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作者： Li, Xing University of Washington

学位级别：Ph.D.

Complex broadband sounds are decomposed by the auditory filters into a series of relatively narrowband signals, each of which conveys information about the sound by time-varying features. The slow changes in the overall amplitude constitute envelope, while the more rapid events, such as zero crossings, constitute temporal fine structure (TFS). Although envelope cues from a small number of channels can support robust speech recognition in quiet, TFS seems to plays a significant role for speech perception in noise, especially in fluctuating background. Fundamental questions about the relative importance of envelope and TFS have been addressed by many studies. The definition of TFS poses a critical issue. Due to the coupling between envelope and phase, it is problematic to isolate the TFS from the envelope for any signal which is not extremely narrowband. Conventionally, a Hilbert transform is used to represent each band as the product of the Hilbert envelope and a frequency-modulated (FM) sinusoidal carrier. The FM component is then taken as the TFS of the band. We show in this dissertation that the Hilbert FM is a distorted representation. To address this concern, we proposed a new distortion-free additive view of signal decomposition, the slow envelope and the fast envelope, using half wave rectification followed by filters reflecting engineering interpretation of neural physiology. The slow envelope is a tool for representing temporal cues that can be coded in the average firing rate of auditory nerve fibers, while the fast envelope instead captures the temporal cues conveyed in neural phase locking patterns. Using this new decomposition and the conventional Hilbert decomposition, we investigated the relative contribution of neural envelope and TFS coding to speech intelligibility in different noise conditions. The neural representation was generated by a simplified peripheral auditory model (Shamma and Lorenzi, 2013). We observed that the distortions in the Hilber

关键词： cochlear implants harmonic coding pitch perception speech perception temporal fine structure timbre recognition Acoustics

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Improved Perception of Music With a harmonic Based Algorithm for Cochlear Implants

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IEEE TRANSACTIONS ON NEURAL SYSTEMS AND REHABILITATION ENGINEERING 2013年第4期21卷 684-694页

作者： Li, Xing Nie, Kaibao Imennov, Nikita S. Rubinstein, Jay T. Atlas, Les E. Univ Washington Dept Elect Engn Seattle WA 98195 USA Univ Washington Virginia Merrill Bloedel Hearing Res Ctr Seattle WA 98195 USA Univ Washington Dept Bioengn Seattle WA 98105 USA

The lack of fine structure information in conventional cochlear implant (CI) encoding strategies presumably contributes to the generally poor music perception with CIs. To improve CI users' music perception, a harmonic-single-sideband-encoder (HSSE) strategy was developed [1]-[3], which explicitly tracks the harmonics of a single musical source and transforms them into modulators conveying both amplitude and temporal fine structure cues to electrodes. To investigate its effectiveness, vocoder simulations of HSSE and the conventional continuous-interleaved-sampling (CIS) strategy were implemented. Using these vocoders, five normal-hearing subjects' melody and timbre recognition performance were evaluated: a significant benefit of HSSE to both melody (p < 0.002) and timbre (p < 0.026) recognition was found. Additionally, HSSE was acutely tested in eight CI subjects. On timbre recognition, a significant advantage of HSSE over the subjects' clinical strategy was demonstrated: the largest improvement was 35% and the mean 17% (p < 0.013). On melody recognition, two subjects showed 20% improvement with HSSE;however, the mean improvement of 7% across subjects was not significant (p > 0.090). To quantify the temporal cues delivered to the auditory nerve, the neural spike patterns evoked by HSSE and CIS for one melody stimulus were simulated using an auditory nerve model. Quantitative analysis demonstrated that HSSE can convey temporal pitch cues better than CIS. The results suggest that HSSE is a promising strategy to enhance music perception with CIs.

关键词： Cochlear implant fine structure harmonic coding pitch perception timbre perception

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Multi-mode harmonic transform coding (MMHTC) for speech and music signals

Multi-mode harmonic transform coding (MMHTC) for speech and ...

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3rd International Conference on Computing, Communications and Control Technologies

作者： Kim, Jong-Hark Shin, Jae-Hyun Lee, In-Sung Chungbuk Natl Univ Dept Radio Engn Cheongju 361763 South Korea

ISBN: (纸本)9806560477

A multi-mode harmonic transform coding (MMHTC) for speech and music signals is proposed. Its structure is organized as a linear prediction model with an input of harmonic and transform-based excitation. The proposed coder also utilizes harmonic prediction and an improved quantizer of excitation signal. To efficiently quantize the excitation of music signals, the modulated lapped transform (MLT) is introduced. In other words, the coder combines both the time domain (linear prediction) and the frequency domain technique to achieve the best perceptual quality The proposed coder showed better speech quality than that of the 8 kbps QCELP coder at a bit-rate of 4 kbps.

关键词： speech coding harmonic coding CELP audio coding

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harmonic vector excitation coding of speech

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ACOUSTICAL SCIENCE AND TECHNOLOGY 2006年第6期27卷 375-383页

作者： Nishiguchi, Masayuki Sony Corp Technol Dev Grp Audio Codec Dev Dept Shinagawa Ku 6-7-35 Kitashinagawa Tokyo 1410001 Japan

A coding algorithm for speech called harmonic vector excitation coding (HVXC) has been developed that encodes speech at very low bit rates (2.0-4.0 kbit/s). It breaks speech signals down into two types of segments: voiced segments, for which a parametric representation of harmonic spectral magnitudes of LPC residual signals is used;and unvoiced segments, for which the CELP coding algorithm is used. This combination provides near toll-quality speech at 4.0 kbit/s, and communication-quality speech at 2.0 kbit/s, thus outperforming FS1016 4.8-kbit/s CELP. This paper discusses the encoder and decoder algorithms for HVXC, including fast harmonic synthesis, time scale modification, and pitch-change decoding. Due to its high coding efficiency and new functionality, HVXC has been adopted as the ISO/IEC International Standard for MPEG-4 audio.

关键词： harmonic coding VXC HVXC CELP Vector quantization MPEG-4

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Weighted vector quantization of harmonic spectral magnitudes for very low-bit-rate speech coding

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ACOUSTICAL SCIENCE AND TECHNOLOGY 2006年第1期27卷 43-49页

作者： Nishiguchi, Masayuki Sony Corp Shinagawa Ku 6-7-35 Kitashinagawa Tokyo 1410001 Japan

harmonic coding is a very powerful technique for the coding of speech at very low bit rates;and the efficient coding of spectral magnitudes sampled at harmonic frequencies is the key to obtaining good coded-speech quality. This paper presents a weighted vector quantization method for spectral vectors composed of a variable number of harmonic magnitudes. It is based on simple, efficient linear dimension conversion and employs a weighted distortion measure that exploits the human auditory sense. A codebook training algorithm using the weighting matrix is also presented. Finally, a low-complexity VQ codebook search technique based on pre-selection is described that reduces the computational complexity to less than 10% of that of an exhaustive search, without perceptible loss of quality. The proposed quantization scheme is used in harmonic Vector eXcitation coding (HVXC), which is a very low-bit-rate speech coding algorithm that combines harmonic and stochastic vector representations of LPC residual signals. Due to the high efficiency of this VQ scheme, HVXC provides good communication-quality speech at bit rates as low as 2-4 kbit/s, and was adopted as the ISO/IEC International Standard for MPEG-4 Audio.

关键词： harmonic coding VXC HVXC CELP Vector quantization MPEG-4

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Integration of harmonic and analysis by synthesis coders

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IEE PROCEEDINGS-VISION IMAGE AND SIGNAL PROCESSING 2002年第6期149卷 321-326页

作者： Katugampala, NN Kondoz, AM Univ Surrey Ctr Commun Syst Res Guildford GU2 7XH Surrey England

Hybrid coding of speech has been proposed to overcome the limitations of a single model in representing the wide variety of characteristics of human speech. A new hybrid coding algorithm, which combines harmonic and analysis by synthesis coding techniques, is presented. To integrate the harmonic and analysis by synthesis coders, novel phase synchronisation and speech classification techniques are developed. The perceptual quality of the speech synthesised using the unquantised hybrid model is almost indistinguishable when compared with 128 kbit/s linear PCM. Two variable rate coders are developed based on the designed hybrid model, by quantising the parameters at different bit rates. Subjective listening tests show that the speech quality of the variable rate hybrid coders outperform the quality of 5.3 kbit/s and 6.3 kbit/s ITU G.723.1 coders, at maximum bit rates of 4 kbit/s and 6 kbit/s respectively.

关键词： variable rate coders 6 kbit/s subjective listening tests variable rate codes analysis by synthesis coding phase synchronisation ACELP 4 kbit/s harmonic excitation Speech and audio coding hybrid coding algorithm harmonic coding speech classification speech coding vocoders perceptual speech quality

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