iLBC语音编解码器核心技术的研究

重点讨论了iLBC编解码器独立于帧的长期预测。独立于帧的长期预测是用来在编码语音没有遭受与传输丢失相关的多帧语音退化情况下,开发斜度标记相关的办法。然后介绍了iLBC,G.729A和G.723.1编解码器的平均主观得分MOS,并用信号为例说明基于独立于帧的长期预测编解码器和CELP编解码器之间的不同,最后用语音重构的例子说明二者语音质量间的差别。     

自适应计算减少QCELP功率

CELP(Qualcomm Code Excited Linear Predictive Coding-Qualcomm码受激线性预示编码)话音压缩算法是TIAIS-95北美宽带CDMA数字蜂窝电话标准的TIAIS-96话音编解码所选的声码器。一个话音编解码器等于一个编码器和解码器对。依此标准进行设计对于便携系统设计人员来讲是个问题,这是因为用通常DSP方法需要异常的功耗。QCELP是一种向量编码器型话音编解码,在话音编解码分类中被称为CELP(码受激线性预示)编码话音压缩算法。CELP编解码采用话音编码的合成分析方法。编码器的任务是确定描述话音音频段的小的参量组,话音音频段…     

码本激励线性预测类和多带激励类话音编码

本文分析比较了混合编码类的CELP类（CELP、VSELP）及MBE类（IMBE、AMBE）编码。给出了一种新型优质的编码器─—AMBE－1000编码器在实际工程中的应用。     

同时实现LD—CELP和EC的研究

本文提出了一种将回波抵消器（ＥＣ）作为ＬＤ－ＣＥＬＰ编解码器的部分工作的方法，将ＥＣ功能融入ＬＤ－ＣＥＬＰ编解码器之中，文中讨论了用两片ＴＭＳ３２０Ｃ３０实时实现ＬＤ－ＣＥＬＰ与ＥＣ于一个系统中的可能性。     

低比特速率话音编码技术

在数字汽车电话中,要有效地利用频率,必须要有高效话音编码技术.汽车电话用的话音编解码器要求作到:在低比特速率下具有用作公众通信的高质量话音,通过无线线路传输时误码率低,抗环境噪声特性优良.现在,世界各地采用 CELP 为中心的混合编码方式达到10kbit/s 左右的速率。并以此进行标准化.此外,为了实现系统的大容量化,正在开展半速率化的研究开发。     

文摘

文摘ＮＥＣ公司开发出数字便携电话用的３Ｖ工作的编解码器－μＰＤ９９３０ＮＥＣ公司开发的线性编解码器μＰＤ９９３０，是数字便携电话机用的编解码器，工作电压为２．７～３．６Ｖ、消耗电流为７ｍＡ。该编解码器具有音频接Ｄ、单音发生、ＰＬＬ、切断电流、ＧＳＭ系...     

一种使用分带——整带复用技术的CELP宽带语音编码算法

介绍了一种新的宽带语音编码算法，由于使用了分带-整带复用技术，克服了两种传CELP宽带语音编码技术（分带技术和整带技术）的不足。该算法的激励码本采用了分带CELP的多带码本，由此克服了整带CELP码本搜索复杂度巨大的缺点；算法中的线性预测以及合成语音误差最小化部分都是采用整带CELP的方案，由此克服了分带CELP频带重叠影响语音质量的短处。     

MPEG音频编解码器:从mp3到xHE-AAC

音频编解码器是现代媒体系统的基础核心之一。没有音频编解码器,就不会有现在的数字广播、流媒体服务及音乐发行。首个同时也仍是最主流的MPEG音频编解码器是1998年面市的mp3。此后,Fraunhofer IIS和其他ISO-MPEG成员参与者开发并制造了多种音频编解码器。每个MPEG音频编解码器已经或将会改变我们消费媒体的方式。本文介绍了MPEG音频编解码器及其应用,     

AC-3和MPEG-2编解码器的研究

介绍了ＡＣ－３算法的编程及ＭＰＥＧ－２算法的具体实现，介绍了编解码器的硬件，软件设计考虑和研制过程及效果，并成功的完成了实验样机。     

MPEG－2编解码器技术

ＭＰＥＧ－２作为当前数字视频传输的关键技术，广泛应用于有线电视传输、家庭影院、多媒体服务等领域，重点介绍ＭＰＥＧ－２编解码器在广播电视骨干传输中的应用。     

Implementation of a LD-CELP Codec with Echo Canceller Functions

ＩｍｐｌｅｍｅｎｔａｔｉｏｎｏｆａＬＤ－ＣＥＬＰＣｏｄｅｃｗｉｔｈＥｃｈｏＣａｎｃｅｌｌｅｒＦｕｎｃｔｉｏｎｓＴｉａｎＷｅｎｓｈｕｎ（ＮａｔｉｏｎａｌＵｎｉｖｅｒｉｓｔｙｏｆＳｍｇａｐｏｒｅ，０５１１，Ｓｉｎｇａｐｏｒｅ）ＮｉＷｅｉｚｈｅｎ（Ｄｅｐａ...     

8kbit/s低时延CELP语音编码的研究

低时延码激励线性预测(LD-CELP)语音编码能够在较低速率把高质量与低时延结合起来。它可以用建立在全搜索算法基础之上的自相关搜索算法来降低计算量,用抗误码码本来提高抗误码能力。此外,对8kbit/s LD-CELP系统进行了模拟。     

15.2kb／sLD—CELP语音编码算法及实时实现

本文以Ｇ．７２８语音编码标准为基础,提出一个１５．２ｋｂ／ｓ ＬＤ－ＣＥＬＰ语音编码算法。通过对滤形码书进行重新设计,降低了算法复杂性,同时保持高质合成语音。最后,用双片ＴＭＳ３２０Ｃ３１高速ＤＳＰ系统全双工实时实现了该算法。     

Codage audio haute qualité (20 Hz-15 kHz) en sous bandes à débit réduit (64 kbit/s) et à faible retard (5 ms)

In this paper, we present a new method for high quality audio coding at low delay and low bit rate for telecommunications applications such as audioconfe-rence or videoconference. The developped coder is adapted to code generic audio signals at a bit rate of 64 kbit/s with a delay close to 5 ms in the 20-15000 Hz bandwidth. The method is based on speech coding as well as audio coding concepts. The coder combines subband decomposition of the input signal and LD-CELP techniques. We introduce in this structure of coding a psychoacoustic model which allows to allocate an optimal bit rate on each subband according to perceptual properties of the human hearing. In order to satisfy the bit rate requirement of the psychoacoustic model and to reduce the complexity of such a coding algorithm, we suggested a new method of vector quantization based on lattice quantization. This method allows to quantify the residual signal in the LD-CELP coder and avoid the complexity of the full search. Objective and subjective tests have been made on a test set of audio signals which is a critical sub-set used by ISO. Formal tests showed that the quality of the proposed coder is comparable to the best implementation of the MPEG-1, Layer II, but our solution has the advantage of reaching a very low delay (5 ms).     

A packet-based CAPDM speech coder for PCN applications

In this paper, we present a median-rate speech coder, the controlled adaptive prediction delta modulation coder (CAPDM), which operates at 16 kb/s with good speech quality and low algorithm complexity. The coder is dedicated to personal communication network (PCN) applications and transmits speech samples on the basis of packets. It combines the features of a one-step looking forward decision, syllabic companding, instantaneous companding, and adaptive prediction. In addition to the use of a short-term prediction filter, CAPDM also exploits the pitch property to predict speech waveform explicitly. With the aid of a pitch prediction filter, the performance of a CAPDM codec improves about 3 dB in segmental signal-to-noise ratio (SEGSNR). The average SEGSNR of CAPDM.FF is about 21 dB, which is 7 dB over traditional CVSD at 16 kb/s. We also utilize an adaptive postfilter (APF) to enhance the perceptual quality of the decoded speech. The mean opinion score (MOS) listening test of CAPDM.FF with APF shows that its average score achieves 4.19, which is as good as G.728 16-kb/s LD-CELP and is comparable with CCITT G.721 32-kb/s ADPCM. The complexity of CAPDM.FF is evaluated to be 8 MIPS, which is much lower than that of LD-CELP and could be further reduced by adopting a smaller correlation window for pitch detection. To solve the problem of packet loss, we developed a packet-based waveform substitution method by reinitializing the codec parameters at the beginning of each packet. The simulation results show that CAPDM.FF could tolerate 5% of packet loss and still keep an SEGSNR at 10 dB and an MOS at about 3.0     

用增益精确值和归一化波形码书改进G.728

分析研究了增益、波形乘积码书结构的缺陷，设计了归一化波形码书和精确表示增益的LD-CELP方案。采用自适应预测和自适应量化对增益的精确值进行量化，在3bit和4bit自适应量化时比G．728固定量化增益分别提高0．5dB和6dB。采用4bit自适应量化和64波形码书比G.728SNR提高约1dB。将G．728综合滤波器由50阶减少到30阶，信噪比不变而算法复杂性降低约20%。     

Impact of network routing on the end-to-end transmission quality ofcellular and PCS connections

Long-distance and especially international calls involve an increasing number of multilink circuits of cellular, personal communications, mobile satellite, and public switched telephone network (PSTN)-type connections incorporating a variety of speech coding devices. In particular, the rapid growth of cellular communications and the impeding emergence of personal communication systems (PCSs) has highlighted the need to characterize the quality of end-to-end connections when cellular and PCS terminals are connected over the PSTN. At the same time, the nonlinear nature of low-rate parametric speech coding has rendered unsuitable known analytical methods for estimating end-to-end quality of interconnected networks. Instead, quantification of the transmission performance requires direct subjective evaluation of the pertinent connections of interest. In this paper, the quality of North American, Japanese, and European digital cellular and PCS terminals is quantified when these are interconnected using 16-kb/s low-delay code-excited linear prediction (LD-CELP) and 32-kb/s adaptive differential pulse code modulation (ADPCM) facilities. From these assessments, it can be concluded that digital cellular networks may meet end-to-end transmission planning criteria when interconnected with the switched network as long as calls are terminated to a wired terminal     

低延迟话音编码算法的增益优化与混合窗设计

G.728标准将增益滤波器的偏移不适当地处理成常数32dB,相当于将20～30ms内短时平稳的语音信号不适当地按整个时间轴上的平稳信号对待。对增益滤波器的改进必然涉及优化混合窗。该文提出一种估计平均分段信噪比的方法用来设计混合窗参数,并导出激励增益的精确表示。以此为基础,将Jayant的自适应量化与G.728的增益预测结合起来,提出一种适合于设计低时延码激励线性预测(LD-CELP)的增益码书的方案。用此方法,激励的精确值经自适应预测和量化后,3bit自适应量化和4bit自适应量化分别比G.728的固定量化有0.5dB和6dB的信噪比的改善。当采用4bit自适应量化和64码字的波形码书,比G.728复杂性降低20%,同时信噪比提高1dB。     

Tools for real-time signal-processing research

The use of digital signal processing (DSP) devices for real-time communication applications is discussed. The authors comment on distinguishing aspects of DSP architecture, describing not so much individual processors as those features common to DSPs and distinct from modern general-purpose processors. They describe three DSP32xx-based machines that support DSP algorithm implementation: SURF-board, HoBo, and DSP3. They also described rtpi, a source-code debugger for workstations and for the AT&T DSP32C signal-processor integrated circuit, and dspx, a collection of subroutines and host programs that provides an execution environment for DSPs akin to the UNIX environment. These tools facilitate the transfer of algorithms from mainframes or workstations to DSP hardware. Included are case studies of two real-time implementations: the low-delay CELP (LD-CELP) speech coder and the decoder side of the perceptual audio coder (PAC), an algorithm that compresses CD-quality audio into a 128-kb/s stream without perceptible distortion