搜索资源列表
LMS_speech_enhancement
- 运用自适应滤波算法实现的语音增强Matlab程序，附带带噪语音，在M文件中自己修改文件名即可使用-The use of adaptive filtering algorithm for speech enhancement Matlab program, with noisy speech, changes in the M file, the file name to use their own
FIR
- 基于线性预测FIR自适应滤波器的语音增强算法-FIR adaptive filter based on linear prediction algorithm for speech enhancement
duixiao
- 本自适应算法，采用RLS算法实现自适应噪声对消。使得语音增强-The adaptive algorithm, RLS algorithm using adaptive noise cancellation. Making the speech enhancement
adaptse1
- 利用MATLAB实现变长自适应滤波，实现语音增强的功能-MATLAB realize the use of variable length adaptive filtering, realize the function of speech enhancement
DNLMSspeechenu
- 双通道语音增强算法，消除环境噪声。采用归一化自适应方法，噪噪声抵消10dB，语音保持较好可懂度。-The dual-channel speech enhancement algorithm to eliminate ambient noise. Using normalized adaptive noise noise offset by 10dB, and the voice to maintain good intelligibility.
GSC
- 采用广义旁瓣抵消（GSC）自适应波束形成方法实现时域和频域滤波，采用LMS自适应算法，最终实现语音增强。（文件中包含纯净语音及不同信噪比的带噪语音）-Generalized sidelobe canceller (GSC) adaptive beamforming method to achieve time-domain filtering using the LMS adaptive algorithm, and ultimately the speech enhancement
Adaptive-Filters
- 该代码是在dsp上实现自适应滤波的语音增强算法-The code is on the DSP to achieve adaptive filter speech enhancement algorithm
Wiener
- 实现维纳滤波用于语音增强。第一步采用频域谱函数计算最佳的维纳滤波器，再对输入带噪信号进行滤波；第二步是在此基础上的改进，用了自适应的思想，及时根据当前帧的状态更新修正滤波器的系统函数，得到更精确的无噪语音信号。第三步是采用自适应LMS方法，在时域迭代计算滤波器系数，迭代次数足够多时可得到较准确的有用信号。-Implement wiener filtering used in speech enhancement. First step optimum wiener filter is obtai
spectral_subtractive
- 谱减法语音增强，包含基本谱减法，多带谱减法，以及自适应谱减法。-Spectral subtraction speech enhancement, contains the basic spectral subtraction, multi-band spectral subtraction, and adaptive spectral subtraction
paper1
- 基于RLS的自适应阵列抗交叉串扰语音增强研究_陈紫强-RLS adaptive array based anti-crosstalk Speech Enhancement _ Chen Zijiang
ss_rdc
- 语音增强，采用自适应增益平均和低延时卷积的谱减，函数可直接调用，简单易行-Speech enhancement, the use of adaptive gain average and low delay convolution spectrum subtraction, the function can be called directly, simple and easy to use
vad
- 关于端点检测的几种方法，语音样本是自己录制的，对传统算法做了一些改进，加入了去噪，去噪之后再进行端点检测，均调通 vad0303：自己设置调整门限为一定值 vad0310：根据能量值和过零率设置门限，自适应门限值 vad0310_2：基于比例因子的门限自调整 vad0310.m加入了噪声，端点检测前都噪声进行了处理 entropy.m：基于自适应子带频谱熵的稳健性语音端点检测 可用于语音增强及端点检测 dbdoor.m：双门限算法，用于语音端点检测。可以通过调整门限值，并