Electrical audio signal processing systems and devices – Including frequency control – Having automatic equalizer circuit
Reexamination Certificate
1997-09-23
2001-06-26
Mei, Xu (Department: 2644)
Electrical audio signal processing systems and devices
Including frequency control
Having automatic equalizer circuit
C381S102000, C381S056000, C381S059000
Reexamination Certificate
active
06252968
ABSTRACT:
TECHNICAL FIELD
The invention relates to the audio reproduction where the quality of the acoustic source is affected by unknown and possibly time-varying characteristics of the reproduction equipment and the environment, and, more particularly, relates to the audio reproduction in mobile multimedia systems where the low-cost speakers and the constantly changing environment introduce distortions to audio signals.
DESCRIPTION OF THE PRIOR ART
Audio reproduction in a mobile multimedia system often suffers from distortions introduced by poor quality speakers, and environmental fluctuations.
The subject of audio quality enhancement has been researched in considerable detail over the years. The articles entitled “Digital Equalization of Room Acoustics” by J. N. Mourjopoulos in the Journal of the Audio Engineering Society, Vol. 42, No. 11, pp. 884-900 (November 1994) and “Digital Car Audio Systems” by J. Kontro in IEEE Transactions on Consumer Electronics, Vol. 39, No. 3, pp. 514-521 (August 1993), and the references contained therein provide some relevant background. The idea of using feedback of the audio source, modeling the reproduction medium as a filter, and inverse filtering (equalizing) the effects of the reproduction medium is central to most of these approaches. The mechanisms for estimation of the medium, and for equalization vary considerably. The aforementioned Mourjopoulos article studies the problems encountered in using inverse filters. Primarily, since the impulse response of the reproduction medium tends to be long, the length of an inverse filter is also long, leading to computationally intensive algorithms. Further, a number of algorithms for implementing inverse filters tend to be unstable. The aforementioned Mourjopoulos article presents a method where the length of the inverse filter is shortened by using all-pole modeling and vector quantization of responses of the reproduction medium. The aforementioned Kontro article describes an audio system using an equalizer for gain control and for compensating for the medium's frequency response. The approach is computationally intensive, and is not intended for adaptive use. Once the medium's frequency response is measured, the equalizer parameters are fixed. This approach is reasonably good, but only for static environments, and in addition, it is quite computationally complex.
SUMMARY OF THE INVENTION
The invention addresses the problem of acoustic quality enhancement for such and similar systems, where the subjective quality of the audio source is affected by unknown and possibly time-varying characteristics of the reproduction equipment and the environment. The invention presupposes that the computational complexity of the proposed solution must be kept to a minimum because mobile systems have limited resources, and that the solution should not result in excessive delays in audio source reproduction. The invention provides a means for estimating and compensating for the undesirable characteristics while minimizing both the computational complexity and the delay in audio source reproduction as required, and allow subsequent reproduction of an audio source that is better matched to the intended audio output.
This invention proposes to estimate the characteristics of the reproduction medium using a training signal consisting of a set of pure frequency tones generated solely for the purpose of training, which also satisfies the low-complexity and short delay requirements described above, since the proposed filters that equalize the characteristics of the reproduction medium have short lengths and the filter coefficients may be calculated with minimal complexity due to the simplicity of the training signal. Furthermore, this invention addresses the problem of acoustic quality enhancement in a dynamic environment, as opposed to the static environments considered in the prior art, since we propose to use the existing microphone and speakers, which form integral components of a mobile multimedia system. Thus, the process of estimating and compensating for the undesirable characteristics of the reproduction medium may be done adaptively and repeatedly as deemed necessary.
Consider an audio source, amplified and then reproduced through a set of speakers. A microphone is used to feed back the reproduced audio source, into a processing mechanism. This processing mechanism in turn, controls subsequent audio reproduction. The processing mechanism may operate in two phases. In the first phase, which is the training phase, the medium's characteristics will be estimated, and a set of filters is constructed, with fixed parameters. The set of filters will subsequently pre-filter the audio source, in order to equalize for the medium's characteristics, during the second phase which is the processing phase. If necessary, the pre-filter parameters may be updated by feedback of the reproduced audio source, even after the initial training period.
According to this invention, during the training phase, unique frequency tones are transmitted (e.g., via speakers), and then recorded (e.g., via a microphone). Each fed-back audio frequency tone is then used to estimate the gain of the reproduction medium at that particular frequency, and the background noise parameters at that frequency are also determined. This invention is used to construct a set of inverse filters, so the original audio source can then be pre-filtered to produce the desired audio output.
During the second phase, which is the processing phase for playing back an audio source, the audio source is decomposed into sub-bands whose center frequencies are the frequency tones used for training. In each sub-band, the audio signal component is pre-emphasized by the gain estimates obtained during training, and also inverse filtered using the parameter estimates obtained during training. The resulting signal is then reconstructed into a full-band signal, resulting in an actual audio output signal that is better matched to the intended audio output.
REFERENCES:
patent: 5297198 (1994-03-01), Butani et al.
patent: 5600718 (1997-02-01), Dent et al.
patent: 5680450 (1997-10-01), Dent et al.
Narasimhan Anand
Ramaswamy Ganesh Nachiappa
Cameron Douglas W.
Doughtery Anne Vachon
International Business Machines Corp.
Mei Xu
LandOfFree
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