In this project we analyze and design the minimum mean-square error (MMSE) multiuser receiver for uniformly quantized synchronous code division multiple access (CDMA) signals in additive white Gaussian noise (AWGN) channels.This project is mainly based on the representation of uniform quantizer by gain plus additive noise model. Based on this model, we derive the weight vector and the output signal-to-interference ratio (SIR) of the MMSE receiver. The effects of Quantization on the MMSE receiver performance is characterized in a single parameter named 鈥漞quivalent noise variance鈥? The optimal quantizer stepsize which maximizes the MMSE receiver output SNR is also determined.
標簽： mean-square multiuser receiver project
上傳時間： 2014-11-21
上傳用戶：ywqaxiwang
摘 要 文章以空間監控系統為背景,深入研究了JPEG圖像壓縮標準的實現方法,并基于FPGA對其進行了實現和優化。文中給出了詳細的實現方法和優化過程,測試表明達到了很好的效果。 簡單介紹了有損靜態圖像壓縮當前有兩種比較流行的標準JPEG和JPEG2000。說明了用JPEG方法壓縮的原因。 介紹JPEG基本原理：JPEG對灰度圖像的壓縮處理過程主要包括:圖像分割,離散余弦變換(DCT),量化(Quantization),“Z”形排序(Zigzag Scan),差分脈沖編碼調制(Differential Pulse Code Modulation,DPCM)對直流系數(DC),行程長度編碼(Run-Length Encoding,RLE)對交流系數(AC),霍夫曼(Huffman)編碼等。 JPEG標準的特點是離散余弦變換。 比較詳細介紹壓縮系統的構成和實現。實現提及步驟， JPEG壓縮模塊設計和編碼模塊實現細節。
上傳時間： 2013-12-25
上傳用戶：410805624
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
INTRODUCTION In the past, adding speech recording and playback capability to a product meant using a digital signal processor or a specialized audio chip. Now, using a simplified Adaptive Differential Pulse Code Modulation(ADPCM) algorithm, these audio capabilities can be added to any PICmicro device. This application note will cover the ADPCM compression and decompression algorithms, performance comparison of all PICmicro devices, and an application using a PIC16C72 micro-controller.DEFINITION OF TERMS step size -value of the step used for Quantization of ana-log signals and inverse Quantization of a number of steps.Quantization -the digital form of an analog input signal is represented by a finite number of steps.adaptive Quantization -the step size of a quantizer is dramatically changed with time in order to adapt to a changing input signal.inverse quantizer -a finite number of steps is converted into a digital representation of an analog signal.
上傳時間： 2022-06-20
上傳用戶：
