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FM radio limitations (?)

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Anonymous
August 13, 2005 12:28:36 PM

Archived from groups: rec.audio.pro (More info?)

I was wondering how a music signal gets compromised when it is
transmitted via FM radio. Other than squash-it-to-death broadcast
limiters, is there something about FM that reduces signal quality (less
stereo separation, distortion, less bass, less dynamic range).

just wondering what gets compromised and why...

More about : radio limitations

Anonymous
August 13, 2005 8:04:36 PM

Archived from groups: rec.audio.pro (More info?)

On 13 Aug 2005 08:28:36 -0700, genericaudioperson@hotmail.com wrote:

>I was wondering how a music signal gets compromised when it is
>transmitted via FM radio. Other than squash-it-to-death broadcast
>limiters, is there something about FM that reduces signal quality (less
>stereo separation, distortion, less bass, less dynamic range).
>
>just wondering what gets compromised and why...

FM needs an infinite bandwidth for perfect distortion, but that means
infinite noise - so it is necessarily a compromise between these two.
Stereo separation has no real inherent limits, nor does bass although
phase locked loop implementations of demodulators in receivers have
tradeoffs of their own in that area. Distortion and dynamic range - as
I have said - are commodities that are tradeable between each other.

d

Pearce Consulting
http://www.pearce.uk.com
Anonymous
August 13, 2005 9:14:12 PM

Archived from groups: rec.audio.pro (More info?)

genericaudioperson@hotmail.com wrote:
> I was wondering how a music signal gets compromised when it is
> transmitted via FM radio. Other than squash-it-to-death broadcast
> limiters, is there something about FM that reduces signal quality (less
> stereo separation, distortion, less bass, less dynamic range).
>
> just wondering what gets compromised and why...

Bandwidth to about 16kHz (in the UK, may be slightly different elsewhere)

Stereo tends to be noisy unless you are getting a good signal -
certainly always noisier than mono.

I don't know how liable the transmission and reception process are to
distortion.

There may be some odd frequency response artifacts if the FM signal is
band limited too much.

--
Anahata
anahata@treewind.co.uk -+- http://www.treewind.co.uk
Home: 01638 720444 Mob: 07976 263827
Related resources
Anonymous
August 13, 2005 10:06:13 PM

Archived from groups: rec.audio.pro (More info?)

genericaudioperson@hotmail.com wrote:

> I was wondering how a music signal gets compromised when it is
> transmitted via FM radio. Other than squash-it-to-death broadcast
> limiters, is there something about FM that reduces signal quality (less
> stereo separation, distortion, less bass, less dynamic range).
>
> just wondering what gets compromised and why...

The IF bandwidth in the rx is often limited to around 300Khz which causes
some distortion.

Transmitter audio bandwidth is limited to 15Khz to protect the pilot at 19K.

The Preemp can be a issue if you are driving the transmitter hard, but it IS
possible to have very good sounding FM radio, just not very LOUD at the
same time!

For compatability with mono receivers, the signal is transmitted as L+R in
the 30Hz->15Khz region, with L+R suppressed carrier modulated onto a
subcarrier at 38K. A 19K pilot is used to both signal the presence of a
stereo broadcast and to sync the 38Khz carrier at the rx end.

Now one of the properties of a band limited FM signal in a noisy channel is
that the noise spectrum of the demodulated signal rises with frequency.
Thus the recovered L-R signal has a much worse SNR under weak signal
conditions then the L+R (mono) signal does.

The rx can use this to degrade gracefully under weak signal conditions by
reducing stereo separation to maintain noise performance. This is good
engineering and makes for a more robust system, however if you want to be
heard on car radios (and in fact on mono portables), then you had better
make sure that your signal is mono compatible! Some modern music
(particularly some drum machines) put out out signals which while
essentially mono are inverted between the left and right outputs. Obviously
a mono rx (or one that has degraded to mono due to weak signal) will
completely fail to reproduce this bass.
Dance music producers, this means you need to watch the jellyfish meters!

FM in built up areas suffers from multipath distortion which is its major
weak point, good aerial positioning helps with this, and it is possible to
detect it by looking for ripple on the pilot.

HTH.

Regards, Dan.
Anonymous
August 13, 2005 10:23:45 PM

Archived from groups: rec.audio.pro (More info?)

On 13 Aug 2005 08:28:36 -0700, in rec.audio.pro
genericaudioperson@hotmail.com wrote:

>I was wondering how a music signal gets compromised when it is
>transmitted via FM radio. Other than squash-it-to-death broadcast
>limiters, is there something about FM that reduces signal quality (less
>stereo separation, distortion, less bass, less dynamic range).
>
>just wondering what gets compromised and why...
havent dug out my books, this is a vaguely correct summary

1) is FM deviation, normally 75Kc/s, this sets the limit the noise
floor in the Rx
2)HF premphasis, 75uS in US 50uSec in EU this limits how much signal
you can pump into the TX without exceed 75 Kc/s deviation. This gives
problems with the spectrum of modern music, which is substantially
topier than muisic in the 50's
3) stereo muxing, this robs some of the TX power from the Mono signal,
reducing Tx range
4) the FM pilot tone it at a low level (see 3), so it is noisy This
carries the S signal, (a-b)
5) an FM transmission should give good result, but the FM stereo mux
will reduce the noise in stereo by about 22dB, BUT not when listening
in Mono. This is a good analogue system
6) THD better than 0.2%. respone 20c/s to 15Kc/s. Noise, depends on
how you measure it ( ie cant remember) crosstalk, limited by the
encoder at the Tx, modern DSP designs should be good


martin
Anonymous
August 13, 2005 10:44:40 PM

Archived from groups: rec.audio.pro (More info?)

<genericaudioperson@hotmail.com> wrote:
>I was wondering how a music signal gets compromised when it is
>transmitted via FM radio. Other than squash-it-to-death broadcast
>limiters, is there something about FM that reduces signal quality (less
>stereo separation, distortion, less bass, less dynamic range).

For the most part, it's horribly mutilated. Most of the stations around
here compress so much that there is less than 6 dB peak/average.

The actual broadcast chain CAN be pretty impressive. 10 Hz to 20 KHz
response with more than 60 dB dynamic range (at full quieting) is easy
on a mono channel. With stereo you have to deal with the 19 KHz pilot
subcarrier, so you have to get brickwall filters around 18 KHz and
consequent phase shift issues. But mono FM can be really frighteningly
clean.

>just wondering what gets compromised and why...

Because it has to be loud. Listeners stop more often on stations that
are loud. The whole goal is to get listeners to stop long enough to
listen to a commercial.
--scott
--
"C'est un Nagra. C'est suisse, et tres, tres precis."
August 14, 2005 1:35:53 AM

Archived from groups: rec.audio.pro (More info?)

>
> >just wondering what gets compromised and why...
>
> Because it has to be loud. Listeners stop more often on stations that
> are loud. The whole goal is to get listeners to stop long enough to
> listen to a commercial.

This may have been true in the past with AM radios where listeners
would tune across the dial and stop when they heard osmething
"loud"....

but today, how many people tune across the dial.... radios are all
digitally controlled, you pick the station by number

I wish the broadcasters would get over it and understand that louder is
not better.

Mark
Anonymous
August 14, 2005 3:56:36 AM

Archived from groups: rec.audio.pro (More info?)

SSJVCmag wrote:
>WFMT made their
> NAME doing stellar quality classical broadcasts.


WFMT sounded breathtaking in the late 1960s, often significantly better
than the best vinyl

--
Bob Olhsson Audio Mastery, Nashville TN
Mastering, Audio for Picture, Mix Evaluation and Quality Control
Over 40 years making people sound better than they ever imagined!
615.385.8051 http://www.hyperback.com
Anonymous
August 14, 2005 3:56:37 AM

Archived from groups: rec.audio.pro (More info?)

"Bob Olhsson" <olh@hyperback.com> wrote in message
news:UMvLe.601927$cg1.349769@bgtnsc04-news.ops.worldnet.att.net...
> SSJVCmag wrote:
>>WFMT made their
>> NAME doing stellar quality classical broadcasts.
>
>
> WFMT sounded breathtaking in the late 1960s, often significantly better
> than the best vinyl

For the Chicago Symphony recordings I was told they ran 30 ips 2-track 1/4
inch. I was in their studios several times in the 60s, but I can't rember
what tape decks they used. Nor can I remember their chief engineer's name.
Mitch Heller, who was a first rate tech as well as recording engineer,
joined them in '69. I have a copy from the master of a Carl Sandburg
one-man program of poetry and song recorded by WFMT. This is at 15 ips, but
it is state-of-the-art sound for that time.

Steve King
Anonymous
August 14, 2005 4:54:16 AM

Archived from groups: rec.audio.pro (More info?)

genericaudioperson@hotmail.com wrote:

> I was wondering how a music signal gets compromised when it is
> transmitted via FM radio. Other than squash-it-to-death broadcast
> limiters, is there something about FM that reduces signal quality (less
> stereo separation, distortion, less bass, less dynamic range).
>
> just wondering what gets compromised and why...

The resulting audio will naturally be subject to noise and distortion
that's a function of received signal strength and signal path.

The receiver requires to do some weird stuff to demodulate the stereo
component that results in further signal degradation.

Bandwidth is limited to 15 kHz by design.

The result can never be better than the receiver though. Here's a typical
spec.

Total Harmonic Distortion
FM Mono 0.25%
FM Stereo 0.5%
Signal/noise ratio
Mono 60dB
Stereo 55dB
Stereo Separation at 1kHz 40dB
Frequency Response ±1.5dB 30Hz - 15kHz

http://www.nadelectronics.com/av_receivers/T773closerlo...

Graham
Anonymous
August 14, 2005 5:32:22 AM

Archived from groups: rec.audio.pro (More info?)

On 8/13/05 7:56 PM, in article
UMvLe.601927$cg1.349769@bgtnsc04-news.ops.worldnet.att.net, "Bob Olhsson"
<olh@hyperback.com> wrote:

> SSJVCmag wrote:
>> WFMT made their
>> NAME doing stellar quality classical broadcasts.
>
>
> WFMT sounded breathtaking in the late 1960s, often significantly better
> than the best vinyl

I rememebr an article showing folks that would record the broadcasts on 1/4"
1/2track stereo.
Anonymous
August 14, 2005 8:12:55 AM

Archived from groups: rec.audio.pro (More info?)

> The receiver requires to do some weird stuff to demodulate
> the stereo component that results in further signal degradation.

There's nothing weird about stereo demodulation. It's a product detector
(CMIIW), commonly used in fancy communications receivers.

By the way, most tuners do not demodulate the stereo component separately.
The left and right channels are directly demodulated. This system has been
in use for over 40 years.


> Total Harmonic Distortion
> FM Mono 0.25%
> FM Stereo 0.5%
> Signal/noise ratio
> Mono 60dB
> Stereo 55dB
> Stereo Separation at 1kHz 40dB
> Frequency Response ±1.5dB 30Hz - 15kHz

I don't know where you got those specs, but they'd have been average, if not
mediocre, 25 years ago. Good tuners show distortion levels one half to one
fifth that, and S/N ratios 10 to 20 dB better
Anonymous
August 14, 2005 9:18:17 AM

Archived from groups: rec.audio.pro (More info?)

On Sat, 13 Aug 2005 18:23:45 +0200, martin griffith
<martingriffith@XXyahoo.co.uk> wrote:

>4) the FM pilot tone it at a low level (see 3), so it is noisy This
>carries the S signal, (a-b)

Of course you don't mean this literally, but it might confuse
the OP as-is.

>5) an FM transmission should give good result, but the FM stereo mux
>will reduce the noise in stereo by about 22dB, BUT not when listening
>in Mono. This is a good analogue system

Stereo is noisier than mono through a range of signal strengths,
then at some large signal strength, they're equal because
limited by equipment residuals.

>6) THD better than 0.2%.

And potentially very much better. The McIntosh MR78 when
properly tweaked can get well below my measurement limit
of .05% midband. Some later things are reputedly even
better.

>modern DSP designs should be good

A very interesting topic, indeed. Thanks,

Chris Hornbeck
Anonymous
August 14, 2005 10:27:41 AM

Archived from groups: rec.audio.pro (More info?)

"Bob Olhsson" <olh@hyperback.com> wrote in message
news:UMvLe.601927$cg1.349769@bgtnsc04-news.ops.worldnet.att.net
> SSJVCmag wrote:
>> WFMT made their
>> NAME doing stellar quality classical broadcasts.
>
>
> WFMT sounded breathtaking in the late 1960s, often
> significantly better than the best vinyl.

One non-audio advantage WFMT had is that you could hear
their work product only once, while you could play the LP
many times until you heard all the flaws.
Anonymous
August 14, 2005 11:52:28 AM

Archived from groups: rec.audio.pro (More info?)

On Sat, 13 Aug 2005 11:28:36 -0400, genericaudioperson@hotmail.com wrote
(in article <1123946916.208492.116650@g44g2000cwa.googlegroups.com>):

> I was wondering how a music signal gets compromised when it is
> transmitted via FM radio. Other than squash-it-to-death broadcast
> limiters, is there something about FM that reduces signal quality (less
> stereo separation, distortion, less bass, less dynamic range).
>
> just wondering what gets compromised and why...
>

Nothing over 15 kHz,

Stereo separation compromised depending on your radio and how well you're
receiving the signal. Some radios "blend" all the way to mono if they can't
get a strong lock on the signal.

The less dynamic range thing is all about how the processors are set up at
the studio or transmission site.

Some FMs process at the studio, then use an RF STL (Studio Transmitter Link)
to get thw audio to the transmitter, More stuff can happen there.

Distortion, yes.
Bass, yes and no.

It's all to keep it legally as loud as possible and get it from here to
there.


Regards,

Ty ford



-- Ty Ford's equipment reviews, audio samples, rates and other audiocentric
stuff are at www.tyford.com
Anonymous
August 14, 2005 2:35:49 PM

Archived from groups: rec.audio.pro (More info?)

In article <y8CdncRAc4FCXGPfRVn-1g@comcast.com> steveSPAMBLOCK@stevekingSPAMBLOCK.net writes:

> > WFMT sounded breathtaking in the late 1960s

> For the Chicago Symphony recordings I was told they ran 30 ips 2-track 1/4
> inch.

I would doubt that. Most broadcast recording was done at 7.5 ips,
2-track on 1/4" tape. If they really cared about quality (as
apparently WFMT did, for their symphony broadcasts) it was likely that
they used 15 ips, probably on an Ampex 350. 30 ips wouldn't give them
enough recording time for a symphony without piecing two reels
together, and 14" reels weren't all that common.

Ed Green (of the famous live recording truck) used to record an
orchestra in the National Gallery of Art in DC every Sunday night for
broadcast. He used a pair of Sony C37 mics and an Ampex. Sounded fine
then (though I was listening on a table radio at the time) and when he
played tapes of those concerts at one of the AES historical
demonstrations a few years back, they sounded fine then, too.


--
I'm really Mike Rivers (mrivers@d-and-d.com)
However, until the spam goes away or Hell freezes over,
lots of IP addresses are blocked from this system. If
you e-mail me and it bounces, use your secret decoder ring
and reach me here: double-m-eleven-double-zero at yahoo
Anonymous
August 14, 2005 2:35:50 PM

Archived from groups: rec.audio.pro (More info?)

In article <itCdnTfZT58AgWLfRVn-pA@comcast.com> arnyk@hotpop.com writes:

> One non-audio advantage WFMT had is that you could hear
> their work product only once, while you could play the LP
> many times until you heard all the flaws.

You could always record the broadcast off the air, and I'm sure some
people did. Some may have even had decent recorders and FM tuners.

There was a station in Boston who, back in the day of the PCM
processor, broadcast a concert series late Sunday nights in
cooperation with their TV station. They broadcast the PCM video signal
so that people with F1 series processors could get the digital
recording off the TV broadcast. Not much to look at, but an
interesting experiment.


--
I'm really Mike Rivers (mrivers@d-and-d.com)
However, until the spam goes away or Hell freezes over,
lots of IP addresses are blocked from this system. If
you e-mail me and it bounces, use your secret decoder ring
and reach me here: double-m-eleven-double-zero at yahoo
Anonymous
August 14, 2005 5:09:17 PM

Archived from groups: rec.audio.pro (More info?)

"Mike Rivers" <mrivers@d-and-d.com> wrote in message
news:znr1124017603k@trad...
>
> In article <y8CdncRAc4FCXGPfRVn-1g@comcast.com>
> steveSPAMBLOCK@stevekingSPAMBLOCK.net writes:
>
>> > WFMT sounded breathtaking in the late 1960s
>
>> For the Chicago Symphony recordings I was told they ran 30 ips 2-track
>> 1/4
>> inch.
>
> I would doubt that. Most broadcast recording was done at 7.5 ips,
> 2-track on 1/4" tape. If they really cared about quality (as
> apparently WFMT did, for their symphony broadcasts) it was likely that
> they used 15 ips, probably on an Ampex 350. 30 ips wouldn't give them
> enough recording time for a symphony without piecing two reels
> together, and 14" reels weren't all that common.
>

You might be right. It was a long time ago. However, Bernie Jacobs, the
then owner, ran WFMT as a no-holds-barred state of the art operation. They
didn't do much of anything like "most broadcasting".

Steve King
Anonymous
August 14, 2005 7:12:27 PM

Archived from groups: rec.audio.pro (More info?)

"Pooh Bear"
>
>> I was wondering how a music signal gets compromised when it is
>> transmitted via FM radio. Other than squash-it-to-death broadcast
>> limiters, is there something about FM that reduces signal quality (less
>> stereo separation, distortion, less bass, less dynamic range).
>>
>> just wondering what gets compromised and why...
>
> The resulting audio will naturally be subject to noise and distortion
> that's a function of received signal strength and signal path.


** But nothing like the problems faced by AM radio.


> The receiver requires to do some weird stuff to demodulate the stereo
> component that results in further signal degradation.


** Weird stuff = switch the audio between L and R outs at 38 kHz ???

> Bandwidth is limited to 15 kHz by design.


** The practical limit of hearing on programme for 95 % of people.

> The result can never be better than the receiver though. Here's a typical
> spec.


** This is actually a VERY poor spec:


> Total Harmonic Distortion
> FM Mono 0.25%
> FM Stereo 0.5%
> Signal/noise ratio
> Mono 60dB
> Stereo 55dB
> Stereo Separation at 1kHz 40dB
> Frequency Response ±1.5dB 30Hz - 15kHz
>


** In the early 1980s, the most popular ICs used to demodulate FM and
decode the stereo signal were the LM 3189 and the LM1800.

The LM3189 is speced as having as 80 dB s/n for mono with THD of 0.1% at
full 75 kHz deviation.

The LM1800 is speced as having a stereo separation of 45 dB and a THD of 0.1
% at full level.

The THD percentages all reduce at lower levels - ie 95 % of the listening
time.

In stereo mode, with a good RF signal, the audio s/n obtained is about 70
dB.

A decent FM tuner is a true hi-fi device with sound quality that rivals the
CD format.

Only a few FM stations that feature classical music or are run on an amateur
basis take advantage of the performance that is easily possible.




............ Phil
Anonymous
August 14, 2005 7:12:28 PM

Archived from groups: rec.audio.pro (More info?)

On Sun, 14 Aug 2005 15:12:27 +1000, "Phil Allison"
<philallison@tpg.com.au> wrote:

>A decent FM tuner is a true hi-fi device with sound quality that rivals the
>CD format.
>
>Only a few FM stations that feature classical music or are run on an amateur
>basis take advantage of the performance that is easily possible.

Sadly, this is possibly even more true in the US, where
very large corporations control national markets.

Fortunately, their's still (knock wood) NPR and, as you
say, the local amateur (who's name comes from "doing it
for the love of it") stations.

Thanks,

Chris Hornbeck
Anonymous
August 14, 2005 7:12:28 PM

Archived from groups: rec.audio.pro (More info?)

"Phil Allison" <philallison@tpg.com.au> wrote in
news:3m825qF15inu7U1@individual.net:

>
> "Pooh Bear"
>>
>
>> The receiver requires to do some weird stuff to demodulate
>> the stereo component that results in further signal
>> degradation.
>
>
> ** Weird stuff = switch the audio between L and R outs at
> 38 kHz ???
>
It's a little more complicated than that, Phil. Audio in a FM
signal is sent as a main carrier as L+R, and a subcarrier of L-R
at 38 KHz. You don't switch between the left and right, you mix
the sum with the uninverted difference to get left, and the sum
and an inverted difference to get the right channel. the
performance issues arise from doubling the 19KHz pilot tone to
get the LO frequency to bring the difference signal back to
baseband.

> A decent FM tuner is a true hi-fi device with sound quality
> that rivals the CD format.

I agree.
>
> Only a few FM stations that feature classical music or are run
> on an amateur basis take advantage of the performance that is
> easily possible.
>
Which is a crying shame.

>
> ........... Phil
>

--
Bob Quintal

PA is y I've altered my email address.
Anonymous
August 14, 2005 7:12:28 PM

Archived from groups: rec.audio.pro (More info?)

Phil Allison wrote:

> A decent FM tuner is a true hi-fi device with sound quality that rivals the
> CD format.

Suggestions for one with good sensitivity? It is getting
hard to find tuners without a bunch of other junk integrated
with it. Digital tuning with presets would be nice but I
haven't been able to find such a simple thing.


Bob
--

"Things should be described as simply as possible, but no
simpler."

A. Einstein
Anonymous
August 14, 2005 7:12:29 PM

Archived from groups: rec.audio.pro (More info?)

>> ** Weird stuff = switch the audio between
>> L and R outs at 38 kHz ???

> It's a little more complicated than that, Phil. Audio in a FM
> signal is sent as a main carrier as L+R, and a subcarrier of L-R
> at 38 KHz. You don't switch between the left and right, you mix
> the sum with the uninverted difference to get left, and the sum
> and an inverted difference to get the right channel.

Actually, "switching" between the channels is a mathematically valid way of
analyzing the modulation. Read a book on sampling theory.

Note the connection with color TV... RCA originally used a dot-sequential
system that rapidly switched among red, green, and blue. It was then pointed
out that the net signal generated was equivalent to transmitting the color
signals continuously on a carrier whose frequency equalled the switching
rate.
Anonymous
August 14, 2005 7:12:29 PM

Archived from groups: rec.audio.pro (More info?)

> Suggestions for one with good sensitivity? It is getting
> hard to find tuners without a bunch of other junk integrated
> with it. Digital tuning with presets would be nice but I
> haven't been able to find such a simple thing.


If I might repeat what experts were saying 30+ years ago... Sensitivity is
the least-important spec, unless you're living a weak-signal area and have
to put up a huge 23-element Yagi to pull in the stations you want.

I'd look at the distortion and S/N specs first. Then the usual RF specs,
like adjacent-channel selectivity, image rejection, etc.
Anonymous
August 14, 2005 7:55:31 PM

Archived from groups: rec.audio.pro (More info?)

"martin griffith"

> 1) is FM deviation, normally 75Kc/s, this sets the limit the noise
> floor in the Rx


** Depends on the FN detector sues - but -80 dB is easily reached.


> 2)HF premphasis, 75uS in US 50uSec in EU this limits how much signal
> you can pump into the TX without exceed 75 Kc/s deviation.


** In practice it has no such effect.


>This gives
> problems with the spectrum of modern music, which is substantially
> topier than muisic in the 50's


** With the exception of hyper-compressed thrash bands - that is horse
poo.


> 3) stereo muxing, this robs some of the TX power from the Mono signal,
> reducing Tx range


** Not true.


> 4) the FM pilot tone it at a low level (see 3), so it is noisy This
> carries the S signal, (a-b)


** Complete bullshit.

The 19kHz pilot is just a tone and carries no programme information.

The L-R signal from the FM detector is in the supersonic range at high
level.


> 5) an FM transmission should give good result, but the FM stereo mux
> will reduce the noise in stereo by about 22dB,


** Under weak signal conditions - maybe.


> 6) THD better than 0.2%. respone 20c/s to 15Kc/s. Noise, depends on
> how you measure it ( ie cant remember) crosstalk, limited by the
> encoder at the Tx,


** Very good stereo reproduction via speakers only needs circa 15 dB
channel separation.

Separation of 40 dB is typical for FM radio.




........... Phil
Anonymous
August 14, 2005 7:55:32 PM

Archived from groups: rec.audio.pro (More info?)

>> 3) stereo muxing, this robs some of the TX power from
>> the Mono signal, reducing Tx range

> ** Not true.

True, Phil. Sorry about that.

The pilot and subcarrier add to the total deviation, thus forcing a slight
reduction of the main signal's deviation.
Anonymous
August 14, 2005 8:56:42 PM

Archived from groups: rec.audio.pro (More info?)

"Mike Rivers" <mrivers@d-and-d.com> wrote in message
news:znr1124017832k@trad
> In article <itCdnTfZT58AgWLfRVn-pA@comcast.com>
> arnyk@hotpop.com writes:
>
>> One non-audio advantage WFMT had is that you could hear
>> their work product only once, while you could play the LP
>> many times until you heard all the flaws.
>
> You could always record the broadcast off the air, and
> I'm sure some people did. Some may have even had decent
> recorders and FM tuners.

All that was avaialble in the 60s were analog tape
recorders. And by the standards of analog tape recorders and
media of the 80s, they weren't all that good. Furthermore
many of us knew it so that if a tape of a broadcast sounded
a little down in the mouth, we blamed it on the taped
transcription.

> There was a station in Boston who, back in the day of the
> PCM processor, broadcast a concert series late Sunday
> nights in cooperation with their TV station. They
> broadcast the PCM
> video signal so that people with F1 series processors
> could get the digital recording off the TV broadcast. Not
> much to look at, but an interesting experiment.

Interesting, but a few miles ahead of the
generally-available technology of the day, or AFAIK even
today. For example the British DAB sequel to FM is AFAIK
perceptually coded.
Anonymous
August 14, 2005 8:56:43 PM

Archived from groups: rec.audio.pro (More info?)

> All that was avaialble in the 60s were analog tape recorders.
> And by the standards of analog tape recorders and media of
> 80s, they weren't all that good. Furthermore, many of us
> knew it so that if a tape of a broadcast sounded a little down
> in the mouth, we blamed it on the taped transcription.

Arny, you're the mirror image of people who adore vinyl -- you have almost
nothing good to say about analog recording. You probably have a photo of
"Fluffy" above your desk, intoning "All else is gaslight."

I agree that analog recording made significant strides in the two-decade
interval you mention. But pro tape recorders of the 60s weren't exactly
chopped liver. Many great recordings were made on them, and even larger
numbers of putrid recordings were made on the better recorders of a decade
or two later. The medium is important, but not nearly so important as how
you use it.

I, too, used to feel I could hear a subtle loss of quality -- mostly
"aliveness" or "immediacy" in recordings. But I didn't run double-blind
tests, so there was no way to prove it. (Yes, Arny, I'm being sarcastic at
your expense.)


> Interesting, but a few miles ahead of the generally-available
> technology of the day, or AFAIK even today. For example,
> the British DAB sequel to FM is AFAIK perceptually coded.

It would /have/ to be perceptually coded, because there's no way to squeeze
CD-format digital sound into a 150kHz channel.

Several Seattle FM stations are using some digital system (not DAB) hidden
in the same channel, along with the regular analog signal. Most are PBS; one
is classical. The latter claimed "CD-quality" sound, and I asked them not to
make that claim, but they declined.
Anonymous
August 14, 2005 9:01:04 PM

Archived from groups: rec.audio.pro (More info?)

"Phil Allison" <philallison@tpg.com.au> wrote in message
news:3m84miF15hi9nU1@individual.net


>> 3) stereo muxing, this robs some of the TX power from
>> the Mono signal, reducing Tx range

> ** Not true.

True. There's a ton of energy in the added carriers,
subcarriers, etc.

>> 4) the FM pilot tone it at a low level (see 3), so it is
>> noisy This carries the S signal, (a-b)
>
>
> ** Complete bullshit.

> The 19kHz pilot is just a tone and carries no programme
> information.

I think its at 10% modulation. Not a lot loss of max
deviation, but finite and significant.

> The L-R signal from the FM detector is in the supersonic
> range at high level.

....the stereo subcarrier sops up deviation very nicely think
you.

http://members.tripod.com/~transmitters/stereo.htm

A Stereo transmission tends to sound noisier that a Mono
signal. This is mainly caused by the noise in the L-R
channel. As the noise in the 23 KHz to 53 KHz segment is
also brought down to the audible 0-15 KHz region by the
decoding process, we now have more noise than receiving the
same signal in Mono. The decoder circuits in the receiver
also contribute extra noise. On top of that, as MPX signals
have more bandwidth than a Mono signal, the station has to
use less modulation with MPX to remain in the deviation
limits. All above tend to increase the noise. (Visit USENET
message link below for more info)

>The last step is to add in about 10% of the 19 KHz Pilot
>Tone. The MPX signals would look quite the same as the ones
>shown above; the Harmonic analyses would show a blip at the
>19 KHz point. As an MPX signal is not really a simple Audio
>signal, you cannot Compress/Limit/Equalise/Pre-emphasise an
>MPX signal. All the audio processing should have been done
>before the MPX process. All you can do now is feed the MPX
>signal to the transmitter instead of the Mono Audio signal
>it had before,
Anonymous
August 14, 2005 9:03:26 PM

Archived from groups: rec.audio.pro (More info?)

"Bob Quintal" <rquintal@sPAmpatico.ca> wrote in message
news:1124029008.6a4884b0867a39bd55ac7e9b73fef389@teranews
> "Phil Allison" <philallison@tpg.com.au> wrote in
> news:3m825qF15inu7U1@individual.net:
>
>>
>> "Pooh Bear"
>>>
>>
>>> The receiver requires to do some weird stuff to
>>> demodulate the stereo component that results in further
>>> signal degradation.
>>
>>
>> ** Weird stuff = switch the audio between L and R
>> outs at 38 kHz ???
>>
> It's a little more complicated than that, Phil. Audio in
> a FM signal is sent as a main carrier as L+R, and a
> subcarrier of L-R at 38 KHz. You don't switch between the
> left and right, you mix the sum with the uninverted
> difference to get left, and the sum and an inverted
> difference to get the right channel. the performance
> issues arise from doubling the 19KHz pilot tone to get
> the LO frequency to bring the difference signal back to
> baseband.

This paper explores how this is analogous to simply
switching the audio at 38 JHz:

http://members.tripod.com/~transmitters/stereo.htm

"It can however be shown ( not by me though ;)  that rapidly
switching between L and R channels at 38 KHz does most of
the hard work and is a near equivalent to the hard way.
Switching channels as above makes a L+R signal and a DSBSC
L-R channel centered around 38 KHz. It also generates a lot
of harmonics but they can be filtered away. All that is left
is to add the Pilot Tone. Maybe the FM MPX standards were
actually decided upon the other way around, the ease of the
switching method lead to it being accepted as the standard ?
Rapidly sampling L and R signals alternatively is called
Time Division Multiplexing (TDM)."

etc.
Anonymous
August 14, 2005 9:24:11 PM

Archived from groups: rec.audio.pro (More info?)

On Sun, 14 Aug 2005 15:55:31 +1000, in rec.audio.pro "Phil Allison"
<philallison@tpg.com.au> wrote:

>
>"martin griffith"
>
>> 1) is FM deviation, normally 75Kc/s, this sets the limit the noise
>> floor in the Rx
>
>
>** Depends on the FN detector sues - but -80 dB is easily reached.
>
>
>> 2)HF premphasis, 75uS in US 50uSec in EU this limits how much signal
>> you can pump into the TX without exceed 75 Kc/s deviation.
>
>
>** In practice it has no such effect.
Sorry, not true, we did some on air experiments with Dolby Labs with
DolbyFM a long time ago. This effectively reduced the premph to 25uS
and DolbyB'd the S signal. This gave us approx 20miles extra radius on
our coverage.

The sales department didn't give a damn about the tech stuff, but this
20 miles increase is not insignificant in terms of hard cash
>
>>This gives
>> problems with the spectrum of modern music, which is substantially
>> topier than muisic in the 50's
>
>
>** With the exception of hyper-compressed thrash bands - that is horse
>poo.
>
>
>> 3) stereo muxing, this robs some of the TX power from the Mono signal,
>> reducing Tx range
>
>
>** Not true.
>
>
>> 4) the FM pilot tone it at a low level (see 3), so it is noisy This
>> carries the S signal, (a-b)
>
>
>** Complete bullshit.
>
>The 19kHz pilot is just a tone and carries no programme information.
>
>The L-R signal from the FM detector is in the supersonic range at high
>level.
>
>
>> 5) an FM transmission should give good result, but the FM stereo mux
>> will reduce the noise in stereo by about 22dB,
>
>
> ** Under weak signal conditions - maybe.
>
>
>> 6) THD better than 0.2%. respone 20c/s to 15Kc/s. Noise, depends on
>> how you measure it ( ie cant remember) crosstalk, limited by the
>> encoder at the Tx,
>
>
>** Very good stereo reproduction via speakers only needs circa 15 dB
>channel separation.
>
> Separation of 40 dB is typical for FM radio.
>
>
>
>
>.......... Phil
>
>
>



martin
Anonymous
August 14, 2005 11:26:43 PM

Archived from groups: rec.audio.pro (More info?)

Mike Rivers wrote:

> In article <y8CdncRAc4FCXGPfRVn-1g@comcast.com> steveSPAMBLOCK@stevekingSPAMBLOCK.net writes:
>
> > > WFMT sounded breathtaking in the late 1960s
>
> > For the Chicago Symphony recordings I was told they ran 30 ips 2-track 1/4
> > inch.
>
> I would doubt that. Most broadcast recording was done at 7.5 ips,
> 2-track on 1/4" tape. If they really cared about quality (as
> apparently WFMT did, for their symphony broadcasts) it was likely that
> they used 15 ips, probably on an Ampex 350. 30 ips wouldn't give them
> enough recording time for a symphony without piecing two reels
> together, and 14" reels weren't all that common.

Recorders that can take 14" reels are pretty rare too. 350s only took 10-1/2" reels max IIRC.

Graham
Anonymous
August 14, 2005 11:30:54 PM

Archived from groups: rec.audio.pro (More info?)

Steve King wrote:

> "Mike Rivers" <mrivers@d-and-d.com> wrote in message
> news:znr1124017603k@trad...
> >
> > In article <y8CdncRAc4FCXGPfRVn-1g@comcast.com>
> > steveSPAMBLOCK@stevekingSPAMBLOCK.net writes:
> >
> >> > WFMT sounded breathtaking in the late 1960s
> >
> >> For the Chicago Symphony recordings I was told they ran 30 ips 2-track
> >> 1/4
> >> inch.
> >
> > I would doubt that. Most broadcast recording was done at 7.5 ips,
> > 2-track on 1/4" tape. If they really cared about quality (as
> > apparently WFMT did, for their symphony broadcasts) it was likely that
> > they used 15 ips, probably on an Ampex 350. 30 ips wouldn't give them
> > enough recording time for a symphony without piecing two reels
> > together, and 14" reels weren't all that common.
>
>
> You might be right. It was a long time ago. However, Bernie Jacobs, the
> then owner, ran WFMT as a no-holds-barred state of the art operation. They
> didn't do much of anything like "most broadcasting".

How long ago was this ?

15 ips 2 track 1/4" used to be the 'gold standard' for a long time until the
mid/late 70s when 30 ips 1/2" took stereo mastering to a new level.

Graham
Anonymous
August 15, 2005 12:41:50 AM

Archived from groups: rec.audio.pro (More info?)

"Pooh Bear" <rabbitsfriendsandrelations@hotmail.com> wrote in message
news:42FF8DDE.41C67FE6@hotmail.com...
> Steve King wrote:
>
>> "Mike Rivers" <mrivers@d-and-d.com> wrote in message
>> news:znr1124017603k@trad...
>> >
>> > In article <y8CdncRAc4FCXGPfRVn-1g@comcast.com>
>> > steveSPAMBLOCK@stevekingSPAMBLOCK.net writes:
>> >
>> >> > WFMT sounded breathtaking in the late 1960s
>> >
>> >> For the Chicago Symphony recordings I was told they ran 30 ips 2-track
>> >> 1/4
>> >> inch.
>> >
>> > I would doubt that. Most broadcast recording was done at 7.5 ips,
>> > 2-track on 1/4" tape. If they really cared about quality (as
>> > apparently WFMT did, for their symphony broadcasts) it was likely that
>> > they used 15 ips, probably on an Ampex 350. 30 ips wouldn't give them
>> > enough recording time for a symphony without piecing two reels
>> > together, and 14" reels weren't all that common.
>>
>>
>> You might be right. It was a long time ago. However, Bernie Jacobs, the
>> then owner, ran WFMT as a no-holds-barred state of the art operation.
>> They
>> didn't do much of anything like "most broadcasting".
>
> How long ago was this ?
>
> 15 ips 2 track 1/4" used to be the 'gold standard' for a long time until
> the
> mid/late 70s when 30 ips 1/2" took stereo mastering to a new level.
>
> Graham

In 1965 Malcolm Chisholm was recording the Fine Arts Quartet using Ampex 300
decks with custom capstans he had made for 30 ips. The Ampex 300
electronics were also modified, since there was no NAB or RIAA standard for
30 ips. In addition Malcolm went through a lot of input stage tubes,
looking for the lowest possible noise. I recall a conversation about his
regret at the loss of extreme low freq. performance, but felt the gain (from
30 ips) in lower tape noise was worth it. I'm sure there were other mods as
well. I think that RCA was also experimenting with 30 ips at about the same
time, probably not with the same decks. I recall that Malcolm and the chief
eng. at WFMT were friends. As you say, half-inch 2-track for mastering came
much later. The Ampex 300 did handle 14" reels as I recall. I didn't work
much with those decks. They and the Magnecord tape destroyers were on the
way out as I and the model 350 were on the way in. The first studio I
worked for had a couple, which were used in the tape duplication room.
Having had to move those a few times, I was convinced that the model name,
300, came about when they weighed the thing. Can you imagine taking one of
those on a remote?

Steve King
Anonymous
August 15, 2005 1:47:46 AM

Archived from groups: rec.audio.pro (More info?)

"Bill Summawank"
>>
>>> 3) stereo muxing, this robs some of the TX power from
>>> the Mono signal, reducing Tx range
>
>> ** Not true.
>
> True, Phil. Sorry about that.
>
> The pilot and subcarrier add to the total deviation, thus forcing a slight
> reduction of the main signal's deviation.


** But that has no effect on " Tx range ".




........... Phil
Anonymous
August 15, 2005 1:47:47 AM

Archived from groups: rec.audio.pro (More info?)

"Phil Allison" <philallison@tpg.com.au> wrote:
|>
|> The pilot and subcarrier add to the total deviation, thus forcing a slight
|> reduction of the main signal's deviation.
|
|
|** But that has no effect on " Tx range ".

A reduction in main signal level causes a like increase in SNR.
Therefore, for a given SNR, this effectively reduces the "TX range"

Phil
Anonymous
August 15, 2005 1:47:47 AM

Archived from groups: rec.audio.pro (More info?)

>> The pilot and subcarrier add to the total deviation, thus forcing
>> a slight reduction of the main signal's deviation.

> ** But that has no effect on " Tx range ".

But isn't the Tx range established in practice by determining how far away
an acceptable signal can be received (ie, adequate quieting)? Anything that
reduces the overall modulation is therefore going to reduce the range.

The pilot is limited to about 10% of peak deviation (ie, it's 20dB below
peak deviation), so I would guess the net loss of S/N from a compensating
reduction of the main program would be less than 1dB. I don't know how much
the subcarrier (or more precisely, the subcarrier sidebands, as there is no
subcarrier, per se) reduces main program modulation.
Anonymous
August 15, 2005 3:27:25 AM

Archived from groups: rec.audio.pro (More info?)

<pgx@pgrahams.com>
> "Phil Allison"
> |>
> |> The pilot and subcarrier add to the total deviation, thus forcing a
> slight
> |> reduction of the main signal's deviation.
> |
> |
> |** But that has no effect on " Tx range ".
>
> A reduction in main signal level causes a like increase in SNR.


** The noise level at the receiver depends only on the *carrier's signal
strength* at its location and then only when it has fallen below a certain
( very low) threshold.

> Therefore, for a given SNR, this effectively reduces the "TX range"


** Nonsense.



......... Phil
Anonymous
August 15, 2005 4:42:46 AM

Archived from groups: rec.audio.pro (More info?)

"Bob Quintal"
> "Phil Allison"
>> "Pooh Bear"
>>
>>> The receiver requires to do some weird stuff to demodulate
>>> the stereo component that results in further signal
>>> degradation.
>>
>>
>> ** Weird stuff = switch the audio between L and R outs at
>> 38 kHz ???
>>
>
> It's a little more complicated than that, Phil.


** Not much really.

FM stereo decoders have mostly worked by synchronous switching since the mid
1970s.



> Audio in a FM
> signal is sent as a main carrier as L+R, and a subcarrier of L-R
> at 38 KHz.


** Actually, the modulation consists of a baseband L+R signal, then a 19
kHz pilot tone at low level, then the L-R signal is added as double sideband
signal centered on a fully suppressed carrier of exactly 38 kHz.


> You don't switch between the left and right,


** Believe it or not - that is exactly how the composite signal I just
described is separated into L and R.


> you mix
> the sum with the uninverted difference to get left, and the sum
> and an inverted difference to get the right channel.


** In the tube days and early transistor days that was indeed so - but
then the switching decoder was developed and National Semiconductor released
their famous LM1310 decoder IC.


> the performance issues arise from doubling the 19KHz pilot tone to
> get the LO frequency to bring the difference signal back to
> baseband.


** The LM1310 used a PLL to phase lock the 38 kHz switching frequency to the
incoming 19 kHz tone - then the composite signal is then steered to L and R
outputs.

Finer points include correction for the fact the L-R signal is slightly
lower in level than the L+R and there is pre-emphasis.

Usually a sharp LP filter comes after the IC to remove the 38 kHz and higher
switching components.



........... Phil
Anonymous
August 15, 2005 4:48:56 AM

Archived from groups: rec.audio.pro (More info?)

"Will Summawank"

>>> The pilot and subcarrier add to the total deviation, thus forcing
>>> a slight reduction of the main signal's deviation.
>
>> ** But that has no effect on " Tx range ".
>
> But isn't the Tx range established in practice by determining how far away
> an acceptable signal can be received (ie, adequate quieting)?


** "Quieting" depends on carrier strength alone.



........... Phil
Anonymous
August 15, 2005 4:48:57 AM

Archived from groups: rec.audio.pro (More info?)

>> But isn't the Tx range established in practice by determining how far
>> away an acceptable signal can be received (ie, adequate quieting)?

> ** "Quieting" depends on carrier strength alone.

But if I reduce the modulation level, the signal is necessarily not as far
above the noise. Hence the S/N ratio drops.
Anonymous
August 15, 2005 5:33:47 AM

Archived from groups: rec.audio.pro (More info?)

"martin griffith"
"Phil Allison"
>
>>> 2)HF premphasis, 75uS in US 50uSec in EU this limits how much signal
>>> you can pump into the TX without exceed 75 Kc/s deviation.
>>
>>
>>** In practice it has no such effect.
>
> Sorry,


** You are not one bit sorry.


> not true, we did some on air experiments with Dolby Labs with
> DolbyFM a long time ago. This effectively reduced the premph to 25uS
> and DolbyB'd the S signal. This gave us approx 20miles extra radius on
> our coverage.


** The Dolby B compression extended the range over which good quality could
be had - you ass.

Sweet FA to do with the change in emphasis curve.


All the other points you ignored are conceded to me as well.



............. Phil
Anonymous
August 15, 2005 8:53:33 AM

Archived from groups: rec.audio.pro (More info?)

On Sun, 14 Aug 2005 13:33:47 -0700, "William Sommerwerck"
<gizzledgeezer@comcast.net> wrote:

>> Suggestions for one with good sensitivity? It is getting
>> hard to find tuners without a bunch of other junk integrated
>> with it. Digital tuning with presets would be nice but I
>> haven't been able to find such a simple thing.
>
>
>If I might repeat what experts were saying 30+ years ago... Sensitivity is
>the least-important spec, unless you're living a weak-signal area and have
>to put up a huge 23-element Yagi to pull in the stations you want.

And even then, the issue of interest should be what RF folks
call "noise figure", which is how much the measured device
degrades the carrier to noise ratio. This varies only very
slightly among good (note waffle injection) modern tuners.

Sensitivity as commonly specified compares unusably weak
signals, and capture ratio is concerned with even weaker
signals (mostly), because noise figure for modern devices
is quite low.

>I'd look at the distortion and S/N specs first. Then the usual RF specs,
>like adjacent-channel selectivity, image rejection, etc.

Tuners, like antennas, are location-specific. If you live in
New York and want to receive your favorite Boston station
despite a strong *adjacent* (200KHz away) local, you just
convince your friend at McIntosh to build you the MR78.

But the first step is antennas, antennas, antennas. No
grief about Latin spellings please. Like initial tracking
in the recording world, some things can't be fixed in the mix.

If you live in the sticks like me, a ribbon dipole on the
floor pulls in the NPR station at the end on my street just
fine. If you need to "DX" 75 miles you have an antenna
issue. If you live in Gotham City you have an antenna issue.


Oh yeah, the FM pilot is at 9%. Everybody remembered pretty good.

Good fortune,

ps: I gave up on your screen Goddess puzzler. Hints?

Chris Hornbeck
Anonymous
August 15, 2005 9:38:06 AM

Archived from groups: rec.audio.pro (More info?)

In article <9bOdnbJaJNxrN2LfRVn-2w@comcast.com> gizzledgeezer@comcast.net writes:

> If I might repeat what experts were saying 30+ years ago... Sensitivity is
> the least-important spec, unless you're living a weak-signal area and have
> to put up a huge 23-element Yagi to pull in the stations you want.

With a new receiver that I bought earlier this year, I couldn't pull
in either of our two local fairly powerful NPR stations with a dipole,
stations that came in fine on the 20+ year old receiver that I needed
to replace. When I asked what the "state" was here, several people
told me that they just weren't making receivers with decent front end
sensitivity any more because manufacturers assumed that they'd be
connected to a cable system. This wasn't a very expensive receiver,
but still you'd think it would pick up local stations pretty well.

On advice of someone, maybe it was Kurt, I bought a Tivoli radio and
that works just fine. Hardly high fidelity in its native form, and not
stereo, but fine for listening to the radio in the shop.


--
I'm really Mike Rivers (mrivers@d-and-d.com)
However, until the spam goes away or Hell freezes over,
lots of IP addresses are blocked from this system. If
you e-mail me and it bounces, use your secret decoder ring
and reach me here: double-m-eleven-double-zero at yahoo
Anonymous
August 15, 2005 10:36:01 AM

Archived from groups: rec.audio.pro (More info?)

"William Sommerwerck" <gizzledgeezer@comcast.net> wrote in
message news:9aydnRGHDsBeT2LfRVn-hA@comcast.com
>> All that was avaialble in the 60s were analog tape
>> recorders. And by the standards of analog tape recorders
>> and media of 80s, they weren't all that good.
>> Furthermore, many of us
>> knew it so that if a tape of a broadcast sounded a
>> little down in the mouth, we blamed it on the taped
>> transcription.
>
> Arny, you're the mirror image of people who adore vinyl
> -- you have almost nothing good to say about analog
> recording.

Something about being and audiophile who stuck with nothing
but analog recordings in the record store from 1959 to 1983,
and then waiting another 10-12 years for digital recording
to become good enough and inexpensive enough.

>You probably have a photo of "Fluffy" above
> your desk, intoning "All else is gaslight."

No.

> I agree that analog recording made significant strides in
> the two-decade interval you mention. But pro tape
> recorders of the 60s weren't exactly chopped liver. Many
> great recordings were made on them, and even larger
> numbers of putrid recordings were made on the better
> recorders of a decade or two later. The medium is
> important, but not nearly so important as how you use it.

I agree with the idea that the correct use of the tool at
hand is paramount, but I also recall my experinces with
trying to make sonically-transparent recordings in the day
of.

> I, too, used to feel I could hear a subtle loss of
> quality -- mostly "aliveness" or "immediacy" in
> recordings. But I didn't run double-blind tests, so there
> was no way to prove it. (Yes, Arny, I'm being sarcastic
> at your expense.)

It was very apparent to me in sighted evaluations at the
time, and when I went back and verified those results in
DBTs I found that I wasn't imagining things.

>> Interesting, but a few miles ahead of the
>> generally-available technology of the day, or AFAIK even
>> today. For example,
>> the British DAB sequel to FM is AFAIK perceptually coded.

> It would /have/ to be perceptually coded, because there's
> no way to squeeze CD-format digital sound into a 150kHz
> channel.

I'm not sure that is absolutely true that there's no way
to squeeze CD-format digital sound into a 150kHz
channel.

It takes about 1.42 megabits per second to do true CD
quality. I suspect that with a sophisticated modem, 1.42
megabits could probably be funneled through 150 KHz
bandwidth. After all we pump 56 KB down analog telephone
lines with ca. 2 KHz bandwidth. Analog FM actually has at
least 53 KHz bandwidth.

No, the problem is that they are still trying stuff the
legacy analog signal through the same pipe.

> Several Seattle FM stations are using some digital system
> (not DAB) hidden in the same channel, along with the
> regular analog signal. Most are PBS; one is classical.
> The latter claimed "CD-quality" sound, and I asked them
> not to make that claim, but they declined.

Hype springs eternal. ;-)
Anonymous
August 15, 2005 5:46:58 PM

Archived from groups: rec.audio.pro (More info?)

"William Sommerwerck"
>
> It would /have/ to be perceptually coded, because there's no way to
> squeeze
> CD-format digital sound into a 150kHz channel.
>


** Shannon's theorem ( aka Shannon-Hartley Theorem) says it certainly can
be done.

See: http://en.wikipedia.org/wiki/Shannon-Hartley_theorem


Using the formula C = 0.332 x B x S/N

C = 1.41 Mb/s and B = 150kHz

Gives a needed S/N of 28.9 dB.




.......... Phil
Anonymous
August 15, 2005 5:46:59 PM

Archived from groups: rec.audio.pro (More info?)

>> It would /have/ to be perceptually coded, because there's no way to
>> squeeze CD-format digital sound into a 150kHz channel.

> ** Shannon's theorem ( aka Shannon-Hartley Theorem) says it
> certainly can be done.

> See: http://en.wikipedia.org/wiki/Shannon-Hartley_theorem

> Using the formula C = 0.332 x B x S/N
> C = 1.41 Mb/s and B = 150kHz
> Gives a needed S/N of 28.9 dB.

You're right. In fact, the multi-level (or multi-phase) system described in
the article is used for dial-up modems, to get 56kbs in a 3kHz connection.

"I know nussing" about the modulation schemes used for digital radio.
Anonymous
August 15, 2005 5:54:17 PM

Archived from groups: rec.audio.pro (More info?)

"Will Summawank"
>
>>> But isn't the Tx range established in practice by determining how far
>>> away an acceptable signal can be received (ie, adequate quieting)?
>
>> ** "Quieting" depends on carrier strength alone.
>
> But if I reduce the modulation level, the signal is necessarily not as far
> above the noise. Hence the S/N ratio drops.


** Nonsense when you take into account the REALITIES of broadcast FM.

The modulation level is NOT fixed at the maximum possible.



.......... Phil
Anonymous
August 15, 2005 5:54:18 PM

Archived from groups: rec.audio.pro (More info?)

> ** Nonsense when you take into account the REALITIES of broadcast FM.

> The modulation level is NOT fixed at the maximum possible.

The reality is that a station is going to try to keep the modulation as high
as possible, whether it's by compression (in which case slight differences
in S/N aren't noticeable) or because the engineer is paying attention to
what he's paid to pay attention to.
Anonymous
August 15, 2005 6:05:32 PM

Archived from groups: rec.audio.pro (More info?)

"Arny Krueger"
> "Phil Allison"
>
>>> 3) stereo muxing, this robs some of the TX power from
>>> the Mono signal, reducing Tx range
>
>> ** Not true.
>
> True. There's a ton of energy in the added carriers, subcarriers, etc.


** Bullshit.

The only energy is in the fixed power carrier.


>>> 4) the FM pilot tone it at a low level (see 3), so it is
>>> noisy This carries the S signal, (a-b)
>>
>>
>> ** Complete bullshit.
>
>> The 19kHz pilot is just a tone and carries no programme
>> information.
>
> I think its at 10% modulation. Not a lot loss of max deviation, but finite
> and significant.


** It is insignificant in terms of the Tx range.


> ...the stereo subcarrier sops up deviation very nicely think you.


** Bullshit.

There is no stereo sub-carrier - it 100% suppressed !!


> http://members.tripod.com/~transmitters/stereo.htm
>
> A Stereo transmission tends to sound noisier that a Mono signal. This is
> mainly caused by the noise in the L-R channel. As the noise in the 23 KHz
> to 53 KHz segment is also brought down to the audible 0-15 KHz region by
> the decoding process, we now have more noise than receiving the same
> signal in Mono. The decoder circuits in the receiver also contribute extra
> noise. On top of that, as MPX signals have more bandwidth than a Mono
> signal, the station has to use less modulation with MPX to remain in the
> deviation limits. All above tend to increase the noise. (Visit USENET
> message link below for more info)


** None of which is even slightly relevant to the claim that the of Tx
range being limited because the max deviation is a little less with stereo.

Sure, stereo is noisier that mono - even under ideal conditions.

But max modulation is NOT the reason.



........... Phil
!