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SparkyMember
Yeah that’s what I thought but wanted to check anyway.
thxSparkyMemberJust wondering why you chose IIR filtering vs. FIR in this application?
February 10, 2009 at 5:45 pm in reply to: Let’s …over-modulate…using high frequencies only!! :) #6547SparkyMemberquote :This forum board has become very powerfull lately,Yes it has, especially with a brilliant software engineer at the controls (Uber coder )
Hey Leif, how about adding another test tone to the set up? 😉
Freq: 9.5kHz, mono, no MPX subcarriers or pilot. 100% modulationWhy this frequency? It’s 1/2 of the pilot tone. When checking a transmitter for group delay symmetry, this tone in conjunction with a modulation meter allows the user to see and adjust for any asymmetry.
When high group delay is present it causes an increase in the even order harmonics of the 9.5kHz tone sidebands. The second harmonic falls squarely within the passband of a modulation meter pilot tone detector (very narrow band filter). If high amounts of harmonic energy is seen on the meter, then asymmetrical group delay is present. The higher the reading the worse it is. Now the user has a simple visual indicator to follow while adjusting the transmitter. Tune and adjust the coupling stages until the 19kHz harmonic nulls out or is reduced as low as it can go. Transmitter interstage group delay symmetry has been restored. 8)February 9, 2009 at 8:41 pm in reply to: Let’s …over-modulate…using high frequencies only!! :) #6545SparkyMemberCamclone,
My comments were not intended to say that you and your station were non professional in scope, rather from the limited information provided I had no way of knowing this, and thought perhaps it was of your own design.
As a +30 year veteran in and out of the broadcast business and a wireless design engineer here are a few points to consider.
Receiver lock. Yes as you have pointed out, different brands of receivers have varying levels of engineering quality built into them. Not all brands follow the nominal design guidelines needed to receive standardized broadcast transmissions. Because of this, some people, due to their individual choice of receiver brand purchased, will have a less than satisfactory listening experience. There is nothing you can do to prevent this except to adhere to the broadcast transmission standards of your country as closely as possible such that the largest majority of listeners who do have adequate receivers will receive your signals properly.
Loss of lock in FM receivers as you describe or have been seeing is not due to a loss of main FM carrier PLL lock. All superhetrodyne receiver designs use an unmodulated local oscillator to convert down the FM signal of choice to a 10.7MHz intermediate frequency. This frequency is then sharply filtered and applied to the modulation recovery circuits (demodulator). The quality of the IF filtering as Leif pointed out will have a significant impact on the quality of the recovered audio and subcarrier signals. All FM broadcast receiver ceramic filters must be engineered to have a sufficiently wide and flat bandpass response, in combination with very low group delay for all of the allotted FM channel bandwidth (+/- 100kHz). Typical ceramic IF filters for FM broadcast reception use anywhere from 150-330kHz of bandwidth depending upon the designers ultimate design goals. In most instances the loss of "PLL lock" you describe can be largely due to issues within the receiver stereo demodulator sub-circuits. A large part of distortion from these circuits come from non-linearity in the phase discriminators, subcarrier filtering, and internally generated synchronous AM of the 19kHz stereo pilot. Signal multipath distortion compounds the problem of synchronous AM of the 19kHz stereo pilot.
However, there are many reasons why a loss of "stereo PLL lock" in a high quality receiver is encountered that are directly caused by the FM station itself.
These are: synchronous AM of the FM carrier and MPX subcarriers, transmitter interstage group delay, MPX tilt at the exciter, passband tilt of the antenna array.
In all my experience in the broadcast engineering business, a lack of symmetrical interstage or amplifier group delay is the most common of the above problems. Many times I personally have seen what this can do to a station’s 100kW FM signal.In an ideal FM transmitter the main carrier deviates symmetrically from center frequency regardless of modulating frequency. This in turn generates modulation sidebands with equal amounts of energy. As these sidebands propagate through various amplifier circuits, each frequency group will pass at a slightly different rate such that all groups are not delayed equally in time. However the design goal of a transmitter engineer is to build symmetrical group delay throughout the system which equates to constant group delay or linear phase shift with modulating frequency. But many times the interstage coupling between the exciter to the IPA or the IPA to the final amplifiers introduces significant group delay causing anything from tiny to massive amounts of distortion seen only at the receiver. Throw in signal multipath distortion it’s any wonder a receiver can faithfully recover any listenable audio. Excessive processing with uncontrolled clipping makes a bad situation that much worse. So if the transmitter modulation group delay is large, the MPX signal will undergo lots of phase shifts that will make the received audio and MPX subcarriers appear to "lose lock". You the station engineer need to perform a sweep analysis of your entire transmitter chain (from exciter all the way through to the antenna) to see if this is indeed the case. If the results of your analysis shown everything is perfect, then you have done your job to ensure the station is transmitting a clean signal. The rest is up to the listener and the uncontrollable effects of how your signal propagates around the coverage area. Can’t do much more than that.
From a design engineering perspective, multi-module MOSFET amplifiers are very much prone to the effects of group delay distortion. Because each amplifier module splits the incoming signal, amplifies it, then recombines the outputs to create the TPO, any slight coupling mismatch will create noticeable group delay, both at the carrier level and MPX modulation sidebands.
February 6, 2009 at 6:24 am in reply to: The advantqages and disadvantages of MPX clipping … FM #6527SparkyMemberThey should outlaw such clippers. With all the clipping commercial FM stations do to the audio nowadays, it won’t be long before they’re sending DC over the airwaves. Can’t get any "louder" then that. Perhaps at some point my Sony walkman radio can be designed to be self powered from all the clipping. It sure would save having to regularly buy batteries… Crystal radio sets live again !
Leif, how many dB is the 38kHz subcarrier clipped?
Also, does your post clipper filtering provide sufficient protection to the RDS and SCA subcarrier spectrum?February 6, 2009 at 6:03 am in reply to: Let’s …over-modulate…using high frequencies only!! :) #6541SparkyMemberWhat a bizarre request… but having read the post a few times I think I understand what his issue is.
Camclone, it appears you have a poorly designed (or broken) PLL synthesizer in your transmitter, specifically the loop filter.
This loop filter is a group of circuits that generates the feedback control voltage keeping the transmitter on frequency.
If the loop filter circuit time constants are improperly chosen or wrong, you will have high instability from audio modulation, especially lower bass frequencies. Changing Breakaway’s frequency equalization to accommodate the faulty transmitter is not the way to fix this problem. To do so would be similar to asking a mechanic to remove the doors off your new car because the air conditioner is broken. 😉Is your transmitter design a kit and was assembled by you? If so, I suggest taking a close look at the component values in this circuit. An incorrect resistor or capacitor value will have a dramatic effect on how well it works. If it’s of your own creation, then I suggest you find more information on loop filter design and try again. Once you have evaluated your circuit I recommend you use the 30Hz burst tone tool found in Breakaway’s test tone settings window. This will help aid in determining if your changes were effective or not. If you have access to one, an oscilloscope is a great tool to view the loop filter output when the 30Hz tone burst is applied. It will quickly show you how well the PLL is working and it’s stability (or lack of) to a low frequency impulse. If the viewed waveform is wildly swing around each time the tone burst hits, the loop filter still needs work. However if you only see a tiny "bump" in an otherwise flat trace then you probably have it right (or very close).
December 10, 2008 at 7:32 pm in reply to: 30Hz… Why is this freq the low end cut off point for FM? #5984SparkyMemberquote :Does anyone still use SCAs?Here in the states SCA’s are still active, but not nearly as much as 25 years ago.
The largest users of audio type SCA programming today are ethnic broadcasters. This is a form of specialized broadcasting serving a small but highly targeted audience. Buying a radio station today is dreadfully expensive, but renting SCA subcarriers for this type of programming is an inexpensive alternative. Large population centers that are ethnically diverse are the most likely areas to have stations of this kind. New York City for example has a large assortment of stations on SCA. http://www.n2nov.net/NYCareaFM_SCA.html
Here in Denver a NPR affiliated public broadcaster has Korean programming on 92kHz.
I found this YouTube post on this very subject. http://www.youtube.com/watch?v=V0Qo7vKYQ-M (note the modified Physicians Radio Network table top radio)The second largest users of SCA is "Radio reading services for the blind". As the name suggests this service targets blind people in providing news and other entertainment needs. Seeing that blind folks can’t use computers and internet connections very easily, it remains quite popular. This service is (or was) also carried on many analog TV station SAP channels. I would imagine it will carry over to HDTV secondary channels (or HD FM broadcasting too).
Most of the elevator music providers like Muzak have largely abandoned SCA and now use satellite distribution methods.
Other specialized audio services that were once active but now defunct are the Physicians Radio Network, and Farm Commodities Information Services.
On occasion, some FM stations use SCA channels for backhaul distribution to affiliated AM transmitters. I’ve even heard stations when on remote location (local ball game coverage, auto dealer grand openings ect…) the use the SCA channel for a studio to remote "cueing" channel. This always makes for interesting and sometimes colorful listening.In the digital realm, a lot of SCA activity is still used for remote transmitter control and metering. Modulation methods are direct FSK of the subcarrier, or modem style audio tones.
Other digital services include things like Microsoft’s "directband" (which has been renamed MSNDirect http://www.msndirect.com), and mobile stock quote messaging (which may be defunct and has transitioned to internet).
quote :I believe it might be possible, at least from a technical standpoint, to generate a 67 kHz SCA in software, make it part of the regular Stereo MPX signal, and output it through a sound cardThis might prove to be an interesting challange. You would probably need to have a audio processor for the SCA programming too. Breakaway SCA?
December 9, 2008 at 7:59 pm in reply to: 30Hz… Why is this freq the low end cut off point for FM? #5980SparkyMemberWow… totally forgot about the old automation tones. Haven’t thought on those line in quite awhile.
On a different note I will add that subsonic tones were used in some instances for remote transmitter metering information sent on SCA subcarriers. the 10-25Hz control tones were commonly mixed with the Muzak program audio. Most SCA receivers on the subscriber end had high pass filters that largely filtered this out. But between music selections and the quality of the receiver and signal received, you could sometimes faintly hear the warbling of the control tone in the grocery store. I forget which transmitter manufacturer used this system (Marti, McMartin, Continental?). This sub-sonic tone system seems so rudimentary compared to today’s high precision digital systems. But it sure was reliable from a reception standpoint.
December 7, 2008 at 6:21 am in reply to: 30Hz… Why is this freq the low end cut off point for FM? #5973SparkyMemberI may be wrong here but I don’t believe elephants truly hear 30Hz. They sense it much like we do.
Whales on the other hand can hear 30Hz or lower largely because of the excellent transmissive properties of water (approx 1500m/sec).
Low frequency energy travels great distances in water due it’s elastic nature, and evolution has given whales "ears" that are perfectly developed to "hear" these frequencies.
You don’t hear whale love songs sung in soprano for a reason 😉December 7, 2008 at 2:44 am in reply to: 30Hz… Why is this freq the low end cut off point for FM? #5971SparkyMemberLeif,
Good points… all of them.
I might add the only known creatures that can truly "hear" 30Hz and lower frequencies are whales.
The only known musical instrument that can truly produce 30Hz note is a pipe organ, usually requiring massive amounts of air and a hefty compressor to pump it.Sparky
SparkyMemberquote :What am I missing?Actually nothing. I was in error in providing the test conditions 😳 . In efforts to recheck my work I came to realize I inadvertentaly generated a spatial stereo white noise file run through BBP w/ pre-emphasis on.
From the cool edit help files.quote :Spatial Stereo
Cool Edit 2000 generates Spatial Stereo noise by using 3 unique noise sources, and spatially encoding them to appear as if one is coming from the left, the other from the center, and the last from the right. When you listen to it with stereo headphones, your mind perceives sound coming from all around, not just in the center. To choose the distance from center of the left and right noise sources, you can enter a delay value in microseconds. About 900 to 1000 microseconds corresponds to the maximum delay perceivable, and a delay of zero is identical to Mono noise (left and right channels are the same).The settings in my test condition was 500uS.
So me bad… 😳
Looks good though. Makes the sidebands really stand out. 8)quote :Alright, you’ve convinced me about the 30 Hz. It’s a worst case scenario test, and if your airchain passes that, it will pass anything. I’ll put it in.Well perhaps user feedback will decide if it’s work keeping. The DJ crowd might find it useful for locating PA system crash points. Are you planning on shaping the 30Hz burst edges?
quote :However, the pre-emph white noise, I need some more explanation on.I personally like to test things in the way in which they will be used. White noise put through the pre-emphasis process as you pointed out is not "white" anymore, and the energy distribution will be less overall but more concentrated in the upper audio freq regions. Because RF modulator non linearity tends to favor the higher frequency components more so then lower, the pre-emphasied signal makes it easier to locate problems. Taking this further it pushes the RF system bandwidths right out to the edges. Any improperly tuned circuits (filters, interstage coupling, narrow band antennas) will show up a bit more easily vs having uniform noise energy over the whole bandwidth. But again this is a matter of preference. I think for most BBP users plain o’l white will be the most useful. 😉
SparkyMemberLooks really good 😀 😀
However…Plain o’l white noise is great for checking the MPX portion of the airchain. Your MPX tool demonstrates this very well, but once the signal is passed on through to an exciter the ideal world abruptly meets reality. Not all exciters are alike. Each have their strengths and limitations. Many introduce sonic artifacts that can are caused by modulator non linearity, high Q circuits exhibiting group delay, and coupling asymmetry in cascaded amplifier stages. I know many exciter designs use high frequency predistortion techniques that try to compensate for some of the non linearity, but many times it fixes one problem by creating another. Offering both options. i.e White and FM Fuchsia noise spectra will give the user that much better tool set to test their system with (IMHO).
I understand your argument regarding the 30Hz options and I agree in part. Any good designer adheres to the KISS principle (Keep It Super Simple). But the main objective of the 30Hz tool is to see how bad the overshoots really are. Remember this should be an "unprocessed" signal. The 30 Hz tone w/ burst option is to see how the exciter AFC circuits handle this kind of impulse energy. A good way to think of it is testing how well your cars’ suspension responds to an impulse like a deep pot hole. A well designed system that is critically damped the driver hardly notices the bump. Over damped and the driver gets kidney damage. Under damped and the car goes bouncing merrily along for a few km well after the pot hole event (ie. cars with no shocks… just springs ).
Exciter PLL control loop filters behave no different. Loop filter dynamics can sonically alter the sound of deep, booty style bass, or heavy bass drum beats. Under damped systems (fast PLL lock times) can create huge RF carrier overshoots that will exceed government regulations, or generate bad asymmetric carrier deviations. Over damped and the base tones get stripped down leaving no low end dynamics. The 30Hz tool is a convenient way to see how your exciter handles such extremes and distortions (if any) it creates. I know many of your customers are or will be future radio pirates that will eagerly embrace BBA as a lower cost way of sounding competitive like their professional FM counterparts. A lot of them will use hand built equipment. Why not give them another tool to help keep the gov radio police at bay? Perhaps as a compromise the 30Hz burst is shaped to help soften the initial overshoot vs. the first 1/2 cycle going right to 100% amplitude. Shaping is more like real world anyway.
Any more thought of using touch tone signal pairs for the preset control mechanism? If you want I could generate a few tone files for you to test FFT processing. Let me know.
SparkyMemberquote :I couldn’t think of a name to call it, so i just made up “JesseG-Weighted”Funny How about shortening it to G-Weighted?
quote :It’s been through pre-emphasis. It might make for a much more even-looking RF spectrum if it was white.This would be true if the RF carrier was moduled 100% white w/ no pre-emphasis.
I’m not sure what to call it seeing that it does not fit any of the standard "noise color" definitions.
Closest is violet noise, or differentiated white noise at 6dB per octive.Perhaps we should name it ourselves. How about FM Fuchsia ?
SparkyMemberFYI.
Attached are two photos of the MPX composite signal with white noise. The noise was generated using Cool Edit (mono 16-bit 44.1kHz) driving the BBP pipeline.First is the direct composite output (w/ 2 SCA’s unmodulated). The L+R and L-R matrix is clearly visible with the pre-emphasis slope on each. If a BBP user has an o-scope with FFT capability then white noise modulation will aid in showing tilt when using a sound card for MPX generation.
[attachment=1:2ogu8chf]White Noise Modulation Of MPX Composite Signal With SCAs.jpg[/attachment:2ogu8chf]The second photo is the actual RF spectra (no SCA’s). Notice how symmetrical the sidebands are relative to center (the pilot and sideband are the two discrete spikes). This is a properly tuned system that has sufficient bandwidth. Any tuning mismatches will show a noticeable skew in the symmetry. Unfortunately I don’t have anything handy to de-tune it enough to show this.
[attachment=0:2ogu8chf]RF Spectra White Noise Composite Modulation.jpg[/attachment:2ogu8chf]SparkyMemberquote :The one major reason I can think of for not simply adding another control for “burst mode” for all test tones, is that I want to make it as simple as possible for the user.Sure makes sense. You should poll the user base to see if they would want this.
quote :There’s already tooltips that pop up when you place the cursor over the bar of interest. There have been all along. Geez, doesn’t anyone notice all the effort I spent on these little thingsOh it didn’t go unnoticed. I was refering to adding the frequency range information for that band/bar that is pointed to. i.e. Band 2 of 6 – 300Hz to 600Hz (made up numbers)
quote :For white noise to modulate L+R and L-R equally, the noise in L and R must be completely uncorrelated. Easy enough to do, perhaps! It’ll be a useful feature indeed — BBP is a broadcast processor after all.Every little bit helps too…
1. make BBP kick ass.
2. Best sounding FM signal for Pro or Pirate alike.quote :had 8 preset recall slots, and I include 8 ready-made MP3 files to recall these slots, I would save myself a lot of grief.Yep exactly as I was thinking. Perhaps use the telephone touch tone frequency combinations (15 extended tone set). Non harmonically related, easier to FFT.
The number of sets could be expanded easily by stringing 2 tone sequences together (like dialing a two digit phone number) to give you 225 possible combinations. Using two tone digits further prevents false positives from music energy.
You may not remember this but satellite program providers used this technique to signal cable TV providers when to insert local programming automatically. These tones were clearly audible. Short 6-8 tone bursts… beep bip boop… click something happened somewhere out in TV land. CBS radio network did the same thing too back in the 70’s during network programming feeds. Worked great for stations that were unstaffed and automated overnight.Now if you want to get really fancy, reserve a few tone sequences for control items outside of audio processing. How about a tone sequence to issue a turn on or off command to a transmitter? With an LAN capable AC power switch, BBP can issue a command packet to the designated IP address and… click. 8)
quote :…but if I just include pre-encoded filesExactly. Encode and name them (tag) for the current presets that are most popular.
quote :This could be really really useful — even more so than dayparting/scheduling. Simply embed the sequence into the theme/intro of the show, and it’ll be all automatic.Glad you like it 😀 I’ve been wanting something like this for a long time. I would recommend seperate files though instead of mixing it into the program start/stop. This keeps it simple and can’t be screwed up.
quote :Also, the detection could be very quick and basically fool-proof this way — the chance is basically nil that music would have exactly that combination of frequencies, especially if I make it a multi-stage sequence.This is why the telephone engineers picked the specific tones for touch tone signaling. These combinations are not harmonically related, and most speech patterns didn’t contain very much of the tone frequency energy at any given time to generate false positives. Smart dudes at Bell Labs in the 60’s
quote :Breakaway Broadcast indeed has plenty of look-ahead to mute the output. Breakaway Live doesn’t, but who cares — the tones don’t need to be muted in the studio. It’s not like they need to be full amplitude, -40dB down should be plenty strong enough.40dB should be fine if you encode the files to maximize SNR. But in reality how much tone signal time do you need to capture what you need? More than the audio signal latency of BA Live?
Cheers
Sparky
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