Statistical Block Protection Coding for DPCM-AQF Speech

The statistical block protection coding scheme [2] for protecting DPCM encoded speech signals through noisy channels has been extended to accommodate DPCM-AQF encoded speech, where AQF stands for adaptive quantization with forward (explicit) transmission of step size. Signal-to-noise ratio (SNR) gains, typically 12 dB, have been achieved over a dynamic range> 20dB for a bit error rate (BER) of 1.4 percent and the SNR improvement is found to increase with BER. Perceptual improvements in decoded speech have a good correspondence with the gains in SNR. The penalty for this substantial enhancement in the performance of the DPCM-AQF system is an increase in transmission bit rate of 3.5 percent and encoding delay of 64 ms.     

基于线性预测系数自适应前后向量化的可变速率MELP语音编码

该文基于LPC的自适应前后向量化技术,提出了一种可变速率的混合激励线性预测MELP语音编码算法。该算法中,采用当前语音帧(前向LPC)或前面某帧已合成语音帧(后向LPC)进行线性预测,当采用后向LPC时,只需传输时间序列编码,故减少了LPC系数的平均编码比特。计算机模拟表明,该算法与标准MELP算法合成的语音质量相当,但显著减少了LPC的传输带宽,从而明显降低了MELP平均编码速率。     

Adaptive nonuniform polar quantization application to high quality speech compression

The logarithmic companding technique has shown to be extremely useful in speech quantization with rate of 8 bits/sample. However, for lower bit rates it is not the ideal solution for high quality speech coding. Because of that, in this paper we establish source coding scheme which enables better spectrum efficiency for input that has a large dynamic range. Since our wish is also to improve signal quality in comparison with quality defined with standards G.711 and G.712, we opt for adaptive technique application to the speech coding. Our research shows that proper design of forward gain-adaptive polar quantization can enable compression of about 1 bit/sample as well as significantly better quality than in case of using coder designed according to standard G.711. Furthermore, performances can be sustained over the whole speech dynamic range. Also, if the requisite speech quality is not supposed to be lower than G.712 standard quality, the achieved compression can be almost 1.5 bits/sample. Besides, we propose knew simple encoding rule which can additionally reduce bit rate for 0.1 bit/sample.     

Sub-band coding with adaptive bit allocation

The paper presents an efficient method for speech encoding which is based on the well known idea of sub-band coding. Typically, the frequency range from 0.3 kc/s to 3.4 kc/s is split into four sub-bands, and the sub-band signals are encoded separately with different accuracies by means of familiar PCM techniques. An adaptive bit allocation scheme is introduced here, in order to replace the usual form of a fixed distribution of the bit rate among the sub-bands. Listening tests have shown that by these means the bit rate can be reduced by more than 2.5 kb/s without degrading speech quality. Accordingly, highly intelligible reproduction of speech is possible at bit rates below 7 kb/s.     

Combining speech coders and entropy coders to reduce the bit rate in the compression of speech

The combination of speech coders and entropy coders is investigated, for bit rate reduction. Three speech coders of the celp (code excited linear prediction) type are considered and the residual correlation in lsp (line spectrum pairs) coefficients and gains in a speech frame is exploited. The lossless entropy coders use Huffman, Lzw (lempel ziv welch) and gzip (LZ-Huffrnan) techniques. The greatest efficiency is provided by the adaptive Huffman approach, with a 15 % gain in each type of compressed parameter and an overall average bit rate reduction of 7 % for the FS1016 coder and 5 % for the Tetra and lbc coders.     

Sample Robbing in Predictive Speech Coders

In this paper, speech bit rate reduction by not transmitting a percentage of samples (i.e., robbing the coder of some samples) has been studied. The technique has been applied to predictive coders, namely differential PCM (DPCM) and adaptive DPCM (ADPCM) coders. A robbed sample is replaced by its estimate so that the prediction process in the feedback loop of the coders continues in a normal manner. After one period delay, when the next sample is decoded, the robbed sample is reestimated using delayed interpolation. Only periodic sample robbing has been considered, such as every fourth, every third, etc. The technique is particularly useful where graceful degradation is required under heavy loading conditions. The technique is found to be useful when the desired bit rate is 24 kbits/s or lower. The technique was evaluated by computer simulation using real-time speech inputs. Improvements of up to 3 dB in the case of a DPCM coder and of up to 1.5 dB in the case of an ADPCM coder have been achieved.     

Capacity enhancement of cellular CDMA by traffic-based control ofspeech 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/∞//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     

Cyclic prefixed single carrier and multicarrier transmission: bit rate comparison

We consider cyclic prefixed single carrier and adaptive multicarrier transmission over a frequency selective channel. We compare the achievable bit rate for a target bit error rate. We analytically prove that the bit rate achieved with multicarrier transmission with adaptive modulation is always higher than that obtained with single carrier transmission when a one-tap frequency domain equalizer is used. We also show that the same adaptively loaded multicarrier scheme reaches the performance of single carrier transmission with a block decision-feedback equalizer.     

Design and simulation of an efficient adaptive delta modulationembedded coder

An embedded coding version of hybrid companding delta modulation (HCDM) is described that operates from 16 to 48 kb/s in 8 kb/s steps. The embedded HCDM coder employs the explicit noise coding technique to transmit an adaptive PCM (APCM) coded version of the HCDM reconstruction error signal as a supplementary bit stream that may be partly or wholly deleted in transmission. SNR performance with speech input depends critically on the design of the supplemental APCM code and two new coding algorithms are investigated. In algorithm 1, the basic cue for step size adaptation is obtained from the RMS slope energy of the HCDM output whereas in algorithm 2, the HCDM reconstruction error is logarithmically compressed before quantisation and the basic step size is derived from peak input magnitudes. Instantaneous adaptation for both algorithms is achieved by using step size multipliers which are optimised for operation at single fixed bit rates and also for decoding with an unknown number of input bit deletions. Simulation results show that SNR performance is significantly enhanced using either algorithm and a graceful reduction of reconstructed speech quality with progressive bit deletion is achieved over the range from 48 kb/s to 16 kb/s. On the whole, the SNR performance of the embedded HCDM system is superior in comparison with conventional HCDM     

Linear and adaptive delta modulation

New results are presented, offering insight into the performance and optimization of linear and adaptive delta modulation, together with a comparison with pulse code modulation. The results are applied to three cases of practical interest: television, speech, and broadband signals. The results are presented as follows: first, a characterization of the quantization noise of linear delta modulation (DM) is given; second, an adaptive DM system which seems promising for television and speech is evaluated; and third, a comparison between PCM and adaptive DM is made for speech, television, and broadband signals. It is concluded that 1) the adaptive system provides DM with a companding capability, 2) adaptive DM offers a bit rate or channel bandwidth reduction capability in comparison with PCM for television signals, 3) adaptive DM appears better suited to television and speech signals than linear DM, 4) the maximum S/N performance of adaptive DM is the same as that of linear DM, 5) the companding improvement offered by adaptive DM is not limited by the same practical considerations as those of PCM, and 6) the S/N performance of adaptive DM is the same for both Gaussian and exponential signal densities.     

极低速率语音编码的新发展与应用

从时域、频域和混合域3方面分别介绍了目前在极低速率语音编码中应用的LPC,CELP,MBE,STC,VQ,MELP,WI,分段声码器和基于语料库的声码器等算法。全面论述了极低速率语音编码算法中运用的方法理论及最新的研究成果,并对这些算法进行了比较和分析,得出结论。     

Very low bit rate speech coding using a diphone-based recognitionand synthesis approach

High compression rates of speech signals may be achieved by coding schemes based on relevant linguistic segments. A system is described that relies on a diphone recogniser as the coder and on a speech synthesiser reproducing speech starting from a diphone codebook as the decoder. The spoken message is encoded in textual (phoneme labels) plus prosody representation. This speech coding technique may be used for voice mail or phone communication over low bit rate channels     

Multiuser rate subband coding incorporating DSI and buffer control

A model of multiuser variable rate subband coding of speech incorporating digital speech interpolation (DSI) is presented and analyzed. A dynamic bit assignment algorithm which minimizes the mean squared reconstruction error is presented. A backward estimation of subband variances which makes the bit assignment algorithm backward-adaptive and greatly reduces the need to send side information is proposed. A novel switched-stepsize adaptation algorithm is used at the coders to improve system performance. With adaptive buffer control, buffer overflow is completely prevented. It has been shown by simulation means that a 2:1 DSI gain can be obtained with a relatively small buffer size     

Entropy Measurements for Three Adaptive Source Encoders

The entropy of the output of three adaptive source encoders, AΔM, APCM, and ADPCM, is measured for speech inputs at low sampling rates. It is attempted to correlate the source encoder output entropies, normalized to the channel bit rate, with the results of informal listening tests. It was hypothesized that the channel bit rate utilization factor would predict subjective ranking of the source encoders. The results of measurements do not support this hypothesis; an intuitive explanation is offered.     

Bit Rate Reduction of Digitized Speech Using Entropy Techniques

We present the results of a study to reduce the bit rate of speech that has been digitized with a continuously variable slope delta modulator (CVSD) operating at 16, 24, and 32 kbits/s. The theoretical reduction is found from the bit stream entropy. The actual reduction, via Huffman coding, is within 1-2 Percent of the theoretical value. The conditional entropy indicates that additional bit rate reduction can be achieved if we use a set of Huffman codes, conditioned on the past CVSD bits. A third technique, tandem coding, using a maximum likelihood predictor in tandem with run length and Huffman coding, is also investigated. Using these entropy techniques, bit rate reductions of 11-25 percent are achieved for the CVSD rates considered. The paper concludes with a study of the buffer requirements needed to support these entropy coders.     

Performance comparison between VBR speech coders for adaptive VoIPapplications

The introduction of new variable bit rate (VBR) speech coders has opened up new perspectives for the implementation of adaptive voice over IP (AVoIP) systems. The paper compares different VBR speech coding techniques in a scenario in which the rate of the single sources is dynamically adapted to the workload conditions. The coders compared are the AMR, the M³ R and the G.729. Using source and rate control mechanism models, performance is evaluated in terms of loss probability, offered throughput and mean CMOS with varying numbers of sources. The use of header compression mechanisms is also evaluated     

Fast vector-sum codebook search method for low bit rate speechcoding

A fast vector-sum codebook search method for low bit rate speech coding is presented. In this method, the codebook search is simplified by designing a vector-sum codebook that consists of orthonormal regular pulse basis vectors. A further simplification is achieved by adopting backward filtering. The method proposed has significantly reduced computational complexity, compared with the conventional VSELP, without producing any additional degradation in the quality of the synthesised speech     

声码器通用硬件平台的实现

基于低速率语音编解码算法实时实现的需求和IP电话终端以及小型IP电话网关的需求,设计开发了通用语音处理平台,实时实现了320b／s高质量极低速率语音编解码系统,通过测试得到了和定点优化后的语音算法完全相同的结果。     

Code-aided adaptive equaliser for mobile communication systems

A new structure for a code-aided adaptive equaliser is proposed for mobile communication systems. The structure aims to improve the tracking capability of a receiver that combines equalisation and coding. Simulation results show that the code-aided adaptive equaliser performs particularly well over mobile channels that fade at a moderate rate. Compared to a conventional equaliser, an order of magnitude reduction in bit error rate is achieved     

On adaptive vector transform quantization for speech coding

Adaptive vector transform quantization (AVTQ) as a coding system is discussed. The optimal bit assignment is derived based on vector quantization asymptotic theory for different PDFs (probability density functions) of the transform coefficients. Strategies for shaping the quantization noise spectrum and for adapting the bit assignment to the changes in the speech statistics are discussed. A good estimate of the efficiency of any coding system is given by the system coding gain over scalar PCM (pulse code modulation). Based on the optimal bit allocation, the coding gain of the vector transform quantization (VTQ) system operating on a stationary input signal is derived. The VTQ coding gain demonstrates a significant advantage of vector quantization over scalar quantization within the framework of transform coding. System simulation results are presented for a first-order Gauss-Markov process and for typical speech waveforms. The results of fixed and adaptive systems are compared for speech input. Also, the AVTQ results are compared to known scalar speech coding systems