Forums

Digital Equalizer Design for a Radio Receiver

Started by Unknown February 7, 2006
Hi all,
  I am currently on a design of an equalizer for Medium/Short wave AM
receiver.

Problem statement:
Medium /Short waver AM receiver suffers from sudden volume reduction
irrespective of the Automatic gain control in receiver. How does this
can be addressed. I am trying to control this audio reduction by an
equalizer.  But how does the actual reduction of audio in the radio
program and reduction of audio due to radio reception addressed.

I have went through bell and shell type digital equalizers available
for a desgin.

Question

[1] How does this problem of sudden reduction in volume irrespective of
the Automatic Gain Control in an Medium wave AM receiver.

[2] Apart from this how does the actual reduction of audio in the radio
program and reduction of audio due to radio reception addressed.

Is my approach correct.

Any relevant information or references will be appreciated.

Ganesan R

<ganernet@yahoo.co.in> wrote in message
news:1139289921.690084.262190@g14g2000cwa.googlegroups.com...
> Hi all, > I am currently on a design of an equalizer for Medium/Short wave AM > receiver. > > Problem statement: > Medium /Short waver AM receiver suffers from sudden volume reduction > irrespective of the Automatic gain control in receiver. How does this > can be addressed. I am trying to control this audio reduction by an > equalizer. But how does the actual reduction of audio in the radio > program and reduction of audio due to radio reception addressed. > > I have went through bell and shell type digital equalizers available > for a desgin. > > Question > > [1] How does this problem of sudden reduction in volume irrespective of > the Automatic Gain Control in an Medium wave AM receiver. > > [2] Apart from this how does the actual reduction of audio in the radio > program and reduction of audio due to radio reception addressed. > > Is my approach correct. > > Any relevant information or references will be appreciated. > > Ganesan R >
I believe you want automatic level control (ALC) or some people call it automatic volume control. At any rate it is just an AGC after the AM demodulator. It operates at the audio rates instead of the IF rates so it can pull up the signal energy even when the RF is fading. You may also want to google "adaptive filters". These have been used for years to clean up HF audio. I worked on a product previously that did automatic notch filtering or speech enhancement based on the user selection. -Clark
ganernet@yahoo.co.in wrote:
> Hi all, > I am currently on a design of an equalizer for Medium/Short wave AM > receiver. > > Problem statement: > Medium /Short waver AM receiver suffers from sudden volume reduction > irrespective of the Automatic gain control in receiver. How does this > can be addressed. I am trying to control this audio reduction by an > equalizer. But how does the actual reduction of audio in the radio > program and reduction of audio due to radio reception addressed. > > I have went through bell and shell type digital equalizers available > for a desgin. > > Question > > [1] How does this problem of sudden reduction in volume irrespective of > the Automatic Gain Control in an Medium wave AM receiver. > > [2] Apart from this how does the actual reduction of audio in the radio > program and reduction of audio due to radio reception addressed. > > Is my approach correct. > > Any relevant information or references will be appreciated. > > Ganesan R
It sounds like you are suffering from the effects of a fading channel. An adaptive equalizer that keeps the received signal envelope within prescribed bounds might help. I don't know if the Constant Modulus Algorithm (CMA) has been used for this, but it is worth a search. John
john wrote:
> ganernet@yahoo.co.in wrote: > > Hi all, > > I am currently on a design of an equalizer for Medium/Short wave AM > > receiver. > > > > Problem statement: > > Medium /Short waver AM receiver suffers from sudden volume reduction > > irrespective of the Automatic gain control in receiver. How does this > > can be addressed. I am trying to control this audio reduction by an > > equalizer. But how does the actual reduction of audio in the radio > > program and reduction of audio due to radio reception addressed. > > > > I have went through bell and shell type digital equalizers available > > for a desgin. > > > > Question > > > > [1] How does this problem of sudden reduction in volume irrespective of > > the Automatic Gain Control in an Medium wave AM receiver. > > > > [2] Apart from this how does the actual reduction of audio in the radio > > program and reduction of audio due to radio reception addressed. > > > > Is my approach correct. > > > > Any relevant information or references will be appreciated. > > > > Ganesan R > > It sounds like you are suffering from the effects of a fading channel. > An adaptive equalizer that keeps the received signal envelope within > prescribed bounds might help. I don't know if the Constant Modulus > Algorithm (CMA) has been used for this, but it is worth a search. > > John
interesting point... how could you make an adaptive equalizer for an analog AM modulated signal? or FM for that matter... The AE has nothing to guage the "correct" signal by... as it would with a QAM signal where it can look at the constellation points...???????? I know there were ghost canclers made for analog AM TV signals but these used reference signal that were included in the video blanking interval... How could you do it for a voice modulated signal?? I think the OP needs a good AGC that keeps the IF level constant before detection followed by an audio AGC that makes up for real or apparent changes to the modulation depth... Mark Mark Mark

Mark wrote:


> > interesting point... > > how could you make an adaptive equalizer for an analog AM modulated > signal? or FM for that matter... > > The AE has nothing to guage the "correct" signal by... as it would with > a QAM signal where it can look at the constellation points...???????? > > I know there were ghost canclers made for analog AM TV signals but > these used reference signal that were included in the video blanking > interval... > > How could you do it for a voice modulated signal??
For the sake of argument, it could be done. For AM, you can tune the equalizer to minimize the dissimilarity between the lower and the upper sidebands. Also you can exploit the statistical properties of the audio signal. Anyway it is going to be a mental exersise for fun without much practical purpose.
> I think the OP needs a good AGC that keeps the IF level constant > before detection followed by an audio AGC that makes up for real or > apparent changes to the modulation depth...
AGC is a trivial solution, which is of limited help. The AGC can't keep the SNR, so it sounds like the signal dives back and forth into the noise. To me, this effect is more irritating then the fading itself. I don't expect much help from the use of the equalizer either. Vladimir Vassilevsky DSP and Mixed-Up Signal Design Consultant http://www.abvolt.com
"Vladimir Vassilevsky" <antispam_bogus@hotmail.com> wrote in message
news:_BaGf.2049$rL5.778@newssvr27.news.prodigy.net...
> > > Mark wrote: > > > > > > interesting point... > > > > how could you make an adaptive equalizer for an analog AM modulated > > signal? or FM for that matter... > > > > The AE has nothing to guage the "correct" signal by... as it would with > > a QAM signal where it can look at the constellation points...???????? > > > > I know there were ghost canclers made for analog AM TV signals but > > these used reference signal that were included in the video blanking > > interval... > > > > How could you do it for a voice modulated signal?? > > For the sake of argument, it could be done. For AM, you can tune the > equalizer to minimize the dissimilarity between the lower and the upper > sidebands. Also you can exploit the statistical properties of the audio > signal. Anyway it is going to be a mental exersise for fun without much > practical purpose. > > > I think the OP needs a good AGC that keeps the IF level constant > > before detection followed by an audio AGC that makes up for real or > > apparent changes to the modulation depth... > > AGC is a trivial solution, which is of limited help. The AGC can't keep > the SNR, so it sounds like the signal dives back and forth into the > noise. To me, this effect is more irritating then the fading itself. > I don't expect much help from the use of the equalizer either. > > Vladimir Vassilevsky > > DSP and Mixed-Up Signal Design Consultant > > http://www.abvolt.com >
The basical principle is you use the adaptive filter to "predict" the actual received signal, i.e. the error signal driving the AF is the difference between the actual signal and the output of the AF. Then if you listen to the output of the AF you get only the components that are statistically predictable. In my case I used the AF output as the speech enhancement output (you can't predict noise or rapid fading, right?) and I used the error signal for tone removal. By tweaking the step size of the AF you can control the statistics it's able to track from highly correlated(tones) to medium correlated(speech). The advantage of the AF is that it automatically expands/contracts the bandwidth based on the SNR of the received signal. There are several HF products out there doing the same thing by the way. -Clark
Clark,

The AF (adaptive filter) you are describing automatically adapts the
overall bandwidth of the signal...OK...good... but does it actually
compensate for selective fading the way an adaptive equalizer does for
digtial modulation...  i.e. if the selective fading causes  a 5 dB tilt
across the channel,  will your AF compensate for that tilt by creating
a reverse tilt or will it just reduce the bandwidth because the SNR is
poor....

Mark


Anonymous wrote:


> > The basical principle is you use the adaptive filter to "predict" the actual > received signal, i.e. the error signal driving the AF is the difference > between the actual signal and the output of the AF. Then if you listen to > the output of the AF you get only the components that are statistically > predictable. In my case I used the AF output as the speech enhancement > output (you can't predict noise or rapid fading, right?) and I used the > error signal for tone removal. By tweaking the step size of the AF you can > control the statistics it's able to track from highly correlated(tones) to > medium correlated(speech).
Are you implementing some kind of noise reduction filter after the AM envelope detector? Or are you trying to compensate for the multipath distortion before the detector? What is the goal?
> The advantage of the AF is that it automatically expands/contracts the > bandwidth based on the SNR of the received signal. There are several HF > products out there doing the same thing by the way.
There is not much bandwidth in AM. It's only about 5kHz. Since the spectrum is so narrow, the fading is not very selective and hits a big part of the spectrum at once. I don't expect much gain from either pre-detector or post-detector processing. Vladimir Vassilevsky DSP and Mixed Signal Design Consultant http://www.abvolt.com
"Mark" <makolber@yahoo.com> wrote in message
news:1139417980.646987.93170@g44g2000cwa.googlegroups.com...
> Clark, > > The AF (adaptive filter) you are describing automatically adapts the > overall bandwidth of the signal...OK...good... but does it actually > compensate for selective fading the way an adaptive equalizer does for > digtial modulation... i.e. if the selective fading causes a 5 dB tilt > across the channel, will your AF compensate for that tilt by creating > a reverse tilt or will it just reduce the bandwidth because the SNR is > poor.... > > Mark >
An AF needs some known component or statistic by which to generate an error term from. With voice AM there really isn't any fixed parameter to lean on. A loss of amplitude could be fading on the RF OR the person speaking could have just stopped talking. If it were FM you could probably uses the constant envelope of the signal, but I don't see any way to do it with AM. Maybe ALC on the audio will work for you I just know that an AF in the predictive mode worked well for us in the past to make HF audio much more tolerable to listen to. There used to be several companies selling little dsp boxes that you hung off your radio speaker to do this. I can recall a name right now. -Clark
"Vladimir Vassilevsky" <antispam_bogus@hotmail.com> wrote in message
news:jYpGf.53436$PL5.14607@newssvr11.news.prodigy.com...
> > > Anonymous wrote: > > > > > > The basical principle is you use the adaptive filter to "predict" the
actual
> > received signal, i.e. the error signal driving the AF is the difference > > between the actual signal and the output of the AF. Then if you listen
to
> > the output of the AF you get only the components that are statistically > > predictable. In my case I used the AF output as the speech enhancement > > output (you can't predict noise or rapid fading, right?) and I used the > > error signal for tone removal. By tweaking the step size of the AF you
can
> > control the statistics it's able to track from highly correlated(tones)
to
> > medium correlated(speech). > > Are you implementing some kind of noise reduction filter after the AM > envelope detector? Or are you trying to compensate for the multipath > distortion before the detector? What is the goal? >
The former. A fast AGC on the IF could compensate the fading but it can't be so fast that it strips the AM modulation, right?
> > The advantage of the AF is that it automatically expands/contracts the > > bandwidth based on the SNR of the received signal. There are several HF > > products out there doing the same thing by the way. > > There is not much bandwidth in AM. It's only about 5kHz. Since the > spectrum is so narrow, the fading is not very selective and hits a big > part of the spectrum at once. I don't expect much gain from either > pre-detector or post-detector processing. >
I basically agree but you can minimize the fatigue on the listener by getting rid of the noise bursts that occur as the fade drives the signal towards the noise floor, right?
> Vladimir Vassilevsky > > DSP and Mixed Signal Design Consultant > > http://www.abvolt.com