videophone circuit diagram
Abstract: h221 8X8 INC 7230 acelp Maximum Likelihood Quantization C161 C163 PSB7230 Q67101-H6864
Text: Design-In Guide for JADE AN 08/97 Audio Ringing Codec Filter Basic Function PSB 7230 The PSB 7230 Joint Audio Decoder Encoder for Analog Videophones JADE AN is a device wich implements voice compression algorithmus using the Algebraic Code Excited Linear Prediction (ACELP) and the Multi-Pulse Maximum Likelihood
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Q67220-H1003
PSB4595/96
C161/
RS232
videophone circuit diagram
h221
8X8 INC
7230
acelp
Maximum Likelihood Quantization
C161
C163
PSB7230
Q67101-H6864
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TC88401F
Abstract: tc511000
Text: I l TC88401F-1 B C1D'Î72MC| 0 Q 5 S 3 Ô 7 Ifib m j O S 3 TOS H IB A 1. UC/UP b4E » GENERAL TC88401F is a voice recording C M O S L S I using the APC-M LQ (Adaptive Predictive Coding w ith Maximum Likelihood Quantization) Method. The voice recording / playback system is configured
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TC88401F-1
TC88401F
TC88401F.
TC3531fi*
82kil
10kil
TC514100ZL
tc511000
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h221
Abstract: TA 7230 P 8X8 INC SZB6120 iit inc acelp
Text: Design-In Guide for JADE AN 08/97 Audio Ringing Codec Filter Basic Function PSB 7230 The PSB 7230 Joint Audio Decoder Encoder for Analog Videophones JADE AN is a device wich implements voice compression algorithmus using the Algebraic Code Excited Linear Prediction (ACELP) and the Multi-Pulse Maximum Likelihood
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SIPB7282
324/H
Q67220-H1003
h221
TA 7230 P
8X8 INC
SZB6120
iit inc
acelp
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verilog code for speech recognition
Abstract: block diagram of speech recognition using matlab circuit diagram of speech recognition block diagram of speech recognition vhdl code for speech recognition VHDL audio codec ON DE2 simple vhdl de2 audio codec interface VHDL audio processing codec DE2 Speech Signal Processing matlab noise vhdl code for voice recognition
Text: SOPC-Based Speech-to-Text Conversion Second Prize SOPC-Based Speech-to-Text Conversion Institution: National Institute of Technology, Trichy Participants: M.T. Bala Murugan and M. Balaji Instructor: Dr. B. Venkataramani Design Introduction For the past several decades, designers have processed speech for a wide variety of applications ranging
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acelp
Abstract: MP-MLQ g723 Maximum Likelihood Quantization code excited linear predictive g.723 speech ICs
Text: a G.723.1 DSP-BASED SPEECH COMPRESSION CODER A nalog Devices’ G.723.1 speech compression algorithm, running on a highperformance 33 MIPS Digital Signal Processor DSP compresses the speech or any other audio signal component of multimedia software or other audio. This speech coder chipset features
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acelp
Abstract: Maximum Likelihood Quantization g723 linear predictive coding SPEECH SYNTHESIS H02161 MP-MLQ
Text: G.723.1 DSP-BASED SPEECH COMPRESSION CODER OVERVIEW Analog Devices’ G.723.1 speech compression algorithm, running on a high-performance 33 MIPS Digital Signal Processor DSP , compresses speech, any other audio signal component of multimedia software or other
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H02161
D-81373
acelp
Maximum Likelihood Quantization
g723
linear predictive coding
SPEECH SYNTHESIS
H02161
MP-MLQ
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h221
Abstract: 16kHz-8kHz acelp videophone circuit diagram
Text: Product Brochure for JADE MM 08/97 Joint Audio Decoder-Encoder PSB 7238, JADE MM The PSB 7238 Joint Audio Decoder Encoder Multimode JADE MM is a device wich implements voice compression algorithms using the Low-Delay Code Excited Linear Prediction (LD-CELP) standard as defined in the ITU-T
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Q6722Q-H1QQ3
h221
16kHz-8kHz
acelp
videophone circuit diagram
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videophone circuit diagram
Abstract: Maximum Likelihood Quantization G.723 codec 7238 h221 mp V1.2 C161 C163 PSB2186 Q67101-H6773
Text: Product Brochure for JADE MM 08/97 Joint Audio Decoder-Encoder PSB 7238, JADE MM The PSB 7238 Joint Audio Decoder Encoder Multimode JADE MM is a device wich implements voice compression algorithms using the Low-Delay Code Excited Linear Prediction (LD-CELP) standard as defined in the ITU-T
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Q67220-H1003
PSB2186
C161/
RS232
videophone circuit diagram
Maximum Likelihood Quantization
G.723 codec
7238
h221
mp V1.2
C161
C163
PSB2186
Q67101-H6773
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rAised cosine
Abstract: rAised cosine FILTER nyquist plot AN-922 pulse interval encoding AN922 fixed point IIR Filter
Text: AN-922 APPLICATION NOTE One Technology Way • P.O. Box 9106 • Norwood, MA 02062-9106, U.S.A. • Tel: 781.329.4700 • Fax: 781.461.3113 • www.analog.com Digital Pulse-Shaping Filter Basics by Ken Gentile INTRODUCTION RECTANGULAR PULSE TIME DOMAIN Data transmission systems that must operate in a bandwidthlimited environment must contend with the fact that constraining
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AN-922
AN06897-0-9/07
rAised cosine
rAised cosine FILTER
nyquist plot
AN-922
pulse interval encoding
AN922
fixed point IIR Filter
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Viterbi Decoder
Abstract: TMS320C25 XDS510 linear predictive coding convolutional mps40 Videophone
Text: Transmission of Still and Moving Images Over Narrowband Channels Application Report Stefan Goss, Wilhelm Vogt, Rodolfo Mann Pelz, Dirk Lappe Communication Research Institute SPRA110 February 1994 Printed on Recycled Paper IMPORTANT NOTICE Texas Instruments TI reserves the right to make changes to its products or to discontinue any
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MPS40
TMS320C4x
Viterbi Decoder
TMS320C25
XDS510
linear predictive coding
convolutional
Videophone
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Adaptive Differential Pulse Code Modulation Decoder
Abstract: MPS40 TMS320C25 XDS510 convolutional ti 261 326XFAST vogt elektronik
Text: Transmission of Still and Moving Images Over Narrowband Channels Application Report Stefan Goss, Wilhelm Vogt, Rodolfo Mann Pelz, Dirk Lappe Communication Research Institute SPRA110 February 1994 Printed on Recycled Paper IMPORTANT NOTICE Texas Instruments TI reserves the right to make changes to its products or to discontinue any
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SPRA110
Adaptive Differential Pulse Code Modulation Decoder
MPS40
TMS320C25
XDS510
convolutional
ti 261
326XFAST
vogt elektronik
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verilog code for speech recognition
Abstract: vhdl code for speech recognition circuit diagram of speech recognition block diagram of speech recognition vhdl code for voice recognition speech to text recognition vhdl vhdl code hamming block diagram of speech recognition using matlab SPEECH RECOGNITION by matlab verilog code hamming
Text: Nios II Embedded Processor Design Contest—Outstanding Designs 2005 Second Prize SOPC-Based Word Recognition System Institution: National Institute Of Technology, Trichy Participants: S. Venugopal, B. Murugan, S.V. Mohanasundaram Instructor: Dr. B. Venkataramani
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MP-MLQ
Abstract: G.723. c code TMS320C62x TMS320C6000 codec723 TMS320C6000 intrinsics list C6000 G723 simulator encoder decoder cd 7231
Text: Application Report SPRA552A G.723.1 Dual Rate Speech Coder: Multichannel TMS320C6000 Implementation Thomas J. Dillon, Jr. C6000 Applications Abstract This application report describes how the G.723.1 Dual Rate Speech Coder has been implemented on the Texas Instruments TI TMS320C6000 digital signal processor (DSP).
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TMS320C6000
C6000
MP-MLQ
G.723. c code
TMS320C62x
codec723
TMS320C6000 intrinsics list
G723
simulator encoder decoder
cd 7231
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acelp
Abstract: Maximum Likelihood Quantization G.723 C6000 G723 TMS320C6000
Text: Application Report SPRA552B- February 2000 G.723.1 Dual-Rate Speech Coder: Multichannel TMS320C62x Implementation Thomas J. Dillon, Jr. C6000 Applications ABSTRACT This application report describes how the G.723.1 Dual-Rate Speech Coder has been implemented on the Texas Instruments TI TMS320C62x digital signal processor (DSP).
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TMS320C62x
C6000
acelp
Maximum Likelihood Quantization
G.723
G723
TMS320C6000
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Maxlinear
Abstract: intel286 vector quantization hmm mfcc coefficient quantization
Text: Intel Recognition Primitives Library Reference Manual Copyright 1995-1997, Intel Corporation All Rights Reserved Issued in U.S.A. Order Number 637785-005 Intel Recognition Primitives Library Reference Manual Order Number: 637785-005 Revision Revision History
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baseband QPSK matlab code
Abstract: qpsk demapper VHDL CODE Wimax in matlab simulink 16qam demapper VHDL CODE simulink 16QAM gsm simulink wimax matlab qpsk modulation VHDL CODE qpsk simulink matlab wimax CHANNEL CODING matlab
Text: Constellation Mapper and Demapper for WiMAX Application Note 439 May 2007, version 1.1 Introduction Altera provides building blocks that can be used to accelerate the development of an IEEE 802.16e-2005 WiMAX compliant basestation. This application note describes a reference design that demonstrates the
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16e-2005
baseband QPSK matlab code
qpsk demapper VHDL CODE
Wimax in matlab simulink
16qam demapper VHDL CODE
simulink 16QAM
gsm simulink
wimax matlab
qpsk modulation VHDL CODE
qpsk simulink matlab
wimax CHANNEL CODING matlab
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lm3824
Abstract: No abstract text available
Text: OBSOLETE LM3822 www.ti.com SNVS064C – JULY 2000 – REVISED APRIL 2013 LM3822 Precision Current Gauge IC with Internal Zero Ohm Sense Element and PWM Output Check for Samples: LM3822 FEATURES DESCRIPTION • • The LM3822 Current Gauge provides easy to use
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SNVS064C
LM3822
lm3824
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SSOP16
Abstract: SSOP20 UDA1324TS
Text: Philips Semiconductors Preliminary specification Ultra low-voltage stereo filter DAC UDA1324TS FEATURES General • Low power consumption • Ultra low power supply voltage 1.9 to 2.4 V • Selectable control via L3 microcontroller interface or via static pin control.
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512fs
384fs
SSOP16)
SSOP16
SSOP20
UDA1324TS
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lm3812
Abstract: No abstract text available
Text: LM3812, LM3813 www.ti.com SNOS028D – JUNE 1999 – REVISED APRIL 2013 LM3812/LM3813 Precision Current Gauge IC with Ultra Low Loss Sense Element and PWM Output Check for Samples: LM3812, LM3813 FEATURES DESCRIPTION • • The LM3812/LM3813 Current Gauges provide easy
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LM3813
SNOS028D
LM3812/LM3813
LM3812
LM3813
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lm3814
Abstract: No abstract text available
Text: OBSOLETE LM3814, LM3815 www.ti.com SNVS045D – JANUARY 1999 – REVISED APRIL 2013 LM3814/LM3815 Fast Current Gauge IC with Ultra Low Loss Sense Element and PWM Output Check for Samples: LM3814, LM3815 FEATURES DESCRIPTION • • The LM3814/LM3815 Current Gauges provide easy
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SNVS045D
LM3814/LM3815
LM3814
LM3815
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Untitled
Abstract: No abstract text available
Text: PCM1733 PCM 173 3 Stereo Audio DIGITAL-TO-ANALOG CONVERTER 18 Bits, 96kHz Sampling TM FEATURES DESCRIPTION ● COMPLETE STEREO DAC: Includes Digital Filter and Output Amp The PCM1733 is a complete low cost stereo audio digital-to-analog converter DAC , operating off of a
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PCM1733
96kHz
PCM1733
256fS
384fS
18-bit
16kHz
96kHz
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Untitled
Abstract: No abstract text available
Text: PCM1733 PCM 173 3 Stereo Audio DIGITAL-TO-ANALOG CONVERTER 18 Bits, 96kHz Sampling TM FEATURES DESCRIPTION ● COMPLETE STEREO DAC: Includes Digital Filter and Output Amp The PCM1733 is a complete low cost stereo audio digital-to-analog converter DAC , operating off of a
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PCM1733
96kHz
PCM1733
256fS
384fS
18-bit
16kHz
96kHz
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TDA1548T
Abstract: bssq S028 SSOP28 TDA1548TZ hi bass boost ic
Text: Philips Semiconductors Product specification Bitstream continuous calibration filter-DAC with headphone driver and DSP TDA1548T FEATURESPRODUCT SPECIFICATION Easy application • Only first-order analog post-filtering required • Headphone amplifiers and digital filter integrated
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TDA1548T
256fs
384fs
SSOP28)
711002b
TDA1548T
bssq
S028
SSOP28
TDA1548TZ
hi bass boost ic
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LQFP48
Abstract: TDA8787 TDA8787HL
Text: INTEGRATED CIRCUITS DATA SHEET TDA8787 10-bit, 3.0 V analog-to-digital interface for CCD cameras Preliminary specification Supersedes data of 1998 Mar 27 File under Integrated Circuits, IC02 1998 Oct 15 Philips Semiconductors Preliminary specification 10-bit, 3.0 V analog-to-digital interface for
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TDA8787
10-bit,
10-bit
TDA8787
SCA60
545104/750/02/pp20
LQFP48
TDA8787HL
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