Hands-free Telephone Sets Research Articles

State detection for hands-free telephone sets by means of fuzzy LVQ and SOM

The step gain of adaptive algorithms for acoustic echo cancellation should vary with time in dependence of the far-end signal, the local signal, and the adaptation quality. It can be described as state-dependent, which is also the case for the control parameters of other algorithms in a hands-free telephone set, such as adaptive loss control or noise reduction. Most of the known state detection methods can only distinguish between subsets of the states, and are unreliable for certain state transitions, so that they should be combined for improved reliability of the state detection. In this contribution, state detection is performed by a fuzzy learning vector quantization/self-organizing map approach to combine several state detectors. The step gain is inferred from state information using fuzzy logic concepts. Finally, adaptation results are compared to the best conventional step-gain control method used in the complete structure.

Control of a hands-free telephone set

In this paper, we discuss different methods of double talk detection used for echo cancellation in a hands-free telephone set. We deduce a general state-space representation of the set, which leads to state-dependent structures for stepsize control. The approach is further generalized by introducing fuzzy logic and fuzzy state memberships. Finally, first results are shown which are obtained with identical detection methods, but applying different control structures.

Perfect sequence excitation of the NLMS algorithm and its application to acoustic echo control

Adaptive echo cancellers are currently being studied for applications such as audio teleconference systems or hands-free telephone sets with high speech quality. The purpose of the echo control is to eliminate the acoustic feedback from the loudspeaker to the microphone. One problem of echo cancellers using e.g. the normalized least mean square algorithm(nlms) for the adaptation of the coefficients is that the convergence properties degrade with colored signal input such as speech signals [8, 9, 17]. One approach to accelerate the convergence speed is to introduce linear prediction filters in order to decorrelate the speech signal [1, 2, 12, 15, 20]. This paper presents a new approach, named the excited lms algorithm orelms algorithm, which prewhitens the input signal applying perfect sequences. Coincidently, the proposed algorithm can be interpreted as a combination of the conventionalnlms algorithm and a system identification approach using m-sequences or related sequences.

Can the VLSI technology meet the needs of acoustic echo control system?

Since the introduction of hands-free telephone sets, acoustic echo cancellation has been a challenge for engineers. While in the past the problem was overridden using gain switching techniques, recently many authors have shown that the best solution for the echo compensation problem is represented by an adaptive filter which iteratively tries to identify the unknown impulse response of the system loudspeaker-room-microphone. Unfortunately the VLSI technology was not suitable to meet the requirements of this solution in a single chip acoustic echo canceller. The present paper, starting from a CMOS 1.2μm technology, will show some figures in terms of speed and silicon area taken for some basic blocks (such as adders, multipliers, memories) required by a canonical solution based on an adaptive FIR filter of 1000 taps with coefficients updated by the NLMS algorithm. Moreover, it will be shown a more efficient solution for the acoustic echo compensation based on a sub-bands approach. Simulated experiments with this architecture will be presented, compared with the ones of the canonical solution for what concerns size, number of operations required and performances.

A new prevention against the disturbance by narrow-band signal in hands-free telephone set with compact echo cancellers

A new prevention against the disturbance by narrow-band signal in hands-free telephone set with compact echo cancellers

Two-way loudspeaking device for telephone stations

A handsfree telephone set uses a single transducer as microphone and speaker. A switching arrangement senses and compares the relative strength between received and transmit signals, and switches signal paths accordingly. By sensing the input of the receive path versus the output of the transmit path, the detectors are thus asymmetrically coupled, to provide a break-in capability from one side (the telephone central or calling party) only.

General aspects of hands-free telephony

The use of a simple hands-free telephone set with distant talking and loudspeaker reception introduces transmission degradation not encountered with hand-set usage. Proper usage and good acoustic surroundings reduce these impairments to a point where satisfactory transmission is obtained on a large percentage of calls in today's plant which is designed to have transmission margin on a good proportion of all calls. There are several potentialities for improving the hands-free telephone set but it is not yet clear whether or not the gain will justify the added complexity or the added restrictions placed on the user. It is to be expected that very excellent performance could be obtained from a unit such as a booth which combines optimum instrument design with favorable environment.