Automatic volume control for communication system

Electrical audio signal processing systems and devices – Including amplitude or volume control – Automatic

Reexamination Certificate

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Details

C381S086000, C381S057000, C381S324000, C381S186000

Reexamination Certificate

active

06674865

ABSTRACT:

FIELD OF THE INVENTION
The present invention relates to improvements in voice amplification and clarification in a noisy environment, such as a cabin communication system, which enables a voice spoken within the cabin to be increased in volume for improved understanding while minimizing any unwanted noise amplification. The present invention also relates to a movable cabin that advantageously includes such a cabin communication system for this purpose. In this regard, the term “movable cabin” is intended to be embodied by a car, truck or any other wheeled vehicle, an airplane or helicopter, a boat, a railroad car and indeed any other enclosed space that is movable and wherein a spoken voice may need to be amplified or clarified.
BACKGROUND OF THE INVENTION
As anyone who has ridden in a mini-van, sedan or sport utility vehicle will know, communication among the passengers in the cabin of such a vehicle is difficult. For example, in such a vehicle, it is frequently difficult for words spoken by, for example, a passenger in a back seat to be heard and understood by the driver, or vice versa, due to the large amount of ambient noise caused by the motor, the wind, other vehicles, stationary structures passed by etc., some of which noise is caused by the movement of the cabin and some of which occurs even when the cabin is stationary, and due to the cabin acoustics which may undesirably amplify or damp out different sounds. Even in relatively quiet vehicles, communication between passengers is a problem due to the distance between passengers and the intentional use of sound-absorbing materials to quiet the cabin interior. The communication problem may be compounded by the simultaneous use of high-fidelity stereo systems for entertainment.
To amplify the spoken voice, it may be picked up by a microphone and played back by a loudspeaker. However, if the spoken voice is simply picked up and played back, there will be a positive feedback loop that results from the output of the loudspeaker being picked up again by the microphone and added to the spoken voice to be once again output at the loudspeaker. When the output of the loudspeaker is substantially picked up by a microphone, the loudspeaker and the microphone are said to be acoustically coupled. To avoid an echo due to the reproduced voice itself, an echo cancellation apparatus, such as an acoustic echo cancellation apparatus, can be coupled between the microphone and the loudspeaker to remove the portion of the picked-up signal corresponding to the voice component output by the loudspeaker. This is possible because the audio signal at the microphone corresponding to the original spoken voice is theoretically highly correlated to the audio signal at the microphone corresponding to the reproduced voice component in the output of the loudspeaker. One advantageous example of such an acoustic echo cancellation apparatus is described in commonly-assigned U.S. patent application Ser. No. 08/868,212. Another advantageous acoustic echo cancellation apparatus is described hereinbelow.
On the other hand, any reproduced noise components may not be so highly correlated and need to be removed by other means. However, while systems for noise reduction generally are well known, enhancing speech intelligibility in a noisy cabin environment poses a challenging problem due to constraints peculiar to this environment. It has been determined in developing the present invention that the challenges arise principally, though not exclusively, from the following five causes. First, the speech and noise occupy the same bandwidth, and therefore cannot be separated by band-limited filters. Second, different people speak differently, and therefore it is harder to properly identify the speech components in the mixed signal. Third, the noise characteristics vary rapidly and unpredictably, due to the changing sources of noise as the vehicle moves. Fourth, the speech signal is not stationary, and therefore constant adaptation to its characteristics is required. Fifth, there are psycho-acoustic limits on speech quality, as will be discussed further below.
One prior art approach to speech intelligibility enhancement is filtering. As noted above, since speech and noise occupy the same bandwidth, simple band-limited filtering will not suffice. That is, the overlap of speech and noise in the same frequency band means that filtering based on frequency separation will not work. Instead, filtering may be based on the relative orthogonality between speech and noise waveforms. However, the highly non-stationary nature of speech necessitates adaptation to continuously estimate a filter to subtract the noise. The filter will also depend on the noise characteristics, which in this environment are time-varying on a slower scale than speech and depend on such factors as vehicle speed, road surface and weather.
FIG. 1
is a simplified block diagram of a conventional cabin communication system (CCS)
100
using only a microphone
102
and a loudspeaker
104
. As shown in the figure, an echo canceller
106
and a conventional speech enhancement filter (SEF)
108
are connected between the microphone
102
and loudspeaker
104
. A summer
110
subtracts the output of the echo canceller
106
from the input of the microphone
102
, and the result is input to the SEF
108
and used as a control signal therefor. The output of the SEF
108
, which is the output of the loudspeaker
26
, is the input to the echo canceller
106
. In the echo canceller
106
, on-line identification of the transfer function of the acoustic path (including the loudspeaker
104
and the microphone
102
) is performed, and the signal contribution from the acoustic path is subtracted.
In a conventional acoustic echo and noise cancellation system, the two problems of removing echoes and removing noise are addressed separately and the loss in performance resulting from coupling of the adaptive SEF and the adaptive echo canceller is usually insignificant. This is because speech and noise are correlated only over a relatively short period of time. Therefore, the signal coming out of the loudspeaker can be made to be uncorrelated from the signal received directly at the microphone by adding adequate delay into the SEF. This ensures robust identification of the echo canceller and in this way the problems can be completely decoupled. The delay does not pose a problem in large enclosures, public address systems and telecommunication systems such as automobile hands-free telephones. However, it has been recognized in developing the present invention that the acoustics of relatively smaller movable cabins dictate that processing be completed in a relatively short time to prevent the perception of an echo from direct and reproduced paths. In other words, the reproduced voice output from the loudspeaker should be heard by the listener at substantially the same time as the original voice from the speaker is heard. In particular, in the cabin of a moving vehicle, the acoustic paths are such that an addition of delay beyond approximately 20 ms will sound like an echo, with one version coming from the direct path and another from the loudspeaker. This puts a limit on the total processing time, which means a limit both on the amount of delay and on the length of the signal that can be processed.
Thus, conventional adaptive filtering applied to a cabin communication system may reduce voice quality by introducing distortion or by creating artifacts such as tones or echoes. If the echo cancellation process is coupled with the speech extraction filter, it becomes difficult to accurately estimate the acoustic transfer functions, and this in turn leads to poor estimates of noise spectrum and consequently poor speech intelligibility at the loudspeaker. An advantageous approach to overcoming this problem is disclosed below, as are the structure and operation of an advantageous adaptive SEF.
Several adaptive filters are known for use in the task of speech intelligibility enhancement. These filters can be broadly classified into two main

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