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Audio signal decoding using complex-valued data

A complex and numerical technology, applied in speech analysis, instruments, etc., can solve the problems of phase adjustment or rotation impossible, SBR algorithm troubles, high cost, etc.

Inactive Publication Date: 2007-02-14
KONINKLIJKE PHILIPS ELECTRONICS NV
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Problems solved by technology

The second reason is that MDCT is a real-valued transform, and this makes phase adjustment or rotation practically impossible
attached figure 1 The disadvantage of the algorithm illustrated in is that, in addition to the base layer decoder, a complex-exponentially modulated filter bank is used, which is expensive both computationally and in terms of storage
The SBR algorithm proposed for mp3 suffers from the same shortcoming

Method used

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  • Audio signal decoding using complex-valued data
  • Audio signal decoding using complex-valued data
  • Audio signal decoding using complex-valued data

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Embodiment Construction

[0027] A typical conventional MPEG-1 layer III encoder (not shown) is arranged to receive a PCM input signal comprising a stream or frame of 1152 audio input samples. The input signal is supplied to a polyphase analysis filter bank which filters the input signal into 32 evenly spaced overlapping frequency bands to produce 32 downsampled subband signal components, each comprising 36 subband samples.

[0028] For each subband signal component, a windowed (forward) MDCT (Modified Discrete Cosine Transform) is performed. Four window types are used to cater for variable time segments. For (quasi-)stationary parts of the signal, so-called normal windows can be used, whereas, for non-stationary parts of the signal, a series of so-called short windows can be used. Two short types of windows (so-called start and stop windows) are defined to prevent discontinuities when switching from normal windows to short windows and vice versa. For normal, start and stop windows, MDCT is performed...

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Abstract

A decoder particularly, but not exclusively, for MPEG-1 layer III data signals, in which recovered spectral coefficients are transformed into time domain signal components, the time domain signal components then being transformed, using a forward transform which is orthogonally modulated with respect to the forward transform that was used at the encoder, to produce a set of second spectral coefficients. In this way, the first and second spectral coefficients may be used as complex-valued spectral coefficients which are amenable to post-processing. In the preferred embodiment, the complex-valued frequency components are, after post-processing, transformed to the time domain using an odd-frequency modulated Discrete Fourier Transform (DFT).

Description

technical field [0001] The invention relates to audio signal coding. The present invention relates particularly, but not exclusively, to decoding MPEG-1 Layer III data signals. Background technique [0002] MPEG-1 Layer III (commonly known as mp3) is a widely adopted audio codec. mp3 is introduced in ISO / IEC JTC1 / SC29 / WG11 MPEG, IS11172-3, Information Technology-Coding of Moving Pictures and Associated Audio for DigitalStorage Media at up to about 1.5Mbit / s, Part3: Audio, MPEG-1, 1992 industry standard. This standard is available from the International Organization for Standardization (ISO) (www.iso.ch) and is incorporated herein by reference. [0003] Advanced Audio Coding (AAC) has been devised to address some of the shortcomings of mp3. The AAC standard is introduced in ISO / IEC JTC1 / SC29 / WG11 MPEG, IS13818-3, Information Technology-Generic Coding of Moving Pictures and Associated Audio, Part3: Audio, MPEG-2, 1994, which is also available from ISO. [0004] As part of...

Claims

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Application Information

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IPC IPC(8): G10L19/02G10L19/26
CPCG10L19/0212G10L19/26G10L19/02G10L19/06
Inventor E·G·P·舒杰斯
Owner KONINKLIJKE PHILIPS ELECTRONICS NV
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