Digital processing of speech signals

A mobile terminal including a body; a touchscreen provided to a front and extending to side of the body and configured to display content; and a controller configured to detect one side of the body comes into contact with one side of an external terminal, display a first area on the touchscreen corresponding to a contact area of the body and the external terminal and a second area including the content, receive an input of moving the content displayed in the second area to the first area, display the content in the first area, and share the content in the first area with the external terminal.

Mobile terminal and method for controlling the same

The notion of perceptual coding, which is based on the concept of distortion masking by the signal being compressed, is developed. Progress in this field as a result of advances in classical coding theory, modeling of human perception, and digital signal processing, is described. It is proposed that fundamental limits in the science can be expressed by the semiquantitative concepts of perceptual entropy and the perceptual distortion-rate function, and current compression technology is examined in that framework. Problems and future research directions are summarized. >

Signal compression based on models of human perception

A novel method, free flow acoustophoresis (FFA), capable of continuous separation of mixed particle suspensions into multiple outlet fractions is presented. Acoustic forces are utilized to separate particles based on their size and density. The method is shown to be suitable for both biological and nonbiological suspended particles. The microfluidic separation chips were fabricated using conventional microfabrication methods. Particle separation was accomplished by combining laminar flow with the axial acoustic primary radiation force in an ultrasonic standing wave field. Dissimilar suspended particles flowing through the 350-μm-wide channel were thereby laterally translated to different regions of the laminar flow profile, which was split into multiple outlets for continuous fraction collection. Using four outlets, a mixture of 2-, 5-, 8-, and 10-μm polystyrene particles was separated with between 62 and 94% of each particle size ending up in separate fractions. Using three outlets and three particle siz...

Free flow acoustophoresis: Microfluidic-based mode of particle and cell separation

The vector sum excited linear prediction speech coder is presented. It utilizes a codebook with a structure that allows for a very efficient search procedure. Other advantages of the VSELP codebook structure are discussed, and a detailed description of an 8-kb/s VSELP coder is given. This coder was selected by the Telecommunications Industry Association (TIA) as the standard for use in North American digital cellular telephone systems. The coder uses two VSELP excitation codebooks, a gain quantizer which is robust to channel errors, and a novel adaptive pre/postfilter arrangement. >

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Vector sum excited linear prediction (VSELP) speech coding at 8 kbps

Vector Sum Excited Linear Prediction falls into the class of speech coders known as Code Excited Linear Prediction (CELP) (also called Vector Excited or Stochastically Excited) [1,4,5]. The VSELP speech coder was designed to achieve the highest possible speech quality with reasonable computational complexity while providing robustness to channel errors. These goals are essential for wide acceptance of low data rate (4.8-8 kbps) speech coding for telecommunications applications.

Vector Sum Excited Linear Prediction (VSELP)

A method (100) includes receiving (101) an input digital audio signal comprising a narrow-band signal. The input digital audio signal is processed (102) to generate a processed digital audio signal. A high-band energy level corresponding to the input digital audio signal is estimated (103) based on an estimated enery of a transition- band of the processed digital audio signal within a predetermined upper frequency range of a narrow-band bandwidth. A high-band digital audio signal is generated (104) based on the high-band energy level and an estimated high-band spectrum corresponding to the high-band energy level.

Method and apparatus for estimating high-band energy in a bandwidth extension system

A method includes defining a transition band for a signal having a spectrum within a first frequency band, where the transition band is defined as a portion of the first frequency band, and is located near an adjacent frequency band that is adjacent to the first frequency band. The method analyzes the transition band to obtain a transition band spectral envelope and a transition band excitation spectrum; estimates an adjacent frequency band spectral envelope; generates an adjacent frequency band excitation spectrum by periodic repetition of at least a part of the transition band excitation spectrum with a repetition period determined by a pitch frequency of the signal; and combines the adjacent frequency band spectral envelope and the adjacent frequency band excitation spectrum to obtain an adjacent frequency band signal spectrum. A signal processing logic for performing the method is also disclosed.

Bandwidth extension method and apparatus for a modified discrete cosine transform audio coder

A digital speech coder includes a long-term filter (124) having an improved sub-sample resolution long-term predictor (FIG. 5 ) which allows for subsample resolution for the lag parameter L. A frame of N samples of input speech vector s(n) is applied to an adder (510). The output of the adder (510) produces the output vector b(n) for the long term filter (124). The output vector b(n) is fed back to a delayed vector generator block (530) of the long-term predictor. The nominal long-term predictor lag parameter L is also input to the delayed vector generator block (530). The long-term predictor lag parameter L can take on non-integer values, which may be multiples of one half, one third, one fourth or any other rational fraction. The delayed vector generator (530) includes a memory which holds past samples of b(n). In addition, interpolated samples of b(n) are also calculated by the delayed vector generator (530) and stored in its memory, at least one interpolated sample being calculated and stored between each past sample of b(n). The delayed vector generator (530) provides output vector q(n) to the long-term multiplier block (520), which scales the long-term predictor response by the long-term predictor coefficient β. The scaled output βq(n) is then applied to the adder (510) to complete the feedback loop of the recursive filter (124).

Digital speech coder having improved sub-sample resolution long-term predictor

In a speech coder (100), excitation source gain information (802) is transmitted along with a coding mode indicator. The coding mode indicator indicates how the gain information is to be interpreted. In one embodiment, the coding mode indicator can also be utilized to control which of a plurality of excitation sources (202, 206-208) are utilized when synthesizing the speech. The coding mode itself is selected as a function of the periodicity of an input speech signal.

Mark A. Jasiuk

Papers

Vector Sum Excited Linear Prediction (VSELP)

Method and apparatus for estimating high-band energy in a bandwidth extension system

Bandwidth extension method and apparatus for a modified discrete cosine transform audio coder

Digital speech coder having improved sub-sample resolution long-term predictor

Provision of speech coder gain information using multiple coding modes