AC3（Audio Coding 3）是Dolby（R）杜比公司于1996年开发的一种先进的音频编码模式，它是DVD音频的标准，其编码完全是数字的，压缩比约为1：9~10，但解码后可得符合人耳声学模型（0~20000Hz）的音频。其编码一般由ffmpeg完成，生成文件为*.AC3或*.A52。
大家来传播AC3高品质音乐吧！！！！！
请相互转载，宣传！！！！！
AC3风暴
AC3简介：
杜比数字AC-3（Dolby Digital Audio Coding 3）是杜比Dolby（R）公司开发的新一代家庭影院多声道数字音频系统。杜比定向逻辑系统是一个模拟系统。它的四个声道是从编码后的两个声道分解出来的，因此难免有分离度不佳、信噪比不高，对环绕声缺乏立体感，并且环绕声的频带窄等缺点。AC（Audio Coding）指的是数字音频编码，它抛弃了模拟技术，采用的是全新的数字技术。
杜比数字AC-3提供的环绕声系统由五个全频域声道加一个超低音声道组成，所以被称作5.1个声道。五个声道包括前置的"左声道"、"中置声道"、"右声道"、后置的"左环绕声道"和"右环绕声道"。这些声道的频率范围均为全频域响应3-20000Hz。第六个声道也就是超低音声道包含了一些额外的低音信息，使得一些场景如爆炸、撞击声等的效果更好。由于这个声道的频率响应为3-120Hz，所以称".1"声道。
杜比数字AC-3是根据感觉来开发的编码系统多声道环绕声。它将每一种声音的频率根据人耳的听觉特性区分为许多窄小频段，在编码过程中再根据音响心理学的原理进行分析，保留有效的音频，删除多作的信号和各种噪声频率，使重现的声音更加纯净，分离度极高。
杜比数字AC-3系统可用前置的左、右音箱，中置音箱产生极有深度感和定位明确的音场，用两个后置或侧置的环绕音箱和超低音箱表现宽广壮阔的音场，而六个声道的信息在制作和还原过程中全部数字化，信息损失的很少。全频段的细节十分丰富，具有真正的立体声。
杜比数字AC-3具有很好的兼容性，它除了可执行自身的解码外，还可以为杜比定向逻辑解码服务。因此，目前已生产的杜比定向逻辑影视软件都可以使用杜比数字AC-3系统重现。由于杜比数字AC-3系统的编码非常灵活，所以它的格式很多。目前它已被美国采用作为高清晰电视（HDTV）音频系统，最新DVD机也包含杜比数字AC-3。因此杜比AC-3环绕声系统可能是极有发展前途的技术。
AC3的特性：
码率构成
[只可为整点码率]：（16 24 32 48 64 72 80 96 112 ）128 （160 192 224 256 320] [384 448 480 640]Kpbs
（16 24 32 48 64 72 80 96 112 ）：此段只适用于Voice Phone等低频低码率的编码，80，96，112Kpbs适合于单声道保真音频
128Kpbs：此段适合于不是过高要求的保真音频（比较适合于劲爆的音乐）和近于Mp3 44100Hz CBR 320kbps的音频（极不适合于纯色音乐，尤其是弦乐）
（160 192 224 256 320] 非常适合于双声道高品质音频的传播，且文件较小，不会产生听觉疲劳（远超出lame MP3中对HiFi的定义:160kbps VBR）,适合于收藏。
[384 448 480 640]：此段对于stereo来说有点奢侈，所以适合于HiFi级多声道听众享受。
频率特性：
这是AC3Filter的信息：
从中可以看出192kbps 48000Hz的stereo AC3文件音域为0~20906Hz，（160 192 224 256 320]都可达此效果
AC3支持的基本频率有8000 12000 16000 22050 24000 32000 44100 48000Hz
8000 12000 16000 22050 24000 32000Hz： 此段压制的音频的频率约为基频的一半减1000Hz（如32000Hz：极限频率=(32000/2)-1000=15000Hz）
44100Hz：（声波已足够平滑）极限频率为19500Hz，近于Mp3 44100Hz CBR 320kbps的音频
48000Hz：（声波非常平滑）极限频率为20906Hz，同时限制了方波的产生，再说高于20906Hz的声音是永远也不能被人耳听见的
声道特性：
AC3支持mono stereo ~5.1声道
AC3的优点
1. 建立了适合人耳的声学模型，不会产生听觉疲劳。
2. 文件小，网上下载的音质较好mp3,wma音乐，约为3~4MB，而AC3 2.0 HiFi的音频也只有4~5MB
3. 压制较为方便
4. 也可以存储内部文件信息
5. 应用降噪技术,解决了模拟音频播放中噪音过大的缺点（尤其是AC97芯片的声卡，其信噪比较小，噪音较大）。
6.
AC3 2.0 压制攻略
1. Media Coder压制：
缺点：无法写入标签（内部文件信息）
2．Foobar（或千千静听TTplayer）[有命令行编码器即可]
以Foobar为例：
首先将ffmpeg.exe等组件复制到foobar的文件夹中（在Media Coder的文件夹中的codecs文件夹中可以找到，将以Lib开头的dll文件都复制到foobar的文件夹中）
然后就可以进行转码了
缺点：速度较Media Coder慢一点
参数的设置：
示例参数 ：
HiFi stereo：
-i - -ab 256000 -aq 100 -ar 48000 -ac 2 -acodec ac3 %d
-i - -ab 224000 -aq 100 -ar 48000 -ac 2 -acodec ac3 %d
-i - -ab 192000 -aq 100 -ar 48000 -ac 2 -acodec ac3 %d
-i - -ab 160000 -aq 100 -ar 48000 -ac 2 -acodec ac3 %d [适用于Foobar或千千静听TTplayer]
-i %s -ab 160000 -aq 100 -ar 48000 -ac 2 -acodec ac3 %d [只适用于Foobar]
Mp3 44100Hz CBR 320kbps：-i - -ab 128000 -aq 100 -ar 441000 -ac 2 -acodec ac3 %d
Voice：-i - -ab 40000 -aq 100 -ar 12000 -ac 1 -acodec ac3 %d
Phone：-i - -ab 32000 -aq 100 -ar 8000 -ac 1 -acodec ac3 %d
参数设置如下：
AC3的播放工具
建议播放工具：Foobar或千千静听TTplayer。通过AC3 filter（可加载在winamp中）解码的AC3效果也很不错（建议配置Speakers：Dolby ProLogic II Sample format：PCM 24Bit）
AC3与其他音频格式（CDA，PCM，LPCM，DTS，APE，AAC，DAC（豪杰公司开发），AtrAc3，Flac，ALS，OGG，RAM，TXT）的对比
PCM，LPCM，DTS：音质完美，可文件巨大无比。LPCM是音乐DVD专用格式，压缩比约为1：1.5。DTS音效超过AC3，但压缩比只有1：2~3，不适合传播。[100~n%CD]
CDA：CD格式，类似于PCM的模拟音频文件（无法复制，可用软件提取）[100%CD]
AAC：音质较mp3好一些，文件也较小，但损失也较大[50~90%CD]
APE, Flac，ALS：网上开始盛行的收藏级音乐，都是以PCM为源码，进行压缩封装，所以压缩比不高，一般为1：1~3[80~98%CD]
AtrAc3：简称AA3，是sony公司为其便携音频设备开发的一种类似于AC3的音频格式，文件较AC3略大[80~95%CD]
DAC（豪杰公司开发）：以AC3为母本开发的音频，但达不到较好的效果，320kpbs时约与AC3 160kpbs差不多，且失真严重。继续努力，有望超过AC3。[40~90%CD]
OGG RAM：音质中等偏下的压缩模式。OGG在码率较小时，表现较好。[40~80%CD]
TXT：绝对无损的典范，以时间为横轴，以电平为纵轴，绝对无损的代表，一般电脑无法播放，文件大到境界。[100~n%CD]
AC3的“缺点”
AC3文件中标签位置与MP3和WMA位置不同，据鄙人所知，能识别AC3标签的播放器只有Foobar或千千静听（TTplayer）
AC3的标签在文件结尾，对于播放器来说学要预读，但字符几乎没有限制（可以算是个弱点吧！不大适合于便携式播放器）
WMA的标签在头部，有较大空间
MP3的v1标签在头部，标签空间较小，v2的标签在尾部，同AC3
http://sv009d./5751933063401783/RGlzazIvNTQvNTQzNjYxNzQyMS83LzcwMDMxMzYyNDM5MDQ1NA../AC3风暴Z.rar或登录http://file./输入提取码：3397078931106753 提取附件
以下为关于音频的英文资料(选自GoldWave Help)：
Appendix A: An Introduction to Digital Audio
Digital Audio Basics
In the digital world, everything is reduced to an on or off state so that it can be stored in computer memory as a single bit of information: 1 or 0. Complex real world things, like images and audio, cannot be directly represented in such a simple manner. An image is rarely composed of black and white dots and audio is rarely just on or off.
Reducing images and audio to a digital state requires an analog-to-digital conversion. Instead of using just one bit of information, many bits are used to more accurately store the state. By using 2 bits, for example, four states are possible: 00, 01, 10, 11. For images, that could be black (00), dark gray (01), light gray (10), and white (11). For audio, that would give four different levels of loudness. Typically, many more bits are used. Most computer video cards use 16 to 32 bits to store a single dot. Sound cards typically use 16 bits for audio levels.
The number of bits to use depends on human perception and bit alignment within computers. Computer tend to bundle bits in groups of 8, called bytes, so using 8, 16, 24, or 32 bits would fit nicely in 1, 2, 3, or 4 bytes respectively. For images, 16 bits do not provide enough states to make the transition from one state to the next imperceptible, so 24 or more bits are used. For audio, 16 bits are adequate, which is what a CD contains, but audio systems using 24 bits will be common in the future.
Samples
Digital audio is composed of thousands of numbers, called samples. Each sample holds the state, or amplitude (loudness), of a sound at a given instant in time. For images, each point of light, or pixel, has a certain brightness and locetion and all pixels combine to make a picture (see figure below). For digital audio, all the samples combine to make a waveform of the sound.
When playing audio, each sample specifies the position of the speaker at a certain time. A small number moves the speaker in and a large number moves the speaker out. This movement occurs thousands of times per second, causing vibration, which we hear as sound.
Digital Audio Attributes
There are several attributes that determine the quality and size of a digital audio file. They are the sampling rate, the bit depth, the number of channels, and the bitrate.
Sampling Rate
The sampling rate is the number of times, per second, that the amplitude level (or state) is captured. It is measured in Hertz (seconds-1, Hz). A high sampling rate results in high quality digital sound in the same way that high resolution video shows better picture quality. Compact disks, for example, use a sampling rate of 44100Hz, whereas telephone systems use a rate of only 8000Hz. If you've ever heard music on the telephone while on hold, you'll notice a big difference in quality when compared to the original music played on a CD player.
Higher sampling rates capture a wider range of frequencies and maintain a smoother waveform. The figure below shows a real world waveform in red and the digital waveform in black at different sampling rates. You can see that increasing the sampling rate makes each step of the digital waveform narrower. The shape more closely follows the real world. In general, the height of each step is reduced as well, but that depends on the number of bits. In simple terms, the sampling rate controls the width of each step.
The rate to use depends upon the type of sound and the amount of storage space available. Higher rates consume a lot of space. In the above example, the CD requires over 5 times the amount of storage as the telephone system for the same digital sound. Certain types of sounds can be recorded at lower rates without loss of quality. Some standard rates are listed in the table.
|
Attributes |
Quality and Usage |
Storage |
|
8000Hz |
Low quality. Used for telephone systems. Good for speech. Not recommended for music. |
0.960 |
|
11025Hz |
Fair quality. Good for speech and AM radio recordings. |
1.323 |
|
22050Hz |
Medium quality. Good for TV and FM radio quality music. |
2.646 |
|
44100Hz |
High quality. Used for audio CDs. Used for digital audio tapes (DAT). |
5.292 |
|
48000Hz |
Very high quality. Used for DVD video or audio. |
5.760 |
|
96000Hz |
Extreme high quality. Used for DVD(Lpcm) audio. |
11.520 |
Channels
Digital audio can have one or more channels. Single channel audio, referred to as a monaural (or mono) audio, contains information for only one speaker and is similar to AM radio. Two channel audio, or stereo audio, contains data for two speakers or two ears, much like FM stereo. Stereo sounds can add depth, but they require twice as much storage and processing time as mono sounds. Most movie theatres have advanced audio systems with 4 or more channels, which are capable of making sounds appear to come from certain directions.
File Compression
Uncompressed audio files tend to be large. CD quality audio requires ten megabytes per minute. That is not a problem with large computer hard drives available today, but it is a problem if you want to save many songs on a portable player or if you want to transfer files over the Internet. Unlike most computer data, audio data does not compress very well using typical compression methods such as those found in programs like PKZIP or WinZip. These methods preserve the data exactly so there is no loss of quality. Such compression is called lossless compression.
To make audio files smaller, complex algorithms have to used. Most of these algorithms sacrifice some quality so that when the data is decompressed, you do not get exactly the same quality you had originally. This type of compression is known as lossy compression. Ideally the quality that is lost is not perceptible, so you do not notice the difference.
The most common method of lossy audio compression is MPEG Layer-3, better known as MP3. It is capable of getting near CD quality audio in less than one tenth the size, which is about one megabyte per minute. More recent algorithms, such as Ogg Vorbis and Windows Media Audio, get even better quality in a smaller size.
Software and hardware that compress audio using complex algorithms are typically referred to as codecs (from coder/decoder). Compression is the same as encoding and decompression is the same as decoding.
Bitrate
Many compressed audio formats measure the compressed size as a bitrate. The bitrate is the number of bits per second (bps) required to store the audio. Usually the number is given in kilobits or one thousand bits. Divide that number by 8 to determine the number of kilobytes required per second.
Internet connection speed (bandwidth) is often measured in bitrates as well. A 56k modem is capable of receiving 56 kilobits per second. If you want people to stream your MP3 audio over a modem, you'll need to compress the file using a maximum bitrate of 56kbps. Due to the connection overhead and Internet protocol, a lower rate would have to be used to ensure the audio can be downloaded fast enough. For DSL and Cable Internet connections, the standard 128kbps MP3 rate can be used.
Audio files can contain a wide range of sounds, from noisy cymbal clashes to silence. Algorithms typically get much better compression on silence or simple audio sections than on complex, noisy audio. This means that the bitrate depending on whether constant bitrate or variable bitrate compression is used.
Constant Bitrate [CBR]
When using constant bitrate, each section of audio compresses to exactly the same size, regardless of the content. If the audio contains silence, then the data may be padded to fill the required bitrate. If the audio contains complex music, then quality may be decreased until it fits within the bitrate.
Constant bitrate is useful for broadcast systems where the transmission rate is fixed. It also make is easy to seek to arbitrary positions within the audio stream or file.
Variable Bitrate [VBR]
Variable bitrate compression uses the smallest size possible for each section of audio. If the audio contains silence, then the bitrate will be very low. If the audio contains complex music, the bitrate will be at its maximum.
Variable bitrate gives the best compression and quality. However, it makes it difficult to seek within the stream or file since there is no direct relation between time and size.
