72 results about "Code-excited linear prediction" patented technology

A multi-rate speech codec supports a plurality of encoding bit rate modes by adaptively selecting encoding bit rate modes to match communication channel restrictions. In higher bit rate encoding modes, an accurate representation of speech through CELP (code excited linear prediction) and other associated modeling parameters are generated for higher quality decoding and reproduction. A speech encoder employing various encoding schemes based upon parameters including an available transmission bit rate. In addition, the speech encoder is operable to identify and apply an optimal encoding scheme for a given speech signal. The speech encoder may be applied code-excited linear prediction when the available bit rate is above a predetermined upper threshold. Pitch preprocessing, including continuous warping, may be applied when it is below a predetermined lower threshold. The encoder considers varying characteristics of the speech signal including the long term prediction mode of a previous frame, and a spectral difference between the line spectral frequencies of a current and a previous frame, a predicted pitch lag, an open loop pitch lag, a closed loop pitch lag, a pitch gain, and a pitch correlation.

A speech encoding method and apparatus in which an input speech signal is divided in terms of blocks or frames as encoding units and encoded in terms of the encoding units, whereby explosive and fricative consonants can be impeccably reproduced, while there is an attenuation of the occurrence of foreign sounds being generated at a transient portion between voiced (V) and unvoiced (UV) portions, so that the speech with high clarity devoid of “stuffed” feeling may be produced. The encoding apparatus includes a first encoding unit for finding residuals of linear predictive coding (LPC) of an input speech signal for performing harmonic coding and a second encoding unit for encoding the input speech signal by waveform coding. The first encoding unit and the second encoding unit are used for encoding a voiced (V) portion and an unvoiced (UV) portion of the input signal, respectively. Code excited linear prediction (CELP) encoding employing vector quantization by a closed loop search of an optimum vector using an analysis-by-synthesis method is used for the second encoding unit. A corresponding decoding method and apparatus is also provided.

The present invention provides a method for compensating transient effects in transform coding and decoding of a combined speech and audio in electronic devices by using a transform based time-frequency domain codec. The method can combine, e.g., a CELP (code excited linear prediction) type speech codec and a transform type audio codec. The invention describes a compensation method to handle the transient (e.g., from the CELP coding to the transform coding) in transform coding when the number of quantized transform coding coefficients is lower than in the output of the transform.

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.

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.

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 and apparatus for DTMF detection and voice mixing in the code-excited linear prediction (CELP) parameter space, without fully decoding and reconstructing the speech signal. The apparatus includes a Dual Tone Multiplexed Frequency (DTMF) signal detection module and a multi-input mixing module. The DTMF signal detection module detects DTMF signals by computing characteristic features from the input CELP parameters and comparing them with known features of DTMF signals. The multi-input mixing module mixes multiple sets of input CELP parameters, that represent multiple voice signals, into a single set of CELP parameters. The mixing computation is performed by analyzing each set of input CELP parameters, determining the order of importance of the input sets, selecting a strategy for mixing the CELP parameters, and outputting the mixed CELP parameters. The method includes inputting one or more sets of CELP parameters and external commands, detecting DTMF tones, mixing multiple sets of CELP parameters and outputting the DTMF signal, if detected, and the mixed CELP parameters.

Codebook indices for a scalable speech and audio codec may be efficiently encoded based on anticipated probability distributions for such codebook indices. A residual signal from a Code Excited Linear Prediction (CELP)-based encoding layer may be obtained, where the residual signal is a difference between an original audio signal and a reconstructed version of the original audio signal. The residual signal may be transformed at a Discrete Cosine Transform (DCT)-type transform layer to obtain a corresponding transform spectrum. The transform spectrum is divided into a plurality of spectral bands, where each spectral band having a plurality of spectral lines. A plurality of different codebooks are then selected for encoding the spectral bands, where each codebook is associated with a codebook index. A plurality of codebook indices associated with the selected codebooks are then encoded together to obtain a descriptor code that more compactly represents the codebook indices.

Provided is a speech coding apparatus and method. A band divider divides an input signal into a high-band signal and a low-band signal, a narrowband encoder encodes the low-band signal using a Code Excited Linear Prediction (CELP)-based narrowband speech codec, a frequency characteristic collector converts the high-band signal to a signal in a frequency domain and obtains Modified Discrete Cosine Transform (MDCT) coefficients, a subband determiner determines subbands in a final stage based on the MDCT coefficients and determines subbands for quantization based on the subbands in a final stage, a gain quantizer performs gain quantization of the subbands, a bit assignment unit assigns bits to the subbands according to the magnitude of the gain quantization, and a shape quantizer performs shape quantization of the subbands in an algebraic method. Accordingly, algorithm consistency can be maintained and a complexity can be reduced by extending a bandwidth with a small number of bits in a speech codec.

A decoding apparatus is provided. The decoding apparatus has a first decoding part for decoding a code word obtained by encoding an input signal using a Code-Excited Linear Prediction encoding method. A second decoding part decodes a code word obtained by encoding a signal with an encoding method other than the Code-Excited Linear Prediction encoding method. A rising-transition detection and notification part has a detection part that detects the existence of a rising-transition of amplitude of the input signal based on time variation of a gain of excitation vectors obtained by the first decoding part, and a notification part that notifies the second decoding part that the rising-transition of the amplitude exists.

Layered (embedded) code-excited linear prediction (CELP) speech encoders/decoders with adaptive plus algebraic codebooks applied in each layer with fixed codebook pulses of one layer used in higher layers. Pulse weightings emphasize lower layer pulses relative to the higher layer pulses.

Audio coding and decoding apparatuses and methods that can optimize the quality of an audio signal including harmonics, and recording mediums storing the methods. An audio coding apparatus includes: a first harmonic coding module performing first harmonic coding on an input audio signal using a pitch lag of the input audio signal and producing a quantized linear prediction coding coefficient; a first detector detecting a first difference audio signal from a difference between an audio signal output from the first harmonic coding module and the input audio signal; a second harmonic coding module performing harmonic coding on the first difference audio signal using the quantized linear prediction coding coefficient and a previous harmonic coding result; a second detector detecting a second difference audio signal obtained from a difference between an audio signal output from the second harmonic coding module and the first difference audio signal; and a code excited linear prediction (CELP) module CELP coding the second difference audio signal using the quantized linear prediction coding coefficient obtained from the first harmonic coding module.

Provided are a fixed codebook search method based on iteration-free global pulse replacement in a speech codec, and a Code-Excited Linear-Prediction (CELP)-based speech codec using the method. The fixed codebook search method based on iteration-free global pulse replacement in a speech codec includes the steps of: (a) determining an initial codevector using a pulse-position likelihood vector or a correlation vector; (b) calculating a fixed-codebook search criterion value for the initial codevector; (c) calculating fixed-codebook search criterion values for respective codevectors obtained by replacing a pulse of the initial codevector each time for respective tracks, and determining a pulse position generating the largest fixed-codebook search criterion value as a candidate pulse position for the respective tracks, respectively; (d) calculating fixed-codebook search criterion values for respective codevectors of all combinations obtained by replacing at least one pulse position of the initial codevector with the candidate pulse positions of the respective tracks, and determining the largest value of the fixed-codebook search criterion values; and (e) comparing the fixed-codebook search criterion value for the initial codevector obtained in step (b) with the largest value determined in step (d) to determine an optimum fixed codevector.

A scalable speech and audio codec is provided that implements combinatorial spectrum encoding. A residual signal is obtained from a Code Excited Linear Prediction (CELP)-based encoding layer, where the residual signal is a difference between an original audio signal and a reconstructed version of the original audio signal. The residual signal is transformed at a Discrete Cosine Transform (DCT)-type transform layer to obtain a corresponding transform spectrum having a plurality of spectral lines. The transform spectrum spectral lines are transformed using a combinatorial position coding technique. The combinatorial position coding technique includes generating a lexicographical index for a selected subset of spectral lines, where each lexicographic index represents one of a plurality of possible binary strings representing the positions of the selected subset of spectral lines. The lexicographical index represents non-zero spectral lines in a binary string in fewer bits than the lengthof the binary string.

A layered code-excited linear prediction (CELP) encoder, an Adaptive Multirate Wideband (AMR-WB) encoder and methods of CELP encoding and decoding. In one embodiment, the encoder includes: (1) a core layer subencoder and (2) at least one enhancement layer subencoder, at least one of the core layer subencoder and the enhancement layer subencoder having first and second adaptive codebooks and configured to retrieve a pitch lag estimate from the second adaptive codebook and perform a closed-loop search of the first adaptive codebook based on the pitch lag estimate.

Layered code-excited linear prediction (CELP) speech encoders have progressively weaker perceptual weighting filters for each of the successive enhancement layers and decoders have progressively weaker short-term postfilters for increased bit rates (increased number of enhancement layers decoded) and a long-term postfilter for all bit rates.

A code excited linear prediction speech decoder includes an adaptive codebook configured to generate an adaptive code vector. The decoder also includes a random codebook configured to generate a random code vector. The decoder also includes a a synthesis filter that receives a signal based on said adaptive code vector and said random code vector, and that is configured to perform linear prediction coefficient synthesis on said signal. The random codebook includes a pulse vector providing system configured to provide a pulse vector having a signed unit pulse. The random codebook also includes a comparing system configured to compare a value of adaptive codebook gain with a preset threshold value. The random codebook further includes a selecting system configured to select a dispersion pattern from a plurality of dispersion patterns stored in a memory in accordance with a result of said comparison. The random codebook additionally includes a generating system configured to generate said dispersed vector by convoluting said pulse vector and said selected dispersion pattern.