共查询到19条相似文献,搜索用时 187 毫秒
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1.6Kb/s类MELP语音压缩编码器的FPGA实现 总被引:2,自引:0,他引:2
基于"CPU软核 模块算法IP"的方法对一个1.6Kb/s类MELP语音压缩编码算法进行了实现,并将整个语音压缩编码器在FPGA上进行了整体验证,实验结果说明本文给出的语音压缩编码器的实现结构是可行的,能够满足语音压缩编码算法对实时性的要求,从而为下一阶段语音压缩编码器的芯片设计提供有力的可行性论据.同时,由于本文给出的语音压缩编码器的实现结构中的各模块算法IP对于许多语音压缩编码算法中都适用,因此该语音压缩编码器的实现结构对不同的语音压缩编码算法具有一定的通用性. 相似文献
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针对CS-ACELP语音编码算法编码复杂度较高、DSP实时实现比较困难的问题,提出了一种可降低CS-ACELP语音编码算法复杂度的优化方法,分析了CS-ACELP语音编码算法原理,详细介绍了优化的CS-ACELP语音编码算法从固定码本搜索上降低算法复杂度的实现,并给出了在16位定点DSP芯片TMS320VC5402上实现CS-ACELP语音编解码方案的硬件及软件设计。实验结果表明,优化的CS-ACELP语音编码算法降低了运算复杂度,提高了运行速度,重建的语音符合标准的编解码要求。 相似文献
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马延飞 《单片机与嵌入式系统应用》2002,(1):276-279
描述MPEGI音频第三层编码(MP3)标准的实现算法;就其音频编码(44.1 kHz的采样频率)和语音编码(16kHz的采样频率)算法的异同作比较;介绍基于DSP的MP3实时编码器.该MP3实时编码器将被用来作为MP3播放器的音频及语音录音功能模块. 相似文献
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马延飞 《单片机与嵌入式系统应用》2002,(2):20-23
描述MPEGI音频第三层编码(MP3)标准的实现算法;就其音频编码(44.1kHz的采样频率)和语音编码(16kHz的采样频率)算法的异同作比较;介绍基于DSP的MP3实时编码器。该MP3实时编码器将被用来作为MP3播放器的音频及语音录音功能模块。 相似文献
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G.729a是ITU-T推出的用于PSTN的第四代语音编码标准,采用了共轭结构-算术码本激励线性预测编码(CS-ACELP)算法,其码率为8Kbps。本文在对G.729a的编解码算法作出扼要介绍后,就如何在定点DSP芯片TMS320C541上实时实现该编码算法做出了具体讨论,包括系统的软硬件设计及关键技术。随后文中给出了详细的实验结果以供分析。根据测试结果,最后得出结论:在'C541上实现一路全双工G.729a编解码器需程序空间7.23K字、数据空间6.7K字,其算法复杂度最大为15.5MIPS。 相似文献
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该文在研究语音编码技术原理的基础上,详细研究了G.729A语音压缩算法实现过程,并在VC和DSP C5000 CCS2.0软件中对其进行了仿真,最后对G.729A语音压缩的结果和性能进行了分析。 相似文献
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G.729语音编码算法研究及基于DSP的实现 总被引:1,自引:0,他引:1
对G.729语音编解码算法的原理进行了简要分析,并提出了一种基于DSP芯片TMS320VC5510的语音编解码算法的实现方法。针对算法特征及体系结构的特点,提出了一些有效的优化措施。实验结果表明,运算复杂度大大降低,且在语音的编解码压缩过程中具有很好的重建效果。 相似文献
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该文根据G.729编解码理论和具体解码流程,结合TI公司DSP产品TMS320C6000系列的TMS320DM642、CCS6000集成开发环境以及G.729的硬件实现平台,提出了DSP传送数据给ARM的算法,以及实现G.729解码算法的主要程序,在最后给出了G.729解码的结果。实验表明:该方案能够成功地实现语音解码。该方法具有低延迟、低速率、高语音质量的优点。 相似文献
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《IEEE transactions on audio, speech, and language processing》2007,15(2):632-640
This paper proposes a new speaker-dependent coding algorithm to efficiently compress a large speech database for corpus-based concatenative text-to-speech (TTS) engines while maintaining high fidelity. To achieve a high compression ratio and meet the fundamental requirements of concatenative TTS synthesizers, such as partial segment decoding and random access capability, we adopt a nonpredictive analysis-by-synthesis scheme for speaker-dependent parameter estimation and quantization. The spectral coefficients are quantized by using a memoryless split vector quantization (VQ) approach that does not use frame correlation. Considering that excitation signals of a specific speaker show low intra-variation especially in the voiced regions, the conventional adaptive codebook for pitch prediction is replaced by a speaker-dependent pitch-pulse codebook trained by a corpus of single-speaker speech signals. To further improve the coding efficiency, the proposed coder flexibly combines nonpredictive and predictive type method considering the structure of the TTS system. By applying the proposed algorithm to a Korean TTS system, we could obtain comparable quality to the G.729 speech coder and satisfy all the requirements that TTS system needs. The results are verified by both objective and subjective quality measurements. In addition, the decoding complexity of the proposed coder is around 55% lower than that of G.729 annex A 相似文献
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文章分析了错误隐藏技术在不同的网络丢包情况下的性能以及不同参数的敏感程度,提出了一种非对称网络丢包错误保护技术,并与G.729语音编码器相结合,提出了一种固定码率12.0kbps的网络自适应语音编码器。实验表明其具有良好的效果。 相似文献
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Sean A. Ramprashad 《International Journal of Speech Technology》1999,2(4):359-372
A two stage hybrid embedded speech/audio coding structure and algorithm are proposed. The first stage of the structure consists of a core speech coder which provides a minimum output bit rate and acceptable performance on clean speech inputs. The second stage is a perceptual/transform based coder which provides a separate optional bitstream for the enhancement of the core stage output.The two stage structure can be used to enhance the quality of an existing codec without modification of the original coding algorithm. In this regard it can be considered a value added option that can be used with a standard (existing) system. The structure can also be used in systems in which many users/systems force the coding algorithm to work simultaneously under multiple constraints of bitrate, complexity, delay, and coding quality.Informal testing of the algorithm has been done using ITU-T standard G.723.1 at 5.3 kb/s as a core coder. The maximum combined bitrate from the core and enhancement stages for the tests is 16 kb/s. The tests show that the second stage significantly improves the quality of the core output in the cases of music and speech with background noise. Compared to the non-embedded fixed rate standard LD-CELP G.728 at 16 kb/s, the quality of the two stage structure is generally lower on these inputs; the embedded feature does affect quality. On clean speech the quality of the two stage structure at 16 kb/s is close to if not better than that of G.728 at 16 kb/s. 相似文献