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本文从MPEG-4标准的主要特点、MPEG-4 标准的技术体系、MPEG-4音频对象(AO)的编码、 MPEG-4视频对象(VO)的编码、MPEG-4交互式AV场景描述、MPEG-4的多媒体应用等方面,分析与概述了 MPEG-4标准的技术内容以及MPEG-4标准的应用情况。 相似文献
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本文从MPEG-4标准的主要特点、MPEG-4标准的技术体系、MPEG-4音频对象(AO)的编码、MPEG-4视频对象(VO)的编码、MPEG-4交互式AV场景描述、MPEG-4的多媒体应用等方面,分析与概述了MPEG—4标准的技术内容以及MPEG-4标准的应用情况。 相似文献
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本文在简要说明MPEG音频编码标准发展史和音频编码的基础之后,重点介绍了MPEG-4音频编码标准,包括:MPEG-4音频编码标准概要;MPEG-4的“型”与“层”;自然语言信号的编码;自然声音信号的编码;MPEG-4音频标准的特殊功能;合成语言信号的编码;合成声音信号的编码;音频景象的描述;现在和将来的应用。最后展望了音频编码方法未来的发展趋势。 相似文献
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曾制定出MPEG-1和MPEG-2视音频压缩标准的运动图像专家组(MPEG)目前正在发展最新的MPEG-4标准,其目标是提供未来的交互式多媒体应用.MPEG-4将制定出与以往不同的、具有高度灵活性和可扩展性的未来新一代国际标准.在音频标准的制定方面,比较以前的音频编码标准,MPEG-4增加了许多新的关于合成内容及场景描述等领域的工作,增加了诸如可分级性、音调变化、可编辑性及延迟等新功能.MPEG-4将以前发展良好但相互分离的高质量音频编码、计算机音乐及合成语音等第一次合成并在一起,在诸多领域内给予高度的灵活性. 相似文献
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在话音和音频编码中,代表数字话音或音频信号的位数尽可能地少,但要保持一个合理的可识别质量水平。这主要通过从信号中去除冗余和非相干信息来实现。本文给出了下一代网络中话音编码标准的分类方法,并详细分析了波形编解码、参量编解码和混合编解码标准的目标与原理,最后研究了话音编码标准的应用环境。 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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Iwadare M. Sugiyama A. Hazu F. Hirano A. Nishitani T. 《Selected Areas in Communications, IEEE Journal on》1992,10(1):138-144
A Hi-Fi audio codec with an improved adaptive transform coding (ATC) algorithm is presented using digital signal processors (DSPs). An audio signal with a 20 kHz bandwidth sampled at 48 kHz is coded at a rate of 128 kb/s. The algorithm utilizes adaptive block size selection, which is effective for preecho suppression. A modified discrete cosine transform (MDCT) with a simple window set is employed to reduce block boundary noise without decreasing the performance of transform coding. In addition, a fast MDCT calculation algorithm, based on a fast Fourier transform, is adopted. Weighted bit allocation is employed to quantize the transformed coefficients. The codec was realized by a multiprocessor system composed of newly developed DSP boards. Subjective tests with the codec show that the coding quality is comparable to that of compact disc signals 相似文献
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利用波形编码和参数编码的优点,设计了一种移动音频编码技术.该技术采用子带滤波、预测编码、变换编码、频带扩展和参数立体声编码技术,以较低的复杂度实现了宽带音频信号的高效率编码. 相似文献
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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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The key factors deterring the use of visual telephony are identified, and an overview of a typical system architecture is given. The video signal formats and video and audio coding algorithms used are described. Video codec implementation is considered, and an implementation based on application-specific integrated circuits is presented. In particular, three key signal processing modules in the video codec are examined: a discrete cosine transform chip, a motion estimation chip, and a variable-length codec chip. Standardization activities in the video coding area are discussed 相似文献
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提出一种适用于无线移动通信的低码率音频压缩算法。该算法基于正弦模型,而且针对极低码率的应用做了修正,提高了重建音频的质量。这些修正包括:自适应变换的分析长度,用于匹配跟踪算法和参数量化的心理声学模型以及频域的无相位音频重建算法。主观测试表明,在0.5bit/抽样的码率要求下,重建信号达到并超过了调幅广播的音频质量。 相似文献