412 results about "Speech code" patented technology

A method of filtering a speech signal is presented. The method involves providing a filter (404) suited for reduction of distortion caused by speech coding, estimating acoustic noise in the speech signal, adapting the filter in response to the estimated acoustic noise to obtain an adapted filter, and applying the adapted filter to the speech signal so as to reduce acoustic noise and distortion caused by speech coding in the speech signal.

There are provided speech coding methods and systems for estimating a plurality of speech parameters of a speech signal for coding the speech signal using one of a plurality of speech coding algorithms, the plurality of speech parameters includes pitch information, the plurality of speech parameters is calculated using a plurality of thresholds. An example method includes estimating a background noise level in the speech signal to determine a signal to noise ratio (SNR) for the speech signal, adjusting one or more of the plurality of thresholds based on the SNR to generate one or more SNR adjusted thresholds, analyzing the speech signal to extract the pitch information using the one or more SNR adjusted thresholds, and repeating the estimating, the adjusting and the analyzing to code the speech signal using one the plurality of speech coding algorithms.

An echo cancellation device (ECD) comprises an echo canceller (EC) including a transfer function estimator (EST, H) and a subtractor (ADD) and a residual echo suppression device (G, ADD2). The residual echo suppression device (G) comprises a residual echo filter (G) having an adjustable filter function (g). This filter function (g) can be adapted to either remove from the subtractor output (TNE') the spectral characteristics relating to the reception signal (RFE) and / or to emphasize in the subtractor output signal (TNE') a background signal spectral content relating to the transmission signal (TNE). A noise generation means (NGM') can be provided at the output of the adaptable filter (G) for injecting a noise process in to the filter output signal (TNE') prior to a speech coding in a speech coder (COD). The noise process masks in the filter output signal a spectral content relating to the reception signal (RFE).

A system, method and access gateway for minimizing the number of transcodings of a speech signal during a Voice-over-IP (VoIP) conference call in a packet-switched network in which Tandem Free Operation (TFO) is utilized. The system includes a first gateway connecting the first mobile subscriber to the network, a second gateway connecting the second subscriber, and a third gateway connecting the third subscriber. The second gateway sends a message to the first gateway indicating a speech coding mode being utilized between the second gateway and the second subscriber. The third gateway sends a message to the first gateway indicating a speech coding mode being utilized between the third gateway and the third subscriber. When a three-way conference call is initiated, the first gateway encodes the call path to the second subscriber with the speech coding mode being utilized between the second gateway and the second subscriber. The first gateway also encodes the speech signal for the call leg to the third subscriber with the speech coding mode being utilized between the third gateway and the third subscriber.

A variable rate compressed voice signal domain transcoder that transcodes a bitstream representing frames of data encoded according to a first voice compression standard to a bitstream representing frames of data according to a second voice compression standard; the second voice compression standard defines a variable-rate voice codec. The method includes unquantizing a bitstream into a first set of parameters compatible with a first compression standard. The first set of parameters in addition to external control commands are then used to determine the frame class and rate for the second compression standard. Next, the first set of parameters are transformed into a second set of parameters compatible with a second compression standard according to the frame-classification and rate determination decision without converting the first set of parameters to an analog or digital voice waveform representation. The transformation approaches can be varied and further optimized based on the characteristics of the pair of first compression standard and the second compression standard. Lastly, the second set of parameters is packed into a bitstream compatible with the second compression standard.

A method and system are provided for synthesizing a number of corrupted frames output from a decoder including one or more predictive filters. The corrupted frames are representative of one segment of a decoded signal (sq(n)) output from the decoder. The method comprises determining a first preliminary time lag (ppfe1) based upon examining a predetermined number (K) of samples of another segment of the decoded signal and determining a scaling factor (ptfe) associated with the examined number (K) of samples when the first preliminary time lag (ppfe1) is determined. The method also comprises extrapolating one or more replacement frames based upon the first preliminary time lag (ppfe1) and the scaling factor (ptfe).

A high-band speech encoding apparatus and a high-band speech decoding apparatus that can reproduce high quality sound even at a low bitrate when wideband speech encoding and decoding using a bandwidth extension function, and a high-band speech encoding and decoding method performed by the apparatuses. The high-band speech encoding apparatus includes: a first encoding unit encoding a high-band speech signal based on a structure in which a harmonic structure and a stochastic structure are combined, if the high-band speech signal has a harmonic component; and a second encoding unit encoding a high-band speech signal based on a stochastic structure if the high-band speech signal has no harmonic components. The high-band speech decoding apparatus includes: a first decoding unit decoding a high-band speech signal based on a combination of a harmonic structure and a stochastic structure using received first decoding information; a second decoding unit decoding the high-band speech signal based on a stochastic structure using received second decoding information; and a switch outputting one of the decoded high-band speech signals received from the first and second decoding units according to received mode selection information.

A high-band speech encoding apparatus and a high-band speech decoding apparatus that can reproduce high quality sound even at a low bitrate when wideband speech encoding and decoding using a bandwidth extension function, and a high-band speech encoding and decoding method performed by the apparatuses. The high-band speech encoding apparatus includes: a first encoding unit encoding a high-band speech signal based on a structure in which a harmonic structure and a stochastic structure are combined, if the high-band speech signal has a harmonic component; and a second encoding unit encoding a high-band speech signal based on a stochastic structure if the high-band speech signal has no harmonic components. The high-band speech decoding apparatus includes: a first decoding unit decoding a high-band speech signal based on a combination of a harmonic structure and a stochastic structure using received first decoding information; a second decoding unit decoding the high-band speech signal based on a stochastic structure using received second decoding information; and a switch outputting one of the decoded high-band speech signals received from the first and second decoding units according to received mode selection information.

A speech coding system (10) and associated method relies on a speech encoder (15) and a speech decoder (20). The speech decoder (20) includes a harmonic generator (70) which modulates the phase of each generated harmonic with a low frequency, low bandwidth signal to remove the buzzy quality of the speech and to produce natural sounding speech. The amplitude of the phase modulating signal is adjusted in accordance with the harmonic magnitude. For harmonics residing in a spectral valley the amplitude of the modulating signal is relatively large and for harmonics residing near spectral peaks, the amplitude of the modulation signal is relatively small.

Linear predictive speech coding system with classification of frames and a hybrid coder using both waveform coding and parametric coding for different classes of frames. Phase alignment for a parametric coder aligns synthesized speech frames with adjacent waveform coder synthesized frames. Zero phase alignment of speech prior to waveform coding aligns synthesized speech frames of a waveform coder with frames synthesized with a parametric coder. Inter-frame interpolation of LP coefficients suppresses artifacts in resultant synthesized speech frames.

There is provided a speech post-processor for enhancing a speech signal divided into a plurality of sub-bands in frequency domain. The speech post-processor comprises an envelope modification factor generator configured to use frequency domain coefficients representative of an envelope derived from the plurality of sub-bands to generate an envelope modification factor for the envelope derived from the plurality of sub-bands, where the envelope modification factor is generated using FAC=αENV / Max+(1−α), where FAC is the envelope modification factor, ENV is the envelope, Max is the maximum envelope, and α is a value between 0 and 1, where α is a different constant value for each speech coding rate. The speech post-processor further comprises an envelope modifier configured to modify the envelope derived from the plurality of sub-bands by the envelope modification factor corresponding to each of the plurality of sub-bands.

A method and system are provided for synthesizing a number of corrupted frames output from a decoder including one or more predictive filters. The corrupted frames are representative of one segment of a decoded signal (sq(n)) output from the decoder. The method comprises determining a first preliminary time lag (ppfe1) based upon examining a predetermined number (K) of samples of another segment of the decoded signal and determining a scaling factor (ptfe) associated with the examined number (K) of samples when the first preliminary time lag (ppfe1) is determined. The method also comprises extrapolating one or more replacement frames based upon the first preliminary time lag (ppfe1) and the scaling factor (ptfe).

A block-constrained Trellis coded quantization (TCQ) method and a method and apparatus for quantizing line spectral frequency (LSF) parameters employing the same in a speech coding system wherein the LSF coefficient quantizing method includes: removing the direct current (DC) component in an input LSF coefficient vector; generating a first prediction error vector by performing inter-frame and intra-frame prediction for the LSF coefficient vector, in which the DC component is removed, quantizing the first prediction error vector by using the BC-TCQ algorithm, and by performing intra-frame and inter-frame prediction compensation, generating a quantized first LSF coefficient vector; generating a second prediction error vector by performing intra-frame prediction for the LSF coefficient vector, in which the DC component is removed, quantizing the second prediction error vector by using the BC-TCQ algorithm, and then, by performing intra-frame prediction compensation, generating a quantized second LSF coefficient vector; and selectively outputting a vector having a shorter Euclidian distance to the input LSF coefficient vector between the generated quantized first and second LSF coefficient vectors.

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.

Linear predictive speech coding system with classification of frames and a hybrid coder using both waveform coding and parametric coding for different classes of frames. Phase alignment for a parametric coder aligns synthesized speech frames with adjacent waveform coder synthesized frames. Zero phase alignment of speech prior to waveform coding aligns synthesized speech frames of a waveform coder with frames synthesized with a parametric coder. Inter-frame interpolation of LP coefficients suppresses artifacts in resultant synthesized speech frames.

A speech coding system is provided with time-domain noise attenuation. The speech coding system has an encoder operatively connected to a decoder via a communication medium. A preprocessor processes a digitized speech signal from an analog-to-digital converter. Speech coding systems are used to encode and decode a bitstream. Gains from the speech coding are adjusted by a gain factor Gf that provides time-domain background noise attenuation.

A method of significantly reducing error propagation due to voice packet loss, while still greatly profiting from long-term pitch prediction, is achieved by adaptively limiting the maximum value of the pitch gain for the first pitch cycle within one frame. A speech coding system for encoding a speech signal, wherein said a plurality of speech frames are classified into said a plurality of classes depending on if the first pitch cycle is included in one subframe or several subframes. The pitch gain is set to a value significantly smaller than 1 for the subframes covering first pitch cycle; wherein the pitch gain reduction is compensated by increasing the coded excitation codebook size or adding one more stage of excitation for the subframes covering the first pitch cycle.

The invention improves the encoding and decoding of speech by focusing the encoding on the perceptually important characteristics of speech. The system analyzes selected features of an input speech signal, and first performing a common frame based speech coding of an input speech signal. The system then performs a speech coding based on either a first speech coding mode or a second speech coding mode. The selection of a mode is based on characteristics of the input speech signal. The first speech coding mode uses a first framing structure and the second speech coding mode uses a second framing structure.

Provided is a highband coding apparatus and method for a wideband coding system. The coding apparatus and method can reduce a pre-echo phenomenon by encoding the highband based on lowband encoding information and Temporal Noise Shaping technique. A highband encoding apparatus includes: a domain converter for converting the domain of an input highband signal into a frequency domain; a linear prediction order determiner for determining a linear prediction order based on the lowband encoding information; a linear prediction analyzer for analyzing a highband signal of the frequency domain based on the determined linear prediction order to thereby generate a linear prediction coefficient; a linear prediction coefficient quantizer for quantizing the linear prediction coefficient based on the lowband encoding information; and a residual signal quantizer for obtaining a residual signal by dequantizing the quantized linear prediction coefficient and quantizing the residual signal.

Provided are a scalable wide-band speech coding / decoding apparatus, method, and medium. An input wide-band speech input signal is first divided into a low-band signal and a high-band signal. The divided low-band signal is then coded using a code excited linear prediction (CELP) method. The divided high-band signal is coded using a harmonic method. A signal representing a difference between a synthetic signal obtained from the low-band and the high band, and a signal input to the low-band and the high-band is then coded using a modified discrete cosine transform (MDCT) method. The coded signal is then multiplexed. The multiplexed signal is then output. Accordingly, high quality speech can be achieved for all layers.

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.

The technical scheme relates to a device for calling a computer program to run by utilizing a Chinese text and belongs to the field of computer man-machine conversation automatic control technologies. The device for calling the computer program to run by utilizing the Chinese text comprises a Chinese text input module (1), a computer system (2), a Chinese text word segmentation module (3), a Chinese character and Chinese speech code conversion module (7), a computer program module (4) which is pre-stored in a computer and bound with input Chinese speech codes or keywords in Chinese characters, a program logic arranging and connecting module (5) which arranges bound programs according to a logic sequence of execution, and a program execution module (6) stored in the computer, and the above modules are sequentially connected in series according to an information processing sequence of the computer. By means of the device for calling the computer program to run by utilizing the Chinese text, a non-computer programmer can call the computer program to run by utilizing Chinese natural language.

A method for speech processing in a code excitation linear prediction (CELP) based speech system having a plurality of modes including at least a first mode and a consecutive second mode. The method includes providing an input speech signal, dividing the speech signal into a plurality of frames, dividing at least one of the plurality of frames into sub-frames including a plurality of pulses, selecting a first number of pulses for the first mode, with a second number of remaining pulses in the frame plus the first number of pulses in the first mode for the second mode, providing a plurality of sub-modes between the first mode and the second mode, forming a base layer, forming an enhancement layer, generating a bit stream including a basic bit stream and an enhancement bit stream, wherein the basic bit stream is used to update memory states of the speech system.

A coding apparatus including a base layer, a speech quality enhancement layer, and a multiplexer. The base layer filters an input speech signal using linear prediction coding and generates an excitation signal corresponding to the filtered speech signal through a fixed codebook search and an adaptive codebook search. The speech quality enhancement layer searches a fixed codebook using parameters obtained through the fixed codebook search in the base layer, or searches the fixed codebook using a target signal, which is obtained by removing a contribution of a fixed codebook of the base layer and a signal which is obtained by synthesizing and filtering a previous fixed codebook of the speech quality enhancement layer from a target signal for the fixed codebook search of the base layer. The multiplexer multiplexes signals generated by the base layer and the at least one speech quality enhancement layer.

In methods and apparatus for encoding a gain parameter in a generalized linear predictive analysis-by-synthesis (GLPAS) coder, a subframe gain parameter is determined for each of a plurality of successive subframes of a frame, and a quantized frame gain parameter is determined for each frame using a delayed decision quantizer operating on the subframe gain parameters. The subframe gain parameters may be treated as components of a gain vector and the gain vector may be vector quantized to determine the quantized frame gain parameter. Encoder parameters are efficiently aligned with decoder parameters to ensure proper end-to-end operation. Alternatively, tree quantization or trellis quantization may be applied to the subframe gain parameters to determine the quantized frame gain parameter. The methods and apparatus are particularly applicable to low bit rate speech coding.

The invention provides a Chinese speech remote control computer program operating device and belongs to the technical field of man-computer conversation automatic control. A remote control of the Chinese speech remote control computer program operating device comprises an input Chinese speech remote control device module 1, a Chinese speech recognition module 2, a network transmission module 3, a computer system 4, a computer program module 5 pre-stored in the computer system 4 and bound with input Chinese speech codes or keywords of Chinese characters, a module 6 for arranging and connecting bound programs according to an executed logic sequence and a program execution module 7 stored in the computer system 4, wherein the modules are sequentially connected in series according to a computer information processing sequence. After adopting the technical scheme, people can conveniently use Chinese speech to control specified computer program operation through various remote control devices in a non-field and non-real-time mode.

The invention relates to a speech information hiding secret communication system which organically integrates techniques, such as information hiding, passwords, communication, and the like. In the invention, MELP2.4KBps secret speech is hidden in the public speech of G.721, G.728, GSM and G.729 of a code-excited linear prediction code by connecting a public telephone network PSTN, mobile communication GSM / CDMA, videoconferences and a VoIP network and utilizing a speech information hiding and extraction algorithm, and a private channel is established in a public communication channel to carry out the real-time secret communication of secret speech information. The system adopts an embedded type technique based on DSP, and designs corresponding functional modules, such as speech coding and decoding, speech encryption and decryption, speech information hiding and extraction, and the like. Under the condition that a certain communication rate and a certain speech quality are ensured, secret speech real-time communication is carried out on various public channels, and the communication quality reaches the requirement of a communication standard.

A method of using music detection to enhance an operation of an echo canceller is provided, wherein the echo canceller includes an adaptive filter and a nonlinear processor. The method comprises receiving an input signal including an echo signal by the echo canceller from a near end device, filtering the input signal using the adaptive filter to eliminate linear components of the echo signal in the input signal and generate an error signal, analyzing the error signal using a music detector to determine existence of a music signal in the error signal, bypassing the nonlinear processor if the analyzing determines the music signal exists in the error signal, and eliminating nonlinear components of the echo signal from the error signal using the nonlinear processor if the analyzing determines the music signal does not exist in the error signal.