利用LPC2300系列ARM芯片的I~2S实现WAV文件播放

I2S(Inter-IC Sound Bus,集成电路内置音频总线)是工业领域或嵌入式系统领域常用的音频总线之一,它只处理音频数据,而其它信号如编码、控制等单独传送.使用I2S总线bus和支持该接口的编解码芯片连接,从而构建一个嵌入式音频系统.     

基于DSP与TLV320AIC23B的音频处理系统

介绍了基于DSP的音频处理技术,提供采用音频编解码芯片TLV320AIC23和DSP理器实现的音频处理系统的典型解决方案.音频编解码芯片完成模拟音频信号与数字信号之间的相互转换,包括语音信号采集和语音信号发送两部分.DSP处理器则完成对经模数转换后的语音信号在数字域处理的过程.该方案可以充分发挥DSP所具有的灵活性好、处理速度快的特点.     

Cirrus Logic推出集成先进音频转换器的32位DSP

Cirrus Logic公司针对音频放大器应用推出高性能的音频系统级芯片（SoC）CS47048。CS47048在单个IC中集成了先进的32位音频DSP、高性能多声道音频编解码器和数字音频接收器／发射器,极大地减少了整个电路板空间,降低了系统成本。     

基于FPGA的多声道数据采集与监测系统

本研究提出了一个音频数据采集和处理的硬件可重构系统.可以完成多声道音频采集和存储.本设计采用系统级芯片技术和软硬件协同设计集成到一个FPGA芯片.     

基于DSP5410与TLC320AD50C的实时音频信号处理研究

介绍一种高保真的音频信号处理系统,给出了前端音频信号调理电路设计,实现了DSP芯片TMS320VC5410与语音转换芯片TLC320AD50C的硬件接口设计,并通过FIR算法实现了对音频信号的处理.测试结果表明,此音频处理系统达到预期要求.     

一种基于DSP的音频采集与回放系统

介绍一种基于DSP的音频信号处理系统。该系统采用TI公司的低功耗数字信号处理器TMS320VC5509A作为主处理器,采用能与之无缝连接的TLV320AIC23作为音频CODEC芯片。在此基础上完成系统硬件平台的搭建和软件设计,可以作为对音频信号处理的通用平台。本文详细介绍DSP与CODEC芯片的接口设计方法,提出一种利用DMA中断的方式进行数据传送的方法,大大提高了处理速度。     

基于ARMLinux的嵌入式音频系统设计

介绍了由处理器S3C2410和语音处理芯片UDA1380组成的基于ARMlinux的音频系统的构建.提出了软硬件设计方案,并就音频文件的录制、传输、播放进行了探讨.利用IIs总线,IIC总线实现了音频数据及控制信号的传输,在驱动程序设计中通过使用DMA传输及缓存分段技术提高了对音频数据的实时处理,达到了较好实时性效果,实现了录放同步的音频系统.     

视频/音频处理器AV4400A简介

介绍了一种 L ucent公司的新一代视频 /音频处理器 AV44 0 0 A的应用特点、主要性能及芯片内部的系统结构和功能框图 ,并简要介绍了以 AV44 0 0 A为核心芯片的 H.3 2 4视频会议系统终端设计     

SMSC推出带嵌入式多信道USB2.0音频控制器的三频数字无线音频处理器

<正>智能型混合信号连接解决方案(Smart Mixed-Signal ConnectivityTM)领导厂商SMSC于2011年1月11日发表新款DARR83芯片,这是业界最完整的数字无线音频处理器,可支持三种频段并带一个嵌入式多信道USB 2.0音频控制器,可适用于各种不同的数字无线音频应用。     

SMSC推出业界完整的带嵌入式多信道USB 2.0音频控制器的三频数字无线音频处理器

智能型混合信号连接解决方案(Smart Mixed-Signal ConnectivityTM)领导厂商SMSC近日发表新款DARR83芯片,这是业界最完整的数字无线音频处理器,支持三种频段并带一个嵌入式多信道USB2.0音频控制器,适用于各种不同的数字无线音频应用。     

嵌入式流媒体系统的设计

介绍了基于ARMlinux的嵌入式流媒体系统的构建。通过RTP/RTCP实现了流媒体数据服务端的传输,客户端利用IIS总线,IIC总线实现了音频数据及控制信号的传输,在驱动程序设计中通过使用DMA传输及缓存分段技术提高了对音频数据的实时处理,达到了较好的实时性效果。     

基于压缩感知和音频指纹的固定音频检索方法

针对现有音频检索中样本音频特征库数据量较大且检索速率慢问题, 本文提出一种基于压缩感知和音频指纹降维的固定音频检索方法. 在音频检索的训练阶段, 首先, 对样本音频信号进行稀疏化处理, 并通过压缩感知算法对稀疏化后的音频数据进行压缩; 其次, 提取压缩信号的音频指纹; 再次, 引入音频指纹离散基尼系数通过计算音频指纹各维度的离散基尼系数对指纹实施降维, 最终得到检索特征库. 在音频检索阶段用和训练阶段相同的算法提取待检音频的特征与音频特征库数据匹配得出检索结论. 实验结果表明, 所提音频检索方法在确保较好的检索准确率的基础上, 大幅度减小了样本音频数据库的存储量, 提高了音频的检索速率.     

基于替代数据检测的音频信号非线性分析

针对轻微含噪的音频信号,本文提出了一种基于替代数据检测的非线性分析方法。该方法首先根据音频信号的线性假设生成多组替代数据,然后分别计算原始数据和替代数据的样本峰度,最终根据假设检验方法判断原始音频中是否包含非线性成分。实验结果表明,所提方法能够有效地验证音频信号的非线性特性。相比于传统基于最大Lyapunov指数的非线性分析方法,该方法能更好地区分音频信号和噪声信号。     

基于VPRSM的音频特征选择

在音频索引中保持音频特征非常重要,但是在很多情况下特征数量又很庞大,直接处理这些海量数据是非常耗时的.特征选择作为数据挖掘的一个处理步骤,在特征维数的减少和非相关数据的约简方面已经有很成功的使用.提出了一种基于变精度粗糙集模型(variable precision rough setmodel,VPRSM)的音频特征选择算法.实验结果表明,该算法能够得到最小约简,并且最大程度地保持了音频数据的特征,提高检索效率.     

Search audio data with the wavelet pyramidal algorithm

To solve the problem of query-by-example in multimedia audio data, we analyze the characteristics of wavelet multiresolution, and propose a method for searching audio data using wavelet multiresolution pyramidal algorithm. First, we preprocess 100 audio data files by transforming them into wavelet coefficient files with audio data and query data. And then we search these wavelet coefficient files using the different hierarchy pyramidal algorithms. By comparing our method with the method using wavelet approximate coefficient algorithms, we found that our method is highly efficient and reduces the search time without influencing the retrieval precision.     

Toward bandwidth-efficient and error-robust audio streaming over lossy packet networks

Bandwidth efficiency and error robustness are essential issues for different multimedia streaming applications. This paper presents strategies for high-quality audio streaming based on fragmenting perceptually coded audio frames and shuffling the data components among multiple packets for transportation. This is done to increase robustness against packet loss. We also address the delivery of audio data consisting of components with different proportional priorities. Our approach is rationalized with streaming tests using the MPEG AAC audio codec in a simulated network environment and formal listening tests to evaluate the resulting audio output. According to the results, the proposed schemes improve audio quality significantly with reasonable increase to network resource utilization compared to traditional error robustness measures.     

基于DSP的便携式综合音频播放器的实现

目前各种便携式媒体播放器比较流行 ,数字音频广播 DAB也正在发展。本文主要介绍了以 DSP数据处理技术为基础 ,如何把媒体音频播放 ,数字音频广播 DAB集成在一起 ,组成新的便携式综合音频播放器。该便携式综合音频播放器将会有良好的市场前景     

Quality-driven secure audio transmissions in wireless multimedia sensor networks

Many audio applications such as audio surveillance and human acoustic health monitoring require security protections for audio streaming over WSNs. The process of watermarking which embeds small amounts of data (i.e., the watermark) into the original audio is an effective technique to ensure the integrity of received audio data at the receiver in energy-constrained WSNs. However, the selection of positions to embed watermark into audio streams is critical to both received audio quality and watermarking authentication performance in error-prone wireless transmission environments. In this paper we propose an approach that dynamically determines the range of middle sub-band components for embedding the watermark with minimum quality distortions, based on psycho-acoustic models and adaptive sub-band thresholds. In addition, through unequal network resource allocation schemes the proposed approach protects both middle sub-bands and high sub-bands, which include the important audio components. Our theoretical analysis and simulation results demonstrate that the proposed quality-driven energy-efficient watermarking approach for audio transmissions can achieve considerable performance gains in WSNs.     

Compensation of audio data with a high frequency components for realistic media FTV

Like the concept of free-viewpoint TV (FTV), the audio data should be rendered according to video data. However, on condition that minimum numbers of microphone are used, it is difficult to acquire accurate audio signal for rendering audio data to the image with the choice of view point. Especially, degradation of high frequency components (HFC) happens due to the characteristic of polar pattern for microphone. The degradation of HFC causes imperfection of signal restoration and leads to the degradation of clarity for hearing. In this paper, a compensation method for the degradation of HFC audio signal is proposed for producing an immersive audio effect at realistic media. Our experimental results show that low frequency components (LFC) of audio signal had a little directional degradation in spite of effect of the polar patterns of microphone and the compensation of HFC can be realized with adapting the attenuation inclination of LFC. This research is expected to be helpful for producing an immersive audio effect for a realistic media.