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GSM speech coder Indirect Identification Algorithm

INFORMATICA 2010年第4期21卷 575-596页

作者： Svecko, Rajko Kotnik, Bojan Chowdhury, Amor Mezgec, Zdenko Univ Maribor Fac Elect Engn & Comp Sci SI-2000 Maribor Slovenia Ultra Doo Res Ctr SI-2000 Maribor Slovenia

This paper presents GSM speech coder indirect identification algorithm based on sending novel identification pilot signals through the GSM speech channel. Each GSM subsystem disturbs identification pilot, while speech coder uniquely changes the tempo-spectral characteristics of the proposed pilot signal. speech coder identification algorithm identifies speech coder with the usage of robust linear frequency cepstral coefficient (LFCC) feature extraction procedure and fast artificial neural networks. First step of speech coder identification algorithm is the exact position detection of the identification pilot signal using normalized cross correlation approach. Next stage is time-domain windowing of the input signal to convolve each frame of the input speech signal and window spectrum. Consecutive step is a short-time Fast Fourier Transformation to produce the magnitude spectrum of each windowed frame. Further, a noise reduction with spectral subtraction based on spectral smoothing is carried out. In last steps we perform the frequency filtering and Discrete Cosine Transformation to receive 24 uncorrelated cepstral coefficients per frame as a result. speech coder identification is completed with fast artificial neural network classification using the input feature vector of 24 LFCC coefficients, giving a result of identified speech coder. For GSM speech coder indirect identification evaluation, the standardized GSM ETSI bit-exact implementations were used. Furthermore, a set of custom tools was build. These tools were used to simulate and control various conditions in the GSM system. Final results show that proposed algorithm identifies the GSM-EFR speech coder with the accuracy of 98.85%, the GSM-FR speech coder with 98.71%, and the GSM-HR coder with 98.61%. These scores were achieved at various types of surrounding noises and even at very low SNR conditions.

关键词： GSM speech coder identification pilot

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A packet-based CAPDM speech coder for PCN applications

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IEEE TRANSACTIONS ON VEHICULAR TECHNOLOGY 2000年第3期49卷 753-765页

作者： Liu, CH Huang, CC Natl Chiao Tung Univ Dept Commun Engn Hsinchu Taiwan

In this paper, we present a median-rate speech coder, the controlled adaptive prediction delta modulation coder (CAPDM), which operates at 16 kb/s with good speech quality and low algorithm complexity [15], The coder is dedicated to personal communication network (PCN) applications and transmits speech samples on the basis of packets. It combines the features of one-step looking forward decision, syllabic companding, instantaneous companding, and adaptive prediction. In addition to the use of a short-term prediction filter, CAPDM also exploits the pitch property to predict speech waveform explicitly. With the aid of a pitch prediction filter, the performance of a CAPDM codec improves about 3 dB in segmental signal-to-noise ratio (SEGSNR), The average SEGSNR of *** is about 21 dB, which is 7 dB over traditional CVSD at 16 kb/s, We also utilize an adaptive postfilter (APF) to enhance the perceptual quality of the decoded speech. The mean opinion score (MOS) listening test of *** with APF shows that its average score achieves 4.19, which is as good as G.728 16-kb/s LD-CELP and is comparable with CCITT G.721 32-kb/s ADPCM, The complexity of *** is evaluated to be 8 MIPS, which Is much lower than that of LD-CELP and could be further reduced by adopting a smaller correlation window for pitch detection. To solve the problem of packet loss, we developed a packet-based waveform substitution method by reinitializing the codec parameters at the beginning of each packet. The simulation results show that *** could tolerate 5% of packet loss and still keep an SEGSNR at 10 db and an MOS at about 3.0.

关键词： adaptive prediction instantaneous companding packet recovery pitch detection speech coder syllabic companding waveform substitution

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Low bit-rate CELP speech coder with low delay

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SIGNAL PROCESSING 1999年第2期72卷 97-105页

作者： Hayashi, S Suguimoto, M Erinnoviar Takushoku Univ Fac Engn Dept Elect & Syst Tokyo 1938585 Japan

This paper presents a speech coding algorithm for a low bit-rate backward prediction coder with a bit-rate of less than 4 kbit/s, and a codebook design with a closed-loop training procedure. The coding system proposed is based on the backward prediction algorithm in LD-CELP and results in a 13.5 ms coding delay, which is 1/7 the length of delay in conventional forward prediction coders at the same transmission bit-rates. The codebook design is carried out in a closed-loop training process to generate the codebook so that we can represent harmonic portions of voiced speech appropriately. Finally, we carried out a simulation for the performance of the low-delay coder system trained in closed-loop method with the proposed initial codebook. We evaluated the effectiveness of the initial codebook with the optimal assignment and found that its reconstructed speech was clear enough to understand though it contained a slight element of noise. (C) 1999 Elsevier Science B.V. All rights reserved.

关键词： speech coder backward prediction low delay low bit-rate closed-loop

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SWITCHED PREDICTOR ADAPTATION OF DPCM-AQB speech coder USING AN IMBEDDED VECTOR QUANTIZATION

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ELECTRONICS LETTERS 1985年第10期21卷 442-444页

作者： RAMAMOORTHY, V Department of Electrical Engineering University of Linköping Linköping WS-58183 Sweden

An adaptive differential pulse-code-modulation speech coder with a switched predictor adaptation scheme is described. The predictor adaptation is done by switching one of the several predetermined predictors stored both at the encoder and the decoder. The selection of the predictor is controlled by a vector quantiser. The tradeoff between the performance and the extra side information that needs to be transmitted is evaluated for bit rates of 16, 24 and 32 Kbit/s.

关键词： filtering and prediction theory encoding side information speech processing speech analysis and processing Pulse-code modulation imbedded vector quantisation Codes bit rates pulse code modulation switched predictor adaptation scheme speech coder adaptive differential pulse-code-modulation

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PLP BASED CELP speech coder

PLP BASED CELP SPEECH CODER

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2nd International Conference on Computer and Electrical Engineering

作者： Najafi, Jafar Marvi, Hossein Shahrood Univ Technol Fac Elect & Robot Engn Shahrood Iran

ISBN: (纸本)9780769539256

Perceptual linear prediction (PLP) is widely used in speech recognition systems as a feature extraction method. Also code-Exited Linear Prediction coder (CELP) is one of the well known speech coders which widely used in communication systems. In this paper the application of PLP in speech coding has been discussed. In the first stage the parameters of formant synthesis filter are determined by applying PLP algorithms. Then these parameters are used in coder, code-exited linear prediction coder, to improve the efficiency of this kind of coder. The experiments show promising result in some cases.

关键词： speech coder Linear Prediction Coefficients Perceptual Linear Prediction CELP coder

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A 2.4kb/s speech coder with Improved Analysis and Synthesis Based on MELP

A 2.4kb/s Speech Coder with Improved Analysis and Synthesis ...

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3rd International Conference on Frontiers of Manufacturing Science and Measuring Technology (ICFMM 2013)

作者： Li, Qiang Zhang, Xiaohong Xia, Xujiu Zhang, Haoran Chongqing Univ Posts & Telecommun Chongqing Key Lab Signal & Informat Proc Chongqing Peoples R China

ISBN: (纸本)9783037858462

This paper describes a speech coder which has an input signal frame interval of 20ms percent in duration, contains 160 voice samples (8,000 samples/s) and 48 bits. A key feature of the coder is a novel Line Spectral Frequencies (LSF) quantization scheme, requiring only 19 bits per frame. This new coder, through algorithmic improvements, enhanced quantization techniques and new harmonic synthesis method, produces better speech quality at 2.4 kb/s transnission rate than the new U.S. Federal Standard MELP coder.

关键词： speech coder MELP LSF quantization

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A proposed method for complexity reduction algorithm of Wideband AMR speech coder

A proposed method for complexity reduction algorithm of Wide...

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3rd IASTED International Conference on Circuits, Signals, and Systems

作者： Yoon, BS Kim, HJ Park, MH Gee, DG Choi, SI ETRI Convergence Terminal Res Team Taejon 305350 South Korea

ISBN: (纸本)0889865078

3(rd) Generation Project Plan (3GPP) has standardized the Wideband Adaptive Multi-rate (AMR) speech coder for 3GPP-based International Mobile Telecommunications 2000 (IMT-2000) system. This paper proposed a new method to reduce the complexity of wideband AMR speech coder. Using this method the complexity can be reduced to half of original method with no speech quality degradation. We describe the efficient software scheme and sophisticated hardware design for real-time implementation of wideband AMR speech. We verified complexity reduced algorithm performance using SNR and ITU-T P.862 (PESQ) measure. We also evaluated its real-time operating performance using hard ware test system.

关键词： speech coder real-time implementation AMR

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Bstj 61: 9. November 1982: a 9.6-Kb/S Dsp speech coder. (Crochiere, R.E.;Cox, R.V.;Johnston, J.D.;Seltzer, L.A.)

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2016年

Bstj 61: 9. November 1982: a 9.6-Kb/S Dsp speech coder. (Crochiere, R.E.; Cox, R.V.; Johnston, J.D.; Seltzer, L.A.) by published by

关键词： (crochiere, 9.6-kb s algorithm bell system block diagram coder coder. compression dsp frequency j.d. pitch pitch detector sampling rate sbc shift register signal speech speech coder step size system technical tdhs tdhs algorithm tdhs compression

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Research of G.723.1 speech coder

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微计算机信息 2002年第8期 72-73+75页

作者： College of Computer & Comniu. Engi. Southwest Jiaotong Univ., Sichuan ?Chengdu 610031 Zhang Shaohua and Li Chengzhong College of Computer & Commu. Engi. Southwest Jiaotong Univ. Sichuan. Chengdu610031

This paper introduces concept of G.723.1 speech coder and analyses its technology and features. We advise to optimize its running time of G.723.1 speech coder. We put forward to improve some modules with large computa... 详细信息

关键词： speech coder linear predictive coding optimization

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Intelligibility evaluation of GSM coder for Mandarin speech using CDRT

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speech COMMUNICATION 2002年第1-2期38卷 161-165页

作者： McLoughlin, I Ding, ZQ Tan, EC Nanyang Technol Univ Sch Comp Engn Singapore 639798 Singapore

The Global system for Mobile telecommunications (GSM) speech-coding algorithm is evaluated by using the newly proposed Chinese Diagnostic Rhyme Test (CDRT) to determine its suitability for coding Mandarin speech. Experimental results show that the algorithm is not effective for all phonetic features in Mandarin and it reduces intelligibility of sibilated consonants. (C) 2002 Elsevier Science B.V. All rights reserved.

关键词： speech coder GSM Chinese Diagnostic Rhyme Test intelligibility

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