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Signal and system with matlab computing and simulink modeling is a good book for communication student and engineer due to the methods and tools mentioned in the book
標簽：
communication
and
computing
simulink
上傳時間：
2013-12-25
上傳用戶：liuchee
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The TAS3204 is a highly-integrated audio system-on-chip (SOC) consisting of a fully-programmable, 48-bit digital audio processor, a 3:1 stereo analog input MUX, four ADCs, four DACs, and other analog functionality. The TAS3204 is programmable with the graphical PurePath Studio? suite of DSP code development software. PurePath Studio is a highly intuitive, drag-and-drop environment that minimizes software development effort while allowing the end user to utilize the power and flexibility of the TAS3204’s digital audio processing core.
TAS3204 processing capability includes speaker equalization and crossover, volume/bass/treble control, signal mixing/MUXing/splitting, delay compensation, dynamic range compression, and many other basic audio functions. Audio functions such as matrix decoding, stereo widening, surround sound virtualization and psychoacoustic bass boost are also available with either third-party or TI royalty-free algorithms.
The TAS3204 contains a custom-designed, fully-programmable 135-MHz, 48-bit digital audio processor. A 76-bit accumulator ensures that the high precision necessary for quality digital audio is maintained during arithmetic operations.
Four differential 102 dB DNR ADCs and four differential 105 dB DNR DACs ensure that high quality audio is maintained through the whole signal chain as well as increasing robustness against noise sources such as TDMA interference.
The TAS3204 is composed of eight functional blocks:
Clocking System
Digital Audio Interface
Analog Audio Interface
Power supply
Clocks, digital PLL
I2C control interface
8051 MCUcontroller
Audio DSP – digital audio processing
特性
Digital Audio Processor
Fully Programmable With the Graphical, Drag-and-Drop PurePath Studio? Software Development Environment
135-MHz Operation
48-Bit Data Path With 76-Bit Accumulator
Hardware Single-Cycle Multiplier (28 × 48)
標簽：
3204
tas
上傳時間：
2016-05-06
上傳用戶：fagong
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This report presents a tutorial of fundamental array processing and beamforming theory relevant to microphone array speech processing. A microphone array consists of multiple microphones placed at different spatial locations. Built upon a knowledge of sound propagation principles, the multiple inputs can be manipulated to enhance or attenuate signals emanating from particular directions. In this way, microphone arrays provide a means of enhancing a desired signal in the presence of corrupting noise sources. Moreover, this enhancement is based purely on knowledge of the source location, and so microphone array techniques are applicable to a wide variety of noise types. Microphone arrays have great potential in practical applications of speech processing, due to their ability to provide both noise robustness and hands-free signal acquisition.
標簽：
Microphone array
Tutorial
Array Signal Processing
上傳時間：
2016-06-12
上傳用戶：halias
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The 4.0 kbit/s speech codec described in this paper is based on a
Frequency Domain Interpolative (FDI) coding technique, which
belongs to the class of prototype waveform Interpolation (PWI)
coding techniques. The codec also has an integrated voice
activity detector (VAD) and a noise reduction capability. The
input signal is subjected to LPC analysis and the prediction
residual is separated into a slowly evolving waveform (SEW) and
a rapidly evolving waveform (REW) components. The SEW
magnitude component is quantized using a hierarchical
predictive vector quantization approach. The REW magnitude is
quantized using a gain and a sub-band based shape. SEW and
REW phases are derived at the decoder using a phase model,
based on a transmitted measure of voice periodicity. The spectral
(LSP) parameters are quantized using a combination of scalar
and vector quantizers. The 4.0 kbits/s coder has an algorithmic
delay of 60 ms and an estimated floating point complexity of
21.5 MIPS. The performance of this coder has been evaluated
using in-house MOS tests under various conditions such as
background noise. channel errors, self-tandem. and DTX mode
of operation, and has been shown to be statistically equivalent to
ITU-T (3.729 8 kbps codec across all conditions tested.
標簽：
frequency-domain
interpolation
performance
Design
kbit_s
speech
coder
based
and
of
上傳時間：
2018-04-08
上傳用戶：kilohorse
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This paper presents a Hidden Markov Model (HMM)-based speech
enhancement method, aiming at reducing non-stationary noise from speech
signals. The system is based on the assumption that the speech and the noise
are additive and uncorrelated. Cepstral features are used to extract statistical
information from both the speech and the noise. A-priori statistical
information is collected from long training sequences into ergodic hidden
Markov models. Given the ergodic models for the speech and the noise, a
compensated speech-noise model is created by means of parallel model
combination, using a log-normal approximation. During the compensation, the
mean of every mixture in the speech and noise model is stored. The stored
means are then used in the enhancement process to create the most likely
speech and noise power spectral distributions using the forward algorithm
combined with mixture probability. The distributions are used to generate a
Wiener filter for every observation. The paper includes a performance
evaluation of the speech enhancer for stationary as well as non-stationary
noise environment.
標簽：
Telecommunications
Processing
Signal
for
上傳時間：
2020-06-01
上傳用戶：shancjb
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In this thesis several asp ects of space-time pro cessing and equalization for wire-
less communications are treated. We discuss several di?erent metho ds of improv-
ing estimates of space-time channels, such as temp oral parametrization, spatial
parametrization, reduced rank channel estimation, b o otstrap channel estimation,
and joint estimation of an FIR channel and an AR noise mo del. In wireless commu-
nication the signal is often sub ject to intersymb ol interference as well as interfer-
ence from other users.
標簽：
Communications
Space-Time
Processing
Wireless
for
上傳時間：
2020-06-01
上傳用戶：shancjb
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An acronym for Multiple-In, Multiple-Out, MIMO communication sends the same data as several signals
simultaneously through multiple antennas, while still utilizing a single radio channel. This is a form of
antenna diversity, which uses multiple antennas to improve signal quality and strength of an RF link. The
data is split into multiple data streams at the transmission point and recombined on the receive side by
another MIMO radio configured with the same number of antennas. The receiver is designed to take
into account the slight time difference between receptions of each signal, any additional noise or
interference, and even lost signals.
標簽：
Understanding_the_Basics_of_MIMO
上傳時間：
2020-06-01
上傳用戶：shancjb
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The goal of this textbook is to support the teaching of digital and statistical signal
processing in higher education. Particular attention is paid to the presentation of the fun-
damental theory; key topics are outlined in a comprehensible way, and all areas of the
subject are discussed in a fashion that aims at simplification without sacrificing accuracy.
標簽：
Statistical
Processing
Digital
Signal
and
上傳時間：
2020-06-10
上傳用戶：shancjb
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Precision, Low Noise, CMOS, Rail-to-Rail,
Input/Output Operational Amplifiers
Data Sheet AD8605/AD8606/AD8608The AD8605, AD8606, and AD86081 are single, dual, and quad rail-to-rail input and output, single-supply amplifiers. They feature very low offset voltage, low input voltage and current noise, and wide signal bandwidth. They use the Analog Devices, Inc. patented DigiTrim? trimming technique, which achieves
標簽：
運算放大器
上傳時間：
2022-02-02
上傳用戶：
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C++ Algorithms for Digital Signal Processing 第4章 濾波器程序
標簽：
Algorithms
Processing
Digital
Signal
上傳時間：
2013-08-01
上傳用戶：eeworm
