Index

 

signalprocessing

DSP

DX-11 kit

test set up

test results

conclusions

 

Fishing in the noise

(published in Electron #7, 2007)

 

 

 

Introduction

 

For many years radio-amateur are trying to communicate especially under adverse contact conditions. A dedicated radio-amateur is most satisfied when he finally does succeed in 'working that difficult station' when conditions are almost prohibitive and 'big guns' have given up long ago. The limitation of the license than no longer will be regarded as a threat but more of an opportunity and a challenge to use every trick in the book to bring the minimal exchange of information to a good end.

 

The challenge is taken on at various levels, starting off at the antenna at the transmitting and receiving side. Loss in the antenna system is the first to avoid trap and a discussion about this subject may be found at "Where does the power go".

Next to the antenna-system, the 'art of listening' is an important instrument that may only be mastered after extensive practice. This art may be boost by clever tools to give the signal an edge over the noise. This last subject will be the topic in the following discussion.

   

 

Signal processing

 

Already from the beginning of radio-communication, listening stations have been confronted with all kinds of natural radio-sources that made life difficult. It therefore did not take long before the 'noise limiter' was invented to limit the deafening burst noise of thunderstorm.

 

In these early radio-years Morse code signals were the mode of operation. Using feed-back receivers it was possible to limit band-width at the receiving side to the absolute minimum as required range for these signals.  Frequency-stability problems of the receivers of those days however made keeping track a tedious adventure. With the super-heterodyne arriving, stability was enhanced as well and filtering to improve signal to noise ratio was shifted to the audio part of the station.

 

It soon appeared to be difficult to design and construct good audio-filtering that was steep enough to prevent unwanted signals to pass without too much added distortion. Filtering technology at the IF section was helping out especially with the aid of 'mechanical' technologies like crystal filters, later to be followed by the famous Collins mechanical filters.

With these technologies firmly established it appeared rather difficult to further improve on analogue filtering since practical limitation proved to be of a rather fundamental nature. Listening practice was again the art to go for.

 

 

Digital signal processing

 

A big step in filtering technology could be made with the micro-processor arriving on the scene to allow Digital Signal Processing (DSP) on real-time signals. By converting analogue signals to digital values using an A/D converter, mathematical signal processing became available with addition, subtraction, multiplication and holding the signal samples for some time (delay) as tools of the trade.

 

Loss of signal quality was related to limited resolution of readily available, real-time A/D converters and the span of the internal registers. In those early processor days these internal registers were four to eight bit wide while the processor was running at a very modest speed of a few mega 'beats' per second. To reliably reproduce digital signals, the signal samples should be taken at just over twice the highest frequency to reproduce. With the high number of clock periods for each mathematical process taken into account it is clear the micro-processors of those day where not fast enough for real-time audio processing.

 

By the end of the '80 of last century the first micro-processors came to the market that were dedicated to real-time, digital signal processing. Registers were 16 to 24 bits wide with on-board high-resolution A/D and D/A converters to provide for the necessary high signal resolution. Furthermore these processors were equipped with dedicated hardware for direct multiplication without the high amount of clock tics. The TMS 310 by Texas Instruments was a good example of such micro-processor and is still at the heart of many DSP processing equipment, although currently in more advanced versions of this TMS series. 

 

In many contemporary receiving systems DSP is 'a must' and a selling argument. Next to build-in processors add-on DSP systems  are on offer by various manufacturers as a 'set-top' box. More information on this subject is also available at the ARRL-site.

 

Many of these DSP processors are limited to analogue-like filtering, although with much enhanced specifications and without the 'ringing'- effects that hinder their analogue counterparts when at low band-width. With these DSP filters it is not unusual to find very steep filter edges e.g. - 3dB @ 2,5 kHz with -60 dB @ 2,8 kHz. It is also possible to select variable (audio) bandwidth with these specifications at any position in the LF spectrum. With this technology also (automatic) notch filters can be created to counteract annoying whistlers.

Most of the above functionality however is still connected to processing in the 'frequency domain', whereas with DSP technology and the right mathematics also processing in time-domain is quite feasible and also time and frequency domain at the same time. Mathematical background to this extend has become available from the military and medical profession opening doors to more, and more versatile audio signal processing.

 

A good example of such a dedicated DSP system is the 'SprachEkstraktor' by Günther Michels, DJ7UP, available as module, DX-11 or as a complete set-top box, DX-21 and may be found at his web-site.   

This system is exploiting the typical characteristics of spoken language as in the limited frequency range and the low frequency repetition of basis speech utterances. By focusing on these characteristics the system is capable of recognizing speech from (other?) noises and enhancing accordingly. This typically is also the power of our biological hearing and speech processing. 

 

Not long ago I visited a local ham-friend that showed me the DX-21 'set-top box'. It indeed did sound OK and was remarkable in its ability to remove the noise from already intelligible speech signals. In my experience 'speech enhancing' systems can improve the quality of speech signals in a noisy environment and will diminish listening 'fatigueness'. The biological processor though is usually better at improving intelligibility at adverse signal conditions, so to me it was not really worth my while to purchase such system.

 

This speech processing system by DJ7UP however appeared to have opened a new window and I was wondering whether this system really was to improve on an already trained biological processor. Therefore I acquired the bare processing board DX-11 to test for myself.

 

DX-11 kit

 

The DX-11 kit as send to me was existing of a professional multilayer printed circuit board, equipped with SMD components amongst which the earlier mentioned TMS320 in a prominent position. On top of this a small amount of additional components was supplied as well to ease integrating this processing system in an existing environment (receiver, speaker) or to construct your own stand-alone cabinet. A picture of the kit is in figure 1. 

 

Beschrijving: Beschrijving: Beschrijving: DX-11 kit

Figure 1: DX-11 kit

 

 

The dimensions of the small footprint DX-11 board may be judged by the size of the additional components that came with the board. It is obvious this small PCD will easily fit in any, not too compactly build, existing cabinet.

 

 

Test set-up

 

To enable an objective, quantifiable and repeatable test, I set up a test environment as in figure 2.

 

 

Figure 2: Test set-up

 

 

In figure 2 the following components may be recognized:

1. At position 1, a source of high quality speech, of unknown content was selected. For this source I tried several different sources like a local FM news broadcasting station and a few 'speaking books' from the local library. These were available in a variety of stories and with different voices by male and female speakers. As it appeared intelligibility also to depend on the particular speaker I did several tests with different speech sources.

2. Amateur receiver set for reception of SSB speech signals. Internal audio bandwidth is switched to 2,4 kHz @ -3dB and 4,4 dB @ -60 dB. This receiver was to be tuned to different types of band-noise, switching noise, TV-rattle, thunderstorm, etc. 

3. Audio filter to limit the 'perfect speech' source to the same audio band bandwidth as the receiver. The signals from 1 and 2 are added and fed to the DX-11 at position 5.

4. The input signals to the DX-11 are also fed to an oscilloscope to ensure peak signals are below the maximum input voltage for the A/D converter in the processor at 900 mVtt. At the same position an additional, slow reacting, analogue rms milliVolt meter is connected to obtain a clear and repeatable reading of all input signals as a single value. 

5. DX-11 that could be controlled by the supplied potentiometer to obtain optimal intelligibility at different noise-types and signal-to-noise conditions.

A switch has been inserted to select signals with and without the speech-processor. The final audio signals were level-optimized each time for good listening conditions. This appeared to be a little bit different for different types of signal.

  

 

Audio tests

 

For each test the maximum levels have been checked and set to ensure maximum peak-peak input signal always remain (just) below 900 mV. The accompanying rms value has been kept constant at each of the successive tests, to ensure the DX-11 to always operate at the same internal resolution.

 

The first series of speech intelligibility tests have been performed using back-ground band noise from all HF amateur frequency bands, because this noise did sound differently in different bands. Next, speech was turned up until this was just, but 100% intelligible. At each of these test the optimal setting at the DX-11 potentiometer was 'tweaked', but this appeared not to be critical; once a setting had been selected, this could be maintained for most of the other tests.

At this 'just 100% intelligibility' position, the noise source (receiver) was switched off, and the speech level was measured. This speech level as compared to the noise measurement above (without speech) is providing the S/N figure at this setting. The test could be repeated (with different speech content, the news or spoken book was not interrupted) with very good accuracy.  

 

Next, the test was repeated without the speech processor, to find the set-point for the trained, naked ear, at the same audio noise level as before. Again the 'just 100%  intelligibility' position is being determined to yield a comparable S/N figure.

Comparing both S/N figures will present an 'objective' presentation as to the quality (improvement / degradation) of this speech processor.  

 

Tests have been performed for various types of 'noise', e.g. 'white' band noise, noise by TV-rattle, jammers, thunderstorm, LF break-in noise, neighboring channel SSB-speech break-in and artificially made 'noise'. For all measurements, different 'voices' have been used, as some were more easily intelligibility or pleasant to the ear. In general, the more high-pitched voices (female), at a more constant speech rate (female) provided highest intelligible in noisy conditions when operating the DX-11 as well as to the naked ear.   

 

Results

 

- White band-noise at a 'clean' part of the 80 m. band, at S 3 on the meter (no antenna is S 0).

* With processor: S/N: -9,3 dB. Without: S/N: -6,3 dB. DX-11 improves the intelligibility by 3 dB.

Some difference could be measured between various male voices with female voice as best. Results at this voice:

* With processor: S/N: -10,9 dB. Without: S/N: -6,3. DX-11 appears to be better tuned to this voice. On average DX-11 improved S/N by around 4 dB.

 

- White band noise in the 80 m. band plus 'TV rattle' at + 1 S-point.

* With processor: S/N: -15,3 dB. Without: S/N: -9,3 dB.

It is remarkable we can hear 'deeper' into the noise with this type of 'colored noise' than without. Also DX-11 is better at this task. In all, DX-11 was beating 'naked hearing' by 6 dB, an improvement of 2 dB over 'band noise' only.

 

Since not all 'TV-rattle' sounds the same, a different and much stronger rattle at + 3 S-points was found at the 80 m. band.

* With processor: S/N: -8,6 dB. Without: S/N: -2,3 dB.

This rattle appeared to be a challenge for DX-11 as well as to the naked ear. Still DX-11 won by about 6 dB.

 

- Same band, same band-noise, this time with added artificial rattle by a LF square wave generator, tuned to a spike repetition of about 5 Hz. (at the human utterance-repetition frequency range).

* With processor: S/N: -6,8 dB. Without: -4.0 dB.

The DX-11 as well as the naked ear clearly had some problems. Still DX-11 won by 3 dB.

 

- Wide band noise at a 'clean' part in the 10 m. band, at S1 on the meter (no signal is S 0)

* With processor: S/N: -6,3 dB. Without: S/N: 4,0 dB.

Noise was more 'white' to the ear with more signal-to-noise needed for full intelligibility. Still DX beat the naked ear by over 2 dB.

 

- Wide band noise on 7,05 MHz. with voice break-in by next-door station (side-band whisper).

Processor nor naked ear could improve on this situation, probably because this 'noise' is also of a voice-type character. This will also happen when trying to break through a (voice) pile-up.

 

 

Some conclusions

 

Above tests and measurements are leading to a few interesting observations:

- A (trained!) human ear is capable of recognizing speech signals that are up to 6 dB below the noise; this may even go up to 9 dB depending on the particular type of noise.

- DX-11 will improve on the naked, but trained ear by 2,5 - 4 dB depending on noise 'coloring' and voice type.

- At a particular type of 'jamming', DX-11 may even improve up to 6 dB.

- If 'jamming' is at the speech repetition rate ( 5 - 10 Hz.), DX-11 improvement is lowering to around 2,5 dB.  

- At 'noise signals' that are speech-related (side channel break-in), no improvement may be noticed.

- At already well intelligible speech signals, DX-11 will further improve speech 'contrast' by lowering the noise considerably. This will make listening to these signals less tiresome. This type of speech enhancement therefore may also be useful in other noisy environments, e.g. communicating at a motorbike  

- Interesting to notice: DX-11 will 'improve' intelligibility of the lead-vocal in an orchestra; apparently the orchestra is also regarded as 'noise' by the DX-11.

- The DX-11 does not improve for other than speech type of signals (e.g. Morse-code, other modes). Apparently the system was not designed for this.

 

General conclusion is justified the DX-11 can make a difference when trying to (voice) communicate under adverse signal to noise conditions. The improvement is in the order of 3 - 6 dB S/N over the naked, but trained ear. With speech signals already intelligible, DX-11 will further improve quality by diminishing background noise. To obtain the same improvement without the DX-11, the antenna should be at least twice as effective in improving the signal or the transmitter should crack-up output power by a factor of 2 - 4, a challenge that is easily won by DX-11 from an economic point of view.

The test results were convincing enough to me to finally build this DX-11 into my ham-bus system, you may also find at this web-site.

 

Bob J. van Donselaar, on9cvd@veron.nl