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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Voiced speech synthesis with a nonlinear glottal model 基于非线性声门模型的浊音合成
Akhil Kumar, A. Gersho
{"title":"Voiced speech synthesis with a nonlinear glottal model","authors":"Akhil Kumar, A. Gersho","doi":"10.1109/SCFT.1997.623867","DOIUrl":"https://doi.org/10.1109/SCFT.1997.623867","url":null,"abstract":"We propose a model for voiced speech synthesis using a threshold autoregressive (TAR) glottal flow model. It reconstructs the pitch periodicity and tracks small pitch period variations even at a low parameter update rate of 20 ms, without requiring explicit pitch information. The model overcomes several limitations of traditional glottal models and has potential for application to high quality speech synthesis and low bit rate speech coding.","PeriodicalId":364273,"journal":{"name":"1997 IEEE Workshop on Speech Coding for Telecommunications Proceedings. Back to Basics: Attacking Fundamental Problems in Speech Coding","volume":"56 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":"124297628","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: 1
Speech excitation modelling for low bit speech coding 低比特语音编码的语音激励建模
Xiaoqin Sun, B. Cheetham
{"title":"Speech excitation modelling for low bit speech coding","authors":"Xiaoqin Sun, B. Cheetham","doi":"10.1109/SCFT.1997.623870","DOIUrl":"https://doi.org/10.1109/SCFT.1997.623870","url":null,"abstract":"This paper examines commonly accepted models for voiced speech production to determine more accurate ways of relating magnitude and phase spectra, taking into account the non-minimum phase nature of the glottal excitation. The implications for low-bit rate coding techniques such as STC, PWI and MBE, which encode magnitude information only, are addressed.","PeriodicalId":364273,"journal":{"name":"1997 IEEE Workshop on Speech Coding for Telecommunications Proceedings. Back to Basics: Attacking Fundamental Problems in Speech Coding","volume":"56 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":"134475865","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
Multi-band CELP coding of speech and music 语音和音乐的多波段CELP编码
A. Ubale, A. Gersho
{"title":"Multi-band CELP coding of speech and music","authors":"A. Ubale, A. Gersho","doi":"10.1109/SCFT.1997.623916","DOIUrl":"https://doi.org/10.1109/SCFT.1997.623916","url":null,"abstract":"A novel low-delay wideband speech coder, called multi-band CELP (MB-CELP), overcomes the major obstacles usually associated with two traditional CELP approaches to wideband speech coding, namely, fullband CELP and splitband CELP. The new MB-CELP coder employs a multiband bank of off-line filtered excitation codebooks, fullband linear prediction synthesis, and minimization of the error between original and synthesized audio signals over the full frequency range. Two versions of MB-CELP coder at 16 and 24 kbps are described. Subjective comparison test results show that these coders perform better than G.722 coder at 48 and 56 kbps respectively.","PeriodicalId":364273,"journal":{"name":"1997 IEEE Workshop on Speech Coding for Telecommunications Proceedings. Back to Basics: Attacking Fundamental Problems in Speech Coding","volume":"38 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":"117228714","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: 1
Audibility of pitch-synchronously modulated noise 音高同步调制噪声的可听性
J. Skoglund, W. Bastiaan Kleijn, P. Hedelin
{"title":"Audibility of pitch-synchronously modulated noise","authors":"J. Skoglund, W. Bastiaan Kleijn, P. Hedelin","doi":"10.1109/SCFT.1997.623891","DOIUrl":"https://doi.org/10.1109/SCFT.1997.623891","url":null,"abstract":"This paper examines the audibility of stationary and cyclostationary narrow-band noise added to voiced speech generated by natural and synthetic excitation. Varying the temporal location of noise within a pitch cycle corresponds to varying its phase spectrum. Exploiting this fact, we find that a change of phase is more perceptible for a low-pitched sound than for a high-pitched sound. Our results support the notion that sinusoidal coders work well for female speech and that CELP coders work well for male speech.","PeriodicalId":364273,"journal":{"name":"1997 IEEE Workshop on Speech Coding for Telecommunications Proceedings. Back to Basics: Attacking Fundamental Problems in Speech Coding","volume":"99 3","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"1997-09-07","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"120883940","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: 21
Modeling and quantization of speech magnitude spectra at low data rates-evaluating design trade-offs 低数据速率下语音幅度谱的建模和量化-评估设计权衡
A.M. Smith, T. Ramabadran, M. McLaughlin
{"title":"Modeling and quantization of speech magnitude spectra at low data rates-evaluating design trade-offs","authors":"A.M. Smith, T. Ramabadran, M. McLaughlin","doi":"10.1109/SCFT.1997.623873","DOIUrl":"https://doi.org/10.1109/SCFT.1997.623873","url":null,"abstract":"Several designs for modeling and quantizing the discrete harmonic amplitudes in voiced speech spectra are investigated. A cumulative distortion measure is computed for each design. This measure is shown to provide useful insight for evaluating design trade-offs.","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":"127944273","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: 1
An improved harmonic-plus-noise decomposition method and its application in pitch determination 一种改进的谐波加噪声分解方法及其在基音确定中的应用
R. Ahn, W. Holmes
{"title":"An improved harmonic-plus-noise decomposition method and its application in pitch determination","authors":"R. Ahn, W. Holmes","doi":"10.1109/SCFT.1997.623886","DOIUrl":"https://doi.org/10.1109/SCFT.1997.623886","url":null,"abstract":"This paper presents an improved method to decompose the harmonic and the noise components of voiced speech. The improvements make the method more accurate and robust by applying noise and harmonic separation in alternative iterations and by including a new pitch detection algorithm. This new pitch detection algorithm utilises the proposed decomposition as a main component. The results show that the improved decomposition converges faster than the original method.","PeriodicalId":364273,"journal":{"name":"1997 IEEE Workshop on Speech Coding for Telecommunications Proceedings. Back to Basics: Attacking Fundamental Problems in Speech Coding","volume":"39 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":"124147450","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
Spectral postfilter design based on LSP transformation 基于LSP变换的谱后滤波设计
H. Tasaki, K. Shiraki, K. Tomita, S. Takahashi
{"title":"Spectral postfilter design based on LSP transformation","authors":"H. Tasaki, K. Shiraki, K. Tomita, S. Takahashi","doi":"10.1109/SCFT.1997.623894","DOIUrl":"https://doi.org/10.1109/SCFT.1997.623894","url":null,"abstract":"A new spectral postfilter that has high flexibility of design is proposed. In this method, two LSP (line spectrum pair) transformation blocks compute ARMA filter coefficients for the postfilter. The characteristics of the proposed postfilter can be flexibly designed in the frequency domain, because an effect of LSP transformation can be estimated in the frequency domain. As examples of design, an approximation to a conventional LPC-based postfilter, enhancement in selected frequency bands, and a suitable design for a tandem coding are demonstrated.","PeriodicalId":364273,"journal":{"name":"1997 IEEE Workshop on Speech Coding for Telecommunications Proceedings. Back to Basics: Attacking Fundamental Problems in Speech Coding","volume":"39 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":"116484084","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: 7
A 1.6 kb/s MELP coder for wireless communications 用于无线通信的1.6 kb/s MELP编码器
A. McCree, J. Carlos de Martin
{"title":"A 1.6 kb/s MELP coder for wireless communications","authors":"A. McCree, J. Carlos de Martin","doi":"10.1109/SCFT.1997.623877","DOIUrl":"https://doi.org/10.1109/SCFT.1997.623877","url":null,"abstract":"This paper describes our new mixed excitation linear predictive (MELP) coder designed for wireless applications. This new coder, through algorithmic improvements and enhanced quantization techniques, produces better speech quality at 1.6 kb/s than the new U.S. Federal Standard MELP coder at 2.4 kb/s. A key feature of the coder is a novel line spectral frequencies (LSF) quantization scheme, requiring only 20 bits per frame. With channel coding (using convolutional codes), the new MELP coder at 3.1 kb/s is capable of maintaining good speech quality even in very degraded channels.","PeriodicalId":364273,"journal":{"name":"1997 IEEE Workshop on Speech Coding for Telecommunications Proceedings. Back to Basics: Attacking Fundamental Problems in Speech Coding","volume":"17 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":"114515317","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: 5
PWI speech coder in the speech domain 语音域的PWI语音编码器
D. Marston, F. Plante
{"title":"PWI speech coder in the speech domain","authors":"D. Marston, F. Plante","doi":"10.1109/SCFT.1997.623881","DOIUrl":"https://doi.org/10.1109/SCFT.1997.623881","url":null,"abstract":"Prototype waveform interpolation (PWI) coding can produce high quality speech at low bit rates. Most PWI coders that have been published carry out the coding in the residual domain, as opposed to the speech domain. This paper discusses a speech domain approach which aims to reduce quantisation complexity and improve speech quality.","PeriodicalId":364273,"journal":{"name":"1997 IEEE Workshop on Speech Coding for Telecommunications Proceedings. Back to Basics: Attacking Fundamental Problems in Speech Coding","volume":"34 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":"134454603","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: 1
A good job well done: the latest wireless codecs deliver wireline quality 干得好:最新的无线编解码器提供有线质量
L. Thorpe, P.V. Coverdale
{"title":"A good job well done: the latest wireless codecs deliver wireline quality","authors":"L. Thorpe, P.V. Coverdale","doi":"10.1109/SCFT.1997.623908","DOIUrl":"https://doi.org/10.1109/SCFT.1997.623908","url":null,"abstract":"The results of a listening test conducted at Nortel on behalf of the CDMA Development Group (CDG) suggest that the new crop of wireless codecs are all able to deliver voice quality comparable to wireline with single encoding over a clean wireless transmission channel. The codecs were tested with clean speech, varying input level, background noise, and tandem encoding. Given these findings, we hope to provoke discussion about the next challenges facing speech coding researchers and codec designers. The potential issues are: codec robustness to errors, noise reduction, and very-low bit rate codecs for a variety of applications.","PeriodicalId":364273,"journal":{"name":"1997 IEEE Workshop on Speech Coding for Telecommunications Proceedings. Back to Basics: Attacking Fundamental Problems in Speech Coding","volume":"68 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":"129082706","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
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