Skip to main content
Log in

The hands-free telephone problem: an annotated bibliography update

PrOblème de la Téléphonie mains-libres: mise à jour de la bibliographie commentée

  • Published:
Annales Des Télécommunications Aims and scope Submit manuscript

Abstract

Providing means for a full-duplex hands-free telephone conversation is still a research topic in many communication and signal processing laboratories around the world. More than ninety publications mainly from the last two and a half years support this statement. The reason for these activities lies in the complexity of the problem: full-duplex telephone communication without hand-sets requires electronic replicas of the loudspeaker-enclosure-microphone systems used by the communicating parties. The impulse response of such a system is typically several 100 ms long and time varying. Therefore, the replica has to be adaptive while the adaptation is controlled by speech signals. Regarding these requirements the hands-free telephone problem may currently be considered as one of the most challenging signal processing problems. This bibliography supplements a bibliography composed approximately two and a half years ago [29]. Brief summaries are given on papers published since then.

Résumé

L’étude des techniques permettant la communication mains-libres en duplex est toujours un sujet de recherche dans de nombreux laboratoires travaillant dans les télécommunications et le traitement du signal de par le monde. Cette observation est étayée par l’existence de plus de quatre-vingt dix publications parues principalement depuis deux ans et demi. La communication téléphonique duplex sans combiné nécessite la modélisation du système constitué par le haut-parleur, la salle et le microphone à chaque extrémité. La réponse impulsionnelle de tels systèmes a une longueur typique de plusieurs centaines de millisecondes et varie au cours du temps. Par conséquent, le modèle doit être adaptatif, l’adaptation étant contrôlée par le signal de parole. Compte tenu de ces diverses contraintes, la téléphonie mains-libres peut être considérée actuellement comme l’un des problèmes les plus difficiles en traitement du signal. La présente bibliographie complète une bibliographie établie il y a environ deux ans et demi [29] ; elle donne un bref résumé des articles publiés depuis lors.

This is a preview of subscription content, log in via an institution to check access.

Access this article

Price excludes VAT (USA)
Tax calculation will be finalised during checkout.

Instant access to the full article PDF.

Similar content being viewed by others

References

  1. Ait Amrane (O.), Moulines (E.), Charbit (M.), Grenier (Y.). Low-delay frequency domain lms algorithm.Proc. ICASSP-92, San Francisco, CA (1992),4, pp. 9–12.

    Google Scholar 

  2. Ait Amrane (O.), Moulines (E.), Grenier (Y.). Structure and convergence analysis of the generalized multi-delay adaptive filter.Proc. EUSIPCO-92, Brussels, Belgium (1992), pp. 115–118.

  3. Acker (C), Vary (P.). Combined implementation of predictive speech coding and acoustic echo cancellation.Proc. EUSIPCO-92, Brussels, Belgium (1992), pp. 1641–1644.

  4. Antweiler (C). Orthogonalisierende Verfahren zur Verbesserung von digitalen Freisprecheinrichtungen.Proc. 8. Aachener Kolloquium Signaltheorie, Aachen, Germany (1994), pp. 283–286.

  5. ArmbrÜster (W.). Wideband acoustic echo canceller with two filter structure.Proc. EUSIPCO-92, Brussels, Belgium (1992), pp. 1611–1614.

  6. Asharif (M. R.), Amano (F.). Hardware implementation of acoustic echo canceller based on fbaf algorithm.IEEE Workshop on VLSI Signal Processing, San Diego, CA (1990), pp. 191–200.

  7. Asharif (M. R.), Amano (F.). Frequency bin adaptive filtering (fbaf) algorithm and its application to acoustic echo cancelling.IEICE Trans., E 74 (1991), pp. 2276–2283.

  8. Benesty (J.), Li (S. W.), Duhamel (P.). A gradient-based adaptive algorithm with reduced complexity, fast convergence and good tracking characteristics.Proc. ICASSP-92, San Francisco, CA (1992),4, pp. 5-8.

  9. Benesty (J.), Duhamel (P.). A fast exact least mean square adaptive algorithm.IEEE Trans. on Signal Processing (1992),40, pp. 2904–2920.

    Article  MATH  Google Scholar 

  10. Binde (S.). Eine Adaptionssteuerung zur Kompensation akustischer Echos in Frequenzteilbändern.Proc. 8. Aachener Kolloquium Signaltheorie, Aachen, Germany (1994), pp. 287–290.

  11. Bisgaard (N.), Dyrlund (O.). dfs — ein neues digitales System zur Rückkopplungsunterdrückung in Hörgeräten.Audiologische Akustik (1991),5, pp. 166–177.

    Google Scholar 

  12. Boray (G. K.), Srinath (M. D.). Conjugate gradient algorithm for adaptive echo cancellation.Proc. ICASSP-92, San Francisco, CA (1992),4, pp. 13–15.

  13. Burger (T.), Schultheiß (U.). A robust acoustic echo canceller for a hands-free voice-controlled telecommunication terminal.Proc. of the Third European Conf. on Speech Communication and Technology, Berlin, Germany (1993), pp. 1809–1812.

  14. BUSTAMANTE (D. K.), WORRALL (T. L.), WILLIAMSON (M. J.). Measurement and adaptivr suppression of acoustic feedback in hearing aids.Proc. ICASSP-89, Glasgow, UK (1989), pp. 2017–2020.

  15. Canagarajah (C. N.), Rayner (P. J. W). Two-stage adaptive filtering techniques for noise cancellation in hearing aids.Proc. ICASSP-92, San Diego, CA (1992), pp. 521–524.

  16. Ceruti (R.), Serano (D.). International workshop on acoustic echo control.Signal Processing (1992),27, pp. 255–256.

    Article  Google Scholar 

  17. Chao (J.), Kawabe (S.), Tsujii (S.). A new iir adaptive echo canceller: give.Proc. IEEE Int. Conf. on Systems Engineering, Kobe, Jap. (1992), pp. 547–551.

  18. Chu (P. L.). Weaver ssb subband acoustic echo canceller.1993 ASSP Workshop on Applications of Digital Signal Processing to Audio and Acoustics, New Paltz, NY (1993).

    Google Scholar 

  19. Egelmeers (G. P. M.), Sommen (P. C. W.). Relation between reduced dimension time and frequency domain adaptive algorithm.Proc. EUSIPOC-92, Brussels, Belgium (1992), pp. 1065–1068.

  20. Egelmeers (G. P. M.). Decoupling of partition factors in partitioned block fdaf.Proc. ProRISC/IEEE Benelux Workshop on Circuits, Systems and Signal Processing, Houthalen, Belgium (1993), pp. 203–208.

  21. Elko (G. W), Goodwin (M. M.). Beam dithering: acoustic feedback control using a modulated-directivity loudspeaker array.Proc. ICASSP-93, Minneapolis (1993),1, pp. 173–176.

  22. Frenzel (R.), Hennecke (M. E.). Using prewhitening and stepsize control to improve the performance of the lms algorithm for acoustic echo compensation.Proc. ISCAS, San Diego, CA (1992), pp. 1930–1932.

  23. Frenzel (R.). Freisprechen in gestörter Umgebung.Fortschritt-Berichte VDI, Reihe 10, Nr. 228, VDI-Verlag, Düsseldorf, Germany (1992).

    Google Scholar 

  24. GIErlich (H. W.). New measurement methods for determining the transfer characteristics of telephone terminal equipment.Proc. ISCAS-92, San Diego, CA (1992), pp. 2069–2072.

  25. Gierlich (H. W.). Principle and application of a new test signal to determine the transfer characteristics of telecommunication systems.1993 ASSP Workshop on Applications of Digital Processing to Audio and Acoustics, New Paltz, NY (1993).

    Google Scholar 

  26. Gilloire (A.), PÉtillon (T.), Theodoridis (S.). Acoustic echo cancellation using fast rls adaptive filters with reduced complexity.Proc. ISCAS-92, San Diego, CA (1992), pp. 2065–2068.

  27. Gilloire (A.), Vetterli (M.). Adaptive filtering in subbands with critical sampling: analysis, experiments and application to acoustic echo control.IEEE Trans. Signal Processing (1992),40, pp. 1862–1875.

    Article  MATH  Google Scholar 

  28. Gudvangen (S.), Flockton (S. J.). Comparison of pole-zero and all-zero modelling of acoustic transfer functions.Electronics Letters (1992),28, pp. 1976–1978.

    Article  Google Scholar 

  29. Hänsler (E.). The hands-free telephone problem. An annotated bibliography.Signal Processing (1992),27, pp. 259–271.

    Article  Google Scholar 

  30. Hänsler (E.). The hands-free telephone problem.Proc. ISCAS-92, San Diego, CA (1992), pp. 1914–1917.

  31. Haneda (Y), Makino (S.), Kaneda (Y). Modeling of a room transfer function using common acoustical poles.Proc. ICASSP-92, San Francisco, CA (1992),2, pp. 213–216.

  32. Hart (J. E.), Naylor (P. A.), Tanrikulu (O.). Polyphase allpass iir structures for subband acoustic echo cancellation.Proc. of the Third European Conf. on Speech Communication and Technology, Berlin, Germany (1993), pp. 1813–1816.

  33. Heitkämper (P.), Walker (M.). Adaptive gain control for speech quality improvement and echo suppression.Proc. ISCAS-93, Chicago, Illinois (1993), pp. 455–458.

  34. Heitkämper (P.), Walker (M.). Adaptive gain control and echo cancellation for hands-free telephone systems.Proc. of the Third European Conf. on Speech Communication and Technology Berlin, Germany (1993), pp. 1077–1080.

  35. Heitkämper (P.). Ein Korrelationsmaßzur Feststellung von Sprecheraktivitäten.Proc. 8. Aachener Kolloquium Signaltheorie, Aachen, Germany (1994), pp. 97–100.

  36. Henkel (T.), Feldmann (J.). Freisprechen als Standardfunktion.NTZ (1992),45, pp. 596–599.

    Google Scholar 

  37. Hirano (A.), Sugiyama (A.). Convergence characteristics of a multi-channel echo canceller with strongly cross-correlated input signals — analytical results.Proc. of the 6th IEICEJ DSP Symposium, Fuji-Yoshida, Jap. (1991), pp. 144–149.

  38. Hirano (A.), Sugiyama (A.). A compact multi-channel echo canceller with a single adaptive filter per channel.Proc. ISCAS, San Diego, CA (1992), pp. 1922–1925.

  39. Hirano (A.), Sugiyama (A.). dsp implementation of a stereo echo canceller with a single adaptive filter per channel.Proc. of the 7th IEICEJ DSP Symposium, Fuji-Yoshida, Jap. (1992), pp. 190–195.

  40. Hirano (A.), Sugiyama (A.). dsp implementation and performance evaluation of a compact stereo echo canceller.Proc. of the Int. Workshop on Intelligent Signal Processing and Communication Systems, Sendai, Jap. (1993), pp. 356–361.

  41. Huhn (T.), Jentschel (H.-J.). Kombination von Geräuschreduktion und Echokompensation beim Freisprechen.Nachrichtentechnik, Elektronik (1993),43, pp. 274–280.

    Google Scholar 

  42. Jensen (S. H.). Acoustic echo canceller for hands-free mobile radiotelephony.Proc. EUSIPCO-92, Brussels, Belgium (1992), pp. 1629–1632.

  43. Kaelin (A.), Moschytz (G. S.). Linear echo cancellation using optimized orthogonal recursive filters.Proc. IEEE Int. Conf. on Systems Engineering, Kobe, Jap. (1992), pp. 173–176.

  44. Kaelin (A.), Lindgren (A. G.), Moschytz (G. S.). Linear echo cancellation using optimized recursive prefiltering.Proc. ISCAS-93, Chicago, Illinois (1993), pp. 463–466.

  45. Kates (J. M.). Feedback cancellation in hearing aids: results from a computer simulation.IEEE Trans. on Signal Processing (1991),39, pp. 553–562.

    Article  Google Scholar 

  46. Kellermann (W.). On the integration of subband echo cancellation into subband coding schemes.Proc. EUSIPCO-92, Brussels, Belgium (1992), pp. 123–126.

  47. Kellermann (W). On the impulse response of a microphone array for acoustic echo cancellation in hands-free telephony.Proc. 8. Aachener Kolloquium Signaltheorie, Aachen, Germany (1994), pp. 295–298.

  48. Van de Kerkhof (L. M.), Kitzen (W. J. W.). Tracking of a time-varying acoustic impulse response by an adaptive filter.IEEE Trans. on Signal Processing (1992),40, pp. 1285–1294.

    Article  MATH  Google Scholar 

  49. Kettler (F.), Hottenbacher (A.), Heinrichs (R.), Gierlich (H.W.). Verfahren zur subjektiven Qualitätsbeurteilung von Freisprecheinrichtungen.Fortschritte der Akustik-D AGA’94, Dresden, Germany (1994).

    Google Scholar 

  50. Kuo (S. M.), Chen (J.). New adaptive iir notch filter and its application to howling control in speakerphone system.Electronics Letters (1992),28, pp. 764–766.

    Article  Google Scholar 

  51. Kuo (S. M.), Pan (Z.). An acoustic echo canceller adaptable during double-talk periods using two microphones.Acoustic Letters (1992),15, pp. 175–178.

    Google Scholar 

  52. Kuo (S. M.), Chen (J.). Multiple-microphone acoustic echo cancellation system with the partial adaptive process.Digital Signal Processing (1993),3, pp. 54–63.

    Article  Google Scholar 

  53. Linhard (K.). Frequenzbereichsverfahren zur Echokompensation bei Störgeräuschen und Gegensprechen.Proc. 8. Aachener Kolloquium Signaltheorie, Aachen, Germany (1994), pp. 291–294.

  54. Mahalanobis (A.), Song (S.), Petraglia (M. R.), Mitra (S.K.). General structural subband decomposition of adaptive filters for system identification problems.IEEE Trans. on Circuits & Systems II (1993),40, pp. 375–381.

    Article  MATH  Google Scholar 

  55. Makino (S.), Kaneda (Y). Exponentially weighted step-size projection algorithm for acoustic echo cancellers.IEICE Trans. Fundamentals, E75-A (1992), pp. 1500–1508.

  56. Makino (S.), Kaneda (Y.), Koizumi (N.). Exponentially weighted stepsize nlms adaptive filter based on the statistics of a room impulse response.IEEE Trans. on Speech & Audio Processing (1993),1, pp. 101–108.

    Article  Google Scholar 

  57. Maloberti (F.), Panini (C.), Roncella (R.), Saletti (P.), Terreni (P.), Toncelli (C.), Torelli (G.), Troiani (M.). ASIC-based acoustic echo-canceller board for vme bus.European Trans. on Telecommunications (1992),3, pp. 125–136.

    Article  Google Scholar 

  58. Martin (R.), Vary (P.). Mehrkanalige Verfahren für die Störgeräuschunterdrückung und die Kompensation akustischer Echos.Proc. 8. Aachener Kolloquium Signaltheorie, Aachen, Germany (1994), pp. 299-302.

  59. Marx (J.). Kompensation akustischer Echos in Räumen.Fortschritte der Akustik-DAGA’94, Dresden, Germany (1994).

    Google Scholar 

  60. Mboup (M.), Bonnet (M.), Bershad (N.). Coupled adaptive prediction and system identification: a statistical model and transient analysis.Proc. ICASSP-92, San Francisco, CA (1992),4, pp. 1–4.

  61. Mboup (M.), Bonnet (M.). On the adequateness of iir adaptive filtering for acoustic echo cancellation.Proc. EUSIPCO-92, Brussels, Belgium (1992), pp. 111–114.

  62. Moustakides (G. V.), Theodoridis (S.). Fast Newton transversal filters — a new class of adaptive estimation algorithms.IEEE Trans. on Signal Processing (1991),39, pp. 2184–2193.

    Article  MATH  Google Scholar 

  63. Murano (K.), Unagami (S.), Amano (F.). Echo cancellation and applications.IEEE Communications Magazine (1990),28, pp. 49–55.

    Article  Google Scholar 

  64. Peng (W.), Kuo (S. M.). Asymmetric crosstalk-resistant adaptive noise canceller and its application in beamforming.Proc. ISCAS-92, San Diego, CA (1992), pp. 513–516.

  65. Pétillon (T.), GILLoire (A.), Theodoridis (S.). A comparative study of efficient transversal algorithms for acoustic echo cancellation.Proc. EUSIPCO-92, Brussels, Belgium (1992), pp. 119–122.

  66. Pétillon (T.), GILLoire (A.), Theodoridis (S.). Complexity reduction in fast rls transversal adaptive filters with application to acoustic echo cancellation.Proc. ICASSP-92, San Francisco, CA (1992),4, pp. 37–40.

  67. Petraglia (M. R.). Efficient adaptive filtering structures based on multirate techniques.Ph.D. Dissertation, University of California, Santa Barbara (1991).

  68. Petraglia (M. R.), MItra (S. K.). Generalized structural subband implementation of adaptive fir filters.Proc. ISCAS-92, San Diego, CA (1992), pp. 2184–2187.

  69. Petraglia (M. R.), Mitra (S. K.). Adaptive fir filter structure based on the generalized subband decomposition of fir filters.IEEE Trans. on Circuits & Systems II (1993),40, pp. 354–362.

    Article  MATH  Google Scholar 

  70. Poltmann (R.). A fast method for the nlms algorithm with time variant decorrelation filters.Submitted European Transactions on Telecommunications.

  71. Schönle (M.), Fliege (N.), Zölzer (U.). Parametric approximation of room impulse responses by multirate systems.Proc. ICASSP-93, Minneapolis, Minnesota (1993),1, pp. 153–156.

  72. Schönle (M.), Fliege (N.), Zölzer (U.). Parametric approximation of room impulse responses based on wavelet decomposition.1993 ASSP Workshop on Applications of Digital Signal Processing to Audio and Acoustics, New Paltz, NY (1993).

    Google Scholar 

  73. Schroeter (J.). Digital signal processing in audio and electroa-coustics.The Electrical Engineering Handbook (R. C. Dorf, Ed.),CRC-Press, Boca Raton (1993), pp. 395–406.

    Google Scholar 

  74. Schütze (H.). Stabilized fast adaptation algorithms for acoustic echo control.Proc. ISCAS-92, San Diego, CA (1992), pp. 1926–1929.

  75. Schütze (H.). Convergence of acoustic echo cancellers for hands-free telephones operating under feedback conditions.IEEE Trans. on Speech & Audio Processing (1993),1, pp. 257–260.

    Article  Google Scholar 

  76. Sommen (P. C. W.). Adaptive filtering methods: on methods to use a priori information in order to reduce complexity while maintaining convergence properties.Ph.D. Dissertation, Technische Universiteit Eindhoven, Netherlands (1992).

  77. Sommen (P. C. W.), Wilde (E. de). Equal convergence conditions for normal- and partitioned-frequency domain adaptive filters.Proc. ICASSP-92, San Francisco, CA (1992),4, pp. 69–72.

  78. Sondhi (M. M.), Morgan (D. R.). Acoustic echo cancellation for stereophonic teleconferencing.1991 ASSP Workshop on Applications of Digital Signal Processing to Audio and Acoustics, New Paltz, NY (1991).

    Google Scholar 

  79. Sugiyama (A.), Hirano (A.), Ma (Z.). A subband adaptive filtering algorithm with adaptive intersubband tap assignment.Proc. of the 8th DSP Symposium, Sendai, Jap. (1993), pp. 103–110.

  80. Thi (J.), Morgan (D. R.). Delayless subband active noise control.Proc. ICASSP-93, Minneapolis, Minnesota (1993),1, pp. 181–184.

  81. Usaoawa (T.), Ebata (M.). Remote control system using speech under noisy environment.Proc. 14th Int. Congr. Acoustic, Beijing, China (1992), G6–1.

  82. Walker (M.). Freisprechen als natürliche Kommunikation.Design & Elektronik (1993),6, pp. 32–38.

    Google Scholar 

  83. Wallace (R. B.), Goubran (R. A.). Parallel adaptive filter structures for acoustic noise cancellation.Proc. ISCAS-92, San Diego, CA (1992), pp. 525–528.

  84. Wang (R.), Hariani (R.). Acoustic feedback cancellation in hearing aids.Proc. ICASSP-93, Minneapolis, Minnesota (1993),1, pp. 137–140.

  85. Wang (R.), Harjani (R.). Suppression of acoustic oscillation in hearing aids using minimum phase techniques.Proc. ISCAS-93, Chicago, Ill (1993), pp. 818–821.

  86. Wehrmann (R.). Concepts of improving hands-free speech communication.Proc. ISCAS-92, San Diego, CA (1992), pp. 1918–1921.

  87. Wehrmann (R.). Acoustic echo control — a perceptual challenge.Signal Processing (1992),27, pp. 253–254.

    Article  Google Scholar 

  88. Yasukawa (H.), Furukawa (I.), Ishiyama (Y). Acoustic echo control for high quality audio teleconferencing.Proc. ICASSP-89, Glasgow, UK (1989), pp. 2041–2044.

  89. Yasukawa (H.), Ogawa (M.), Nishino (M.). Echo return loss required for acoustic echo controller based on subjective assessment.IEICE Trans. E (1991),74, pp. 692–705.

    Google Scholar 

  90. Yasukawa (H.), Shimada (S.). An acoustic echo canceller using subband sampling and decorrelation methods.IEEE Trans. on Signal Processing (1993),41, pp. 926–930.

    Article  Google Scholar 

  91. Zimmermann (S.), Williamson (G. A.). Performance properties of fixed pole adaptive filters.Proc. ISCAS-93, Chicago, Ill (1993), pp. 56–59.

Download references

Author information

Authors and Affiliations

Authors

Rights and permissions

Reprints and permissions

About this article

Cite this article

HÄnsler, E. The hands-free telephone problem: an annotated bibliography update. Ann. Télécommun. 49, 360–367 (1994). https://doi.org/10.1007/BF02999423

Download citation

  • Received:

  • Issue Date:

  • DOI: https://doi.org/10.1007/BF02999423

Key words

Mots clés

Navigation