Data processing: speech signal processing – linguistics – language – Audio signal time compression or expansion – With content reduction encoding
Reexamination Certificate
2000-09-08
2003-06-03
{haeck over (S)}mits, T{overscore (a)}livaldis Ivars (Department: 2641)
Data processing: speech signal processing, linguistics, language
Audio signal time compression or expansion
With content reduction encoding
C704S200100, C704S205000, C704S229000
Reexamination Certificate
active
06574602
ABSTRACT:
TECHNICAL FIELD
This invention relates in general to digital signal processing of audio signals, such as music signals. More particularly, the invention relates to the implementation of a high quality dual-channel digital audio encoder, based on the psychoacoustic model of the human auditory system, for digital storage or transmission.
BACKGROUND ART
In order to more efficiently broadcast or record audio signals, the amount of information required to represent the audio signals may be reduced. In the case of digital audio signals, the amount of digital information needed to accurately reproduce the original pulse code modulation (PCM) samples may be reduced by applying a digital compression algorithm, resulting in a digitally compressed representation of the original signal. The goal of the digital compression algorithm is to produce a digital representation of an audio signal which, when decoded and reproduced, sounds the same as the original signal, while using a minimum of digital information for the compressed or encoded representation.
Recently, the use of psychoacoustic models in the design of audio coders has led to high compression ratios while keeping audible degradation in the compressed signal to a minimum. Description of one such method can be found in the Advanced Television Systems Committee (ATSC) Standard document entitled “Digital Audio Compression (AC-3) Standard”, Document A/52, Dec. 20, 1995. In the basic approach, the time domain signal is first converted to frequency domain using a bank of filters. Frequency domain masking of human auditory system is then exploited to maximise perceived fidelity of the signal transmitted at a given bit-rate.
Further compression can be successively obtained by use of a well known technique called coupling. Coupling takes advantage of the way the human ear determines directionality for very high frequency signals, in order to allow a reduction in the amount of data necessary to code an audio signal. At high audio frequency (approximately above 2 KHz) the ear is physically unable to detect individual cycles of an audio waveform, and instead responds to the envelope of the waveform. Consequently, the coder combines the high frequency coefficients of the individual channels to form a common coupling channel. The original channels combined to form the said coupling channel are referred to as coupled channels.
A basic encoder can form the coupling channel by simply taking the average of all the individual channel coefficients. A more sophisticated encoder can alter the sign of individual channels before adding them into the sum so as to avoid phase cancellations.
The generated coupling channel is next sectioned into a number of frequency bands. Frequency sub-bands are grouped together to form coupling bands. For each such band and each coupled channel a coupling co-ordinate is transmitted to the decoder. To obtain the high frequency coefficients in any frequency band, for a particular coupled channel, from the said coupling channel, the decoder multiplies the coupling channel coefficients in that frequency band by the coupling co-ordinate of that channel for that particular frequency band. For a dual channel implementation of such a decoder, a phase flag bit may also be provided for each coupled band of the coupling channel. A final step of phase-correction is then performed, by the decoder, in which the coefficients in each band are multiplied by the phase flag bit for that band.
The standard does not outline any specific method for determination of the phase flag bits. Ad hoc methods do exist but, due to their very nature, do not guarantee any assured performance, nor can be relied upon to provide minimum error between the original coefficients at encoder and the reconstructed, phase corrected, coefficients at the decoder.
SUMMARY OF THE INVENTION
The phase flag for a bard is, essentially, a function of the coefficients of the original channel and the coefficients of the coupling channel, in that band. Embodiments of the invention aim to minimise the difference between the original coefficients at the encoder and the reconstructed coefficients at the decoder.
In accordance with the present invention, there is provided a method for computing a phase reconstruction coefficient in a dual channel digital audio encoder having first and second encoded channels and a coupling channel, comprising computing transform coefficients for said first and second channels, computing coupling coefficients from the transform coefficients of the first and second channels, and computing a dot product of the corresponding transform coefficients for one of the first and second channels and the coupling coefficients, and determining the sign of the computed dot product.
The present invention also provides a method for computing a phase reconstruction coefficient in a dual channel digital audio encoder having first and second encoded channels and a coupling channel, comprising determining transform coefficients for one of the first and second channels, determining coupling coefficients from the first and second channels, and computing the sign of the sum of corresponding transform and coupling coefficients over a predetermined frequency range of the coefficients.
Preferably the method includes computing a phase reconstruction coefficient for each of a plurality of coupling frequency bands for said one of the first and second channels.
Preferably the phase reconstruction coefficient computation is independent of a method used for computing the coupling coefficients, and independent of a method used for computing coupling coordinates for said first and second channels.
The present invention also provides a dual channel encoder for coding of audio information which generates a coupling channel with at least one coupling band, comprising means for computing a dot product of input channel transform coefficients and coupling channel coefficients in said at least one coupling band, and means for determining the sign of said dot product for use as a phase flag corresponding to the at least one coupling band.
The present invention also provides a dual channel encoder for coding of audio information which generates a coupling channel and phase estimation data such that a difference between original coupled channel coefficients generated at the encoder and channel coefficients estimated at a compatible decoder have a least square error.
There is further provided, in accordance with the present invention, an encoder for transform coding digital audio information from first and second channels, the encoder producing a coupling channel arranged in at least one frequency band and at least one phase flag corresponding to the at least one frequency band, wherein the at least one phase flag is computed according to:
phase flag=sign(&Sgr;(
b
i
*c
i
))
wherein
b
i
represents transform coefficients for one of the first and second channels,
c
i
represents transform coefficients for the coupling channel, and
index i extends over the frequency range of the band.
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Advanced Television Systems Committee (ATSC) Digital Audio Compression (AC-3) Standard, Doc. A/52, Dec. 20, 1995, p. 1-130.*
Steve Vernon, “Design and Implementation of AC-3 Coders,” IEEE Trans. Consumer Electronics, vol. 41, No. 3, Aug. 1995, p. 754-759.
Absar Mohammed Javed
Alvarez-Tinoco Antonio Mario
George Sapna
Iannucci Robert
Jorgenson Lisa K.
Seed IP Law Group PLLC
STMicroelectronics Asia Pacific PTE Limited
{haeck over (S)}mits T{overscore (a)}livaldis Ivars
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