Voice activity detection with noise reduction and long-term spectral divergence estimation
噪声估计算法.doc
太原理工大学毕业设计(论文)任务书毕业设计(论文)题目;2004年Rangachari和Loizou提出了一种快速估讣方法,不仅使得带噪语音子带中语音出现概率讣算更准确,而且噪声谱的更新在连续时间内不依赖固怎时间的窗长,但是在语音或噪声能量过高时噪声的估计就会慢下来,而且如果时间大于0.5s时,就会削弱一些语音能量。
因此,噪声估计算法有待更进一步的改进。
2009年余力,陈颖琪提出了一种基于DCT变换的自适应噪声估计算法,采用DCT系数作为块均匀度的度量,较好地适应了高低噪声的情况,算法复杂度不大,能适用于各种实时图像视频处理系统。
理论分析及实验结果表明本算法不仅在低噪声的图片中表现出良好的性能,而且在高噪声的图片中依旧有效。
此外,还能适应不同质量的图像。
通过对基于最小统讣量的噪声估计方法和改进的最小统计量控制递归平均噪声估计算法研究发现这些噪声估计方法可以在语音存在段进行噪声估计,能够有效地跟踪非平稳噪声。
但是,这些算法在各个频带进行噪声估计,算法复杂度髙,噪声估计方差大。
于是在考虑各频带间的相关性上提岀了在巴克域进行噪声估计,减小了噪声估il•方差,提髙了噪声估计的准确性,并极大地减小算法运算量和存储量。
而且,在巴克域进行噪声估计更符合人耳听觉特性,增强语音具有更好的质量。
英他类似的方法还有低能量包络跟踪和基于分位数的估计方法,后者噪声的估计是基于带噪语音未平滑功率谱的分位数,而不是提取平滑功率谱的最小值,但是此方法il•算复杂度很高,且需要很大的内存来存储过去的功率谱值。
毕业设计(论文)主要内容:本文分为五章,具体章节内容如下:第一章绪论。
噪声估计算法研究的目的和意义及国内外研究的现状。
第二章几种经典的噪声估计的算法。
Martin的最小统计量的估计算法,Cohen和Berdu旳提出的基于最小统计量控制递归平均算法,通过比较最终提出改进的最小统计量控制递归平均算法,仿真结果表明,这种方法在非平稳噪声条件下,也具有较好的噪声跟踪能力和较小的噪声估计误差,可以有效地提髙语音增强系统的性能。
语音信号当中降噪算法的实现方法
语音信号当中降噪算法的实现方法1.语音信号的降噪算法可以通过滤波器来实现。
The noise reduction algorithm of speech signals can be implemented through filters.2.降噪算法可以利用数字信号处理技术进行实现。
The noise reduction algorithm can be implemented using digital signal processing techniques.3.常见的降噪算法包括中值滤波和小波变换。
Common noise reduction algorithms include median filtering and wavelet transforms.4.中值滤波是一种简单且有效的降噪技术。
Median filtering is a simple and effective noise reduction technique.5.小波变换可以将信号分解成不同频率的子信号进行处理。
Wavelet transform can decompose the signal into sub-signals of different frequencies for processing.6.降噪算法的实现需要考虑运算速度和处理效果的平衡。
The implementation of noise reduction algorithm needs to consider the balance between computational speed and processing effect.7.降噪算法的性能评价可以使用信噪比等指标进行量化。
The performance evaluation of noise reduction algorithm can be quantified using metrics such as signal-to-noise ratio.8.自适应滤波是一种根据信号特性进行动态调整的降噪技术。
华为IAD配置
介绍IAD101S/101/102/104H 设备的命令行使用方法,以及常用配置操作。
登陆配置环境Telnet到待配置的IAD101S/101/102/104H 设备。
此时出现如下界面,并提示输入用户名和密码。
设备默认有一个超级管理者级别的用户,用户名为root,密码为admin。
屏幕上显示“TERMINAL>”的提示符,表示登录系统成功。
IAD101S/101/102/104H 命令行提示符由两部分组成:“固定字符串+命令模式标识符”。
命令模式标识符表明命令当前所在模式,如“>”表示普通用户模式,“#””表示特权模式。
固定字符串可以在特权模式下使用hostname 配置,系统缺省的固定字符为“TERMINAL”。
例如:将系统缺省的固定主机名修改为iad101h,操作如下。
TERMINAL#hostname iad101hiad101h#命令执行成功后,主机名已经更改为iad101h。
IAD101S/101/102/104H 的命令模式包括:普通用户模式、特权模式、全局配置模式、高级配置模式和以太网交换机模式。
特权模式和全局配置模式具有向下兼容的特点:特权模式下,可执行普通用户模式中的所有命令。
全局配置模式下,可执行普通用户模式和特权模式中的所有命令。
2常用命令IAD101S/101/102/104H常用命令描述如下:进入特权配置模式TERMINAL>enable进入全局配置模式TERMINAL#configure terminal进入以太网交换机配置模式TERMINAL(config)#lanswitch退出当前命令模式进入用户模式disable(任意模式)退出当前命令模式,进入上一级命令模式,也可以退出配置环境exit(任意模式)立即保存TERMINAL #write设置主机名TERMINAL#hostname显示设备版本TERMINAL>display version更改用户密码TERMINAL(config)#terminal user设置本机系统时间TERMINAL#time设置设备所在地域信息TERMINAL(config)# region { CN | HK |BR | EG | GB | CL | SG | US }设置命令行超时时间间隔TERMINAL#terminal timeout注意:为防止系统复位或掉电等意外情况导致数据丢失,请及时使用write 命令保存配置数据。
语音质量评估
语⾳质量评估语⾳质量评估,就是通过⼈类或⾃动化的⽅法评价语⾳质量。
在实践中,有很多主观和客观的⽅法评价语⾳质量。
主观⽅法就是通过⼈类对语⾳进⾏打分,⽐如MOS、CMOS和ABX Test。
客观⽅法即是通过算法评测语⾳质量,在实时语⾳通话领域,这⼀问题研究较多,出现了诸如如PESQ和P.563这样的有参考和⽆参考的语⾳质量评价标准。
在语⾳合成领域,研究的⽐较少,论⽂中常常通过展⽰频谱细节,计算MCD(mel cepstral distortion)等⽅法作为客观评价。
所谓有参考和⽆参考质量评估,取决于该⽅法是否需要标准信号。
有参考除了待评测信号,还需要⼀个⾳质优异的,没有损伤的参考信号;⽽⽆参考则不需要,直接根据待评估信号,给出质量评分。
近些年也出现了MOSNet等基于深度⽹络的⾃动语⾳质量评估⽅法。
语⾳质量评测⽅法以下简单总结常⽤的语⾳质量评测⽅法。
主观评价:MOS[1], CMOS, ABX Test客观评价有参考质量评估(intrusive method):ITU-T P.861(MNB), ITU-T P.862(PESQ)[2], ITU-T P.863(POLQA)[3], STOI[4], BSSEval[5]⽆参考质量评估(non-intrusive method)传统⽅法基于信号:ITU-T P.563[6], ANIQUE+[7]基于参数:ITU-T G.107(E-Model)[8]基于深度学习的⽅法:AutoMOS[9], QualityNet[10], NISQA[11], MOSNet[12]此外,有部分的⽅法,其代码已开源::该仓库包括MOSNet, SRMR, BSSEval, PESQ, STOI的开源实现和对应的源仓库地址。
ITU组织已公布⾃⼰实现的P.563: 。
GitHub上⾯的微⼩修改版使其能够在Mac上编译。
在语⾳合成中会⽤到的计算MCD:此外,有⼀本书⽤来具体叙述评价语⾳质量:Quality of Synthetic Speech: Perceptual Dimensions, Influencing Factors, and Instrumental Assessment (T-Labs Series in Telecommunication Services)[13]。
多通道语音增强方法简介
多通道语音增强方法简介【摘要】由于多麦克风越来越多地部署到同一个设备上,基于双麦克风和麦克风阵列的多通道语音增强研究有了较大的应用价值。
介绍了自适应噪声对消法、FDM等双通道语音增强方法和波束形成、独立分量分析等麦克风阵列语音增强方法,对各个方法的原理、发展和优缺点进行了详细分析和总结,对多通道语音增强深入研究有一定帮助。
【关键词】语音增强;双通道;麦克风阵列;波束形成1.引言语音是人们通讯交流的主要方式之一。
我们生活的环境中不可避免地存在着噪声,混入噪声的语音会使人的听觉感受变得糟糕,甚至影响人对语音的理解。
在语音编码、语音识别、说话人识别等系统中,噪声也会严重影响应用的效果。
语音增强成为研究的一个问题,其模型如图1所示。
图1 语音增强模型按照采集信号的麦克风数量分类,语音增强方法可被分为单通道(single channel)、双通道(dual-channel)、麦克风阵列(microphone array)三种类型。
一般来说,麦克风越多,去噪的效果越好。
早期,大部分通信/录音终端都只配有一个麦克风,因此单通道语音增强吸引了大量研究者的目光,方法较为成熟。
但单通道方法的缺点是缺少参考信号,噪声估计难度大,增强效果受到限制。
近年来随着麦克风设备的小型化和成本的降低,双麦克风和麦克风阵列越来越多地被部署。
研究者的注意力也在从单通道语音增强向双通道和麦克风阵列语音增强转移,这里对已有的多通道语音增强算法作以简单介绍。
2.双通道语音增强方法在语音增强中,一个关键的问题就是获得噪声。
在单通道语音增强中,噪声是通过从带噪语音信号中估计得到的,估计算法较为复杂且估计噪声总是与真实噪声存在差异,这就限制了增强效果的提高。
为了获得真实噪声,简单的做法就是增加一个麦克风来采集噪声。
从带噪语音信号中减去采集噪声来得到语音信号,这种方法叫做自适应噪声对消法(ANC,adaptive noise canceling),是最原始的最简单的双通道语音增强算法。
IP Camera噪声抑制,回声消除方案--总结
方案二:寻求现成的已经裁剪语音处理模块代码,进行移植。 (1)优点:移植时所需改动较少,遇到的编译和链接问题相对较少。 (2)缺点:项目代码不是最新,可能会遇到在最新代码上已经被解决的问 题。也无法使用新功能。
最终我选择了方案二。
移植--代码下载
代码下载路径:/software/pulseaudio/webrtc-audioprocessing/ 代码结构如下:
编写测试程序--LD_LIBRARY_PATH
如果使用静态链接库libwebrtc_audio_processing.a,不需要指定。 如果使用动态链接库libwebrtc_audio_processing.so,则需要指定程序执行 时,动态链接库的查找路径。
可通过“读取任意pcm文件,测试语音降噪功能”来测试程序是否可以正 常执行。
关于stream_delay的说明(1)
This must be called if and only if echo processing is enabled. Sets the |delay| in ms between AnalyzeReverseStream() receiving a far-end frame and ProcessStream() receiving a near-end frame containing the corresponding echo. On the client-side this can be expressed as
delay = (t_render - t_analyze) + (t_process - t_capture)
- t_analyze is the time a frame is passed to AnalyzeReverseStream() - t_render is the time the first sample of the same frame is rendered by the audio hardware. - t_capture is the time the first sample of a frame is captured by the audio hardware. - t_process is the time the same frame is passed to ProcessStream().
Poly CCX 350 Microsoft Teams集成IP办公电话说明书
POLY CCX 350 - ENTRY-LEVEL IP DESK PHONEGet everything you want from a desk phone: The CCX 350 Microsoft Teams-integrated IP phone offers industry-leading HD audio, giving you incredible value at a great price. Shut out distracting background noise with our exclusive Acoustic Fence and NoiseBlockAI technologies. Durable construction and seamless keypad design make this a phone for any environment. It’s even an advanced speakerphone with world-class echo cancellation to keep your conversations on track. Stylish enough to brighten up any desktop or wall, versatile enough for any office or home office, and feature-rich enough to surpass far more expensive models, it offers enormous value in a slim, streamlined package.• Poly HD Voice with Acoustic Clarity technologies. • Traditional dial pad experience.• Poly Acoustic Fence and NoiseBlockAI technologies.• Full-duplex speakerphone operation and echo cancellation.• Full Microsoft Teams integration on a dial-pad phone.A SMALL-BUSINESS PHONE THAT OFFERS BIG-BUSINESS VALUEBENEFITS• Great for Microsoft Teams users who want (or need) a dial pad and full MSFT integration.• Ideal for glove-hand use in retail, warehouse, and manufacturing locations, thanks to durable design and debris resistance.• Easier telephony deployment and IT support with Poly Lens robust provisioning and management.• Make any reception area look world-class with next-gen modern phone design.• Easy to place anywhere with a two-position stand and optional wall mount (sold separately).POLY CCX 350 IP PHONEUSER INTERFACE FEATURES• 2.8” color LCD screen(640 x 480 pixels)• Dedicated Microsoft Teams button• Rugged surfaces to aid in durability and debris resistance• Voicemail support• One USB-C port (2.0 compliant)for media and storage applications• Adjustable font size selection (regular, medium, large)• Normal and dark mode• Multilingual user interface including Arabic, Chinese, Czech, Danish, Dutch, English (Canada/ US/UK), French, German, Hungarian, Italian, Japanese, Korean, Norwegian, Polish, Portuguese, Romanian, Russian, Slovenian, Spanish, and SwedishAUDIO FEATURES• Poly HD Voice technology delivers lifelike voice quality for each audio path; handset, hands-free speakerphone, and optional headset• Poly Acoustic Fence technology removes background office noise heard by far end when using handset or wired headsets• Poly Acoustic Clarity technology providing full-duplex speakerphone conversations, acoustic echo cancellation and Poly NoiseBlock AI background noise suppression• Poly NoiseBlock AI technology removes most background noise when using the speakerphone even while you are talking• TIA-920 wideband audio, type 1 compliant (IEEE 1329 full duplex)• Frequency response—150 Hz—7 kHz forhandset, optional headset and hands-freespeakerphone modes• Codecs: G.711 (A-law and μ-law),G.729AB, G.722 (HD Voice), G.722.1,G.722.1C, Opus, iLBC, Siren7, Siren14,G.719, SILK• Individual volume settings with visualfeedback for each audio path• Voice activity detection• Comfort noise generation• DTMF tone generation(RFC 2833 and in-band)• Low-delay audio packet transmission• Adaptive jitter buffers• Packet loss concealment• OPUS supportHEADSET AND HANDSET COMPATIBILITY• Dedicated RJ-9 headset port• Hearing aid compatibility to ITU-T P.370and TIA 504A standards• Compliant with ADA Section 508Subpart B 1194.23 (all)• Hearing aid compatible (HAC) handsetfor magnetic coupling to hearing aids• Compatible with commercially-availableTTY adapter equipment• USB headset supportNETWORK AND PROVISIONING• Two-port gigabit Ethernet switch• 10/100/1000Base-TX across LANand PC ports• Manual or dynamic host configurationprotocol (DHCP) network setup• Time and date synchronizationusing SNTP• FTP/TFTP/HTTP/HTTPSserver-based central provisioningfor mass deployments• Provisioning and call serverredundancy supported• RTCP and RTP support• Event logging• Syslog• Hardware diagnostics• Status and statistics reporting• IPv4 and IPv6• UDPSECURITY• SRTP media encryption using AES-256• Transport layer security (TLS)• Encrypted configuration files• Digest authentication• Password login• Support for URL syntax with passwordfor boot server address• HTTPS secure provisioning• Support for signed software executablesPOWER• Built-in auto sensing IEEE 802.3afPower- over-Ethernet (Class 3) 13 W (max)• External Universal AC/DCAdapter(optional) 5 VDC @ 3 A (15 W)• ENERGY STAR® ratedSPECIFICATIONSREGULATORY APPROVALS• Australia RCM• Brazil ANATEL• Canada ICES and NRTL• China RoHS 2.0• EEA CE Mark• Indonesia SDPPI• Japan VCCI• Mexico NYCE• NZ Telepermit• Saudi Arabia CITC• South Africa ICASA• South Korea KC• USA FCC and NRTL• UKCA markSAFETY• UL 62368 -1• CAN/CSA C22.2 No. 62368-1:19• EN 62368 -1• AS/NZS 62368 -1EMC• FCC Part 15 Class B• ICES-003 Class B• EN 55032 Class B• CISPR32 Class B• VCCI Class BOPERATING SYSTEM• Android 9• Android 12 (post FA)OPERATING CONDITIONS• Temperature: 0 to 40 °C (+32 to 104 °F)• Relative humidity: 5% to 95%,noncondensingSTORAGE TEMPERATURE• -40 to + 70 °C (-40 to + 160 °F)POLY CCX 350 BUSINESS MEDIA PHONECOMES WITH:• Console• Handset with handset cord• Network (LAN) cable—CAT-5E• Desk Stand• Setup SheetSIZE UNIT• 18.8 cm x 21.4 cm x 4.2 cm WxHxD(7.4 in x 8.4 in x 1.6 in WxHxD)WEIGHT• Unit box weight: 1.043 kg (2.23 lbs)UNIT BOX DIMENSIONS• 22.2 cm x 24.3 cm x 6.9 cm WxHxD(8.7 in x 9.6 in x 2.7 in WxHxD)MASTER CARTON QUANTITY• Ten (10)WARRANTY• One (1) Year©2023 Poly. All trademarks are the property of their respective owners. 3.23 1895923LEARN MOREFor more information on CCX 350 visit /ccx。
微芯片 dsPIC30F 音频回声消除库文档说明书
DS70148B-14dsPIC30FAcoustic Echo Cancellation LibraryDevices SupporteddsPIC30F6014dsPIC30F6014A dsPIC30F6012dsPIC30F6012AdsPIC30F5013 (for a max. of 32 ms echo delay)dsPIC30F5011 (for a max. of 32 ms echo delay)SummaryThe dsPIC30F Acoustic Echo Cancellation (AEC) Library provides a function to eliminate echo generated in the acoustic path between a speaker and a microphone. This function is useful for speech and telephony applications in which a speaker and a microphone are located in close proximity to each other and are susceptible to signals propagating from the speaker to the microphone resulting in a perceptible and distracting echo effect at the far-end. It is especially suitable for these applications:• Hands-free cell phone kits • Speakerphones • Intercoms• Teleconferencing systemsFor hands-free phones intended to be used in compact environments, such as a car cabin, this library is fully compliant with the G.167 standard for acoustic echo cancellation.The AEC Library is written entirely in assembly language and is highly optimized to make extensive use of the dsPIC30F DSP instruction set and advanced addressing modes. The algorithm avoids data overflow. The AEC Library provides an “AcousticEchoCancellerInit ” function for initializing the various data structures required by the algorithm and an “AcousticEchoCanceller ” function to remove the echo component from a 10 ms block of sampled 16-bit speech data. The user can easily call both functions through a well-documented Application Programmer’s Interface (API).The “AcousticEchoCanceller ” function is primarily a Time Domain algorithm. The received far-end speech samples (typically received across a communication channel such as a telephone line) are filtered using an adaptive Finite Impulse Response (FIR) filter. The coefficients of this filter are adapted using the Normalized Least Mean Square (NLMS) algorithm, such that the filter closely models the acoustic path between the near-end speaker and the near-end microphone (i.e., the path traversed by the echo). Voice Activity Detection (VAD) and Double Talk Detection (DTD) algorithms are used to avoid updating the filter coefficients when there is no far-end speech and also when there is simultaneous speech from both ends of the communication link (double talk). As a consequence, the algorithm functions correctly even in the presence of full-duplex communication. A Non-Linear Processor (NLP) algorithm is used to eliminate residual echo.The dsPIC30F Acoustic Echo Cancellation Library uses an 8 kHz sampling rate. However, the library includes a sample rate conversion function that ensures interoperability with libraries designed for higher sampling rates (9.6 kHz, 11.025 kHz or 12 kHz). The conversion function allows incoming signals at higher sampling rates to be converted to a representative 8 kHz sample. Similarly, the conversion function allows the output signal to be converted upward from 8 kHz to match the user application.FeaturesKey features of the Acoustic Echo Cancellation Librar include:• All functions can be called from either a C or assembly applicationprogram• Five user functions:– AcousticEchoCancellerInit – AcousticEchoCanceller – InitRateConverter – SRC_upConvert – SRC_downConvert• Full compliance with the Microchip MPLAB® C30 C Compiler,assembler and linker• Simple user interface – one library file and one header file• Highly optimized assembly code, utilizing DSP instructions andadvanced addressing modes• Echo cancellation for 16, 32 or 64 ms echo delays or ‘tail lengths’(configurable)• Fully tested for compliance with G.167 specifications for in-carapplications• Audio bandwidth: 0-4 kHz at 8 kHz sampling rate• Convergence rate: Up to 43 dB/sec., typically > 30 dB/sec.• Echo cancellation: Up to 50 dB, typically > 40 dB• Can be used together with the Noise Suppression (NS) Library, sincethe same processing block size (10 ms) is used• dsPIC30F Acoustic Echo Cancellation Library User’s Guide is included • Demo application source code is provided with the library• Accessory Kit available for purchase includes an audio cable,headset, oscillators, microphone, speaker, DB9 M/F RS-232 cable, DB9M-DB9M null modem adapter and can be used for library evaluation64 16.5 6 5.7 32 10.5 6 3.4 16 7.5 6 2.6Sample Rate ConversionComputational requirements: 1 MIPS Program Flash memory: 2.6 KB RAM: 0.5 KBNote: The user application might require an additional 2 to 2.5 KB of RAM for data buffering (application-dependent)。
G.729AB介绍
以前我的QSPhone一直在用G729A编码,最近遇到一个问题,有用户在使用的时候出现类似破音的情况,我测试也发现这样很不爽,(奇怪的是在XP下没问题,但在Win7下就出现)。
经过多次测试发现在说话时质量很好,当有空闲时出现破音。
到时就考虑静音检测问题,于是查了一下G729与静音检测,找到了问题所在,是双方编码有偏差,G729ab<->G729A 间通信,本来是可以兼容的,只怪判断编码是加了一default的,而舒适噪音生成(comfort noise generator,CNG)的RTP Payload是13,走到了default里面播放出现破音。
关于最近学习G729ab资料总结一下:1.G729编码系列:G729:最基本的G.729 标准协议,原始版G729A: 精简版的G729,兼容原始版G729,对G.729 codec 一些算法进行简单处理,相当于降低了算法的复杂度G729B: 加入了语音端点检测模块,在编码前对语音进行语音和静默音进行检测,然后分别对不同情况进行编码G729AB: 这是有静音检测的G729A版本,而且相容G729B。
2.一些缩略语VAD: Voice Activity Detection 语音行为检测,是一种当被在语音端口或拨号对等体上被开启时,只有可以听见的语音能够被传输的功能。
当VAD 被开启时,语音的质量的级别会被稍微降低,但此种连接占用的宽带相对要少一些。
CNG: comfort noise generator舒适噪音生成,在静音的时候产生一定的语音,一直静音下,人的感官觉得不适,为电话通信产生背景噪声。
DTX: 不连续发射(Discontinuous Transmission ),在语音间歇期关闭发射,仅发射静音指示帧,接收端码变换器产生舒适噪声。
BFI:Bad Frame Idicator,坏帧指示,静音帧标识。
SID: Silence Insertion Descriptor,2个字节表示。
Poly Edge E300系列IP桌面电话说明书
DATA SHEET POLY EDGE E300 SERIES IP DESK PHONESIt’s time for a desk phone that makes hybrid work easy. The Poly Edge E300 Series with eight-line keys offers more ways to connect, plus unbelievable audio, in a seriously sharp package. Built for hot-desking or the home office, these phones provide simple mobile phone pairing, plus bring together Poly’s famous noise reduction technology and a cutting-edge design for the perfect user experience. Our most versatile phones to date, the Poly Edge E300 Series is packed with features you didn’t even know you needed, like text-to-speech and antimicrobial protection. No matter what size your business, this is the phone that upgrades your office with style, bringing your organization into the future. • 8-line keys supporting up to 32 lines, features and contacts.• Integrated Bluetooth® 5.0 for mobile phone or headset pairing (E320,E350).• Poly NoiseBlockAI and Poly Acoustic Fence technologies.• Bright 3.5” color IPS LCD display.• Text-to-speech and new accessibility options.CUTTING-EDGE EXPERIENCEBENEFITS• Stay informed with style with the surround light status indicator with RGB mixing.• Easy to install in home offices orhard to cable locations with included Wi-Fi (E350).• Designed for the hybrid office with NFC technology for advanced integrations. • Your phones stay cleaner for longer with integrated Microban®antimicrobial protection.POLY EDGE E300 SERIESPOLY EDGE E300POLY EDGE E320POLY EDGE E350IDEAL FOR Cubicle workers, call centersand small businesses Cubicle workers, hybrid officeworkers, call centers andsmall businessesRemote call center workers,hybrid office workers and smallbusinesses needing phones in hardto cable locations or home officesLINE KEYS888LINES, CONTACTS8 displayed / 32 supported8 displayed / 32 supported8 displayed / 32 supported PAIRING BLUETOOTH® HEADSETAND MOBILE PHONENo Yes YesWI-FI No No Yes SPECIFICATIONSLINES / FEATURE KEYS• 8-line keys supporting up to 32-line key assignments for lines, contacts, and features supported with pagination • 4 context-sensitive “soft” keys• 4-way navigation key cluster with center “Select” key• Home and back feature keys• Pagination key for additional lines/ contacts• Volume + / - control keys• Hold and Transfer keys• Headset select key• Speakerphone select key• Mute key (illuminated when muted)• Voicemail keyUSER INTERFACE FEATURES• Color 3.5” IPS LCD display (320x240 pixel resolution)• Voicemail support1• WebKit-based browser• Two position desk stand (an optional wall mount kit can be ordered separately)• Unicode UTF-8 character support• One USB Type-C ports (2.0 compliant) for media, storage applications and headset connectivity• Surround Lighting for the status indicator (RGB with color mixing) and future applications • Integrated Bluetooth 5.0 (Edge E320,Edge E350)• NFC Support• Multilingual user interface including2Arabic (UAE), Chinese (Traditional/Simplified), Czech, Danish, Dutch,English (Canada/ US/UK), French(France/Canadian), German, Hungarian,Italian, Japanese, Korean, Norwegian,Polish, Portuguese, Romanian, Russian,Slovenian, Spanish, and Swedish AUDIO FEATURES• Poly HD Voice technology deliverslifelike voice quality for each audio path:handset, hands-free speakerphone, andoptional headset• Poly Acoustic Clarity technologyprovides full duplex conversations,acoustic echo cancellation, andbackground noise suppression• Poly Acoustic Fence technologyeliminates background noise when usinga handset or wired headset• Poly NoiseBlockAI technology removesmost background noise when using thespeakerphone• Poly Computer Audio Connector appinstalled on your PC (Windows only)enables selecting your phone for PCaudio in/out to use the phones handset,optional headset and handsfreespeakerphone with PC applications• Frequency response—150 Hz-14 kHz forhandset, optional headset and handsfreespeakerphone modes• Codecs: G.711 (A-law and μ-law),G.729AB, G.722 (HD Voice), G.722.1,iLBC, OPUS• TIA-920 wideband audio, type 1compliant (IEEE 1329 full duplex)• Individual volume settings with visualfeedback for each audio path• Voice activity detection• Comfort noise generation• DTMF tone generation (RFC 2833 andin-band)• Low delay audio packet transmission• Adaptive jitter buffers• Packet loss concealmentHEADSET AND HANDSET COMPATIBILITY• Dedicated RJ-9 headset port• Hearing aid compatibility to ITU-T P.370and TIA 504A standards• Compliant with ADA Section 508Subpart B 1194.23 (all)• Hearing aid compatible (HAC) handsetfor magnetic coupling to hearing aids• Compatible with commercially availableTTY adapter equipment• USB headset 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ratedSAFETY• UL62368-1• CAN/CSA C22.2 No 62368-1-14• EN 60950-1/62368-1• IEC 60950-1/62368-1• AS/NZS 60950.1/62368.1REGULATORY APPROVALS(POLY EDGE E300)³• FCC Part 15 (CFR 47) Class B• ICES-003 Class B• EN55032 Class B• CISPR32 Class B• VCCI Class B• EN55024• EN61000-3-2; EN61000-3-3• UK - UKCA• NZ Telepermit• UAE TRA• Eurasian Customs Union EAC• Brazil ANATEL• Australia RCM• South Africa ICASA• Saudi Arabia CITC• Indonesia SDPPI• S.Korea KC• Mexico NOM ANCE• RoHS Compliant• CE Mark• TAAREGULATORY APPROVALS(POLY EDGE E320)³• Argentina ENACOM• Australia RCM• Brazil ANATEL• Canada ICES• China SRRCDATA SHEETLEARN MOREFor more information on Poly Edge E300 Series visit /edge-e300DATA SHEET• China RoHS 2.0• EEA CE Mark• Eurasian Customs Union EAC • India WPC • Indonesia SDPPI • Israel MOC• Japan MIC and VCCI • Malaysia SIRIM• Mexico IFETEL and NYCE • NZ Telepermit • Saudi Arabia CITC • Singapore IMDA • South Africa ICASA • South Korea KC • Taiwan NCC • UAE TRA • UK - UKCA • USA FCCEMC (POLY 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supported by the server. Please contact your IP PBX/Softswitch vendor or service provider for a list of supported features.2 Planned localizations 3 Planned compliances4 O rdering an optional power supply unit will be necessary if not powered over Ethernet with PoE (i.e. Using Wi-Fi for network)©2023 Poly. All trademarks are the property of their respective owners. The Bluetooth trademark is owned by Bluetooth SIG, Inc. and any use of the mark by Poly is under license. 4.23 1835654。
ESP32 ESP-SR 用户手册说明书
ESP32ESP-SR用户手册Release master乐鑫信息科技2023年10月24日Table of contentsTable of contents i 1入门指南31.1概述 (3)1.2准备工作 (3)1.2.1必备硬件 (3)1.2.2必备软件 (3)1.3编译运行一个示例 (3)2AFE声学前端52.1AFE声学前端算法框架 (5)2.1.1概述 (5)2.1.2使用场景 (5)2.1.3选择AFE Handle (8)2.1.4输入音频 (8)2.1.5输出音频 (9)2.1.6使能唤醒词识别WakeNet (9)2.1.7使能回声消除算法AEC (9)2.1.8资源消耗 (9)2.2乐鑫麦克风设计指南 (10)2.2.1麦克风电器性能推荐 (10)2.2.2麦克风结构设计建议 (10)2.2.3麦克阵列设计建议 (10)2.2.4麦克风结构密封性建议 (10)2.2.5回声参考信号设计建议 (11)2.2.6麦克风阵列一致性验证 (11)3唤醒词133.1WakeNet唤醒词模型 (13)3.1.1概述 (13)3.1.2WakeNet的使用 (14)3.1.3资源消耗 (15)3.2乐鑫语音唤醒方案客户定制流程 (15)3.2.1唤醒词定制服务 (15)3.2.2硬件设计与测试服务 (16)4命令词174.1MultiNet命令词识别模型 (17)4.2命令词识别原理 (17)4.3命令词格式要求 (18)4.4自定义命令词方法 (18)4.4.1MultiNet6定义方法: (18)4.4.2MultiNet5定义方法: (18)4.4.3通过调用API修改 (19)4.5MultiNet的使用 (20)4.5.1MultiNet初始化 (20)4.5.2MultiNet运行 (21)4.5.3MultiNet识别结果 (21)4.6资源消耗 (21)5TTS语音合成模型235.1简介 (23)5.2简单示例 (23)5.3编程指南 (24)5.4资源消耗 (24)6模型加载256.1配置方法 (25)6.1.1使用AFE (25)6.1.2使用WakeNet (25)6.1.3使用MultiNet (26)6.2模型使用 (26)6.2.1模型数据存储在Flash SPIFFS (27)7性能测试结果297.1AFE (29)7.1.1资源消耗 (29)7.2WakeNet (29)7.2.1资源消耗 (29)7.2.2性能测试 (29)7.3MultiNet (30)7.3.1资源消耗 (30)7.3.2Word Error Rate性能测试 (30)7.3.3Speech Commands性能测试(空调控制场景) (30)7.4TTS (30)7.4.1资源消耗 (30)7.4.2性能测试 (30)8测试方法与测试报告318.1测试场景要求 (31)8.2测试案例设计 (31)8.3乐鑫测试与结果 (32)8.3.1唤醒率测试 (33)8.3.2语音识别率测试 (33)8.3.3误唤醒率测试 (33)8.3.4唤醒打断率测试 (33)8.3.5响应时间测试 (34)9术语表359.1通用术语 (35)9.2特别术语 (35)本文档仅包含针对ESP32芯片的ESP-SR使用。
AMRWB语音端点检测功能分析研究及定点C代码仿真
(3) 用 C 代码在 PC 机上设计实现了 AMR-WB 定点仿真环境 该 仿真环境为研究验证相关算法(特别是 VAD 算法)的可行性和 有效性提供了有效的实验平台
本文由五部分组成 第一章前言 对当前使用的语音编解码器 的概念 发展状况及存在的问题进行了阐述 第二章对介绍 第三章对 AMR-WB 宽带语音编解码器的语音端点检测算法 VAD 及其具体实现方法进行 了详细分析 提出了改进方法并对改进后的 VAD 方法进行了实验验 证 第四章对 AMR-WB 宽带语音编解码器仿真系统设计进行了阐 述 包括系统模型 各模块分析以及算法设计 并且对实测数据进 行了详细分析 第五章为总结和展望
宽带语音编解码器的自然度在高保真电话 扩展的通讯业务如声音会议 电视广播中是一个重要的特征 以下简要介绍几种宽带语音的应用领域
(1) 3G 移动通讯系统 提供多媒体服务是 3G 无线通讯系统的一个主要功能 这也意味着在多媒 体部分要使用高质量的声音和语音 即使在说话声音的电话应用中 宽带语音 也是无线服务提供商可以提供比传统公共交换电话网(PSTN)更高语音质量的重
关键词 AMR-WB 语音编解码器 ACELP 端点检测 鲁棒性
MECHANISM STUDY OF VOICE ACTIVITY DETECTOR (VAD) AND C SIMULATION OF AMR-WB CODEC
ABSTRACT
At present in mobile communication system AMR-WB (Adaptive Multi-rate Wideband) CODEC makes speech frequency extended to 7kHz and sample frequency extended to 16kHz, greatly breakthroughs the restriction of the bandwidth than the narrow band CODEC. Therefore, AMR-WB CODEC will ameliorate greatly in many aspects such as speech naturalness and musical processing. This paper expatiates the analysis of the VAD (Voice Activity Detection) algorithm of the AMR-WB CODEC in the noise background condition. Based on this research, author gives a more robust improved VAD algorithm. On the basis, author develops a practical AMR-WB CODEC simulation system according to demands from consigning unit.
一种基于噪声场景识别与多特征集成学习的活动语音检测方法
一种基于噪声场景识别与多特征集成学习的活动语音检测方法TIAN Y,ZHANG X B.An Approach for Voice Activity Detection Based on Noise Recognition and Multi-feature Ensemble44(6):28 - 31,39.DOI:10. 16311/j. audioe. 2020. 06. 006一种基于噪声场景识别与多特征集成学习的活动语音检测方法田 野,张小博中国电子科技集团公司第三研究所,北京为了提高动态噪声环境下语音检测的准确性,提出一种基于噪声场景识别与多特征集成学习的活动语音检测方法。
针奇异值等时频域特征,采用随机森林优选出可分性更好的特征组合。
实验表明,所提方法的语音检测准确率提升显著,随机森林An Approach for Voice Activity Detection Based on Noise Recognition and Multi-feature Ensemble LearningTIAN Ye, ZHANG Xiaobo(The Third Research Institute of China Electronics Technology Corporation, Beijing 100015, China) To improve the accuracy of voice activity detection under dynamic noise environments, this paper proposes an approach based on noise recognition and multi-feature ensemble learning. Considering the nonstationary and complex signals, time-frequency features, like wavelet energies and singular values, are extracted and selected using Random Forest(RF) for better separability. Case studies based on different noises and different noise powers demonstrate the higher accuracy and stability of the proposed approach.voice activity detection; noise recognition; feature selection; random forest在一段语音信号中往往会存在着停顿、间歇等干扰语音处(Voice Activity技术的目标是从信号中检测出真两个环节构成。
vad 方案
1. 简介语音活动检测(Voice Activity Detection,简称VAD)是一种技术,用于在语音信号中检测和区分活动状态和非活动状态。
VAD技术在语音信号处理、音频编解码、语音识别以及语音通信等领域中具有广泛的应用。
本文将介绍VAD的概念、原理、算法以及在实际应用中的方案。
2. VAD的概念VAD是指在语音信号中检测和判断活动状态,即判断说话人是否在说话。
活动状态表示语音信号中的有效说话部分,非活动状态则表示语音信号中的噪声或静音部分。
VAD的主要目标是抑制非活动状态的信号,从而提高语音处理系统的性能。
3. VAD的原理VAD的原理是基于语音信号的能量和频率特征进行判断。
常用的VAD算法包括基于能量门限、基于短时能量和短时过零率、基于倒谱系数等。
3.1 基于能量门限的VAD基于能量门限的VAD算法是一种常见的简单检测方法。
该算法通过设置一个能量门限,当语音信号的能量超过门限值时,判断为活动状态;当能量低于门限值时,则判断为非活动状态。
3.2 基于短时能量和短时过零率的VAD基于短时能量和短时过零率的VAD算法是一种较为复杂的方法。
该算法通过计算短时能量和短时过零率,判断语音信号的活动状态。
当语音信号的短时能量和短时过零率超过一定阈值时,判断为活动状态;当短时能量和短时过零率低于阈值时,则判断为非活动状态。
3.3 基于倒谱系数的VAD基于倒谱系数的VAD算法是一种较为复杂和高精度的方法。
该算法通过提取语音信号的倒谱系数特征,并使用机器学习算法进行分类和判断。
根据倒谱系数的分布和特征,判断语音信号的活动状态。
4. VAD的实际应用方案VAD在语音处理和语音通信中有着广泛的应用。
以下是VAD的几种实际应用方案。
4.1 音频编解码中的VAD在音频编解码中,VAD能够提高编码效率和压缩率。
通过检测语音信号的活动状态,可以忽略非活动状态的信号,从而减少需要编码和传输的数据量。
4.2 语音识别中的VAD在语音识别中,VAD能够提高识别的准确性和效率。
vad方案
VAD方案什么是VAD?VAD(Voice Activity Detection)是一种音频信号处理技术,用于检测和识别语音活动部分。
VAD方案通常用于语音通信应用和语音识别系统中,以提高语音识别的准确性和效率。
VAD方案的作用VAD方案可以在语音信号中准确地检测出语音活动的部分,从而实现以下功能:1. 节省计算资源:在语音处理和语音识别过程中,对非语音活动部分进行处理是浪费资源的。
VAD方案可以减少对非语音活动部分的处理,从而提高计算效率。
2.准确识别语音边界: VAD方案可以帮助准确地识别语音的开始和结束,这对于语音识别和音频存储等应用非常重要。
3.降噪处理: VAD方案可以将噪声和语音信号区分开来,从而帮助降噪算法更好地处理语音信号。
常用的VAD方案以下列举了一些常用的VAD方案:1. 基于能量的VAD基于能量的VAD方案是最简单和常见的VAD方法之一。
该方法通过计算音频信号的瞬时能量,并设置一个阈值,来区分语音和非语音区段。
当信号的能量超过阈值时,被认为是语音活动。
2. 基于短时过零率的VAD基于短时过零率的VAD方案是通过计算音频信号在短时窗口内的过零率来判断语音活动。
过零率指的是信号穿过零点的频率。
在语音活动期间,音频信号的过零率较高。
3. 基于频域特征的VAD基于频域特征的VAD方案采用频域分析技术,如FFT(Fast Fourier Transform),将音频信号转换到频率域,并根据频域特征来判断语音活动。
4. 基于机器学习的VAD基于机器学习的VAD方案使用训练好的模型来判断语音活动。
常用的机器学习算法包括支持向量机(SVM)、随机森林(Random Forest)和深度神经网络(DNN)等。
5. 基于端到端的VAD基于端到端的VAD方案是近年来的研究热点。
它直接从原始音频信号中学习语音活动和非语音活动的区别,不需要人工设计特征和训练模型。
如何选择适合的VAD方案?选择适合的VAD方案需要考虑以下因素:1.应用场景:根据实际应用场景的需求,选择最合适的VAD方案。
vad方案
vad方案VAD(Voice Activity Detection,语音活动检测)是一种用于检测语音信号是否存在的技术。
在实际应用中,VAD技术广泛应用于语音识别、语音通信、音频编解码等各种领域。
VAD方案主要包括以下几个方面:1. 基于能量的VAD方案:此方案根据语音信号的能量来判断是否存在语音活动。
通常,当语音信号的能量超过一定阈值时,认为存在语音活动;当能量低于阈值时,认为不存在语音活动。
这种方案的优点是简单易实现,但在存在噪音的环境中容易误判。
2. 基于过零率的VAD方案:此方案根据语音信号通过零点的次数来判断是否存在语音活动。
通常,当语音信号通过零点的次数超过一定阈值时,认为存在语音活动;当通过零点的次数低于阈值时,认为不存在语音活动。
这种方案的优点是对于纯语音信号的判别效果较好,但对于存在噪音的环境中的判别效果较差。
3. 基于能量和过零率的联合VAD方案:此方案结合了基于能量和过零率的判别方法,通过同时判断语音信号的能量和过零率来确定是否存在语音活动。
这种方案的优点是综合考虑了能量和过零率两个指标,相对于单一指标的判别方法,判别效果更加准确。
4. 基于梅尔频谱的VAD方案:此方案基于语音信号的梅尔频谱特征进行判别。
通过计算语音信号的梅尔频谱图,并结合相关模型进行判断,判断该信号是否为语音活动。
这种方案的优点是可以更准确地判别语音活动和非语音活动。
总的来说,VAD方案的选择应根据具体的应用场景和需求来确定。
对于需要快速响应的实时应用,基于能量的VAD方案可能更加适用;而对于需要准确判断的高精度应用,可以采用基于梅尔频谱的VAD方案。
综合考虑各种因素,可以根据具体需求设计和选择相应的VAD方案。
一种提高恶劣环境中电台话音质量的方法
一种提高恶劣环境中电台话音质量的方法姚元飞;陈洪瑀;黄鹤【摘要】针对恶劣环境中电台话音质量差这一难题,提出了一种有效的解决方法.首先在最小值控制的递归平均Ⅱ型(MCRA-2)算法基础上,提出了一种改进的MCRA-2算法.该算法采用自适应平滑参数和双检测门限,能有效减小音乐噪声,提高噪声估计准确性;然后将该噪声估计算法与对数最小均方误差估计器相结合,实现话音降噪,提高恶劣环境下电台话音质量;最后利用噪声估计结果和降噪后的话音信号,采用信噪比判决算法,自适应估计静噪门限,实现自适应静噪,进一步提高话音舒适度.实验表明,采用该方法的电台在大噪声、强干扰等恶劣环境中使用时,能明显改善话音质量,有效减轻收听者听觉疲劳.【期刊名称】《电讯技术》【年(卷),期】2019(059)004【总页数】6页(P437-442)【关键词】电台话音质量;自适应噪声估计;话音降噪;自适应静噪【作者】姚元飞;陈洪瑀;黄鹤【作者单位】中国西南电子技术研究所,成都610036;成都天奥信息科技有限公司,成都611731;中国西南电子技术研究所,成都610036;成都天奥信息科技有限公司,成都611731;中国西南电子技术研究所,成都610036;成都天奥信息科技有限公司,成都611731【正文语种】中文【中图分类】TN9241 引言电台是无线电台的通称,话音通信是电台最重要的通信方式之一。
然而,随着电台应用普及程度的增大,应用场所也越来越多,其中有些场所环境比较恶劣。
比如地空通信中,不管是飞机驾驶舱还是机场,噪声都比较大,如果飞机出现驾驶舱玻璃破碎、机械故障等紧急情况,噪声更是巨大无比。
因此,提高恶劣环境下电台话音质量是提高电台性能的一个重要研究方向。
其中,话音质量也可称为通话质量,是表示收听到的对方讲话声音的优劣程度,是收听者对话音可懂度和舒适度的主观感受的反映,可用话音信噪比和失真度客观地衡量。
在这些恶劣环境中,噪声比较大,为了保证正确无误的话音通信,一般会将电台音量调到很大,这时如果话音可懂度低,那么听到的基本上都是噪声,收听者肯定是难以忍受的。
话音激活检测(VAD)算法的研究
(2-3)
1 N →∞ N
∑ s(n) = 0 ,则 C
n =0
N −1
3y
(t1 , t2 ) = C3 s (t1 , t2 ) + C3v (t1 , t2 ) [3]。 C3 s 为语音信号的
三阶累积量; C3v 为噪声信号的三阶累积量。若 v( n) 是高斯或者对称的随机过程,则
话音激活检测(VAD)算法研究
王凡
北京邮电大学,北京(100876)
E-mail:Wangfan614@
摘 要:话音激活检测(VAD)算法能够区分传输语音信号中的语音信号和背景噪音,避免 无用信号的传输,从而节省有限网络资源,因此对 VAD 算法的研究有重要的意义。由于一般 静默压缩方法仅考虑高信噪比和平稳背景噪声这种理想通信条件,为了在复杂的背景噪声下 进行 VAD 检测,本文应用高阶统计理论的特点设计了一种新的话音激活检测算法,并从客观 和主观的角度对算法进行分析。结果表明,本文提出的算法简单有效的作出 VAD 检测。 关键词:自适应多速率编码;话音激活检测;静默压缩;三阶累积量
图 2-1 男声信号时域波形
-2-
如果规定 20ms(160 个样本点)[8]为一帧信号的长度,则图 2-1 中的信号共有 187 帧, 图 2-2 和图 2-3 分别为从图 2-1 的电平信号中取出的一帧噪声信号和一帧话音信号,由(3-4) 由此可以 式取 (t1 , t2 ) = (2, 4) 可以计算出它们对应的三阶累积量分别为 0.14375 和 374.61875, 看出噪声信号的三阶累积量接近于零,而话音信号的三阶累积量数值很大,用此方法来实现 话音激活检测有很强的可行性。
2.3 程序设计的思路及各功能模块的描述
多种噪声环境下语音检测算法
多种噪声环境下语音检测算法胡海波;任立伟;刘柏森【摘要】提出针对语音检测的一种新的算法-傅立时谱矩阵,该方法以傅立叶变换为基础,对含噪语音信号直接进行分帧,对每帧做傅立叶变换后以帧为单位,构建能反映语音特征频段及能量的傅立叶谱矩阵.在傅立叶谱矩阵中找到语音信号能量的集中区域,从而找出所需的语音特征谱矩阵的范围,达到对语音的有效检测.【期刊名称】《黑龙江工程学院学报(自然科学版)》【年(卷),期】2012(026)003【总页数】4页(P36-39)【关键词】信号处理;语音检测;傅立叶谱矩阵;傅立叶变换【作者】胡海波;任立伟;刘柏森【作者单位】黑龙江工程学院电气与信息工程学院,黑龙江哈尔滨150050;黑龙江工程学院电气与信息工程学院,黑龙江哈尔滨150050;黑龙江工程学院电气与信息工程学院,黑龙江哈尔滨150050【正文语种】中文【中图分类】TN912近年来,随着科学的发展和计算机的普及,人们对与计算机的交流方式提出了更高的要求,这促进了语音识别技术的发展并使之成为语音处理领域中的一个重要研究方向。
目前,在实验室环境下,语音识别系统的识别率已经达到很高的水平,也有一些产品出现,但是由于受实际环境噪声的影响,系统的识别率大幅度下降。
因此,噪声是语音识别技术广泛实用化的最大障碍,对噪声环境下语音识别系统的研究也就变得尤为重要。
而语音检测是实现语音识别最重要的一部分,所以语音检测及去噪越来越受关注,出现了许多好的方法[1-2]。
谱减法[3]是基于人的感觉特性,即语音信号的短时幅度比短时相位更容易对人的听觉系统产生影响,从而对语音短时幅度谱进行估计,适用于受加性噪声污染的语音。
基于隐马尔可夫能够有效地提取时序特征和小波神经网络细分类的基础上,通过实验表明该方法在不同信噪比的情况下比传统的隐马尔可夫(HMM)的识别率更高。
以上两种方法为常用算法,然而存在很大的缺陷。
谱减法属于模型补偿法,受噪音影响较大,且可能在去噪的过程中把原有的语音信号成分去掉,处理后不能很好地重构原始语音。
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VOICE ACTIVITY DETECTION WITH NOISE REDUCTION AND LONG-TERM SPECTRAL DIVERGENCE ESTIMATIONJ.Ram´ırez,J.C.Segura,C.Ben´ıtez,A.de la Torre,A.RubioDept.of Electronics and Computer TechnologyUniversity of Granada,SpainABSTRACTThis paper is mainly focussed on an improved voice activity detec-tion algorithm employing long-term signal processing and maxi-mum spectral component tracking.The benefits of this approach has been analyzed in a previous work with clear improvements in speech/non-speech discriminability and speech recognition per-formance in noisy environments.Two clear aspects are consid-ered in this paper.Thefirst one,which improves the performance of the V AD in low noise conditions,considers an adaptive length frame window to track the long-term spectral components.The second one reduces misclassification errors in high noisy environ-ments by using a noise reduction stage before the long-term spec-tral tracking.Experimental results show clear improvements over different V AD methods in speech/pause discrimination and speech recognition performance.Particularly,the proposed V AD reported improvements in recognition rate when replaced the V ADs of the ETSI Advanced Front-end(AFE)for distributed speech recogni-tion(DSR).1.INTRODUCTIONModern applications of speech technology are demanding increased levels of performance in many areas.With the advent of wireless communications new speech services are becoming a reality with the development of modern robust speech processing technology. An important obstacle affecting most of the environments and ap-plications is the environmental noise and its harmful effect on the system performance.Most of the noise compensation algorithms often require to estimate the noise statistics by means of a precise voice activity detector(V AD).The detection task is not as trivial as it appears since the increasing level of background noise degrades the clas-sifier effectiveness.During the last decade numerous researchers have studied different strategies for detecting speech in noise and the influence of the V AD decision on speech processing systems [1,2].Most of them have focussed on the development of robust algorithms,with special attention on the study and derivation of noise robust features and decision rules[3,4,5,6].This paper presents several improvements over a previous work on voice activity detection[7]that has been shown to be very ef-fective for noise suppression and speech recognition in noisy envi-ronments.The algorithm assumes that the most significant infor-mation for detecting speech in noise remains on the time-varying signal spectrum.The main contributions of this paper are:i)the increased non-speech detection accuracy in low noise conditions This work was partly supported by the Spanish Government under the CICYT project TIC2001-3323.by making the long-term window length adaptive to the measured noise energy,and ii)a noise reduction stage previous to tracking the long-term spectral envelope that improves the V AD effective-ness in high noise environments.The algorithm is evaluated on the context of the AURORA project and the recently approved Advanced Front-End(AFE)[8]for distributed speech recognition (DSR).Other standard V ADs such as the ITU G.729[9]or ETSI AMR[10]are also used as a reference.2.LONG-TERM SPECTRAL ESTIMATION V ADA block diagram of the proposed V AD is shown in Fig.1.Noise reduction is performedfirst and the V AD decision is formulated on the de-noised signal.The noisy speech signal x(n)is decom-posed into25-ms frames with a10-ms window shift.Let X(k,l) be the spectrum magnitude for the k-th band at frame l.The de-sign of the noise reduction block is based on Wienerfilter theory being its attenuation dependent on the signal-to-noise ratio(SNR) of the processed signal.Finally,the V AD decision is formulated in terms of the de-noised speech signal,being its spectrum Y(k,l), processed by means of a(2N+1)-frame window.2.1.Noise reductionThe noise reduction block consists of four stages:i)Spectrum smoothing.The power spectrum is averaged over two consecu-tive frames and two spectral bands.ii)Noise estimation.The noise spectrum N e(k)is updated by means of a1st order IIRfil-ter based on the smoothed spectrum Y s(k,l),that is,N e(k)=λN e(k)+(1−λ)Y s(k,l)whereλ=0.99.iii)Design of the Wienerfilter(WF).First,the clean signal S(k)is estimated by spectral subtraction:S(k,l)=XβS (k,l)+(1−Xβ)max(Y s(k,l)−N e(k),0)(1) and the Wienerfilter H(k)is calculated by:η(k,l)=maxS(k,l)N e(k),ηminH(k,l)=η(k,l)1+η(k,l)(2)whereηmin is selected so that thefilter yield a20dB maximum attenuation,and Xβ=0.98.Finally,S (k,l),that is assumed to be zero at the beginning of the process,is defined to be:S (k,l)=max[Y(k,l)H(k,l),16](3) Thefilter H(k,l)is smoothed in order to eliminate rapid changes between neighbor frequencies that may often cause musical noise.X k l(,)Fig.1.Block diagram of the VAD.The smoothing is performed by truncating the impulse response of the corresponding causal FIRfilter to17taps using a Hanning window.iv)Frequency domainfiltering.Thefilter is applied in the frequency domain to obtain the de-noised spectrum Y(k,l).2.2.V AD decision ruleOnce the input speech has been de-noised,its spectrum magni-tude Y(k,l)is processed by means of a(2N+1)-frame window. Spectral changes around an N-frame neighborhood of the actual frame are analyzed using the N-order Long-Term Spectral Enve-lope(LTSE)that was defined in[7]as:LT SE(k)=max{Y(k,l+j)}j=+Nj=−N(4) where l is the actual frame for which the V AD decision is made and k=0,1,...,NFFT-1,is the spectral band.Note that the noise suppression block have to perform the noise reduction of the block{X(k,l−N),X(k,l−N+1), ...,X(k,l−1),X(k,l),X(k,l+1),...,X(k,l+N)}before the LTSE at the l-th frame can be computed.This is carried out as follows.During the initialization,the noise suppression algorithm is applied to thefirst2N+1frames and,in each iteration,the (l+N+1)-th frame is de-noised,so that Y(k,l+N+1)become available for the next iteration.The V AD decision rule is formulated in terms of the long-term spectral divergence(LTSD)calculated as the deviation of the LTSE respect to the residual noise spectrum N(k)and defined by:LT SD=10log101NF F TNF F T−1k=0LT SE2(k)N2(k)(5)If the LTSD is greater than an adaptive thresholdγ,the actual frame is classified as speech,otherwise it is classified as non-speech.A hangover delays the speech to non-speech transition in order to prevent low-energy word endings being misclassified as silences.On the other hand,if the LTSD achieves a given thresh-old LT SD0,the hangover algorithm is turned off to improve non-speech detection accuracy in low noise environments.The V AD is defined to be adaptive to time-varying noise en-vironments with the following algorithm for updating the noise spectrum during non-speech periods being used:N(k)=αN(k)+(1−α)N K(k)(6) where N K is the average spectrum magnitude over a K-frame neigh-bourhood:N K(k)=12K+1Kj=−KY(k,n−j)(7) and k=0,1,...,NFFT/2.2.3.Initialization of the algorithmFor the initialization of the algorithm,thefirst frames of the input utterance are assumed to be non-speech and the decision threshold γand N are adapted to the measured noise energy E by:γ=γ0−γ1E0−E1E+γ0−γ0−γ11−E1/E0N=roundN0−N1E0−E1E+N0−N0−N11−E1/E0(8)where E0and E1are the average noise energy for clean and high noise conditions,respectively.Sinceγand N are critical param-eters for the algorithm,they are restricted to be bounded in the intervals[γ0,γ1]and[N0,N1],respectively.3.EXPERIMENTAL FRAMEWORKSeveral experiments were conducted to evaluate the performance of the V AD.The analysis is focused on the assessment of misclas-sification errors at different SNR levels and the influence of the V AD decision on an automatic speech recognition(ASR)system.3.1.Receiver operating characteristics(ROC)curvesThe ROC curves are frequently used to completely describe the V AD error rate.The AURORA subset of the original Spanish SpeechDat-Car(SDC)database[11]was used in this analysis. This database contains4914recordings using close-talking and distant microphones from more than160speakers.As in the whole SDC database,thefiles are categorized into three noisy conditions: quiet,low noisy and highly noisy conditions,which represent dif-ferent driving conditions with average SNR values between25dB, and5dB.The non-speech hit rate(HR0)and the false alarm rate (FAR0=100-HR1)were determined in each noise condition being the“real”speech frames and“real”speech pauses determined by hand-labelling the database on the close-talking microphone.The parameters used for the V AD were:N0=3N1=6,α=0.95,K=3, LT SD0=40,HO=3(hang-over length).Fig.2.a shows the ROC curve for recordings from the close-talking microphone with low noise.The working points of the G.729,AMR and AFE V ADs are also included.It can be de-rived from thefigure that the improvements considered in this work yield higher speech/non-speech discrimination.This fact is mainly motivated by using a shorter frame window for low noise environments.Fig.2.b shows the ROC curve for recordings from the distant microphone in high noisy conditions.It also shows improvements in detection accuracy over standard V ADs such as G.729,AMR and AFE and over a representative set of recently reported V AD algorithms[3,6,4,5].20304050607080900.00.51.0 1.52.0 2.53.0 3.54.0FALSE ALARM RATE (FAR0)P A U S E H I T R A T E (H R 0)(a)020*********102030405060FALSE ALARM RATE (FAR0)P A U S E H I T R A T E (H R 0)(b)Fig.2.ROC curves.(a)Close talking microphone (stopped car,motor running).(b)Distant microphone (high speed,good road).It can be concluded from these results that:i)The working point of the G.729V AD shifts to the right in the ROC space with decreasing SNR.ii)AMR1works on a low false alarm rate point of the ROC space but exhibits poor non-speech hit rate.iii)AMR2yields clear advantages over G.729and AMR1exhibiting impor-tant reduction of the false alarm rate when compared to G.729and increased non-speech hit rate over AMR1.iv)The V AD used in the AFE for noise estimation yields good non-speech detection accuracy but works on a high false alarm rate point on the ROC space.It suffers from rapid performance degradation when the driving conditions get noisier.On the other hand,the V AD used in the AFE for frame-dropping has been planned to be conserva-tive since it is only used in the DSR standard for frame-dropping.Thus,it exhibits poor non-speech detection accuracy working on a low false alarm rate point of the ROC space.Among all the V AD examined,our V AD yields the lowest false alarm rate for a fixed non-speech hit rate and also,the highest non-speech hit rate for a given false alarm rate.The ability of the adaptive LTSE V AD to tune the detection threshold by means the algorithm described in Eq.8enables working on the optimal point of the ROC curve for different noisy conditions.3.2.Speech recognition performanceAlthough the ROC curves are effective to evaluate a given al-gorithm,the influence of the V AD in an ASR system was alsostudied.The reference framework (Base)is the ETSI AURORA project for distributed speech recognition [12]while the recog-nizer is based on the HTK (Hidden Markov Model Toolkit)soft-ware package [13].The task consists on recognizing connected digits which are modelled as whole word HMMs (Hidden Markov Models)with the following parameters:16states per word,simple left-to-right models,mixture of 3Gaussians per state while speech pause models consist of 3states with a mixture of 6Gaussians per state.The 39-parameter feature vector consists of 12cepstral coefficients (without the zero-order coefficient),the logarithmic frame energy plus the corresponding delta and acceleration coef-ficients.Two training modes are defined for the experiments con-ducted on the AURORA-2database:i )training on clean data only (Clean Training),and ii )training on clean and noisy data (Multi-Condition Training).The influence of the V AD decision on the performance of a feature extraction scheme incorporating Wiener filtering (WF)as noise suppression method and non-speech frame-dropping (FD)to the Base system [12]was assessed.Table 1shows the recognition results as a function of the SNR for the Base system and for the different V ADs that were incorporated to the feature extraction al-gorithm.These results are averaged over the three test sets of the AURORA-2recognition experiments.An estimation of the 95%confidence interval (CI)is also provided.Notice that,particularly,for the recognition experiments based on the AFE V ADs,we have used the same configuration used in the standard [8]which present different V ADs for WF and FD.The proposed V AD outperforms the standard G.729,AMR1,AMR2and AFE V ADs in both clean and multi condition training/testing experiments.Table 2compares recently reported V AD algorithms to the proposed one in terms of the average word accuracy for clean and multicondition training/test experiments.The proposed algorithm also outperforms the V ADs used as a reference being the one that is closer to the “ideal”hand-labelled speech recognition perfor-mance.Finally,in order to compare the proposed method to the best available results,the V ADs of the full AFE standard (including both the noise estimation and frame dropping V ADs)were re-placed by the proposed LTSE V AD and the AURORA recogni-tion experiments were conducted.The results are shown in Table 3.The word error rate is reduced from 13.07%to 12.42%for the clean training experiments and from 8.14%to 7.88%in multi-condition when the V ADs of the original AFE are replaced by the proposed V AD.4.CONCLUSIONSThis paper has shown an improved V AD algorithm for increasing speech detection robustness in noisy environments and the perfor-mance of speech recognition systems.The V AD is based on the es-timation of the long-term spectral envelope and the measure of the spectral divergence between speech and noise.Two improvements have been considered over the base system.The first one,which improves the performance of the V AD in low noise conditions,considers a variable length frame window to track the long-term spectral components.The second one reduces misclassification errors in high noisy environments by using a noise reduction stage before the long-term spectral tracking.With this and other innova-tions the proposed V AD has demonstrated an enhanced ability to discriminate speech and silences and to be well suited for robust speech recognition.Table1.Average Word Accuracy for the AURORA-2database.Base Multicondition Clean conditionV AD used None G.729AMR1AMR2AFE Proposed G.729AMR1AMR2AFE Proposed Clean99.0397.5096.6798.1298.3998.8398.4197.8798.6398.7899.11 20dB94.1996.0596.9097.5797.9898.4083.4696.8396.7297.8298.03 15dB85.4194.8295.5296.5896.9497.5971.7692.0393.7695.2896.44 10dB66.1991.2391.7693.8093.6395.4959.0571.6586.3688.6791.51 5dB39.2881.1480.2485.7285.3288.4943.5240.6670.9771.5577.25 0dB17.3854.5053.3662.8163.8967.4927.6323.8844.5841.7849.27 -5dB8.6523.7323.2927.9230.8033.3614.9414.0518.8716.2322.90 Average60.4983.5583.5687.2987.5589.4957.0865.0178.4879.0282.50 C.I.(95%)±0.24±0.18±0.18±0.16±0.16±0.15±0.24±0.23±0.20±0.20±0.19Table2.Average Word Accuracy for the AURORA-2database.V AD Used Clean Multi Average Woo76.6485.5481.09Li78.6385.5882.11 Marzinzik82.1588.3285.23Sohn79.4988.1183.80 Proposed82.5089.4986.00Hand-labelled84.8388.8886.86Table3.Average Word Accuracy for the AURORA-2database.Clean training Multicondition training AFE AFE+LTSE AFE AFE+LTSE Set A87.5188.1292.2992.55Set B87.0687.7992.1092.41Set C85.4286.1090.5190.70 Average86.9387.5891.8692.125.REFERENCES[1]L.Karray and A.Martin,“Towards improving speech de-tection robustness for speech recognition in adverse environ-ments,”Speech Communitation,vol.40,no.3,pp.261–276, 2003.[2] A.Sangwan,M.C.Chiranth,H.S.Jamadagni,R.Sah,R.V.Prasad,and V.Gaurav,“V AD techniques for real-time speech transmission on the internet,”in IEEE International Confer-ence on High-Speed 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