Adaptive Filters – Introduction
Gerhard Schmidt Christian-Albrechts-Universität zu Kiel Faculty of Engineering Institute of Electrical and Information Engineering Digital Signal Processing and System Theory
Contents of the Lecture Today: Boundary conditions
of the lecture
Contents Literature hints Exams
Notation Example of an adaptive
Filter Examples from speech and audio signal processing
Digital Signal Processing and System Theory| Adaptive Filters | Introduction
Slide I-2
Contents of the Lecture Entire Semester: Introduction with examples for speech and audio processing Wiener Filter Linear Prediction Algorithms for adaptive filters LMS und NLMS algorithm Affine projection RLS algorithm
Control of adaptive filters
Signal processing structures Applications of linear prediction Examples for speech and audio processing
Digital Signal Processing and System Theory| Adaptive Filters | Introduction
Slide I-3
Literature English and German Books: Statistical signal theory: A. Papoulis: Probability, Random E. Hänsler: Statistische
Variables, and Stochastic Processes, McGraw Hill, 1965
Signale: Grundlagen und Anwendungen, Springer, 2001
(in German)
Adaptive filters: E. Hänsler, G. Schmidt: Acoustic S. Haykin: Adaptive
Echo and Noise Control, Wiley, 2004
Filter Theory, Prentice Hall, 2002
A. Sayed: Fundamentals
of Adaptive Filtering, Wiley, 2004
Speech processing: L. R. Rabiner, R. W. Schafer: Digital Processing
of Speech Signals, Prentice Hall, 1978 P. Vary, R. Martin: Digital Speech Transmission, Wiley, 2006 L. R. Rabiner, R. W. Schafer: Introduction to Digital Speech Processing, Now, 2008
Digital Signal Processing and System Theory| Adaptive Filters | Introduction
Slide I-4
Boundary Contition of the Lecture Credit Points, Exams, Exercises, and Lecture Notes Credit points: 4 ECTS points
Oral exam: About 30 minutes per student In the exams period
Exercises: Two Matlab exercises during the semester
Talks: Duration about 10 minutes (afterwards short discussion) Topics will be offered during the lectures (own
suggestions are welcome)
Lecture notes: Printed versions will be spread at the beginning of each lecture In the internet via www.dss.tf.uni-kiel.de
Digital Signal Processing and System Theory| Adaptive Filters | Introduction
Slide I-5
Notation – Part 1 Scalars and Vectors Scalars:
Discrete time index
Signals:
Coefficient index
Impulse responses
(time-variant):
Example for a (real) convolution:
Vectors:
Boldface and lowercase
Signal vectors: Impulse response
vectors (time-variant) :
Example for a real convolution:
Matrices:
Boldface and uppercase Digital Signal Processing and System Theory| Adaptive Filters | Introduction
Slide I-6
Notation – Part 2 Random Processes Random variables and processes: Notation:
No differences between deterministic signals and random processes – different writing styles: Probability density function: Stationary random processes:
Expected values of stationary random processes:
Digital Signal Processing and System Theory| Adaptive Filters | Introduction
Slide I-7
Notation – Part 3 Correlation
Auto and cross correlation for real, stationary random processes: Auto-correlation function:
Cross-correlation
function:
(Auto) power spectral density:
(Cross)
power spectral density:
Digital Signal Processing and System Theory| Adaptive Filters | Introduction
Slide I-8
Notation – Part 4 White Noise Stationary white noise: Auto-correlation function:
Auto power spectral density:
Digital Signal Processing and System Theory| Adaptive Filters | Introduction
Slide I-9
A First Example of an Adaptive Filter – Part 1 Basic Structure
Unknown impulse response
Adaptive filter
+ +
Digital Signal Processing and System Theory| Adaptive Filters | Introduction
Local signals
Unknown system Slide I-10
A First Example of an Adaptive Filter – Part 2 Matlab Demo
Digital Signal Processing and System Theory| Adaptive Filters | Introduction
Slide I-11
Applications of Adaptive Filters Selected Application Areas Speech coding (e.g. GSM, UMTS)
Speech enhancement (hands-free systems, hearing aids, public address systems) Equalization (sending antennas, radar, loudspeakers) Anti-noise systems (cars and airplanes) Multi-channel signal processing (beamforming,
submarine localization, layer of earth analysis)
Missile control Medical applications (fetal heart rate monitoring, dialysis) Processing of video signals (cancellation of distortions, image analysis)
Antenna arrays
Digital Signal Processing and System Theory| Adaptive Filters | Introduction
Slide I-12
Basis Structures of Adaptive Filters – Part 1 System Identification
Unknown system
+ +
Adaptive filter
Examples: Line echo cancellation Cancellation of acoustical echoes
Digital Signal Processing and System Theory| Adaptive Filters | Introduction
Slide I-13
Basis Structures of Adaptive Filters – Part 2 Inverse Modelling
Unknown system
Adaptive filter
+ Delay Distortions are not depicted!
Examples: Equalization of amplifiers of transmission antennas Loudspeaker equalization
Digital Signal Processing and System Theory| Adaptive Filters | Introduction
Slide I-14
Basis Structures of Adaptive Filters – Part 3 Prediction
Delay
Adaptive filter
+
Examples: Speech coding in the GSM and UMTS networks Suppression of carrier signals after demodulation
Digital Signal Processing and System Theory| Adaptive Filters | Introduction
Slide I-15
Basis Structures of Adaptive Filters – Part 4 Cancellation of Undesired Signals
+
Adaptive filter
Example: Automotive speech signal enhancement via cancellation
of engine harmonics
Digital Signal Processing and System Theory| Adaptive Filters | Introduction
Slide I-16
Examples from Speech and Audio Processing Contents Part 1: Automotive hands-free telephone systems Basics Solutions Examples
Part 2: In-car communication systems Basics Solutions
Examples
Digital Signal Processing and System Theory| Adaptive Filters | Introduction
Slide I-17
Examples from Speech and Audio Processing Part 1
Automotive Hands-Free Telephone Systems
Digital Signal Processing and System Theory| Adaptive Filters | Introduction
Slide I-18
Automotive Hands-Free Telephone Systems Basics – Electro-Acoustic Transducers Microphones: Integrated in the rear-view
mirror (example)
Up to four microphones
Loudspeakers: Loudspeakers of the car stereo (head unit) coupling > 0 dB Volume adjustable by the passengers Digital Signal Processing and System Theory| Adaptive Filters | Introduction
Slide I-19
Automotive Hands-Free Telephone Systems Basics – Loudspeaker Enclosure Microphone (LEM) Systems – Part 1 Signal of the remote communication partner
: Excitation signal
: Echo (desired) signal : Local speech signal : Background noise
: Microphone signal
Microphone signal
Digital Signal Processing and System Theory| Adaptive Filters | Introduction
Slide I-20
Automotive Hands-Free Telephone Systems Basics – Loudspeaker Enclosure Microphone (LEM) Systems – Part 2
Assumption: FIR filter
The loudspeaker enclosure microphone system (LEM system) can be modeled as a linear system with finite memory.
+
+
Digital Signal Processing and System Theory| Adaptive Filters | Introduction
Slide I-21
Automotive Hands-Free Telephone Systems Basics – Loudspeaker Enclosure Microphone (LEM) Systems – Part 3
0.4
Boundary conditions: Volume
of a passenger compartment: 5 … 15 m³
0.3 0.2 0.1
Properties: Short delay
0
Direct -0.1
Early reflections Diffuse sound (decays
-0.2
logarithmically in amplitude)
-0.3 -0.4 0
sound after 3 … 4 ms
5
10
15
20 Time in ms
25
30
35
Digital Signal Processing and System Theory| Adaptive Filters | Introduction
40
Slide I-22
Automotive Hands-Free Telephone Systems Basics – Background Noise and its Components
External components: Engine noise
Wind noise Tire noise
Internal components: Air conditioning Defrost
Digital Signal Processing and System Theory| Adaptive Filters | Introduction
Slide I-23
Automotive Hands-Free Telephone Systems A Basic System With Two Adaptive Filters
Echo cancellation filter
Noise suppression filter
+
Digital Signal Processing and System Theory| Adaptive Filters | Introduction
Slide I-24
Automotive Hands-Free Telephone Systems An Adaptive Filter for Cancellation of Acoustical Echoes Loudspeaker
enclosure microphone system FIR model
Adaptive echo cancellation filter
+
(system parameters are unknown, only input and output signals are measurable)
+
+
Digital Signal Processing and System Theory| Adaptive Filters | Introduction
Slide I-25
Automotive Hands-Free Telephone Systems Maximal Achievable Echo Reduction – Part 1 Derivation during the lecture …
Digital Signal Processing and System Theory| Adaptive Filters | Introduction
Slide I-26
Automotive Hands-Free Telephone Systems Maximal Achievable Echo Reduction – Part 2 5
Boundary conditions: Maximum echo attenuation in relation to the filter length
0
White noise as
excitation signal
-5
Ideal convergence,
-10
meaning that all filter coefficients of the adaptive filter are equal to the corresponding ones of the impulse response.
dB
-15 -20 -25 -30
Linear loudspeakers, -35 -40
microphones, and amplifiers 0
50
100
150
200
250
300
350
400
450
Filter length
Digital Signal Processing and System Theory| Adaptive Filters | Introduction
Slide I-27
Automotive Hands-Free Telephone Systems A Basic System With Two Adaptive Filters
Echo cancellation filter
Noise suppression filter
+
Digital Signal Processing and System Theory| Adaptive Filters | Introduction
Slide I-28
Automotive Hands-Free Telephone Systems Residual Echo and Noise Suppression Remaining echoes....
... and local background noise
+
Local speech signal
Approach according to Wiener (next lecture):
Cross power spectral density of the distorted input signal and the desired output signal
Auto power spectral density of the distorted input signal Digital Signal Processing and System Theory| Adaptive Filters | Introduction
Slide I-29
Automotive Hands-Free Telephone Systems A Basic System With Two Adaptive Filters – Audio Examples
Stereo signals (16 kHz): Left:
Right:
Received signal ...
Sent signal ...
... of the remote communication partner
Initial filter convergence: Adaptation at the beginning of the call
Transmission to the communication partner (channel delay: about 180 ms)
Received signal („Hearing channel“ of the remote communication partner)
Double talk:
Remote communication partner
Enclosure dislocations:
Both partners speak simultaneously
Digital Signal Processing and System Theory| Adaptive Filters | Introduction
Without Wiener filter With Wiener filter Slide I-30
Automotive Hands-Free Telephone Systems Enhanced Systems
Improvements: Improved noise suppression
by adaptive combination of several microphone signals (beamforming)
Further
improvements by applying adaptive filters for different kinds of distortions
Digital Signal Processing and System Theory| Adaptive Filters | Introduction
Slide I-31
Automotive Hands-Free Telephone Systems Microphone Array Using Four Sensors (Integrated into the Rear-View Mirror)
Rear-view mirror
Microphone
module
Cheap realization by means of an integrated microphone module.
A fixed steering direction can be used for the driver – the steering angle varies only in a small range (62° - 75°).
The array can be used for the driver and for the passenger simultaneously.
Cardioid microphones are usually applied (± 3 dB sensitivity).
Digital Signal Processing and System Theory| Adaptive Filters | Introduction
Slide I-32
Automotive Hands-Free Telephone Systems Beamforming – Introduction
Desired signal
Distortion Adaptive filters
Beamformer: Minimizing the output power with respect to one or more constraints (signals from a desired direction must pass the structure without distortion) The desired direction is known in automotive applications (at least approximately) The performance of adaptive filtering is limited by sensor tolerances and multipath propagation within the passenger compartment Digital Signal Processing and System Theory| Adaptive Filters | Introduction
Slide I-33
Automotive Hands-Free Telephone Systems Beamforming – Adaptive Structure Output of the so-called generalized sidelobe canceller
Summation path
+
+ Delay
+
Adaptive filter
Blocking path
„Griffith-Jim“ beamformer (generalized sidelobe canceller)
Digital Signal Processing and System Theory| Adaptive Filters | Introduction
Slide I-34
Automotive Hands-Free Telephone Systems Beamforming – Audio Examples
4-channel beamformer
Loudspeaker on the passengers seat (undesired signal)
Adaptive filtering of the microphone signal results in an SNR improvement of about 15 dB.
Single microphone Fixed beamformer Adaptive beamformer
Digital Signal Processing and System Theory| Adaptive Filters | Introduction
Slide I-35
Automotive Hands-Free Telephone Systems Beamforming – Performance of Speech Recognition Systems Automotive wind, engine, tire noise Basic commands (120and km/h)
Noise a defroster Basic produced commands by (defrost on)
With permission from Eberhard Hänsler, Gerhard Schmidt, Acoustic Echo and Noise Control, Wiley, 2004
Speech and noise were mixed artificially to obtain different signal-to-noise ratios. About 30 command words for controlling the radio and phone system were used. 16 subjects (9 male, 7 female) participated in the test.
Digital Signal Processing and System Theory| Adaptive Filters | Introduction
Slide I-36
Automotive Hands-Free Telephone Systems Involved Signal Processing Units – Start
Microphone array
Acoustic coupling from the loudspeaker to the microphone(s)
Digital Signal Processing and System Theory| Adaptive Filters | Introduction
Telephone or speech dialog system
Slide I-37
Automotive Hands-Free Telephone Systems Involved Signal Processing Units – Bandwidth Extension Bandwidth extension Missing frequency components were estimated and resynthesized. Effect: The speech quality (not the intelligibility) of the received signal is improved. Microphone array
Acoustic coupling from the loudspeaker to the microphone(s)
Telephone or speech dialog system
Bandwidth extension
Digital Signal Processing and System Theory| Adaptive Filters | Introduction
Slide I-38
Automotive Hands-Free Telephone Systems Involved Signal Processing Units – Automatic Gain and Equalization Adjustment Volume and equalization control The (broadband) playback volume is adjusted automatically with respect to the noise measured in the car. In addition also the spectrum can be shaped in order to improve the perceived signal quality. Microphone array
Acoustic coupling from the loudspeaker to the microphone(s)
Telephone or speech dialog system
Adaptive volume and equalization control
Digital Signal Processing and System Theory| Adaptive Filters | Introduction
Bandwidth extension
Slide I-39
Automotive Hands-Free Telephone Systems Involved Signal Processing Units – Adaptive Limiter Adaptive limiter Adaptive adjustment of the parameters of a limiter in order to avoid microphone clipping by those loudspeakers that are close to the microphones (e.g. so-called center speaker). Microphone array
Acoustic coupling from the loudspeaker to the microphone(s)
Telephone or speech dialog system
Adaptive limiter
Adaptive volume and equalization control
Digital Signal Processing and System Theory| Adaptive Filters | Introduction
Bandwidth extension
Slide I-40
Automotive Hands-Free Telephone Systems Involved Signal Processing Units – Echo Cancellation Echo cancellation The signals emitted by the loudspeakers are reflected by windows, etc. These reflected signals as well as directly coupled signals are also recorded by the microphones. Microphone array
Acoustic coupling from the loudspeaker to the microphone(s)
To decouple the electro-acoustic system, the echo signals are estimated and subtracted from the microphone signal.
Telephone or speech dialog system
Echo cancellation
Adaptive limiter
Adaptive volume and equalization control
Digital Signal Processing and System Theory| Adaptive Filters | Introduction
Bandwidth extension
Slide I-41
Automotive Hands-Free Telephone Systems Involved Signal Processing Units – Beamforming Beamforming The microphone signals are filtered such that a predefined direction is kept open, while other directions are attenuated as much as possible. Effect: Directional distortions can be suppressed. Microphone array
Acoustic coupling from the loudspeaker to the microphone(s)
Beamforming
Telephone or speech dialog system
Echo cancellation
Adaptive limiter
Adaptive volume and equalization control
Digital Signal Processing and System Theory| Adaptive Filters | Introduction
Bandwidth extension
Slide I-42
Automotive Hands-Free Telephone Systems Involved Signal Processing Units – Noise and Residual Echo Suppression Background noise and residual echo suppression Despite beamforming and echo cancellation several remaining undesired signal components are still audible. Effect: Stationary background noise and residual echoes can be suppressed. Microphone array
Acoustic coupling from the loudspeaker to the microphone(s)
Noise and Beamforming echo suppression
Telephone or speech dialog system
Echo cancellation
Adaptive limiter
Adaptive volume and equalization control
Digital Signal Processing and System Theory| Adaptive Filters | Introduction
Bandwidth extension
Slide I-43
Automotive Hands-Free Telephone Systems Involved Signal Processing Units – Wind Buffet Removal Wind buffet suppression Open windows and defrost on might cause wind buffets. Effect: A detection optimized for those undesired signals finds wind buffets and replaces the signal with so-called comfort noise. Microphone array
Acoustic coupling from the loudspeaker to the microphone(s)
Noise and Wind buffet Beamforming echo suppression removal
Telephone or speech dialog system
Echo cancellation
Adaptive limiter
Adaptive volume and equalization control
Digital Signal Processing and System Theory| Adaptive Filters | Introduction
Bandwidth extension
Slide I-44
Automotive Hands-Free Telephone Systems Involved Signal Processing Units – Removal of “Transients” Suppression of transients Transient signal, such as the noise of an indicator or a wind shield wiper, cause problems for voice recognitions signals (voice activity detection). Effect: Short impulsive distortions are suppressed. Microphone array
Acoustic coupling from the loudspeaker to the microphone(s)
Noise and Wind buffet Beamforming echo suppression removal
Suppression of transients
Telephone or speech dialog system
Echo cancellation
Adaptive limiter
Adaptive volume and equalization control
Digital Signal Processing and System Theory| Adaptive Filters | Introduction
Bandwidth extension
Slide I-45
Automotive Hands-Free Telephone Systems Involved Signal Processing Units – Adaptive Equalization Adaptive equalization For compensation of different microphone-speaker distances and room characteristics, a (blind) equalization can be performed adaptively. Effect: The signal sounds more natural. Microphone array
Acoustic coupling from the loudspeaker to the microphone(s)
Noise and Wind buffet Beamforming echo suppression removal
Suppression of transients
Adaptive equalization
Telephone or speech dialog system
Echo cancellation
Adaptive limiter
Adaptive volume and equalization control
Digital Signal Processing and System Theory| Adaptive Filters | Introduction
Bandwidth extension
Slide I-46
Automotive Hands-Free Telephone Systems Involved Signal Processing Units – Summary Bandwidth extension
Volume and equalization control
Adaptive limiter
Missing frequency components were estimated and resynthesized. Effect: The speech quality (not the intelligibility) is improved.
The (broadband) playback volume is adjusted automatically with respect to the noise measured in the car. In addition also the spectrum can be shaped in order to improve the perceived signal quality.
Adaptive adjustment of the parameters of a limiter in order to avoid microphone clipping by those loudspeakers that are close to the microphones (e.g. so-called center speaker).
Echo cancellation
Beamforming
Noise and residual echo suppression
To decouple the electro-acoustic system, the echo signals are estimated and subtracted from the microphone signal.
The microphone signals are filtered such that a predefined direction is kept open, while other directions are attenuated.
Despite beamforming and echo cancellation several remaining undesired signal components are still audible.
Effect: Directional distortions can be suppressed.
Effect: Stationary background noise and residual echoes can be suppressed.
Wind buffet suppression
Suppression of transients
Adaptive equalization
Open windows and defrost on cause might cause wind buffets.
Transient signal, such as the noise of an indicator or a wind shield wiper, cause problems for voice recognitions signals.
For compensation of different microphonespeaker distances and room characteristics, a (blind) equalization can be performed adaptively.
Effect: A detection optimized for those signals finds wind buffets and replaces the signal with so-called comfort noise.
Effect: Short impulsive distortions are suppressed.
Digital Signal Processing and System Theory| Adaptive Filters | Introduction
Effect: The signal sounds more natural.
Slide I-47
Examples from Speech and Audio Processing Part 2
In-Car Communication Systems
Digital Signal Processing and System Theory| Adaptive Filters | Introduction
Slide I-48
In-Car Communication Systems Motivation Passenger compartment
Current situation: Communication between
passengers is difficult, because of the acoustic loss (especially front to rear).
Driver turns around – road safety is reduced.
Driving direction
-5 … -15 dB*
Front
passengers have to speak louder than normal – longer conversations will be tiring.
Solutions: Improve
the speech quality and intelligibility by means of an intercom system.
*Acoustic loss (referred to the ear of the driver)
Application:
Mid and high-class automobiles, which are already equipped with the necessary audio and signal processing devices.
Vans, etc. – systems with reduced complexity.
Digital Signal Processing and System Theory| Adaptive Filters | Introduction
Slide I-49
In-Car Communication Systems Algorithmic Overview
the speech quality and intelligibility by means of an ICC system. The ICC system records the speech by means of microphones and improves the communication by playing back the signals via those loudspeakers that are close to the listening passengers.
Loudspeakers
Front passenger
Solution: Improve
Rear passengers Microphones Driver Loudspeakers Passenger compartment Feedback and noise suppression
Mixer
Adaptive splitter, Automatic equalizer, delay, gain control, noise dependent limiter gain adjustment
Clipping detection, highpass filtering, speaker localization, beamforming
Feedback cancellation
Digital Signal Processing and System Theory| Adaptive Filters | Introduction
Slide I-50
In-Car Communication Systems Results of a Comparison Mean Opinion Score (CMOS) Test 0 km/h, car parked close to a motorway 19.7 % prefer the system
to be switched off 29.7 % have no preference
50.6 % prefer an activated
system
130 km/h, on a motorway
4.3 % prefer the system to be switched off
7.1 % have no preference
88.6 % prefer an activated
system With permission from Eberhard Hänsler, Gerhard Schmidt (eds.), Topics in Acoustic Echo and Noise Control, Springer, 2006
Digital Signal Processing and System Theory| Adaptive Filters | Introduction
Slide I-51
In-Car Communication Systems Diagnostic Rhyme Tests (DRT) and Modified Rhyme Tests (MRT) On a parking area beside motorway (0 km/h): No significant difference (95.2 system off versus 95.0 % system on). Due to the automatic gain adjustment the intercom system operates with
only very small gain at these noise levels.
On a motorway (130 km/h): Significant improvement
of the DRT error rate. Nearly 50 % error
reduction
(85.4 % correct answers increased to 92.2 % correct answers).
With permission from Eberhard Hänsler, Gerhard Schmidt (eds.), Topics in Acoustic Echo and Noise Control, Springer, 2006
Digital Signal Processing and System Theory| Adaptive Filters | Introduction
Slide I-52
Adaptive Filters – Introduction Summary and Outlook
This week: Boundary conditions of the lecture Contents Literature hints Exams Notation Example of an adaptive Filter Examples from speech and audio signal processing
Next week: Wiener filter Noise suppression
Digital Signal Processing and System Theory| Adaptive Filters | Introduction
Slide I-53