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Variable Rate Characteristic Waveform Interpolation Speech Coder Based on Phonetic Classification
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作者 王晶 匡镜明 赵胜辉 《Journal of Beijing Institute of Technology》 EI CAS 2007年第2期187-192,共6页
A variable-bit-rate characteristic waveform interpolation (VBR-CWI) speech codec with about 1.8 kbit/s average bit rate which integrates phonetic classification into characteristic waveform (CW) decomposition is p... A variable-bit-rate characteristic waveform interpolation (VBR-CWI) speech codec with about 1.8 kbit/s average bit rate which integrates phonetic classification into characteristic waveform (CW) decomposition is proposed. Each input frame is classified into one of 4 phonetic classes. Non-speech frames are represented with Bark-band noise model. The extracted CWs become rapidly evolving waveforms (REWs) or slowly evolving waveforms (SEWs) in the cases of unvoiced or stationary voiced frames respectively, while mixed voiced frames use the same CW decomposition as that in the conventional CWI. Experimental results show that the proposed codec can eliminate most buzzy and noisy artifacts existing in the fixed-bit-rate characteristic waveform interpolation (FBR-CWI) speech codec, the average bit rate can be much lower, and its reconstructed speech quality is much better than FS 1 016 CELP at 4.8 kbit/s and similar to G. 723.1 ACELP at 5.3 kbit/s. 展开更多
关键词 variable bit rate speech coding characteristic waveform interpolation phonetic classification
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Multi-Level Error Detection and Concealment Algorithm to Improve Speech Quality in GSM Full Rate Speech Codecs
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作者 王林芳 刘加 +1 位作者 刘小青 李明 《Tsinghua Science and Technology》 SCIE EI CAS 2011年第3期247-255,共9页
Digital mobile telecommunication systems, such as the global system for mobile (GSM) system, want to further improve speech communication quality without changing the channel encoders and decoders. Speech quality is... Digital mobile telecommunication systems, such as the global system for mobile (GSM) system, want to further improve speech communication quality without changing the channel encoders and decoders. Speech quality is most affected by residual bit errors in received speech frames. Conventional methods use binary decision strategies for error detection and concealment in frames. This paper presents a multi-level error detection and concealment algorithm for GSM full rate speech codec systems. The algorithm uses multi-source knowledge to detect and conceal speech frame errors at the frame, parameter, and even bit levels. Tests show that most corrupted frames can be appropriately concealed by this algorithm, resulting in MOS gains of more than 50% for real-world data tests. 展开更多
关键词 speech coding error detection error concealment speech quality global system for mobile (GSM)
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A New Speech Codec Based on ANN with Low Delay
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作者 YANG Zhen (Nanjing University of Posts & Telecommunications, Nanjing 210003, P.R.China) 《The Journal of China Universities of Posts and Telecommunications》 EI CSCD 2002年第4期1-7,共7页
The author designs a new speech codec in this paper, which is based on ANN tocarry out nonlinear prediction . This new codec synthesizes speeches with better quality than theconventional waveform or hybrid codecs does... The author designs a new speech codec in this paper, which is based on ANN tocarry out nonlinear prediction . This new codec synthesizes speeches with better quality than theconventional waveform or hybrid codecs does at the same bit rate. Moreover, the most importantcharacteristic of this codec is the low coding delay, which will benefit the enhancement of thespeech communication QoS when we transmit speech signals in IP or ATM networks. 展开更多
关键词 speech coding low delay ANN nonlinear prediction
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Realtime robust speech communication based on iterative joint source-channel decoding and demodulation algorithm for MELP vocoder
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作者 彭坦 Cui Huijuan Tang Kun 《High Technology Letters》 EI CAS 2010年第2期111-116,共6页
Realtime speech communications require high efficient compression algorithms to encode speech signals. As the compressed speech parameters are highly sensitive to transmission errors, robust source and channel decodin... Realtime speech communications require high efficient compression algorithms to encode speech signals. As the compressed speech parameters are highly sensitive to transmission errors, robust source and channel decoding and demodulation schemes are both important and of practical use. In this paper, an it- erative joint souree-channel decoding and demodulation algorithm is proposed for mixed excited linear pre- diction (MELP) vocoder by both exploiting the residual redundancy and passing soft information through- out the receiver while introducing systematic global iteration process to further enhance the performance. Being fully compatible with existing transmitter structure, the proposed algorithm does not introduce addi- tional bandwidth expansion and transmission delay. Simulations show substantial error correcting perfor- mance and synthesized speech quality improvement over conventional separate designed systems in delay and bandwidth constraint channels by using the joint source-channel decoding and demodulation (JSCCM) algorithm. 展开更多
关键词 speech coding joint souree-channel coding and modulation (JSCCM) iterative decoding
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弹幕语言的语码转换探析——以韩剧TV中的弹幕语言为例
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作者 杨倩 《现代语文》 2024年第5期11-16,共6页
弹幕语言是网络语言新形式之一,存在着形式多样的句内语码转换现象。以韩剧TV中的弹幕语言为例,弹幕语言语码转换的语表形式主要表现在四个方面:方言与普通话的语码转换、英语与普通话的语码转换、韩语与普通话的语码转换、多语间的语... 弹幕语言是网络语言新形式之一,存在着形式多样的句内语码转换现象。以韩剧TV中的弹幕语言为例,弹幕语言语码转换的语表形式主要表现在四个方面:方言与普通话的语码转换、英语与普通话的语码转换、韩语与普通话的语码转换、多语间的语码转换。弹幕语言语码转换的语用功能则表现在七个方面:信息传递功能、顺应功能、评价功能、委婉表达功能、共情功能、戏谑功能、填补功能。韩语影视剧和综艺节目网络言语社区是一个典型的网络言语社区,它可以作为上位概念,包含具有类似功能、表达习惯、心理认同的其他网络语言社区。弹幕语言的语码转换是该社区群体成员身份标记和心理认同的外显形式。通过使用语码转换,该网络言语社区的内部一致性与排他性特点得到凸显。 展开更多
关键词 弹幕语言 语码转换 语表形式 语用功能 言语社区
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基于带阈值的BPE-dropout多任务学习的端到端语音识别
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作者 马建 朵琳 +1 位作者 韦贵香 唐剑 《吉林大学学报(理学版)》 CAS 北大核心 2024年第3期674-682,共9页
针对语音识别任务中出现的未登录词问题,提出一种带阈值的BPE-dropout多任务学习语音识别方法.该方法采用带随机性的字节对编码算法,在形成子词时引入带字数阈值的策略,将子词作为建模单元,编码器部分采用Conformer结构,与链接时序分类... 针对语音识别任务中出现的未登录词问题,提出一种带阈值的BPE-dropout多任务学习语音识别方法.该方法采用带随机性的字节对编码算法,在形成子词时引入带字数阈值的策略,将子词作为建模单元,编码器部分采用Conformer结构,与链接时序分类和注意力机制相结合.为进一步提升模型性能,引入动态参数对损失函数进行动态调节,并同时进行多任务训练和解码.实验结果表明,该方法采用子词作为建模单元可有效解决未登录词问题,在多任务学习框架下进一步提升了模型的识别性能.在公开数据集THCHS30和ST-CMDS上,该模型实现了超过95%的识别准确率. 展开更多
关键词 语音识别 多任务学习 字节对编码 动态调节参数
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A Review of Speech Coding 被引量:3
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作者 Bao ChangchunAssociate professor of Information Engineering, Beijing Polytechnic University, Ph.D, CIE senior member (Department of Electronic Engineering, Beijing Polytechnic University, Beijing 100022) Fan ChangxinProfessor with Xidian University, C 《通信学报》 EI CSCD 北大核心 1998年第5期45-56,共12页
AReviewofSpechCodingBaoChangchun(DepartmentofElectronicEngineering,BeijingPolytechnicUniversity,Beijing10... AReviewofSpechCodingBaoChangchun(DepartmentofElectronicEngineering,BeijingPolytechnicUniversity,Beijing100022)FanChangxin?.. 展开更多
关键词 语音编码 线性估计 综合分析 波形编码
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Noise Feedback Coding Revisited:Refurbished Legacy Codecs and New Coding Models 被引量:2
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作者 Stéphane Ragot Balázs Kvesi Alain Le Guyader 《ZTE Communications》 2012年第2期34-44,共11页
Noise feedback coding (NFC) has attracted renewed interest with the recent standardization of backward-compatible enhancements for ITU-T G.711 and G.722. It has also been revisited with the emergence of proprietary ... Noise feedback coding (NFC) has attracted renewed interest with the recent standardization of backward-compatible enhancements for ITU-T G.711 and G.722. It has also been revisited with the emergence of proprietary speech codecs, such as BV16, BV32, and SILK, that have structures different from CELP coding. In this article, we review NFC and describe a novel coding technique that optimally shapes coding noise in embedded pulse-code modulation (PCM) and embedded adaptive differential PCM (ADPCM). We describe how this new technique was incorporated into the recent ITU-T G.711.1, G.711 App. III, and G.722 Annex B (G.722B) speech-coding standards. 展开更多
关键词 speech coding noise shaping noise feedback coding G.711 G.722
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Lattice Vector Quantization Applied to Speech and Audio Coding 被引量:1
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作者 Minjie Xie 《ZTE Communications》 2012年第2期25-33,共9页
Lattice vector quantization (LVQ) has been used for real-time speech and audio coding systems. Compared with conventional vector quantization, LVQ has two main advantages: It has a simple and fast encoding process,... Lattice vector quantization (LVQ) has been used for real-time speech and audio coding systems. Compared with conventional vector quantization, LVQ has two main advantages: It has a simple and fast encoding process, and it significantly reduces the amount of memory required. Therefore, LVQ is suitable for use in low-complexity speech and audio coding. In this paper, we describe the basic concepts of LVQ and its advantages over conventional vector quantization. We also describe some LVQ techniques that have been used in speech and audio coding standards of international standards developing organizations (SDOs). 展开更多
关键词 Vector quantization lattice vector quantization speech and audio coding transform coding
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Improved Multiple Descriptions Sinusoidal Coder Adaptive to Packet Loss Rate
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作者 郎玥 王晶 +1 位作者 赵胜辉 匡镜明 《Journal of Beijing Institute of Technology》 EI CAS 2008年第2期202-207,共6页
To make the multiple descriptions codec adaptive to the packet loss rate, which can minimize the final distortion, a novel adaptive multiple descriptions sinusoidal coder (AMDSC) is proposed, which is based on a sin... To make the multiple descriptions codec adaptive to the packet loss rate, which can minimize the final distortion, a novel adaptive multiple descriptions sinusoidal coder (AMDSC) is proposed, which is based on a sinusoidal model and a noise model. Firstly, the sinusoidal parameters are extracted in the sinusoidal model, and ordered in a decrease manner. Odd indexed and even indexed parameters are divided into two descriptions. Secondly, the output vector from the noise model is split vector quantized. And the two sub-vectors are placed into two descriptions too. Finally, the number of the extracted parameters and the redundancy between the two descriptions are adjusted according to the packet loss rate of the network. Analytical and experimental resuits show that the proposed AMDSC outperforms existing MD speech coders by taking network loss characteristics into account. Therefore, it is very suitable for unreliable channels 展开更多
关键词 speech coding multiple descriptions coding sinusoidal model
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A New Method of Designing Waveform Codebook
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作者 Zhang Xueying Zhang Gang (TaiYuan University of Technology, TaiYuan 030024) 《通信学报》 EI CSCD 北大核心 1998年第5期93-96,共4页
ANewMethodofDesigningWaveformCodebookZhangXueyingZhangGang(TaiYuanUniversityofTechnology,TaiYuan030024)Abstr... ANewMethodofDesigningWaveformCodebookZhangXueyingZhangGang(TaiYuanUniversityofTechnology,TaiYuan030024)AbstractThecodebooksea... 展开更多
关键词 语音编码 失量量化 编码激励 线性估计 波形编码本
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A Novel Low-bit-rate Speech Coding Based on Decomposition of the Pitch-cycle Waveform of the Linear Predictive Residual
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作者 Bao ChangchunAssociate professor of Information Engineering, Beijing Polytechnic University, Ph.D, CIE senior member (Department of Electronic Engineering, Beijing Polytechnic University, Beijing 100022) Fan ChangxinProfessor of Information Engineerin 《通信学报》 EI CSCD 北大核心 1998年第5期39-44,共6页
ANovelLowbitrateSpechCodingBasedonDecompositionofthePitchcycleWaveformoftheLinearPredictiveResidualBaoCh... ANovelLowbitrateSpechCodingBasedonDecompositionofthePitchcycleWaveformoftheLinearPredictiveResidualBaoChangchun(Departm... 展开更多
关键词 线性估计 语音编码 失量量化 分解 节圈波形
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A FRACTAL INTERPOLATION SPEECH CODING ALGORITHM
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作者 周志杰 胡光锐 《Journal of Shanghai Jiaotong university(Science)》 EI 1997年第1期55-59,共5页
It is supposed that speech is the output of a LPC filter which is excited by LPC residual. Consequently, speech can be reproduced if a signal, which occupies main characteristics of the LPC residual, excites the LPC f... It is supposed that speech is the output of a LPC filter which is excited by LPC residual. Consequently, speech can be reproduced if a signal, which occupies main characteristics of the LPC residual, excites the LPC filter. Based on this hypothesis, a new speech coding algorithm is proposed. Its excitation of synthesizer is the fractal interpolation of down sampled LPC residual with the same fractal dimension of LPC residual. Computer simulation shows that this speech coding algorithm can provide high quality coded speech at bit rate of 6.4 kb/s. Some essential issues are also presented to demonstrate this algorithm such as the calculation of fractal dimension, the implementation of fractal interpolation. 展开更多
关键词 speech CODING FRACTAL CHAOS
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A REAL-TIME IMPLEMENTATION OF 4.2Kb/s CELP SPEECH CODING
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作者 Bao Changchun Dai Yisong Fan Changxin(information Science Institute, Xidian University, Xi’an 710071) (Dept. of Electronic Eng., Jilin University of technology 130025) 《Journal of Electronics(China)》 1997年第1期52-58,共7页
This paper presents a real-time implementation of 4.2Kb/s CELP speech coding on single DSP chip. An algorithm reducing search complexity for adaptive codebook is suggested; the solving method that the parameters are c... This paper presents a real-time implementation of 4.2Kb/s CELP speech coding on single DSP chip. An algorithm reducing search complexity for adaptive codebook is suggested; the solving method that the parameters are changed into LSP parameters is discussed. The realtime implementation process of this coding on a commercial development board with a single TMS320C30 is described. 展开更多
关键词 speech CODING LINEAR prediction VECTOR QUANTIZATION
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A SINGLE PROCESSOR MULTI-RATE VOCODER
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作者 Wang Dusheng Zhang Jiankang Fan Changxin(information Science Institute, Xidian university, Xi’an 710071) 《Journal of Electronics(China)》 1997年第1期59-62,共4页
This paper presents the design of a full-duplex multi-rate vocoder which implements an LPC-10, CELPC and VSELPC algorithms in real time. A single commercially available digital signal processor IC, the TMS320C25, is u... This paper presents the design of a full-duplex multi-rate vocoder which implements an LPC-10, CELPC and VSELPC algorithms in real time. A single commercially available digital signal processor IC, the TMS320C25, is used to perform the digital processing. The channel interfaces are configured with the design of ASIC, and including timing and control logic circuits. 展开更多
关键词 MULTI-RATE VOcodeR speech CODING Digital SIGNAL PROCESSOR
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High Performance Speech Compression System 被引量:6
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作者 Ke Liu, Zhichun Mu, Zhong Wang Information Engineering School, University of Science & Technology Beijing, Beijing 100083, China 《Journal of University of Science and Technology Beijing》 CSCD 2001年第3期229-233,共5页
Since Pulse Code Modulation emerged in 1937, digitized speech has experienced rapid development due to its outstanding voice quality, reliability, robustness and security in communication. But how to reduce channel wi... Since Pulse Code Modulation emerged in 1937, digitized speech has experienced rapid development due to its outstanding voice quality, reliability, robustness and security in communication. But how to reduce channel width without loss of speech quality remains a crucial problem in speech coding theory. A new full-duplex digital speech communication system based on the Vocoder of AMBE-1000(TM) and microcontroller ATMEL 89C51 is introduced. It shows higher voice quality than current mobile phone system with only a quarter of channel width needed for the latter. The prospective areas in which the system can be applied include satellite communication, IP Phone, virtual meeting and the most important, defence industry. 展开更多
关键词 digital signal processing digital speech compression digital communication full-duplex coding rate
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Web Voice Browser Based on an ISLPC Text-to-Speech Algorithm
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作者 LIAO Rikun JI Yuefeng LI Hui 《Wuhan University Journal of Natural Sciences》 CAS 2006年第5期1157-1160,共4页
A kind of Web voice browser based on improved synchronous linear predictive coding (ISLPC) and Text-toSpeech (TTS) algorithm and Internet application was proposed. The paper analyzes the features of TTS system wit... A kind of Web voice browser based on improved synchronous linear predictive coding (ISLPC) and Text-toSpeech (TTS) algorithm and Internet application was proposed. The paper analyzes the features of TTS system with ISLPC speech synthesis and discusses the design and implementation of ISLPC TTS-based Web voice browser. The browser integrates Web technology, Chinese information processing, artificial intelligence and the key technology of Chinese ISLPC speech synthesis. It's a visual and audible web browser that can improve information precision for network users. The evaluation results show that ISLPC-based TTS model has a better performance than other browsers in voice quality and capability of identifying Chinese characters. 展开更多
关键词 improved synchronous linear predictive coding (ISLPC) Text-to-speech (TTS) Web voice browser voice quality
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基于语音识别技术的日语在线考试系统
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作者 牛立保 王振铎 《信息技术》 2023年第8期8-12,共5页
针对高校线上线下混合式教学逐渐普及,而在线考试系统建设滞后的现状,设计开发了一款日语在线考试系统。系统前端使用微信小程序,后端采用SpringMVC框架,MySQL数据库存储业务数据。系统允许教师建立题库、发布试卷和统计成绩,学生能够... 针对高校线上线下混合式教学逐渐普及,而在线考试系统建设滞后的现状,设计开发了一款日语在线考试系统。系统前端使用微信小程序,后端采用SpringMVC框架,MySQL数据库存储业务数据。系统允许教师建立题库、发布试卷和统计成绩,学生能够在线测试、考试及查看成绩。采用自动组卷、考试时间控制技术防止作弊现象。利用二维码技术发布考试链接,系统还使用语音识别技术提供词汇朗读比对功能,帮助学习者矫正发音。该系统使用简单、功能可靠,对学生掌握日语知识点、进行碎片化学习有一定实用性,适应了“互联网+”教育的发展趋势。 展开更多
关键词 日语 在线考试 组卷 语音识别 二维码
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一种基于CVSD编码特征的失步检测方法
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作者 廖蓉晖 许志强 张鹤鸣 《通信技术》 2023年第12期1359-1363,共5页
首先,介绍了连续可变斜率增量调制(Continuously Variable Slope Delta,CVSD)的语音编码工作原理,从编码特征的角度对CVSD算法进行了研究;其次,通过对自然界各类声音信号的仿真分析提出CVSD算法的特征码型;最后,通过对编码数据的分析对... 首先,介绍了连续可变斜率增量调制(Continuously Variable Slope Delta,CVSD)的语音编码工作原理,从编码特征的角度对CVSD算法进行了研究;其次,通过对自然界各类声音信号的仿真分析提出CVSD算法的特征码型;最后,通过对编码数据的分析对比可知,所提方法可在无须插入同步码组、不占用冗余信道资源的情况下实现通信同失步状态检测,并具备较好的抗信道误码能力。该方法的思路新颖、算法简单,无论采用软件、硬件均可实现,具有较好的推广应用价值。 展开更多
关键词 连续可变斜率增量调制 脉冲编码调制 语音编码 无线语音系统 编码特征
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基于DNA编码和混沌系统的语音加密算法 被引量:1
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作者 饶攀 张烨 《南昌大学学报(理科版)》 CAS 北大核心 2023年第1期85-94,共10页
提出了一种基于DNA编码和混沌系统的语音加密算法。该算法首先将原始语音分成两个部分,利用Lorenz混沌系统生成的混沌序列,分别对第一部分语音做动态DNA编码和异或运算,对第二部分语音做异或运算,以实现语音数据的扩散;其次利用Logisti... 提出了一种基于DNA编码和混沌系统的语音加密算法。该算法首先将原始语音分成两个部分,利用Lorenz混沌系统生成的混沌序列,分别对第一部分语音做动态DNA编码和异或运算,对第二部分语音做异或运算,以实现语音数据的扩散;其次利用Logistic混沌系统生成的混沌序列将扩散后的语音索引置乱,再对置乱后语音的极性置乱,使得样本值的直方图的分布更为均匀。仿真结果和理论分析表明,该加密算法具有密钥空间大、密钥灵敏度高、抗穷举攻击、统计攻击和选择明文攻击能力强等优点。 展开更多
关键词 语音加密 DNA编码 混沌系统 置乱
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