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melppp
- 在语音编码算法中,混和激励线性预测(MELP)算法因为能更好的模拟自然语言特征,在低速率上能合成较高质量的语音,而成为现代低速率语音编码中最有潜力的算法之一。-In the speech coding algorithm, the mixed excitation linear prediction (MELP) algorithm because it can better simulate the natural language features can be synthesized at
Efficient-LSP-quantization-algorithm
- 为在极低速率下实现高质量的语音编码,提出了一 种新的有效的线谱对( LSP)参数量化算法——P-RS- MSM Q 算法。此算法以多帧联合矩阵量化作为基本框架,引入了基 于超级帧模式的均值去除和帧间预测策略、 矩阵分裂和子矩 阵多级量化策略 同时提出了基于语音帧短时谱能量的帧 内加权和基于超级帧中各子帧重要性的帧间加权策略等。-Very low rates in high-quality speech coding, a new and effective LSP (LSP)
gettingStartedV40_006b
- straight是一个高质量的语音识别、转换、合成工具，构建于matlab编程工具。-STRAIGHT[1] is a high-quality speech analysis, modification and synthesis system based on a simple channel VOCODER concept (in other words a source filter model).
SP-ENH-DOS-BUILD
- WAV Speech Enhancer can be used to improve the signal to noise ratio of bad quality speech recordings: - Dynamic expansion - Pink noise attenuation - Low frequency noise (50-60Hz) suppression-WAV Speech Enhancer can be used to improve the signal to n
xiaobo
- 利用小波变换中的阈值方法对含噪信号作消噪处理，获得语音增强的效果，提高语音的质量-Threshold in the wavelet transform to the noisy signal de-noising processing, speech enhancement effect and improve the quality of voice
psola
- tpdsola algorithm,Synthesized speech can be produced by several different methods. All of these have some benefits and deficiencies that are discussed in this and previous chapters. The methods are usually classified into three groups: Articula
prioriposterioripowersubtractionalgo
- Algorithm to improve the quality of speech signals by improving the SNR
sinusoidal-model
- 对语音信号进行分析仿真，合成声音轻微的音质下降-The speech signal analysis and simulation, a slight loss of quality of the synthesized voice
Tchxuziph
- 本程序用于测量增强语音与纯净语音的对数谱距离，用于语音增强。对数谱距离越小小，表明增强语音与纯净语音越接近，增强语音质量越好；反之，则增强语音质量越差。 -The procedure used to measure to enhance the voice and the pure voice of the logarithmic spectral distance for speech enhancement. More small log spectral distance, the c
P862
- p.862是ITU标准提案，主要是用来对窄带电话网的端到端语音质量和语音编码器解码的语音质量进行评估。中有p.862的标准文档和实现软件。-p.862 is ITU standard proposal, is mainly used to evaluate the narrowband telephone network and the end-to-end voice quality and voice quality of the decoded speech encoder. P.862
New-Folder-(4)
- Low bit-rate speech coders are important in many critical communication systems such as military, police, fire departments, homeland security and other first responders. The need for elimination, reduction of the redundancy or irrelevant information
speech_voice
- 用matlab实现语音合成，并修改基频、lpc、lsp参数，使合成质量有所提高-Using matlab speech synthesis, and modify the baseband, lpc, lsp parameters, the quality has improved synthesis
FPTTS_demo
- 语音合成，又称文语转换（Text to Speech）技术，能将任意文字信息实时转化为标准流畅的语音朗读出来，相当于给机器装上了人工嘴巴。 我们的实验室实现的英语语音合成技术音质好，自然度高，可懂度高。语音合成并且整个系统基于C开发，可以很方便的移植到其他移动平台。下一步我们会开发中文语音合成技术，以及在 提高合成语音的自然度、丰富合成语音的表现力、降低语音合成技术的复杂度、多语种文语合成方面做进一步的研究。-Speech synthesis, also known as TTS (Text
FoxPigeonHMMTTS
- Fox & Pigeon Laboratory 发布一款基于隐马尔可夫模型的英语语音合成软件（HMM based English Text-To-Speech, TTS）：FoxPigeonTTS alpha版。 近年来，基于HMM的语音合成系统得到广泛的重视和应用。我们实验室实现的基于HMM的语音合成系统基本不需要任何语言学知识指导系统训练，构建时间短，构建过程基本不需要人工干预，而由于系统属于参数化合成方法，系统的合成结果灵活多变，可以很容易的应用于多个发音人，多种发音风格，多种
s16p06
- power quality monitoring has advanced to an important tool for system evaluation.the increased amount of recorded data requires more sophisticated analysis methods. the paper describes the development and operation of a system for automated classific
AMBE-3000R_manual
- AMBE-3000R_manual.rar，音频芯片，做音频方面可以参考-Digital Voice System s AMBE-3000™ Vocoder Chip is an extremely flexible, voice compression solution that sets a new standard for quality, high-performance speech compression hardware. The AMBE-3000™ Vo
ch6example10
- pcm编解码模型，测试指定误码条件下PCM解码语音信号的音质和可懂度-pcm codec model, sound quality and intelligibility test conditions specified BER PCM decoded speech signal
TotalCommanderPortable
- 客观语音质量评价，对于合成语音检测效果显著，对原始算法进行约减了。-Objective voice quality uation, for the synthesis of speech detection significant effect on the Reduction of the original algorithm.
ZNJJ
- 控制核心采用8位廉价51系列单片机完成系统信号分析、密码管理、人机对话以及相应控制。 2. 采用DTMF专用芯片集解码的MT8870芯片进行DTMF信号的解码。 3. 语言提示电路：电话远程控制系统利用语音进行人机交流。语音提示电路采用美国ISD公司出品的新型600段单片优质语音录放电路ISD2500，完成语言提示部分电路设计 。 -The control core uses 8 bit cheap 51 Series MCU to complete the system sign
Remove-noise
- 在现场直播节目的录制过程中，经常会由于现场录制工作人员的操作不当，会使MIC中串入交流干扰，从而造成语音信号的采集过程中的噪声干扰。语音降噪是针对噪声问题的一种有效的处理技术，它的目的就是为了消除噪声的影响，提高语音清晰度、改善语音的质量。-During recording, live programs, often as a result of improper handling live recording of staff, will MIC in series AC interferen