摘要:
This disclosure describes the use of non-dyadic discrete cosine transform (DCT) sizes for performing a DCT. Similarly, this disclosure describes the use of non-dyadic inverse discrete cosine transform (IDCT) sizes for performing an IDCT. Using non-dyadic transform sizes may be less computationally expensive compared to using conventional dyadic transform sizes. Aspects of this disclosure may be useful in any device or system that performs a DCT or IDCT.
摘要:
A more efficient encoder/decoder is provided in which an N-point MDCT transform is mapped into smaller sized N/2-point DCT-IV, DST-IV and/or DCT-II transforms. The MDCT may be systematically decimated by factor of 2 by utilizing a uniformly scaled 5-point DCT-II core function as opposed to the DCT-IV or FFT cores used in many existing MDCT designs in audio codecs. Various transform factorizations of the 5-point transforms may be implemented to more efficiently implement a transform.
摘要:
A more efficient encoder/decoder is provided in which an N-point MDCT transform is mapped into smaller sized N/2-point DCT-IV and/or DCT-II transforms with isolated pre-multiplications which can be moved to a prior or subsequent windowing stage. That is, the windowing operations may be merged with first/last stage multiplications in the core MDCT/IMDCT functions, respectively, thus reducing the total number of multiplications. Additionally, the MDCT may be systematically decimated by factor of 2 by utilizing a uniformly scaled 5-point DCT-II core function as opposed to the DCT-IV or FFT cores used in many existing MDCT designs in audio codecs. The modified windowing stage merges factors from a transform stage and windowing stage to obtain piece-wise symmetric windowing factors, which can be represented by a sub-set of the piece-wise symmetric windowing factors to save storage space. Such features offer appreciable reduction in complexity and less memory usage than the prior art.
摘要:
A complex analysis filterbank is implemented by obtaining an input audio signal as a plurality of N time-domain input samples. Pair-wise additions and subtractions of the time-domain input samples is performed to obtain a first and second groups of intermediate samples, each group having N/2 intermediate samples. The signs of odd-indexed intermediate samples in the second group are then inverted. A first transform is applied to the first group of intermediate samples to obtain a first group of output coefficients in the frequency domain. A second transform is applied to the second group of intermediate samples to obtain an intermediate second group of output coefficients in the frequency domain. The order of coefficients in the intermediate second group of output coefficients is then reversed to obtain a second group of output coefficients. The first and second groups of output coefficients may be stored and/or transmitted as a frequency domain representation of the audio signal.
摘要:
A more efficient encoder/decoder is provided in which an N-point MDCT transform is mapped into smaller sized N/2-point DCT-IV and/or DCT-II transforms with isolated pre-multiplications which can be moved to a prior or subsequent windowing stage. That is, the windowing operations may be merged with first/last stage multiplications in the core MDCT/IMDCT functions, respectively, thus reducing the total number of multiplications. Additionally, the MDCT may be systematically decimated by factor of 2 by utilizing a uniformly scaled 5-point DCT-II core function as opposed to the DCT-IV or FFT cores used in many existing MDCT designs in audio codecs. The modified windowing stage merges factors from a transform stage and windowing stage to obtain piece-wise symmetric windowing factors, which can be represented by a sub-set of the piece-wise symmetric windowing factors to save storage space. Such features offer appreciable reduction in complexity and less memory usage than the prior art.
摘要:
A complex analysis filterbank is implemented by obtaining an input audio signal as a plurality of N time-domain input samples. Pair-wise additions and subtractions of the time-domain input samples is performed to obtain a first and second groups of intermediate samples, each group having N/2 intermediate samples. The signs of odd-indexed intermediate samples in the second group are then inverted. A first transform is applied to the first group of intermediate samples to obtain a first group of output coefficients in the frequency domain. A second transform is applied to the second group of intermediate samples to obtain an intermediate second group of output coefficients in the frequency domain. The order of coefficients in the intermediate second group of output coefficients is then reversed to obtain a second group of output coefficients. The first and second groups of output coefficients may be stored and/or transmitted as a frequency domain representation of the audio signal.
摘要:
This disclosure presents techniques for implementing a fast algorithm for implementing odd-type DCTs and DSTs. The techniques include the computation of an odd-type transform on any real-valued sequence of data (e.g., residual values in a video coding process or a block of pixel values of an image coding process) by mapping the odd-type transform to a discrete Fourier transform (DFT). The techniques include a mapping between the real-valued data sequence to an intermediate sequence to be used as an input to a DFT. Using this intermediate sequence, an odd-type transform may be achieved by calculating a DFT of odd size. Fast algorithms for a DFT may be then be used, and as such, the odd-type transform may be calculated in a fast manner
摘要:
A more efficient encoder/decoder is provided in which an N-point MDCT transform is mapped into smaller sized N/2-point DCT-IV, DST-IV and/or DCT-II transforms. The MDCT may be systematically decimated by factor of 2 by utilizing a uniformly scaled 5-point DCT-II core function as opposed to the DCT-IV or FFT cores used in many existing MDCT designs in audio codecs. Various transform factorizations of the 5-point transforms may be implemented to more efficiently implement a transform.
摘要:
An encoder may include a core MDCT filterbank that can be used to implement an advanced audio coding (AAC) algorithm, an AAC-enhanced low delay (ELD) algorithm or both algorithms. For the AAC algorithm, a sequence of input samples is sent directly to the MDCT filterbank to obtain a sequence of output samples. For the AAC-ELD algorithm, the signs of input samples of the sequence of input samples are inverted, the MDCT analysis filterbank is applied to the sign-inverted sequence of input samples to obtain a sequence of output samples, the order of the sequence of output samples is reversed, and the signs of alternating output samples of the sequence of output samples are inverted. Similarly, a decoder may include a core IMDCT synthesis filterbank that can be used to implement AAC-ELD or both AAC and AAC-ELD algorithms. The steps for the decoder are merely the reverse of the encoder.