编码解码器主要特点-中英对照
Speex的主要特点可以总结有以下几点：
•免费软件/开放源码、免专利费和版权费
•采用嵌入型的比特流来集成窄频带和宽频带
•适用的比特率的范围很广（从2.15 kbps 到44 kbps）
•动态比特率转换（AMR）和可变比特率（VBR）运算
•声音活动探测（VAD和VBR整合）和不连续传送（DTX）
•可变的复杂性
•嵌入式的宽频带结构（可扩展的采样率）
•32kHz的超宽频带的采样率
•强度立体声编码选项
•定点的实现
2.3 预处理器
这一部分引用了在1.1.x 分支中介绍的预处理器模块。预处理器是设计用来在运行编码器之前来处理声音的。预处理器提供了三个主要的功能：
•噪音抑制
•自动增益控制（AGC）
•声音活动探测（VAD）
图2.1 声学回音模型
降噪器可以用来减少出现在输入信号中的背景噪音的数量。不论这降噪以后的信号是不是由Speex来进行编码，这过程都提供了更高质量的语音。然而，在编码解码器使用降噪的信号的时候，都会得到附加的好处。语音编码解码器通常（也包括Speex）不能很好地处理嘈杂的输入，即会倾向于放大噪音。而降噪器则会大大的减少这个影响。
自动增益控制（AGC）是一种用来处理下面这种情况的特性：由于不同的设置之间存在大量的差别，所以记录的音量可能会有差别。AGC提供了一种将某一信号调节到参考音量的方法。这对于网络语音电话很有用，因为它免除了人工调节麦克风增益的需求。另外一个优势怎是通过将麦克风的增益设置到一个保守（低的）水平，从而更容易的避免剪音。
由预处理器所提供的声音活动探测（VAD）比由编码解码器所直接提供的要更加先进。
2.4 自适应抖动缓冲器
当传送中的声音（或者就此而言的任何内容）超过了UDP或者RTP，数据包可能会丢失，或者经过不同的延迟而到达，或者甚至发生故障。抖动缓冲器的目的是给数据包重新排序并且使它们缓冲足够长的时间（但是不会超过所必须得时间）从而使它们能够传送从而解码。
2.5 回声消除器
在任何的免手持的通讯系统中（图2.1），来自远端的语音都是在本地的扬声器上播放，然后传送到房间里并且由麦克风所捕捉到。如果由麦克风所捕捉的音频直接发送到远端的话，那么远端的用户会听到他自己声音的回音。因此回声消除器的作用就是在回声被发送到远端之前将其消除。回声消除器的目的是改善远端的通话质量，理解这一点是很重要的。
2.6 重采样器
在一些情况下，将音频从一种采样率转换成另一种采样率可能是一种很有用的方法。这么做是有很多种原因的。这样可以混合不同采样率的数据流，从而来支持声卡所不能支持的采样率，或者来进行转码等等。这就是为什么现在重采样器会成为Speex项目的一部分的原因。这个重采样器可以用来在任何两种任意的频率（比率只能是一个有理数）之间相互转换，并且可以在质量/复杂性之间达到平衡。
2.2 Codec
The main characteristics of Speex can be summarized as follows:
• Free software/open-source, patent and royalty-free
• Integration of narrowband and wideband using an embedded bit-stream
• Wide range of bit-rates available (from 2.15 kbps to 44 kbps)
• Dynamic bit-rate switching (AMR) and Variable Bit-Rate (VBR) operation
• Voice Activity Detection (VAD, integrated with VBR) and discontinuous transmission (DTX)
• Variable complexity
• Embedded wideband structure (scalable sampling rate)
• Ultra-wideband sampling rate at 32 kHz
• Intensity stereo encoding option
• Fixed-point implementation
2.3 Preprocessor
This part refers to the preprocessor module introduced in the 1.1.x branch. The preprocessor is designed to be used on the
audio before running the encoder. The preprocessor provides three main functionalities:
• noise suppression
• automatic gain control (AGC)
• voice activity detection (VAD)
8
2 Codec description
Figure 2.1: Acoustic echo model
The denoiser can be used to reduce the amount of background noise present in the input signal. This provides higher quality
speech whether or not the denoised signal is encoded with Speex (or at all). However, when using the denoised signal with the
codec, there is an additional benefit. Speech codecs in general (Speex included) tend to perform poorly on noisy input, which
tends to amplify the noise. The denoiser greatly reduces this effect.
Automatic gain control (AGC) is a feature that deals with the fact that the recording volume may vary by a large amount
between different setups. The AGC provides a way to adjust a signal to a reference volume. This is useful for voice over
IP because it removes the need for manual adjustment of the microphone gain. A secondary advantage is that by setting the
microphone gain to a conservative (low) level, it is easier to avoid clipping.
The voice activity detector (VAD) provided by the preprocessor is more advanced than the one directly provided in the
codec.
2.4 Adaptive Jitter Buffer
When transmitting voice (or any content for that matter) over UDP or RTP, packet may be lost, arrive with different delay,
or even out of order. The purpose of a jitter buffer is to reorder packets and buffer them long enough (but no longer than
necessary) so they can be sent to be decoded.
2.5 Acoustic Echo Canceller
In any hands-free communication system (Fig. 2.1), speech from the remote end is played in the local loudspeaker, propagates
in the room and is captured by the microphone. If the audio captured from the microphone is sent directly to the remote end,
then the remove user hears an echo of his voice. An acoustic echo canceller is designed to remove the acoustic echo before it
is sent to the remote end. It is important to understand that the echo canceller is meant to improve the quality on the remote
end.
2.6 Resampler
In some cases, it may be useful to convert audio from one sampling rate to another. There are many reasons for that. It can
be for mixing streams that have different sampling rates, for supporting sampling rates that the soundcard doesn’t support, for
transcoding, etc. That’s why there is now a resampler that is part of the Speex project. This resampler can be used to convert
between any two arbitrary rates (the ratio must only be a rational number) and there is control over the quality/complexity
tradeoff.
2013.2.5