Disclosed is a voice code conversation apparatus to which voice code obtained by a first voice encoding method is input for converting this voice code to voice code of a second voice encoding method. The apparatus includes a code separating unit for separating, from the voice code based upon the first voice encoding method, codes of a plurality of components necessary to reconstruct a voice signal, code converters for dequantizing the codes of each of the components and then quantizing the dequantized values by the second voice encoding method to thereby generate codes, and a code multiplexer for multiplexing the codes output from respective ones of the code converters and transmitting voice code based upon the second voice encoding method.

Voice code conversion apparatus

A method and apparatus for CELP-based to CELP-based vocoder packet translation. The apparatus includes a formant parameter translator and an excitation parameter translator. The formant parameter translator includes a model order converter and a time base converter. The method includes the steps of translating the formant filter coefficients of the input packet from the input CELP format to the output CELP format and translating the pitch and codebook parameters of the input speech packet from the input CELP format to the output CELP format. The step of translating the formant filter coefficients includes the steps of converting the model order of the formant filter coefficients from the model order of the input CELP format to the model order of the output CELP format and converting the time base of the resulting coefficients from the input CELP format time base to the output CELP format time base.

CELP-based to CELP-based vocoder packet translation

In the method and device for interoperating a first station using a first communication scheme and comprising a first coder and a first decoder with a second station using a second communication scheme and comprising a second coder and a second decoder, communication between the first and second stations is conducted by transmitting signal-coding parameters related to a sound signal from the coder of one of the first and second stations to the decoder of the other station. The sound signal is classified to determine whether the signal-coding parameters should be transmitted from the coder of one station to the decoder of the other station using a first communication mode in which full bit rate is used for transmission of the signal-coding parameters. When classification of the sound signal determines that the signal-coding parameters should be transmitted using the first communication mode and when a request to transmit the signal-coding parameters from the coder of one station to the decoder of the other station using a second communication mode designed to reduce bit rate during transmission of the signal-coding parameters is received, a portion of the signal-coding parameters from the coder one station is dropped and the remaining signal-coding parameters are transmitting to the decoder of the other station using the second communication mode. The dropped portion of the signal-coding parameters are regenerated before the decoder of the other station decodes the signal-coding parameters.

Method and device for efficient in-band dim-and-burst signaling and half-rate max operation in variable bit-rate wideband speech coding for cdma wireless systems

The invention relates to a method for improving the coding accuracy and transmission efficiency of an audio signal. According to the method, a part of the audio signal to be coded is compared with earlier stored samples of the audio signal and a reference sequence of samples that best corresponds to the audio signal to be coded is identified. Predicted signals are produced from the reference sequence by means of long-term prediction, using at least two different LTP orders (M), a group of pitch predictor coefficients (b(k)) being formed for each pitch predictor order. The predicted signals for each pitch predictor order are compared with the audio signal to be coded in order to determine a prediction error. The amount of information required to code the predicted signals is compared with the amount of information required to code the original signal and a coding method that provides the best representation of the audio signal while minimising the amount of data required is selected.

Method for improving the coding efficiency of an audio signal

An apparatus for processing adaptive codebook pitch lag from one CELP based standard to another CELP based standard. The apparatus has various modules that perform at least the functionality described herein. The apparatus includes a time-base subframe checker inspection module, which is adapted to associate one or more incoming subframes with an outgoing subframes of a destination codec. The apparatus also has a decision module coupled to the time-base subframe inspection module. The decision module is adapted to determine a desired pitch lag parameter from a plurality of pitch lag parameters among respective two or more incoming subframes. The apparatus has a pitch lag selection module coupled to the decision module. The pitch lag selection module is adapted to select the desired pitch lag parameter.

Method for adaptive codebook pitch-lag computation in audio transcoders

PURPOSE: To provide the code conversion system which allows a telephone conversation without reconverting a speech into a reproduced speech temporarily as to a telephone conversation between speech encoding systems that are different in quantized value or quantizing method. CONSTITUTION: When there are a 1st speech encoding method and a 2nd speech encoding method which perform encoding processes differing in quantized value or quantizing method as code excitation linear predictive encoding methods, the code converting device converts multiplexed codes of the 1st speech encoding method into multiplexed codes of the 2nd speech encoding method. A code separation part 510 inputs the multiplexed codes encoded by the 1st speech encoding method and separates them into individual codes, and a conversion part 500 converts the individual separated codes into respective codes of the 2nd speech encoding method according to the correspondence relation between the codes of the 1st speech encoding method and the codes of the 2nd speech encoding method. A multiplexing part 570 multiplexes the respective codes of the 2nd speech encoding method which are converted by the conversion part 500.

Device and system for code conversion

PURPOSE: To provide a speech CODEC which allows decoding corresponding to each coding method in a network where plural kinds of coding methods (full rate/half rate) are intermingled and to share the processing. CONSTITUTION: A speech CODEC 27 of full rate 20 and half rate 21 is provided to switch the coding method depending on whether the coding method in the communication device of the party uses the full rate 20 or the half rate 21. Each processing module of linear prediction factor calculation, leg calculation, filtering, filtering renewal, and coding processing of transmission line is shared at the full rate and the half rate in the coders. Each processing module of deinterleave, punctured decoding, Viterbi decoding and filtering is also shared at the full rate and the half rate in the decoders.

Speech coder and decoder provided with plural kinds of coding methods

PURPOSE: To previously display warning indicating the anxiety of communication disablement when error generation frequency is high in digital mobile radio data communication. CONSTITUTION: In the case of executing error detection in each frame from a error-corrected sound encoding parameter by a detector 22, error generation frequency in a prescribed frame period is monitored by a monitor 23, and when the error generation frequency exceeds a previously set threshold, anxiety about the generation of communication disablement is displayed on a warning device 24 as warning even when the communication disablement state is not generated yet, so that a communicator can prepare a proper counterplan. COPYRIGHT: (C)1995,JPO

Digital mobile radio data communication method and its device

PURPOSE:To operate the telephone communication terminal equipment executing a call in voice. CONSTITUTION:A telephone terminal equipment is made up of an antenna 201, a branching filter 202, a radio section 203, a base band section 204, a control section 205, a speaker 206, a microphone 207, a key entry display section 208, a sound volume measurement equipment 209, threshold level generators 210, 213, comparators 211, 214, an analyzer 212, and an opposite party number storage device 215. The sound measurement device 209 always measures the sound volume of the input signal of a voice received from the microphone 207. The acoustic measurement device 209 conducts processing such as calculation of signal power for each prescribed period. Then an output of the sound volume measurement equipment 209 is compared with an output of the threshold level generator 210 outputting a preset threshold level at the comparator 211 and when the output of the sound volume measurement 209 is higher, the comparison result is given to the control section 205. Upon the receipt of both the outputs, the control section 205 discriminates the arrival of the input signal satisfying the condition set by the threshold level and reads a telephone number registered in advance in the opposite party number storage device 215 and sends it.

隆二郎村松

Papers

Device and system for code conversion

Speech coder and decoder provided with plural kinds of coding methods

Digital mobile radio data communication method and its device

Telephone communication terminal equipment

隆二郎 村松

Papers

Device and system for code conversion

Speech coder and decoder provided with plural kinds of coding methods

Digital mobile radio data communication method and its device

Telephone communication terminal equipment

隆二郎村松