Abstract:
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.
Abstract:
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 (e0, e1, . . . en−1) which is mapped to a new exponent set (e0′, e1′, . . . , e′n−1) after coding, so as to satisfy: ∥e′i+1−e′i∥
Abstract:
A method and apparatus for subband phase flag determination for coupling of channels in a dual channel audio encoder is based on a psychoacoustic model of the human auditory system. The method and apparatus are applicable to audio encoders which utilize a coupling channel to combine certain frequency components of the input audio signals. The method ensures a least square error between the original channel frequency coefficients at the encoder and the estimated coefficients at the decoder by determining the sign of the dot product of the coefficients for one of the channels and the coupling coefficients. No restriction is placed on the strategy utilized for generating the coupling channel coefficients or the coupling coordinates.
Abstract:
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, Dmax, of said partial decimation vector to determine whether Dmax is less than one; e) testing a minimum decimation value, Dmin, of said partial decimation vector to determine whether Dmin is greater than one; and f) if said feasibility criterion is satisfied and Dmax is not less than one and Dmin is 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.
Abstract translation:一种搜索用于非均匀滤波器组的抽取因子的最佳匹配抽取向量的方法,所述最佳匹配向量允许非均匀滤波器组的输入信号的完美或接近完美的重建,所述方法包括以下步骤: :a)选择具有数字l抽取因子的部分抽取向量,其中l不超过所述最佳匹配抽取向量的抽取因子的最大数目K; b)测试所述l个抽取因子以确定所述部分抽取向量是否满足可行性标准; c)测试所述l个抽取因子的最小公式倍数值,以确定所述最小公倍数值是否大于预定值; d)测试所述部分抽取向量的最大抽取值D max max,以确定D MAX max是否小于1; e)测试所述部分抽取向量的最小抽取值D min min以确定D min min是否大于1; 以及f)如果满足所述可行性标准,并且D最大值不小于1,并且D分钟不大于1,则将数字l的抽取因子加1 在部分抽取向量中重复步骤b)至e)多次。
Abstract:
Channel coupling for an AC-3 encoder, using mixed precision computations and 16-bit coupling coefficient calculations for channels with 32-bit frequency coefficients.
Abstract:
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.
Abstract:
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.
Abstract:
A system, method and computer-readable medium are disclosed for improving the performance of a compiler. A set of source code instructions are processed to generate a plurality of source code instruction subsets, each of which is respectively associated with a mathematical operator. The source code subsets are then reordered to “hoist,” or place, a source code instruction subset associated with a product operator before a source code instruction subset associated with a summation operator. The plurality of source code instruction subsets are iteratively reordered until no source code instruction subset associated with a summation operator precedes a source code instruction subset associated with a product operator. A compiler is then used to compile the resulting reordered plurality of source code instruction subsets into a set of optimized object code instructions.
Abstract:
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.
Abstract:
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.