共查询到19条相似文献,搜索用时 109 毫秒
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语音编码方案的选取对移动通信系统的通话质量、信道容量等有重要影响。文章对自适应多速率(AMR)技术、自适应机制、技术优点等进行了介绍,并结合实际测试结果分析了GSM网络中应用AMR技术以达到提高网络容量、改善无线链路性能的目的,最后介绍AMR-HR在实际网络应用中所采用的干扰消除技术、设备厂商支持情况、终端支持情况以及在现网应用中要注意的地方。 相似文献
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文章分析了自适应多速率(AMR)语音编码器的编/解码器的原理;讨论了AMR模式选择的自适应机制;对WCDMA系统的语音信道编码进行了详细地分析,并给出链路匹配参数,并以此为基础进行了AMR的性能分析。为AMR在WCDMA系统中语音传输的应用和实现提供了有益的参考。 相似文献
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分析了在无线通信系统中引入AMR技术的优点,以及时系统容量和语音质量的影响。在对GSM/GPRS系统仿真的基础上,得出了一些反映无线通信系统性能参数的仿真结果,如阻塞率、语音质量在引入AMR技术后的改善情况。 相似文献
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介绍了目前第3代移动通信系统中最主要的2种可变速率语音编码器——AMR和SMV的原理,同时从编码方式、自适应原理、复杂度、误帧处理、系统延迟和合成语音质量这几个方面对这2种多速率语音编码器进行了比较。 相似文献
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自适应多速率语音编码是3GPP制定的主要应用于第三代移动通信W—CDMA系统中的语音压缩编码,当无线环境中的信道质量发生改变时,能通过自适应地改变语音编码方式对其进行差错纠正和保护。对比了传统GSM与AMR2种编码技术,对其在不同环境下的部分性能进行测试并分析结果;结合现有AMR算法原理,提出了一种有效降低运算量的代数码本快速搜索的方法,最后对其结果进行仿真验证。 相似文献
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A new system enhancement method is proposed for the EIA/TIA-136 system offering both channel operational range extension and improved performance within the current operational range. The existing time-division multiple-access (TDMA) (136) speech codec, the IS-641 enhanced full rate vocoder, operates at a fixed bit rate and does not allow the reallocation of bits to channel error protection as channel conditions degrade. The research presented here investigates the application of the narrow-band adaptive multirate (NB-AMR) speech codec and the wide-band AMR (WB-AMR) codec, both originally designed for the 200 kHz GSM channel, in the TDMA (TIA/EIA-136) 30-kHz system. In particular, we investigate adaptively allocating bits between NB/WB speech coding and error control coding within the limited channel bandwidth. Four modes out of 17 have been carefully chosen for the new TDMA/AMR system. Switching between codec rates as channel conditions change produces range extension below a C/I of 15 dB while also improving performance in the existing operational range above 15 dB. We keep the time slot formats unchanged so that our method is completely compatible with existing 136 systems. 相似文献
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In this paper, block constrained trellis coded vector quantization (BC‐TCVQ) is presented for quantizing the line spectrum frequency parameters of the wideband speech codec. Both a predictive structure and a safety‐net concept are combined into BC‐TCVQ to develop the predictive BC‐TCVQ. The performance of this quantization is compared with that of the linear predictive coding vector quantizer used in the AMR‐WB codec, demonstrating reductions in spectral distortion. 相似文献
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This letter proposes a new embedded speech coding structure based on the Adaptive Multi‐Rate Wideband (AMR‐WB) standard codec. The proposed coding scheme consists of three different bitrates where the two lower bitrates are embedded into the highest one. The embedded bitstream was achieved by modifying the algebraic codebook search procedure adopted for the AMR‐WB codec. The proposed method provides the advantage of scalability due to the embedded bitstream, while it inevitably requires some additional computational complexity for obtaining two different code vectors of the higher bitrate modes. Compared to the AMR‐WB codec, the embedded coder shows improved speech qualities for two higher bitrate modes with a slightly increased bitrate caused by the decreased coding efficiency of the algebraic codebook. 相似文献
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Sung-Min Oh Sunghyun Cho Jae-Hyun Kim Jonghyung Kwun 《Communications Letters, IEEE》2008,12(5):374-376
This letter proposes an efficient uplink scheduling algorithm for voice over Internet protocol (VoIP) services with adaptive multi-rate (AMR) speech codec in IEEE 802.16e/m systems. The proposed scheduling algorithm adopts the random access scheme during silent-period to reduce the waste of uplink bandwidth considering the characteristics of AMR speech codec. The numerical results show that the proposed algorithm can increase the maximum supportable number of voice users by 26% compared to the conventional extended real-time polling service (ertPS). 相似文献
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基音周期搜索的准确性将直接影响到语音编码器的编码质量和效率。本文根据AMR—WB+标准中基音周期搜索算法会发生检测基音倍频和半频错误,提出了开环基音搜索算法。该算法以白相关函数为基础,利用基音周期的平滑性,引入基音周期全局参考作为基音周期判断的辅助条件,有效解决了基音周期加倍的问题并在基音周期预测中体现基音周期的平滑性,实验结果表明本文算法性能优于AMR—WB+中的算法性能。该算法已应用到AVS—P10移动音频编解码框架中。 相似文献
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AMR语音编解码在DSP上的实现与优化 总被引:1,自引:0,他引:1
介绍了自适应多速率(AMR)语音编解码算法在ZSP500DSP上的代码优化。ZSP500数字信号处理器是LSILogic公司生产的第二代ZSP数字信号处理器,它是一种低功耗、高性能的DSP,广泛应用于音频、视频处理领域。简要介绍了AMR以及ZSP500的基本特性,详细讨论了基于3GPPTS26.073V6.0.0给出的C程序源代码转化为汇编在速度和空间上的优化技巧,并给出了优化结果。 相似文献
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分析了Adaptive Multi Rate(简称AMR)编码方案与G.729编码方案的异同,在此基础上介绍了一种从AMR编码到G.729编码的参数层直接转换方案。与传统的编码转换方案相比,在语音质量的损失可以接受的前提下,算法复杂度有较大的降低。 相似文献