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在matlab环境下的详细的turbo的编码和解码过程，包括循环编码过程...
在matlab环境下的详细的turbo的编码和解码过程，包括循环编码过程-In the matlab environment detailed turbo encoding and decoding process, including the cycle of coding process
- 2022-07-08 19:11:10下载
- 积分：1
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txt2wav.rar文本语音（Text
txt2wav.rar文本语音（Text-to-Speech，以下简称TTS），它的作用就是把通过TTS引擎把文本转化为语音输出。代码示范了如何运用Microsoft Speech SDK 建立自己的文本语音转换应用程序。应该事先下载微软的语音SDK里面含微软的语音开发工具原来是sapi5.1现在好像升级了微软的MSDN上有下-txt2wav.rar text pronunciation (Text-to-Speech, hereafter
refers to as TTS), its function is transforms through the TTS engine
the text as the pronunciation output. How did the code demonstrate has
established own using Microsoft Speech SDK text pronunciation
transformation application procedure. Should beforehand download
Microsoft inside pronunciation SDK to contain Microsoft the
pronunciation development kit originally is sapi5.1 now looks like
promotes on Microsoft s MSDN to have next
- 2022-05-16 09:16:38下载
- 积分：1
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matlab中的wave文件添加功能，欢迎使用
matlab中的wave文件添加功能，欢迎使用-Matlab the paper added wave function, welcomed the use of
- 2023-04-28 20:10:03下载
- 积分：1
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directshow 实现wave文件的播放
directshow 实现wave文件的播放
-directshow achieve wave broadcast documents
- 2022-07-04 06:33:35下载
- 积分：1
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ITU
ITU-T G.729B标准源代码解码端，已创建工程文件，方便初学者，已测试，运行良好-ITU-T G.729B standard decoder source code, project files have been created for beginners, have been tested, well-functioning
- 2023-05-03 18:15:03下载
- 积分：1
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可以讲mulaw语音数据转换成线性pcm数据
可以讲mulaw语音数据转换成线性pcm数据-can speak mulaw voice data converted into linear data pcm
- 2022-08-12 12:45:30下载
- 积分：1
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LMS filter (minimum mean square error filter), learning modern digital signal pr...
LMS滤波器（最小均方误差滤波器），学习现代数字信号处理理论的人应该用的上。-LMS filter (minimum mean square error filter), learning modern digital signal processing theory should be used on.
- 2022-08-06 03:50:26下载
- 积分：1
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数字音频格式有很多种，其质量与采样频率和采样精度两个参数有关。频率的表示单位为赫兹〔Hz〕，它表示每秒采样次数。采样频率越高，音质就越好。采样精度为每次采样所存...
数字音频格式有很多种，其质量与采样频率和采样精度两个参数有关。频率的表示单位为赫兹〔Hz〕，它表示每秒采样次数。采样频率越高，音质就越好。采样精度为每次采样所存储的数据数量，它决定每个数字信号所能够表示的离散振幅的数量。存储每个样本的数据越多，音质就越好。但是高品质的声音需要占用大量的内存和磁盘空间。考虑到网络带宽，在Internet连接上传输就需要花费很长的时间。对于Applet来说，保证声音文件的最小化是极为重要的。-digital audio format there are many, the quality and frequency of sampling and sampling accuracy of the two parameters. The frequency of Hertz unit of [Hz], it said the number of samples per second. The higher sampling frequency, the sound quality better. For each sampling precision sampling by the amount of data storage, it was decided that each digital signal can be expressed as the number of discrete amplitude. Each sample storage of more data, better sound quality. But high-quality voice would take a lot of memory and disk space. Consideration of the network bandwidth, the transmission on the Internet connection on the need to spend a very long time. Applet for instance, to ensure that voices of the smallest document is very important.
- 2022-08-08 10:30:29下载
- 积分：1
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在语音中提取基音频率的程序。输入为.wav格式的语音文件，输出各帧基音频率。...
在语音中提取基音频率的程序。输入为.wav格式的语音文件，输出各帧基音频率。-This is a project of pitch extract of voice. When input is a wave file(.wav), the output will be the fundamental frequency of each frame.
- 2023-05-09 09:05:03下载
- 积分：1
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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...
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.
- 2023-04-29 12:00:03下载
- 积分：1
