Showing papers on "Code-excited linear prediction published in 2004"

Gain quantization for a CELP speech coder

Abstract: A speech encoder that analyzes and classifies each frame of speech as being periodic-like speech or non-periodic like speech where the speech encoder performs a different gain quantization process depending if the speech is periodic or not. If the speech is periodic, the improved speech encoder obtains the pitch gains from the unquantized weighted speech signal and performs a pre-vector quantization of the adaptive codebook gain GP for each subframe of the frame before subframe processing begins and a closed-loop delayed decision vector quantization of the fixed codebook gain GC. If the frame of speech is non-periodic, the speech encoder may use any known method of gain quantization. The result of quantizing gains of periodic speech in this manner results in a reduction of the number of bits required to represent the quantized gain information and for periodic speech, the ability to use the quantized pitch gain for the current subframe to search the fixed codebook for the fixed codebook excitation vector for the current subframe. Alternatively, the new gain quantization process which was used only for periodic signals may be extended to non-periodic signals as well. This second strategy results in a slightly higher bit rate than that for periodic signals that use the new gain quantization strategy, but is still lower than the prior art's bit rate. Yet another alternative is to use the new gain quantization process for all speech signals without distinguishing between periodic and non-periodic signals.

Method and apparatus for improved quality voice transcoding

Abstract: A method and apparatus for a voice transcoder that converts 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 using perceptual weighting that uses tuned weighting factors, such that the bitstream of a second voice compression standard to produce a higher quality decoded voice signal than a comparable tandem transcoding solution. The method includes pre-computing weighting factors for a perceptual weighting filter optimized to a specific source and destination codec pair, pre-configuring the transcoding strategies, mapping CELP parameters in the CELP parameter space according to the selected coding strategy, performing Linear Prediction analysis if specified by the transcoding strategy, perceptually weighting the speech using with tuned weighting factors, and searching for adaptive codebook and fixed-codebook parameters to obtain a quantized set of destination codec parameters.

KLT-based adaptive classified VQ of the speech signal

Abstract: Compared to scalar quantization (SQ), vector quantization (VQ) has memory, space-filling, and shape advantages. If the signal statistics are known, direct vector quantization (DVQ) according to these statistics provides the highest coding efficiency, but requires unmanageable storage requirements if the statistics are time varying. In code-excited linear predictive (CELP) coding, a single "compromise" codebook is trained in the excitation-domain and the space-filling and shape advantages of VQ are utilized in a nonoptimal, average sense. In this paper, we propose Karhunen-Loe/spl grave/ve transform (KLT)-based adaptive classified VQ (CVQ), where the space-filling advantage can be utilized since the Voronoi-region shape is not affected by the KLT. The memory and shape advantages can be also used, since each codebook is designed based on a narrow class of KLT-domain statistics. We further improve basic KLT-CVQ with companding. The companding utilizes the shape advantage of VQ more efficiently. Our experiments show that KLT-CVQ provides a higher SNR than basic CELP coding, and has a computational complexity similar to DVQ and much lower than CELP. With companding, even single-class KLT-CVQ outperforms CELP, both in terms of SNR and codebook search complexity.

Bi-directional predicting method for video coding/decoding

Abstract: The invention discloses a bi-directional prediction method for video coding/decoding. When bi-directional prediction coding at the coding end, firstly the given forward candidate motion vector of the current image block is obtained for every image block of the current B-frame; the backward candidate motion vector is obtained through calculation, and the candidate bi-directional prediction reference block is obtained through bi-directional prediction method; the match is computed within the given searching scope and/or the given matching threshold; finally the optimal matching block is selected to determine the final forward motion vector, and the backward motion vector and the block residual. The present invention achieves the object of bi-directional prediction by coding a single motion vector, furthermore, it will not enhance the complexity of searching for a matching block at the coding end, and may save amount of coding the motion vector and represent the motion of the objects in video more actually. The present invention realizes a new prediction coding type by combining the forward prediction coding with the backward.

Lossless video coding using variable block-size MC and 3D prediction optimized for each frame

Abstract: This paper proposes an efficient lossless coding scheme for video signals. The coding scheme utilizes a novel block-adaptive 3D prediction method which predicts a video signal at each pel based on both the current frame and the motion-compensated previous frame. The resulting prediction errors are encoded using a kind of context-adaptive arithmetic coding. In order to improve coding efficiency, not only prediction coefficients of the 3D linear prediction but also motion vectors in all the blocks are iteratively optimized for each frame so that a coding rate of prediction errors can have a minimum. Moreover, a variable block-size motion compensation technique is employed for efficient representation of motion information. Experimental results show that coding rates of the proposed scheme are 18–55% lower than those of the JPEG-LS based intra-frame coding scheme.

Arranging CELP information of one frame in a second packet

Abstract: Packets of real-time information are sent with a source rate greater than zero kilobits per second, and a time or path or combined time/path diversity rate initially being zero kilobits per second. This results in a quality of service QoS, optionally measured at the sender or the receiver. When the QoS is on an unacceptable side of a threshold of acceptability, the sender sends diversity packets at an increased rate. Increasing the diversity rate while either reducing or maintaining the overall transmission rate is new. CELP-based multiple-description data partitioning sends the base or important information plus a subset of fixed excitation in one packet and sends the base or important information plus the complementary subset of fixed excitation in another packet. Reconstruction produces acceptable quality when only one of the two packets is received and better quality when both packets are received. Reconstruction provides for single and multiple lost packets.

Voice activity detection algorithm using radial basis function network

Abstract: A voice activity detection (VAD) algorithm using the radial basis function (RBF) network is proposed. The k-means clustering and least mean square algorithms are used to update the RBF network to the underlying speech condition. The inputs for RBF are code excited linear prediction coder parameters, which are stable under background noise. The RBF network output is compared to a threshold to determine the nature of the period (voice or silence). Experimental results show that the proposed VAD algorithm achieves better performance than G.729 Annex B at any noise level.

Speech coding using trajectory compression and multiple sensors.

Abstract: This paper presents a new method of multi-frame speech coding based upon polynomial approximation of speech feature trajectories incorporating multiple sensor signals from microphones, accelerometer, electro-glottograph, and microradar. The trajectory polynomial approximation exploits the inter-frame information redundancy encountered in natural speech. The trajectory method is applicable to features such as spectral parameters, gain, and pitch. The method is suitable for application to a frame vocoder to further reduce the transmission bit rate. Multiple transducers increase the intelligibility and quality of the coded speech in noisy environments. Experimental results are obtained by embedding the new method into an enhanced mixed-excitation linear prediction vocoder. The resulting vocoder operates at 1533 bps and preliminary intelligibility and quality tests show results comparable to those of the original 2400 bps vocoder.

Voicing index controls for celp speech coding

Abstract: An approach for improving quality of speech synthesized using analysis-by-synthesis (ABS ) coders is presented. An unstable perceptual quality in analysis-by-synthesis type speech coding (e.g. CELP) may occur because the periodicity degree in a voiced speech signal may vary significantly for different segments of the voiced speech. Thus the present invention uses a voicing index, which may indicate the periodicity degree of the speech signal, to control and improve ABS type speech coding. The voicing index may be used to improve the quality stability by controlling encoder and/or decoder in: fixed-codebook (301) short-term enhancement including the spectrum tilt; perceptual weighting filter; sub-fixed codebook determination; LPC interpolation (304); fixed-codebook pitch enhancement; post-pitch enhancement; noise injection into the high-frequency band at decoder; LTP sync window; signal decomposition, etc.

Dereverberation of speech signals based on linear prediction.

Abstract: This paper proposes an algorithm for the blind dereverberation of speech signals based on a two-channel linear prediction. Traditional dereverberation methods usually achieve good performance when the input signal is white noise. However, when dealing with colored signals generated by an autoregressive (AR) process such as speech, the generating AR process is deconvolved causing excessive whitening of the signal. This paper proposes a blind dereverberation algorithm that recovers speech signals suffering from deterioration due to the reverberation in a room. We overcome the whitening problem faced by traditional methods by estimating the generating AR process and applying this estimated AR process to the whitened signal. Simulation results show the great potential of the proposed method.

Signal decomposition of voiced speech for CELP speech coding

Abstract: An approach for improving quality of synthesized speech is presented. The input speech or residual is first separated into a voiced portion and a noise portion. The voice portion is coded using CELP methods. The noise portion of the input speech may be estimated at the decoder since it contains minimal voiced speech components. The separation is frequency dependent and is adaptive to the input speech. The separation may be accomplished using a lowpass/highpass filter combination. The information regarding bandwidth of the lowpass/highpass is presented to the decoder to facilitate reproduction of the noise portion of the speech.

Backward Linear Prediction for Lossless Coding of Stereo Audio

Abstract: Lossless audio coding aims at achieving the lowest possible bitrate for transmission or storage of audio without any loss of information. This is usually done by first removing redundancy from the audio signal, and then applying entropy coding to the residual signal. Linear prediction (LP), when applied to monophonic signals, is a very effective way to remove redundancy. It produces minimum-phase predictors that are efficiently compressed by combining vector quantization with a meaningful representation of the LP coefficients (such as the LSFs). When applied to stereo signals however, joint channel prediction often produces non-minimum-phase predictors, whose quantization requires a high bit rate and poses stability problems. In this paper, we show that backward estimation of the LP coefficients (where those are estimated on the past decoded signal) solves most of the problems associated with the use of joint channel prediction in a lossless audio coder.

Fine granularity scalability for MP-CELP based speech coding with HPDR technique

Abstract: In this work, based on the MP-CELP speech coding with HPDR technique, fine granularity scalability (FGS) is introduced by adjusting the amount of transmitted fixed excitation information. The FGS feature aim at changing the bit rate of the conventional coding more finely and more smoothly. Through performance analysis and computer simulation, the quality of scalability of the MP-CELP coding is presented with an improvement from conventional scalable MP-CELP. The HPDR technique is also applied to the MP-CELP to use for tonal language, meanwhile it can support the core coding rate of 4.2, 5.5, 7.5 kbps and additional scaled bit rates.

Optimizations of ITU G.729 speech codec

Abstract: Modern mobile communications require optimum bandwidth utilization with minimum loss, delay and good quality of speech transmission This triggered the usage of low bit rate voice codecs, among which CELP codecs inherit the merits of both waveform and source codecs, like toll quality, low bit rate etc In this paper several hardware and software optimization techniques are presented for efficient implementation of ITU G729 standard (CS-ACELP, conjugate structure algebraic code excited linear prediction) of 8 Kbit/s bit rate on a real time digital signal processor (DSP), with the aim of overcoming the limitation of computational burden and also scaling this application for enhanced speed to process more channels These techniques are in general applicable to any speech codec and DSP processor platform

Generalised candidate scheme for the stochastic codebook search of scalable CELP coders

Abstract: MPEG-4 CELP speech coders configured with multiple bit rates and coding layers provide SNR scalability, and hence may be operated effectively in variable bandwidth environments. A generalised candidate (GC) scheme is proposed to reduce the computational complexity of the stochastic codebook search of CELP coders. The experimental results demonstrate that the proposed GC scheme, incorporated with the verification model (VM) of the MPEG-4 CELP coder, enables a reduction of over 50% of the computational load without suffering any subjective quality degradations. The proposed GC scheme is not only suitable for the base layer and enhancement layers, but also facilitates computational scalability, and is therefore suitable for different working platforms and integrated media source services.

Device and method for speech decoding

Abstract: PROBLEM TO BE SOLVED: To improve the subjective quality of a decoded signal in a CELP (Code Excited Linear Prediction) type speech encoding method. SOLUTION: A pseudo-stationary noise generator 122 generates a pseudo-stationary noise signal. A gain adjuster 123 receives noise section decision information sent from an encoding side to calculate a gain coefficient with which the pseudo-stationary noise signal is multiplied. A multiplier 124 multiplies the pseudo-stationary noise by the gain determined by the gain adjuster 123 and outputs the result to an adder 125. The adder 125 adds the pseudo-stationary noise signal after gain adjustment to the output of a speech decoding device 101. A scaling part 126 uses the decoded speech signal after the pseudo-stationary noise signal is added and the decoded speech signal before the pseudo-stationary noise signal is added to perform scaling processing so that both signals become nearly equal in energy. A stationary noise feature extraction part 127 calculates a mean LSP parameter and signal energy in a stationary noise section. COPYRIGHT: (C)2005,JPO&NCIPI

Low-cost smart transcoding algorithm between ITU-T G.729 (8 kbit/s) and 3GPP NB-AMR (12.2 kbit/s)

Abstract: Networks interconnection causes interoperability problems between different speech coding formats. Today, tandem transcoding solutions (decoding/re-encoding) are generally used in communication chains. However intelligent transcoding solutions which exploit similarities between formats have been recently proposed in order to overcome tandem drawbacks (computational complexity, algorithmic delay, speech degradation). This paper gives an overview of these intelligent transcoding methods for CELP coders. Then, a low-cost intelligent transcoding algorithm between ITU-T G.729 (at 8 kbit/s) and 3GPP NB-AMR (at 12.2 kbit/s) is proposed. It is composed of four parts corresponding to the four CELP parameters conversions: LSP coefficients, fractional pitch lags, fixed codevectors and gains. A novel and computationally efficient method is described here for the fixed codevectors. The ACELP search is strongly focused on privileged positions, what considerably reduces the number of tested combinations. Objective and subjective quality tests show that this smart transcoding algorithm achieves a quality very close to that of the tandem while strongly reducing complexity with a shorter algorithmic delay.

Low-band extension of CELP speech coder by harmonics recovery

Abstract: Most of the telephone speech being transmitted in public networks is bandlimited to 300-3400 Hz. Narrowband speech is lacking in the information from low-band (0-300Hz) and high-band (3.4-8kHz) that are found in wideband speech (0-8kHz). As a result, narrowband speech is characterized by the reduced intelligibility and muffled quality, and degraded speaker identification. Therefore a bandwidth extension method, one of the techniques providing wideband speech quality without any additional information has been explored. By using the conventional technique to recover the missing high-band speech, we estimated the spectral envelope and the excitation for high-band using statistical recovery and spectral folding, respectively. In addition, we proposed an approach that exploits the harmonic synthesis method for the reconstruction of low-band speech over the CELP speech coding. Spectral distortion measurement and listening test are performed to evaluate the proposed method. Evaluation with synthesized speech shows the quality improvement.

Property vector based distortion estimation

Abstract: This paper addresses rate-distortion optimal mode selection, based on estimates of the instantaneous coding distortion, for multi-mode source coders. We propose a two stage estimation framework, including a property extraction, and a Gaussian mixture model based distortion estimation. We also suggest accompanying measures for evaluating the goodness of a given property vector. The proposed framework is evaluated for a set of properties in a CELP coding framework. The results suggest that the proposed framework is capable of doing rate distortion optimal mode selection in multi-mode source coders.

Speech encoding system and speech decoding system

Abstract: PROBLEM TO BE SOLVED: To encode a speech signal with high quality at a bit rate designating the speech signal and encoding delay. SOLUTION: An encoding parameter control circuit 31 calculates a frame length from the bit rate and encoding delay and outputs the frame length to a CELP encoding circuit 32. The encoding parameter control circuit selects a control parameter according to the bit rate from a table where a plurality of control parameters for controlling the operation of the CELP encoding circuit on the basis of the calculated frame length are described. Further, the encoding parameter control circuit outputs a sub-frame length and the number of bits allocated to a multipulse signal to a multipulse forming parameter setting circuit 33. The multipulse encoding parameter circuit calculates the number of pulses indicating a multipulse excitation signal, the number of pulse candidate positions of the respective pulses and the candidate positions from the sub-frame length and the number of bits of the multipulse signal. COPYRIGHT: (C)2004,JPO&NCIPI

CELP-based speaker verification: an evaluation under noisy conditions

Abstract: We propose a text-independent speaker verification method based on a speech coding scheme. The proposed method utilizes CELP parameters which are used in speech coding schemes for mobile communication systems, and verifies a speaker only with the encoded speech information. The reliability of the proposed method under noisy conditions is mainly discussed with some simulation results.

A low-bit-rate coding method and apparatus for unvoiced speed

Abstract: A low-bit-rate coding technique [502-530] for unvoiced segments of speech, without loss of quality compared to the conventional code Excited Linear Prediction (CELP) method operating at a much higher bit rate A set of gains are derived from a residual signal after whitening the speech signal by a linear prediction filter These gains are then quantized and applied to a randomly generated sparse excitation The excitation is filtered, and its spectral characteristics are analyzed and compared to the spectral characteristics of the original residual signal Based on this analysis, a filter is chosen to shape the spectral characteristics of the excitation to achieve optimal performance A low-bit-rate coding technique for unvoiced segments of speech A set of gains are derived from a residual signal after whitening the speech signal by a linear prediction filter These gains are then quantized and applied to a randomly generated sparse excitation The excitation is filtered, and its spectral characteristics are analyzed and compared to the spectral characteristics of the original residual signal Based on this analysis, a filter is chosen to shape the spectral characteristics of the excitation to achieve optimal performance

Receiver for encoding speech signal using a weighted synthesis filter

Abstract: A method for processing speech in a spread spectrum communication system uses CELP speech encoded signals. A speech input receives samples of a speech signal and a codebook analysis block for selects an index of a code from each of a plurality of codebooks. A weighted synthesis filter is used in the generation of a prediction error between a predicted current sample and a current sample of the speech samples. The index is transmitted to the receiver to enable reconstruction of the speech signal at the receiver.

Code excited linear prediction codec for electrocardiogram

Abstract: In This work we propose a CELP ECG codec for medical telemetry. The encoding algorithm is based on code-excited linear prediction (CELP). The general framework proposed is: QRS detection, calculation of LPC parameter, generation of residual error signal, codebook generation, MSE (mean square error) search. The codebook is generated for residual error. The indices of the codebook and corresponding LPC parameters are transmitted where the minimum MSE occurs. A replica of the transmitter codebook is present at the receiver. Corresponding to the received index value residual error coefficients are retrieved from the receiver codebook. The ECG signal is reconstructed from the retrieved code word.

Joint spatial-temporal-orientation-scale prediction and coding of motion vectors for rate-distortion-complexity optimized video coding

Abstract: Several prediction and coding schemes are combined to optimize performance in terms of the rate-distortion-complexity tradeoffs. Certain schemes for temporal prediction and coding of Motion Vectors (MVs) are combined with a new coding paradigm of over­ complete wavelet video coding. Two prediction and coding schemes are set forth herein. A first prediction and coding scheme employs prediction across spatial scales. A second prediction and coding scheme employs a motion vector prediction and coding across different orientation sub-bands. A video coding scheme utilizes joint prediction and coding to optimize the rate, distortion and the complexity simultaneously.

Method for searching codebook in CELP Vocoder using algebraic codebook

Abstract: PURPOSE: A method of searching a codebook of a code excited linear prediction (CELP) vocoder using an algebraic codebook is provided to search the algebraic codebook with a small amount of calculations using a search tree restricting method CONSTITUTION: A method of searching an algebraic codebook of an algebraic CELP vocoder using a depth-first tree search method includes a step(100) of searching up to a specific level of trees in order to predict a tree where an optimum pulse is positioned, a step(200) of selecting a predetermined tree according to the search result and removing other trees, and a step(300) of searching only the selected tree to select an optimum algebraic code