amidon toroid core q curve
Abstract: MC 4538 6132 RAM amidon 43 toroid core amidon toroid core balun ADC804 ADC604JH 4069 not CMOS amidon HP3325A
Contextual Info: AUDIO, COMMUNICATIONS, A/D CONVERTERS Or, Call Customer Service at 1-800-548-6132 USA Only 12-Bit 5MHz Sampling ANALOG-TO-DIGITAL CONVERTER FEATURES APPLICATIONS • • • • • • • DIGITAL RECEIVERS • RADAR SIGNAL ANALYSIS • FFT SPECTRUM ANALYSIS
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adc60412-bit-83dbc46-pin+120808hs6061-t6amidon toroid core q curvemc 45386132 ramamidon 43 toroid coreamidon toroid core balunadc804adc604jh4069 not cmosamidonhp3325a
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ic 7905
Abstract: bruel kjaer 1051 DATA SHEET OF IC 7905 HP3325A 6730S 74HC574 ADC701 ADC701JH ADC701KH SHC702
Contextual Info: ADC701 SHC702 16-Bit 512kHz SAMPLING A/D CONVERTER SYSTEM FEATURES APPLICATIONS ● CONVERSION RATE: to 512kHz Over Temp ● NO MISSING CODES AT 16 BITS ● MEDICAL IMAGING ● SONAR ● PROFESSIONAL AUDIO RECORDING ● AUTOMATIC TEST EQUIPMENT ● HIGH PERFORMANCE FFT SPECTRUM
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adc701shc70216-bit512khz+10107dbab-133ic 7905bruel kjaer 1051data sheet of ic 7905hp3325a6730s74hc574adc701jhadc701kh
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realtime PID controller
Abstract: assembly language programs for fft algorithm TMS320C30 10 band graphic equalizer Application of dsp in sonar Stereo Graphic Equalizer 340201B-01 Motorola 68020
Contextual Info: Application Note 023 Digital Signal Processing Fundamentals D. Koenig Introduction1 Recent advances in digital signal processing DSP technology make it easier for scientists to develop powerful personal computer-based data acquisition and analysis systems. This application
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tms320c30realtime pid controllerassembly language programs for fft algorithm10 band graphic equalizer+4application of dsp in sonarstereo graphic equalizer340201b-01motorola 68020
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91-tap
Abstract: ADSP-2100 ADSP-2101 AN-334 remez exchange algorithm bk 9435 linear convolution
Contextual Info: 1. _ 1 ANALOG ► DEVICES AN-334 APPLICATION NOTE ONE TECHNOLOGY WAY • P.O. BOX 9106 • NORWOOD, MASSACHUSETTS 02062-9106 • 617/329-4700 Digital Signal Processing Techniques D ig it a l F il t e r in g Real-time digital filtering is one of the most powerful tools of DSP. Apart from the
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an-334e1329-5-9/89adsp-210091-tap+4adsp-2101remez exchange algorithmbk 9435linear convolution
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MTL 728
Abstract: 204S ADSP-2100 ADSP-2101 AN-334 8 point fft audio spectrum analyzer fixed point IIR Filter 1941m
Contextual Info: P i ANALOG U DEVICES AN-334 APPLICATION NOTE ONE TECHNOLOGY WAY • P.O. BOX 9106 • NORWOOD, MASSACHUSETTS 02062-9106 • 617/3294700 Digital Signal Processing Techniques D ig ita l F ilt e r in g Real>time digital filtering is one of the most powerful tools of DSP. Apart from the
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an-334e1329-5-9/89adsp-2100mtl 728+5204sadsp-21018 point fft audio spectrum analyzerfixed point iir filter1941m
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spra884
Abstract: SPRU565 TMS320C64x dsp64xe TMS320C64X dsp dsp64x C64X
Contextual Info: Application Report SPRA925B − September 2003 Big-Endian Digital Signal Processing Library DSPLIB for TMS320C64x Chris Chung, Emil Lenchak Software Applications ABSTRACT The digital signal processing library (DSPLIB) is a set of C-callable, assembly-optimized
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spra925btms320c64xspra884spru565+4dsp64xetms320c64x dspdsp64xc64x
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assembly language programs for fft algorithm
Abstract: cc770 uPD77016 uPD77110 uPD77111 RX77016 Speech Recognition nec g.729 codec chip
Contextual Info: NEW PRODUCTS 2 COMPILER FOR µPD77016/µPD77111 FAMILY OF DSPs CC77016 Koji Doi Introduction DSPs* 1 were originally used as accelerator devices for realizing specific signal processing such as FFT*2 by software at high speed, but as the performance of processors increases, the applications that are
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pd77016/pd77111cc77016pd77xxx+13pd772xxpd771xxpd7705xpd77111/112pd77113/114assembly language programs for fft algorithmcc770upd77016upd77110upd77111rx77016speech recognition necg.729 codec chip
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implementation of data convolution algorithms
Abstract: digital filter calculus geology z transform DSP hearing aid image compression using neural networks linear convolution Civil Engineering data sheet design of Electrical Power Distribution transform display king
Contextual Info: The Scientist and Engineer's Guide to Digital Signal Processing Second Edition Be sure to visit the book’s website at: www.DSPguide.com The Scientist and Engineer's Guide to Digital Signal Processing Second Edition by Steven W. Smith California Technical Publishing
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implementation of data convolution algorithmsdigital filter calculusgeologyz transform+6dsp hearing aidimage compression using neural networkslinear convolutioncivil engineering data sheetdesign of electrical power distribution transformdisplay king
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74HC14 oscillator application note
Abstract: 74HC14 oscillator application diagram 74HC14 HIF3G-50P-2.54DSA 64-128fs HIF3G JP13-2 74LS07 pin details for 74HC14 AK4114
Contextual Info: [AKD4141-A] AKD4141-A Evaluation board Rev.2 for AK4141 GENERAL DESCRIPTION The AKD4141 is an evaluation board for the AK4141, digital stereo decoder with stereo sample rate converter, digital switches and sound processing functions. The AKD4141 has the analog/digital audio interface and can achieve the interface with analog/digital
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akd4141-aak4141akd414110pin+11hif3g-50p-274hc14 oscillator application note74hc14 oscillator application diagram74hc14hif3g-50p-2.54dsa64-128fshif3gjp13-274ls07pin details for 74hc14ak4114
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SLAA114
Abstract: digital audio pwm modulation demodulation audio analyzer pulse amplifier measurement AES17 tunable notch filter Low-pass Passive Filter Design Techniques Passive Low-pass Filter Passive Low-pass Filter Design Method
Contextual Info: Application Report SLAA114 – January 2001 Digital Audio Measurements Claus Neesgaard Digital Audio, TI Copenhagen ABSTRACT To perform correct measurements on a true digital audio amplifier TDAA , it is important to understand the characteristics of the device, especially in comparison with conventional
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slaa114digital audio pwm modulation demodulationaudio analyzerpulse amplifier measurement+5aes17tunable notch filterlow-pass passive filter design techniquespassive low-pass filterpassive low-pass filter design method
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hearing chip
Abstract: GA3220 GC5055 GC505 process Instrument Sound Design hearing transducers for engines GENNUM DSP
Contextual Info: GA3220 Programmable Signal Processing Platform for Hearing Instruments OVERVIEW GA3220 is a system-in-a-package SiP signal processing platform specifically designed for hearing instruments which require high precision processing and ultra low power consumption. The platform
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ga3220gc5002provi00204mw/mmac+842mipshearing chipgc5055gc505process instrumentsound design hearingtransducers for enginesgennum dsp
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sharc iir filter
Abstract: 7.1 theater system sharc accelerator IIR pdf str 5643 5.1 home theatre circuit datasheet circuit for 7.1 home theatre system IIR SIMD sharc iir filter IIR Accelerator
Contextual Info: Hardware Accelerators Boost the Performance of Next-Generation SHARC Processors By Paul Beckmann, DSP Concepts, LLC Summary The recently announced Analog Devices SHARC ADSP-2146x processor incorporates hardware accelerators for implementing three widely used signal processing operations: FIR finite impulse response ,
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adsp-2146xsharc iir filter7.1 theater systemsharc accelerator iir+5pdf str 56435.1 home theatre circuit datasheetcircuit for 7.1 home theatre systemiir simdsharc iir filter iir accelerator
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LPC1300
Abstract: Decoding DTMF tones using M3 DSP library FFT function RDB1768 LPC1768 AN10943 NXP LPC1768 multi tone buzzers VFD-S LPC1700 cortex m3 256-Point
Contextual Info: AN10943 Decoding DTMF tones using M3 DSP library FFT function Rev. 1 — 17 June 2010 Application note Document information Info Content Keywords M3, LPC1300, LPC1700, DSP, DFT, FFT, DTMF Abstract This application note and associated source code example demonstrates
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an10943lpc1300lpc1700decoding dtmf tones using m3 dsp library fft function+7rdb1768lpc1768nxp lpc1768multi tone buzzersvfd-slpc1700 cortex m3256-point
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verilog code for twiddle factor ROM
Abstract: vhdl code for speech recognition VHDL audio codec ON DE2 verilog code for speech recognition lms algorithm using verilog code lms algorithm using vhdl code VHDL FOR FFT TO SPEECH RECOGNITION ON DE2 block diagram of speech recognition using matlab circuit diagram of speech recognition Speech Recognition filter noise matlab
Contextual Info: Nios II-Based Audio-Controlled Digital Oscillograph Third Prize Nios II-Based Audio-Controlled Digital Oscillograph Institution: Xian Jiao Tong University Participants: Wan Liang, Zhang Weile, and Wang Wei Instructor: Penghui Zhang Design Introduction The oscillograph is a common instrument that plays a key role in many experiments. Because of its
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x1/10ep2c35f672c6verilog code for twiddle factor romvhdl code for speech recognition+8vhdl audio codec on de2verilog code for speech recognitionlms algorithm using verilog codelms algorithm using vhdl codevhdl for fft to speech recognition on de2block diagram of speech recognition using matlabcircuit diagram of speech recognitionspeech recognition filter noise matlab
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AD1895
Abstract: AD1896 SRC4190 SRC4190IDB SRC4190IDBR SRC4192 SSOP-28 CP-1201 SRC419
Contextual Info: SRC4190 1 SRC 419 SBFS023A – JUNE – REVISED JULY 2003 192kHz Stereo Asynchronous Sample Rate Converters FEATURES APPLICATIONS ● AUTOMATIC SENSING OF THE INPUT-TOOUTPUT SAMPLING RATIO ● WIDE INPUT-TO-OUTPUT SAMPLING RANGE: 16:1 to 1:16 ● SUPPORTS INPUT AND OUTPUT SAMPLING
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30MHZ
Abstract: PGA100 STV0300 STV0300S 64 point FFT radix-4
Contextual Info: STV0300 FAST SINGLE CHIP OF 8192 COMPLEX POINT FAST FOURIER TRANSFORM . . . . . . . . . . . . . PROGRAMMABLE FFT SIZE FROM 2 TO 8192 COMPLEX WORDS 16K WORDS OF ON-CHIP MEMORY NO EXTERNAL MEMORY NEEDED . HALF OF THE MEMORY ACTS AS AN INPUT BUFFER TO ALLOW SERIAL PROCESSING OF ADJACENT DATA BLOCKS.
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stv030020mhz8192-points4092-points+430mhzpga100stv0300s64 point fft radix-4
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Contextual Info: SRC4190 1 SRC 419 SBFS023A – JUNE – REVISED JULY 2003 192kHz Stereo Asynchronous Sample Rate Converters FEATURES APPLICATIONS ● AUTOMATIC SENSING OF THE INPUT-TOOUTPUT SAMPLING RATIO ● WIDE INPUT-TO-OUTPUT SAMPLING RANGE: 16:1 to 1:16 ● SUPPORTS INPUT AND OUTPUT SAMPLING
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Contextual Info: SRC4190 1 SRC 419 SBFS023B – JUNE 2003 – REVISED SEPTEMBER 2007 192kHz Stereo Asynchronous Sample Rate Converters FEATURES APPLICATIONS ● AUTOMATIC SENSING OF THE INPUT-TOOUTPUT SAMPLING RATIO ● WIDE INPUT-TO-OUTPUT SAMPLING RANGE: 16:1 to 1:16 ● SUPPORTS INPUT AND OUTPUT SAMPLING
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AD1895
Abstract: AD1896 SRC4190 SRC4190IDB SRC4190IDBR SRC4192 SSOP-28
Contextual Info: SRC4190 1 SRC 419 SBFS023A – JUNE – REVISED JULY 2003 192kHz Stereo Asynchronous Sample Rate Converters FEATURES APPLICATIONS ● AUTOMATIC SENSING OF THE INPUT-TOOUTPUT SAMPLING RATIO ● WIDE INPUT-TO-OUTPUT SAMPLING RANGE: 16:1 to 1:16 ● SUPPORTS INPUT AND OUTPUT SAMPLING
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AD1895
Abstract: AD1896 SRC4190 SRC4190IDB SRC4190IDBR SRC4192 SSOP-28
Contextual Info: SRC4190 1 SRC 419 SBFS023A – JUNE – REVISED JULY 2003 192kHz Stereo Asynchronous Sample Rate Converters FEATURES APPLICATIONS ● AUTOMATIC SENSING OF THE INPUT-TOOUTPUT SAMPLING RATIO ● WIDE INPUT-TO-OUTPUT SAMPLING RANGE: 16:1 to 1:16 ● SUPPORTS INPUT AND OUTPUT SAMPLING
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varispeed -616PC5/616P5
Abstract: CS8421 varispeed CDB8421 CS8421-CNZ CS8421-CZZ CS8421-DZZ
Contextual Info: CS8421 32-bit, 192 kHz Asynchronous Sample Rate Converter Features General Description No programming required 175 dB Dynamic Range –140 dB THD+N Flexible 3-wire serial digital audio input port 16, 20, 24, or 32-bit data I/O Supports 128*Fs Master Mode only ,
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Digital Audio Receivers
Contextual Info: SRC4190-Q1 www.ti.com . SBFS034 – SEPTEMBER 2008 192-kHz STEREO ASYNCHRONOUS SAMPLE-RATE CONVERTER
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Contextual Info: SRC4190-Q1 www.ti.com SBFS034A – SEPTEMBER 2008 – REVISED OCTOBER 2009 192-kHz STEREO ASYNCHRONOUS SAMPLE-RATE CONVERTER Check for Samples: SRC4190-Q1 FEATURES 1 • • • • • • • • • • • • • Qualified for Automotive Applications
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CS8421
Abstract: varispeed -616PC5/616P5 CS8421-DZZR AN270 AN282 CDB8421 varispeed
Contextual Info: CS8421 32-bit, 192 kHz Asynchronous Sample Rate Converter Features 175 dB Dynamic Range Bypass Mode –140 dB THD+N Time Division Multiplexing TDM Mode No Programming Required Attenuates Clock Jitter No External Master Clock Required
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