新型宽带语音编解码器AMR-WB的研究

该文介绍了一种应用于第三代移动通讯系统的编解码器,同时也是第一个可同时用于无线和有线应用的编解码器,该编解码器的语音带宽拓展为50Hz到7000Hz,编码后语音的自然度很高,用在3G移动通讯系统的多媒体服务、宽带包交换网络、音频和视频会议等等。由于AMR-WB为一个全新的宽带编解码器,其标准在2001年3月刚刚通过,国外对其的研究也属于起步阶段,还没有真正进入实用阶段。尤其在国内,至今尚未见到相关的研究。故对其进行全面的分析和深入的研究是必要的,会对今后的研究打下良好的基础。     

新型宽带语音编解码器AMR-WB的语音端点检测(VAD)研究

AMR-WB是一种应用于3G上的一种新型宽带语音编解码器,该编解码器将语音带宽拓展到7kHz,采样频率为16kHz,编码后的语音在自然度、音乐方面有较大的改善,将在3G移动通信系统、ISDN上的高保真电话、数字无线广播等各个方面有广泛的应用.文章着重介绍该语音编解码器中有关语音端点检测(VAD)算法,并给出了测试结果.     

一种可重构体系结构的声码器ALU设计

文章介绍了采用可重构体系结构的TR600语音编解码器中的ALU设计。重点讨论了ALU的资源部件、数据通路、指令及在设计中的平衡规则。该ALU采用VHDL语言描述,经过仿真、综合和FPGA验证后,完全符合设计要求。     

矿井下基于蓝牙技术的语音通讯控制器设计

提出一种基于蓝牙技术的矿井下无线语音通讯控制器设计方案,解决了蓝牙语音传输在特殊环境下容易受到外界干扰的问题;在语音通讯控制器系统结构中,采用ROK101 007蓝牙芯片做为无线通信模块,使用MSM7570芯片完成语音编解码;详细介绍了软、硬件的设计,使该语音通讯控制器达到本安型标准,从而适合在矿井下的特殊场合使用,经试用后,该语音通讯控制器传输话音清晰稳定。     

G.726语音编解码器在SoPC中的实现

在对G.726语音编解码标准分析的基础上给出了基于FPGA的DSP设计流程,利用MATLAB/Simulink、DSPBuilder和SOPCBuilder工具设计了G.726语音编解码器,通过仿真实验验证了所设计的编解码器模型的正确性,实现了编解码器在SoPC系统中的综合。     

G.723.1语音编解码器优化策略的研究

在语音编解码器的实际应用中,语音编解码器的编解码时间是一个关键的性能指标。本文通过对G.723.1语音编解码器客观测试和使用VC^＋＋的工具profile对G.723.1语音编解码器测试结果的分析,提出对G.723.1语音编解码器的优化应着重于对计算复杂度较大的模块进行优化。如基音估计、自适应码本搜索、固定码本搜索等模块。并针对这些模块给出相应的优化方法。最后对根据优化策略优化的代码和未优化代码进行了测试,结果显示优化代码比未优化代码的运行时间减少了24％－31％。     

浅谈以程控交换机为枢纽实现三网灵活通讯

本文通过实践总结,介绍了如何打破技术壁垒,实现跨网灵活通讯,通过挖掘程控交换机功能,实现海关语音IP专网、移动手机虚拟网、网通固定电话虚拟网S-N灵活通讯方式,在降低通讯费用方面进行了有益的探索。     

G.723.1语音编码器在Blackfin平台上的实时实现①

ITU-TG.723.1是一种用于多媒体通信的双码率语音编码标准,几乎在所有的语音网关设备上面g723.1音频编解码器都是必须支持的一个标准编解码器。针对G.723.1音频编解码算法尚未在BF532＋uClinux平台上实时实现的情况,基于BF532＋uClinux平台提出了该算法实时实现的优化方案。方案减少了编解码的时延,降低了算法的复杂度,编解码整体性能提升约10倍,满足了BF532＋uClinux平台的实时性要求,并全部通过ITU测试向量的测试。最后将优化好的G723.1编解码器应用到嵌入式语音网关中,实验表明语音通话效果良好。     

AMR语音编解码器在C55xDSP上的实现

本文主要介绍了在手持终端上实现自适应多速率(AMR)语音业务功能的DSP解决方案,并分析了AMR语音编解码器标准及TMS320VC5510 DSP的性能特征.     

ITUG．726语音编码器在DSP上的实现

ITUG．726建议的语音编码方案是一套从16kbit/s到40kbit/s的完整速率ADPCM算法。本文介绍了有关的算法原理和在一片ADSP－2181定点DSP芯片上实时实现该编解码器过程中的软、硬件结构。     

AMR mode selection enhancement in 3G networks

This paper describes methods for mode selection in multirate speech codecs, such as the AMR (Adaptive Multi-Rate), that is the mandatory speech codec selected in 3GPP (3rd Generation Partnership Project) defined mobile networks. Originally, the multirate functionality has been developed for coping with changing radio conditions. The algorithms described in this paper find applicability in IP-based mobile networks, where speech encoded data is encapsulated using the RTP (Real Time Protocol). The main advantages offered by these techniques are improved speech quality and congestion control along the network path between two mobile terminals.     

Recognizing voice over IP: a robust front-end for speechrecognition on the world wide web

The Internet Protocol (IP) environment poses two relevant sources of distortion to the speech recognition problem: lossy speech coding and packet loss. In this paper, we propose a new front-end for speech recognition over IP networks. Specifically, we suggest extracting the recognition feature vectors directly from the encoded speech (i.e., the bit stream) instead of decoding it and subsequently extracting the feature vectors. This approach offers two significant benefits. First, the recognition system is only affected by the quantization distortion of the spectral envelope. Thus, we are avoiding the influence of other sources of distortion due to the encoding-decoding process. Second, when packet loss occurs, our front-end becomes more effective since it is not constrained to the error handling mechanism of the codec. We have considered the ITU G.723.1 standard codec, which is one of the most preponderant coding algorithms in voice over IP (VoIP) and compared the proposed front-end with the conventional approach in two automatic speech recognition (ASR) tasks, namely, speaker-independent isolated digit recognition and speaker-independent continuous speech recognition. In general, our approach outperforms the conventional procedure, for a variety of simulated packet loss rates. Furthermore, the improvement is higher as network conditions worsen     

一种引入多脉冲提高AMR_WB丢包处理质量的前向纠错技术

GSM AMR_WB是唯一可以作为有线和无线通用的语音编码标准,提高AMR_WB在丢包环境下的语音质量至关重要.本文提出了一种改进的AMR_WB丢包处理方法,发送端利用基于多脉冲的前向纠错技术,解决丢包时自适应码本带来的错误传播问题.接收端采用基音轮廓修改方法消除语音帧不同步在帧边界带来的恼人噪声.提出的方法在AMR_...     

Packet based echo cancellation for VoIP networks

In today’s modern telephony network, VoIP is fast emerging as one of the main communication techniques. However, the performance and the quality of VoIP are affected by echo. Packet Based Echo Canceller (PBEC) is introduced, as a solution to cancel echo in the VoIP network. PBEC can replace the current echo cancellers, which are located in the Public Switched Telephony Network (PSTN) central switches. The operating principle of the PBEC is explained and its advantages are highlighted. The performance of the PBEC using different speech codecs is also studied. Using the PBEC, a maximum Echo Return Loss Enhancement (ERLE) of 37.39 dB has been achieved when used with the Pulse Code Modulation (PCM) based speech codec. From the simulation results, it can be seen that the performance of the Adaptive Differential Pulse Code Modulation (ADPCM) clearly matches the performance of the PCM based speech codec. The other major problem affecting the VoIP network is the issue of packet loss. This issue of packet loss has been successfully addressed in this paper by the insertion of random values. With the insertion of random values, the ERLE increases by 4.81 dB compared to when there is no insertion of random value. The PBEC with the utilization of random values would make the VoIP a better communication tool.     

矿用数字漏泄移动通信系统中语音编解码的研究与实现

结合矿井的实际环境讨论了井下数字漏泄移动通信系统中语音编解码方法的选择原则；分析了LD—CELP算法的特点和优势，研究了该算法的性能；给出了在TMS320DM643平台上LD—CELP编解码方法的具体设计方案。     

Impairment Factor Framework for Wide-Band Speech Codecs

A new method is described for quantifying the quality degradation introduced by wide-band speech codecs via a one-dimensional impairment factor. The method is based on auditory listening-only tests, but the resulting impairment factors may be used for predicting speech quality in an instrumental way, e.g., for network planning purposes. Following the method, auditory test results are first transformed to an overall quality rating scale, and then adjusted to rule out test-specific effects. The derived impairment factors fit into the common framework which is defined by the E-model for narrow-band telephone networks, and which is hereby extended towards wide-band speech transmission. This paper presents the necessary auditory test data, describes the derivation and adjustment methodology, and provides numerical values for a range of wide-band speech codecs. The values are tested for their robustness in case of codec tandems and adjusted to represent the effects of packet loss.     

基于抖动激励的VoIP终端拥塞控制机制

针对基于H．323的VoIP系统，提出了一种抖动激励式终端拥塞控制机制，将语音分组的延迟抖动参数与RTCP协议相结合，控制动态改变语音编码方式及传输分组大小，从而缓解网络拥塞，降低传输延时和丢包率。     

Embedded bandwidth scalable wideband codec using hybrid matching pursuit harmonic/CELP scheme

Recent advances in speech coding have made wideband coding feasible at the bit-rates sufficient for mobile communication. Here we propose a novel hybrid harmocic Code Excited Linear Prediction (CELP) scheme for highband coding of band-split scalable wideband codec, where the low-band (0–4?kHz) is critically subsampled and coded selectively using existing narrowband codecs such as 5.4 kbps and 6.3 kbps G.723.1, 8 kbps G.729, and 11.8 kbps G.729E. The high-band signal is divided into stationary mode (SM) and non-stationary mode (NSM) components based on its unique characteristics. In the SM portion, the high-band signal is compressed using a multi-stage coding that combines the sinusoidal model and CELP. The first stage coding applies the damping factor matching pursuit (MP) algorithm without either the Over-Lap-Add (OLA) or smoothly interpolative synthesis schemes and the second stage utilizes CELP with the circular codebook. In the NSM portion, the high-band signals are coded by CELP with both pulse and circular codebooks by applying the complexity-reduced algorithm. To ensure scalability in highband coding, two enhancement layers are used to increase the number of pulses and control the quantizing sinusoidal parameter numbers. This paper describes the new algorithm and discuses novel techniques for efficient bandwidth wideband speech coding and subjective quality performance. For efficient bit allocation and enhanced performance, the pitch of the high-band codec is estimated using the quantized pitch parameter in low-band codec. An informal listening test, rated the subjective speech quality as comparable to that obtainable with G.722.2 as the fullband wideband codec and G.722.2 as the highband codec, the recent standardized band-split wideband codec.