Adaptive Digital Filters

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9.6

• Adaptive digital filters are self learning filters, whereby an FIR (or IIR) is designed based on the characteristics of input signals. No other frequency response information or specification information is available. • There are a large number of applications suitable for the implementation of adaptive digital filters. • An adaptive digital filter is often represented by a signal flow graph with adaptive nature of weights shown:
x(k-1) x(k) Input w0 w1 w2 x(k-2) Adaptive Weights d(k)

y(k) Output

+
e(k) Error

August 2007, Version 3.8/21/07 For Academic Use Only. All Rights Reserved

Notes:

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An adaptive digital filter (FIR or IIR) will therefore “adapt” to its environment. The environment will be defined by the input signals x(k) and d(k) to the adaptive digital filter.

Developed by:

8/21/07 For Academic Use Only.7 “The aim is to adapt the digital filter such that the input signal x(k) is filtered to produce y(k) which when subtracted from desired signal d(k). Version 3.y(k) y(k) = Filter(x(k)) August 2007. will minimise the power of the error signal e(k).Intuitive Adaptive DSP • General Closed Loop Adaptive Signal Processor: Top 9.” desired signal input signal d(k) + Σ e(k) error signal x(k) Adaptive FIR Digital Filter Adaptive Algorithm y(k) Output signal - e(k) = d(k) . All Rights Reserved .

the length of the adaptive filter. Adaptive Filter Performance Obviously the key aim of the adaptive filter is to minimise the error signal e(k). The success of this minimisation will clearly depend on the nature of the input signals. IIR (recursive) or even a non-linear filter. adaptive IIR filters have become increasingly used in stable forms and in a number of real world applications (notably active noise control. As with any feedback system. Current research has highlighted a number of useful non-linear adaptive filters such as Volterra filters. music. At this stage of abstraction no information is given on the actual input signals which could be anything: speech. and the adaptive algorithm used. Filter Type The adaptive filter could be FIR (non-recursive). In the last few years however. The arrow through the adaptive filter is standard notation to indicate that the filter is adaptive and that all of the digital filter weights can be updated as some function of the error signal. desired. stability is a concern. y(k) and e(k) respectively is standard in most textbooks and papers and will be used in this presentation. d(k). hence care must be taken to ensure that any adaptive algorithm is stable.Notes: Naming Conventions and Notation: Top The naming of the signals as input. predefined noise and so on. Developed by: . digital data streams. and ADPCM techniques). vibration signal. output and error and denoted as x(k). and some forms of simple artificial neural networks. Most adaptive filters are FIR for reasons of algorithm stability and mathematical tractability.

All Rights Reserved .8 • The general adaptive signal processor can be implemented for real time applications using standard DSP components.Analogue Interfacing Top 9. Version 3.8/21/07 For Academic Use Only. DSP Board ADC fs ADC x(k) DSP Processor Adaptive Digital Filter Adaptive Algorithm fs fs d(k) y(k) + -Σ e(k) DAC DAC Analogue Signal Digital Signal fs August 2007.

.. Applications are discussed below.Sampling Frequency Top To implement the general adaptive signal processor.Analogue to Digital Converter. Developed by: . 20 bits. up to two ADC and two DAC converters are required.g. The algorithm designer must also ensure that the ADCs and DACs provide enough accuracy (e. 8 bits.Notes: ADC.) and most importantly that the DSP processor can operate fast enough to implement the required filter. fs . 16 bits. It is no use designing an adaptive filter that requires 1000 weights at a sampling rate of 48000 Hz unless a DSP processor architecture capable of performing this is available. DAC .Digital to Analogue Converter.. For example in a system identification scenario where a unknown system is being modelled there is no need for the DACs at y(k) and e(k). However an application will not always require these ADCs and DACs. The actual result of the filtering operation is the actual filter weights which model the response of the unknown system.

. s(k) + n)k) d(k) n’(k) x(k) Top 9.8/21/07 For Academic Use Only.. All Rights Reserved .9 Delay d(k) Adaptive Filter y(k) - Σ + e(k) s(k) Unknown System x(k) Adaptive Filter y(k) - Σ + e(k) Noise Cancellation Unknown System d(k) x(k) Inverse System Identification d(k) Adaptive Filter y(k) - Σ + e(k) s(k) Delay x(k) Adaptive Filter y(k) - Σ + e(k) System Identification Prediction August 2007.. Version 3.Architectures.

inverse system identification.Notes: In each of the above architectures the general adaptive signal processor can be clearly seen. and noise cancellation. For example the set up below includes elements of system identification. and inverse modelling “Unknown System 2”.to minimise the error signal e(k). To simplify the figures. then if s ( k ) is uncorrelated with r ( k ) then the error signal is likely to be e ( k ) ≈ s ( k ) : + Delay s(k) + Unknown System 1 d(k) r(k) Unknown System 2 x(k) Adaptive Filter y(k) + e(k) Developed by: . If the adaptive filter is successful in modelling “Unknown System 1”. A particular application may have elements of more than one single architecture. Top In each architecture the aim of the general adaptive signal processor is the same . the ADCs and DACs are not explicitly shown.

Application Examples • System Identification: • Channel identification. August 2007. CMDA interference suppression.8/21/07 For Academic Use Only. All Rights Reserved .10 • Inverse System Identification: • Digital communications equalisation. Version 3. Interference cancellation for CDMA • Prediction: • Periodic noise suppression. Periodic signal extraction. Speech coders. • Noise Cancellation: • Active Noise Cancellation. Echo Cancellation Top 9.

Sonar ~ 50 . Top Adaptive filtering has a tremendous range of applications from everyday equalisers on modems to less obvious applications such as adaptive tracking filters for predicting the movement of human eyes when following a moving stimulus! (research undertaken by S. • • • • • • • • Hi fidelity audio ~ 48kHz. Biomedical DSP~ 500 to 2000Hz.Notes: The sampling rate will vary depending on the particular application and the range of signal frequencies. Ultrasonic applications ~ MHz. MRC. Teleconferencing type applications ~ 16 kHz. Developed by: .100 kHz Radar ~ MHz. Voiceband telecommunications ~ 8 kHz. London). Low frequency active noise control ~ 1000Hz.Goodbody.

8/21/07 For Academic Use Only. DAC Comms Channel ADC d(k) x(k) Broadband Signal (White noise) Adaptive Filter y(k) + Σ - e(k) General Adaptive Signal Processor August 2007. and therefore produce a digital filter model of the room.11 • Applying a broadband input signal the adaptive filter will adapt to minimise the error. All Rights Reserved . Version 3.Channel Identification Top 9.

Adaptive System Identification Room Acoustic Identification If the architecture in the previous slide does indeed adapt. public address system design and so on. and record the impulse with a microphone and tape recorder. the adaptive filter will have the same same impulse (and frequency) response as the room. Developed by: . is to apply an impulse using “clappers” or a starting pistol. loudspeaker design. Improved impulse responses can be found by taking an ensemble average of around 20 impulses. h(t) Impulse Response H(f) DFT time frequency IDFT Calculating the impulse response is important to audio engineers working in applications such as architectural acoustics. it will travel to a specific point by the direct path. When an impulse is generated in a room. x(k). This technique can however be difficult and time consuming in practice. consider that the channel is a simple acoustic channel (from a loudspeaker to a microphone). One traditional method of finding the impulse response of a room. then over the frequency range of the input signal. and white noise correlation techniques are more likely.Notes: Top To intuitively appreciate the above example. Because the adaptive filter and the room were excited by the same signal. and also by many (first) echo or reflection paths and then by echoes of echoes (reverberation). Clearly the dimensions and the walls of the room will influence the impulse response. echo control systems. To find the frequency response of the room. car interior acoustics. the Fourier transform of the impulse response is taken. then the error will become very small.

Version 3.Echo Cancellation “. A Input Signal x(k) ADC Echo Path “.12 • Local line echo cancellation is widely used in data modems (V-series) and in telephone exchanges for echo reduction.g. Hybrid Telephone Connection A B “Hello” y(k) e(k) Simulated echo of A Σ d(k) + ADC B + echo of A “Hello” August 2007..8/21/07 For Academic Use Only..morning” .morning” Top 9. All Rights Reserved London Paris “..morning” Adaptive Filter Output Signal B “Hello” DAC e.

For V32 modems (9600 bits/sec with Trellis code modulation) an echo reduction ratio of 52dB is required. In general local echo cancellation (where the adaptive echo canceller is inside the consumer’s telephone/data communication equipment) is only used for data transmission and not speech. Therefore speaker A will hear an echo of their own voice which can be particularly annoying if the echo path from the near and far end hybrids is particularly long. Adaptive echo cancellers at telephone exchanges have however helped to solve this problem. At the other end of the line. To cancel both near-end and far-end an echo canceller is often presented in two sections. This is power reduction of around 160. telephone user B can also have an echo canceller. This is inconvenient for speakers who must take it in turn to speak. Further information on telecommunication echo cancellers can be found the textbooks referenced earlier. then a negative simulated echo can be added to cancel out the speaker A echo.) If we can suitably model the echo generating path with the adaptive filter.Notes: Top When speaker A (or data source A) sends information down the telephone line.000 in the echo. and the local hybrid is deliberately mismatched. Minimum specifications for the modem V series of recommendations can be found in the ITU (formerly CCITT) Blue Book. one for the near end echo. For long distance telephone calls where the round trip echo delay is more than 0. and one for the far end echo.1 seconds and suppressed by less than 40dB (this is typical via satellite or undersea cables) line echo on speech can be a particularly annoying problem. however for data transmission echo is very undesirable and must be removed. Before adaptive echo cancellers this problem would be solved by setting up speech detectors and allowing speech to be half duplex. Hence the requirement for a powerful DSP processor implementing an adaptive echo cancelling filter. (Some echo to the earpiece is often desirable for telephone conversation. Developed by: . mismatches in the telephone hybrids can cause echoes to occur.

...Acoustic Echo Cancellation Top 9. etc.) August 2007... Version 3..8/21/07 For Academic Use Only.etc A H1(f) “feedback” + d(k) y(k) Adaptive Filter x(k) Σ e(k) Adaptive Filter A “feedback” H2(f) - B Σ + B Room 1 B(t) + echo of A(t-1) + echo of B(t--2). A(t) + echo of B(t-1) + echo of A (t-2). (Note ADCs/DACs not shown above.13 • Speakerphone acoustic echo cancellation is a very suitable application for adaptive DSP:. Room 2 • With the loudspeaker and microphone in the same room the direct acoustic feedback path may cause problems. All Rights Reserved .

amplifiers. Hence unless the loudspeaker and microphones in each room are acoustically isolated. It is often thought that speakerphones are a relatively new innovation. and transmitted back into room 1 via loudspeaker 1.hence no acoustic echo problems!) Developed by: . and so on. Because the reverberation time of a an office type room can be up to a second or more. there is a direct feedback path which may cause stability problems and hence failure of the full duplex speakerphone. the adaptive filters are white noise trained when a connection is set up and may thereafter adapt on-line to an changes in the environment. which in turn is picked up by loudspeaker 1. Teleconferencing. When speaker A in room 1 speaks into microphone 1. ADC. or hands free telephony has a significant required for good adaptive filters. communication channels etc have been omitted to allow the problem to be clearly defined. However the speech from loudspeaker 2 will be picked up by microphone 2. For some commercially available teleconferencing systems. They are not. the speech will appear at loudspeaker 2 in room 2.Notes: Top In the above example. DACs. it is not unknown for the adaptive filters to have 1000’s of weights. Speakerphones in emergency telephone equipment for roadside public use have been available since the 1920s (albeit the phone at the exchange was not a speakerphone .

Mains Hum Noise Suppression Top 9.8/21/07 For Academic Use Only. Version 3. electrical mains hum can be removed from the ECG (electrocardiograph.14 • Using a 50 Hz noise reference. time Signal + Noise ADC d(k) ADC x(k) Adaptive Filter + y(k) - Σ e(k) time Noise Reference Signal time August 2007. heartbeat signal). All Rights Reserved .

NJ. Prentice Hall. Additional information on this area of work can be found in: W.Notes: Top The ECG main’s hum noise canceller is a classic example first presented by Widrow et al. EEGs EMGs). Englewood Cliffs. In this case the input signal was a reference of the mother’s heartbeat.. Developed by: . Baby’s + Mother’s heartbeat ADC d(k) Mother’s heartbeat ADC x(k) Adaptive Filter y(k) + Σ - e(k) “Baby’s heartbeat” Many companies now make very high resolution (> 22 bits) ADCs suitable for a wide variety of biomedical DSP applications (ECGs. Widrow also completed work on noise cancellation for foetal heart monitoring. Biomedical Digital Signal Processing. and frequently quoted in many texts and papers for example purposes. 1993.J. Hence the dominant mother’s heartbeat could be subtracted out to allow the doctor’s to observe the baby’s heartbeat. and the desired input was the foetus heartbeat plus the mother’s heartbeat sensed at the mother’s stomach. Tompkins.

a car or helicopter. Version 3. for example.Background Noise Suppression Top 9. All Rights Reserved .15 • Inside.8/21/07 For Academic Use Only. background engine noise can be reduced from a radio or telephone signal if an uncorrelated reference signal is available: Primary Microphone ADC Reference Microphone s(k)+n(k) n’(k) NOISE ADC Reference Microphone x(k) d(k) Adaptive Filter + y(k) Σ e(k) ≈ s(k) August 2007.

This signal should contain as little of the speech as possible otherwise the adaptive noise canceller will also try to cancel the speech signal.Notes: Top To reduce the engine noise picked up by the microphone. Therefore ideally the reference microphone is acoustically isolated from the primary microphone. or perhaps using a judiciously placed accelerometer in place of the reference microphone. a reference noise signal is required. This may be accomplished by using particular types of microphone. Reasonable levels of noise reduction can be achieved for noise cancellation architectures. In general the background noise will quasi-periodic (consider any type of reciprocating or rotating engine noise). Developed by: . and can improve voiceband communications from incoherent to understandable.

All Rights Reserved . August 2007.16 • To improve the bandwidth of a channel we can attempt to equalise a communication channel: Training Sequence “Virtual wire” d(k) x(k) Δ Training Sequence DAC Telephone Channel ADC Adaptive Filter y(k) + - Σ e(k) s(k) NEW YORK. USA GLASGOW. UK • Training sequence could be a PRBS standard.8/21/07 For Academic Use Only.Channel Equalisation s(k) Top 9. Version 3.

and hence a complex adaptive algorithm is required. Data Channel Equalisation The process of data channel equalisation is one of the most exploited areas of adaptive signal processing. IEEE Comms. pp. then when symbols are transmitted the impulse response will cause a symbol to spread over many time intervals. pp 32-45. See also the general adaptive DSP textbooks for more information. 1985. Ahmed. 1349-1387. In the last few years the availability of fast adaptive equalisers has led to modems capable of more than 28800 bits/s communication with 115200 bits/s (also using data compression) modems on the horizon. Most digital data communications (V32 modems for example) use some form of data channel equaliser. 29. Vol. In general for channels where the impulse response changes slowly. Recent advances in DSP Systems. Developed by: . Further Information R.. 45. Qureshi. a decision directed adaptive data equaliser is used. 5. it should be noted that a data equaliser only requires to equalise the channel at the symbol sampling instants rather than over all time. Lucky’s paper on adaptive equalisation (Bell Syst Tech. thus introducing intersymbol interference (ISI). May 1991. Compared to simple channel equalisation. If the above telephone channel is a (stationary) communication channel with a continuous time impulse response.M. J. A useful introduction can be found in: H. Adaptive Equalisation. the data is complex.. rather than the raw stochastic data (as in the slide). Vol. Hence the problem can be posed with data symbols as inputs. 73. No. Proceedings of the IEEE. It is also worth noting that for many data transmission systems. Mag. The aim of a data equaliser is to remove this ISI. A more recent paper is: S. Vol.Notes: Top If the above architecture successfully adapts (error is minimised) then the adaptive filter will produce an approximately inverse transfer function of the telephone channel. whereby a slicer is used to produce a retraining signal. 1966) defined the LMS algorithm for equalisation of digital communications and still is very relevant today.

Version 3. Δ. resulting in a filter which extracts the narrowband periodic signal at filter output y(k) (or removes the periodic noise from a wideband signal at e(k) ). All Rights Reserved .Adaptive Line Enhancer (ALE) Top 9.17 • The delay. d(k) time s(k) Δ s(k – Δ) Adaptive Filter y(k) - + Σ e(k) time Δ − decorrelation parameter time August 2007.8/21/07 For Academic Use Only. is long enough to decorrelate the broadband “noise-like” signal.

If the decorrelation parameter. whereas the additive noise is stochastic. Alternatively note that the ALE can be used to extract the periodic noise from the stochastic signal by observing the output e(k). Δ. an ALE could be used to extract periodic DTMF signals from very high levels of stochastic noise. In a telecommunications system. τ Correlation of a periodic (sine wave) signal −Δ Δ Lag. r(τ) r(τ) Lag. Perhaps one of the most imaginatively named of all signal processing algorithms was an adaptive line enhancer developed in the 1980s by Kevin Buckley and Lloyd Griffiths and named BASS-ALE! Developed by: . τ Correlation of a stochastic signal Typically an ALE may be used in communication channels or in radar and sonar applications where a low level sinusoid is masked by white or coloured noise. is long enough then the stochastic noise presented to the d(k) input is uncorrelated with the noise presented to the x(k) input.Notes: Top An adaptive line enhancer (ALE) exploits the knowledge that the signal of interest is periodic. however the periodic noise remains correlated.

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