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单击此处编辑母版标题样式,单击此处编辑母版文本样式,第二级,*,Technical English,For Information Science and Electronic Engineering,0,Unit 10,Digital Audio Compression,1,Part I,MPEG Audio Layer 3,2,New Words,advent,到来，来临,sampler,样品,album,照相簿，歌曲集,amplitude,幅度，广阔,precision,精度，精确的,stereo,立体声,megabyte,兆字节,modem,调制解调器,imperative,势在必行的,psycho-acoustic,心理声学的,pitch,音调，程度,threshold,门限，阈值,mono,单一，单声道,bitrate,比特率,Huffman encoding Huffman,编码,bitcode,比特字，比特码,underestimate,低估,minority,少数,mainstream,主流,genre,流派，类型,3,New Words,multitude,多数，大众,a multitude of ,种种的，众多的,unsigned band,未签约的乐队,promote,推销，促进,rip,撕，拉，劈,royalty,版税，庄严，王权,clout,权力，影响，力量,spell,拼写，迷住,advocate,提倡者，拥护者,stem,阻止，堵住,rival,对手,lax,宽松的，不严格的,4,1,With the advent of the Internet, there is a desire for more and more information to be transmitted across phone lines.,Audio information is one form that is increasingly downloaded, be it a sampler for a bands album, a radio program, or sound as part of a video.,1,As bandwidth in a telephone wire is limited, this has led to a need for information (including audio) to be compressed.,音频信息是一种愈来愈多被下载的（多媒体）形式，无论是乐队的唱片选曲，无线电节目，还是视频伴音。,5,2,The traditional method of storing digital audio, used in CDs and digital TV, samples the amplitude of the sound a set number of times per second, and records this.,2,用于,CD,和数字电视中存储数字音频的传统方法是每秒抽取并记录一定次数的声音幅度值。,6,2,The precision of the amplitude is determined by the number of bits used to store the amplitude. So the bandwidth (or memory) consumed by the audio signal is dependent on three factors:,the number of samples taken per second (Frequency), the number of bits used to store the amplitude (Bit Depth) and the length of the signal (Time).,每秒钟的采样次数（频率），用于存储幅度的比特位数（比特深度）以及信号的长度（时间）,7,2,When we know these three things, the memory used becomes simple to calculate:,Memory = Frequency,Bit Depth,Time.,Additionally,if the signal is in stereo, then this must be doubled as two signals are in fact used,.,如果信号是立体声的，内存就乘以二，因为立体声实际上用了两个信号,8,3,This equation can be used to demonstrate why transmitting high-quality audio across the Internet requires compression. CD audio uses 16-bit stereo sampled at 44,100 Hz. This means that one minute of CD audio uses 44,100,16,60,2 = 84,672,000 bits, or slightly over 10 megabytes.,9,3,A standard 56 kbps modem would take 84,672,000/57344 = 1477 seconds or about 25 minutes!,25 minutes is a long time to wait for one minute of audio, so an alternative is imperative,. That alternative was MPEG Audio Layer 3, or MP3.,等待,1,分钟的音频要,25,分钟这么长的时间，因此必须有另一个选择,10,4 The codec,The human ear can only hear a limited range of frequencies. The codec therefore removes all sounds outside this range, as they will not be heard anyway.,11,5,A psycho-acoustic model is then applied to the sound.,If a high-pitched sound is played, then the decibel threshold for sounds of lower frequencies to be made audible is increased.,3,The psycho-acoustic model removes any and all sounds that will be masked in this way.,在播放音调高的声音时，要提高使低频声能被听到的临界分贝数（分贝阈值）。,12,6,The next stage is Joint Stereo.,The human brain is unable to place the directions of sounds at low frequencies, so sounds below this threshold are encoded in mono,. If some sections of the signal are still above the required bitrate, then the quality of these sections will be decreased. Finally, Huffman encoding is applied.,人脑无法估计低频声音的方向，所以这个阈值之下的声音都用单声道编码,13,6,This replaces all bitcodes with unique bitcodes of varying length according to the frequency of the pattern occurrence.,4,For instance, the most commonly occurred bit pattern would be encoded as 01, while the next common would be 010 and the next would be 011, and so on.,将所有比特码字根据其出现的频率换成独特的变长比特码。,14,7 Social and economic effects,The social effects of MP3 cannot be underestimated. It is allowing new, unsigned bands to distribute free music over the Internet.,Those with minority tastes can obtain experimental or alternative music far more easily than previously, as mainstream record shops do not tend to stock these genres of music.,5,具有非主流口味（喜好）的人获取实验性或不同寻常的音乐比以往容易得多，主流唱片店一般不经销这些流派的音乐。,15,7,Portable hardware MP3 players may now be purchased at low prices, and the price is still falling. Sites such as MP3 distribute free MP3s from a multitude of unsigned bands. Others sell albums from unsigned bands, with a free MP3 track from some albums, allowing the consumer to sample the bands music before purchase.,另外一些网站销售未签约乐队的专辑，用一些专辑上的免费,MP3,音乐让消费者在购买之前进行试听,16,7,Some mainstream acts have also taken to distributing some free tracks in order to promote albums. There is, unfortunately, a darker side to this revolution.,Illegal sites distribute tracks illegally ripped from the albums of established artists, who subsequently lose out on royalties.,6,非法网站发行从已成名艺术家的唱片中非法窃取的音乐，这些艺术家因而失去版税。,17,8,The economic implications of MP3 are closely tied to its social effects.,The major record companies are essentially running scared of the effect this is likely to have.,7,主要的唱片公司差不多因为这可能产生的社会影响而惊恐万状。,18,8,They refuse to sell MP3 albums, except in those cases where the artist has enough clout to force them.,They are hurriedly trying to establish a standard for a secure music format that cannot be used on more than one machine.,8,他们急欲建立一种“安全”音乐格式标准，使之不能用于一台以上的机器（仅能在一台机器上播放）。,19,8,Microsoft recently attempted, and failed, to do just that.,Their format, WMT4, was cracked within less than 24 hours of its release,. In fact, the same techniques used to crack WMT4 can be applied to any music format, no matter how secure.,People believe that MP3 may spell the end for the traditional record company,.,它们的格式,WMT4,，在发布之后不到,24,小时就被破解了,人们相信，,MP3,可能是传统唱片公司的末日,20,8,Most, however, foresee the record companies realizing that they cannot win this particular war, and beginning to distribute MP3 albums themselves.,In fact, in this age MP3 is probably not that much less secure than the CD itself,.,实际上，在这个时代，,MP3,在安全性上大概并不比,CD,差很多,21,9 Conclusion,In conclusion, MP3 can compress digital audio by a high factor. This makes it ideal for the distribution of audio across the Internet. It has found popularity among unsigned and experimental bands, as well as some established artists who are advocates of the technology.,22,9,Although the record companies refuse to back it, there is little they can do to stem the tide,. They have found themselves very much in the position of King Canute, although this will undoubtedly change in the long term.,King Canute was a British monarch who believed that he could force the tide back. Funnily enough, he failed.,尽管唱片公司拒绝对其进行支持，但是它们也没办法阻止这个潮流,23,10,The MP3 file format is an extremely efficient compression standard, which has already seen off challenges from WMT4 and other rival standards.,9,Due to its extremely lax licensing terms, MP3 seems unlikely to lose its popularity.,MP3,文件格式是一种效率极高的压缩标准，它已经赢得（告别）了,WMT4,和其他对手发起的挑战。,24,Part II,Digital Audio Compression Standard AC3,25,New Words,coordination,协调,voluntary,自发的，自愿的,executive,执行的，执行者,annex,附件,multiplex,多样的，多路复用,herein,在此，如此,motivation,动机，推动力,algorithm,算法,dynamic range,动态范围,fractional,部分的，分数的,woofer,低音喇叭,transponder,应答器，转发器,demodulate,解调,terrestrial,地面的，地球上的,consistent,一致的,normative,规范的，标准的,syntax,句法,informative,提供信息的,overlap,重叠,decimate,抽取,26,New Words,coefficient,系数,exponent,指数,mantissa,尾数,envelope,包络,allocation,分配，指定,synchronize,使同步，同时发生,resolution,分辨率,parameter,参数,inverse,反转的，逆,unpack,解开,rematrix,重新进行矩阵变换,dematrix,求矩阵反变换,27,1 Foreword,The United States Advanced Television Systems Committee (ATSC),was formed by the member organizations of the Joint Committee on InterSociety Coordination (JCIC),recognizing that the prompt, efficient and effective development of a coordinated set of national standards is essential to the future development of domestic television services,.,认识到迅速有效地制定一套相互协调的国家标准对于美国电视业今后的发展至关重要,28,1,The United States Advanced Television Systems Committee, December 20, 1995,The JCIC is presently composed of: the Electronic Industries Association (EIA), the Institute of Electrical and Electronic Engineers (IEEE), the National Association of Broadcasters (NAB), the National Cable Television Association (NCTA), and the Society of Motion Picture and Television Engineers (SMPTE).,29,2,One of the activities of the ATSC is,exploring the need for,and, where appropriate,coordinating the development of,voluntary national technical standards for Advanced Television Systems (ATV).,1,ATSC,（先进电视系统委员会）的工作之一就是寻求对,ATV,（先进电视系统）的非强制性国家技术标准的需要并在适当的情况下协调这种开发工作。,30,2,The ATSC Executive Committee assigned,the work of documenting the U.S. ATV standard,to a number of specialist groups working under the Technology Group on Distribution (T3). The Audio Specialist Group (T3/S7) was charged with documenting the ATV audio standard.,起草美国,ATV,标准的工作,31,3,This document was prepared initially by the Audio Specialist Group,as part of its efforts to document the United States Advanced Television broadcast standard,. It was approved by the Technology Group on Distribution on September 26, 1994, and by the full ATSC Membership as an ATSC Standard on November 10, 1994.,作为制定美国,ATV,广播标准的一部分,32,3,Annex A, “AC-3 Elementary Streams in an MPEG-2 Multiplex,” was approved by the Technology Group on Distribution on February 23, 1995, and by the full ATSC Membership on April 12, 1995. Annex B, “AC-3 Data Stream in IEC958 Interface,” and Annex C, “AC-3 Karaoke Mode,” were approved by the Technology Group on Distribution on October 24, 1995 and by the full ATSC Membership on December 20, 1995.,33,3,ATSC Standard A/53,Digital Television Standard for HDTV Transmission,references this document and describes how the audio coding algorithm described herein is applied in the U.S. ATV standard,.,引用了本文件并叙述了本文件所述的音频编码算法应如何用于美国,ATV,标准,34,4,At the time of release of this document,the system description contained herein had not been verified by the transmission of signals from independently developed encoders to separately developed decoders,.,本文所述系统尚未经过由独立开发的编码器到分别开发的解码器之间进行信号传输的验证,35,5 Motivation,In order to more efficiently broadcast or record audio signals,the amount of information required to represent the audio signals may be reduced,.,需要减少用于表示音频的信息量,36,5,In the case of digital audio signals, the amount of digital information needed to accurately reproduce the original pulse code modulation (PCM) samples may be reduced by applying a digital compression algorithm, resulting in a digitally compressed representation of the original signal.,2,对于数字音频信号，用于精确重建原始脉冲编码调制样本所需要的数字信息量可以通过应用数字压缩算法来减少，由此产生原信号的数字压缩形式。,37,5,(,The term compression used in this context means the compression of the amount of digital information which must be stored or recorded, and not the compression of dynamic range of the audio signal,.),压缩一词在这里是指压缩必须存储或记录的数字信息量，而不是压缩音频信号的动态范围,38,5,The goal of the digital compression algorithm is to produce a digital representation of an audio signal which,when decoded and reproduced,sounds the same as the original signal, while using a minimum of digital information (bitrate) for the compressed (or encoded) representation,.,在解码和回放时，给出与原始信号相同的听觉效果，而其压缩（或编码）形式使用最少的数字信息量（比特率），,39,5,The AC-3 digital compression algorithm specified in this document,can encode from 1 to 5.1 channels of source audio from a PCM representation into a serial bit stream at data rates ranging from 32 kbps to 640 kbps,. The 0.1 channel refers to a fractional bandwidth channel intended to convey only low frequency (subwoofer) signals.,可以将,1,至声道的,PCM,形式音频源信号编码为,32kbps,至,640bps,的串行比特流,40,6,A typical application of the algorithm is shown in Figure 10.1. In this example, a 5.1 channel audio program is converted from a PCM representation requiring more than 5 Mbps (6 channels 48 kHz 18 bits = 5.184 Mbps) into a 384 kbps serial bit stream by the AC-3 encoder.,Satellite transmission equipment converts this bit stream to an RF transmission which is directed to a satellite transponder,.,卫星发射设备将比特流转换成射频，送到卫星收发器,41,6,Figure 10.1 Example application of AC-3 to satellite audio transmission,42,6,The amount of bandwidth and power required by the transmission,has been reduced by more than a factor of 13,by the AC-3 digital compression. The signal received from the satellite is demodulated back into the 384 kbps serial bit stream, and decoded by the AC-3 decoder. The result is the original 5.1 channel audio program.,下降了,13,倍以上,43,7,Digital compression of audio is useful,wherever there is an economic benefit to be obtained,by reducing the amount of digital information required to represent the audio. Typical applications are in satellite or terrestrial audio broadcasting,delivery of audio over metallic or optical cables, or storage of audio on magnetic, optical, semiconductor, or other storage media,.,只要,具有经济效益,经过电缆或光缆传输音频，或在磁、光、半导体或其它存储介质上存储音频信号,44,8 Encoding,The AC-3 encoder accepts PCM audio and,produces an encoded bit stream,consistent with this standard,. The specifics of the audio encoding process are not normative requirements of this standard.,产生符合本标准的比特流,45,8,Nevertheless, the encoder must produce a bit stream matching the syntax described in Section 5, which, when decoded according to Sections 6 and 7,produces audio of sufficient quality for the intended application,. Section 8 contains informative information on the encoding process. The encoding process is briefly described below.,对于具体应用产生音质足够好的音频信号,46,9,The AC-3 algorithm achieves high coding gain (the ratio of the input bit-rate to the output bit-rate),by coarsely quantizing a frequency domain representation of the audio signal,. A block diagram of this process is shown in Figure 10.2.,通过对音频信号的频域形式进行粗量化,47,9,Analysis Filter Bank,Mantissa Quantization,AC-3 Frame Formatting,Bit Allocation,Spectral Envelop Encoding,PCM Time Samples,Exponents,Bit Allocation Information,Mantissas,Quantized Mantissas,Encoded Spectral Envelope,Encoded AC-3 Bit-Stream,Figure 10.2 The AC-3 encoder,48,9,The first step in the encoding process is to,transform the representation of audio from a sequence of PCM time samples into a sequence of blocks of frequency coefficients,. This is done in the analysis filter bank. Overlapping blocks of 512 time samples are,multiplied by a time window,and transformed into the frequency domain.,将音频信号从一系列,PCM,时域样本形式转换为一系列频率系数的块,被乘以一个时间窗函数,49,9,Due to the overlapping blocks, each PCM input sample is represented in two sequential transformed blocks. The frequency domain representation may then be decimated by a factor of two so that each block contains 256 frequency coefficients.,The individual frequency coefficients are represented in binary exponential notation as a binary exponent and a mantissa.,3,各频域系数以二进制指数形式表示成二进制指数和二进制尾数。,50,9,The set of exponents is encoded into,a coarse representation of the signal spectrum,which is referred to as the spectral envelope. This spectral envelope is used by the core bit allocation routine which determines,how many bits to use to encode each individual mantissa,.,信号频谱的粗放表示形式,对每一尾数编码时所需要的比特数,51,9,The spectral envelope and the coarsely quantized mantissas for 6 audio blocks (1536 audio samples) are formatted into an AC-3 frame. The AC-3 bit stream is a sequence of AC-3 frames.,52,10,The actual AC-3 encoder is more complex than indicated in Figure 10.2. The following functions not shown above are also included:,A frame header is attached which contains information (bit- rate, sample rate, number of encoded channels, etc.) required to synchronize to and decode the encoded bit stream.,4,在每一帧前面加上头部，其中包含与编码比特流实现同步并将它解码所需的信息，如比特率、采样频率、编码声道数等。,53,10,Error detection codes are inserted in order to allow the decoder to verify that a received frame of data is error free.,The analysis filter bank spectral resolution may be dynamically altered so as to better match the time/frequency characteristic of each audio block.,5,The spectral envelope may be encoded with variable time/frequency resolution.,分析滤波器组的频谱分辨率可以动态地改变以便更好地匹配每一音频（信号）块的时频特性。,54,10,A more complex bit allocation may be performed, and parameters of the core bit allocation routine modified so as to produce a more optimum bit allocation,.,The channels may be coupled together at high frequencies in order to,achieve higher coding gain for operation at lower bit-rates,.,可以进行更复杂的比特分配，可以修改核心比特分配例程的参数，以生成更优的比特分配,在低比特率工作时实现更高的编码增益,55,10,In the two-channel mode a rematrixing process may be selectively performed in order to provide additional coding gain, and to allow improved results to be obtained in the event that the two-channel signal is decoded with a matrix surround decoder.,6,在双声道模式中，可以选择执行一个重新进行矩阵变换的过程以便提供附加的编码增益，并在使用矩阵环绕声解码器解码双声道信号时给出改进的效果。,56,11 Decoding,The decoding process is basically the inverse of the encoding process. The decoder, shown in Figure 10.3, must,synchronize to the encoded bit stream, check for errors, and de-format the various types of data such as the encoded spectral envelope and the quantized mantissas,.,与编码的比特流同步，检查有无误码，将各种数据如编码的频谱包络和量化的尾数解格式,57,11,Synthesis Filter Bank,Mantissa De-quantization,AC-3 Frame Synchronization, Error Detection and Frame De-formatting,Bit Allocation,Spectral Envelop Decoding,PCM Time Samples,Exponents,Bit Allocation Information,Mantissas,Quantized Mantissas,Encoded Spectral Envelope,Encoded AC-3 Bit-Stream,Figure 10.3 The AC-3 decoder,58,11,The bit allocation routine is run and,the results used to,unpack and de-quantize the mantissas. The spectral envelope is decoded to produce the exponents. The exponents and mantissas are transformed back into the time domain to produce the decoded PCM time samples.,省略“,are”,59,12,The actual AC-3 decoder is more complex than indicated in Figure 10.3. The following functions not shown above are included:,Error concealment or muting may be applied,in case a data error is detected,.,Channels which have had their high-frequency content coupled together must be decoupled.,检测到误码时,60,12,Dematrixing must be applied (in the 2-channel mode) whenever the channels have been rematrixed.,The synthesis filter bank resolution must be dynamically altered,in the same manner as the encoder analysis filter bank had been during the encoding process,.,以编码时分析滤波器组所用的同样方式,61,
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