Stereo Tool

Hans, have you reviewed the information and links on this legacy AM Stereo fan page?

http://www.amstereo.org/

Hans,

Since AM stereo is not used in Europe i don't know mutch about it.
Best thing to do is to look at the decoders specsheet.

Motorola made an AMQ decoder IC MC13020 and on the datasheet is probably some usefull information for you to check out.
As sayd this is the DEcoder chip but some signal fundamentals are explained in the data sheet.

http://www.e-ele.net/DataSheet/MC13020.pdf

Found this too: http://www.radiomuseum.org/forumdata/up ... ereo_2.pdf

It's in german but probably some usefull info

Hope this clears things a bit more.

Kind regards, Jeroen.

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Not know the transmission in AM Stereo, because in my country there.
But I provide a link that you could help.

This is AM Stereo:
http://maisonthenezay.fr/Maintenance_HF ... s/4-02.PDF

and this is the complete index, which is the complete Handbook:
http://maisonthenezay.fr/Maintenance_HF/NAB.html

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by GAP
"Less is More" (Bob Katz)

I don`t know why they not make AM stereo instead FM stereo in the same band . Since the AM is the most "right" modulation, i can`t see something wrong. May be AM stereo with 15Khz bandwidth per channel will cause more than 150KHz modulation ? I think NO ....

STAND ALONE: http://www.stereotool.com/download/ster ... 43-010.exe
WINAMP/DSP: http://www.stereotool.com/download/dsp_ ... 43-010.exe

CHANGED:
Improved limiter behavior. The limiter (in all compressors) now works much tighter so it causes less pumping and at the same time it causes less distortion.

TO DO:
- Show Lowpass filter freq in AM panel?
- C-QUAM?
- Add Pirate License (3 days)
- Add auto-start with Windows
- Add tests for exception handling

OLDER CHANGES:
- Support ASIO as well in the watchdog
- Improve restart behavior (make sure that the old instance is really closed).
- Memory usage of stand alone version reduced.
- Merged in older changes in a separate branch. should not have any noticeable effects.
- Lower CPU load when using streaming with low pre-emphasis values. "Hole protection" must be set to LQ mode if it's disabled for non-preemphasized audio.
- Make SCA frequencies configurable for HAM radio
- Real watchdog behavior: Regularly check for sound card input & output
- Added restart-on-crash behavior (enable 'Watchdog' in the Configuration panel).
- Fixed one weird hangup in VLC output when passing bad parameters
- Added option to pass extra parameters to input VLC (just type them after the URL).
- Carrier at +/- 19khz on the AM band. Just turning the FM off doesn't kill the stereo pilot. (fixed in 004)
- I again will say here that there is something wrong with Calibration section . Three options are identical with what they do : Test right silence , Test SSB lower and Test SSB higher. Nevermind what option you check, they causes same display behavour and same result as audio. --> NO, they are very different if you look at the actual output. Display is generated before calibration. Actual output does contain SSB or LSB audio.
- At the new Normal output settings there is a problem with "Input without processing" option . Sound is gagged/strange / I'm not at my Stereotool pc at the moment but when using a pipeline input and normal & fm output (192 kHz) , I noticed a strange effect when switching the normal output to bypass. Sounded like it could be a sampling rate thing? - Worked fine in some cases but not in others. I think it's fixed now.
- I think I also noticed a small typo - check the AM transmitter window.
- On the thorium preset it plays fine and then gets garbled and sounding like a 45rpm song playing on 33 rpm. cpu load is just 26% the rest of the presets work fine. (waiting for feedback, unable to reproduce it). --> Multipath Clipper is enabled in this preset. That explains the higher CPU load.
- VLC output still hangs on outputs that don't work (for example an invalid ShoutCast address). Probably because no data is sent through the output path, which causes the same stop hangup again. --> If Status is STOPPED, ERROR or NOTHINGSPECIAL, I don't call STOP or RELEASE. This seems to cause a small memory leak (a few kB probably) every time it happens, so it's not ideal. But better than a complete hangup. Still waiting for feedback on a question in VLC's forum.
- Add VLC output error detection & re-connect behavior (if possible without resetting the other inputs/outputs).
- VLC input: But something is strange sometimes plays like 1 or more samples delay between channels, occurs highs cancelation when switch to mono. I checked everything including exact mono sound input. -> Output was already running before connection was established. That's ok, but the start sample must be left channel -> fixed.
- VLC input with Low Quality Low Latency output hangs -> Solved by disabling LQLL output when VLC is used for input (which makes it impossible to reach low latencies anyway)
- VLC MP3 output. No room in current settings window... -> You can now add more options in the SOUT string (space separated)
4. Turn AM stuff off if FM is enabled (?) but what about asymmetrical clipping/limiting for FM? -> Extra switch in Expert mode
5. Add pre-emphasize output button for AM pre-emphasis
Fixed the mentioned and some other bugs in the previous beta. All combinations of settings seem to work now.
- Normal Output can now be partially separately processed for FM output. So you can generate, for example, a composite clipped 75 us pre-emphasized FM signal and a not or far less pre-emphasized signal for streaming, or a separate FM signal without composite clipping to feed remote transmitters, in the same Stereo Tool instance - no more need to run a separate copy for the stream! Only the clipping and limiting is done separately, so the CPU load does not go up much. Note that the lowpass filter can not be specified separately (lowpass filtering happens much earlier in the sequence and using different values would require doing almost all the processing twice).
- AM section now contains an 'AM Transmitter' Enabled switch.
- Headroom slider added to the AM section. No more need to abuse the Post Amp slider to create headroom for asymmetrical output. (Note: It's also not possible anymore, so you need to update your settings if you use AM).
- AM pre-emphasis checkbox added in the AM section, so you don't have to abuse the FM pre-emphasis anymore.
- AM pre-emphasis curve (which is slightly different from the FM curve) is now correctly implemented.
- AM signal can now go through the FM sound card output (it's now basically an FM/AM sound card), including tilt correction etc.

This watchdog behavior works as follows: If you enable the check, it will act if no audio passes through an enabled audio input or output for more than 1 minute. If so, it will trigger a 'RESTART SOUND CARDS' button click. It will attempt this 4 times, if NO DATA passes through the sound card AT ALL during all these attempts, if the 'Restart' watchdog setting is enabled Stereo Tool will close and a new instance will be started.

Definitely! I now finally understand *how* they did it - it's actually pretty easy but before I couldn't really grasp how they keep the spectrum within limits (the answer is: They don't!!! AM Stereo causes the signal to get outside the 10 kHz bandwidth in the US, which is apparently acceptable because the signal outside this range is weak and the band isn't that full. In Europe, because of stricter regulations AM stereo is completely impossible according to this document. (I actually disagree slightly - it must be possible to make a limiter and clipper that adhere better to the mask but stereo separation will be less strong - the Stokkemask filter in Stereo Tool for FM stations is an example of a similar technique. The effect on the audio will be MUCH bigger on AM though.

So, here's what happens:
1. We create a carrier which gets phase modulated based on the L-R signal. Basically this means FM modulation on AM! (unless I'm completely misunderstanding things)
2. We modulate the L+R signal with that carrier.

1. is the reason that the spectrum gets too wide (as it does on FM). If the L-R signal gets stronger or moves faster (ie. has higher frequency content) the signal gets wider. Loud high frequency tones have the biggest effect. Since the AM spectrum itself is already determined by the L+R signal, loud and high frequency tones in the L+R spectrum will already push the spectrum near the edge, and the L-R phase modulation will make that worse, pushing it over the edge.

In fact this means that an ideal clipper/limiter for C-QUAM would do a bit more than simply limiting L+R and L-R, basically, at points where the L+R signal has a lot of content around the edges (loud high frequency tones), I think the L-R signal should be reduced. And high frequency L-R content should be avoided in all cases. Basically, the more high frequency content there's in the L+R signal, the lower the frequency should be above which the audio should be blended to mono or limited below a certain threshold.

That would all be great, but it falls well beyond my idea to 'simply' add C-QUAM limiting to the processing. I also don't know if others do this (?) - it's not mandatory but if it reduces problems to neighboring stations it could enable more stations to broadcast in AM stereo.

And that one helps as well, because it finally describes how to limit/clip the signal. I did find another NAB handbook but it lacked the info that's in this one. I'm still trying to fully understand what's happening (just read it for the first time) but I think all the info that I need is here!

Edit: Just re-read it. It describes WHAT to do but not WHY, which is what makes it a bit difficult. Anyway, the short version:

1. If you clip/limit L and R separately, as I currently do in Stereo Tool, in a worst case situation (1 channel audio, 1 channel silence) you would loose about 6 dB of loudness in the L+R signal, which most (mono) radio's receive. For AM that immediately means a reduced reception area etc.

2. To avoid that, L+R and L-R are processed separately. That fixes the problem, but there's another weird situation: If you really have 1 channel audio and 1 channel silence, then the channel with audio will be twice as loud (+6 dB, because you get 0 dB mono reception which is then moved to a single channel) than it would be in a mono broadcast.

3. For some reason, in the NEGATIVE direction, peaks this big aren't allowed. If we call this +6 dB per channel (maximum possible peak level per channel for one-channel content) 100%, so mono audio would lead to 50% per channel, then each channel may not peak below -75%. Since this will rarely happen and the difference isn't that big anyway, the simple solution would be to add a limiter after the clipper which limits the left and right channel below 75% - we can probably just as well do that symmetrically.

Fortunately I have just improved my limiter a lot, which means that I can immediately use it for this


Edit: Oddly, the NAB manual from gpagliaroli says the opposite of phantomfm's German document: According to the NAB document there is no difference in bandwidth between mono and stereo AM broadcasts. I'm confused again. But never mind - I'll just add the limiting and then it's at least compliant. If I want to improve things later I'll dive into this again...

for AM limit and clip is a MUST too. it cannot be endless, right ?

Hans , you may ask some radioamateurs that work in the 3.5/7 Mhz band . Their transmitters works in AM LSB mode.

Also you can check in WebSDR.org on some receiver to see that good transmitter have limited signal .Transmitters without limits are taking more "space" on the watefall.

As i understand well , the AM stereo uses DSB ?
See Left is well limited , but on right is not limited .

Wooh. Ok, I've implemented it all now - AM L+R and L-R clipping, and limiting of the L and R signals. But, a few things are BAD.

First, the old situation. To generate a C-QUAM compliant signal, you need to enable AM, and disable asymmetry (because the L-R signal may not be clipped asymmetrical).

Compared to this, it looks like I can easily squeeze about 4 dB of extra audio in the signal with a C-QUAM optimized clipper and limiter. According to the documentation that I read, it should be rare that a stereo signal peaks above 75% (using the definition above, where 50% = what you get with normal left and right clipping/limiting). However, I sent some music through it and it constantly peaks to very close to 100%. I had expected to be able to use a limiter to remove some spurious peaks, not that it would be active all the time.

There are more issues. With filtering L+R asymmetrical and L-R symmetrical, with only audio on one channel, the results are similar except for the peaks on one side. Combining to L and R again gives crackling sounds on the silent channel. Turning asymmetry off fixes this. But I don't want to do that!

The solution to this is very simple, but it's not so easy to implement because a lot of CPU optimizations were done to the clipper that need to be 'undone'. That's because for each clipping step I need both the left and the right channel audio. And I don't have those - I've split the processing over multiple cores and they are completely separate.

Since only few people will probably use this, I don't want to spend this much time on it now. If it turns out that this is a popular feature I might in the future, but not now with only a few people who said 'it would be cool to have it'. So I want to use a different solution: I first clip the left and right channels so they don't exceed -75%, then I go to the normal clipping, and finally I use L/R limiting which will remove some small peaks but not much more. This will result in a higher CPU load and it won't fix the crackling sounds in 100% audio on one channel (unless you disable asymmetry). It will give you something very close to maximum loudness though.

It would be very useful to know if the cause is in VLC or in HiFi cables. If you've seen it multiple times it's probably easy to reproduce - can you choose one of the two and hit 'RESTART SOUND CARDS' a number of times? That does something very similar to switching (it closes everything and re-opens it). But that way it will be clear who caused it.

O, if this happens, does the audio work? It looks like - assuming that you made the screenshot without changing other things first - you switched from VLC to HiFi, got this message but HiFi works. If that's the case, then it must have been the closing of VLC that caused it, since opening HiFi went ok. If it's that then reproducing it is less needed (especially if you know if it happens every time when leaving VLC mode).

All you need for that is a good lowpass filter. Except that for AM stereo things are apparently very different, and you'll NEVER get a steep cutoff (the image on the right looks like spectra for AM stereo transmitters as shown in the German PDF file).