Rate Speech Coding Research Articles

Quad-band excitation for low bit rate speech coding.

The excitation parameters of a multiband excitation model for speech production consist of a pitch period and a series of voiced/unvoiced (v/uv) decisions for a number of fixed frequency bands around the harmonics of the fundamental. In this paper, it is shown that, at most, four v/uv variable-length nonoverlapping frequency bands spanning over the telephone bandwidth are adequate to represent the excitation spectrum that provides an improved modeling of mixed voicing and noisy speech. A quad-band excitation (QBE) method is presented which generates excitation source for a linear prediction vocoder for low bit rate transmission of speech signal and the DRT scores validate the potential of the method.

Capacity enhancement of cellular CDMA by traffic-based control of speech bit rate

Variable rate speech coding is now recognized as an important system component for high-capacity cellular networks because it exploits speech statistics to reduce the average bit rate, which results in reduced interference and increased capacity. Once a variable rate capability is available, an additional capacity enhancement can be achieved by introducing network control of the user bit rate in response to changing traffic levels. We introduce the concept of network control of rate and propose a particular network-control method for code-division multiple access (CDMA) systems. Based on an M/M//spl infin///M queueing model applied to a cell under heavy traffic conditions and a new performance measure called averaged speech quality, we obtain simulation results to demonstrate how network control of rate can achieve improved speech quality or increased capacity for a given quality objective.

Trellis decoding of Reed-Solomon codesin low bit rate speech communications

The residual redundancy that persists in the transmitted parameters of a low bit rate speech coder are exploited to reduce the computational complexity of a Reed-Solomon (RS) trellis decoder. The use of RS codes for mobile telephony provides the opportunity for avoiding channel interleaving and so reduces one-way delay.

A mixed excitation LPC vocoder model for low bit rate speech coding

Traditional pitch-excited linear predictive coding (LPC) vocoders use a fully parametric model to efficiently encode the important information in human speech. These vocoders can produce intelligible speech at low data rates (800-2400 b/s), but they often sound synthetic and generate annoying artifacts such as buzzes, thumps, and tonal noises. These problems increase dramatically if acoustic background noise is present at the speech input. This paper presents a new mixed excitation LPC vocoder model that preserves the low bit rate of a fully parametric model but adds more free parameters to the excitation signal so that the synthesizer can mimic more characteristics of natural human speech. The new model also eliminates the traditional requirement for a binary voicing decision so that the vocoder performs well even in the presence of acoustic background noise. A 2400-b/s LPC vocoder based on this model has been developed and implemented in simulations and in a real-time system. Formal subjective testing of this coder confirms that it produces natural sounding speech even in a difficult noise environment. In fact, diagnostic acceptability measure (DAM) test scores show that the performance of the 2400-b/s mixed excitation LPC vocoder is close to that of the government standard 4800-b/s CELP coder.< <ETX xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">&gt;</ETX>

Achieving high quality speech at low bit rates: Fundamental issues in psychophysics

Excellent progress has been made in recent years in synthesizing high quality natural sounding speech at bit rates as low as 8 kb/s. Rapidly increasing demand for speech transmission over digital radio channels requires that the bit rates be decreased even further—to 4 kb/s or lower. But, one is not able to achieve high quality speech at these low bit rates. Why? Low bit rate speech coders exploit auditory masking to mask the quantizing noise in the coded signal. The current models of auditory masking are valid only for relatively stationary segments, such as vowels, in the speech signal. Much of the distortions we heard in low bit rate coders are introduced during transitional segments, resulting from lack of psychophysical data on perception of such distortions. A human listener is able to detect such distortions in the coded version without comparing the coded version with the original. Why is it so? How does our hearing system know that the coded signal is distorted? Such questions must be answered before progress can be made in achieving high quality speech at low bit rates.

A variable rate multipulse speech coder for CDMA cellular systems

Variable Rate (VR) speech coders are classified into: source-controlled VR coders where the rate is selected depending on the local character of the speech, and network-controlled VR coders where an external control signal selects the coding rate. The first category benefits from the variable rate channels used by Code Division Multiple Access (CDMA) mobile communications. The second category is indispensable for the right behaviour of the CDMA systems under conditions as high traffic levels. The VR speech coder presented in this communication exhibits both types of control. The source control is achieved by means of a Voice Activity Detector (VAD) and a phonetic classifier. The network control acts on the selection procedure of the multipulse excitation sequence to the synthesis filter. This is the main advantage of our VR MultiPulse speech coder because by means of an external signal the bit rate can be changed only every 4 msec, without transitions or distortions. Considering one-way communication, six different operating rates can be externally selected ranging from 4.8 to 9.1 kbps for the active frames; an average bit rate of 380 bps is required for the noise frames.

Split-dimension vector quantization of Parcor coefficients for low bit rate speech coding

A novel split vector quantization (SVQ) scheme for low bit rate coding of speech signals is proposed. In this scheme, the LPC parameter vector, which is represented by Parcor coefficients, is split into small-dimension subvectors, and each subvector is sequentially quantized according to a multistage structure that resembles a segmented lattice filter. The forward and backward prediction residuals in the segmented filter are coupled across VQ stages. The quantizer in each stage operates on the principle of minimizing the forward and backward prediction error energies similar to linear predictive analysis. Simulation results show that the new split VQ scheme can achieve transparent quantization of LPC parameters at 25 b/frame.< <ETX xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">&gt;</ETX>

Pitch-synchronous frame-by-frame and segment-based articulatory analysis by synthesis

This paper presents a pitch-synchronous analysis-by-synthesis procedure for estimating model parameters for voiced speech. These model parameters describe the vocal-tract shape and the time derivative of the glottal area function. The excitation waveform is derived from the glottal area function by incorporating source-tract interaction using the current vocal-tract input impedance. The corresponding analysis procedure for estimating the model parameters once every pitch period is outlined. A significant improvement in quality was obtained for the new pitch-synchronous analysis/synthesis procedure relative to the fixed-frame-length-based scheme used previously. It was also found that the new pitch-synchronous articulatory analysis/synthesis scheme achieves lower rms spectral distortion values than the 2.4 kb/s. Federal standard LPC-10E algorithm. A segment-based procedure for estimating the vocal-tract model parameters at a rate much lower than the current pitch is described. In this segment-based analysis-by-synthesis approach, the model parameters are estimated every 50–100 ms. The parameters for the intermediate pitch periods are derived by interpolation. The segments are selected using a maximum likelihood segmentation algorithm that segments an utterance into diphonelike units. A segment-based parameter optimization scheme could lead to a highly economical representation of the speech signal for potential applications in very low bit rate speech coding and speech storage. The above schemes were optimized for a pilot test sentence and then evaluated using eight test sentences for a log area and the Coker articulatory model representation of the vocal tract. Nine listeners were asked to judge the quality of the synthesis in a paired-comparison test and the results were analyzed using a nonparametric one-tailed sign test. For the log-area representation of the vocal tract, we found a significant degradation in speech quality for the segment-based optimization procedure relative to the frame-based procedure. However, for the Coker model representation, the degradation was found to be insignificant. This shows that unlike cross-sectional areas, the movement of various articulators in the vocal tract during speech production can be described with sufficient accuracy by specifying the position of these articulators and by using an interpolation function at time intervals much longer than a pitch period.

CELP based mixed-source model for very low bit rate speech coding

A CELP based mixed-source model is described. It uses a mixed excitation which combines a lowpass-filtered adaptive source and a highpass-filtered stochastic source. In addition, one more stochastic source is newly employed for more natural sounding speech. In informal listening tests, the proposed model at 3 kbit/s shows very good performance both in speech quality and intelligibility.

New multistage scheme for vector quantisation of parcor coefficients

A new multistage scheme for vector quantisation of PARCOR coefficients for low bit rate speech coding is proposed. The novelty of this scheme is that the lattice filter structure is preserved after the quantisation. This is achieved by coupling the forward and backward prediction errors across vector quantisation (VQ) stages. Simulation results show that the new scheme can achieve lower average distortion than other multistage VQ schemes.

Finite state CELP for variable rate speech coding

The performance of a variable rate code excited linear predictor system is investigated. The coding system is based on a finite state CELP (FSCELP) frame work. Each individual state is primarily identified with a LPC model order, LPC coefficients bit allocation, excitation code book population density and state encoding rate. Successive input speech vectors are encoded at a rate that depends on the current state of the FSCELP system and the input vector characteristics. The use of a finite state system involves implicit clustering of speech signals. The lower rate states are selected during highly correlated steady state speech segments when relatively few bits are required to obtain adequate fidelity. For speech signals with a strong glottal excitation, unvoiced signals and transient speech segments, a relatively greater quantisation accuracy is needed to obtain good fidelity and therefore higher rate states of the system are used. Further improvement is obtained by using gamma populated excitation codebooks, for those states that are mainly used to encode speech signals with a strong underlying glottal excitation pulses. Experiments focus on investigation of the varying encoding requirements of the excitation signal for low pass, voiced, unvoiced and transient speech signals. The parameters of the finite state CELP system are designed to match the encoding requirements of typical speech signals. The greater part of the coding gain is obtained from variable rate encoding of the excitation signal. Using a six-state FSCELP, good quality speech is obtained at an average, maximum and minimum bit rates of 4 kbit/s, 10 kbit/s and 2 kbit/s, respectively.

Real-time speech segmentation using pitch and convexity jump models: application to variable rate speech coding

Convexity jump models are combined with period identification in the time domain to provide an efficient segmentation technique adapted to the needs of coders. The algorithm presented works online, with no training, and is speaker-independent. Structural problems found in previous systems (blank detection spots, long decision delays) have been overcome. Both the simplicity and the generality (information theory basis for the divergence test) allow applications of the automatic online segmentation method in other areas of signal processing. >

Performance of packet reservation multiple access for digital mobile radio using variable rate speech coder

The advantages of using adaptive bit rate voice coders to improve the performance of PRMA for digital cellular radio are considered. It is shown that such a technique forces a fairer distribution of packet loss between mobiles and is able to accommodate fluctuations in traffic levels within a cell.

On a Real Time Implementation of LPC Speech Coder on a Bit-Slice Microprocessor Based Digital Signal Processor

Low bit rate (2.4 kb/s) speech coders are very important for secure speech transmission over ordinary telephone lines and for compact storage applications. In this connection, Linear Predictive Coding (LPC) technique has attracted particular attention because of its efficacy and amenability to DSP approach in implementation.This paper presents a real time implementation of a LPC system on a dedicated bit slice microprocessor based signal processor. It lays stress on the microprogrammed software developed for the purpose. A discussion of the results of the subjective evaluation of the LPC coder is included.

A research trend of very low bit rate speech coding. (&amp;ltspecial isssue&amp;gtRecent trend of speech information processing.)

A research trend of very low bit rate speech coding. (&amp;ltspecial isssue&amp;gtRecent trend of speech information processing.)

Dynamic data compression algorithms in speech coding

Speech coding is to use a series of bits to encode speech signals, such that it is much easier and more efficient to digitally transmit or store speech signals. Bit rate is the number of bits required for every second of speech signal. Based upon parametric descriptions of speech signals, the linear prediction analysis provides a successful method for low bit rate speech coding. Due to the high redundancy among the parameters of adjacent speech signal frames, further data compression becomes possible. In this paper, various data compression techniques and algorithms are studied and two sets of variable frame rate linear predictive vocoders are developed based on frame interpolation and frame repeating concepts, respectively. Simulation results indicate that almost identical speech quality can be achieved even when the bit rate is reduced from 2,400 bps to 1,200 bps.

Speech transmission capacity of standard ethernet systems

This paper is concerned with the estimation of the speech traffic capacity of a standard ethernet system by simulation techniques. The particular interest in this work is the possible trade-off of speech coding rate against number of simultaneous users during peak traffic conditions. The results of simulation studies are given and it is concluded that the decisive factor in such a trade-off is the maximum transmission delay which may be accepted and also the percentage loss of transmitted information which occurs as a result.