Abstract: Estimating a signal power in a compressed audio signal [A] is provided, the audio signal comprising blocks of quantized samples, a given block being provided with a set of scale factors. The estimating is performed by extracting the set of scale factors from the compressed audio signal, and estimating the signal power in the given block based on a combination of the scale factors. Advantageously, the extracting step and estimating step are performed on only a sub-set of the set of scale factors. The signal power estimation may be used in a silence detector (11) for use in a receiver (1).

Abstract: A method and system for testing voice applications, such as VoiceXML applications, is provided. The system provides a run-time simulation environment for voice applications that simulates and automates user interaction. A user simulation script is provided in a customized mark-up language. The voice application is processed to derive a nominal output of the voice application. The user simulation script is processed to generate a simulated output for the voice application corresponding to the nominal output. The user simulation script is also processed to generate a simulated input for the voice application corresponding to a pre-determined user input to the voice application. The simulated inputs and outputs are executed in conjunction with the voice application in real-time to test the application. Execution time rates for text based input and output are calculated based on empirical speaking rates of human users.

Abstract: An information processing apparatus for inputting a pronunciation symbol corresponding to an English notation includes pronunciation symbol information holding means for holding pronunciation symbol information indicating a relationship between a predetermined alphabet and a pronunciation symbol that starts from the predetermined alphabet, pronunciation symbol statistical information holding means for holding statistical information associated with the probability of occurrence of each pronunciation symbol immediately after a predetermined pronunciation symbol, display means for extracting pronunciation symbols corresponding to an input alphabet from the pronunciation symbol information, and displaying the extracted pronunciation symbols while sorting them on the basis of the statistical information, and determination means for determining a pronunciation symbol corresponding to the English notation from the displayed pronunciation symbols.

Abstract: A processing unit and method are described herein that are capable of estimating a quality of a speech signal transmitted through a wireless network. The processing unit uses a logistic function to map a score output from an objective voice quality method (PESQ algorithm) into a mean of opinion (MOS) score which is an estimation of the quality of the speech signal that was transmitted through the wireless network. The logistic function has the form: y=1+4/(1+exp(?1.7244*x+5.0187)) where x is the score from the PESQ algoritm which is in the range of ?0.5 to 4.5 and y is the mapped MOS score which is in the range of 1 to 5 wherein if y=5 then the quality of the speech signal is considered excellent and if y=1 then the quality of the speech signal is considered bad.

Abstract: A voice signal interpolation apparatus is provided which can restore original human voices from human voices in a compressed state while maintaining a high sound quality. When a voice signal representative of a voice to be interpolated is acquired by a voice data input unit 1, a pitch deriving unit 2 filters this voice signal to identify a pitch length from the filtering result. A pitch length fixing unit 3 makes the voice signal have a constant time length of a section corresponding to a unit pitch, and generates pitch waveform data. A sub-band dividing unit 4 converts the pitch waveform data into sub-band data representative of a spectrum. A plurality of sub-band data pieces are averaged by an averaging unit 5 and thereafter a sub-band synthesizing unit 6 converts the sub-band data pieces into a signal representative of a waveform of the voice by a sub-band synthesizing unit 6.

Abstract: In an audio coding system, an encoding transmitter represents encoded spectral components as normalized floating-point numbers. The transmitter provides first and second control parameters that may be used to transcode the encoded spectral parameters. A transcoder uses first control parameters to partially decode the encoded components and uses second control parameters to re-encode the components. The transmitter determines the second control parameters by analyzing the effects of arithmetic operations in the partial-decoding process to identify situations where the floating-point representations lose normalization. Exponents associated with the numbers that lose normalization are modified and the modified exponents are used to calculate the second control parameters.