A Multi-Channel SpeechSilence Detector based on Time Delay Estimation and Fuzzy Classificat
SGM321 SGM358 SGM324 1MHz, 60μA, Rail-to-Rail I O
SGM321/SGM358/SGM3241MHz, 60μA, Rail-to-Rail I/OCMOS Operational Amplifiers GENERAL DESCRIPTIONThe SGM321 (single), SGM358 (dual) and SGM324(quad) are low cost, rail-to-rail input and output voltagefeedback amplifiers. They have a wide input commonmode voltage range and output voltage swing, and takethe minimum operating supply voltage down to 2.1V.The maximum recommended supply voltage is 5.5V.All are specified over the extended -40℃to +85℃temperature range.The SGM321/358/324 provide 1MHz bandwidth at alow current consumption of 60μA/amplifier. Very lowinput bias currents of10pA enable SGM321/358/324 tobe used for integrators, photodiode amplifiers andpiezoelectric sensors. Rail-to-rail input and output areuseful to designers for buffering ASIC in single-supplysystems.Applications for this series of amplifiers include safetymonitoring, portable equipment, battery and powersupply control, and signal conditioning and interfacingfor transducers in very low power systems.The SGM321 is available in Green SOT-23-5 andSC70-5 packages. The SGM358 is available in GreenSOIC-8, MSOP-8 and DIP-8 packages. The SGM324 isavailable in Green SOIC-14 and TSSOP-14 packages.FEATURES•Low Cost•Rail-to-Rail Input and Output•Input Offset Voltage: 5mV (MAX)•Unity Gain Stable•Gain-Bandwidth Product: 1MHz•Very Low Input Bias Current: 10pA•Supply Voltage Range: 2.1V to 5.5V•Input Voltage Range:-0.1V to 5.6V with V S = 5.5V•Low Supply Current: 60μA/Amplifier•Small Packaging:SGM321 Available in SOT-23-5 and SC70-5SGM358 Available in SOIC-8, MSOP-8 and DIP-8SGM324 Available in SOIC-14 and TSSOP-14APPLICATIONSASIC Input or Output AmplifierSensor InterfacePiezoelectric Transducer AmplifierMedical InstrumentationMobile CommunicationAudio OutputPortable SystemSmoke DetectorNotebook PCPCMCIA CardBattery-Powered EquipmentDSP InterfacePACKAGE/ORDERING INFORMATIONMODELPACKAGE DESCRIPTIONSPECIFIED TEMPERATURERANGE ORDERING NUMBER PACKAGE MARKINGPACKING OPTION SGM321SC70-5-40℃ to +85℃ SGM321YC5/TR 321 Tape and Reel, 3000 SOT-23-5 -40℃ to +85℃ SGM321YN5/TR 321 Tape and Reel, 3000 SOT-23-5 -40℃ to +85℃ SGM321BYN5/TR 321B Tape and Reel, 3000 SGM358SOIC-8-40℃ to +85℃ SGM358YS/TR SGM358YS XXXXX Tape and Reel, 4000 MSOP-8 -40℃ to +85℃ SGM358YMS/TR SGM358 YMS XXXXX Tape and Reel, 3000 DIP-8 -40℃ to +85℃ SGM358YP SGM358YP XXXXX 20 Tube (1000pcs) SGM324SOIC-14-40℃ to +85℃SGM324YS14/TRSGM324YS14 XXXXX Tape and Reel, 2500 TSSOP-14-40℃ to +85℃SGM324YTS14/TRSGM324 YTS14 XXXXXTape and Reel, 3000MARKING INFORMATIONNOTE: XXXXX = Date Code and Vendor Code.Date Code - Week Vendor CodeDate Code - YearX XX X XGreen (RoHS & HSF): SG Micro Corp defines "Green" to mean Pb-Free (RoHS compatible) and free of halogen substances. If you have additional comments or questions, please contact your SGMICRO representative directly.ABSOLUTE MAXIMUM RATINGSSupply Voltage, +V S to -V S ................................................. 6V Input Common Mode Voltage Range .................................................... (-V S ) - 0.3V to (+V S ) + 0.3V Package Thermal Resistance @ T A = +25℃ SC70-5, θJA .............................................................. 333℃/W SOT-23-5, θJA .......................................................... 190℃/W SOIC-8, θJA .............................................................. 125℃/W MSOP-8, θJA ............................................................ 216℃/W Junction Temperature ................................................. +150℃ Storage Temperature Range ....................... -65℃ to +150℃ Lead Temperature (Soldering, 10s) ............................ +260℃ ESD Susceptibility HBM ............................................................................. 4000V MM ................................................................................ .400VRECOMMENDED OPERATING CONDITIONSOperating Temperature Range ....................... -40℃ to +85℃OVERSTRESS CAUTIONStresses beyond those listed in Absolute Maximum Ratings may cause permanent damage to the device. Exposure to absolute maximum rating conditions for extended periods may affect reliability. Functional operation of the device at any conditions beyond those indicated in the Recommended Operating Conditions section is not implied.ESD SENSITIVITY CAUTIONThis integrated circuit can be damaged by ESD if you don’t pay attention to ESD protection. SGMICRO recommends that all integrated circuits be handled with appropriate precautions. Failure to observe proper handling and installation procedures can cause damage. ESD damage can range from subtle performance degradation to complete device failure. Precision integrated circuits may be more susceptible to damage because very small parametric changes could cause the device not to meet its published specifications.DISCLAIMERSG Micro Corp reserves the right to make any change in circuit design, or specifications without prior notice.PIN CONFIGURATION SSGM321YN5/SGM321BYN5 (TOP VIEW)+V S-IN-V S +IN OUT+V S-IN-V S+INOUTSOT -23-5/SC70-5SOT -23-5SGM358 (TOP VIEW)OUTA -INA+INA-V S +V S -IND+IND -INB+INB -INC OUTBOUTC+INC OUTDOUTA OUTB +INB+V S-INA +INA -V S-INBSOIC -8/MSOP -8/DIP -8TSSOP -14/SOIC -14ELECTRICAL CHARACTERISTICS(At V S = +5V, R L = 100kΩ connected to V S/2, and V OUT = V S/2, unless otherwise noted.)TYPICAL PERFORMANCE CHARACTERISTICSAt T A = +25℃, V S = 5V, and R L = 100kΩ connected to V S /2, unless otherwise noted.Supply Current vs. TemperatureOpen-Loop Gain vs. TemperatureCommon Mode Rejection Ratio vs. Temperature Power Supply Rejection Ratio vs. TemperatureSupply Current vs. Supply Voltage Short-Circuit Current vs. Supply Voltage50556065707580-50-25255075100125150S u p p l y C u r r e n t /A m p l i f i e r (μA )Temperature (℃)60708090100110120-50-25255075100125150O p e n –L o o p G a i n (d B )Temperature (℃)60708090100110120-50-25255075100125150C M R R (d B )Temperature (℃)60708090100110120-50-25255075100125150P S R R (d B )Temperature (℃)0102030405060708001234567S u p p l y C u r r e n t /A m p l i f i e r (μA )Supply Voltage (V)-2002040608010012001234567S h o r t -C i r c u i t C u r r e n t (m A ) Supply Voltage (V)TYPICAL PERFORMANCE CHARACTERISTICS (continued)At T A = +25℃, V S = 5V, and R L = 100kΩ connected to V S /2, unless otherwise noted.Output Voltage Swing vs. Output CurrentOutput Voltage Swing vs. Output CurrentMaximum Output Voltage vs. Frequency CMRR and PSRR vs. FrequencyInput Voltage Noise Density vs. Frequency Open-Loop Gain and Phase vs. Frequency0123456020406080100120140160O u t p u t V o l t a g e (V )Output Current (mA)102030405060O u t p u t V o l t a g e (V )Output Current (mA)0123456110100100010000O u t p u t V o l t a ge (V p -p )Frequency (kHz)01020304050607080901000.010.1110100100010000C M R R a n d P S R R (d B )Frequency (kHz)1010010000.010.1110100Frequency (kHz)I n o u t V o l t a g e N o i s e D e n s i t y (n V /√H z )-180-150-120-90-60-300-40-20020*******.1110100100010000P h a s e (D e g r e e )O p e n -L o o p G a i n (d B )Frequency (kHz)TYPICAL PERFORMANCE CHARACTERISTICS (continued)At T A = +25Small-Signal Overshoot vs. Load CapacitanceSmall-Signal Step ResponseLarge-Signal Step ResponseTime (2μs/div)Time (10μs/div)Overload Recovery Time2.5V0V500mV0VTime (2μs/div)010203040506010100100010000S m a l l -S i g n a l O v e r s h o o t (%)Load Capacitance (pF)010203040506010100100010000S m a l l -S i g n a l O v e r s h o o t (%)Load Capacitance (pF)G = +1 C L = 100pF R L = 100k ΩG = +1 C L = 100pF R L = 100k ΩO u t p u t V o l t a g e (20m V /d i v )O u t p u t V o l t a g e (500m V /d i v )V S= 5VG = -5V IN = 500mVAPPLICATION NOTESDriving Capacitive LoadsThe SGM321/SGM358/SGM324 can directly drive 250pF in unity-gain without oscillation. The unity-gain follower (buffer) is the most sensitive configuration to capacitive loading. Direct capacitive loading reduces the phase margin of amplifiers and this results in ringing or even oscillation. Applications that require greater capacitive driving capability should use an isolation resistor between the output and the capacitive load like the circuit in Figure 1. The isolation resistor R ISO and the load capacitor C L form a zero to increase stability. The bigger the R ISO resistor value, the more stable V OUT will be. Note that this method results in a loss of gain accuracy because R ISO forms a voltage divider with the R LOAD.V IN V OUTFigure 1. Indirectly Driving Heavy Capacitive LoadAn improved circuit is shown in Figure 2. It provides DC accuracy as well as AC stability. R F provides the DC accuracy by connecting the inverting input with the output. C F and R ISO serve to counteract the loss of phase margin by feeding the high frequency component of the output signal back to the amplifier’s inverting input, thereby preserving phase margin in the overall feedback loop.VOUTFigure 2. Indirectly Driving Heavy Capacitive Load withDC Accuracy For non-buffer configuration, there are two other ways to increase the phase margin: (a) by increasing the amplifier’s closed-loop gain or (b) by placing a capacitor in parallel with the feedback resistor to counteract the parasitic capacitance associated with inverting node.Power Supply Bypassing and LayoutThe SGM321/SGM358/SGM324 can operate from either a single 2.1V to 5.5V supply or dual ±1.05V to ±2.75V supplies. For single-supply operation, bypass the power supply +V S with a 0.1µF ceramic capacitor which should be placed close to the +V S pin. For dual-supply operation, both the +V S and the -V S supplies should be bypassed to ground with separate 0.1µF ceramic capacitors. 2.2µF tantalum capacitor can be added for better performance.V NV PSV NV P-V S (GND)Figure 3. Amplifier with Bypass CapacitorsTYPICAL APPLICATION CIRCUITS Differential AmplifierThe circuit shown in Figure 4 performs the difference function. If the resistor ratios are equal (R4/R3 = R2/R1), then V OUT = (V P - V N) × R2/R1 + V REF.V NV PV OUTFigure 4. Differential AmplifierInstrumentation AmplifierThe circuit in Figure 5 performs the same function as that in Figure 4 but with a high input impedance.V NV PV OUT Figure 5. Instrumentation Amplifier Active Low-Pass FilterThe low-pass filter shown in Figure 6 has a DC gain of (-R2/R1) and the -3dB corner frequency is 1/2πR2C. Make sure the filter bandwidth is within the bandwidth of the amplifier. Feedback resistors with large values can couple with parasitic capacitance and cause undesired effects such as ringing or oscillation in high-speed amplifiers. Keep resistor values as low as possible and consistent with output loading consideration.VV OUTFigure 6. Active Low-Pass FilterREVISION HISTORYNOTE: Page numbers for previous revisions may differ from page numbers in the current version.APRIL 2019 ‒ REV.E.1 to REV.E.2 Page Added Open-Loop Gain and Phase vs. Frequency (6)MARCH 2017‒ REV.E to REV.E.1 Page Updated Package/Ordering Information section (2)NOVEMBER 2015 ‒ REV.D.4to REV.E Page Updated Packing Option of DIP-8 (2)Updated SOIC-14 and TSSOP-14 packages ............................................................................................................................................... 14, 15 JANUARY 2013 ‒ REV.D.3to REV.D.4 Page Added Tape and Reel Information section ................................................................................................................................................... 16, 17PACKAGE OUTLINE DIMENSIONS SOT-23-5Symbol Dimensions In Millimeters Dimensions In Inches MIN MAX MIN MAX A 1.050 1.250 0.041 0.049 A1 0.000 0.100 0.000 0.004 A2 1.050 1.150 0.041 0.045 b 0.300 0.500 0.012 0.020 c 0.100 0.200 0.004 0.008 D 2.820 3.020 0.111 0.119 E 1.500 1.700 0.059 0.067 E1 2.6502.9500.1040.116e 0.950 BSC 0.037 BSC e1 1.900 BSC 0.075 BSC L 0.300 0.600 0.0120.024 θ0°8°0°8°RECOMMENDED LAND PATTERN (Unit: mm)PACKAGE OUTLINE DIMENSIONS SC70-5Symbol Dimensions In Millimeters Dimensions In Inches MIN MAX MIN MAX A 0.900 1.100 0.035 0.043 A1 0.000 0.100 0.000 0.004 A2 0.900 1.000 0.035 0.039 b 0.150 0.350 0.006 0.014 c 0.080 0.150 0.003 0.006 D 2.000 2.200 0.079 0.087 E 1.150 1.350 0.045 0.053E1 2.1502.4500.0850.096e 0.65 TYP 0.026 TYP e1 1.300 BSC 0.051 BSC L 0.525 REF 0.021 REF L1 0.260 0.460 0.010 0.018 θ0°8° 0°8°RECOMMENDED LAND PATTERN (Unit: mm)PACKAGE OUTLINE DIMENSIONS SOIC-8Symbol Dimensions In Millimeters Dimensions In Inches MIN MAX MIN MAX A 1.350 1.750 0.053 0.069 A1 0.100 0.250 0.004 0.010 A2 1.350 1.550 0.053 0.061 b 0.330 0.510 0.013 0.020 c 0.170 0.250 0.006 0.010 D 4.700 5.100 0.185 0.200 E 3.800 4.000 0.150 0.157 E1 5.8006.2000.228 0.244 e 1.27 BSC0.050 BSCL 0.400 1.270 0.016 0.050 θ0°8° 0°8°RECOMMENDED LAND PATTERN (Unit: mm)PACKAGE OUTLINE DIMENSIONS MSOP-8Symbol Dimensions In Millimeters Dimensions In Inches MIN MAX MIN MAX A 0.820 1.100 0.032 0.043 A1 0.020 0.150 0.001 0.006 A2 0.750 0.950 0.030 0.037 b 0.250 0.380 0.010 0.015 c 0.090 0.230 0.004 0.009 D 2.900 3.100 0.114 0.122 E 2.900 3.100 0.114 0.122 E1 4.750 5.050 0.187 0.199 e 0.650 BSC 0.026 BSCL 0.400 0.800 0.016 0.031θ0°6°0°6°cRECOMMENDED LAND PATTERN (Unit: mm)PACKAGE OUTLINE DIMENSIONS SOIC-14Symbol Dimensions In Millimeters Dimensions In Inches MIN MAX MIN MAX A 1.35 1.75 0.053 0.069 A1 0.10 0.25 0.004 0.010 A2 1.25 1.65 0.049 0.065 A3 0.55 0.75 0.022 0.030 b 0.36 0.49 0.014 0.019 D 8.53 8.73 0.336 0.344 E 5.80 6.20 0.228 0.244 E1 3.804.000.150 0.157 e 1.27 BSC 0.050 BSCL 0.450.80 0.0180.032L1 1.04 REF 0.040 REF L2 0.25 BSC 0.01 BSCR 0.07 0.003 R1 0.07 0.003 h 0.30 0.50 0.012 0.020 θ0°8° 0°8°RECOMMENDED LAND PATTERN(Unit: mm)PACKAGE OUTLINE DIMENSIONS TSSOP-14Symbol Dimensions In Millimeters DimensionsIn Inches MIN MAX MIN MAX A 1.200 0.047 A1 0.050 0.150 0.002 0.006 A2 0.800 1.050 0.031 0.041 b 0.190 0.300 0.007 0.012 c 0.090 0.200 0.004 0.008 D 4.860 5.100 0.1910.201 E 4.300 4.500 0.169 0.177 E1 6.250 6.550 0.246 0.258 e 0.650 BSC 0.026 BSCL 0.5000.7000.020.028H 0.25 TYP 0.01 TYP θ1°7° 1°7°RECOMMENDED LAND PATTERN (Unit: mm)PACKAGE OUTLINE DIMENSIONS DIP-8Symbol Dimensions In Millimeters Dimensions In Inches MIN MAX MIN MAX A 3.710 4.310 0.146 0.170 A1 0.510 0.020 A2 3.200 3.600 0.126 0.142 b 0.380 0.570 0.015 0.022 b1 1.524 BSC 0.060 BSCc 0.204 0.360 0.008 0.014 D 9.000 9.400 0.354 0.370 E 6.200 6.600 0.244 0.260 E1 7.320 7.920 0.288 0.312 e 2.540 BSC 0.100 BSCL 3.000 3.600 0.118 0.142 E28.4009.0000.3310.354TAPE AND REEL INFORMATIONNOTE: The picture is only for reference. Please make the object as the standard.KEY PARAMETER LIST OF TAPE AND REELPackage Type Reel DiameterReel WidthW1(mm)A0(mm)B0 (mm) K0 (mm) P0 (mm) P1 (mm) P2 (mm) W (mm) Pin1 QuadrantDD0001SOT-23-5 7″ 9.5 3.20 3.20 1.40 4.0 4.0 2.0 8.0Q3 SC70-5 7″ 9.5 2.25 2.551.20 4.0 4.02.0 8.0 Q3 SOIC-8 13″ 12.4 6.40 5.40 2.10 4.0 8.0 2.0 12.0 Q1 MSOP-8 13″ 12.4 5.203.30 1.504.0 8.0 2.0 12.0 Q1 SOIC-14 13″ 16.4 6.60 9.30 2.10 4.0 8.0 2.0 16.0 Q1TSSOP-1413″12.46.955.601.204.08.02.012.0Q1REEL DIMENSIONS TAPE DIMENSIONS DIRECTION OF FEEDCARTON BOX DIMENSIONSNOTE: The picture is only for reference. Please make the object as the standard. KEY PARAMETER LIST OF CARTON BOXReel Type Length(mm) Width(mm)Height(mm) Pizza/CartonDD00027″ (Option)368 227 224 8 7″442 410 224 18 13″386 280 370 5。
启英泰伦双麦离在线语音识别模组硬件规格书 CI-B03ST01S-BK说明书
双麦离在线语音识别模组硬件规格书型号:CI-B03ST01S-BK版本:V1.0文件历史跟踪DOCUMENT HISTORY PAGE版本号Rev.NO.发起者Originator描述Description日期Date V1.0启英泰伦新建文档2020/05/10声明本手册由成都启英泰伦科技有限公司版权所有,未经许可,任何单位和个人都不得以电子的、机械的、磁性的、光学的、化学的、手工的等形式复制、传播、转录和保存该出版物,或翻译成其他语言版本。
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目录1概述 (5)2模组主芯片描述 (7)3模组实物图 (10)4硬件接口定义 (12)5电路设计参考 (15)5.1电源 (15)5.2ADC接口 (15)5.3升级接口 (15)5.4PWM (15)5.5ESD设计 (15)5.6GPIO (16)5.7UART (16)5.8I2S音频接口 (16)5.9调试接口 (16)5.10DAC输出 (16)6电气参数 (17)7使用注意事项 (18)Chipintelli Technology Co.,Ltd.CONFIDENTIAL ALL RIGHTS RESERVED.This document is not to be reproduced,modified,adapted,published, translated in any material form in whole or in part nor disclosed to any third party without the prior writtenpermission of Chipintelli Technology Co.,Ltd.1概述启英泰伦双麦离在线语音识别模组CI-B03ST01S-BK是针对离在线应用开发的一款通用、便携、低功耗高性能语音识别模组,该模组具有以下功能和特点:⏹模组搭载高性能CI1103语音识别芯片,通过该芯片内置的基于深度神经网络(DNN)的硬件处理器,内置100条命令词离线语音识别功能,不依赖网络,时延小,且可保护用户隐私。
《2024年基于麦克风阵列的语音增强研究》范文
《基于麦克风阵列的语音增强研究》篇一一、引言随着人工智能技术的快速发展,语音识别和语音交互技术已成为人们日常生活和工作中不可或缺的一部分。
然而,在复杂多变的实际环境中,语音信号常常受到各种噪声的干扰,导致语音质量下降,进而影响语音识别的准确性和语音交互的体验。
因此,如何有效地进行语音增强,提高语音信号的信噪比(SNR),成为了一个重要的研究课题。
麦克风阵列技术因其能够通过多个麦克风的协同作用,实现空间滤波和声源定位,为语音增强提供了新的解决方案。
本文将就基于麦克风阵列的语音增强研究进行深入探讨。
二、麦克风阵列技术概述麦克风阵列是由多个麦克风按照一定几何结构排列组成,通过采集声波到达各个麦克风的相位差和幅度差,实现声源定位和语音信号处理。
麦克风阵列技术具有空间分辨率高、抗干扰能力强、适用于复杂环境等优点,在语音识别、语音交互、机器人听觉等领域有着广泛的应用。
三、基于麦克风阵列的语音增强方法基于麦克风阵列的语音增强方法主要包括波束形成、噪声抑制和语音分离等技术。
1. 波束形成波束形成是麦克风阵列技术中常用的一种方法,它通过加权求和各个麦克风的信号,形成指向性波束,从而提高目标语音的信噪比。
常见的波束形成方法包括延迟求和波束形成、相位变换波束形成等。
2. 噪声抑制噪声抑制是针对麦克风阵列接收到的语音信号中的噪声进行处理,以降低噪声对语音质量的影响。
常见的噪声抑制方法包括谱减法、非负矩阵分解等。
在麦克风阵列中,可以通过空间滤波和声源定位,更准确地识别并抑制噪声。
3. 语音分离语音分离是通过分析多个声源的信号特征,将不同声源的语音信号分离出来。
在麦克风阵列中,可以利用声源定位技术,确定各个声源的位置,然后通过信号处理技术将不同声源的语音信号分离出来。
四、实验与分析为了验证基于麦克风阵列的语音增强方法的有效性,我们进行了相关实验。
实验结果表明,通过波束形成、噪声抑制和语音分离等技术,可以有效提高语音信号的信噪比,改善语音质量。
ffmpeg silencedetect 原理
ffmpeg silencedetect 原理【原创版】目录1.介绍 ffmpeg2.详述 ffmpeg 的 silencedetect 功能3.分析 silencedetect 的原理4.总结 ffmpeg silencedetect 的优点和不足正文一、介绍 ffmpegffmpeg 是一套跨平台的音视频处理软件,可以用于转换、编码、解码、录制、处理各种音视频格式。
它的音视频处理能力非常强大,广泛应用于电影制作、视频剪辑、音频转换等领域。
二、详述 ffmpeg 的 silencedetect 功能ffmpeg 的 silencedetect 功能可以检测视频中的静音部分,即音频信号低于指定阈值的部分。
这个功能对于处理视频中的无声片段非常有用,例如在剪辑视频时,可以快速找到无声的部分并进行处理。
三、分析 silencedetect 的原理silencedetect 功能的原理是基于音频信号的能量计算。
ffmpeg 会读取视频文件的音频流,并计算每一帧音频信号的能量。
当音频信号的能量低于指定阈值时,ffmpeg 就会认为这一帧是静音,将其标记为无声片段。
四、总结 ffmpeg silencedetect 的优点和不足ffmpeg 的 silencedetect 功能可以帮助用户快速找到视频中的无声片段,提高剪辑效率。
但是,这个功能也有一定的局限性。
首先,它只能检测到音频信号低于阈值的部分,对于静音时间较长的片段,可能无法准确检测。
其次,silencedetect 功能依赖于音频信号的能量计算,当音频信号质量较差时,可能会影响检测结果的准确性。
总的来说,ffmpeg 的 silencedetect 功能在处理视频无声片段方面非常有用,但也存在一定的局限性。
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)。
录音截断器测试方案
录音截断器测试方案1. 引言录音截断器是一种用于处理音频信号的电子设备,可以将长时间的音频信号截断成多个短片段。
本文档将介绍录音截断器的测试方案,以确保其功能的正确性和稳定性。
2. 测试目标录音截断器的测试目标是验证其在不同输入条件下的截断功能,以及确保其输出的音频片段符合预期的要求。
3. 测试环境搭建搭建一个测试环境来模拟真实的录音场景是测试录音截断器的重要步骤。
下面是搭建测试环境的步骤:1.准备一台录音设备:可以是专业的录音设备或者智能手机等。
2.准备音频信号源:可以是各种类型的音乐、语音等。
3.连接录音设备和音频信号源。
4.确保录音设备能够正常录制音频。
4. 测试方案4.1 输入测试输入测试是验证录音截断器在不同输入条件下的截断功能。
以下是输入测试的步骤:1.启动录音截断器。
2.设置截断参数,如截断时间间隔和截断时长。
3.开始录制音频信号。
4.根据设定的时间间隔,截断音频信号,并记录截断时间点。
5.检查截断的音频片段是否符合要求:长度是否达到设置的时长,截断时间点是否准确。
4.2 输出测试输出测试是验证录音截断器输出的音频片段是否符合预期要求。
以下是输出测试的步骤:1.启动录音截断器。
2.设置截断参数,如截断时间间隔和截断时长。
3.开始录制音频信号。
4.根据设定的时间间隔,截断音频信号,并记录截断时间点。
5.检查截断的音频片段是否符合要求:长度是否达到设置的时长,截断时间点是否准确。
6.播放截断的音频片段,检查音频质量是否正常,没有噪音或失真等问题。
4.3 性能测试性能测试是验证录音截断器在大量音频输入时的稳定性和性能表现。
以下是性能测试的步骤:1.准备大量不同类型的音频信号作为输入。
2.启动录音截断器。
3.设置适当的截断参数。
4.开始录制音频信号。
5.持续运行录音截断器,记录截断的音频片段数目和时间点。
6.检查截断的音频片段是否符合预期要求。
同时,观察录音截断器的运行状态,包括内存和CPU占用情况等。
基于傅里叶卷积的多通道语音增强
doi:10.3969/j.issn.1003-3106.2024.03.009引用格式:孙思雨,张海剑,陈佳佳.基于傅里叶卷积的多通道语音增强[J].无线电工程,2024,54(3):580-588.[SUNSiyu,ZHANGHaijian,CHENJiajia.Multi channelSpeechEnhancementBasedonFourierConvolution[J].RadioEngineering,2024,54(3):580-588.]基于傅里叶卷积的多通道语音增强孙思雨,张海剑,陈佳佳(武汉大学电子信息学院,湖北武汉430072)摘 要:神经波束形成器(NeuralBeamformer)的构建是处理多通道语音增强任务的主要方法之一,其通过求解波束权值对多通道信号进行滤波从而获得纯净语音。
与传统波束求解空间协方差矩阵的原理类似,频谱信息和空间线索在神经波束形成器的波束权值估计中也起着至关重要的作用。
由于缺乏对频谱和空间信息的充分学习,现有许多工作无法对波束权值进行最优估计。
为应对这一挑战,构建了一种基于傅里叶卷积的上下文特征提取器,在频率轴上具有全局感受野,并加入时频卷积模块对时间上下文信息建模,增强对输入频谱图上下文信息的学习;采用了一种新的卷积循环网络(ConvolutionalRecurrentNetwork,CRN)结构,其编解码模块中嵌入了所提的上下文特征提取器,并在跳连接中嵌入卷积注意力模块(ConvolutionalBlockAttentionModule,CBAM)。
所提出的CRN结构能充分从输入特征频谱图中捕获时频上下文信息以及跨通道的空间信息。
实验结果表明,该方法参数量仅1.14M,与目前先进的基线系统对比达到最优性能。
关键词:多通道;语音增强;神经波束形成器;傅里叶卷积;深度学习中图分类号:TN911.7文献标志码:A开放科学(资源服务)标识码(OSID):文章编号:1003-3106(2024)03-0580-09Multi channelSpeechEnhancementBasedonFourierConvolutionSUNSiyu,ZHANGHaijian,CHENJiajia(SchoolofElectronicInformation,WuhanUniversity,Wuhan430072,China)Abstract:Theconstructionofneuralbeamformerisoneofthemainmethodstodealwithmulti channelspeechenhancementtasks,whichfiltersthemulti channelsignalstoobtaintargetspeechbysolvingthebeamweights.Similartotheprincipleofthesolutionofspatialcovariancematrixintraditionalbeamforming,spectral spatialinformationalsoplaysacrucialroleinthebeamweightspredictionofneuralbeamformer.However,duetothelackofadequatelearningofspectral spatialinformation,manyexistingeffortsfailtooptimallypredictthebeamweights.Inordertodealwiththischallenge,acontextfeatureextractorbasedonFourierconvolutionisproposed,withwhichaglobalreceptivefieldonthefrequencyisinvolved.Besides,themodelingoftemporalcontextinformationisalsorealizedbyaddingatime frequencyconvolutionalmoduletoboostthelearningofcontextfrominputspectrograms.Inaddition,aConvolutionalRecurrentNetwork(CRN)structureisapplied,inwhichtheproposedcontextfeatureextractorisembeddedintheencodersanddecoders,andaConvolutionalBlockAttentionModule(CBAM)isinvolvedintheskipconnection.TheproposedCRNstructurecancapturethetime frequencycontextinformationandcross channelspatialfeaturessufficientlyfromtheinputspectrograms.Experimentalresultsshowthattheparameterquantityoftheproposedapproachisonly1.14M,whichindicatesgreatsuperiorityovertheexistingadvancedbaselinesystems.Keywords:multi channel;speechenhancement;neuralbeamformer;Fourierconvolution;deeplearning收稿日期:2023-08-15基金项目:湖北省自然科学基金(2022CFB084)FoundationItem:HubeiProvincialNaturalScienceFoundationofChina(2022CFB084)0 引言语音增强问题是语音信号处理领域的研究热点,广泛应用于助听器、远场语音识别等语音通信场景[1-4]。
语音识别中的声学特征提取使用教程
语音识别中的声学特征提取使用教程语音识别技术已经在我们的生活中发挥着越来越重要的作用。
为了提高语音识别系统的精确性和准确性,声学特征提取是一个至关重要的步骤。
在本篇文章中,我将详细介绍语音识别中声学特征提取的使用教程。
无论您是新手还是有经验的开发者,都可以从中获得一些有用的信息。
声学特征提取是将语音信号转换为一系列数学特征的过程。
这些特征包含了语音信号的重要信息,可以帮助识别和区分不同的语音单元。
在语音识别中常用的声学特征提取方法包括梅尔频率倒谱系数(MFCC)、线性预测编码(LPC)、梅尔频率包络(MFE)等。
下面将逐一介绍这些常用的声学特征提取方法。
首先,梅尔频率倒谱系数(MFCC)是一种广泛应用的声学特征提取方法。
它模拟了人耳对声音频率的感知特性。
MFCC的主要步骤包括预加重、分帧、加窗、快速傅里叶变换(FFT)、梅尔滤波器组和离散余弦变换(DCT)。
其中,预加重通过对语音信号进行高频增强,可以提高MFCC的提取效果。
分帧将长时间的语音信号分割为短时间的帧,通常采用20-30毫秒的帧长。
加窗是将每个帧进行加窗处理,常用的窗函数有汉明窗、汉宁窗等。
FFT将每个帧从时域转换到频域,得到每个频率的幅度谱。
梅尔滤波器组将频域的振幅谱映射到梅尔频率的刻度上,以模拟人耳对声音频率的感知。
最后,DCT将梅尔滤波器组的输出转换为倒谱系数,作为最终的声学特征。
其次,线性预测编码(LPC)是一种基于线性预测模型的声学特征提取方法。
LPC通过建立语音信号的线性预测模型,将语音信号分解为预测误差和线性预测系数。
LPC的主要步骤包括帧分割、自相关函数计算、勒维尔算法(递归最小二乘法)求解线性预测系数、预测误差计算等。
帧分割和加窗与MFCC类似,自相关函数计算用于求解线性预测系数,勒维尔算法通过最小化预测误差的平方和来求解线性预测系数。
而预测误差则表示了语音信号与线性预测模型之间的差异。
最后,梅尔频率包络(MFE)是一种在声学特征提取中不那么常用的方法。
迈克罗纳 DX2000USB 专业级7通道低噪声DJ混音器说明书
DX2000USBProfessional, 7-channel, ultra-low noise DJ mixer with state-of-the-art phono preamps45-mm infi nium “contact-free” optical crossfader with adjustable friction for years of use 5 dual stereo inputs plus 2 mono mic/line channels with ULN mic preamps, Phantom Power, Gain control and Clip LED Built-in USB interface for recording and playback of any digital music fi le. Works with your PC or Mac computer— no setup or drivers requiredMassive software bundle including Audacity vinyl restoration and recording, Podifi er and Golden Ear podcasting software downloadable at Ultra-musical 3-band kill EQ (-32 dB) with EQ on/off switch on all stereo channelsLong-wearing 100-mm faders and sealed rotary controls on all input channelsHeadphone output with Level and Balance (PFL/Main) controls, switchable split mode and level displayAuto-talkover function with separate Sensitivity, Time and Damping controlsMain Dim, main Boost, Punch and Cut functions for awesome mix optionsThese days a DJ gig can command a lot more than a mic and a couple of turntables. You may need toaccommodate a multimedia presentation, a variety of audio platforms (vinyl, CDs, mp3) and multiple mics—all at the same event! That’s why BEHRINGER is proud to present the 7-channel, rack-mountable DX2000USB with its built-in USB/audio interface and contact-free optical crossfader—an invaluable tool for the DJ-of-all-trades.The inclusion of the X1 opticalcrossfader in the DX2000USB represents tremendous added value. The “feel” of this crossfader can be custom-tailored to accommodate your individual mixing style, and the contact-free technology means silky-smooth performance for many years to come. We off er the X1 as a premium upgrade for our DDM4000 digital DJ mixers at $74.99 USD, but it comes standard with the DX2000USB at no additional cost. Please visit our website to fi nd out more about the X1 optical fader.Live Large With More Mediums!If it can be clicked, spun or spoken, the new DX2000USB can mix it! You get 5 dual stereo inputs plus 2 mono mic/line channels with ultra low-noise mic preamps, Phantom Power, Gain control andClip LED. The DX2000USB also has enough built-in phono preamps to handle up to three turntables. Finally, the built-in USB/audio interface gives you the power to add your computer’s digital music library to the mix. Hey, we even include the rack-ears, so you’re ready-to-go right out of the box!Old-school feel,new-school performanceThe DX2000USB is packed with cutting- edge versatility, but we didn’t lose sight of the basic features that make a DJ mixer a lasting part of your rig. This new breed of DJ mixer features super-smooth, long-wearing 100 mm faders and the infi nium optical contact-free crossfader forProfessional 7-Channel DJ Mixer with infi nium “Contact-Free” VCACrossfader and USB/Audio InterfaceContinued on next pagePRO MIXER DX 2000USBDJ Mixersutmost reliability and smooth audio performance. It is important to note that since this premium crossfader iscontact-free, there is none of the typically “wear and tear” that plagues mechanical faders, so your crossfades are always fl awless—and will be for years to come. The DX2000USB comes armed with a switchable 3-band kill EQ and Gain control per stereo channel.Mono channels feature an additional Low Cut switch and an FX switch that routes the channel to the FX SEND and RETURN loop, so you can add a touch of outboard eff ects. These channels feature balanced XLR mic inputs, as well as ¼ " line jacks, a Channel On button and aPFL switch for headphone monitoring. The handy Talkover button allows you to make announcements without killing the music via three adjustable parameters, Sensitivity (speech level threshold), Time (speed at which the music level returns to normal) and Damping (depth the music is attenuated by the mic signal).For the DJ in need of mammoth bass, the DX2000USB features an XLR subwoofer output (perfect for connecting to an active subwoofer) with adjustable (30 - 200Hz) Crossover Frequency dial and Level control.Directly above, the DX2000USB provides a ¼" main mono ouput for synching your music’s strongest pulses with aSubbass Out for separate subwoofer, additional Zone output for second room/area“Planet Earth” switching power supply for maximum fl exibility (100 - 240 V~), noise-free audio, superior transient response plus low power consumption for energy savingRack mount brackets included for ultimate fl exibility High-quality components and exceptionally rugged construction ensure long lifeConceived and designed by BEHRINGER GermanyContinued on next pageTRS ZONE outputs Main XLR outputs Ground lugs for turntablesRCA LINE inputs Built-in USB audio interfaceRCA RECORD OUT jacks TRS monitor outputsLINE/PHONO button optimizes PHONO/LINE jacks for use with turntable o r C D p layerTRS LINE INPHONO/LINE RCA inputsLEVEL knob adjusts output of SUB BASS OUT jack for use with subwooferCROSSOVERFREQUENCY knobadjusts frequency of low pass fi lter for use with subwooferFX SEND/RETURN jacksXLR monitoroutputs IEC Power socket and on/off switch TRS REMOTE CONTROL jackCD RCA inputs TRS LINE inputs Stereo Main Insert Balanced/Unbalanced mono output to external lighting controllerXLR MIC INlighting system, provided your lighting system has “sound-to-light” capability(e.g. BEHRINGER LC2412 Lighting Console). The Main Output section off ers balanced ¼" TRS and XLR stereo outputs, as well as ¼ " TRS Main Insert jacks for connecting to outboard EQ and dynamics processors, such as limiters and compressors. This section also features RCA outputs for recording your set. The Monitor/Zone Output sectiongives you separate stereo ¼ " and XLR monitor outputs, as well as ¼ " stereo zone outputs, each with their own dedicated level controls. This feature gives the DX2000USB incredible zone mixing versatility, allowing you to send output signals to a variety of destinations, such as the lobby, restaurant or waiting areas, all the while maintaining total control.Transforming with Punch and Cut Transforming is a DJ term used to describe the chopping up of sound to create some rather dramatic eff ects. Traditionally this is done by rapidlymoving a crossfader to give a stuttering or “gated” eff ect, either between twomusic sources, or one source andsilence. Another DJ trick is to use the channel faders, or channel on buttons, to chop one music track over another. These methods are still valid, but if you’re looking for instant gratifi cation, check out our ergonomic alternative— a pair of big assignable non-latching PUNCH/CUT buttons. We’re sure you will grow to love their speed and ease of operation.The Punch and Cut modes are available via the TRANSFORM MODE button, with a pair of LEDs that let you know which function is active. The X and Y Punch/Cut buttons are conveniently located right next to the crossfader with which they are designed to work.INPUTbutton selects channel’s audio sourceLOW CUT button removes unwanted low frequencies EQ knobs PAN knobpositions channel in stereo fi eld TALKOVER button CHANNEL ON sends channel to PFL bus CROSSFADER button routes signal to crossfader CHANNEL fader adjusts channel volume ASSIGN button assigns active channel to X or Y side of crossfaderGainFX button routes channel to FX SEND and RETURN loopCD/PHONO input selectorDAMPING adjusts music level reduction when talkover is engaged TIME knob adjusts howquickly the music volume recovers after talkover has been engagedSENSITIVITYknob adjusts how loud mic signal must be to engage talkover LEVEL METER displays left and right MAIN signal, PFL signal levelLIGHT LEVEL adjusts output level of mono LIGHT OUTPUTZONE LEVEL adjusts output of ZONE OUTPUTS FX RETURN MAIN/PFL selects main stereo mix or PFL mix for phones PHONES LEVEL BALANCE adjusts blend of PFL and main signals in headphones MONITOR fader adjusts output of MONITOR OUTPUTSMAIN fadersinfi nium optical X/Y CROSSFADERPUNCH/CUT buttonsMAIN DIM button reduces the main mix by -20 dBMAIN BOOST raises the main mix by +4 dBTRACK START button allows remote control of commercial playback decks via built-in remote jacksTRANSFORM MODE determines whether PUNCH/CUT buttons operate in CUT or PUNCH modeEQ ON Continued on next pageCUT mode enables the big buttons to be used as mutes for interesting gating eff ects, temporarily silencing the X or Y output.In PUNCH mode the X button introduces the X signal to the mix, while the Y button brings in the Y signal. This means you can add in bits and beats from X on top of Y and vice versa, opening up your scope to extremely creative mixing.MAIN DIM and MAIN BOOSTYou can add even more dramatic eff ects by using the MAIN BOOST and MAIN DIM buttons for a temporaryboost (+4 dB) or cut (-20 dB) of the main output volume. For instance, you could use MAIN DIM for audience sing-along bits—or you could use MAIN BOOST to emphasize beats, a special musical passage, etc.Remote ControlIf you use external audio devices, such as CD players, CART machines, etc.,in the course of your performance, you’ll be pleasantly surprised by DX2000USB’s remote control feature. Many commercial playback decks off er remote start and stop functionality. If your player supports this feature (please check your device’s owner’s manual), you can use DX2000USB’s TRACK START buttons, conveniently located below the faders on channels 3-7, to control up to three similarly equipped devices. This may seem like a minor detail, but it can become a major hassle when you’re trying to produce the perfect show!B212D Active PA LoudspeakersFEX800 FX ProcessorXM8500Dynamic MicsB315D Active PA LoudspeakersHPX4000DJ HeadphonesB1800D-PRO Active Subwoofer CD PlayerTurntableFX2000Digital Effects ProcessorLaptopComplete Mobile DJ SystemContinued on next pageSuper SoftwareYou get incredible multitracking software and more when you hook up with the DX2000USB. This amazing DAW (digital audio workstation) makes it easy to manipulate your audio and MIDI fi les, turning song ideas into stunning CD or web-ready recordings. There’s even a built-in drum machine and internal multi-FX processor. If you want toincorporate sounds from old, damaged vinyl, the included Audacity freeware audio editor with vinyl restoration capability will remove pops and othernoise. You even get Golden Ear and Podifi er software so you can instantly share your creations via podcast.ValueWhen you add the DX2000USB to your rig, you connect to unrivaled versatility and performance at a price that will leave you with cash to spare. Its superior build quality and rugged components mean this mixer will serve you well for years to come. Drop in at your nearest BEHRINGER dealer and fi nd out why so many DJs are making magic with BEHRINGER.XL3200Live MixerB212D Active PA LoudspeakersXM8500Dynamic MicsHPX4000DJ HeadphonesCD Player FX2000Digital Effects ProcessorTurntableLaptopMP3playerSimple Setup with Club PA SystemFor service, support or more information contact the BEHRINGER location nearest you:EuropeMUSIC Group Services EU GmbHTel.: +49 2154 9206 4149 / Fax: +49 2154 9206 4199USA/Canada MUSIC Group Services US Inc.Tel.: +1 425 672 0816 / Fax: +1 425 673 7647SingaporeMUSIC Group Services SG (Pte.) LtdTel.: +65 6845 1800 / Fax: +65 6214 0275AustraliaMUSIC Group Services AU Pty LtdTel.: +61 03 9877 7170 / Fax: +61 03 9877 7870JapanMUSIC Group Services JP K.K.Tel.: +81 3 5281 1180 / Fax: +81 3 5281 1181TECHNICAL SPECIFICATIONS AND APPEARANCES ARE SUBJECT TO CHANGE WITHOUT NOTICE AND ACCURACY IS NOT GUARANTEED. BEHRINGER IS PART OF THE MUSIC GROUP (MUSIC-GROUP .COM). ALL TRADEMARKS ARE THE PROPERTY OF THEIR RESPECTIVE OWNERS.MUSIC GROUP ACCEPTS NO LIABILITY FOR ANY LOSS WHICH MAY BE SUFFERED BY ANY PERSON WHO RELIES EITHER WHOLLY OR IN PART UPON ANY DESCRIPTION, PHOTOGRAPH OR STATEMENT CONTAINED HEREIN. COLORS AND SPECIFICATIONS MAY VARY FROM ACTUAL PRODUCT. MUSIC GROUP PRODUCTS ARE SOLD THROUGH AUTHORIZED FULFILLERS AND RESELLERS ONLY. ALL RIGHTS RESERVED. © 2011 MUSIC Group IP Ltd. Trident Chambers, Wickhams Cay, P .O. Box 146, Road Town, Tortola, British Virgin Islands. 985-10000-00466Specifi cationsMono Input ChannelsMic input Electronically balanced, discrete input confi guration Gain+10 to +60 dBFrequency response 10 Hz to 80 kHz, +/-3 dB THD0.08 % typ. @ -30 dBu, 1 kHzLine inputGain-10 to +40 dBFrequency response 10 Hz to 80 kHz, +/-3 dB THD 0.08 % typ. @ -10 dBu, 1 kHz S/N ratio85 dB, unweightedEQLow +/-12 dB @ 50 Hz Mid +/-8 dB @ 750 Hz High +/-12 dB @ 10 kHz Low cut75 Hz, 18 dB/octStereo Input ChannelsPhone/Line/CD input Unbalanced input GainLine/CD +17/-20 dBPhone +17/-20 dBFrequency response Line/CD 10 Hz to 130 kHz, +/-3 dB Phone 20 Hz to 20 kHz, RIAA THD Line/CD 0.05 % typ @ 0 dBu, 1 kHz Phone 0.1 % typ @ -40 dB, 1 kHz S/N ratio Line/CD -82 dB, unweighted Phone -78 dB, unweighted Kill EQ Low +6/-25 dB @ 50 Hz Mid +6/************High+6/-18 dB @ 15 kHzConnectorsMaster outJack 0 dBXLR +6 dB Monitor out Jack 0 dB (max.10 dB gain)Zone out Jack 0 dB (max.10 dB gain)Insert send 0 dB Insert return 0 dB Eff ect send 0 dBUSBAudio Stereo In/Out Connector Type B Sample rate48 kHzPower SupplyMains Voltages USA/Canada 120 V ~, 60 H UK/Australia 240 V ~, 50 Hz Europe230 V ~, 50 HzGeneral Export Model 100 - 120 V ~, 200 - 240 V ~,50 - 60 HzPhysicalDimensions (Hx W x D) 6.29 x 13.38 x 17.32"160 x 340 x 440 mm Weight 14.77 lbs / 6.7 kgPlease note these specifications are preliminary and conceptual in nature, and as such are subject to change as product development progresses. This information is supplied for market research purposes only and is not to be made public in any manner. This document is solely the property of The MUSIC Group, or one of its subsidiaries, and must be surrendered upon request of the owner.DX2000USB。
惠普 电脑 说明书
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目录安装电脑 (1)组装电脑 (1)连接到电脑 (2)首次启动您的电脑 (3)建立并连接到您的 Internet 帐户 (4)安装软件和硬件设备 (5)使用 HP 产品 (5)将旧电脑中的信息和文件转移到新电脑中 (5)使用桌面图标 (6)使用 HP Organize 软件 (6)使用键盘 (7)自定义键盘按钮 (9)使用电脑机箱顶部 (11)使用支撑架 (11)设置声音 (15)使用扬声器 (15)设置 3 接口的声音 (16)为 DVD 播放机设置多声道音频输出 (18)设置 6 接口的声音 (18)确定音频配置软件 (20)使用 Sound Effect Manager(6 接口) (20)使用 Sound Effect Manager(6接口)配置扬声器 (20)配置声音以使用 Sound Effect Manager 录音(6接口) (21)使用 Realtek HD Sound Effect Manager(6 接口、多音源音频) (23)配置 Realtek HD Sound Effect Manager(6 接口、多音源音频) (23)使用 Realtek HD Sound Effect Manager(6 接口、多音源音频)配置用于录音的声音设置 (24)目录v使用 Realtek HD Sound Effect Manager(6 接口、多音源音频)重新定义音频接口 (24)配置多音源音频(6 接口、多音源音频) (24)设置多音源音频 (26)使用读卡器 (29)使用读卡器 (29)媒体插入指南 (30)使用安全删除硬件实用程序 (32)将电视用作显示器 (33)确认您可能用到的电缆 (33)连接到电视机 (33)在电视屏幕上观看电脑画面 (34)nView 选项卡 (35)Ge Force 选项卡 (35)显示选项卡 (36)禁用电视选项 (36)nView 选项卡(禁用) (37)Ge Force 选项卡(禁用) (37)显示选项卡(禁用) (37)断开与电视的连接 (38)使用 HP Personal Media Drive (39)连接硬盘驱动器 (39)插入硬盘驱动器 (40)定位硬盘驱动器并为之分配一个盘符 (41)使用硬盘驱动器 (42)断开硬盘驱动器 (43)处理数码图像 (45)处理数码图像 (45)使用 HP Image Zone (45)播放 CD 和 DVD (49)用 iTunes 播放音乐 CD (49)用 Windows Media Player 播放 CD 和 DVD (50)使用 InterVideo WinDVD 播放 DVD 影片 (50)使用国家/地区编码 (52)使用 InterVideo WinDVD 播放 VCD(影碟) (52)使用 Windows Media Player 播放 VCD(影碟) (53)vi入门指南制作音频和数据光盘 (55)刻录前请擦除光盘上的内容 (56)制作音频 CD (56)验证是否音频文件刻录无误 (57)音频 CD 提示 (57)制作音频 CD (58)制作 jukebox 光盘 (58)制作数据 DVD 和 CD (59)验证数据光盘刻录是否无误 (59)数据光盘提示 (60)制作数据光盘 (60)复制 DVD 或 CD (61)将文件归档至 CD 或 DVD (62)处理映像文件 (62)制作映像文件 (63)从映像文件进行烧录 (63)使用 LightScribe 制作光盘标签 (63)Lightscribe 需求 (64)使用 Sonic Express Labeler 烧录 LightScribe 标签 (64)使用 iTunes 烧录后刻录 LightScribe 标签 (65)使用胶粘光盘标签 (66)兼容性信息 (66)光盘特性及兼容性图表 (67)光驱快速参考图表 (68)软件快速参考图表 (69)制作视频光盘和影片 (71)开始捕获影片之前 (71)Sonic MyDVD Video 项目 (72)制作视频项目 (72)添加将文件到视频项目 (73)为视频项目捕获视频 (73)将幻灯片添加到视频项目 (74)将子菜单添加到视频项目 (74)编辑视频项目的样式 (75)编辑影片文件 (75)将视频项目刻录到光盘 (76)使用 Direct-to-Disc 向导制作视频项目 (77)视频质量和光盘尺寸 (78)DVD 刻录时间 (79)目录vii使用 muvee autoProducer 制作影像 (81)制作影像的基本步骤 (81)查找 muvee autoProducer (82)使用 muvee autoProducer (83)入门 (83)从数码视频摄像机截取影像 (84)加入影像 (85)加入图片 (86)加入音乐 (86)选择风格 (87)更改设置 (87)加入开场标题和结束署名 (88)制作影像 (89)保存影像项目 (89)将影像项目刻录到光碟 (90)获取帮助 (91)查找联机指南 (91)从网络上查找指南 (91)使用联机的“帮助和支持中心” (92)使用 HP 更新信息 (92)获取信息 (92)关闭消息 (93)重新激活消息 (93)使用 PC 帮助和工具 (93)索引 (95)viii入门指南安装电脑警告:安装电脑并将其连接到电源之前,请阅读《保修和支持指南》中的“安全信息”。
音频设备滤波器修改[发明专利]
专利名称:音频设备滤波器修改
专利类型:发明专利
发明人:A·莫吉米,W·贝拉迪,D·克里斯特申请号:CN201880008841.3
申请日:20180126
公开号:CN110268470A
公开日:
20190920
专利内容由知识产权出版社提供
摘要:一种具有若干个麦克风的音频设备,该若干个麦克风被配置成麦克风阵列。
与麦克风阵列通信的音频信号处理系统被配置为从多个麦克风导出多个音频信号,使用先前的音频数据来操作处理音频信号的滤波器拓扑结构以使得该阵列对期望声音比对不期望声音更敏感,将所接收的声音分类为期望声音或不期望声音之一,并且使用分类的所接收的声音和所接收的声音的类别来修改滤波器拓扑结构。
申请人:伯斯有限公司
地址:美国马萨诸塞州
国籍:US
代理机构:北京市金杜律师事务所
更多信息请下载全文后查看。
四面音频信号混合器(SISM)用户手册说明书
Shifting Inverting Signal Mingler (SISM)from 4ms CompanyEurorack Module User ManualThe Shifting Inverting Signal Mingler (SISM) is a 4-channel voltage manipulator that can scale, invert, attenuate, shift (DC offset), mix, split, and slice your CV and audio signals.Fourteen red and blue LEDs show exactly what each output jack is doing: red = negative voltage, blue = positive. Each of the four channels are independent and have an input and output jack, a knob for attenuating/inverting (Scale), a knob for applying positive or negative DC offset (Shift), and two LEDs.Four jacks on the bottom combine the four channels in different ways. Mix is a straight mix of all four shifted inputs, Switched Mix is a mix of all channels that have nothing plugged into the OUT jack, Positive Slice is the sum of all output voltages greater than zero volts, and Negative Slice is the sum of all negative voltages. The SLICE jacks can be used for sub-mixes, half-wave rectifying, “dirty panning”, and creating new CV signals from existing ones. Each channel has a jumper to make the Scale knob uni-polar (no inverting, attenuate only). This is useful for mixing audio instead of CV. Another jumper connection allows for a maximum of 2x gain. Headers allow for connections to a QCD, QPLFO, VCA Matrix and other manufacturer's modules.DOWNLOAD MOST RECENT MANUAL AT:/sism.phpBasic features:•Four independent voltage processor channels and four mix/mingle outputs•Each channel has:◦IN put jack◦SCALE knob (from inverted unity, to zero, to unity). Center-detented.◦SHIFT knob (adds/subtracts DC offset from -9.5V to +9.5V). Center-detented.◦OUT put jack◦Red and blue LEDs indicate positive and negative signal, respectively. Brighter means more voltage, dimmer means less.•MIX jack outputs sum of all channels•MIX SW itch jack outputs sum of all channels with nothing plugged into their output jack•+ SLICE jack outputs positive portion of mix (voltages greater than 0V)•- SLICE jack outputs negative portion of mix (voltages less than 0V)•Red and blue LEDs on all four mix/mingle jacks indicate positive and negative signal, respectively. Brighter means more voltage, dimmer means less.Advanced features:•Inputs for channels 1+2 and 3+4 are normalized together:•Plugging into Channel 1 (or 3) will split the signal to channels 1 and 2 (or to 3 and 4)• A jumper on the back enables/disables this normalization•Scale knob has three ranges that are selectable with jumpers:•Bi-polar: -1x to +1x (inverting unity to unity) — default•Uni-polar: 0 to +1x (level fader)•Boost: 0 to +2x (twice the voltage gain)•Input Header (2x4):•Connects the output of a QCD or QPLFO to the SISM's inputs•Plugging into the SISM input jacks overrides the signal on the header•Output Header (2x4)•Connects the SISM's channel outputs to the input of a VCA Matrix or other compatible module (Mixiplexer or Intermix)•PAD breakout header (2x4)•Can connect to an external DIY module that enables a pre-set pad (attenuation) on each channel. Usage:•Attenuator (quad)•Inverter (quad)•Level shifter/DC offset (quad)•Mixer (4 inputs)•Splitter (dual 1->2, or single 1->4 using a stackcable or passive mult)•Bipolar audio mixing: Useful for mixing various filter outputs with the original signal to create new filter curves•Voltage slicer: + and - portions are separated and outputted to the + and - SLICE jacks•"Dirty" panning: apply a large slow wave to one channel, and the other channels will "pan" between the SLICE jacks. The signals are half-wave rectified as they span both outputs•LFO mingler: connect QPLFO outputs to SISM inputs•Sequencer: connect QCD outputs to SISM inputs (or any clock divider/multiplier) and take the sequencer output from the MIX jack.•Add attenuation and inversion to any CV jack on other modules•Keep all your fine-tuning controls for several modules in one place•Manual voltage generator. With no input connected, each channel outputs a steady DC voltage controlled by the SHIFT knob.Specifications•12 HP Eurorack format module•0.86” (22 mm) maximum depth•10-pin to 16-pin power cable (included)•+12V rail: 51mA max•-12V rail: 51mA max•The 5V rail is not usedControlsThere are four identical channels on the SISM. Each channel has two controls: SCALE and SHIFT.SCALEControls gain, attenuation, inversion.The SCALE knob changes the gain of the input signal. If theknob is all the way up (clockwise, or 100%) then the signal willpass un-changed. This is called unity gain (1x).If the knob is turned down a little bit, then the signal will beattenuated. For example, if the input voltage was a wave thatranged from +1V to +5V, the output might range from +0.5V to+2.5V (50% attenuation). If you were controlling the pitch of aVCO, then instead of a 4-octave sequence, you would have a 2-octave sequence, your melody would be “squished”.As the knob approaches center, the signal will become smaller and smaller, until it's zero (silent).As the knob turns down past center, the signal will start to get louder, but will be inverted. For example, a +1V to+5V wave might become -0.25V to -1.25V (inverted and then attenuated to 25%).When the knob is fully down (counter-clockwise), the signal will be the same amplitude as the original, but inverted. Our +1V to +5V example wave would now be -1V to -5V. If this was a sequence being fed into a VCO, it would be same as the original, but it would be shifted down 6 octaves and the intervals would be inverted (meaning an upward step would turn into a downwards step, and vice-versa). Note: you could shift this back up 6 octaves using the SHIFT knob (see next section), and the result would be an inverted melody in the same octave range.The Scale knob has three modes (see “Jumpers” section for how to set the mode). By default the Scale knob ranges from negative unity to positive unity (-1x to +1x), as shown in the diagram above. This is called “Bi-polar” mode because the gain ranges between two “poles” or extremes: negative and positive.Another mode is called “uni-polar”, which means the gain ranges between 0 and one “pole” (positive). This means that the signal will be zero (silent) with the Scale knob turned all the way down, and will be unity (1x) when the knob is turned all the way up. This is often useful with mixing audio if you don't need to invert the signal and/or you want to quickly turn a channel down without finding the center point.The third mode is also uni-polar, but ranges from 0 to 2x. This means the signal is boosted by a factor of 2 with the knob all the way up. When the knob is centered, the gain is unity (1x). For example, a +1V to +5V wave would become a +2V to +10V wave (at maximum).SHIFTAdds or subtracts DC offset, shifts signal. Provides steady voltage.After the signal is scaled, it's shifted up or down along the voltageaxis. This is known as applying a “DC offset”. The shape andamplitude of the signal are not changed, it's merely shifted up ordown.With the knob at center, no voltage is added, and the signalpasses unchanged.Turning the knob up adds a positive voltage to the signal. If, for instance, the input signal ranged from 0V to +3V, applying a DC offset of 1.5V would make it range from 1.5V to4.5V. If you were using this to control the 1V/octave pitch of a VCO (for example), the VCO would play one-and-a-half octaves higher (+1.5V = +1.5octaves).Turning the knob down from center subtracts a steady DC voltage from the signal. Looking at this another way, a negative voltage is added to the signal. If you were to take a signal that ranges from 0V to +3V, subtracting 2V from it would result in a signal that ranges from -2V to +1V. If you were using this to control a VCO, the pitch would drop down 2 octaves.The SHIFT knob can add or subtract up to -9.5V to +9.5V. This is a large range and can be used to introduce clipping into the signal. The maximum output range of the SISM is -11.4V to +11.4V. Attempting to produce any voltage outside this range will clip the signal to about +11.4V or -11.4V.So if you had a signal that was -5V to +5V, and you applied +8V of SHIFT, the result should be +3V to +13V. However, 13V is out of range, so the SISM will clip the top of the wave, resulting in an actual waveform of +3V to +11.4V.Clipping can be useful. One trick is to use oneSISM channel to force clipping on a signal, andthen use a second channel to scale and shift theclipped signal into a useful range. This addsdistortion and can be used to create newwaveshapes. It also can be used to ensure awaveform stays within a certain range. Forexample, if you want to make sure your signalnever exceeds +5V, you could SHIFT it up to clipat +11.4V, then use a second channel to SHIFT itdown by -6.4V. The result will be a voltage that isclipped at +5V (11.4 - 6.4 = 5).Mix/Mingle OutputsBesides the output jacks for each channel, there are four output jacks on the bottom that are combinations of the four inputs.MIXThe MIX jack is a mix of the four channels. All aspects (DC component and AC waveshape) are combined. The MIX output is an arithmetic sum of the four channel outputs (post-SCALE and post-SHIFT).Keep in mind the maximum and minimum voltage of the SISM is +/-11.4V. It's easy to go outside this range when adding four voltages together, so if you have four loud/hot signals on the SISM and are not attenuating them, the MIX output will probably be clipped. If you wish to avoid clipping, simply adjust the SCALE knob on the channels (starting with the loudest) until clipping disappears. Note that the LEDs indicate precense of signal, not clipping. The only way to measure clipping would be to use an oscilloscope such as the Jones O'Tool.MIX (SW)This is identical to the MIX output, but it's the sum of the channels that have nothing plugged into the OUT jack. The slight difference between the MIX and MIX (SW) jack allows for some convenient features. See the Mixer with Master Channel, and the Sub-mixer patch ideas in the Patch Ideas section.+ SLICE and – SLICEThe + SLICE jack (positive SLICE) and the – SLICE jack (negative SLICE) output the positive and negative portions of the signal on the MIX jack, respectively. Whenever the voltage on the MIX jack is greater than 0V, the signal will output on the + SLICE jack. Whenever the voltage on the MIX jack is less than 0V, the signal will output on the – SLICE jack. If you were to combine the two SLICEs, the result would be the same as the MIX jack.Note: The SLICE jacks actually slice at approximately -250mV (+ SLICE) and +250mV (- SLICE). Thus, there is a small range of overlap where both jacks are outputting.“Dirty Panning”Dirty Panning: Triangle wave on Ch.1 is “panned” between + SLICE and -SLICE using the sawtooth on Ch.2Dirty Panning can be done using two channels and the SLICE jacks. One channel should contain the source signal to be panned (say, an audio source or an LFO). The other channel should contain the panning control signal. As the panning control signal goes from a positive voltage to a negative voltage, the source signal will shift from the + SLICE jack to the – SLICE jack. As it crosses the threshold between potitive and negative, different portions of the source wave will appear on the two jacks. The source wave will be half-wave rectified at a variable point. This is why it's called “dirty panning”, because musically it sounds distorted, yet the source signal will appear to “pan” from one output to the other.It's critical to adjust the SHIFT knob on the control channel so that the control signal ranges from an extreme positive voltage to an extreme negative voltage. Watch the LEDs on the SLICE jacks so that they light up alternately, in tempo with the control signal.Patch ideasCV processorMany modules do not have attenuators or inverters on their CV inputs.Patch a CV signal into one channel of the SISM, and patch the output to the CV input of your module. Adjust SCALE and SHIFT to precisely control the modulation. Bonus: after patching four channels, run the Mix and SLICE jacks to other modules.Mixer (4 channel)Run four signals into the SISM inputs. The MIX out will be a mix of the four signals. The level of each signal can be controlled by the SCALE knob. For audio signals, keep SHIFT at center (no DC offset).Patch ideas (continued)Mixer with Master ChannelRun up to three signals into channels 1-3. Patch the MIX (SW) jack into Channel 4 IN. Use Channel 4 OUT as your Master output. Channel 4 is now your master fader and level shifter. Since you have a cable patched into Channel 4 OUT, the MIX (SW) jack will be the sum of channels 1-3.Manual voltage generatorThis is useful for remotely controlling a VCA, testing a CV input on a module, or setting a sweet-spot. At any point, you can fade in a modulation signal instead of a manual voltage, simply by patching something into the IN jack. Plug nothing into the IN jack, and plug the OUT into something you want to control manually. Adjust the SHIFT knob to vary the voltage between -9.5V and +9.5V. For instance, you can simulate a trigger by turning the SHIFT knob up and then down. Or you can adjust a parameter of another module to find sweet-spots or troubleshoot. Note: If you are doing this patch on Channel 2, make sure nothing is plugged into Channel 1. If this isn't possible, then plug a dummy cable into the IN jack of Channel 2, or remove the LINK jumper on the back (see LINK Jumpers section). The same applies to Channels 4 and 3, respectively. The reason is that the LINK normalization copies Channel 1 input into Channel 2 and Channel 3 input into Channel 4, and this patch requires a null input.Precision manual voltage generator (fine tune pitch of a VCO)This allows you to create and control a very precise voltage. A common use is fine-tuning the pitch of a VCO.> Plug the OUT of Channel 1 into the IN of Channel 2. Run the OUT of Channel 2 into the module you want to control precisely (for example: the 1V/oct or FM input on a VCO).Make sure nothing is plugged into the IN jack of Channel 1 (see the Note on the previous patch idea above).Turn Channel 2's SCALE knob just a hair up past center so that it's attenuating heavily.Channel 2's SHIFT knob will be your coarse adjustment (sets the center of the range).Channel 1's SHIFT knob will be your fine-tuning adjustment (sets the precise voltage within the range).Channel 2's SCALE knob determines the level of precision of Channel 1's SHIFT. The more Channel 2 SCALE attenuates, the more precise Channel 1 SHIFT will be.Hint: You can increase precision by patching Ch2 OUT to Ch3 IN, and use Ch3 OUT as your final output.Bi-polar audio mixer (filters)By inverting and mixing various filtered versions of an audio signal, new roll-off curves can be obtained. For example, the Boogie Filter from Malekko has four different sloped curve outputs (-6dB to -24dB) which can be inverted and mixed together to create interesting filter effects. Another idea is to take a Band-Pass output from a filter such as the KOMA SVF-1, invert it with the SCALE knob, and mix it with the original unfiltered signal. This will create a Band-Reject filter.Sub-mixerSay you are using the top two SISM channels to scale/shift some modulation sources before running them into CV inputs of some other module. Say you also want to mix together two audio waves. You can use the two empty channels on the SISM by using the MIX (SW) jack. Just make sure Channels 1 and 2 OUTs have something plugged into them, and then run your two audio signals into Channels 3 and 4 INs. The MIX (SW) jack will be a mix of Channels 3 and 4 only.Modulation control centerBy using all four channels of the SISM as CV processors, you can control parameters of multiple modules from just one spot. This can help make a complex patch be easier to adjust and change, as well as making live performance more simple. The red and blue LEDs on the SISM allow you to see what's going on in your patch without having to scan multiple modules. For very large patches, consider several SISMs side-by-side.Dirty PannerSee “Dirty Panning” in the +/- SLICE sectionSequencerRun four clock signals into the four SISM inputs. Use an RCD, SCM, or use the headers to connect a QCD to the SISM inputs (see Connectivity section). Adjust the SCALE knobs of each channel to control how much thatclock/beat will change the sequence. Adjust SCALEModulation MinglerRun four LFOs or other modulation sources into the four SISM inputs. Plug at least one channel OUT jack into something. The MIX, MIX (SW), + SLICE, and – SLICE jacks will all output different signals.Half-wave rectifierPlug the signal you want to rectify into one channel. Adjust the SHIFT knob until you see both SLICE jacks lighting up (watch the red and blue LEDs). The positive half-wave rectified signal will appear on the + SLICE jack, and the negative will appear on the – SLICE jack.JumpersThe SISM has several advanced features accessed by jumpers on the rear of the module.LINK jumpersBy default, if nothing is plugged into Channel 2's input jack, this channel will copythe input of Channel 1. The same applies to Channels 3 and 4.A thin line on the faceplate between IN jacks tacitly indicates these connections.This feature is made possible by jumpers installed on the back of the PCB.To disable linking channels 1 and 2, remove the jumper labeled “LINK 1+2”.To disable linking channels 3 and 4, remove the jumper labeled “LINK 3+4”.If you are using the INS header to connect another module to the SISM inputs,it's best to remove the LINK jumpers, or else channels 1+2 and 3+4 will besummed at their inputs (in most cases, this is not desirable).Plugging into Channel 2's input jack will disable the LINK 1+2 feature, andlikewise, plugging into Channel 4's input jack will disable the LINK 3+4 feature.The feature is useful for splitting a signal: plug the signal into Channel 1 (or 3)and take the two outputs from Channel 1 and Channel 2 OUT. Each signal canbe SCALE'd and SHIFT'ed independently.SCALE MODESFour jumpers towards the top of the PCB are labelled “SCALEPOTS”. Below the header are the numbers “4 3 2 1” whichindicates which channel each jumper controls. See “SCALE”section above for more discussion.Jumper on: Default. The SCALE pot will range from negativeunity to unity (-1x to +1x). Center is zero.Jumper off: Uni-polar mode. The SCALE pot will range from zeroto unity (0 to 1x).Jumper wire from bottom pin to ground. Ground can be found onthe bottom row of pins on the INS, OUTS, and PADBO headers.Install a wire jumper from the corresponding pin on the bottomrow of the SCALE POTS header, to any unused ground pin onthe PCB (bottom row of any other header). The SCALE pot willrange from zero to double (0 to 2x).ConnectivityINSThe INS header is a 2x4 header for connecting the output ofcertain modules to the inputs of the SISM.This gives a behind-the-scenes connection between the SISMand another module such as the QPLFO or the QCD.To use, connect an 8-pin ribbon cable from the LFO-OUTSheader on the QPLFO or the CLOCK-OUTS header on theQCD, to the INS header on the SISM. The four outputs from thatmodule will now automatically be fed into the SISM. To overridethis connection, just plug something into the input jack on theSISM. This disables the connection for that channel only.For instance, if you find yourself often patching the four LFOoutputs of the QPLFO into the SISM (for shifting and scaling the LFOs before running them to other modules), connecting the two modules with a ribbon cable will connect all four channels without any patch cables. If you decide you need a SISM channel for something else, simply patch it in as normal—the behind-the-scenes connection will be disabled automatically.The top row of pins on this header attach directly to the switch tabs of the SISM input jacks. The bottom row of pins are all ground.OUTSThe OUTS header is a 2x4 header for connecting the channeloutputs of the SISM to the inputs of a compatible module.To use, connect an 8-pin ribbon cable from the OUTS header onthe SISM to the corresponding header on the destinationmodule. See the manual of that manufacturer's module fordetails.Compatible modules include the VCA Matrix (DAISYCHAININPUTS header), the Toppobrillo Mixiplexer, and the CircuitAbbey Intermix.The top row of pins on this header attach to the output of the final opamp on each channel. The bottom row of pins are all ground.DO NOT JUMPER THE OUTS HEADER!PAD B/OThe PAD B/O header can be used to connect to a breakoutmodule for providing a set amount of attenuation (pad) for eachchannel. The breakout module should apply a simple resistancebetween each pin on the top row and ground (the bottom row ofpins are all ground).A simple DIY module would have a switch for each channel thatconnected a resistor between the corresponding header pin andground. This would turn on/off a pre-set pad.100k → -3.5dB33k → -8dB10k → -15dB4k7 → -21dB。
判断音频中静音的代码(没测试)
判断⾳频中静⾳的代码(没测试)/** _______ _____ _____ _____* |__ __| | __ \ / ____| __ \* | | __ _ _ __ ___ ___ ___| | | | (___ | |__) |* | |/ _` | '__/ __|/ _ \/ __| | | |\___ \| ___/* | | (_| | | \__ \ (_) \__ \ |__| |____) | |* |_|\__,_|_| |___/\___/|___/_____/|_____/|_|** -------------------------------------------------------------** TarsosDSP is developed by Joren Six at IPEM, University Ghent** -------------------------------------------------------------** Info: http://0110.be/tag/TarsosDSP* Github: https:///JorenSix/TarsosDSP* Releases: http://0110.be/releases/TarsosDSP/** TarsosDSP includes modified source code by various authors,* for credits and info, see README.**/package be.tarsos.dsp;/*** The continuing silence detector does not break the audio processing pipeline when silence is detected.*/public class SilenceDetector implements AudioProcessor {public static final double DEFAULT_SILENCE_THRESHOLD = -70.0;//dbprivate final double threshold;//dbprivate final boolean breakProcessingQueueOnSilence;/*** Create a new silence detector with a default threshold.*/public SilenceDetector(){this(DEFAULT_SILENCE_THRESHOLD,false);}/*** Create a new silence detector with a defined threshold.** @param silenceThreshold* The threshold which defines when a buffer is silent (in dB).* Normal values are [-70.0,-30.0] dB SPL.* @param breakProcessingQueueOnSilence*/public SilenceDetector(final double silenceThreshold,boolean breakProcessingQueueOnSilence){this.threshold = silenceThreshold;this.breakProcessingQueueOnSilence = breakProcessingQueueOnSilence;}/*** Calculates the local (linear) energy of an audio buffer.** @param buffer* The audio buffer.* @return The local (linear) energy of an audio buffer.*/private double localEnergy(final float[] buffer) {double power = 0.0D;for (float element : buffer) {power += element * element;}return power;}/*** Returns the dBSPL for a buffer.** @param buffer* The buffer with audio information.* @return The dBSPL level for the buffer.*/private double soundPressureLevel(final float[] buffer) {double value = Math.pow(localEnergy(buffer), 0.5);value = value / buffer.length;return linearToDecibel(value);}/*** Converts a linear to a dB value.** @param value* The value to convert.* @return The converted value.*/private double linearToDecibel(final double value) {return 20.0 * Math.log10(value);}double currentSPL = 0;public double currentSPL(){return currentSPL;}/*** Checks if the dBSPL level in the buffer falls below a certain threshold. ** @param buffer* The buffer with audio information.* @param silenceThreshold* The threshold in dBSPL* @return True if the audio information in buffer corresponds with silence, * false otherwise.*/public boolean isSilence(final float[] buffer, final double silenceThreshold) { currentSPL = soundPressureLevel(buffer);return currentSPL < silenceThreshold;}public boolean isSilence(final float[] buffer) {return isSilence(buffer, threshold);}@Overridepublic boolean process(AudioEvent audioEvent) {boolean isSilence = isSilence(audioEvent.getFloatBuffer());//break processing chain on silence?if(breakProcessingQueueOnSilence){//break if silentreturn !isSilence;}else{//never break the chainreturn true;}}@Overridepublic void processingFinished() {}}。
双角色分离阵列麦克风
双角色分离阵列麦克风
品牌:浦喆
1、内置角色分离算法,可以把嫌疑人与办案人员声音独立隔离出2路独立的音频,分别提供给语音识别系统做语音识别。
2、发言方向指示灯:不同人员发言时,指示灯会变换不同的颜色,提示当前是嫌疑人或办案人员正在发言。
3、静音丢弃功能:软件可以把对应只有嫌疑人声音的音频单独录音,并在办案人员说话时的嫌疑人独立音频的静音部分丢弃,生成嫌疑人有效说话的(无办案人员声音无静音)未经压缩的原始音频,可以直接用于声纹识别、语音情绪分析。
4、内置扬声器:用于语音合成后向嫌疑播报法律法规。
声学参数:
阵列单元:两组麦克风阵列
采样率:16KHz
采样深度:16位
有效频响范围:100Hz-8KHz
输出接口:USB
输出声道:双声道
语音识别有限范围:3米内
灵敏度:≥50mV/Pa 本机噪声级:≤30dB 信噪比:≥50dB。
拾音器解决方案
拾音器解决方案一、引言拾音器是一种用于采集声音信号并将其转换为电信号的设备。
它被广泛应用于语音识别、语音合成、音频录制等领域。
本文将介绍一种拾音器解决方案,旨在提供高质量的声音采集和信号转换能力。
二、技术原理该拾音器解决方案基于MEMS(Micro-Electro-Mechanical Systems)技术,利用微机电系统的集成化特点,实现了高度集成化的声音采集和信号转换功能。
其主要技术原理如下:1. MEMS麦克风:采用MEMS麦克风作为声音采集的传感器,具有小型化、低功耗、高灵敏度等优点。
通过MEMS麦克风,可以将声音信号转换为电信号。
2. 音频放大器:采用高性能的音频放大器,可以放大MEMS麦克风输出的微弱电信号,提高信号的强度和稳定性。
3. 数字转换器:通过高精度的数字转换器,将摹拟电信号转换为数字信号,以便进一步处理和存储。
4. 信号处理芯片:采用专用的信号处理芯片,对数字信号进行降噪、滤波、增益控制等处理,提高声音采集的质量和稳定性。
三、产品特点该拾音器解决方案具有以下特点:1. 高质量的声音采集:采用高灵敏度的MEMS麦克风,能够捕捉细微的声音细节,保证声音采集的质量。
2. 低功耗设计:采用低功耗的MEMS麦克风和音频放大器,有效延长设备的使用时间。
3. 高度集成化:通过集成MEMS麦克风、音频放大器、数字转换器和信号处理芯片等功能模块,实现了设备的高度集成化,减小了设备体积。
4. 稳定可靠:采用高性能的音频放大器和信号处理芯片,能够提供稳定可靠的声音采集和信号转换能力。
5. 灵便可定制:该解决方案支持多种接口和通信协议,可以根据用户需求进行定制化设计。
四、应用领域该拾音器解决方案可广泛应用于以下领域:1. 语音识别:用于智能助理、语音控制、语音翻译等场景,提供准确可靠的声音输入。
2. 语音合成:用于智能音箱、语音导航等设备,提供高质量的声音输出。
3. 音频录制:用于录音设备、音频采集设备等,实现高保真的音频录制和采集。
拾音器解决方案
拾音器解决方案一、背景介绍拾音器是一种用于音频采集的设备,广泛应用于语音识别、语音合成、音频录制等领域。
为了满足不同应用场景和需求,设计一个高效、稳定、易用的拾音器解决方案至关重要。
二、解决方案概述本文将介绍一种拾音器解决方案,包括硬件和软件两个方面。
硬件方面,我们将设计一种高性能的拾音器电路,以确保音频采集的准确性和稳定性。
软件方面,我们将开辟一套完整的拾音器驱动程序和应用程序,以提供丰富的功能和良好的用户体验。
三、硬件设计1. 拾音器电路设计我们将采用高灵敏度的麦克风作为音频采集元件,并使用低噪声的摹拟前置放大器对音频信号进行放大。
为了提高抗干扰能力,我们将采用差分输入和抗干扰滤波器。
此外,我们还将加入电源滤波电路和防静电保护电路,以确保电路的稳定性和可靠性。
2. 电路布局和封装设计我们将采用四层板设计,合理布局电路元件,减少信号干扰。
同时,我们将使用高品质的封装材料,以提高电路的抗干扰能力和耐用性。
四、软件开辟1. 驱动程序设计我们将设计一套高效、稳定的拾音器驱动程序,以实现音频采集、数据处理和设备控制等功能。
驱动程序将支持多种操作系统和开辟平台,如Windows、Linux、Android等。
2. 应用程序开辟我们将开辟一款功能丰富、易于使用的拾音器应用程序。
用户可以通过应用程序进行音频录制、音频回放、音频剪辑等操作。
应用程序还将提供实时音频监测、音频格式转换等功能,以满足用户的不同需求。
五、性能测试与优化我们将对拾音器解决方案进行全面的性能测试和优化。
测试内容包括音频采集的准确性、音频信噪比、音频延迟等指标。
根据测试结果,我们将对硬件电路和软件程序进行优化,以提高整体性能和用户体验。
六、成果展示我们将制作一份详细的技术文档,包括硬件设计、软件设计、性能测试等内容。
同时,我们将制作一份演示视频,展示拾音器解决方案的功能和性能。
这些成果将有助于向客户和合作火伴展示我们的技术实力和解决方案的优势。
一种微型双麦克风语音增强算法
一种微型双麦克风语音增强算法
曾庆宁;王红丽;龙超
【期刊名称】《现代电子技术》
【年(卷),期】2022(45)6
【摘要】语音增强的目的是尽可能地从带噪语音中提取纯净的原始语音。
传统的单麦克风语音增强算法在非平稳噪声、低信噪比等复杂情况下性能显著降低;而双麦克风因其可以抑制方向性噪声,整体降噪性能更好而得到广泛应用。
针对微型双麦克风,文中提出一种有效的语音增强算法。
该算法首先通过对两通道含噪语音信号进行差分运算来抑制方向噪声,然后采用基于语音活动检测的改进自适应噪声抵消算法进一步消除剩余噪声,再对畸变的语音信号进行恢复运算,最后使用对数最小均方误差算法进一步消除残留噪声。
在对畸变的语音信号进行恢复运算的过程中,提出一种时域恢复算法,得到比已有频域恢复算法更小的运算量和时延。
实验结果表明,文中所提算法能有效地抑制噪声,改善语音质量,其降噪效果良好而且易于实时实现。
【总页数】7页(P58-64)
【作者】曾庆宁;王红丽;龙超
【作者单位】桂林电子科技大学信息与通信学院
【正文语种】中文
【中图分类】TN912.3-34;TP912.35
【相关文献】
1.子带MCRASC-MGSC微型麦克风阵语音增强算法
2.一种新的语音和噪声活动检测算法及其在手机双麦克风消噪系统中的应用
3.一种可跟踪移动声源方向的麦克风阵列语音增强算法
4.一种基于麦克风阵列的宽带语音增强算法研究
5.双麦克风语音增强算法研究与实现
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ABSTRACT
Discontinuous transmission based on speech/pause detection represents a valid solution to improve the spectral efficiency of new-generation wireless communication systems. In this context, robust Voice Activity Detection (VAD) algorithms are required, as traditional solutions present a high misclassification rate in the presence of the background noise typical of mobile environments. The Fuzzy Voice Activity Detector (FVAD) recently proposed in [1], shows that a valid alternative to deal with the problem of activity decision is to use methodologies like fuzzy logic. In this paper we propose a multichannel approach to activity detection using both fuzzy logic and time delay estimation. Objective and subjective tests confirm a significant improvement over traditional methods, above all in terms of a reduction in activity increase for non stationary noise.
2. THE FUZZY VAD (FVAD)
A Multi-Channel Speech/Silence Detector based on Time Delay Estimation and Fuzzy Classification
Francesco Beritelli, ቤተ መጻሕፍቲ ባይዱalvatore Casale, Alfredo Cavallaro Istituto di Informatica e Telecomunicazioni - University of Catania V.le A. Doria 6, 95125 Catania - Italy e-mail: {beritelli, casale, acavallaro}@iit.unict.it
1. INTRODUCTION
A Voice Activity Detector (VAD) aims to distinguish between speech and several types of acoustic background noise even with low signal-to-noise ratios (SNRs). Therefore, in a typical telephone conversation, a VAD, together with a comfort noise generator (CNG), achieves a silence compression. In the field of multimedia communications, silence compression allows the speech channel to be shared with other information, thus guaranteeing simultaneous voice and data applications. In a cellular radio system that uses the Discontinuous Transmission (DTX) mode, such as the Global System for Mobile communications (GSM), a VAD reduces co-channel interference (increasing the number of radio channels) and power consumption in portable equipment. Moreover, a VAD is vital to reduce the average bit rate in future generations of digital cellular networks, such as the Universal Mobile Telecommunication Systems (UMTS), which provide for variable bit-rate (VBR) speech coding. Most of the capacity gain is due to the distinction between speech activity and inactivity [2]. The performance of a speech coding approach based on phonetic classification, however, strongly depends on the classifier, which must be robust to every type of background noise [2]. As is well known, for example the performance of a VAD is critical for the overall speech quality, above all with low SNRs. When some of speech frames are detected as noise, intelligibility is seriously impaired due to speech clipping in the conversation. If, on the other hand, the percentage of noise detected as speech is high, the potential advantages of silence compression are not obtained. In the presence of background noise it may be difficult to distinguish between speech and silence, so for voice activity detection in wireless environments more efficient algorithms are needed [1][2][3]. Though the Fuzzy Voice Activity Detector (FVAD) recently proposed in [1] performs better than the solutions presented in literature [4][5], it exhibits an activity increase, above all in the presence of non-stationary noise. This paper therefore deals with the problem of speech activity detection in noisy environments, using parameters available only in a multichannel system, such as delay estimation and the difference in the power level between the channels. Referring to systems which use arrays of microphones for various purposes,
such as privileging a certain incoming signal direction or detecting and locating a signal source to be found in the reception area [6][7][8][9][10], we thought that a multi-channel system could be useful to obtain new information to be used in the detection of speech activity in noisy environments. This idea is also confirmed by recent studies on the central nervous system. It has been observed, in fact, that the medial nucleus of the superior olive, the structure whose neurons receive signals from both ears, is related with the location of sounds, which are said to be distinguished according to the differences in their arrival times. These time differences, in fact, are one of the elements used to locate sounds in space: a sound coming from a certain side reaches the ear on that side first and then, a few tens of milliseconds later, the other ear. Starting from this consideration, VADs have been developed which use a fuzzy system whose input is the Time Delay Estimation (TDE), the difference in the levels of the signals on the two microphones, and the continuous output of the FVAD, thus obtaining greater precision in detecting between activity and non-activity. Finally, objective and subjective tests demonstrate that the new approach, with a negligible increase in complexity, considerably increase the spectral resources, maintaining the same perceptive quality as the ITU-T G.729 VAD.