Multi-channel signal encoding and decoding

Data processing: speech signal processing – linguistics – language – Speech signal processing – For storage or transmission

Reexamination Certificate

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C704S219000, C704S222000

Reexamination Certificate

active

06393392

ABSTRACT:

TECHNICAL FIELD
The present invention relates to encoding and decoding of multi-channel signals, such as stereo audio signals.
BACKGROUND OF THE INVENTION
Existing speech coding methods are generally based on single-channel speech signals. An example is the speech coding used in a connection between a regular telephone and a cellular telephone. Speech coding is used on the radio link to reduce bandwidth usage on the frequency limited air-interface. Well known examples of speech coding are PCM (Pulse Code Modulation), ADPCM (Adaptive Differential Pulse Code Modulation), sub-band coding, transform coding, LPC (Linear Predictive Coding) vocoding, and hybrid coding, such as CELP (Code-Excited Linear Predictive) coding. See A. Gersho, “Advances in Speech and Audio Compression”, Proc. of the IEEE, Vol. 82, No. 6, pp. 900-918, June 1994; A. S. Spanias, “Speech Coding: A Tutorial Review”, Proc. of the IEEE, Vol. 82, No. 10, pp. 1541-1582, October 1994.
In an environment where the audio/voice communication uses more than one input signal, for example a computer workstation with stereo loudspeakers and two microphones (stereo microphones), two audio/voice channels are required to transmit the stereo signals. Another example of a multi-channel environment would be a conference room with two, three or four channel input/output. These types of applications are expected to be used on the internet and in third generation cellular systems.
From the area of music coding it is known that correlated multi-channels are more efficiently coded if a joint coding technique is used, an overview is given in P. Noll, “Wideband Speech and Audio Coding”, IEEE Commun. Mag. Vol. 31, No. 11, pp. 34-44, 1993. In B. Grill et al., “Improved MPEG-2 Audio Multi-Channel Encoding”, 96
th
Audio Engineering Society Convention, pp. 1-9, 1994, W. R. Th. Ten Kate et al., “Matrixing of Bit Rate Reduced Audio Signals”, Proc. ICASSP, Vol. 2, pp. 205-208, 1992, and M. Bosi et al., “ISO/IEC MPEG-2 Advanced Audio Coding”, 101
st
Audio Engineering Society Convention, 1996 a technique called matrixing (or sum and difference coding) is used. Prediction is also used to reduce inter-channel redundancy, see B. Grill et al., “Improved MPEG-2 Audio Multi-Channel Encoding”, 96
th
Audio Engineering Society Convention, pp. 1-9, 1994, W. R. Th. Ten Kate et al., “Matrixing of Bit Rate Reduced Audio Signals”, Proc. ICASSP, Vol. 2, pp. 205-208, 1992, M. Bosi et al., “ISO/IEC MPEG-2 Advanced audio Coding”, 101
st
Audio Engineering Society Convention, 1996, and EP 0 797 324 A2, Lucent Technologies, Inc., “Enhanced stereo coding method using temporal envelope shaping”, where the prediction is used for intensity coding or spectral prediction. Another technique known from WO 90/16136, British Teleom., “Polyphonic Coding” uses time aligned sum and difference signals and prediction between channels. Furthermore, prediction has been used to remove redundancy between channels in waveform coding methods. See WO 97/04621, Robert Bosch Gmbh, “Process for reducing redundancy during the coding of multi-channel signals and device for decoding redundancy reduced multi-channel signals”. The problem of stereo channels is also encountered in the echo cancellation area, an overview is given in M Mohan Sondhi et al., “Stereophonic Acoustic Echo Cancellation—An Overview of the Fundamental Problem”, IEEE Signal Processing Letters, Vol. 2, No. 8, August 1995.
From the described state of the art it is known that a joint coding technique will exploit the inter-channel redundancy. This feature has been used for audio (music) coding at higher bit rates and in connection with waveform coding, such as sub-band coding in MPEG. To reduce the bit rate further, below M (the number of channels) times 16-20 kb/s, and to do this for wideband (approximately 7 kHz) or narrowband (3-4 kHz) signals requires a more efficient coding technique.
SUMMARY OF THE INVENTION
An object of the present invention is to reduce the coding bit rate in multi-channel analysis-by-synthesis signal coding from M (the number of channels) times the coding bit rate of a single (mono) channel bit rate to a lower bit rate.
This object is solved in accordance with the appended claims.
Briefly, the present invention involves generalizing different elements in a single-channel linear predictive analysis-by-synthesis (LPAS) encoder with their multi-channel counterparts. The most fundamental modifications are the analysis and synthesis filters, which are replaced by filter blocks having matrix-valued transfer functions. These matrix-valued transfer functions will have non-diagonal matrix elements that reduce inter-channel redundancy. Another fundamental feature is that the search for best coding parameters is performed closed-loop (analysis-by-synthesis).


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