Nuance owns key pieces of technology for entering information onto mobile phones via both voice and touch (including the T9 text prediction algorithm used on most feature phones).
Nuance则拥有手机信息输入方面在语音及触摸方式上的关键技术(包括大多数功能型手机使用的T9文本预测算法)。
Voice conversion experiments are performed to evaluate the tone codebook mapping algorithm.
声音转换实验评估了声调码本映射算法的性能。
This paper describes a DSP system which based on multi-band excitation voice compression coding algorithm - multi-rate vocoder.
论文设计了一个基于多带激励语音压缩编码算法的DSP系统-多速率声码器。
This paper mainly focuses on the research and improvement of the fast code word search algorithm, speech enhancement and voice activity detector key technologies.
本文主要对语音编码中矢量量化码字搜索算法、语音增强、语音激活检测技术进行研究并进行了改进。
On the base of LF-4 derivative glottal flow, an improved algorithm of LF-4 derivative model is proposed to synthesize voice source.
在LF - 4微分声门波模型的基础上，提出了一种LF - 4模型的改进算法，进行嗓音源的合成。
A robust voice command recognition based on SDCN algorithm was presented.
提出了一种基于SDCN算法的鲁棒性语音命令识别。
The technology of voice transformation was introduced. The algorithm of voice transformation based on harmonic sinusoidal modal was analyzed.
介绍了语音变换的相关技术，分析了利用正弦谐波模型实现语音变换的算法及流程。
The characteristics of digital voice short energy and ZRC is studied, and a new voice activity detection algorithm based on finite state machine is presented.
研究了数字语音短时能量和过零率特点，提出了基于有限状态机的端点检测新算法。
This system realized a voice communication system based on the AMBE2000, this chip USES the MBE algorithm to obtain low changeable communication rate, and high speech quality.
本次系统实现了基于AMBE2000的低速率语音通信，该芯片采用改进的多带激励(MBE)算法，能实现可变速率低比特率、高语音音质的语音压缩编码。
In order to improve the robustness of voice activity detection (VAD), the use of an algorithm based on complex Gaussian mixture model under nonstationary noisy environments was presented.
针对语音激活检测的鲁棒性问题，提出在非平稳噪声环境下使用基于复高斯混合模型的鲁棒语音激活检测算法。
According to the characteristic of discontinuous voice signal, this compression algorithm provides the function of voice active detection and improves the compress ratio of voice signal.
同时，根据语音信号不连续的特点，压缩算法具有静音识别功能，进一步提高了语音信号的压缩率。
A new algorithm based on the energy and discrimination information was developed to improve the performance of the voice activity detection system in real-time speech communications.
为提高实时通信中语音端点检测系统的性能，提出了一种基于能量和鉴别信息的端点检测算法。
This paper does analysis and debugging of the algorithm software and build up the platform of collection and analysis of voice data.
进行了该算法软件的分析、调试，建立了语音数据的采集、分析平台。
Based on the feasibility study and the analysis to the encryption algorithm, the paper presents a method of the real-time multi-point voice communication in network, which USES ADPCM and UDP.
在对加密算法实现过程进行可行性研究和分析的基础上，阐述了应用adpcm编码调制技术和UDP协议进行网络环境下实时多点语音通信的方法。
Blind signal separation system can capture the true voice signals and use a different algorithm to separate.
盲信号分离系统可以采集真实语音信号，然后使用不同的盲信号分离算法分离。
Studies on the problems of playing digitized voice at a reduced speed while reserving its origingal tunes has been made, and an interpolation algorithm is presented.
对数字化语音减速保调播放问题进行了研究并提出了一种插值算法。
This paper present a novel kind of voice hiding algorithm based on quantization coding.
本文提出了一种基于量化编码技术的语音隐藏算法。
By discussing the existing voice and audio encoding technology, the thesis designs an CNG algorithm which is based on the mask effect theory.
本文在充分讨论现有语音及音频编码技术的基础上，提出了一种基于掩蔽效应的CNG技术，并实现了该算法。
In this paper, an improved voice quality enhancement algorithm based on traditional spectrum subtraction method is discussed.
讨论了一种基于传统谱相减算法的改进方法。
ICON USES an algorithm that digitally fuses together the best audio from its two electret microphones and its patented Voice Activity Sensor to create a single, natural sounding speech signal.
图标使用了一个算法，数字融合了来自两个驻极体麦克风和声音传感器专利的最佳音频创建一个单独的，自然的声音的语音信号。
In this paper, we suggest a new near -end voice detection method, which is based on maximum length correlation algorithm.
本文提出了一种基于最大长度序列相关算法的近端语音检测算法。
Here, an adaptive jitter buffering algorithm is presented to promote the quality of voice communication with spike delay.
对于这个问题，提出针对突发大时延存在下的自适应语音缓冲算法。
The project do some work in the improving algorithm of voice recognition, design of system of voice recognition, hardware design and software programming.
本课题进行了语音识别系统设计、算法改进、硬件电路设计和软件编写。
A voice conversion approach with a sinusoidal plus noise model is introduced and a parametric conversion algorithm based on phoneme segments is discussed in this paper.
提出一种基于正弦加噪声模型的说话人转换方法，着重讨论通过修改音素段内的声学参数实现说话人的转换。
The key for these problems is to used quit good and efficient voice coding algorithm.
解决这一问题的关键是采用一种有效的数字语音编码压缩技术。
To detect the voice change of speakers, an improved BIC algorithm is proposed in this paper.
该文针对多人说话改变点检测问题，提出一种新的改进型bic话者改变检测算法。
The final simulation results on recorded multi-voice-sources in the experimental environment show that this algorithm is also feasible in the actual noise environment.
最后用实测数据进行仿真，仿真结果表明此算法在实际噪声环境中也是可行的。
The advantage of the improved algorithm will be more obvious with the increasing number of the voice signal to be recognized.
随着待识别语音信号数量的增多，该算法优势更加明显。
The advantage of the improved algorithm will be more obvious with the increasing number of the voice signal to be recognized.
随着待识别语音信号数量的增多，该算法优势更加明显。
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