Showing papers on "Code-excited linear prediction published in 2011"

Estimation of speech energy based on code excited linear prediction (CELP) parameters extracted from a partially- decoded CELP-encoded bit stream and applications of same

Abstract: Methods, systems, and non-transitory computer readable media for estimating speech energy of an encoded bit stream based on coding parameters extracted from the partially-decoded bit stream are disclosed. According to one aspect, a method for estimating speech energy of an encoded bit stream based on coding parameters extracted from the partially-decoded bit stream includes receiving a CELP-encoded bit stream, partially decoding the bit stream, and estimating the speech energy of the bit stream based a set of four or fewer CELP parameters extracted from the partially decoded bit stream. According to another aspect, a method for estimating speech energy of an encoded bit stream based on coding parameters extracted from the partially-decoded bit stream includes receiving a CELP-encoded bit stream, partially decoding the bit stream, extracting at least one CELP parameter from the partially-decoded bit stream, and estimating the speech energy of the bit stream based on the extracted at least one CELP parameter without calculating a linear prediction coding (LPC) filter response energy.

MFCC and CELP to detect turbine engine faults

Abstract: A fault detection and diagnosis method for a gas turbine engines comprises collecting a sensor signal from an acoustic or vibrational sensor at the gas turbine engine, pre-processing the sensor signal to remove predictable background, and extracting a feature set from the sensor signal using Mel-Frequency Cepstral Coefficients (MFCC) algorithms and/or Code Excited Linear Prediction (CELP) algorithms. Fault and non-fault states are reported based on comparison between the feature set and a library of fault and non-fault feature profiles corresponding to fault and non-fault states of the gas turbine engine.

Audio encoding device and audio decoding device

Abstract: Provided is an audio encoding device that can suppress degradation of audio quality. Spectral coefficients of synthesized signal from CELP core layer are utilized to fulfill spectral gaps in error signal spectrum coefficients from a transform coding layer. By both spectral coefficients, decoded signal spectral coefficients are generated. The decoded signal spectral coefficients and the input signal spectral coefficients are divided into a plurality of sub bands. In each sub band, the energy of the input signal spectral coefficient corresponding to a zero decoded error signal spectral coefficient is calculated, and the energy of the decoded signal spectral coefficient corresponding to the zero decoding error signal spectral coefficient is calculated, and their energy ratio is calculated and is quantized and transmitted.

Burst packet loss concealment using multiple codebooks and comfort noise for CELP-type speech coders in wireless sensor networks.

Abstract: In this paper, a packet loss concealment (PLC) algorithm for CELP-type speech coders is proposed in order to improve the quality of decoded speech under burst packet loss conditions in a wireless sensor network. Conventional receiver-based PLC algorithms in the G.729 speech codec are usually based on speech correlation to reconstruct the decoded speech of lost frames by using parameter information obtained from the previous correctly received frames. However, this approach has difficulty in reconstructing voice onset signals since the parameters such as pitch, linear predictive coding coefficient, and adaptive/fixed codebooks of the previous frames are mostly related to silence frames. Thus, in order to reconstruct speech signals in the voice onset intervals, we propose a multiple codebook-based approach that includes a traditional adaptive codebook and a new random codebook composed of comfort noise. The proposed PLC algorithm is designed as a PLC algorithm for G.729 and its performance is then compared with that of the PLC algorithm currently employed in G.729 via a perceptual evaluation of speech quality, a waveform comparison, and a preference test under different random and burst packet loss conditions. It is shown from the experiments that the proposed PLC algorithm provides significantly better speech quality than the PLC algorithm employed in G.729 under all the test conditions.

Automatic Speech Codec Identification with Applications to Tampering Detection of Speech Recordings

Abstract: In this paper we explored many versions of CELP codecs and studied different codebooks they use to encode noisy part of residual. Taking advantage of noise patterns they generated, an algorithm was proposed to detect GSM-AMR,EFR,HR and SILK codecs. Then it’s extended to identify subframe offset to do tampering detection of cellphone speech recordings.

Audio signal bandwidth extension in celp-based speech coder

Abstract: A method for decoding an audio signal having a bandwidth that extends beyond a bandwidth of a CELP excitation signal in an audio decoder including a CELP-based decoder element. The method includes obtaining a second excitation signal having an audio bandwidth extending beyond the audio bandwidth of the CELP excitation signal, obtaining a set of signals by filtering the second excitation signal with a set of bandpass filters, scaling the set of signals using a set of energy-based parameters, and obtaining a composite output signal by combining the scaled set of signals with a signal based on the audio signal decoded by the CELP-based decoder element.

Environmental sound recognition with CELP-based features

Abstract: In this work, we propose the use of a set of new features based on CELP (Code Excited Linear Prediction) to enhance the performance of the environmental sound recognition (ESR) problem. Traditionally, Mel Frequency Cepstral Coefficients (MFCC) have been used for the recognition of structured data like speech and music. However, their performance for the ESR problem is limited. An audio signal can be well preserved by its highly compressed CELP bit streams, which motivates us to study the CELP-based features for the audio scene recognition problem. We present a way to extract a set of features from the CELP bit streams and compare the performance of ESR using different feature sets with the Bayesian network classifier. It is shown by experimental results that the CELP-based features outperform the MFCC features in the ESR problem by a significant 9% margin in average and the integrated MFCC and CELP-based feature set can even reach a correct classification rate of 95.2% using the Bayesian network classifier.

One-Pulse FEC Coding for Robust CELP-Coded Speech Transmission Over Erasure Channels

Abstract: In this paper, we present an improved quantization scheme for the redundancy data of a forward error correction (FEC) technique proposed for the transmission of code-excited linear prediction (CELP)-coded speech over erasure channels. The use of a FEC-based error protection scheme is motivated by the well-known fact that, after a frame erasure, the previous excitation is not available and a desynchronization between the encoder and the decoder long-term prediction (LTP) filters appears, causing an additional distortion which is propagated to subsequent frames. LTP synchronization can be recovered by means of a single-pulse representation of the previous excitation. No additional delay is introduced by this technique which only requires a small transmission bandwidth increase. In this paper, we focus on the efficient encoding of this pulse. Thus, an optimization procedure, which takes into account the overall synthesis error, is proposed in order to provide better pulse-position and pulse-amplitude quantization codebooks. Moreover, by extending the previous procedure, an efficient joint position-amplitude quantization can be obtained. Objective quality tests applied to our proposal show that, by means of the proposed codebooks, the number of bits required to represent the resynchronization pulse is effectively reduced. In addition, a discontinuous transmission mechanism is derived from the cost functional used during joint position-amplitude quantization, further reducing the bit-rate.

Predictive coding with block shapes derived from a prediction error

Abstract: The present invention relates to block-wise coding and decoding of a video signal including at least two color components. The first component is coded by using prediction and the second component is segmented to different parts used for its coding according to the prediction error.

Low-delay unified speech and audio codec

Abstract: A unified speech and audio decoder is described, which comprises a frame buffer configured to buffer a sub-part of a datastream composed of consecutive frames in units of the frames so that the subpart continuously comprises at least one frame, each frame representing a coded version of a respective portion of consecutive portions of an audio signal, and each frame comprising a mode identifier assigning the respective frame to a respective one of a plurality of coding modes comprising a CELP (codebook excitation linear prediction) coding mode and a transform coded excitation linear prediction coding mode. Further, the unified speech and audio decoder comprises a CELP decoder configured to decode the frames to which the CELP coding mode is assigned to reconstruct the respective portions, and a transform coded excitation linear prediction decoder configured to decode the frames to which the transform coded excitation linear prediction coding mode is assigned, to reconstruct the respective portions, wherein the frame buffer is configured to distribute the frames buffered to the CELP decoder and the transform coded excitation linear prediction decoder under removal of the respective frames from the frame buffer, frame-wise.

Effect of Low Bit Rate Speech Coding on Epoch Extraction

Abstract: Speech coding is one of the major degradation involved in building the speech systems in mobile environment. In this paper, we are exploring the effect of low bit rate speech coding on the accuracy of detection of epochs. Epoch is referred as the instant of significant excitation of the vocal-tract system during production of speech. Many speech applications depend on the the accurate estimation of the epoch locations. Epoch extraction from speech signal is challenging due to time-varying characteristics of the excitation source and vocal-tract system. For determining the epochs, two recently developed accurate methods (i) zero frequency filter (ZFF) (ii) Dynamic programming projected phase slope (DYPSA) are used. Most of the epoch extraction methods except ZFF method, attempt to remove the characteristics of the vocal-tract system, in order to emphasize the excitation characteristics in the residual. ZFF method extracts the epoch locations directly from the speech signals using impulse like nature of excitation. Speech coders used in this study are GSM full rate (ETSI 06.10), CELP (FS-1016), and MELP (TI 2.4 kbps). Performance of epoch extraction methods is evaluated using CMU-Arctic data using the epoch locations from electro-glottograph as reference.

Implementation of CELP CODER and to evaluate the performance in terms of bit rate, coding delay and quality of speech

Abstract: Factors serving as constraints in today's wireless communication system include bandwidth and power. In wireless systems that require the transmission of speech, these goals are addressed by developing efficient methods of reducing the amount of information required to transmit and receive quality speech. For this reason, speech coding has been, and remains, the topic of aggressive research. This paper discusses the implementation of CELP CODEC and its analytical evaluation of performance in terms of bit rate, coding delay and Quality of speech. The CELP coder is one of the best methods for producing high quality speech at bit rates between 4.8 and 9.6 Kbps.

Comparison of coding schemes for genetic algorithms

Abstract: Designing a rational coding scheme for a concrete problem is one of application difficulties of genetic algorithms,but up to now there is no uniform solution to itThis paper focused on analyzing and comparing several popular coding schemes for genetic algorithms such as binary coding,real coding,matrix coding,tree coding and quantum coding,then summarized their principles,advantages and disadvantages,application scopes and application trendsFurthermore,pointed out some future research directions for coding schemes

Hybrid Scalar/Vector Quantization of Mel-Frequency Cepstral Coefficients for Low Bit-Rate Coding of Speech

Abstract: In this paper, we propose a low bit-rate speech codec based on a hybrid scalar/vector quantization of the mel-frequency cepstral coefficients (MFCCs). We begin by showing that if a high-resolution mel-frequency cepstrum (MFC) is computed, good-quality speech reconstruction is possible from the MFCCs despite the lack of explicit phase information. By evaluating the contribution toward speech quality that individual MFCCs make and applying appropriate quantization, our results show perceptual evaluation of speech quality (PESQ) of the MFCC-based codec matches the state-of-the-art MELPe codec at 600 bps and exceeds the CELP codec at 2000 -- 4000 bps coding rates. The main advantage of the proposed codec is in distributed speech recognition (DSR) since speech features based on MFCCs can be directly obtained from code words thus eliminating additional decode and feature extract stages.

Encoder apparatus and encoding method

Abstract: Provided is an encoder apparatus that can suppress the quality degradation of encoding processes. An ultimate selection candidate limiting unit (109) uses the spectrum of an input signal and a residual spectrum to designate a given number of pre-selected suppression factors to a CELP component suppressing unit (104); the CELP component suppressing unit (104) uses the designated suppression factors to generate a suppressed spectrum; a CELP residual signal spectrum calculating unit (105), to which the suppressed spectrum is input, calculates a residual spectrum; a conversion encoding unit (110) uses the residual spectrum to performs a second encoding process; and a distortion evaluating unit (112) determines one of the designated suppression factors by use of the spectrum of a second decoded signal generated by decoding a second code obtained by the second encoding process and further by use of the suppressed spectrum and the spectrum of the input signal.

Speech compression exploiting linear prediction coefficients codebook and hamming correction code algorithm

Abstract: In this paper, a replacement algorithm for Linear Prediction Coefficients (LPC) along with Hamming Correction Code based Compressor (HCDC) algorithms are investigated for speech compression. We started with an CELP system with order 12 and with Discrete Cosine Transform (DCT) based residual excitation. Forty coefficients with transmission rate of 5.14 kbps were first used. For each frame of the testing signals we applied a multistage HCDC, we tested the compression performance for parities from 2 to 7, we were able to achieve compression only at parity 4. This rate reduction was made with no compromise in the original CELP signal quality since compression is lossless. The compression approach is based on constructing dynamic reflection coefficients codebook, this codebook is constructed and used simultaneously using a certain store/retrieve threshold. The initial linear prediction codec we used is excited by a discrete cosine transform (DCT) residual, the results were tested using the MOS and SSNR, we had acceptable ranges for the MOS (average 3.6), and small variations of the SSNR (±5 db).

Flexible and scalable combined innovation codebook for use in CELP coder and decoder

Abstract: In a CELP coder, a combined innovation codebook coding device comprises a pre-quantizer of a first, adaptive-codebook excitation residual, and a CELP innovation-codebook search module responsive to a second excitation residual produced from the first, adaptive-codebook excitation residual. In a CELP decoder, a combined innovation codebook comprises a de-quantizer of pre-quantized coding parameters into a first excitation contribution, and a CELP innovation-codebook structure responsive to CELP innovation-codebook parameters to produce a second excitation contribution.

Encoding device, decoding device, spectrum fluctuation calculation method, and spectrum amplitude adjustment method

Abstract: Disclosed is an encoding device whereby it is possible to improve the quality of an encoded signal, even when encoding music signals. In the encoding device, a Code-Excited Linear Prediction (CELP) encoder (101) generates first encoded data by encoding an input signal, a CELP decoder (102) generates a decoded signal by decoding the first encoded data input from the CELP encoder (101), and a characteristic parameter encoder (106) calculates a parameter that expresses the degree of fluctuation in the ratio of the peak components and the floor components between the spectra of the decoded signal and the input signal.

Decoder, encoder, and methods thereof

Abstract: Disclosed is a decoder capable of improving the sound quality of a decoded sound signal in an encoding method which combines speech encoding and music encoding in a hierarchical structure. A transform-encoding decoding unit (202) decodes transform-encoded data to generate a spectrum of a decoded transform-encoded signal. A band decision unit (203) uses the spectrum of the decoded transform-encoded signal to decide whether each of a plurality of bands in which frequency components of an input signal are divided constitute a first band in which a transform encoded pulse is not established or a second band in which said pulse is established. A CELP component suppression unit (207) suppresses the spectrum of a CELP decoded signal, which is the frequency component of a decoded signal of CELP encoded data, to the extent that suppression in the first band is weaker than suppression in the second band.

Compressed domain speech enhancement based on the joint modification of codebook gains

Abstract: A compressed domain speech enhancement method based on the joint modification of adaptive and algebraic codebook gains for the codec of ITU-T G.722.2 is proposed in this paper. First the power of excitation signal corresponding to the noise is estimated by the method of minimum statistics. Then the decision-directed approach is used to get an estimate of the a priori SNR. And the algebraic codebook gain is modified by multiplying a Wiener-type modification factor. In order to solve the problem of power loss in voiced segment, the modified adaptive codebook gain is got by keeping the power of the modified excitation signal equal to the scaled version of the noisy one. The result of performance evaluation under ITU-T G.160 shows that, in comparison with the method that only modifies the algebraic codebook gain, the proposed method could provide larger amount of noise reduction in both white and colored noise with smaller attenuation on the speech level, and the objective speech quality is improved evidently.

The Equivalence of ADPCM and CELP Coding

Abstract: to Remix – to adapt this work Subject to conditions outlined in the license. This work is licensed under the Creative Commons Attribution 3.0 Unported License. To view a copy of this license, visit

An interactive speech coding tool using LabVIEW

Abstract: Code Excited Linear Prediction (CELP) is a closed-loop analysis-by-synthesis speech coding algorithm that has been standardized in Federal Standard-1016. Variants of the CELP algorithm form the core of many speech coding standards that exist today. In this paper, we discuss the development of an interactive speech coding tool in National Instruments LabVIEW™ software for the Federal Standard-1016 CELP algorithm. A brief description of the speech coding algorithm and the features of the LabVIEW speech coding tool are presented. Illustrations demonstrating the use of the interactive software tool in analyzing the speech coding algorithm are provided. This tool can be used to teach the various modules of the CELP based speech coders to undergraduate and graduate students.

Analysis of Fundamental Frequency Contour of Coded Speech Based on Multi-Pulse Based Code Excited Linear Prediction Algorithm

Abstract: Problem statement: In low-bit-rate speech communication, speech coding deteriorates the characteristics of the coded speech significantly. An important feature of the speech is the fundamental frequency contour which determines the pitch information of the speech. It has been known that pitch information is one of the core parameter of the multi-pulse based code excited linear prediction (MPCELP) speech coder. Therefore the study of the deteriorated fundamental frequency contour should be conducted properly. Approach: This study proposes an analysis of the fundamental frequency contour of the coded speech based on MP-CELP speech coder. The comparison of the fundamental frequency contour of the natural speech and that of the coded speech has been performed. The MP-CELP with three levels of bitrate scalability is selected as the core speech coder. The speech material includes a hundred of male speech utterances and a hundred of female speech utterances. Results: The experimental results show that the speech coder causes the deterioration of the fundamental frequency contour empirically. The Root Mean Square Error (RMSE) between the fundamental frequency contour of the natural speech and that of the coded speech for three different bitrates has been conducted. The lower bitrate causes the higher value of RMSE. Conclusion: From the study, it is a proved that the MP-CELP speech coder deteriorates the fundamental frequency contour of the transmitted speech.

Encoding method, decoding method, encoding device, decoding device, and program

Abstract: PROBLEM TO BE SOLVED: To improve encoding quality when a frame to be encoded by a first encoding system does not continue when each frame is encoded by the first encoding system such as a CELP (Code-Excited Linear Prediction) encoding system or a second encoding system such as a frequency encoding system. SOLUTION: When a frame is encoded by the first encoding system or when a code encoded by the first encoding system is decoded, a decoded signal corresponding to a time-series signal at each point belonging to a past frame is linearly predicted and analyzed, and the obtained residual signal is used as a substitute for an excitation signal. COPYRIGHT: (C)2011,JPO&INPIT

Encoder apparatus and method that perform preliminary signal selection for transform coding before main signal selection for transform coding

Abstract: An encoder apparatus is provided that suppresses the quality degradation of encoding processes. An ultimate selection candidate limiting unit uses the spectrum of an input signal and a residual spectrum to designate a given number of pre-selected suppression factors to a CELP component suppressing unit, which uses the designated suppression factors to generate a suppressed spectrum. A CELP residual signal spectrum calculating unit, to which the suppressed spectrum is input, calculates a residual spectrum. A conversion encoding unit uses the residual spectrum to perform a second encoding process. A distortion evaluating unit determines one of the designated suppression factors by use of the spectrum of a second decoded signal generated by decoding a second code obtained by the second encoding process, and further by use of the suppressed spectrum and the spectrum of the input signal.

Complexity scalability design in coding of the adaptive codebook for ITU-T G.729 speech coder

Abstract: Using the adaptive codebook and stochastic codebook, the code-excited linear prediction (CELP) speech coders can achieve a good speech quality at low bit rate. When the complexity reduction of the stochastic codebook search reaches a bottleneck, the only way to reduce the complexity of the speech codec would be simplifying the computation of the adaptive codebook. This paper introduces and describes the complexity scalability design for the adaptive codebook search of the G.729 speech coder. Two schemes, the search-range reduction and the dimension-reduced method, for complexity reduction will be discussed. Simulation results show that the proposed approaches can effectively reduce the computational complexity of the close-loop pitch search with imperceptible degradation of the speech quality.

Sparse signal approximation algorithms in a CELP coder

Abstract: This work presents a series of sparse signal modeling algorithms implemented in a typical CELP coder in order to compare their performances at a reasonable computational load. New algorithms are proposed, based on cyclic and parallel use of a fast implementation of the optimized orthogonal matching pursuit algorithm, i.e. the recursive modified Gram - Schmidt algorithm. These algorithms yield a statistically significant reduction of signal approximation error at a controllable computational complexity.

Frame error-robust MDCT bit reduction scheme using inter-frame correlation for G.729.1

Abstract: In this paper, a new algorithm for encoding spectral envelope in G.729.1 for VoIP is proposed. In the TDAC part of G.729.1, the spectral envelope and MDCT coefficients extracted in the weighted CELP coding error (lower-band) and the higher-band input signal are encoded. In order to reduce allocation bits for spectral envelope coding, a new algorithm using sub-band correlation between adjacent frames is proposed. In addition, a new bit allocation strategy using reduced bits from the proposed algorithm is also proposed. The performance of the proposed algorithm is evaluated by sound quality and bit reduction rates in clean and frame loss conditions.

Chaos for speech coding and production

Abstract: The presence of nonlinearity and chaotic behavior in the human speech production system has been reported previously; however to date chaotic dynamics has not been widely exploited in speech coding and artificial speech production algorithms. In this paper we illustrate how we can utilize chaotic dynamics in speech coding and synthesis and discuss how it can improve the performance of these processes. As an example we choose code-excited linear predictive coding and, instead of an excitation codebook consisting of Gaussian random waveforms, we use chaotic systems to produce chaotic excitations. This simple technique has the potential to greatly improve the efficiency of modern communication devices such as cell phones. We call the resulting scheme chaos-excited linear predictive coding.

Encoding apparatus, decoding apparatus, encoding method, and decoding method for adjusting a spectrum amplitude

Abstract: An encoding device is provided for increasing the quality of an encoded signal, even when encoding music signals. In the encoding device, a Code-Excited Linear Prediction (CELP) encoder generates first encoded data by encoding an input signal, and a CELP decoder generates a decoded signal by decoding the first encoded data input from the CELP encoder. Additionally, a characteristic parameter encoder calculates a parameter that expresses the degree of fluctuation in the ratio of the peak components and the floor components between the spectra of the decoded signal and the input signal.