Optimize Air Chain (Page 2 of 3)
Requirements For Studio-Transmitter Links ("STL")
If the STL is prior to the audio processor, the STL's signal-to-noise ratio must be sufficient to pass unprocessed audio. If the processor precedes the STL, its frequency response must be flat (+/-0.1dB) throughout the operating frequency range. The group delay must be essentially constant throughout this range (deviation from linear phase <+/-10 degree). Phase correction can be applied to meet the requirement at high frequencies. These requirements are necessary to preserve stereo separation and peak modulation control.
At low frequencies, by far the best way to achieve the specification is to extend the -3dB frequency of the STL to 0.16Hz or lower (as discussed above) and to eliminate any peaking in the infrasonic frequency response prior to the rolloff. Poor AFC-loop design in STL transmitters is all too common, and this is the most likely cause of low-frequency response problems. Such problems can be corrected by applying prior to the STL transmitter equalization that is complementary to existing low-frequency rolloff, such that the overall system frequency response rolls off smoothly at 0.16Hz or below. This solution is much better than clipping the tilt-induced overshoots after the STL receiver because the clipping will introduce non-linear distortion, while the equalizer is distortion-free.
Digital STL systems using lossy bit-rate-reduction schemes do not pass processed audio accurately. For example, measurements have shown that APT-XTM at 256kps introduces as much as 3db of overshoot with processed audio and ISO/MPEG Layer II at 384kbps introduces as much as 1dB. Overshoots increase markedly as bit rate is reduced. While these overshoots can be clipped or limited, such processing will cause audible side-effects. On the other hand, if the audio processor is located at the transmitter, its input can be fed without difficulty from an STL using a lossy bit-rate-reduction scheme because it is unnecessary to preserve the waveshape of unprocessed input audio.
Peak Modulation Control
The audio processor must control the peak modulation of the RF carrier to the standards required by the governing authority, such as the FCC in the United States. In FM, the peak deviation of the carrier must be controlled so that the modulation monitor specified by the governing authority does not indicate overmodulation. Because the rules often permit the modulation monitor to ignore very brief overshoots, the instantaneous peak deviation might exceed the peak modulation as indicated on the modulation monitor.
The requirements for peak control and spectrum control tend to conflict, which is why sophisticated non-linear filters are required to achieve highest performance. Applying a peak-controlled signal to a linear filter almost always causes the filter to overshoot and ring because of two mechanisms: spectrum truncation and time dispersion. One can build a square wave by summing its Fourier components together with correct amplitude and phase. Analysis shows that the fundamental of the square wave is approximately 2.1dB higher than the amplitude of the square wave itself. As each harmonic is added in turn to the fundamental, a given harmonic's phase is such that the peak amplitude of the resulting waveform decreases by the largest possible amount. Simultaneously, the rms value increases because of the addition of the power in each harmonic. This is the fundamental theoretical reason why simple clipping is such a powerful tool for improving the peak-to-average ratio of broadcast audio: clipping adds to the audio waveform spectral components whose phase and amplitude are precisely correct to minimize the waveform's peak level while simultane ously increasing the power in the waveform.
If a square wave (or clipped waveform) is applied to a low-pass filter with constant time delay at all frequencies, the higher harmonics that reduce the peak level will be removed, increasing the peak level and, with it, the peak-to-average ratio. Thus even a perfectly phase-linear low-pass filter will cause overshoot. There is no sharp-cutoff linear low-pass filter that is overshoot-free; overshoot-free spectral control to FCC or CCIR standards must be achieved with filters that are embedded within the processing, such that the non-linear peak-controlling elements in the processor can also control the overshoot.
If the sharp-cutoff filter is now allowed to be minimum-phase, it will exhibit a sharp peak in group delay around its cutoff frequency. Because the filter is no longer phase-linear, it will not only remove the higher harmonics required to minimize peak levels, but will also change the time relationship between the lower harmonics and the fundamental. They become delayed by different amounts of time, causing the shape of the waveform to change. This time dispersion will therefore further increase the peak level.
When a square wave is applied to a linear-phase filter, overshoot and ringing will appear symmetrically on the leading and trailing edge of the waveform. If the filter is minimum-phase, the overshoot will appear on the leading edge and will be about twice as large. In the first case, the "overshoot and ringing" are in fact caused by spectrum truncation which eliminates harmonics necessary to minimize the peak level of the wave at all times; in the second case, the overshoot and ringing are caused by spectrum truncation and by distortion of the time relationship between the remaining Fourier components in the wave.
Overshoots in Composite STL Systems & FM Exciters
It is well known that processed, peak-controlled program material causes most composite STL systems and FM exciters to produce overshoot in FM composite baseband signals. The heavier the processing the more they overshoot. This overshoot occurs even in properly band-limited systems that utilize STL paths without multipath. Accordingly, loudness is compromised because average modulation must be reduced to prevent illegal peak overmodulation caused by this overshoot. Previous attempts to eliminate this overshoot have degraded system performance.
It is essential that the measuring equipment have transient accuracy at least as good as the equipment being measured. Many popular modulation monitors introduce false tilt and overshoot into their readings. To avoid such inaccuracy, we used a Belar Electronics Wizard™ FM Modulation Analyzer - a known-accurate instrument - to plot peak modulation versus time on the output of an aggressive audio processing system and several contemporary STL systems and FM exciters. The results reveal the extent of the problem. The same program material and time segment was used for all plots. Fig. 1 shows the audio processor output peak modulation directly. Figs. 2 and 3 show the peak modulation of STL System 2 and FM Exciter 2 respectively. Note that both the STL System and FM Exciter suffer from overshoot causing over-modulation. To prevent the resulting over-modulation, the modulation level must be reduced over 13%, losing almost 1.5dB of loudness!
Fig. 1 AUDIO PROCESSOR OUTPUT PEAK MODULATION
Fig. 2 STL SYSTEM 2 PEAK MODULATION-Before Modification
Moseley Associates, Inc. PCL-6010/C
Fig. 3 FM EXCITER 2 PEAK MODULATION-Before Modification
Continental Electronics Mfg. Co., Inc. 802A
The FM Stereo System
The world-standard FM stereo "pilot-tone" system encodes the sum of the channels (L+R) in the frequency range of 30-15,000Hz in the stereo baseband - the "stereo main channel." It encodes the difference between the channels (L-R) on a double-sideband suppressed-carrier sub-channel centered at 38kHz, and occupying 23kHz to 53kHz in the stereo baseband - the "stereo sub-channel." A pilot tone at 19kHz tells the receiver that a stereophonic transmission is being received, and provides a phase and frequency reference to permit the receiver to regenerate the 38kHz sub-carrier to use in its stereo demodulator. Any energy that appears in the frequency range from 30 to 19,000Hz caused by a signal in the stereo sub-channel is termed "sub-channel-to-main channel crosstalk." Any energy that appears in the frequency range from 19 to 57kHz caused by a signal in the stereo main channel is termed "main-channel-to-sub-channel crosstalk."
When the stereo encoder is driven by a pure right-only or left-only signal, "stereophonic separation" can be measured at the stereo decoder as the ratio between the desired and undesired signal levels, where the "desired" signal is the signal appearing in the decoder output channel corresponding to the channel driven at the encoder, and the "undesired" signal is the signal caused by the desired signal that appears at the remaining output.
Ideally, crosstalk is non-existent and stereophonic separation is infinite. In practice, both linear and non-linear errors cause these characteristics to deteriorate.
In the linear domain, separation and crosstalk are mathematically orthogonal. Phase and frequency errors that cause one to deteriorate will not affect the other. For example, phase or frequency errors in the composite signal channel will cause separation to deteriorate, but cannot affect crosstalk, since the stereo main and sub-channels are already separated in frequency and changes in phase or amplitude response in the composite channel cannot affect this frequency separation. Conversely, mismatches between the linear response of the left and right signal paths prior to the stereo encoder will cause crosstalk, but cannot affect separation.
Non-linearities in the composite channel can cause both separation and crosstalk to deteriorate because such errors cause harmonic and intermodulation distortion that introduce new frequencies into the baseband. These new frequencies are likely to inject power into a part of the baseband spectrum that will be decoded by the stereo decoder in spatial locations different than the locations of the original sound sources. Further, these new frequencies are perceived by the ear not as changes in spatial localization, but as highly offensive distortion.
This is somewhat analogous to "aliasing distortion" in a sample-data audio system. In such a system, any input frequencies greater than one-half of the sampling frequency (the "Nyquist frequency") are encoded with the wrong frequency: they "fold around" the Nyquist frequency and appear at the decoder as frequencies unrelated to the program material that produced them. The ear perceives this "aliasing" as offensive distortion.
Previous Non-Linear Solution
The most common technique for reducing FM composite baseband signal overshoot has been composite baseband clipping. Composite clipping has been disparaged because it causes signal degradation and because early implementations that clipped the pilot could violate the FCC rules. No implementation prevents dynamic signal degradation.
The composite baseband clipper is a non-linear device. Thus, it generates distortion and aliasing products that contaminate the composite baseband signal, degrading dynamic stereo separation and causing audible dynamic distortion. It also produces distortion products in the sub-carrier region, reducing or destroying the sub-carrier's market value and reducing revenue potential.
While it is possible to lowpass-filter the clipped baseband (thus protecting the sub-carriers), such filtering does nothing to eliminate intermodulation distortion in the stereo baseband region below 57kHz, and will also tend to increase peak modulation, partially negating the peak control provided by the composite clipper. Such filters do not protect the 19kHz pilot tone from interference caused by the clipper-induced distortion, which can cause problems in receivers' stereo decoders.
If the FM exciter is the source of overshoot (as opposed to the STL), or if the clipper precedes the STL, then the composite baseband clipper cannot control overshoot. Instead, it can actually increase overshoot because the clipping process produces increased amounts of infrasonic intermodulation distortion products.
Some have argued that composite baseband clipping increases loudness more than audio clipping in the left and right channels. But this loudness increase is accompanied by degraded dynamic stereo separation and crosstalk. To preserve dynamic stereo performance, the spectra of the stereo main channel and sub-channel must be completely isolated: the main channel must not have any energy above 19kHz, and the sub-channel must not have any energy below 19kHz.
One consequence of such frequency separation is this: in a system that achieves high dynamic separation and low crosstalk, it must be impossible for the system's final filter/limiter to reproduce any approximation to a square wave if the square wave's fundamental frequency is higher than one-third the cut-off frequency of the low-pass filter prior to stereo encoding (typically 15kHz). This is because the third harmonic of the square wave is three times the frequency of the fundamental, so the low-pass filter removes it (and all higher harmonics too); any square wave above 5kHz will emerge from the receiver as a sine wave. Because they generate spurious harmonic and intermodulation products, composite baseband clippers do not meet this criterion and thus compromise dynamic stereo performance.
Composite clipping has one potential advantage. Conventional wisdom holds that the peak modulation of the composite baseband is the greater of the left or right channel levels, plus the pilot. However, this is only an approximation because the pilot is correlated in phase with the 38kHz suppressed sub-carrier. This causes the total composite modulation to decrease slightly when the left and right channels are unequal in level. Assuming 10% pilot injection and holding the left channel at 100% modulation, decreasing the right channel from 100% to 0% modulation will cause the composite modulation to decrease by 2.8%. Perfectly accurate peak limiting in the audio domain, prior to stereo encoding, can only control the composite modulation to an accuracy of -2.8%/+0%. Only a process that is aware of the total peak composite modulation (including the pilot and any subcarriers) can control composite modulation accurately. Since composite baseband clipping controls the peak deviation of the composite signal precisely (assuming the pilot is also clipped), it can theoretically be louder than peak limiting in the audio domain. But the "advantage" is an imperceptible 0.24dB at best! Composite baseband clippers that do not clip the pilot (which are the only clippers legal for use in the U.S.) do not eliminate the interleave error, and therefore produce no loudness advantage at all!
Left channel modulation 100%.
Right channel modulation from 0% to 100%.
10% pilot injection.
Peak deviation of composite shown with 100% normalized to L=R with pilot present.
Right % Modulation Composite % Modulation
100.0% 100.00%
99.5% 99.79%
99.0% 99.59%
98.0% 99.26%
97.0% 98.99%
96.0% 98.77%
95.0% 98.59%
94.0% 98.45%
93.0% 98.33%
92.0% 98.14%
90.0% 98.06%
80.0% 97.65%
70.0% 97.48%
60.0% 97.39%
50.0% 97.33%
40.0% 97.29%
30.0% 97.26%
20.0% 97.24%
10.0% 97.22%
0.0% 97.20%
Fig. 5 INTERLEAVING ERROR IN THE FM STEREO SYSTEM
Fig. 5 is a plot of the interleaving error in % of modulation versus right channel modulation, with the left channel modulation held constant at 100%.
In these digital times, bad audio quality has become unacceptable to formerly unsophisticated consumers. It is absurd that composite baseband clippers are being used to degrade system performance below that of some of the least expensive receivers! Composite clipping is a very easy, unsophisticated method of increasing apparent loudness of the broadcast signal, but it compromises quality in a way that is unacceptable to any broadcaster trying to compete with CD or the newer recordable digital media.