Low-Delay Speech Coding at 16 kb/s and Below (original) (raw)
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Proceedings of the Second International Conference on Research in Intelligent and Computing in Engineering, 2017
A fair level of speech quality is desired in speech transmission for mobile voice services. The effective utilization of bandwidth and higher bit rate is must for a best quality speech coder. But at a time the both requirements are not fulfilled in desired format. The research is ongoing in the area of designing speech coder's. In general the CELP is an algorithm to design a good quality speech coder. From 80's to present the advancement in this technique is going on. In this paper a wide band speech coding technique is proposed using LD-CELP algorithm. The overall performance of LD-CELP (16Kbps) is summarized and computed on MATLAB version R2016a with parameters MSE and SNR. In conclusion we observe that SNR for LD-CELP is not much better and enhancement in this is necessary.
Code-excited linear prediction(CELP): High-quality speech at very low bit rates
1985
We describe in this paper a code-excited linear predictive coder in which the optimum innovation sequence is selected from a code book of stored sequences to optimize a given fidelity criterion. Each sample of the innovation sequence is filtered sequentially through two time-varying linear recursive filters, one with a long-delay (related to pitch period) predictor in the feedback loop and the other with a short-delay predictor (related to spectral envelope) in the feedback loop. We code speech, sampled at 8 kHz, in blocks of 5-msec duration. Each block consisting of 40 samples is produced from one of 1024 possible innovation sequences. The bit rate for the innovation sequence is thus 1/4 bit per sample. We compare in this paper several different random and deterministic code books for their effectiveness in providing the optimum innovation sequence in each block. Our results indicate that a random code book has a slight speech quality advantage at low bit rates. Examples of speech produced by the above method will be played at the conference.
A New Deterministic Codebook Structure for Celp Speech Coding
Proceedings. IEEE International Symposium on Information Theory, 1993
Low bit rate, high quality speech coding is a vital part in voice telecommunication systems. The introduction of CELP (1984) (Codebook Excited Linear Prediction) speech coding provided a feasible way to compress speech data to 4.8 kbps with high quality, but the formidable computational complexity required for real-time processing has prevented its wide application. Using the new deterministic codebook, we reduce the computational complexity of codebook search, which originally accounted for 2/3 of the computational complexity, to negligible. Based on this reduction, we produce an algorithm with complexity about 5 MIPS. It can be implemented in even inexpensive DSP chips, while maintaining the same high quality. In addition to extremely simple encoding and decoding schemes, this codebook also provides optimal error tolerance and it doesn't require codebook storage. We hope that this contribution can finally make CELP speech coding a widely applicable and practical technology.
QUALITY EVALUATION OF LPC BASED LOW BIT RATE SPEECH CODERS
IAEME PUBLICATION, 2021
Significant improvements are being reported these days in the coding of speech signals “with high quality at low bit rates. The need for low-bitrate speech coding algorithms continues, supported by the ever-increasing number of users” in wireless communication networks. Linear Prediction coders form important class speech coders with low bitrates. This paper describes the software level simulation and performance evaluation of CELP (4.8Kbps), LD-CELP (16 Kbps), MELP (2.4 Kbps), and CS-ACELP (8Kbps, 6.4 Kbps) speech coders. Even though all e coder outputs match with the input signal, the results from the error resilience test and MOS score show that the output quality of CS-ACELP is much better than the other three coders. The CS-ACELP coder performed well both in a quiet environment and in a noisy environment at a bitrate of 6.4Kbps.
Low-Delay 16 kb/s Wideband Speech Coder with Fast Search Methods
In this paper, a low delay 16 kb/s wideband speech coder with a buffering delay of 1.25 ms is introduced. This coder is basically inspired from the G.728 LD-CELP standard for narrowband speech signals. Our main goal is to reduce the implementation complexity of our wideband G728 -like -coder, which is mainly due to the search of the optimal excitation (gain-shape) in codebook. For this reason an algebraic codebook is proposed and an exhaustive optimal search and suboptimal full position and Joint Position and Amplitude Search techniques are implemented. Objective and subjective measures on a large corpus of a testing speech database show that the JPAS (Joint Position and Amplitude Search) multistage search technique proposed recently can yield to a very important reduction of coder computational load, without decreasing the perceived quality of wideband speech signals compared to other search techniques.
Techniques for improving the performance of CELP type speech coders
1991
Techniques for improving the performance of CELP (code excited linear prediction) type speech coders while maintaining reasonable computational complexity are explored. A harmonic noise weighting function which enhances the perceptual quality of the processed speech is introduced. The combination of harmonic noise weighting and subsample resolution pitch significantly improves the coder performance for voiced speech. A 6.9 kb/s VSELP speech coder which incorporates subsample resolution pitch and harmonic noise weighting is described. Complexity reduction techniques are discussed which allow the coder to be implemented using a single fixed point digital signal processor
IEE Proceedings - Vision, Image, and Signal Processing, 2005
This paper presents several strategies to improve the performance of very low bit rate speech coders and describes a speech codec that incorporates these strategies and operates at an average bit rate of 1.2 kb/s. The encoding algorithm is based on several improvements in a mixed multiband excitation (MMBE) linear predictive coding (LPC) structure. A switched-predictive vector quantiser technique that outperforms previously reported schemes is adopted to encode the LSF parameters. Spectral and sound specific low rate models are used in order to achieve high quality speech at low rates. An MMBE approach with three sub-bands is employed to encode voiced frames, while fricatives and stops modelling and synthesis techniques are used for unvoiced frames. This strategy is shown to provide good quality synthesised speech, at a bit rate of only 0.4 kb/s for unvoiced frames. To reduce coding noise and improve decoded speech, spectral envelope restoration combined with noise reduction (SERNR) postfilter is used. The contributions of the techniques described in this paper are separately assessed and then combined in the design of a low bit rate codec that is evaluated against the North American Mixed Excitation Linear Prediction (MELP) coder. The performance assessment is carried out in terms of the spectral distortion of LSF quantisation, mean opinion score (MOS), A/B comparison tests and the ITU-T P.862 perceptual evaluation of speech quality (PESQ) standard. Assessment results show that the improved methods for LSF quantisation, sound specific modelling and synthesis and the new postfiltering approach can significantly outperform previously reported techniques. Further results also indicate that a system combining the proposed improvements and operating at 1.2 kb/s, is comparable (slightly outperforming) a MELP coder operating at 2.4 kb/s. For tandem connection situations, the proposed system is clearly superior to the MELP coder.
Linear Predictive Coding for Speech Compression
Telecommunication industry is growing and different services are rapidly introduced by different competitors to attract the users. Speech communication and its quality conservation is the most prevalent and common service provided by almost all companies. The objective of this project is the development of a LPC (Linear Predictive Coding) based voice coder. Attributes for speech like pitch, voiced and unvoiced decision and silence were extracted and speech was modeled using LDR (Levinson Durbin Recursion) and SDA (Steepest Descent Algorithm). LPC filter is analyzed and its model is implemented. LPC's different attributes complexity, delay and bitrate are deliberated and tradeoffs are highlighted. The results were analyzed and quality of speech was determined using spectrograph and by listening to the synthesized speech. At the end quality of original and synthesized speech is discussed and shown graphically and a soft comparison between both above mentioned technique is also added.
Design and description of CS-ACELP: a toll quality 8 kb/s speech coder
IEEE Transactions on Speech and Audio Processing, 1998
This paper describes the 8 kb/s speech coding algorithm G.729 which has been recently standardized by ITU-T. The algorithm is based on a conjugate-structure algebraic CELP (CS-ACELP) coding technique and uses 10 ms speech frames. The codec delivers toll-quality speech (equivalent to 32 kb/s ADPCM) for most operating conditions. This paper describes the coder structure in detail and discusses the reasons behind certain design choices. A 16-b fixed-point version has been developed as part of Recommendation G.729 and a summary of the subjective test results based on a real-time implementation of this version are presented.