AMR-WB+技术性能分析和测试

第三代移动通信系统提供的语音短信、流媒体和数字广播服务有着广阔的商业空间和发展潜力,所以一种高效低码率的音频编码方法对移动通信就有着极其重要的意义.3GPP已经把AHR-WB 格式作为其3G流媒体音频部分的编码标准.在低编码速率范围内(JA10Kbps到24Kbps),AHR-WB 编解码格式对音频的处理表现出独特的性能.本文对AMR-WB 进行较深入的研究,并分析和测试AHR-WB 算法的改进带来的音质改善.     

低比特率音频编解码在嵌入式平台中的移植与优化

本文描述了AMR—WB＋音频编解码的结构框架,复杂性,移植和优化。AMR—WB＋编解码在低码率范围内表现出相当优秀的品质,并且在各种音频类型上都有一致的高性能。因此被3GPP和DVB选定为移动网络中低码率音频指定编解码算法。     

The adaptive multirate wideband speech codec: system characteristics, quality advances, and deployment strategies

Wideband speech is the major differentiation and attraction of third-generation network services in both the circuit and packet switched domain. Increased audio bandwidth introduces a significant leap in perceived quality of service compared to currently utilized narrowband telephony in second-generation mobile communications and the PSTN. The adaptive multirate wideband (AMR-WB) speech codec is the service enabler for improved user experience. It is an established 3GPP and ITU-T wideband speech codec standard and represents the state-of-the-art in speech quality as well as robustness in error prone radio channels. It is also the first codec algorithm standardized for wideband speech for mobile communications.     

G.718: A New Embedded Speech and Audio Coding Standard with High Resilience to Error-Prone Transmission Channels

This article is an overview of the standardization, architecture, and performance of the new ITU-T Recommendation G.718. G.718 is an embedded variable bit rate codec providing a scalable solution for compression of 8 and 16 kHz sampled speech and audio signals at rates between 8 kb/s and 32 kb/s. It comprises five layers where higher-layer bitstreams can be discarded without affecting the lower layersiquest decoding. The codec also has an optional core layer interoperable with ITU-T G.722.2 (3GPP AMR-WB) at 12.65 kb/s. G.718 was designed to provide high speech quality at low bit rates and to be robust to significant rates of frame erasures or packet losses. It is also targeting good quality for generic audio at higher rates.     

Challenges of 16 kHz in acoustic pre- and post-processing for terminals

The standardization of the AMR-WB speech codec by 3GPP opens a new area on audio pre-and post-processing integrated in telecommunication terminals. It appears that the transition from narrowband to wideband signals is not a trivial task, but can rather be seen as a challenge, as acoustic quality requirements, scalability and computational limitations have to be faced at the same time. This article presents key examples to address the transition issue as well as proposals for low-complexity implementations of acoustic echo cancellation and noise reduction.     

On the architecture, operation, and applications of VMR-WB: the new cdma2000 wideband speech coding standard

This article is an overview of the architecture and operation of the VMR-WB5 a source- and network-controlled variable-rate multimode codec designed for robust processing of wideband speech. To enable a smooth transition from legacy narrowband voice services, VMR-WB is also capable of processing conventional telephone-bandwidth speech. The VMR-WB codec is interoperable with AMR-WB at certain bit rates, thus eliminating quality degradation and additional delay due to transcoding.     

基于信号规整和稀疏变换的语音与音频分层编码方法

基于语音和音频信号的固有周期性特征,本文构建了一种适合语音和音频信号的统一分析/合成模型,并分别在24kbps和32kbps码率下,实现了对宽带语音和音频信号的高质量分层编码.首先,本文将具有时变周期的输入信号规整为具有固定周期的信号,并对规整后的周期信号构建规整矩阵;其次,对规整矩阵的行和列分别进行调制叠接变换(MLT)和离散余弦变换(DCT),完成规整矩阵的稀疏化;最后,利用分带量化和矢量哈夫曼编码完成稀疏矩阵元素的量化和编码.主客观测试结果表明,本文所提方法的语音、音频及其混合信号的编码质量均优于同等速率下的ITU-T G.722.1和AMR-WB编码器.     

宽带编码中的有效感觉加权滤波器

论文描述了为得到高质量的宽带语音而使用的有效编解码方法。CELP(Code Excited Linear Prediction)技术应用于窄带语音编码时已获得了很高的语音质量,然而直接应用于宽带语音信号编码时不能有效地保持高质量语音,因此需要在CELP模型上添加额外的方法,以使其在宽带信号上亦取得高质量。文章所讨论的提高CELP模型性能的有效技术是感觉加权滤波器,其技术也被用于3GPP所选的AMR-WB(Adaptive Multi-Rate Wideband)[1][2]声码器中。     

AVS音频编解码性能分析

为了对AVS标准音频部分的编解码质量有一个客观评价，2005年8月，AVS标准工作组组织会员单位对AVS音频和LAME MP3进行了内部主观测试，测试结果表明：AVS音频强于LAME MP3；2005年底，AVS音频技术和EVD的ExAC音频技术进行了融合，为了验证融合的效果，又进行了一次主观测试，测试结果表明，融合后的AVS音频编解码质量得到小幅提高。     

下一代网络话音编码标准研究

在话音和音频编码中,代表数字话音或音频信号的位数尽可能地少,但要保持一个合理的可识别质量水平。这主要通过从信号中去除冗余和非相干信息来实现。本文给出了下一代网络中话音编码标准的分类方法,并详细分析了波形编解码、参量编解码和混合编解码标准的目标与原理,最后研究了话音编码标准的应用环境。     

AMR-WB算法仿真及其在TMS320VC5509的优化实现

自适应多速率宽带语音编码标准(AMR-WB)是第一个同时用于有线和无线通用标准的宽带语音编码器.介绍了AMR-WB算法井进行了AMR-WB算法定点C代码的仿真实现以及将该算法移植到TMS320VC5509DSP芯片的实现过程,具体包括C5509存储器的分配,C5509与PC机兼容性问题的处理;并对该算法进行初步优化,基...     

从有损到无损的音频编解码框架

无损音频编码技术是用于数字音频数据存档,高质量音频编码的一种重要技术,提出了一种兼容有损编解码器的无损音频编解码系统设计。包括通过有损编码残差进行无损扩展和独立的无损编解码两部分。系统采用了声道去相关、整型提升小波、线性预测、残差处理和算术熵编码等技术。在相当复杂度条件下,达到与国际主流无损音频编码技术相当的压缩性能。     

一种大功率数字音频系统设计

音频编解码芯片WM8731因其高性能、低功耗等优点在很多音频产品中得到了广泛应用。本文提出了WM8731与FPGA的音频编解码系统,并嵌入大功率D类功放技术作为音频系统的功率放大应用,使得本系统效率高,体积小,音质高,性能显著。     

G.719: The First ITU-T Standard for High-Quality Conversational Fullband Audio Coding

This article presents an overview of the recently standardized ITU-T G.719 codec, its key technologies, and their impact on audio quality. These technologies, while leading to exceptionally low complexity and small memory footprint, result in high fullband audio quality, making the codec a great choice for any kind of communication devices, from large telepresence systems to small low-power devices for mobile communication.     

A SEMI-OPEN-LOOP CODING MODE SELECTION ALGORITHM BASED ON EFM AND SELECTED AMR-WB＋ FEATURES

To solve the problems of the AMR-WB+ (Extended Adaptive Multi-Rate-WideBand) semi-open-loop coding mode selection algorithm, features for ACELP (Algebraic Code Excited Linear Prediction) and TCX (Transform Coded eXcitation) classification are investigated. 11 classifying features in the AMR-WB+ codec are selected and 2 novel classifying features, i.e., EFM (Energy Flatness Measurement) and stdEFM (standard deviation of EFM), are proposed. Consequently, a novel semi-open-loop mode selection algorithm based on EFM and selected AMR-WB+ features is proposed. The results of classifying test and listening test show that the performance of the novel algorithm is much better than that of the AMR-WB+ semi-open-loop coding mode selection algorithm.     

DRA算法及其实时解码器设计

DRA是一种新的音频编解码标准,其在每声道64 Kbit/s时重建的音质达到ITU-R规定的"人耳不可识别损伤"的主观音质评定.研究了DRA的编码与解码原理,在对解码算法进行了优化的基础上.设计并实现了基于PXA270平台的DRA实时解码器.主观听音测试结果表明.该解码器音质良好.满足实时解码应用的要求.     

AVS-P10移动音频编解码标准与关键技术

为改变数字音视频的核心技术和专利长期受制于人的现状,AVS工作组指定由武汉大学牵头制定了移动音频编解码技术标准——AVS-P10,其先进的ACELP/TVC混合编码框架能保证在中低码率下对各种复杂信号进行编码,测试表明其性能与AMR-WB+相当。AVS-P10具有高效灵活的压缩特性、强大的错误保护机制、低复杂度模式及自主的知识产权状况,在3G通信、移动和网络多媒体信息业务等领域具有广泛的应用前景。     

MPEG-4音频压缩技术的新进展--ALS

在语音信号处理的许多应用中,采用了无损音频压缩的方法.回顾了MPEG-4音频技术的现状,MPEG-4 ALS(Audio Lossless Coding)标准化过程以及编解码技术.详细介绍了编码器中的线性预测、量化、熵编码等模块,展望了MPEG-4 ALS的应用前景.     

低码率音频编码技术综述

介绍了当今主流低码率音频编码标准及其关键技术,包括EAAC 及其SBR技术,PS技术,AMR-WB 及TCX技术,G.729.1及嵌入式分层编码技术.     

Subband-multipulse digital audio broadcasting for mobile receivers

The audio quality, robustness and implementational complexity of a novel mobile digital audio broadcast scheme are addressed. The audio codec proposed is based on an efficient combination of subband coding (SBC) and multipulse excited linear prediction coding (MPLPC). The bit allocation is dynamically adapted according to both the signal power in different subbands and a perceptual hearing model. Typically a segmental signal to noise ratio (SEGSNR) in excess of 30 dB associated with high fidelity subjective quality was achieved for 2.67-b/sample transmissions at a bit rate of 86 kb/s. Perceptually unimpaired audio quality was achieved for a bit error rate (BER) of about 10^-4, when injecting random errors, which was degraded for increased BERs. In order to provide robust error protection, the audio codec was also subjected to a rigorous bit sensitivity analysis. Four different forward error correction schemes were investigated in order to explore the complexity, bit rate, and robustness tradeoffs