Intro
It is a well-known fact that if one looks at a speech waveform using an oscilloscope it is quite commonly "lopsided." By this we mean that one side of the waveform, say the top side, has a greater peak amplitude than the other side. The degree of this asymmetry is highly dependent on the voice of the individual involved. This paper will explore this phenomena. We will outline possible problems which may arise from it and methods for reducing or negating those problems.Possible problems - entirely linear systems
In a high quality public address system where the speaker's voice is simply being augmented by a power amplifier and speaker there would be no problem if the volume levels are such that the system is entirely linear. The lopsided waveform would pass unscathed and indeed unnoticed.If such a lopsided waveform were to be used to modulate an AM transmitter in a noncompetitive situation, and if the modulation is adjusted to be relatively low, then such a waveform would offer no problem. It has been found best in ordinary AM systems to set the polarity of modulation such that the peaks with the greater amplitude are "upward." This minimizes or even eliminates distortion due to the typical envelope demodulator in the typical radio receiver.
In a purely linear system asymmetric waveforms are not in themselves a problem.
In systems using a fast-acting AGC loop
There was at one time a kind of contest between various equipment manufacturers to see which one could develop the fastest-acting AGC system to control the modulation in a transmitter. The devices were commonly called volume limiters or volume limiting amplifiers. They generally reacted to an overload situation within a millisecond or perhaps less. Following the overload they would restore the gain to normal over a period of time of perhaps a few seconds.It was impressive to view the output of these devices on an oscilloscope. There was no visible clipping or other artifacts added to the waveform, just a (usually) well-controlled modulation level. The winner in the war of speed used a system which had a zero attack time. It used a delay line to delay the audio signal while the gain-controlling voltage was being generated.
But in every one of these units there was an annoying tendency for the device to respond to signals which were not the same as what the human ear responded to. Rephrased, they were controlling modulation, not volume. Given a transmitter which could handle the level-controlled waveform properly, these units would control the modulation levels nicely. But they did not maximize volume, which was becoming an issue.
If a lopsided waveform were to be applied to one of these units the peak with the greatest magnitude would cause generation of AGC voltage. If the peaks (positive and negative) could be made equal then less AGC voltage would be generated and modulation would increase. This must be done, however, in a manner which does not increase the peak-to-peak value.
In systems involving clipping of transients
Research showed that if the AGC loop could have a reaction time (attack time) of several milliseconds and a release or recovery time of perhaps 100 to 200 milliseconds that such an AGC loop would match the ear quite nicely. Such a system would control modulation and it would match the ear, allowing the volume to be maximized. There was a drawback to such a scheme: the relatively long attack time required that the basic AGC loop must be followed by a clipper to catch the transients which escaped it.In a system with such a relatively long attack time an asymmetrical applied waveform causes less AGC voltage generation than in a short or fast attack time. However, the signal from the AGC loop is then applied to a clipper. If one peak is clipped more than the other we have a DC (or at least subaudible/syllabic) component exiting from the clipper. In an AM transmitter this appears as a form of "carrier shift." In an FM transmitter this appears as a center frequency shift. It ends up interfering with the transmitter's AFC system. It is certainly disconcerting to watch an analog frequency meter kick violently when such a unit is used to control the modulation with an applied asymmetric waveform.
To minimize this problem the waveform should be processed in a manner such that prior to application to the clipper the peaks are made symmetrical, but without increasing the peak-to-peak value.
Let us look at ways to handle the problem.
A standardized waveform
First let us generate a standard waveform that can be reproduced easily and will allow us to compare various approaches to processing. The proposed signal here has an approximate 10 dB (3:1 voltage-wise) positive to negative ratio but it has no DC component. The areas under the curve for the positive and for the negative portions of the waveform are equal. This signal has a fundamental frequency of 200 Hz and is a believable replica of a steady speech signal, having a similar spectrum. Such a modulating waveform is illustrated here:
The waveform as shown would approximate the sound "ohhh."
There is about 10 dB of asymmetry in this waveform and there is no DC component; as can be seen, the areas above and below the centerline are precisely equal. At the end of this paper is a PSpice net list to enable recreation of the waveform.
If used to modulate the transmitter directly the upward peaks would require 10 dB more power than the downward peaks. If the transmitter could not handle this degree of asymmetry then the modulation level would have to be reduced until the positive peaks were in a linear region and the negative peaks would be reduced in amplitude. This is an inefficient use of transmitter capability.
Use a highpass filter
One way to accomplish the task is to apply the modulating signal to a highpass filter. Indeed, this may be a part of the speech-processing chain already. Such a highpass filter would be doing double duty. It would be removing those components that serve no purpose if transmitted, and in fact may be causing mischief. The highpass filter can also make the positive and negative peaks more nearly equal. If the speech signal were applied to a highpass filter and then to a clipper, there would be similar amounts of clipping applied to the positive and negative peaks. There would be a reduction in axis shift due to any subaudible components generated by asymmetric clipping. And the ear normally tolerates clipping of both modulating waveform peaks better than clipping only one side of the waveform.If that signal is applied to a 100 Hz highpass filter we will obtain the signal shown in the following graphic:
This is the appearance of the waveform after it has been passed through a 100 Hz highpass with a rolloff rate of 18 dB per octave. It does make the positive and the negative peaks have values closer to one another, but at the expense of increasing the value of the smaller peak. This is not our object.
Further, the cutoff frequency of this highpass filter would have to be readjusted for different voices to maintain its effectiveness. This is a conflict with its original intended purpose of passing what we usually consider the "needed" speech components.
Use an allpass network
Another way to process the speech signal is to apply it to a network which rearranges the relative phases of the signal to make it less asymmetric (more symmetrical). Ideally this network will have a flat amplitude response. This can be done with a chain of simple circuits using opamps, as shown in this schematic:
Note that the time-constants in the stages are staggered. This allows the circuit to work regardless of the frequency of the speech waveform fundamental. Changing the frequency of the fundamental of our computer-generated signal results in a fairly uniform correction across the audio frequency band.
That is in fact one of the channels of a real-world phasing network for SSB generation by the phasing method.
The phase shift through that cascade of two individual allpass networks is shown here:
The phase shift is from zero degrees at DC through N times 180 degrees at infinity. Here we have a total of 360 degrees of rotation as we go through the audio spectrum (2 times 180).
The output of the allpass network is shown here:
The signal's peak-to-peak amplitude is not changed. The magnitude of the higher-amplitude peak has been reduced while at the same time the magnitude of the lower-amplitude peak has been increased. There is no axis shift and no subaudible components have been added. The areas under the curve above and below the zero axis are equal. Clipping of such a waveform would cause a minimum of "mischief" compared with clipping of the original asymmetric waveform.
Failure
The use of an allpass network is not a cure-all. If the applied waveform is symmetrical top-to-bottom (i.e. has no even-order harmonic content) then the allpass can actually increase the signal's peak-to-peak amplitude. Here we see an example of an applied waveform with top-to-bottom symmetry:
This signal consists of a 200 Hz fundamental and an equal amount of 600 Hz harmonic.
Now let us apply that signal to an allpass network. The signal at the output of that network can appear as shown here:
This signal still has top-to-bottom symmetry but the peak amplitude has increased. This can be seen by comparison of this waveform with the previous. This is not necessarily catastrophic but it illustrates a point: If the waveform to be corrected has no even-order components (unlikely in practice) and so is symmetrical top-to-bottom in the first place, then the use of an allpass network might not be beneficial. This aspect of the allpass is usually glossed over and is being pointed out for completeness.
RMS-sensing AGC and clipping
If the modulation level in the transmitter is controlled by a peak-sensing audio AGC unit ("limiter"), that AGC system will respond to the peak with the highest instantaneous magnitude. But if the modulation level is controlled by an RMS-sensing AGC unit, asymmetry does not enter into the picture at all as regards the AGC portion of the speech processing. But an RMS-sensing AGC unit must be followed by a clipper to catch those waveform excursions which escape the AGC unit. Be advised those excursions will be of significance. But if the clipper operates on one side of the waveform more than the other, a DC or subaudible component will be developed by the clipper. This will normally cause trouble in the modulator proper. We have a situation wherein the modulator must be direct-coupled to properly handle the signal.Here again the allpass can come to the rescue. Insert the allpass between the RMS-sensing AGC block and the clipper. By adding the allpass at this point the clipper will clip symmetrically and no subaudible components will be involved.
Suggestion
An allpass block should be placed in an appropriate point in any speech processor used in a radio transmitter:It should be placed ahead of an AGC circuit which is essentially instantaneous-acting
or
It should be placed ahead of a clipper if the preceding AGC circuit is low-acting or especially if it is RMS-sensing.
The allpass network is inexpensive and generally insures lower distortion by virtue of less clipping or at least symmetrical clipping (which sounds less offensive to the human ear). Symmetrical clipping always causes less "mischief" than does asymmetrical clipping. Placement of the allpass block should always be prior to the point where clipping might occur or where it does occur.
History
The concept of using an allpass network to make an audio signal more symmetrical is certainly not new. The first commercial product was called the SymmetraPeak and was placed on the market by Kahn Communications. During the 70s an active equivalent was installed in the circuitry inside various audio "processors," including those produced by Circuit Research Labs in Phoenix and by Orban in San Francisco. The amateur radio fraternity apparently has been relatively slow to pick up on the idea. The only radio amateur writing on this subject seen by ye scribe has been by Gary Blau, W3AM.
W3AM Website
W3AM Allpass
The netlist
This netlist was written for the PSpice® Spice simulator and is proposed as a generator of a "standardized" speech waveform for allpass network evaluation.
ALLPASS.cir
* 4 1 July 21 June 2001 J. Tonne
* Waveform generator; raw pulse is at 1;
* smoothed "voice" signal is at 100;
* output of highpass filter is at 200;
* output of allpass network is at 303 (for now).
* Raw pulse generator:
Vin 1 0 PULSE -.38 1 0 .0002 .001 .0008 .005
Rin 1 0 1G
* Response shaper to form voice signal:
E1 100 0 VALUE = { 1000 * ( V(3)-V(100) ) }
R1 1 2 10k
R2 2 3 10k
C1 2 100 .15u
C2 3 0 .003u
* Highpass driven by voice signal:
E2 200 0 VALUE = { 1000 * ( V(6)-V(200) ) }
R3 4 0 5.7158k ; These resistor values
R4 5 200 2.2469k ; make a Butterworth
R5 6 0 39.238k ; highpass filter
C3 100 4 .2u
C4 4 5 .2u ; .2uF yields -3 @ 100 Hz
C5 5 6 .2u ; with resistors shown
* Allpass driven by voice signal:
R6 100 301 10k
R7 301 303 10k
R8 302 0 10k
C6 100 302 .1u
E3 303 0 VALUE = { 1000 * ( V(302)-V(301) ) }
.TRAN 10u .06 0 10u
.PROBE
.END
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