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Improved excitation signal bandwidth extension

A technology of excitation signal and bandwidth, which is applied in the field of speech or voice decoding to achieve the effect of improving computing efficiency

Active Publication Date: 2014-04-16
TELEFON AB LM ERICSSON (PUBL)
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  • Abstract
  • Description
  • Claims
  • Application Information

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Problems solved by technology

However, replacing the actual residual (HB excitation) with white noise leads to perceptual degradation, since in certain parts of the speech signal the cycle continues in HB

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  • Improved excitation signal bandwidth extension
  • Improved excitation signal bandwidth extension
  • Improved excitation signal bandwidth extension

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Embodiment Construction

[0033] In the figures, elements having the same or similar functions will be given the same reference signs.

[0034] Before describing in detail several exemplary embodiments of the invention, reference will be made to figure 1 -5 A brief description will facilitate some of the concepts of this description.

[0035] figure 1 is a simple block diagram showing the general principles of source filter model based audio signal coding. The excitation signal e(k) is computed by filtering the waveform x(k) by an all-zero filter 110 having a transfer function A(z) defined by the filter coefficients a(j). In block 12, the filter coefficients a(j) are determined by linear prediction (LP) analysis. In this type of encoding, the input waveform of a signal x(k) is represented by an excitation signal e(k) and filter coefficients a(j), which are sent to the decoder.

[0036] figure 2 is a simple block diagram showing the general principle of source filter model based audio signal deco...

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Abstract

An apparatus for generating a high band extension of a low band excitation signal (eLB) defined by parameters representing a CELP encoded audio signal includes the following elements: upsamplers (20) configured to upsample a low band fixed codebook vector (uFCB) and a low band adaptive codebook vector (uACB) to a predetermined sampling frequency. A frequency shift estimator (22) configured to determine a modulation frequency (Ω) from an estimated measure representing a fundamental frequency (F0) of the audio signal. A modulator (24) configured to modulate the upsampled low band adaptive codebook vector (uACB↑) with the determined modulation frequency to form a frequency shifted adaptive codebook vector. A compression factor estimator (28) configured to estimate a compression factor. A compressor (34) configured to attenuate the frequency shifted adaptive codebook vector and the upsampled fixed codebook vector (uFCB↑.) based on the estimated compression factor. A combiner (40) configured to form a high-pass filtered sum of the attenuated frequency shifted adaptive codebook vector and the attenuated up-sampled fixed codebook vector.

Description

technical field [0001] The present invention relates generally to speech or speech decoding and, in particular, to bandwidth extension (BWE) of the excitation signal used in the decoding process. Background technique [0002] In many types of codecs, the input waveform is split into a spectral envelope and an excitation signal (also known as a residual), which are encoded and transmitted independently. At the decoder, a waveform is synthesized from the received envelope and excitation information. [0003] An efficient way to parameterize the spectral envelope is through the linear prediction (LP) coefficients a(j). The separation into spectral envelope and excitation signal e(k) consists of two main steps: 1) estimating the LP coefficients, and 2) passing through an all-zero filter [0004] A ( z ) = 1 - Σ j = 1 ...

Claims

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Application Information

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Patent Type & Authority Patents(China)
IPC IPC(8): G10L19/12G10L21/038
CPCG10L19/12G10L21/038
Inventor 西格德尔·斯维里森斯特凡·布鲁恩沃洛佳·格兰恰诺夫
Owner TELEFON AB LM ERICSSON (PUBL)
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