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语音信号的频域处理,语音虽然是一个时变、非平稳的随机过程。但在短时间内可近似看作是平稳的。因此如果能从带噪语音的短时谱中估计出“纯净”语音的短时谱，即可达到语音增强的目的。由于噪声也是随机过程，因此这种估计只能建立在统计模型基础上。利用人耳感知对语音频谱分量的相位不敏感的特性，这类语音增强算法主要针对短时谱的幅度估计。 -voice signals in the frequency domain processing, voice is a time-varying, nonstationary random process. But in a short period of time can be approximated as smooth. So if Noisy Speech from the short-term spectrum estimate "pure" voice of the short-term spectrum, and reached speech enhancement purposes. As the noise is random process, which can only be estimated based on statistical models based on. Use ear perception of voice spectrum component of the phase sensitive to the characteristics of such speech enhancement algorithms targeted at the rate of short-term spectral estimation.

此文件是好将声音发送到 dsk 6700 和处理的任何声音。您可以配置此代码由 dsk 海缆无源探测格式然后通过调整可以将声音发送到 dsk 的 aic23 编解码器寄存器的端口处理它，然后回 aic23 和 dsk 的输出中听到它。它是任何一个想要使用 dsk 来学习如何初始化 dsk 的好榜样。

tone encode and decode

这里有fft,ifft,conj,xcorr,resample的用法。-Here are fft, ifft, conj, xcorr, resample usage.

语音识别程序，通过麦克风输入语音信号，系统会识别语音，并跟读。-speech recognition process, through the microphone input speech signal, the system will identify voice, and impersonally.

自已编写的语音信号端点检测程序，采用短时能量与短时过零率的方法。-authorship of the speech signal endpoint detection procedures, using short-term over short-term energy and the rate of zero.

基于最大信噪比的语音分离程序，有需要的朋友可以借鉴-Based on the maximum signal to noise ratio of the speech separation process, there is a need of a friend can learn from

G711语音压缩源码-G711 voice compression FOSS

这是一个基于C++的语音识别系统,包括语音输入,识别,输出等功能-This is a C-based speech recognition systems, including voice input and recognition, output function

LPC10 2.4Kpbs 语音压缩定点运算C语言源程序OPENLPC-Voice compression LPC10 2.4Kpbs computing fixed-point C language source code OPENLPC