Speech Coding Algorithms Research Articles

CODING AND ANALYSIS OF SPEECH IN COCHLEAR IMPLANT: A REVIEW

About 5% of the total population of world has hearing impairment and this is 6.3% in India.The hearing impairment in patients can be corrected by using a hearing aid or cochlear implant (CI) .Hearing aid is a traditional way while CI, which is surgically implanted, is a modern way to correct the hearing impairment. In comparison to hearing aids CI increases speech intelligibility at a greater level. Also speech intelligibility decreases in noise though acceptable in noise-free environments. With the advancement of technology over past decades, it shows increase in speech intelligibility with new emerging speech coding strategies. In this paper the basic speech processing strategies for speech coding and different techniques and algorithms, which aimed to increase the speech intelligibility, are being reviewed. Speech will be analyzed based on some measures like Analysis of Variation (ANOVA), Root Mean Square Error (RMSE) etc. The focus will be on speech perception and intelligibility enhancement.

An Algorithm for Simple Differential Speech Coding Based on Backward Adaptation Technique

This paper presents a simple differential speech signal coding algorithm, based on backward adaptation. The considered algorithm is executed in frame by frame manner, by implementing predictive and adaptive quantization techniques. Both prediction and adaptation are performed backward, based on the previously quantized input signal frame. This enables us to obtain high quality output signal, without increasing the bit rate. This research puts emphasis on the quantizer design, with the optimal support limit determination, and theoretical performance evaluation. Objective quality of the output signal is evaluated through signal to quantization noise ratio (SQNR). We perform theoretical and experimental analysis of the algorithm performance and provide comparative results of implementing speech signal coding techniques with similar complexity. Experimental results show that our simple differential speech coding algorithm satisfies the G.712 Recommendation for high-quality speech coding at the bit rate of 6 bits per sample. This indicates that the algorithm can be successfully implemented in high quality speech signal coding.

高能率符号励振線形予測符号化と帯域選択型変形離散余弦変換符号化を用いた32kbit/sスケーラブル広帯域音声・オーディオ符号化方式の開発

高能率符号励振線形予測符号化と帯域選択型変形離散余弦変換符号化を用いた32kbit/sスケーラブル広帯域音声・オーディオ符号化方式の開発

Research and Analysis on the Implementation of MELP Algorithm on DSP

This paper briefly analyses the principle of MELP vocoder algorithm, and TI's TMS320C5509(C5509) DSP is selected as an implementation platform of 2.4kbps MELP speech-coding algorithm. To ensure the rational and efficient utilization of the limited memory resources, this paper introduces the processing method of MELP algorithm based on frame structure and gives a profound analysis on how to configure its memory space of the selected DSP by analyzing its memory structure and considering the specific circumstances of the MELP vocoder algorithm. Finally, this paper gives the memory configuration during the implementation of MELP vocoder on TI's TMS320C5509 DSP.

A Low-Complexity 3.6kbps Speech Coding Algorithm Based on Mixed Excitation

A low-complexity 3.6kb/s speech coding algorithm based on mixed excitation is presented in this paper. It uses the parameter encoding and mixed excitation technology to ensure the quality of speech. Through adopting the scalar quantization of Line Spectrum Frequency (LSF), the algorithm reduces the storage and computational complexity. Meanwhile, improved frame type with dynamic Unvoiced/Voiced (U/V) thresholds make a reduction of the traditional U/V decision error and the sudden transformation of U/V frame. A modified bit allocation table is introduced and the PESQ-MOS test shows that the synthetic speech quality has been improved and reached the quality of communication, especially for high frequency female speakers with new frame type.

Speech Coding Algorithm with Dynamic Weighted Inter-Frame Linear Prediction

A new multi-frame joint quantization algorithm with dynamic weighted inter-frame linear prediction based on mixed excitation linear prediction (MELP) is proposed in this paper. In encoding stage, a super-frame consists of three adjacent single-frames. Fourier magnitudes and aperiodic jitter flag are eliminated. The other parameters are jointly quantized. LSF of the first and third frame are quantized as a 20-dimensional vector. According to the BPVs of super-frame, pitch is quantized with codebooks of dynamic size. In decoding stage, parameters are indexed from corresponding codebook. The LSF of middle frame are predicted from the first and third frame. The weighted factors keep changing in accordance with the BPVs of adjacent five frames. Results show that the reconstruction accuracy of LSF is significantly improved using dynamic weighted inter-frame linear prediction. Meanwhile the coding bit rate is reduced to 0.6 kbps.

Realtime Joint Speech Coding and Transmission Algorithm for High Packet Loss Rate Wireless Channels

In digital speech communication over noisy high packet loss rate wireless channels, improving the overall performance of the realtime speech coding and transmission system is of great importance. A novel joint speech coding and transmission algorithm is proposed by fully exploiting the correlation between speech coding, channel coding and the transmission process. The proposed algorithm requires no algorithm delay and less bandwidth expansion while greatly enhancing the error correcting performance and the reconstructed speech quality compared with conventional algorithms. Simulations show that the residual error rate is reduced by 84.36% and the MOS (Mean Opinion Score) is improved over 38.86%.

Lloyd–Max's Algorithm Implementation in Speech Coding Algorithm Based on Forward Adaptive Technique

In this paper a detail analysis of speech coding algorithm based on forward adaptive technique is carried out. We consider an algorithm that works on frame-by-frame basis, where a frame consists of a certain number of speech samples. Buffering frame-by-frame an estimation of the gain defined as squared root of the frame variance is enabled. The information about the gain (side information) and the code book of a nonadaptive quantizer, which is designed for the unit variance case of the input signal, are further used when designing an adaptive quantizer. In such a way better quantizer adaptation to the varying input statistics is provided. Observe that the goal of this paper is to investigate the preference that for the wide range of variance change could be achieved when implementing in the forward adaptive speech coding algorithm, the recently developed effective method for the Lloyd-Max's algorithm initialization, which provides optimal Lloyd-Max's quantizer performances for the unit variance case of the input signal. We destine to consider the speech coding algorithm based on forward adaptive technique since the backward adaptation provides SQNR (signal to quantization noise ratio) within 1 dB of the forward adaptation. We provide theoretical and experimental results (performances of our algorithm) which are compared with the optimal results. Additionally, we discuss the performances of speech coding schemes designed according to G. 711 standard and we point out the benefits that can be achieved by using our algorithm. Finally, in order to find better solution for implementation of the proposed algorithm in practice we consider the performances of our algorithm when log-uniform as well as uniform scalar quantizer are used for gain quantizing.

Bandwidth Extension for Hierarchical Speech and Audio Coding in ITU-T Rec. G.729.1

Recommendation G.729.1 is a new ITU-T standard which was approved in May 2006. This recommendation describes a hierarchical speech and audio coding algorithm built on top of a narrowband core codec. One challenge in the codec design is the generation of a wideband signal with a very limited additional bit rate (less than 2 kb/s). In this paper, we describe the respective codec layer, which extends the transmitted acoustic bandwidth from the narrowband frequency range (50 Hz-4 kHz) to the wideband frequency range (50 Hz-7 kHz). The underlying algorithm uses a fairly coarse parametric description of the temporal and spectral energy envelopes of the high frequency band (4-7 kHz). This parameter set is quantized with a bit rate of 1.65 kb/s. At the decoder side, the high-frequency components are regenerated by appropriately shaping a synthetically generated ldquoexcitation signal.rdquo Apart from the algorithmic description and a discussion, we state a complexity evaluation as well as some listening test results.

Intelligibility evaluation of GSM coder for Mandarin speech using CDRT

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.

A combined wideband speech and audio coder using human articulatory and auditory models

The modern high-quality speech and audio coding algorithms are mostly based on human articulatory models. On the other hand, high-quality audio coders are mostly based on models of human auditory system. However, neither of these two methods can provide acceptable performance for both music and speech. In this paper a coder is proposed which operates for both wideband speech and audio signal. The structure of the coder is based on both human articulatory and auditory models. In the first part the LPC parameters are drawn using the conventional autocorrelation method in order to simulate the human articulatory system. With LPC parameters the redundancy of input signal in time domain is removed. In the second part of the coder, the LPC residual signal is further transformed into frequency domain through MLT (modulated lapped transform). The MLT coefficients are then quantized in frequency domain using the human auditory properties such as masking property. The informal listening test has shown that the proposed coder provides very good quality for wideband speech and for most music signals.

Objective speech quality assessment and the RPE-LTP coding algorithm in different noise and language conditions

The formulation of reliable signal processing algorithms for speech coding and synthesis require the selection of a prior criterion of performance. Though coding efficiency (bits/second) or computational requirements can be used, a final performance measure must always include speech quality. In this paper, three objective speech quality measures are considered with respect to quality assessment for American English, noisy American English, and noise-free versions of seven languages. The purpose is to determine whether objective quality measures can be used to quantify changes in quality for a given voice coding method, with a known subjective performance level, as background noise or language conditions are changed. The speech coding algorithm chosen is regular-pulse excitation with long-term prediction (RPE-LTP), which has been chosen as the standard voice compression algorithm for the European Digital Mobile Radio system. Three areas are considered for objective quality assessment which include: (i) vocoder performance for American English in a noise-free environment, (ii) speech quality variation for three additive background noise sources, and (iii) noise-free performance for seven languages which include English, Japanese, Finnish, German, Hindi, Spanish, and French. It is suggested that although existing objective quality measures will never replace subjective testing, they can be a useful means of assessing changes in performance, identifying areas for improvement in algorithm design, and augmenting subjective quality tests for voice coding/compression algorithms in noise-free, noisy, and/or non-English applications.

An Asymmetric Speech Coding Algorithm Using Vector Quantization and Silence Deletion

AbstractIn store‐and‐forward and multi‐media speech applications, e.g. in voice‐mail or when adding verbal comments to a document, the speech encoder can be allowed more time and higher complexity than in real‐time speech coding. Decoding however must be simple when a low‐cost terminal (e.g., a PC) is used. This kind of coding algorithm is called an asymmetric algorithm. Asymmetric algorithms exploit encoding delay to code the signal efficiently. An existing asymmetric coding algorithm is Vector Quantization (VQ). A new asymmetric algorithm is proposed, based on the coding of the time intervals between zero‐crossings (frames) using VQ and silence deletion. The algorithm takes advantage of the empirical fact that infinitely clipped speech retains its intelligibility. The algorithm has been implemented using software and its performance was evaluated.

Efficient channel coding for CELP using source information

Since simultaneous optimization of source and channel coding is often not practical for speech-coding algorithms, it is useful to develop channel codes optimal for a particular speech coding algorithm operating under a specified range of channel-error conditions. Such source-dependent channel-error codes can be obtained by minimizing an appropriate speech distortion criterion. To carry out this minimization, we use a simulated-annealing procedure. The resulting channel codes are efficient, since they provide error correction of non-uniform accuracy (highly probable quantization levels receive more accurate correction) and/or non-uniform error detection (errors which greatly impact the speech quality are more likely to be detected). An optimal trade-off between error correction and error detection can be obtained. Source-dependent channel codes aimed at counteracting low, random error rates (up to 2%) are applied to the CELP algorithm and the resulting performance is reported. It is found that a small allocation of codewords (equivalent to less than one bit) for the protection of a particular parameter often results in a large performance improvement.