This clipper is still xylophone safe — it sounds as clean as BBP, in fact a bit cleaner due to the vast added treble headroom — and much more open.
I know, this looks and sounds too good to be true. So, check out a demo file comparing my L/R back-end (which is in BBP) to my composite back-end (unreleased). 🙂
http://bredband.leif.cx/browse/compositeclip
Play in MpxTool demo version. As usual, highlight both files in explorer and drop them together onto the MpxTool window, to enable direct A/B comparison. You can press Enter to set a bookmark — A/B will then start from the bookmark position.
A few more useful tricks to analyze the MPX:
Right-click the oscilloscope, choose De-modulated L or R, and compare the LR/CC files. You’ll see that the LR clipped file follows the 100% line perfectly, and the composite clipped file goes over like crazy, like the line isn’t even there.
Then, switch the oscilloscope back to Composite, and right-click the area between the scope and spectrum. You’ll see the MPX filter menu. (This menu can also be accessed from the Main page, but it’s easier to right-click here.. Just a shortcut.)
Remove Pilot and notice that modulation INCREASES when the Pilot is removed. (Yes, this should be impossible, and with every other composite clipper — except those actually clipping the pilot — modulation DECREASES when Pilot is removed.)
Then, Keep pilot only. Notice how it’s rock solid — not fluttering like it would be if the composite clipper was clipping the pilot.
Both of those, in combination, should be impossible to accomplish other than by "Smoke and mirrors". Except of course that there are none. You’ve got the MPX files — analyze yourself. Most of what MpxTool does CAN be duplicated in Adobe Audition (it just takes a hell of a lot longer to do it manually). You could also use the MpxTool composite output (and tilt / eq controls) and broadcast the file, and see what the Belar Wizard says about it..
Actually, I know what the Belar Wizard says about it. The L/R level meters get stuck at 127%. It can’t display any higher.
I showed this to an industry friend, and he was like, "Dude, you broke the Belar!". 😀
For comparison, the composite clippers in either of the two big O’s will give you at most about 110% L/R modulation (even at full +3dB composite drive), and they’ll be sporadic peaks — not dense, consistent "overshoots" as with mine. If you don’t have a recording of those to try, you can download that from MpxTool.com too.
Me brag? Neeeeeeeever 😉.
Will this be released? You betcha.
In what form? I don’t know yet.
Best,
///LeifHi Parelduiker!
Could it be this one?
I’ve recommended it a couple of times. It’s NOT perfect — it has aliasing problems for example, but if you’re looking for that slick, glossy treble, it does work — and it’s readily available, and free.
///Leif
Camclone, I’m sorry but that’s just not how it works. Overmodulation is overmodulation whether it’s bass or treble swinging the carrier — and in fact, I believe mids/treble will be more likely than bass to cause receivers to lose lock, simply because the carrier moves faster, has more sidebands, and is harder to lock onto.
///Leif
Hi Luk!
Yes — it’s definitely possible. In most situations, it’s cleanest without the plug-ins!
However, try some 80s, or classic rock — something that does not have a lot of bass, and then turn on the BASS-EFX plug-in — I think you will agree that it creates incredible bass punch on things that do not already have it.
Best,
///Leif
