Digital Speech Communication Research Articles

Method for testing the stability of an autoregressive model of the vocal tract and adjusting its parameters

Within the framework of the traditional direction of research in the field of acoustic measurements, an autoregressive model of the vocal tract as a key link in the human speech apparatus is considered. The acute problem of ensuring the stability of the autoregressive model in systems with adaptation of its parameters to the observed speech signal of short duration is pointed out. To overcome this problem, the task was set of testing the stability of the autoregressive model and adjusting its parameters based on the results of this testing. The study is based on the author’s method of formant analysis of vowel sounds of speech through the synthesis of a recursive shaping filter in the free oscillation mode. To solve sated task, a method is proposed for testing the stability and adjusting the parameters of the autoregressive model of the vocal tract based on a two-stage algorithm for its transformation. At the first stage of transformation, the stability of the autoregressive model is tested using the impulse response of the shaping filter. At the second stage, if the stability of the autoregressive model is violated, its impulse response is modified by element-by-element multiplication by a variable exponential value that asymptotically converges to zero. A regular algorithm has been developed for recalculating the modified impulse response into an adjusted vector of autoregressive parameters at the second stage of transformation. Based on the results of experimental testing of the proposed method, it was concluded that guaranteed stability of the autoregressive model of the vocal tract has been achieved with minimal distortion in the frequency domain. The results obtained are useful in the development and modernization of automatic speech recognition systems, digital speech communications, artificial intelligence and other information systems that use data compression and speech coding based on an autoregressive model of the vocal tract in automatic speech signal processing.

Method for asynchronous analysis of a glottal source based on a two-level autoregressive model of the speech signal

The task of analyzing a glottal source over a short observation interval is considered. The acute problem of insufficient performance of known methods for analyzing a glottal source is pointed out, regardless of the mode of data preparation: synchronous with the main tone of speech sounds or asynchronous. A method for analyzing the glottal source based on a two-level autoregressive model of the speech signal is proposed. Its software implementation based on the high-speed Burg-Levinson computational procedure is described. It does not require synchronization of the sequence of observations used with the main tone of the speech signal and is characterized by a relatively small amount of computational costs. Using the described software implementation, a full-scale experiment was set up and conducted, where the vowel sounds of the control speaker’s speech were used as the object of study. Based on the results of the experiment, the increased performance of the proposed method was confirmed and its requirements for the duration of the speech signal during voice analysis in real time were formulated. It is shown that the optimal duration is in the range from 32 to 128 ms. The results obtained can be used in the development and research of digital speech communication systems, voice control, biometrics, biomedicine and other speech systems where the voice characteristics of the speaker’s speech are of paramount importance.

Efficient audio cryptosystem based on chaotic maps and double random phase encoding

This paper presents an efficient audio cryptosystem based on combining chaotic maps with optical encryption. The proposed audio cryptosystem is designed with the potential of enforcing the security level for digital speech communication via shared networks. The proposed audio cryptosystem is built using two security phases. In the first phase, it utilizes a chaotic system using either Baker map or cat map for providing the first security level. In the second phase, it utilizes optical encryption using double random phase encoding (DRPE) for providing the second security level. This second security level represented with DRPE is a physical security that is very hard to attack. The proposed audio cryptosystem is implemented and its performance is evaluated using different audio encryption/decryption quality metrics. The results demonstrated that the proposed audio cryptosystem increases the level of voice security with high degree of confidentiality.

The Design of Adaptive Noise Cancellation Filter Based on DSP Chip

The speech signal usually could not be extracted correctly from the digital speech communication system with strong interference. As for this kind of system, the common fixed coefficient digital filters (FIR, IIR) are unable to achieve the best effect of filtering. Whereas the adaptive filter could extract the available signals properly by adjusting the filter coefficient automatically without knowing the change characteristics of the noise signal. In this paper, we designed an adaptive noise cancellation filter based on LMS algorithm on the DSP chip and verification of the filter was done on the TMS320C5509 platform. The results show that the adaptive noise cancellation designed in this paper could extract the available signals properly and improve the quality of the speech communication.

Tonal Language Speech Compression Based on a Bitrate Scalable Multi-Pulse Based Code Excited Linear Prediction Coder

Problem statement: Speech compression is an important issue in the modern digital speech communication. The functionality of bitrates scalability also plays significant role, since the capacity of communication system varies all the time. When considering tonal speech, such as Thai, tone plays important role on the naturalness and the intelligibility of the speech, it must be treated appropriately. Therefore these issues are taken into account in this study. Approach: This study proposes a modification of flexible Multi-Pulse based Code Excited Linear Predictive (MP-CELP) coder with bitrates scalabilities for tonal language speech in the multimedia applications. The coder consists of a core coder and bitrates scalable tools. The high pitch delay resolutions are applied to the adaptive codebook of core coder for tonal language speech quality improvement. The bitrates scalable tool employs multi-stage excitation coding based on an embedded-coding approach. The multi-pulse excitation codebook at each stage is adaptively produced depending on the selected excitation signal at the previous stage. Results: The experimental results show that the speech quality of the proposed coder is improved above the speech quality of the conventional coder without pitch-resolution adaptation. Conclusion: From the study, the proposed approach is able to improve the speech compression quality for tonal language and the functionality of bitrates scalability is also developed.

A Study on Digital Language Communication and Speech Communication

A Study on Digital Language Communication and Speech Communication

Wireless chaotic speech communication via digital signal processor ——system design and hardware implementation

提出了一种基于数字信号处理器（DSP）的语音无线混沌数字通信系统设计与硬件实现的新方案. 根据Runge-Kutta算法和变量比例扩张变换，以多涡卷广义Jerk系统为例，对其连续混沌系统作离散化处理，产生作为语音数据加密与解密的混沌数字序列. 在芯片型号TMS320VC5509APGE的DSP技术平台上，利用无线发送器和无线接收器nRF2401，实现了语音无线混沌数字通信. 给出了技术设计与硬件实现结果，证实了该方案的可行性.

Threshold Testing: Improving Yield for Nanoscale VLSI

Yields for digital very-large-scale-integration chips have been declining in the recent years, and the decline is accelerating as the technology moves deep into nanoscale. Recently, we have proposed the notion of error tolerance to improve yields for a wide range of high-performance digital applications, including audio, speech, video, graphics, visualization, games, and wireless communication. Error tolerance systematically codifies the fact that chips used in such applications can be acceptable despite having defects that produce erroneous outputs, provided that the errors are guaranteed to be of certain types and have severities within thresholds specified by the application. In this paper, we propose a new testing approach called threshold testing to practically exploit the notion of error tolerance for applications where errors with absolute numerical magnitudes lower than an application-specified threshold are acceptable. We propose a new automatic test pattern generator (ATPG) for threshold testing for single stuck-at faults. This test generator embodies several completely new techniques, including new approaches for directing the search for a test vector, new types of objectives, new types of necessary conditions, and new approaches to identify and exploit these conditions. We demonstrate that threshold testing can enhance yield and that it is practical in terms of test generation effort and test application costs. We also propose threshold fault simulators and ATPG for bridging and transition delay faults. We use these tools to show that the stuck-at-fault model is indeed a suitable model for threshold testing. This opens the way for developing low-cost tools for threshold testing that will provide high threshold coverage for realistic faults and defects and hence help provide higher yields in future nanoscale processes at low costs.

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%.

A Technique of Lossless Steganography for G.711

This study proposes a technique of lossless steganography for G.711, the most common codec for digital speech communications systems such as VoIP. The proposed technique exploits the characteristics of G.711 for embedding steganogram information without degradation. This paper shows the capacity of the proposed technique.

A Band Extension Technique for G.711 Speech Using Steganography

This study investigates a band extension technique for speech data encoded with G.711, the most common codec for digital speech communications system such as VoIP. The proposed technique employs steganography for the transmission of the side information required for the band extension. Due to the steganography, the proposed technique is able to enhance the speech quality without an increase of the amount of data transmission. From the results of a subjective experiment, it is indicated that the proposed technique may potentially be useful for improving the speech quality, compared with the conventional technique.

Fifty years of progress in speech waveform coding

Over the past 50 years, sustained research in speech coding has made it possible to encode speech with high speech quality at rates as low as 4 kb/s. The technology is now used in many applications, such as digital cellular phones, personal computers, and packet telephony. The early research in speech coding was aimed at reproducing speech spectra using a small number of slowly varying parameters. The focus of research shifted later to accurate reproduction of speech waveforms at low bit rates. The introduction of linear predictive coding (LPC) led to the development of new algorithms, such as adaptive predictive coding, multipulse and code-excited LPC. Code-excited LPC has become the method of choice for low bit rate speech coding and is used in most voice transmission standards. Digital speech communication is rapidly moving away from traditional circuit-switched to packet-switched networks based on IP protocols (VoIP). The focus of speech coding research is now on providing to low cost, reliable, and secure transmission of high-quality speech on IP networks.

Speech coding research at Bell Laboratories

The field of speech coding is now over 70 years old. It started from the desire to transmit voice signals over telegraph cables. The availability of digital computers in the mid 1960s made it possible to test complex speech coding algorithms rapidly. The introduction of linear predictive coding (LPC) started a new era in speech coding. The fundamental philosophy of speech coding went through a major shift, resulting in a new generation of low bit rate speech coders, such as multi-pulse and code-excited LPC. The semiconductor revolution produced faster and faster DSP chips and made linear predictive coding practical. Code-excited LPC has become the method of choice for low bit rate speech coding applications and is used in most voice transmission standards for cell phones. Digital speech communication is rapidly evolving from circuit-switched to packet-switched networks to provide integrated transmission of voice, data, and video signals. The new communication environment is also moving the focus of speech coding research from compression to low cost, reliable, and secure transmission of voice signals on digital networks, and provides the motivation for creating a new class of speech coders suitable for future applications.

DIGITAL SPEECH COMMUNICATION BY TRUNCATED CHAOTIC SYNCHRONIZATION

The method of truncated chaotic synchronization is suggested for digital speech communication. The sequence, composed by truncating some byte segments from the low-dimensional chaotic attractor, demonstrates complicated dynamics and high dimension. Predicting the dynamics of the truncated map through the local approximation method becomes very difficult. However, synchronization lets the message be exactly recovered. The synchronization condition is deduced to synchronize two systems of the truncated one-way coupled ring maps. The sensitivity of synchronization to parameter mismatch is discussed for the security in communication. For the mismatch of parameters or keys, the probability of decoding the message through guessing the keys is below 2-200. The scheme of truncated chaotic synchronization has strong security in encoding message.

Softbit speech decoding: a new approach to error concealment

In digital speech communication over noisy channels there is the need for reducing the subjective effects of residual bit errors which have not been eliminated by channel decoding. This task is usually called error concealment. We describe a new and generalizing approach to error concealment as part of a modified robust speech decoder. It can be applied to any speech codec standard and preserves bit exactness in the case of an error free channel. The proposed method requires bit reliability information provided by the demodulator or by the equalizer or specifically by the channel decoder and can exploit additionally a priori knowledge about codec parameters. We apply our algorithms to PCM, ADPCM, and GSM full-rate speech coding using AWGN, fading, and GSM channel models, respectively. It turns out that the speech quality is significantly enhanced, showing the desired inherent muting mechanism or graceful degradation behavior in the case of extreme adverse transmission conditions.

Low complexity speech coding at 1.2 to 2.4 kbps based on waveform interpolation

Low-rate speech coding technology has recently made a significant progress with the introduction of new interpolative algorithms (Shoham, 1993; Kleijn and Haagen, 1995). The inherent complexity of these algorithms is, however too high too be commercially useful for low-cost applications. In this paper we propose new approaches to low-complexity speech coding at coding rates of 1.2 and 2.4 kbps. The proposed methods utilize all the advantages of interpolatve coding but greatly simplify the analysis and synthesis operations to a point where low-cost two-way digital speech communication can be easily implemented on DSP or host platforms. At 2.4 kbps, the complexity of the proposed coder is about 7.5 and 2.5 MFLOPS for the encoder and decoder, respectively. At 1.2 kbps, the complexity is about 6 and 2.3 MFLOPS for the encoder and decoder, respectively. The small computational load of these coders make them suitable for multi-tasking environment and low-cost terminals. Informal subjective evaluation shows that, at 2.4 kbps, good communication quality is obtained. Communication quality is less than toll quality but the perceived coding effects are not annoying and do not prevent long sustained two-way conversation with high degree of intelligibility. The quality does not significantly degrade at 1.2 kbps and it is considered sufficient for messaging applications.

Digital communication by active-passive-decomposition synchronization in hyperchaotic systems

In this paper we investigate the digital speech communication by active-passive-decomposition (APD) synchronization in a hyperchaotic system with a one-way coupled ring map lattice with length m. We present an analysis of the synchronous principle, synchronization time, and the sensitivity of synchronization to the parameter differences. Experiments as well as theory demonstrate that the APD method leads to secure encoding, short time synchronization, and recovery without error.

Implementation of secure digital communication byhyperchaotic synchronisation

By active-passive-decomposition synchronisation, the application of the one-way coupled ring map lattice hyperchaotic system in secure digital speech communication is demonstrated. Synchronisation guarantees strong security and recovery without error.

Shared time division duplexing: an approach to low-delay high-quality wireless digital speech communications

Various strategies to provide low-delay high-quality digital speech communications in a high-capacity wireless network are examined. Various multiple access schemes based on time-division and packet reservation are compared in terms of their statistical multiplexing capabilities, sensitivity to speech packet dropping, delay, robustness to lossy packet environments, and overhead efficiency. In particular, a low-delay multiple access scheme, called shared time-division duplexing (STDD) is proposed. This scheme allows both the uplink and downlink traffic to share a common channel, thereby achieving high statistical multiplexing gain even with a low population of simultaneous conversations. The authors also propose a choice of low delay, high quality speech coding and digital modulation systems based on adaptive DPCM, with QDPSK or pseudo-analog transmission (skewed DPSK), for use in conjunction with the STDD multiple access protocol. The choice of the alternative systems depends on required end-to-end delay, recovered speech quality and bandwidth efficiency. Typically, with a total capacity of 1 MBaud, 2 ms frame and 8 kBaud speech coding rate, low delay STDD is able to support 48 pairs of users compared to 38, 35, and 16 for TDMA with speech activity detection, basic TDMA and PRMA respectively. This corresponds to respective gains of 26%, 37% and 200%.< <ETX xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">&gt;</ETX>

Secure digital speech communication

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