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基于国际电信联盟标准化组织(ITU-T)编码标准G.729.1,本文提出了一种嵌入式变速率立体声语音与音频编码方法.本算法利用G.729.1和改进的调制叠接变换(Modulated Lapped Transform,MLT)编码技术对输入信号的中值与边带信息进行分层编码,形成具有嵌入式结构的码流.编码器可处理宽带和超宽带的立体声信号,宽带立体声信号编码的最大码率为48kb/s,超宽带立体声信号编码的最大速率为64kb/s.实现结果表明,本编码器的编码质量均达到了ITU-T对G.EV-VBR立体声编码的指标要求. 相似文献
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In March 2008 the ITU-T approved a new wideband speech codec called ITU-T G.711.1. This Recommendation extends G.711, the most widely deployed speech codec, to 7 kHz audio bandwidth and is optimized for voice over IP applications. The most important feature of this codec is that the G.711.1 bitstream can be transcoded into a G.711 bitstream by simple truncation. G.711.1 operates at 64, 80, and 96 kb/s, and is designed to achieve very short delay and low complexity. ITU-T evaluation results show that the codec fulfils all the requirements defined in the terms of reference. This article presents the codec requirements and design constraints, describes how standardization was conducted, and reports on the codec performance and its initial deployment. 相似文献
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《Communications Magazine, IEEE》2008,46(10):116-122
In May 2005 the ITU-T approved ITU-T G.722.1 Annex C. This Annex extends the G.722.1 codec from 7 kHz audio bandwidth to 14 kHz, 1 octave better than the current ITU-T wideband algorithms. It is a low-complexity audio codec operating at 24, 32, and 48 kb/s. Like the main ITU-T G.722.1 recommendation, its main application is videoconferencing. This article presents the standardization stages from the definition of the foreseen applications and specification of the requirements to the performance evaluation. The adaptation and design of new software tools (included in the ITU-T software tool library, STL), and the utilization of ITU-T P-series recommendations (dealing with transmission characteristics and performance characterization of telephone transmission quality) are also presented. 相似文献
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ITU-T.G.723.1为国际电信联盟(ITU)制定的5·3bit/s和6.3kbit/s双速率语音编码建议,分别采用代数码激励线性预测(ACELP)算法和多脉冲最大似然量化(MP-MLQ)算法。在阐述G.723.1建议编译码算法的原理和实现的基础上,重点介绍了在开发基于TMS320VC5409实时实现该建议的全双工编译器过程中所做的工作。该语音编译码器通过了G.723.1所有测试矢量的验证。 相似文献
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This paper presents several strategies to improve the performance of very low bit rate speech coders and describes a speech codec that incorporates these strategies and operates at an average bit rate of 1.2 kb/s. The encoding algorithm is based on several improvements in a mixed multiband excitation (MMBE) linear predictive coding (LPC) structure. A switched-predictive vector quantiser technique that outperforms previously reported schemes is adopted to encode the LSF parameters. Spectral and sound specific low rate models are used in order to achieve high quality speech at low rates. An MMBE approach with three sub-bands is employed to encode voiced frames, while fricatives and stops modelling and synthesis techniques are used for unvoiced frames. This strategy is shown to provide good quality synthesised speech, at a bit rate of only 0.4 kb/s for unvoiced frames. To reduce coding noise and improve decoded speech, spectral envelope restoration combined with noise reduction (SERNR) postfilter is used. The contributions of the techniques described in this paper are separately assessed and then combined in the design of a low bit rate codec that is evaluated against the North American Mixed Excitation Linear Prediction (MELP) coder. The performance assessment is carried out in terms of the spectral distortion of LSF quantisation, mean opinion score (MOS), A/B comparison tests and the ITU-T P.862 perceptual evaluation of speech quality (PESQ) standard. Assessment results show that the improved methods for LSF quantisation, sound specific modelling and synthesis and the new postfiltering approach can significantly outperform previously reported techniques. Further results also indicate that a system combining the proposed improvements and operating at 1.2 kb/s, is comparable (slightly outperforming) a MELP coder operating at 2.4 kb/s. For tandem connection situations, the proposed system is clearly superior to the MELP coder. 相似文献
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High-quality speech codec modules operating at 16 and 8 kb/s have been developed using an adaptive predictive coding with adaptive bit allocation (APC-AB) scheme. An optimized APC-AB algorithm is studied that reduces processing complexity while maintaining speech quality. The coding algorithm is implemented in two digital signal processors (DSPs). The DSP chips, a framing LSI circuit, a PCM codec, and some peripheral ICs are integrated in each of two compact packages, i.e. codec modules, operating at 16 or 8 kb/s. The codec module size is as small as 80 mm×50 mm×12 mm, and its typical power consumption is 500 mW using 2-μm CMOS LSI technology. At 16 kb/s this APC-AB codec achieves high speech quality, close to that of a 7-bit μ-law PCM. The codec modules are expected to be used for various applications such as customer premises multiplexers for digital leased lines, digital mobile radio, and stored-and-forward-message systems (voice-mail systems) 相似文献
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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. 相似文献
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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. 相似文献
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Low bit-rate speech coders for multimedia communication 总被引:10,自引:0,他引:10
The International Telecommunications Union (ITU) has standardized three speech coders which are applicable to low-bit-rate multimedia communications. ITU Rec. G.729 8 kb/s CS-ACELP has a 15 ms algorithmic codec delay and provides network-quality speech. It was originally designed for wireless applications, but is applicable to multimedia communications as well. Annex A of Rec. G.729 is a reduced-complexity version of the CS-ACELP coder. It was designed explicitly for simultaneous voice and data applications that are prevalent in low-bit-rate multimedia communications. These two coders use the same bitstream format and can interoperate. The ITU Rec. G.723.1 6.3 and 5.3 kb/s speech coder for multimedia communications was designed originally for low-bit-rate videophones. Its frame size of 30 ms and one-way algorithmic codec delay of 37.5 ms allow for a further reduction in bit rate compared to the G.729 coder. In applications where low delay is important, the delay of G.723.1 may be too large. However, if the delay is acceptable, G.723.1 provides a lower-complexity alternative to G.729 at the expense of a slight degradation in quality. This article describes the attributes of speech coders such as bit rate, complexity, delay, and quality. Then it discusses the basic concepts of the three new ITU coders by comparing their specific attributes. The second part of this article describes the standardization process for each of these coders 相似文献
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Advances in speech and audio compression 总被引:4,自引:0,他引:4
Gersho A. 《Proceedings of the IEEE. Institute of Electrical and Electronics Engineers》1994,82(6):900-918
Speech and audio compression has advanced rapidly in recent years spurred on by cost-effective digital technology and diverse commercial applications. Recent activity in speech compression is dominated by research and development of a family of techniques commonly described as code-excited linear prediction (CELP) coding. These algorithms exploit models of speech production and auditory perception and offer a quality versus bit rate tradeoff that significantly exceeds most prior compression techniques for rates in the range of 4 to 16 kb/s. Techniques have also been emerging in recent years that offer enhanced quality in the neighborhood of 2.4 kb/s over traditional vocoder methods. Wideband audio compression is generally aimed at a quality that is nearly indistinguishable from consumer compact-disc audio. Subband and transform coding methods combined with sophisticated perceptual coding techniques dominate in this arena with nearly transparent quality achieved at bit rates in the neighborhood of 128 kb/s per channel 相似文献
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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 相似文献
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为有效解决现有单一模型编码器无法在中低速率对语音和音频信号进行高质量通用编码的问题,本文借助语音与音频信号的谐波特性,建立了一种对语音和音频信号统一编码的方法。首先,本文利用经验模态分解(Empirical Mode Decomposition, EMD)提取输入信号的谐波成分;其次,利用感知匹配追踪算法,并结合正弦参数建模对谐波成分进行参数提取与量化;第三,对于量化谐波后的残差进行抖动格型矢量量化,以提升重建音频的主观听觉质量,并最终实现一套包含24kbps和32kbps码率的宽带语音与音频通用编码器;最后,对所提算法进行了客观PESQ/PEAQ和主观A/B测试,并与ITU-T G.722.1和G.722.2编码器进行了比较,实验结果表明,所提编码器对语音和音频信号的编码质量均优于参考编码器。 相似文献
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1998年 9月,ITU-T在 G.729的基础上制定了 G.729D。 G.729D是 G.729的低速率扩展标准,当速率为 6.4kb/ S时仍能保持很好的话音质量。文章介绍了 G.729D基本原理,对G.729D与G.729进行了比较和分析,并给出了主观测试结果。 相似文献
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Entropy coding principles are applied to the 16 kbit/s ITU G.728 speech codec. It is shown that the average bit rate can be reduced to 14.5 kbit/s without a significant increase in the codec complexity. In very low bit rate audiovisual communication applications such as the videophone, the saved bits can be used to improve the output video quality 相似文献
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G.719: The First ITU-T Standard for High-Quality Conversational Fullband Audio Coding 总被引:1,自引:0,他引:1
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. 相似文献