共查询到18条相似文献,搜索用时 140 毫秒
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实现了G.723.1语音压缩编码在数字对讲机基带系统的应用。其创新在于充分利用了DSP的处理能力以及CPLD硬件上的高速、高集成度和可编程性进行硬件电路设计,在对讲机频带和DSP资源有限的条件下,对G.723.1的定点C代码进行深度优化,最终在实际电路上,收端可以播放出发端传来的实时、连续和清晰的语音。 相似文献
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对使用通用DSP芯片取代现有数字对讲机所采用的专用语音芯片来实现对讲机的语音处理的方案进行了研究.方法是采用TMS320C5402作为语音处理芯片,采用MSP430单片机作为控制芯片,语音算法采用G.729.该方案不仅可以取代现有专用语音芯片,而且可以使得对讲机的语音处理具有更好的通用性和可扩展性,以便实现其他多种语音算法. 相似文献
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提出了一种灵活的硬件结构模型——数字语声处理器DSP,它是实现以IS-96为代表的几种CELP语音编码算法的核心。数字语声处理器DSP可以灵活地实现多种语声编码算法。本文给出了这种数字语声处理器的结构,分析了主要的部件,并用VHDL硬件描述语言对部分部件进行了描述。 相似文献
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本文给出了一种以9.6Kb/s数据率实时工作的新的数字语音编码器.该系统由时域谐波定标(TDHS)及子频带编码(SBC)两种不同的语音压缩算法组成.阐述了TDHS的基本原理,简述了几种不同的音调检测方法.本文最后对用DSP硬件实时执行该系统的几种方式作了讨论. 相似文献
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AMR(自适应多速率)语音编码标准由于其低码率和高质量,在通信和多媒体领域得到广泛应用.针对AMR语音编码标准的算法特点,提出了一种"音频DSP软核+硬件加速器"的VLSI实现结构.这种结构能够有效地实现编码算法,同时达到低成本、低功耗的要求.综合后的电路在满足语音编码实时性的要求下能工作在50MHz频率以下,功耗仅有不到50mW,和78 k门的芯片面积.最后通过FPGA整体验证,证明这种方案是可行有效的.同时优化后的音频DSP软核和硬件加速器中的模块复用设计,使得本设计方案对G.7xx系列编码算法具有通用和可移植性. 相似文献
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语音信号压缩编码是数字语音信号处理的主要方面.在现有的语音编码中,G.729A算法在8kb/s的码率下取得了较好的语音质量,具有广阔应用前景,因此提出采用PicoBlaze和ML7204实现G.729A语音压缩/解压详细的软硬件实现方案,并描述了G.729A语音编解码器ML7204的工作原理、性能、接口,以及FPGA内嵌IP核微处理器PicoBlaze的特点和使用方法。给出硬件电路设计原理,以及各部分的具体实现方法和原理图。并给出软件流程和主要代码。实验结果表明,系统提供话音点到点的时延仅为25mS,而语音质量平均意见MOS值达到4.2。在可懂度和清晰度等性能优异,该系统设计可应用于无线移动网、数字多路复用系统和计算机通信系统。 相似文献
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当下,地震应急救援时使用对讲机通话是必不可少的通信方式。但救援人员通话时,必然要用手按住PTT才行,这就影响救援人员工作时便利性。基于此,研究采用最新数字运算处理技术以及编入了只识别人类声音的算法,使得外界声音和冲击等不被识别。当救援人员讲话时,自动识别产生PTT,就可以实现通话。救援人员的双手给彻底解放出来,讲话时对讲机自动发射,这对于通信方法优化,提高救援效率,都是有重大益处的。 相似文献
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针对安防楼宇对讲系统中回声难于消除的问题,简要阐述了回声产生的原因及原理和当前主要的回声消除技术,重点介绍了Conexant CX20707 SPoC回声消除芯片的基本特性,及在模拟楼宇对讲系统和数字楼宇对讲系统中CX20707的音频流处理设计流程。实践中基于CX20707设计的楼宇对讲产品,其声学回声和线路回声都得到... 相似文献
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Yohtaro Yatsuzuka Tomohiro Yamazaki Shigeru Hzuka 《International Journal of Satellite Communications and Networking》1986,4(4):193-202
This paper describes applications of adaptive predictive coding (APC) with maximum likelihood quantization (MLQ) which can cover a wide range of coding rates from 4.8 to 16 kb/s for low C/N satellite communication systems, such as maritime, aeronautical mobile and thin-route satellite communication systems, and also for speech and data integration, including digital circuit multiplication equipment (DCME) in business communication systems, such as INTELSAT business services (IBS). A 16 kb/s APC–MLQ hardware codec has been implemented by NEC–7720 DSP chips and the performance has been confirmed in subjective quality of speech through conversational tests. The objective performance has also been evaluated for non-voice signals, such as single and multi-frequency tones, and 1200 and 2400 b/s voiceband data signals. The APC-MLQ codec can transmit the voice-band data at 1200 b/s over two asynchronous tandem links and at 2400 b/s over one link. It was noted that the APC-MLQ codec is superior in speech performance at 16 kb/s to a narrow-band companded FM and meets requirements for low C/N satellite communication systems. For voice and data integration into 16 kb/s for 64 kb/s links, we propose a multi-media multiplexing for low C/N digital satellite communication systems and also a small-scale circuit multiplication system for business use. In these systems, a variable rate coding of APC-MLQ from 4.8 to 16 kb/s can be effectively introduced for voice and data integration. 相似文献
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Voice is the preferred method of human communication. Although there have been times when it seemed that the voice communications problem was solved, such as when the PSTN was our primary network or later when digital cellular networks reached maturity, such is not the case today. This paper addresses the challenges and opportunities starting from the basic issues in speech coder design, developing the important speech coding techniques and standards, discussing current and future applications, outlining techniques for evaluating speech coder performance, and identifying research directions. The most prominent speech coding standards are presented and their properties, such as performance, complexity, and coding delay, analyzed. Particular networks and applications for each standard are included. Further, reflecting upon the issues and developments highlighted in this paper, it becomes evident that there is a diverse set of challenges and opportunities for research and innovation in speech coding and voice communications. 相似文献
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Speech coding in mobile radio communications 总被引:1,自引:0,他引:1
Budagavi M. Gibson J.D. 《Proceedings of the IEEE. Institute of Electrical and Electronics Engineers》1998,86(7):1402-1412
Speech coding, the efficient representation of speech in digital form, is one of the key technologies in current and evolving digital cellular and wireless voice communications offerings. The speech coders in existing standards exhibit a level of sophistication and performance unimaginable just 15 years ago. We outline the characteristics of the mobile communications problem with respect to speech coders and point out the principal issues in speech coder design for these applications. Speech coding methods in existing mobile communications standards are described and contrasted. The limitations imposed by the wireless channel and by background impairments are discussed, and approaches to addressing their resulting effects are presented. Suggestions for future research in speech coding for the mobile communications problem are outlined 相似文献
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通信技术的快速发展已经彻底改变了我们的生活习惯,同时也在深刻影响着煤炭企业的管理模式和生产方式。基于通信技术,文章设计了一种基于光纤传输的数字信号传输的煤矿井下语音通信系统与广播网络系统,系统充分考虑煤矿井下应用环境,采用光纤对数据进行传输,有效提高了系统的传输效率和抗干扰能力,该语音通信系统以STM32为网络对讲模块的主控芯片,用来实现对各个模块的驱动和控制,通过以太网协议的编写与设计实现系统的通信功能。该语音通信系统实现了一对一、多对多、单向扩音广播、双向对讲、录音等功能,做到“一种系统,多种功能”,为煤矿企业节省成本,减小井下通信难度。经实验验证,该系统运行稳定,具有一定的可行性和实用性。通过对系统对讲模块的优化设计,使系统噪声大幅下降。 相似文献