摘要:
Adaptive speech coding includes receiving an original speech signal, performing on the original speech signal a current coding operation, and adapting the current coding operation in response to information used in the current coding operation. Adaptive speech decoding includes receiving coded information, performing a current decoding operation on the coded information, and adapting the current decoding operation in response to information used in the current decoding operation.
摘要:
In producing from an original speech signal a plurality of parameters from which an approximation of the original speech signal can be reconstructed, a coded signal of the original speech signal is generated. At least one of the parameters is determined using first and second differences between the original speech signal and the coded signal. The first difference is a difference between a waveform associated with the original speech signal and a waveform associated with the coded signal, and the second difference is a difference between an energy parameter derived from the original speech signal and a corresponding energy parameter associated with the coded signal.
摘要:
A post-processing method for a speech decoder which outputs a decoded speech signal in the time domain provides high frequency resolution based on a frequency spectrum having non-harmonic and noise deficiencies. This is obtained by transforming the decoded time domain signal to a frequency domain signal by using a high frequency resolution transform (FFT). Then an analysis of the energy distribution of the frequency domain signal is made throughout its frequency area (4 kHz) to find the disturbing frequency components and to prioritize such frequency components which are situated in the higher part of the frequency spectrum. Next, the suppression degree for the disturbing frequency components is found based on prioritizing. Finally the steps of controlling a post-filtering of the transform in dependence of the finding, and inverse transforming the post-filtered transform in order to obtain a post-filtered decoded speech signal in the time domain are performed.
摘要:
A speech encoding/decoding apparatus. A speech encoding apparatus has a coding portion for receiving input information related to an uncoded signal representative of an original speech signal, the coding portion including a fixed coding portion for receiving the input information and producing a first coded signal estimate, and an adaptive coding portion for receiving the input information and producing a second coded signal estimate. A controller is connected to the fixed coding portion and the adaptive coding portion for receiving information indicative of speech characteristics of the uncoded signal and generates a control signal; and a code modifier receives the first coded signal estimate from the fixed coding portion and the control signal from the controller and produces a modified signal estimate.
摘要:
An apparatus and method for reducing sparseness in a coded speech signal. Sparse codebook values are generated from a codebook. An anti-sparseness operation is performed on the sparse codebook values to produce output codebook values having a greater density of non-zero values than the sparse codebook values. The output codebook values are processed by a speech processor to generate an encoded speech signal during an encoding operation or a decoded speech signal during a decoding operation.
摘要:
Sparseness is reduced in an input digital signal which includes a first sequence of sample values. An output digital signal is produced in response to the input digital signal. The output digital signal includes a second sequence of sample values, which second sequence of sample values has a greater density of non-zero sample values than the first sequence of sample values.
摘要:
The quality of comfort noise generated by a speech decoder during non-speech periods is improved by modifying comfort noise parameter values normally used to generate the comfort noise. The comfort noise parameter values are modified in response to variability information associated with a background noise parameter. The modified comfort noise parameter values are then used to generate the comfort noise.
摘要:
A linear predictive analysis-by-synthesis encoder includes a search algorithm block (50) and a vector quantizer (58) for vector quantizing optimal gains from a plurality of subframes in a frame. The internal encoder states are updated (50, 52, 54, 56) using the vector quantized gains.
摘要:
A channel optimized vector quantization apparatus includes a device for weighting a sample vector x by a weighting matrix A and a device for weighting a set of code book vectors ĉr by a weighting matrix B. Device form a set of distance measures {dw(Ax,Bĉr)} representing the distance between the weighted sample vector Ax and each weighted code book vector Bĉr. Other device form a set of distortion measures {&agr;i(x)} by multiplying each distance measure by a channel transition probability Pr|i that an index r has been received at a decoder when an index i has been sent from an encoder and adding together these multiplied distance measures for each possible index r. Finally device determine an index imin corresponding to the smallest distortion measure &agr;i(x) and represents the sample vector by this index imin.
摘要:
A method and an apparatus for indicating presence of a transient noise in a call are provided. The method comprises the steps of determining activity at an endpoint of the call by monitoring presence of a signal input from the endpoint into the call and monitoring presence of a potential source of transient noise at the endpoint. Further, based on the activity determination and the monitoring of the presence of a potential source of transient noise, a signal representative of the presence of a transient noise in the call is sent. The present invention is advantageous in that it enables improvement of the quality of the call.