1997 IEEE Workshop on Speech Coding for Telecommunications Proceedings. Back to Basics: Attacking Fundamental Problems in Speech Coding最新文献

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A robust low rate voice codec for wireless communications 一种用于无线通信的鲁棒低速率语音编解码器
K. Swaminathan, S. Nandkumar, U. Bhaskar, N. Kowalski, S. Patel, G. Zakaria, J. Li, V. Prasad
{"title":"A robust low rate voice codec for wireless communications","authors":"K. Swaminathan, S. Nandkumar, U. Bhaskar, N. Kowalski, S. Patel, G. Zakaria, J. Li, V. Prasad","doi":"10.1109/SCFT.1997.623903","DOIUrl":"https://doi.org/10.1109/SCFT.1997.623903","url":null,"abstract":"The design, implementation and performance of a high quality low bit rate speech codec for wireless communication is presented. The codec is based on the CELP model. Generalized analysis-by-synthesis, algebraic fixed codebooks, and multistage LSF techniques are used, resulting in robustness to transmission errors and high quality across changing speech levels and background noise conditions. The bit allocations for the quantization of LSF, pitch and the excitation are chosen in a mode specific manner based on a robust mode classification scheme. A 4.8 kb/s version has been implemented and subjective tests show speech quality that is equivalent to or better than most cellular standard codecs. Performance is also consistent across speech levels and transmission errors.","PeriodicalId":364273,"journal":{"name":"1997 IEEE Workshop on Speech Coding for Telecommunications Proceedings. Back to Basics: Attacking Fundamental Problems in Speech Coding","volume":"16 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1997-09-07","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"114545403","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
引用次数: 4
A cross-language performance study of vector quantisation 向量量化的跨语言性能研究
J. J. Parry, I. Burnett, J. Chicharo
{"title":"A cross-language performance study of vector quantisation","authors":"J. J. Parry, I. Burnett, J. Chicharo","doi":"10.1109/SCFT.1997.623905","DOIUrl":"https://doi.org/10.1109/SCFT.1997.623905","url":null,"abstract":"This paper investigates the performance of split vector quantisation (VQ) of the line spectral frequencies (LSFs) across a set of 10 modern languages. Spectral quantisation accounts for a significant portion of the bit allocation in low-rate speech coding. Split VQ of the LSFs can achieve transparent quantisation of the linear prediction coefficients at 24 bits/frame. The codebooks are trained on individual languages and the cross-language VQ performance was measured using spectral distortion (SD). The results show that the spectral structure of the codebook training language influences the performance of the VQ. The number of bits/frame required for transparent speech varied by as much a 2 bits across languages.","PeriodicalId":364273,"journal":{"name":"1997 IEEE Workshop on Speech Coding for Telecommunications Proceedings. Back to Basics: Attacking Fundamental Problems in Speech Coding","volume":"230 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1997-09-07","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"123873993","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
引用次数: 2
A 0.75 kbps speech codec using recognition and synthesis schemes 使用识别和合成方案的0.75 kbps语音编解码器
Heng-Chou Chen, Chin-Yung Chen, Kai-Ming Tsou, O.T.-C. Chen
{"title":"A 0.75 kbps speech codec using recognition and synthesis schemes","authors":"Heng-Chou Chen, Chin-Yung Chen, Kai-Ming Tsou, O.T.-C. Chen","doi":"10.1109/SCFT.1997.623879","DOIUrl":"https://doi.org/10.1109/SCFT.1997.623879","url":null,"abstract":"In this paper, we proposed a very low bit-rate speech codec using recognition and synthesis schemes. The 2512 speech units, including 48 phones and 2463 diphones are utilized in the recognition process. The three-state continuous hidden Markov model, excluding the start and final states, is applied to model these speech units. In addition to the recognized phonetic index, the corresponding phonetic frame length is also the compressed information. In order to obtain a better quality of the reconstructed speech, pitch periods and pitch gains are realized to preserve the speaker's personal characteristics. In the synthesis process, the time-domain pitch-synchronous overlapped addition scheme is utilized to synthesize a high-quality speech waveform. In our tests, a more than 90% recognition accuracy can be achieved when the user speaks in a normal behavior. The reconstructed speech quality can be above a mean opinion score of 3.0, and a diagnostic rhyme test score of 92.","PeriodicalId":364273,"journal":{"name":"1997 IEEE Workshop on Speech Coding for Telecommunications Proceedings. Back to Basics: Attacking Fundamental Problems in Speech Coding","volume":"1 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1997-09-07","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"121932927","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
引用次数: 8
A 6.4 kbit/s extension to the G.729 speech coder G.729语音编码器的6.4 kbit/s扩展
A. Kataoka, S. Kurihara, S. Hayashi
{"title":"A 6.4 kbit/s extension to the G.729 speech coder","authors":"A. Kataoka, S. Kurihara, S. Hayashi","doi":"10.1109/SCFT.1997.623900","DOIUrl":"https://doi.org/10.1109/SCFT.1997.623900","url":null,"abstract":"This paper proposes a 6.4-kbit/s extension to the G.729 speech coder. Each G.729 module was investigated to determine which bits could be removed without hurting the speech quality, then a 6.4-kbit/s coder was designed. This coder has an 11-bit algebraic codebook that uses two or three pulses. We also propose a conditional orthogonalized search for the fixed codebook to improve the speech quality. The conditional orthogonalized search chooses one of two search methods (orthogonalized or non-orthogonalized) based on the optimum pitch gain. Subjective tests have indicated that the quality of the proposed coder is equivalent to that of the 6.3-kbit/s G.723.1 coder.","PeriodicalId":364273,"journal":{"name":"1997 IEEE Workshop on Speech Coding for Telecommunications Proceedings. Back to Basics: Attacking Fundamental Problems in Speech Coding","volume":"67 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1997-09-07","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"131465780","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
引用次数: 3
A noise annoys an oyster: evaluating the subjective effectiveness of single-input noise reduction techniques 噪音惹恼牡蛎:评估单输入降噪技术的主观有效性
L. Thorpe
{"title":"A noise annoys an oyster: evaluating the subjective effectiveness of single-input noise reduction techniques","authors":"L. Thorpe","doi":"10.1109/SCFT.1997.623909","DOIUrl":"https://doi.org/10.1109/SCFT.1997.623909","url":null,"abstract":"Single-input noise reduction techniques show enormous potential for improving the quality of signals that originate in noisy environments or that become mixed with noise along the way. These techniques are of particular interest to developers of speech compression codecs since the quality of noise processed through the codec is often a limiting factor for codec performance. This paper considers the assessment of the subjective performance of noise reduction (NR) techniques for telephony applications. Parameters of interest include the selection of the source material (such as noise type and level), voice characteristics, input impairments, the presence of other DSP processing on the channel, and the choice of test procedure. Some results from absolute category rating tests are presented, showing the subjective improvement found with NR techniques based on spectral subtraction.","PeriodicalId":364273,"journal":{"name":"1997 IEEE Workshop on Speech Coding for Telecommunications Proceedings. Back to Basics: Attacking Fundamental Problems in Speech Coding","volume":"25 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1997-09-07","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"131172021","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
引用次数: 0
Robust voiced/unvoiced speech classification using fuzzy rules 基于模糊规则的鲁棒浊音/浊音分类
F. Beritelli, S. Casale
{"title":"Robust voiced/unvoiced speech classification using fuzzy rules","authors":"F. Beritelli, S. Casale","doi":"10.1109/SCFT.1997.623868","DOIUrl":"https://doi.org/10.1109/SCFT.1997.623868","url":null,"abstract":"The paper presents a robust voiced/unvoiced speech classifier based on fuzzy logic. More specifically, the classification is based on a pattern recognition approach in which the matching phase is performed using a set of 5 fuzzy rules obtained by training. Certain interesting statistical properties of the fuzzy system allow the transition threshold to be adapted to the level of background noise. The results show that the performance of the fuzzy classifier in the presence of various types of background noise is better than that of traditional methods.","PeriodicalId":364273,"journal":{"name":"1997 IEEE Workshop on Speech Coding for Telecommunications Proceedings. Back to Basics: Attacking Fundamental Problems in Speech Coding","volume":"29 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1997-09-07","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"122062311","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
引用次数: 19
A robust low complexity voice activity detection algorithm for speech communication systems 一种用于语音通信系统的鲁棒低复杂度语音活动检测算法
A. Benyassine, E. Shlomot, H. Su, E. Yuen
{"title":"A robust low complexity voice activity detection algorithm for speech communication systems","authors":"A. Benyassine, E. Shlomot, H. Su, E. Yuen","doi":"10.1109/SCFT.1997.623914","DOIUrl":"https://doi.org/10.1109/SCFT.1997.623914","url":null,"abstract":"We present a robust voice activity detection algorithm which enables further bit rate reduction for practical speech communication systems. The voice activity decision is based on a set of energy and spectral measures. The local frame measures are compared to a long term estimation of the noise. The voice activity decision is obtained using a pattern recognition approach with additional decision smoothing. This algorithm was adopted as part of Annex B for the ITU Recommendation G.729.","PeriodicalId":364273,"journal":{"name":"1997 IEEE Workshop on Speech Coding for Telecommunications Proceedings. Back to Basics: Attacking Fundamental Problems in Speech Coding","volume":"32 3 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1997-09-07","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"123213382","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
引用次数: 14
Poincare section techniques for speech 讲话的庞加莱分段技术
Gernot Kubin
{"title":"Poincare section techniques for speech","authors":"Gernot Kubin","doi":"10.1109/SCFT.1997.623869","DOIUrl":"https://doi.org/10.1109/SCFT.1997.623869","url":null,"abstract":"When speech signals are modeled as the output of a nonlinear oscillator, a phase space representation can be obtained by combining several delayed versions of the signal into a trajectory in multi-dimensional space. Due to the cyclic nature of voiced speech, the trajectory approximately repeats itself after every pitch period such that at each point in phase space we see a synchronized bundle of trajectories. Cross-sections through these bundles are referred to as Poincare sections and allow us to assess the signal's deviation from strict periodicity. This gives rise to a cyclostationary description of voiced speech. An extra benefit of the Poincare section approach is the possibility for synchronization of pitch cycles during intervals with strongly irregular pitch period.","PeriodicalId":364273,"journal":{"name":"1997 IEEE Workshop on Speech Coding for Telecommunications Proceedings. Back to Basics: Attacking Fundamental Problems in Speech Coding","volume":"151 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1997-09-07","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"122984872","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
引用次数: 13
Waveform interpolation and analysis-by-synthesis-a good match? 波形插值和分析合成-一个很好的匹配?
I. Burnett, J. Ni
{"title":"Waveform interpolation and analysis-by-synthesis-a good match?","authors":"I. Burnett, J. Ni","doi":"10.1109/SCFT.1997.623880","DOIUrl":"https://doi.org/10.1109/SCFT.1997.623880","url":null,"abstract":"This paper considers the incorporation of analysis-by-synthesis (A-by-S) techniques in waveform interpolation (WI) coding architectures. It proposes an altered A-by-S mechanism which overcomes the non-synchronous input/output speech which is characteristic of WI coders. The proposed architectures operate on a prototype-by-prototype basis, optimising parameter choices within each frame. Three possible architectures are considered, with varying degrees of A-by-S integration. These low-rate, 2.4 kb/s coders are tailored to the transmission of no prototype phase information. It is also shown that the incorporation of A-by-S in WI allows exploitation of CELP style perceptual weighting techniques. The conclusion is that the use of A-by-S can improve the perceptual performance of WI coders.","PeriodicalId":364273,"journal":{"name":"1997 IEEE Workshop on Speech Coding for Telecommunications Proceedings. Back to Basics: Attacking Fundamental Problems in Speech Coding","volume":"54 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1997-09-07","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"127747832","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
引用次数: 2
Spectral quantization using statistics of static and dynamic features 光谱量化使用统计静态和动态特征
K. Koishida, K. Tokuda, T. Masuko, T. Kobayashi
{"title":"Spectral quantization using statistics of static and dynamic features","authors":"K. Koishida, K. Tokuda, T. Masuko, T. Kobayashi","doi":"10.1109/SCFT.1997.623875","DOIUrl":"https://doi.org/10.1109/SCFT.1997.623875","url":null,"abstract":"We propose a vector quantization (VQ) method using statistics of static and dynamic features. The distortion between the input vector and codeword is measured by the likelihood defined by statistics. The output vector is determined so that its likelihood is maximized. The proposed method enables us to obtain the quantized spectra which vary quite smoothly. To evaluate the proposed method, we applied it to encoding of the LSP parameters. From the subjective test, it was shown that the use of the proposed method results in a significant improvement of subjective quality.","PeriodicalId":364273,"journal":{"name":"1997 IEEE Workshop on Speech Coding for Telecommunications Proceedings. Back to Basics: Attacking Fundamental Problems in Speech Coding","volume":"130 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1997-09-07","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"124249938","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
引用次数: 8
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