Stmicroelectronics Asia Pacific Pte Limited

Mohammed Javed Absar

Sapna George

Antonio Mario Alvarez-Tinoco

Anotonio Mario Alvarez-Tinoco

AC-3 is a high quality audio compression format widely used in feature films and, more recently, on Digital Versatile Disks (DVD). For consumer applications the algorithm is usually coded into the firmware of a DSP Processor, which due to cost considerations may be capable of only fixed point arithmetic. It is generally assumed that 16-bit processing is incapable of delivering the high fidelity audio, expected from the AC-3 technology. Double precision computation can be utilized on such processors to provide the high quality; but the computational burden of such implementation will be beyond the capacity of the processor to enable real-time operation. Through extensive simulation study of a high quality AC-3 encoder implementation, a multi-precision technique for each processing block is presented whereby the quality of the encoder on a 16-bit processor matches the single precision 24-bit implementation very closely without excessive additional computational complexity.

Multi-precision technique for digital audio encoder

A method of searching for a best-match decimation vector of decimation factors for non-uniform filter bank, the best match vector allowing perfect or near-perfect reconstruction of an input signal of the non-uniform filter bank, the method including the steps of: a) selecting a partial decimation vector having a number, l, of decimation factors, where l does not exceed a maximum number, K, of decimation factors of said best-match decimation vector; b) testing said l decimation factors to determine whether said partial decimation vector satisfies a feasibility criterion; c) testing a least common multiplier value of said l decimation factors to determine whether said least common multiplier value is greater than a predetermined value; d) testing a maximum decimation value, D, of said partial decimation vector to determine whether Dis less than one; e) testing a minimum decimation value, D, of said partial decimation vector to determine whether Dis greater than one; and f) if said feasibility criterion is satisfied and Dis not less than one and Dis not greater than one, then incrementing by one the number l of decimation factors in the partial decimation vector and repeating steps b) to e) for a plurality of times.

Non-uniform filter bank implementation

A unified filter bank for use in encoding and decoding MPEG-1 audio data, wherein input audio data is encoded into coded audio data and the coded audio data is subsequently decoded into output audio data. The unified filter bank includes a plurality of filters, with each filter of the plurality of filters being a cosine modulation of a prototype filter. The unified filter bank is operational as an analysis filter bank during audio data encoding and as a synthesis filter bank during audio data decoding, wherein the unified filter bank is effective to substantially eliminate the effects of aliasing, phase distortion and amplitude distortion in the output audio data.

Unified filter bank for audio coding

A method of parametrically encoding a transient audio signal, including the steps of: determining a set V of the N largest frequency components of the transient audio signal, where N is a predetermined number; determining an approximate envelope of the transient audio signal; and determining a predetermined number P of samples W of the approximate envelope for use in generating a spline approximation of the approximate envelope, whereby a parametric representation of the transient audio signal is given by parameters including V, N, P and W, such that a decoder receiving the parametric representation can reproduce a received approximation of the transient audio signal.

Method and system for parametric characterization of transient audio signals

A method for effecting aliasing cancellation in an audio effects algorithm using a delay modulated signal, derived from interpolation of a delay modulator at an instantaneous sampling frequency, including: determining the instantaneous sampling frequency 1/Tand band limiting an input signal, to which the audio effects algorithm is to be applied to &#xbd; Tprior to interpolation.

Aliasing cancellation in audio effects algorithms

A method and apparatus for assigning an exponent coding strategy in a digital audio transform coder. Different coding strategies having different differential coding limits may be assigned to different set of transform exponents according to the frequency domain characteristics of the audio signal. A neural network processing system is utilised to perform an efficient mapping of each exponent set to an appropriate coding strategy.

Neural network based method for exponent coding in a transform coder for high quality audio

A technique for computationally efficient evaluation of the Modified Discrete Cosine Transform (MDCT) using the Fast Fourier Transform (FFT) method is presented in which the input of the FFT block consists of a sequence of N complex numbers, and this complex data is evaluated using an N/2-Point FFT only, thereby descreasing computation burden almost by two.

Fast modified discrete cosine transform method

A method for conversion of input audio frequency data, at an input sample frequency, to output audio frequency data, at an output sample frequency. The input data is subjected to expansion to produce expanded data at an output sample frequency. The expanded data is interpolated to produce output data. In one embodiment of the invention the interpolation is effected by a process that also filters the output data. In another embodiment, the input data is sampled by an integer factor to produce expanded data, the expanded data is then interpolated to produce the output data. Also disclosed is a method of transition of a signal output, at one frequency, to a signal output at another frequency. The signal output at said one frequency is faded out over a period, and the signal output at said other frequency is faded in over that period. Both signal outputs are combined to produce the signal output over said period. Apparatus for effecting the methods is also disclosed.

Universal sampling rate converter for digital audio frequencies

AC-3 is a high quality audio compression format widely used in feature films and, more recently, on Digital Versatile Disks (DVD). For consumer applications the algorithm is usually coded into the firmware of a DSP Processor, which due to cost considerations may be capable of only fixed point arithmetic. Commercial AC-3 Encoders have been successfully implemented on 20-bit and 24-bit word-length processors. However, it is generally assumed that 16-bit processing is incapable of delivering the high fidelity audio, expected from the AC-3 technology. Double precision computation can be utilised on such processors to provide the high quality; but the computational burden of such implementation will be beyond the capacity of the processor to enable real-time operation. Through extensive simulation study of a high quality AC-3 Encoder implementation, a multi-precision technique for each processing block is presented whereby the quality of the encoder on a 16-bit processor matches the single precision 24-bit implementation very closely without excessive additional computational complexity.

Channel coupling for an AC-3 encoder, using mixed precision computations and 16-bit coupling coefficient calculations for channels with 32-bit frequency coefficients.

Channel coupling for an AC-3 encoder

A method for encoding an audio signal, including providing a masking function, representative of psychoacoustic masking; setting a quality value for data of the encoded signal, adjusting the masking function dependent upon the quality value; and allocating bits for quantization of the encoded signal based on the incremental masking function.

Method of encoding an audio signal using a quality value for bit allocation

In a transform encoder for audio data, encoded data in the form of mantissas, exponents and coupling data is packed into fixed length frames in an output bitstream. The fields within the frame for carrying the different forms of data are variable in length, and apace within the frame must be allocated between them to fit all of the required information into the frame. The space required by the various data types depends on certain encoding parameters, which are calculated for a particular frame before the data is encoded, thus ensuring that the encoded data will fit into the frame before the computationally expensive encoding process is carried out. Information in relation to, for example, transform length, coupling parameters and exponent strategy are determined, which allows the space required for the coupling and exponent data to be calculated. The mantissa encoding parameters can then be iteratively determined so that the encoded mantissas will fit into the frame with the other encoded data. The determined encoding parameters are stored and the audio data is encoded according to those parameters after it has been determined that the encoded data will fit into the frame.

Fast frame optimization in an audio encoder

A method and apparatus for coding audio data in a frequency transform digital audio coder employing differential frequency coefficient exponent coding. Differential coding of exponents places constraints on possible values an exponent can take, which can lead to distortion in the decoded and reconstructed audio signal. The method and apparatus herein can overcome this restriction by mapping the input exponent set to a new set of values which satisfy the differential constraint as well as reducing information loss, thereby minimizing overall signal distortion due to coding restrictions.

Method and apparatus for spectral exponent reshaping in a transform coder for high quality audio

A method for encoding frequency coefficients in an AC-3 Encoder. The method includes: representing frequency coefficients in theform of a respective exponent and mantissa; coding the exponents; and shifting the mantissas to compensate for changes in the exponent values, wherein the exponents comprise an original exponent set (e , e , . . . e ) which is mapped to a new exponent set (e &prime;, e &prime;, . . . , e&prime; ) after coding, so as to satisfy: &par;e&prime; &minus;e&prime; &par;<D, where i&equals;0, . . . , n and D is a maximum allowed difference between two consecutive exponents, and e&prime; &lE;e .

Method of encoding frequency coefficients in an AC-3 encoder

A method and apparatus for determining parameters for coupling of channels in a digital audio encoder. A frequency range of two audio channels is coupled together in a coupling channel, and a systematic method of determining optimum coupling parameters is employed. Sub-bands of the channels are processed individually, and a measure of the power of each sub-band is used to determine a coupling coefficient generation scheme for each individual sub-band. Adjacent ones of the sub-bands using the same coupling scheme are combined to form bands in the coupling channel, which dictate the generation of the coupling coordinates for the audio channels. The arrangement of the sub-bands in bands also facilitates the generation of phase flags for each band, on the basis of the coupling scheme used in the band.

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Singapore, Singapore

Mohammed Javed Absar