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THE EFFECT OF NARROW-BAND DIGITAL PROCESSING AND BIT ERROR RATE ON THE INTELLIGIBILITY OF ICAO SPELLING ALPHABET WORDS

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IEEE TRANSACTIONS ON ACOUSTICS SPEECH AND SIGNAL PROCESSING 1987年第8期35卷 1101-1115页

作者： SCHMIDTNIELSEN, A Naval Research Laboratory Inc. Washington D.C. DC USA

The diagnostic rhyme test (DRT) is widely used to evaluate digital voice systems. Would-be users often have no reference frame for interpreting DRT scores in terms of performance measures that they can understand, e.g., how many operational words are correctly understood. This research was aimed at providing a better understanding of the effects of very poor quality speech on human communication performance. It is especially important to determine how successful communications are likely to be when the speech quality is severely degraded. This paper compares the recognition of ICAO spelling alphabet words (ALFA, BRAVO, CHARLIE, etc.) to DRT scores for the same conditions. Confusions among the spelling alphabet words are also given. The voice conditions included unprocessed speech, speech processed through the DoD standard linear predictive coding algorithm operating at 2400 bits/s with random bit error rates of 0, 2, 5, 8, and 12 percent, and an 800 bit/s pattern matching algorithm. The results suggest that with distinctive vocabularies like the ICAO spelling alphabet, word intelligibility can be expected to remain very high even when DRT scores fall into the poor range; but once the DRT scores fall below about 75, the intelligibility can be expected to fall off rapidly; and at scores below 50, less than half the words will also be understood.

关键词： Narrowband Bit error rate Speech processing Speech coding System testing Humans Degradation linear predictive coding Prediction algorithms Pattern matching

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ON THE APPLICATION OF MIXTURE AR HIDDEN MARKOV-MODELS TO TEXT INDEPENDENT SPEAKER RECOGNITION

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IEEE TRANSACTIONS ON SIGNAL PROCESSING 1991年第3期39卷 563-570页

作者： TISHBY, NZ AT&T Bell Lab. Murray Hill NJ USA

linear predictive hidden Markov models have proved to be an efficient way for statistically modeling speech signals. The possible application of such models to statistical characterization of the speaker himself is described and evaluated. The results show that even with a short sequence of only four isolated digits, a speaker can be verified with an average equal-error rate of less than 3%. These results are slightly better than the results obtained using speaker dependent vector quantizers, with comparable numbers of spectral vectors. The small improvement over the vector quantization approach indicates the weakness of the Markovian transition probabilities for characterizing speaker dependent transitional information.

关键词： Hidden Markov models Speaker recognition predictive models Vector quantization Testing Error analysis Speech recognition linear predictive coding Cepstral analysis Distortion measurement

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Lossy pole-zero modeling for speech signals

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IEEE TRANSACTIONS ON SPEECH AND AUDIO PROCESSING 1996年第2期4卷 81-88页

作者： Lim, IT Lee, BG LG Electronics Research Center Seoul South Korea Department of Electronics Engineering Seoul National University Seoul South Korea

This paper presents an acoustic model-based lossy pole-zero modeling for speech signals, which overcomes the limitation in the existing lossless pole-zero model that forced the numerator part of the pole-zero transfer function to be symmetric, We derive the lossy pole-zero model and its transfer function by employing the wave digital filter (WDF) adaptor formulas and by converting the fixed termination value -1 to a loss factor mu(0)(c) is an element of (-1, 1). Then we discuss how to determine the reflections coefficients of the lossy pole-zero model, For this we first employ a well-performing ARMA modeling algorithm for a pole-zero type estimation of the given speech signal and then fit the transfer function of the lossy pole-zero model to that of the ARMA model under Euclidean cost function. This procedure is demonstrated by an example using the Steiglitz-McBride ARMA estimation method with a synthetic speech signal. The lossy pole-zero modeling yields a new filter structure-namely, three-branch lattice structure-which consists of three lattice branches with the third branch terminated by the loss factor mu(0)(c) is an element of [-1, 1] and with the three branches connected by a three-port wave adaptor characterized by the area ratio sigma is an element of [0, 1]. The three-branch lattice structure is a general filter structure which becomes tbe lossless pole-zero structure when mu(0)(c) = -1 and becomes the existing all-pole lattice structure when sigma = 0.

关键词： Transfer functions Lattices linear predictive coding Digital filters Termination of employment Reflection Cost function Speech analysis Speech synthesis

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Digital underwater acoustic voice communications

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IEEE JOURNAL OF OCEANIC ENGINEERING 1996年第2期21卷 181-192页

作者： Woodward, B Sari, H Department of Electronic and Electrical Engineering Loughborough University of Technology Leicestershire UK

This paper describes the design of an underwater acoustic diver communication system controlled by a digital signal processor. The speech signal transmission rate is compressed by using linear predictive coding (LPC) and the extracted parameters are transmitted through the water to a synchronized receiver by employing digital pulse position modulation (DPPM). The pulse position in each time frame is estimated by an energy detection and decision algorithm which enables the received LPC parameters to be recovered and used to synthesize the speech signal.

关键词： Underwater acoustics Underwater communication linear predictive coding Acoustic pulses Pulse modulation Speech synthesis Signal design Digital control Communication system control Control systems

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A VLSI chip for isolated speech recognition system

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IEEE TRANSACTIONS ON CONSUMER ELECTRONICS 1996年第3期42卷 458-467页

作者： Kim, SN Hwang, IC Kim, YW Kim, SW Department of Electronics Engineering Korea University Seoul South Korea

A VLSI processor is designed for the small-scale isolated speech recognition applications. It is a dedicated processor which detects endpoint, extracts LPC(linear predictive Coefficient) cepstral coefficients from the speech signal, and computes the spectral distances using a dynamic time warping(DTW) technique. The designed chip can recognize 1000 isolated words per second with an average recognition accuracy of 90.3%. It is designed in a 0.8 mu m CMOS technology, includes 66,760 gates, and runs with a 10MHz clock.

关键词： Very large scale integration Speech recognition CMOS technology Process design linear predictive coding Cepstral analysis Speech processing Signal processing Isolation technology Clocks

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A new phase model for sinusoidal transform coding of speech

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IEEE TRANSACTIONS ON SPEECH AND AUDIO PROCESSING 1998年第5期6卷 495-501页

作者： Ahmadi, S Spanias, AS Nokia Mobile Phones Inc San Diego CA 92121 USA

A phase modeling algorithm for sinusoidal analysis-synthesis of speech is presented, where short-time sinusoidal phases are approximated using a combination of linear prediction, spectral sampling, delay compensation, and phase correction techniques. The algorithm is different than phase compensation methods proposed for source-system LPC in that it has been tailored to sinusoidal representation of speech. Performance analysis on a large speech data base reveals improvement in temporal and spectral signal matching, as well as in the subjective quality of reconstructed speech. The method can be applied to enhance phase matching in low bit rate sinusoidal coders, where underlying sine wave amplitudes are extracted from an all-pole model. Preliminary subjective results are presented for a 2.4 kb/s sinusoidal coder.

关键词： linear prediction linear predictive coding phase modeling sinusoidal representation speech coding

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Forensic speaker and gender identification from voice samples recorded through mobile phones and social media applications: A statistical and machine learning approach

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APPLIED ACOUSTICS 2024年 222卷

作者： Gouri, Gulshan Sharma, Arushi Sharma, Vishal Panjab Univ Inst Forens Sci & Criminol Chandigarh India

The proliferation of the Internet and instant messaging has elevated the use of voice communication, posing challenges in legal and forensic contexts. This study delves into the impact of four social media platforms (WhatsApp, Instagram, Snapchat, and Telegram) on the acoustic properties of vowel sounds. Forty participants, evenly split between 20 males and 20 females, were recorded producing English monophthongs vowels under five conditions: using a mobile phone, WhatsApp, Instagram, Telegram, and Snapchat. Utilizing Multispeech and Praat speech acoustic software, we analyzed formant frequencies, employed statistical F tests, and applied machine learning techniques to assess the influence of social media applications on formant frequency (F1, F2, F3, and F4). The study findings demonstrated that F1, F2, and F3 effectively distinguished vowels, with accuracy rates ranging from 100% to 75% when considering formant frequency. However, in the formant listing, F2, F3, and F4 emerged as reliable markers for identifying vowels, achieving similar accuracy rates. Notably, vowel spaces constructed from mean formant frequencies exhibited distinct patterns, thereby accentuating differences between male and female speakers. Random Forest proved to be the top -performing machine learning approach in gender differentiation, consistently delivering high accuracy rates (ranging from 88% to 93%) and Area Under the Curve (AUC) values between 0.92 and 0.96. Remarkably, Random Forest exhibited strong performance across various algorithms, maintaining its effectiveness despite formant variations.

关键词： Speech linear predictive coding Formant listing Vowel plots Statistical analysis Machine learning

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LPC SPEECH coding BASED ON VARIABLE-LENGTH SEGMENT QUANTIZATION

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IEEE TRANSACTIONS ON ACOUSTICS SPEECH AND SIGNAL PROCESSING 1988年第9期36卷 1437-1444页

作者： SHIRAKI, Y HONDA, M Speech Research Group of the Basic Research Laboratories NTT Musashino Tokyo Japan

A low-bit-rate linear predictive coder (LPC) that is based on variable-length segment quantization is presented. In this vocoder, the speech spectral-parameter sequence is represented as the concatenation of variable-length spectral segments generated by linearly time-warping fixed-length code segments. Both the sequence of code segments and the segment lengths are efficiently determined using a dynamic programming procedure. This procedure minimizes the spectral distance measured between the original and the coded spectral sequence in a given interval. An iterative algorithm is developed for designing fixed-length code segments for the training spectral sequence. It updates the segment boundaries of the training spectral sequence using an a priori codebook and updates the codebook using these segment sequences. The convergence of this algorithm is discussed theoretically and experimentally. In experiments, the performance of variable-length segment quantization for voice coding is compared to that of fixed-length segment quantization and vector quantization.< >

关键词： linear predictive coding Speech coding Quantization Algorithm design and analysis Iterative algorithms Vocoders Speech processing Signal processing algorithms Bit rate Convergence

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Impact of network routing on the end-to-end transmission quality of cellular and PCS connections

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IEEE TRANSACTIONS ON VEHICULAR TECHNOLOGY 1998年第4期47卷 1227-1233页

作者： Dimolitsas, S Corcoran, FL Ravishankar, C Comsat Labs Clarksburg MD 20871 USA

Long-distance and especially international calls involve an increasing number of multilink circuits of cellular, personal communications, mobile satellite, and public switched telephone network (PSTN)-type connections incorporating a variety of speech coding devices, In particular, the rapid growth of cellular communications and the impeding emergence of personal communication systems (PCS's) has highlighted the need to characterize the quality of end-to-end connections when cellular and PCS terminals are connected over the PSTN. At the same time, the nonlinear nature of low-rate parametric speech coding has rendered unsuitable known analytical methods for estimating end-to-end quality of interconnected networks. Instead, quantification of transmission performance requires direct subjective evaluation of the pertinent connections of interest. In this paper, the quality of North American, Japanese, and European digital cellular and PCS terminals is quantified when these are interconnected using 16-kb/s low-delay code-excited linear prediction (LD-CELP) and 32-kb/s adaptive differential pulse code modulation (ADPCM) facilities. From these assessments, it can be concluded that digital cellular networks may meet end-to-end transmission planning criteria when interconnected with the switched network as long as calls are terminated to a wired terminal.

关键词： adaptive codes adaptive modulation cellular radio delays differential pulse code modulation digital radio internetworking linear predictive coding mobile satellite communication personal communication networks speech coding telecommunication network plan cellular radio Personal communication networks end-to-end transfer differential pulse code modulation Information Network Transmission Characteristics Digital radio mobile satellite communication linear Prediction coding Network routing Adaptive coding speech coding Personal communication systems adaptive modulation internetworking

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Audio Watermarking for Security and Non-Security Applications

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IEEE ACCESS 2022年 10卷 12654-12677页

作者： Charfeddine, Maha Mezghani, Eya Masmoudi, Salma Ben Amar, Chokri Alhumyani, Hesham Univ Sfax Natl Engn Sch Sfax ENIS REGIM REs Grp Intelligent Machines Sfax 3038 Tunisia Taif Univ Coll Comp & Informat Technol Dept Comp Engn At Taif 21944 Saudi Arabia

The digitization of audiovisual data is significantly increasing. Thus, to guarantee the protection of the intellectual properties of this digital content, watermarking has appeared as a solution. Watermarking can be used in reality in several types of applications that target two different contexts: the first for security applications and the second for non-security ones. In this paper, we carry a big interest in studying these two types of applications. Moreover, we propose a first digital watermarking scheme for security copyright protection applications, where we have involved neural network architecture in the insertion and detection processes, and integrated some masking phenomena of the human psychoacoustic model with linear predictive coding spectral envelope estimation of the audio file. Experiments proved the efficiency of exploiting perceptual masking with spectral envelope consideration in terms of imperceptibility and robustness results. In addition, we suggest a second audio watermarking technique for non-security content characterization applications based on a deep learning classification architecture. In this scheme, the extracted watermark advises about the audio class: music or speech, speaker gender, and emotion. The reported results indicated that the suggested scheme achieved a higher performance at the classification level, as well as at the watermarking properties.

关键词： Watermarking Security Robustness Standards Estimation Computer architecture Transform coding Copyright protection human psychoacoustic model linear predictive coding audio content characterization deep learning architecture

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