Showing papers on "Code-excited linear prediction published in 2018"

CELPB: A Cache Invalidation Policy for Location Dependent Data in Mobile Environment

Abstract: Location dependent information services (LDIS) can be characterized as the applications that coordinate a cell phone's area or position with other data to give enhanced value of services to the client at right place in the right time from anywhere. In this paper, an algorithm Caching Efficiency with Next Location Prediction Based (CELPB) has been developed that uses a newly developed metric i.e. caching efficiency with next location prediction (CELP) for the computation of valid scope in prediction interval. This metric takes account the future movement behavior of client with the help of Sequential Pattern Mining and Clustering. The mobility rules have also been framed for the prediction of an accurate next location, which can be used in estimating the future movement path (edges) of client if he reached in valid scope area of any data item. Simulation results show that proposed policy achieves up to 10 percent performance improvement compared to earlier cache invalidation policy (CEBAB) for LDIS.

SILK Steganography Scheme Based on the Distribution of LSF Parameter

Abstract: SILK, as a speech codec for real-time packet-based voice communications, which is widely used in many popular mobile Internet application, such as Skype, WeChat, QQ, WhatsApp, etc. It will be a novel and ideal carrier for information hiding. In this paper, a secure steganography scheme for SILK is proposed, which embeds secret message by modifying the LSF (Line Spectral Frequency) quantization indices based on the statistical distribution of LSF Codebook. The experimental results show that the auditory concealment of the proposed scheme is excellent, the decrease in PESQ is very small. The average hiding capacity can achieve 129 bps and 223 bps under the sampling rate of 8 kHz and 16 kHz respectively. More importantly, the proposed scheme has good statistical security. In this scheme, the statistical distribution of LSF Codebook is considered as a constraint condition to make the distribution of stego's codeword close to that of the cover audio. Under the steganlysis scheme which is referenced from the existing steganlysis scheme for G.723.1, the average correct detection rate is under 55.4% for both cover and stego audio. To the best of our knowledge, this is the first work to hide information in SILK. Based on the similar principle of speech compression, the method can be extended to other CELP codec, such as G.723.1, G.729, AMR, etc.

A Simulation-Based Comparison on Code Excited Linear Prediction (CELP) Coder at Different Bit Rates

Abstract: A better speech quality is necessary for transmission of speech in mobile communication systems in Digital Telephony. Nowadays, the conversion of analog speech signals into digital format is required for effective transmission of speech over different channels at far ends. The technique for conversion of speech in digital form is very old and ordinary one which is termed as pulse code modulation (PCM), but the bandwidth of the digitally converted data is too large, so a better level of compression is needed to reduce the bandwidth and enhance the capacity of channel. The compression of speech in nowadays is performed by a procedure called speech coding. In this paper, the code excited linear predictive (CELP) coding is summarized with different bit rates. The MATLAB R2016a version is used for simulating the 9.6 and 16 kbps CELP coder, and performance analysis is done with parameters MSE and SNR.

An investigation on the degradation of different features extracted from the compressed American English speech using narrowband and wideband codecs

Abstract: Speech coding facilitates speech compression without perceptual loss that results in the elimination or deterioration of both speech and speaker specific features used for a wide range of applications like automatic speaker and speech recognition, biometric authentication, prosody evaluations etc. The present work investigates the effect of speech coding in the quality of features which include Mel Frequency Cepstral Coefficients, Gammatone Frequency Cepstral Coefficients, Power-Normalized Cepstral Coefficients, Perceptual Linear Prediction Cepstral Coefficients, Rasta-Perceptual Linear Prediction Cepstral Coefficients, Residue Cepstrum Coefficients and Linear Predictive Coding-derived cepstral coefficients extracted from codec compressed speech. The codecs selected for this study are G.711, G.729, G.722.2, Enhanced Voice Services, Mixed Excitation Linear Prediction and also three codecs based on compressive sensing frame work. The analysis also includes the variation in the quality of extracted features with various bit-rates supported by Enhanced Voice Services, G.722.2 and compressive sensing codecs. The quality analysis of extracted epochs, fundamental frequency and formants estimated from codec compressed speech was also performed here. In the case of various features extracted from the output of selected codecs, the variation introduced by Mixed Excitation Linear Prediction codec is the least due to its unique method for the representation of excitation. In the case of compressive sensing based codecs, there is a drastic improvement in the quality of extracted features with the augmentation of bit rate due to the waveform type coding used in compressive sensing based codecs. For the most popular Code Excited Linear Prediction codec based on Analysis-by-Synthesis coding paradigm, the impact of Linear Predictive Coding order in feature extraction is investigated. There is an improvement in the quality of extracted features with the order of linear prediction and the optimum performance is obtained for Linear Predictive Coding order between 20 and 30, and this varies with gender and statistical characteristics of speech. Even though the basic motive of a codec is to compress single voice source, the performance of codecs in multi speaker environment is also studied, which is the most common environment in majority of the speech processing applications. Here, the multi speaker environment with two speakers is considered and there is an augmentation in the quality of individual speeches with increase in diversity of mixtures that are passed through codecs. The perceptual quality of individual speeches extracted from the codec compressed speech is almost same for both Mixed Excitation Linear Prediction and Enhanced Voice Services codecs but regarding the preservation of features, the Mixed Excitation Linear Prediction codec has shown a superior performance over Enhanced Voice Services codec.

Noise filling without side information for celp-like coders

Abstract: FIELD: physicsSUBSTANCE: invention relates to means for encoding audio Audio decoder for providing decoded audio information on the basis of encoded audio information comprising linear prediction coefficients, comprises a tilt adjuster configured to adjust a tilt of background noise using tilt information; a decoder core configured to decode the audio information of the current frame using the linear prediction coefficients of the current frame to obtain the decoded main output of the encoder; and a noise inserter configured to add the adjusted background noise to the current frame in order to perform noise filling, wherein the tilt adjuster is configured to obtain tilt information by calculating the increment g of the linear prediction coefficients of the current frameEFFECT: technical result is improved quality of audio coding17 cl, 11 dwg

Low Complex Simple Measurement Matrix for Sparse Recovery of Speech Signal

Abstract: Compressed Sensing (CS), the methodology of signal capturing, allows sampling at flexible rates below Nyquist, with the constraint that the sparsifying basis and the level of sparsity are known in advance for the signal of interest. Many speech codecs based on CS frame work are developed using Linear Predictive Coding (LPC), Discrete Cosine Transform (DCT) and Code Excited Linear Prediction (CELP). In most of them, Gaussian random matrix is used for deriving the observation vector which is computationally complex and has large memory requirements. In this paper, a modified binary sensing matrix, specifically for speech signal is proposed, which has low coherence with the sparsifying bases used for reconstruction. The Signal-to-Noise Ratio (SNR) improvement goes beyond 3-4 dB and it is more significant at very high compression ratios. The application of the proposed sensing matrix to CS based codecs using CELP and dynamic DCT&LPC bases shows significant improvement in the perceptual quality of the reconstructed speech. This enables the functioning of these codecs at lower bit rates without compromising the quality.

Improvement of classification between time-domain coding and frequency-domain coding

Abstract: FIELD: data processing.SUBSTANCE: invention relates to means for coding signals. In the method, frequency-domain coding or time-domain coding is selected based on the bit-rate of the coding, which must be used to encode the digital signal, and to detect a short lag of the digital signal's pitch. Further, the detection of a short pitch lag comprises detecting whether the digital signal contains a short pitch signal, for which the pitch lag is shorter than the restriction of the pitch lag, wherein the pitch lag restriction is the minimum permissible pitch for the code excited linear prediction (CELP) algorithm for encoding a digital signal.EFFECT: technical result is to improve the classification efficiency between time-domain coding and frequency-domain coding.14 cl, 15 dwg

Improvement in non-voice content about low rate celp decoder

Abstract: PROBLEM TO BE SOLVED: To provide a method and a device for modifying a composite of time domain excitation decoded with a time domain decoder.SOLUTION: In a method and a device for modifying a composite of time domain excitation decoded with a time domain decoder, the composite of the decoded time domain excitation is classified into one of categories. The decoded time domain excitation is converted into frequency domain excitation which is then modified in accordance with the category where the composite of the decoded time domain excitation is classified thereinto. The modified frequency domain excitation is converted into modified time domain excitation which is then supplied to a composite filter to generate the modified composite of the decoded time domain excitation.SELECTED DRAWING: Figure 1