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
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)
上傳時間： 2016-05-06
上傳用戶：fagong
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
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
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
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
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
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
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：
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
