Sawstudio

Many of today's digital effects processors offer you considerable control over the creation of artificial ambiences for your music, and if you know how reverberation works in real spaces, you'll be better equipped for designing fake ones. HUGH ROBJOHNS boldly goes...（在今天很多的数字效果处理器经常在为你的歌曲带来大量人造的气氛，如果你知道真是空间中的混响是如何工作的，你会更好地去做一个“假”混响。）

Reverberation is something that few people are consciously aware of, yet it is one of the most fundamental aspects of a room's sound character. If you were to blindfold someone, take them to an unfamiliar building and lead them through a succession of rooms, clapping or shouting in each one, they would almost certainly be able to give a pretty accurate description of the size of each room. If they were being particularly perceptive, they might even be able to suggest where they were standing in each room and probably even give some idea of what there was in terms of wall coverings, curtains, soft furnishings and so on! In other words, it is reverberation that gives your brain most of the information it needs to create an aural picture of your immediate environment.
很多人都知道什么是混响，他表现了房间的许多基本特征。如果你带着一个蒙上眼罩的人，带他们来到一座他们不熟悉的建筑物，以及带他们穿过一连串的房间，每间房子都是鼓掌声和叫喊声，他们几乎都能精确的描述出每一个房间的大小。如果的听觉特别好（或者其他感觉比较好），他们可能可以说出自己站在房间的什么位置，以及他们能大概给出这段时间他们所经过房间的墙壁遮盖物，是否有挂门帘，家具有那些等等。换句话说，这是混响的声音为你的脑袋提供了大量的信息，让你的听觉立刻能想象出周围的环境。

Every room has its own sound or 'acoustic', and part of the job of a recording engineer is to assess whether a room's characteristic sound is worth using in a recording. If the dimensions, layout and fabric of the room enhance the recorded sound quality, all well and good, but if not, the microphone technique used should minimise any room sound so that an artificial acoustic can be added later from a reverb processor.
每一间房子都拥有他们自己的独特的声音和“声学”（如果你能找到两间REVERB声音一摸一样的，你就NB了！）在某些工作中，录音工程师们都会去评定一下房间中的特征，看看这些特征（对声音的影响）是否值得（需要）在录音中被运用到。如果房间的尺寸、布局和构造（材料）能提高你房间的声音质量，那就是最好的了。但如果相反，影响音质，那么所有录音室的特征声音（对录制有影响的）都会被降到最小，然后用MIC录制，再之后通过混响效果器加入人造混响。

In this age of digital technology, artificial reverberation is not only more affordable than ever before, but can also be stunningly realistic and very controllable. With a good understanding of the physics of natural reverberation, and the fundamental operational principles of reverb processors, it is possible to quickly create the illusion of any acoustic environment you can imagine.
这个时代的数字技术中，人造混响不一定比以前的好，但同样令人觉得非常真是以及非常容易控制。很好地理解自然混响的物理特性，以及混响效果器的基本运算法则，能帮助你很快的幻想出所有假想的声学空间。

Remember, though, that if you choose to use artificial reverb, it is essential that the recording has the absolute minimum of the recording venue's room sound. If the original environment can be heard, adding extra 'fake' reverbs will just result in a cluttered sound and the mix will often become confused and indistinct.
记住，尽管你选择了人造混响，他的本质是录制房间特征被消除以后的声音（就是消除所有反射声等等影响录音的声音）如果原是环境的声音依然还能被听到，那么在加入“假”REVERB的时候，立刻就能让你的声音变得混乱，以及整个MIX在这样得情况下经常都会变得混响和不清晰。

To understand what a reverberant sound actually is and what information our hearing system is able to extract from it, you need to think about how sound waves travel and what happens when they encounter various surfaces.
去了解一下混响声音实际上是任何形成的，以及我们的听觉系统能从中获得什么其他信息，有必须去思考声音是如何传播的，以及他们遇见不同材料的平面后，会发生什么问题。

The first thing to consider is how fast sound travels in air. My old school physics books said 760mph and the new ones probably say 340 metres per second, but I find it hard to relate these numbers to anything meaningful (other than the average speed of traffic around the Evesham bypass in the morning....).
首先要做的时候去了解声音在空气中的传播速度。我以前学校的物理课本说是760MPH，而新书就说一般情况下340m/S，但是这不能用于所有的情况（这是平均速度）。
A much more useful figure to tuck away in the dark and dusty recesses of your mind is that sound travels roughly a foot each millisecond. Assuming that you're not too young to relate to feet as a valid dimensional measurement, this rule of thumb will allow you to calculate and set one of the most critical parameters of any reverb processor. One quick side note: the speed of sound varies with the condition of the air. Temperature, humidity and pressure (ie. altitude) all have significant effects on the speed of sound, and in certain applications the 1ft/ms guide is not sufficiently accurate. However, as far as dialling up room sounds on a reverb unit is concerned, it's close enough. Imagine that you're standing in the middle of a very large, brick-walled barn with a deep covering of straw on the ground. You have a spontaneous urge to clap your hands: what happens? Well, the very first thing you'll hear is the direct sound of your hand clap, and it's this direct sound that the brain uses to pinpoint the direction of the sound source.
一个更加常用的描述是在你认为黑暗而且干燥幽静的地方畅饮，那么声音传播的速度就是1步长＝1MS。这个单凭经验的办法能让你去计算以及设置所有REVERB的参数。从侧面可以知道：声音的速度受空气条件的影响而出现变化。温度、湿度和气压（例如海拔高度）都对声音的速度有重要的影响，而运用1FT/MS作为标准依据也不是非常准确的。但不管怎样，在REVERB效果器上改变声音的远近，和上面的说法都有非常密切的关系。假设你站在一个用砖瓦搭建的空空荡荡的房间中间，以及铺上一层比较厚的稻草。你随意的鼓掌几下：会发生什么事情呢？首先你会听到你鼓掌的时候，发出的直达声，脑袋就考这个直达声来确定声音的方向来源。


Assuming that the hand-clap radiates sound waves in all directions simultaneously, the next thing you'll hear will be reflected sound from the various room boundaries or nearby objects. As the floor of this imaginary barn is covered with a deep layer of straw, there will be no significant reflection from the ground (although in practice this is often a dominant source), so the first 'room sounds' will be reflections from the side walls and ceiling. If the barn measures 40ft by 60ft and is 20ft high, and assuming we're standing in the middle of it, the very first reflection will come from the roof, after about 30 milliseconds.
同时，鼓掌声同时向所有方向辐射出去，然后你就能听到来自房间所有角落或者其他物体反射回来的声音。而这个房子地面铺上一层稻草，这样使得地面不会有太大的反射声音产生（实际上地面的反射回来的声音经常是反射声的主要来源），所以最早听到的反射声是来自周围的墙壁和天花板。如果这个房子的规格是40ft(步长)*60ft（步长）*20ft（步长），以及我们站在房子的中间，那么最初的反射生来自房顶（距离最短嘛），大概用了30MS。

This time delay can be estimated by working out the distance the sound wave has to travel -- if you're a man standing up in the barn, your hands and ears are likely to be about four and six feet respectively above the floor, so (20-4) + (20-6) = 30. Sound travels roughly 1 foot a millisecond, so it will take 30ms for the sound of the clap to reach the ceiling and return to your ears. Similar calculations reveal that the side-wall reflections occur 40 and 60ms after the initial direct sound.
这个时间上的延迟能在声音传播的距离上作出估计——如果你站立在这个房子里面，你的手和你的耳朵分别高于地面大概4到6步长，所以（20－4）＋（20－6）＝30（这里20－4是手到屋顶的距离，20－6是屋顶的声音反射回你耳朵的距离，声音传播是1步长/MS,声音的传播方向是手掌－>屋顶－>你的耳朵）。利用同样的就算方法，四周的墙壁反射声音大概会在直达声音之后40ms和60ms左右出现。

"...artificial reverberation is not only more affordable than ever before, but can also be stunningly realistic and very controllable."

The time taken for these first reflections to be heard goes a long way towards defining the perceived acoustic size of the room. Short delays imply small rooms, and long delays large rooms. To mimic this natural characteristic, artificial reverberation units normally allow the user to set the time delay between the direct sound and the very first reflection with a parameter called Pre-Delay. This is generally adjustable in millisecond increments (or finer) over an astonishingly wide range.
初次反射声音在传播一段距离之后被你所听到，这段距离所消耗的时间被认为是有利于定义这个房间的声学感觉的大小，从而判断这个房间的大小。短延迟代表这个是小房间，而长延迟代表这个是大房间。在模拟这个自然特征上，混响效果起一般都允许用户去设定直达声和初次反射声（这里用VERY是特制最最最早的那些，但是在小房间里面比较难区分，大空间就比较好区分了）之间的一个时间延迟，这个参数控制我们称为PRE-DELAY（预延迟）。混想器一般情况下，通过调节MS来增加（或者变得更好）声音的宽度。

So the first reflection comes from the closest surface, and is followed by others from the adjacent room boundaries, the whole ensemble being known as Early Reflections. The timing, relative amplitudes and timbre of these individual reflections are determined by three things: the placing, angles and nature of reflective surfaces; the location of the sound source; and the position of the listener. Moving any of these will change the relative timing of the early reflections quite significantly, but our hearing system is remarkably good at extracting even the most subtle information. Consequently, these initial reflections and their relative timing are very important in defining an imaginary acoustic space.
所以初次反射声是来自最接近的反射面，然后是紧接着来自其他临近墙壁的反射声，而所有这些声音就被称为EARLY REFLECTIONS（早反射，非硬件/效果卡支持的PLUGINS好像就只有WAVES有这个设置，而WIZOO是采样REVERB，是采样决定这个ER，所以没有很明确的关于这个参数的设置）。时间上的选择、相对幅度以及这些独立反射声的音调都是由三样事物所决定：第一是现场、房间里面的角落和自然反射面；第二是声源的位置；第三是观众的位置。改变这些东西中的任何一样东西，都会对早反射的相对时间有十分重要的改变，但我们的听觉系统非常善于发现这些微小的变化。因此，这些初次反射声和他们的相对时间对于定义假设声学空间是非常重要的（REVERB PLUGIN就是在做一个假设的声学空间）。

The better reverb units allow the user to alter not only the value of the pre-delay, but also the number, grouping, timing, amplitudes and tonal qualities of the first reflections. In some cases, these parameters are preset by the manufacturer and are simply selected from a list of programmed options, such as Hall, Chamber, or Plate, although occasionally they are derived from measurements taken in genuine acoustic environments. The most sophisticated machines allow the user to specify the cubic volume of the imaginary room, or even its precise dimensions, together with the source and the listening positions!
一个优良的混响效果器都允许用户改变预延迟参数重不只是PRE-DELAY，还有数量（ER的数量，参考WAVES IR1）,种类（不同的类型选择，例如HALL / PLATE等等）、时间设置、振幅和音调的选择（这里在非硬件支持重比较少有）。所以大家可以在REVERB效果起重设置不同的房间参数，来改变房间的特性，变成HALL/CHAMBER/PLATE等等，甚至有些REVERB效果器还能选择声源位置和听者的位置。

[ 本帖最后由 Ying 于 2006-7-31 21:17 编辑 ]

TAILS

So first we hear the direct sound; then, a short time later, a number of discrete reflections return from the various surfaces in the room. However, these reflections don't just stop when they reach the listener -- they continue until they reach other surfaces, where they instigate more reflections. These reflections start even more reflections and the sound density becomes too great to allow us to distinguish the reflections as separate events.
所以首先我们先听到直达声；然后，一个短时间的延迟后，一连串不连续的反射声在房间的不同反射面之间反射回来。当然，这些反射声不是到达听众就立刻停止传播的，他们会继续传播，直到碰到另外一个反射面，然后产生更多的反射声。这些反射声音引发了更多的反射声，以及这些声音密度变得十分大，直到让我们能区分出反射声的所有个别事件（反射声的所有细节）。

At each reflective surface, some of the sound energy is absorbed, and more is lost as the sound travels through the air -- which is why reverberation gradually dies away. This reverberation 'tail' may last for anything from, say, 0.3 seconds (for a dead-sounding room), through to several seconds in a church or big concert hall.
在每一个反射面，声音的一部分能量被吸收了，以及更多的能量是在声音传播中被消耗了——这就是声音逐渐变弱消失。一个混响的声音的长度和房间的空间和材料有关系，如果在一个建筑材料吸音比较厉害的地方，声音的“尾巴”可能比较小，只有0.3s，而在一个大的教堂或者演奏厅就能走上N秒（一般这些建筑的材料都是能提高空间混响的，不会吸收太多的声音能量）。

The length of the reverberation tail is usually specified in terms of its 'RT60'. This is defined as the time taken for the reverberation to fall by 60dB in level below the original direct sound. Every reverb unit allows this time to be adjusted, normally through a parameter called Decay Time.
混响的tail（尾音）长度经常称为RT60(WAVES 的 IR1就有这个设置)。这个定义是：混响声降低到低于直达声60dB的过程中所消耗的时间。每一个混响效果起都允许对这个RT60做调整，一般情况下也被成为DECAY TIME.

It's important to note that the reverberation tail lasts for different durations at different frequencies. High-frequency sound waves have a lot of trouble persuading air molecules to vibrate quickly enough to pass the sound energy onwards. Consequently, high-frequency sounds tend to die away, as they travel, much faster than mid-frequency sounds. On top of that, high-frequency sounds are absorbed by soft furnishings (which includes people and even wallpaper!). On the other hand, high frequencies reflect strongly from a wide range of surfaces, such as windows, sound desks, equipment racks, and so on. At the other frequency extreme, low frequencies are only reflected by large and very solid objects, so there may be little LF in the reverberation at all in some circumstances, but a definite bass 'bloom' in cave-like rooms!
有一点是非常重要的，必须记住，混响的尾音中，随着持续时间的变化（或者是在衰减过程中），频率都会随着发生变化。高频在空气分子中难以传播，因此高频的声音渐渐趋向消亡，而在传播中，高频的传播速度比中频的快。另外，高频的声音容易被软家具所吸收（其中还包括我们的身体还有所有的墙纸！）。另一方面，比较宽的反射面（宽度超过高频的波长）就能产生十分强烈的反射，例如窗户，坚硬的桌子，设备架，或者其他反射面比较大的物体。而在频率的另一个极端中，低频只会被巨大或者非常坚硬的物体所反射（根据物理的特性，物体的反射面必须大于波长才产生反射，希望我初二物理没忘记），所以无论任何情况都有可能需要对混响声做一点LF，不然你的低频在混响中就好像来自山洞一样。

To help provide this level of realism, most reverb units allow you to adjust the reverberation time for high (and sometimes low) frequencies relative to middle frequencies, and introduce some kind of overall equalisation to the reverberation tail.
许多的混响效果起都给予相关的参数对TAIL的频率做调整。这里建议大家能在REVERB之前加入一个EQ，先对进入REVERB之前的声音做一次低频清洗运动（适当就好，不要太过激），然后反复SOLO REVERB的声音，确定低频在经过REVERB后没有变得溷浊模糊。

[ 本帖最后由 Ying 于 2006-7-31 21:20 编辑 ]

^_^又出好文章了

Let's recap on the parameters that today's digital reverb processors are likely to offer for the simulation of real acoustic spaces. Firstly, although I haven't previously mentioned it, there is usually a means of balancing the direct sound against the reverb. The direct sound is often referred to as 'dry' and the reverberation as 'wet', so a wet/dry control will probably be in there somewhere. Some of you will know how unpleasant it sounds when an analogue-to-digital (A/D) converter is overloaded, so a critical control on digital reverbs is the input level control, and its associated headroom meter.
以上一段主要介绍干声和湿声的概念，干声＝直达声，湿声＝混响声。在A/D 转换中，声音可能出现过载，所以数字混响的电路控制是输入级别的控制，而且是并联的HEADROOM的METER.（这里我自己也不是太明白）。

The first control which defines the reverb character is pre-delay, which effectively defines the distance of the first reflective surface. This will be followed by one or more parameters for controlling the number, timing, amplitude, and timbre of the other early reflections. Some machines provide controls called 'Pattern', 'Level' and 'Room Size'; others might simply offer preset venue simulations ('Hall', 'Chamber', 'Jazz Club', and so on).
在调整混响中，首先要调整好混响中的PRE-DELAY。这个可是影响声源和第一反射面的距离啊（同样也是定义出空间的大小）。这个PRE-DELAY可能会由一个或者多个参数来控制，NUMBER（我觉得这里是数量，可以参考IR1里面的那个图形你就明白了），时间设置，振幅以及其他早反射的音调（音品）。一些混响器给出的控制参数有“混响的图像（类似TC或者IR1）”，“音量”以及“房间大小”；其他就可能只是给出一个经常用的模拟参数，例如”HALL”,”CHAMBER”,”JAZZ CLUB”或者其他。（在IR1中你能发现DELAY和早反射之间有非常亲密的关系！）
在早反射之后，就是设置好TAIL，也就是某些REVERB PLUGIN中的DECAY TIME，相关的参数和TAIL中的频率、音调有关，只要设置好这些参数，就能在你自己的房间录制一个歌曲，然后通过调整混响器，模拟出很多声学空间来，对于初学的你，能节省不少装修的房子的金币啊。



You cannot create an artificial acoustic space if you don't know what a real one sounds like to start with, and the only way you can find out is to actively listen to sounds in as wide a range of environments and circumstances as possible. Everyone has a very detailed subconscious knowledge of how different rooms sound, and although few are able to analyse the reverberation structure, most spot incongruities in artificial reverberation very easily.
整段话的意思是：你没有感受过真是声学空间中所产生的自然混响，你是没有办法创造出一个非常优秀的模拟真实声学空间的。通过不停的学习，你慢慢的基础过不同的真是声学空间，你的知识就越来越丰富，对于你创建一个模拟的声学空间就越容易。

It's very revealing and informative to consciously listen to the sound of different rooms as you move around in a building -- even in places with which you thought you were familiar. Try to analyse in your own mind what sort of pre-delay, decay time, early reflections and high-/low-frequency decays naturally occur to create the 'sound' of that room. Don't just listen to indoor reverberations, either -- try to assess the reverberant features of the local high street, the great outdoors, a wood or forest, or wherever you happen to be. You will find reverberation in places you didn't expect it, and may be surprised to discover that places you assumed to be reverberant actually are not!
去凝听不同建筑物不同房间所产生的混响声音是非常有意义的，这些空间能为你提供非常有益的信息。尝试凝听房间所营造的PRE=DELAY/DECAY TIME/ERALY REFLECTIONS 和HIGH-/LOW-FREQUENCY DECAY（高低频衰减）。但不要只是在房间里面（室内）去凝听，尝试到马路上，大的广场，宽阔的户外，树林和森林（国内是比较难的了，大家保护绿化啊），看看会发生什么事情，会出现什么状况？

[ 本帖最后由 Ying 于 2006-7-31 21:21 编辑 ]

The very nature of genuine reverberation is that it tends to come at you from all over the place, but particularly from the sides of the room. This has significant effects on the compatibility between stereo and mono versions of your mix, since the mono listener is effectively denied any information from the sides of the stereo image.
真实的混响是非常自然的，在你所处于的空间里面，混响声从四面八方反射到达你所在地方，但其中最显著的是从房间的四周反射回来的声音。在你的MIX中Stereo和Mono两种版本的兼容性是非常重要的（其实非常建议你在MIX完一个工程以后，你应该作M和S之间的转换，看看在Stereo中听到的乐器，在MONO中是否还能清晰的听到。如果不行的话，大家就要找找原因了），以前的MONO听众能非常有效的拒绝两边立体声的“图像”信息。

To see how this happens, consider a simple M&S (Middle & Side) stereo microphone technique being used to record something in a reverberant room: the stereo listener hears the full acoustic in all its glory, but the mono listener hears only the forward-facing 'M' microphone, not the sideways-facing 'S' microphone -- and guess which one picks up the bulk of the room sound? This absence of reverberation in mono afflicts artificial reverb processors as well as natural acoustics. In practice, the amount of reverberation heard in mono may be substantially less than that in the stereo balance, and if mono listeners are likely to be an important part of your music audience, always check for mono compatibility. In general, you almost always have to compromise the balance in some way because either the mono will be too dry, or the stereo will be too wet!
看看下面的设置会发生什么问题，考虑运用一个简单的M&S（Middle & Side）立体声拾音技术运用与录制房间的回响：立体声能让听众听到所有的声学反应（反射声等等），但MONO的听众就只能听到前面‘M’所拾取的声音，没有听到任何两边‘S’所录取的声音——现在猜想一下那个拾取到更多的房间反射声？MONO的烦恼就是没有录制到回响的声音，需要用混响效果器去模拟一个真实的声学空间，效果同样也能做到自然真实的。在实际运用中，利用MONO所拾取到的混响声，是要少于STEREO的，而如果有人喜欢用MONO去收听你的歌曲，那么你就要去检查一下MONO的兼容性了（在MIX后用MONO来检查一下STEREO里面的元素是否还能清晰听到）。但某些时候你还是要作出一些妥协，毕竟MONO让声音非常DRY，而STEREO让声音变得非常的WET.


"You cannot create an artificial acoustic space if you don't know what a real one sounds like to start with...""You will find reverberation in places you didn't expect it."

Tricks worth trying include reducing the stereo width of the reverb (turn the pan-pots in a bit towards the centre instead of having the reverb returns running out to full left and right), or mix in a small amount of reverb from another reverb processor, panned centrally. The extra reverb should be set up with the same parameter values as the stereo reverb, although a slightly shorter pre-delay and longer HF decay time often work well. The balance between the dry sound, the mono reverb, and the stereo reverb needs to be adjusted carefully, while you continuously switch between mono and stereo listening to find the most uniform results in the two modes.
诀窍是尝试降低STEREO的混响宽度（把PAN的参数调节到靠中间的位置，以代替混响设置为极左和极右），或者由另外一个混响器输出混响效果，PAN已经被设置了（后者不是采集自然混响了）。尽管极短的预延迟和比较长的HF（高频滤波）衰减经常能得出很好的效果，但Stereo混响产生的额外混响都应该是由相同的参数所形成的（相同是指上面的PAN设置，PAN设置太宽的原因）。平衡MONO混响和Stereo混响之间的之间的混响声，我们需要非常小心的调节，不停地在MONO和STEREO之间聆听选择，直到你找出他们两个模式中最好得结果。

In matrix surround systems (such as Dolby Surround), real or artificial stereo reverb tends to spread across the rear channel quite naturally as a result of the way in which the rear-channel information is encoded and decoded. Altering its stereo width controls the front-back balance, narrowing the reverb pulls it to the front, and increasing the width pushes more to the sides and rear.

Many stereo digital reverb units have a single input and a stereo output, and this often causes people to wonder how the reverberation can be 'true stereo' with only a mono input. The answer is simple if you consider the real situation of a sound source within a reverberant space.
许多STEREO数字混响器都是单输入而输出是STEREO，这样经常导致人们非常惊奇，只是单声道输入，为什么反射声居然如此的“TRUE STEREO”（好像真实的立体声一样）。如果你想象一个声源在一个反射空间中的表现，那么答案就非常简单啦。

If someone claps, there's only one sound source, yet the reverberation will come from all directions and could be captured by a simple stereo microphone array -- a mono input to the room and a stereo output from it. Of course, in a more complex situation with, say, a string quartet in the room, there are multiple sound sources and each will have slightly different pre-delays and early reflection patterns, but this is usually a very subtle distinction, and in practice the mono-in, stereo-out system of most digital reverb units works perfectly adequately.
你在一个空间鼓掌，而反射声从多方面的反射回来，那么这样和单声道输入，STEREO输出是非常类似的效果。而且其他情况也是相同的，例如房间中进行STRING四重奏，这是多声源，以及每一个声源的PRE-DELAY和EARLY REFLECTOIN都会有一点的轻微差异，在实际中，运用MONO-IN,STEREO-OUT模式的数字混响，能很好的完成以上工作。

Something few people ever check is the line-up of a stereo reverb unit. However, it is a stereo source and should be treated in just the same way as any other stereo signal, which means making sure that the left and right reverb outputs have the same gain and equalisation through your mixer. I find that a quick, easy and reliable method of doing this is to simply dial up a 3- or 4-second decay time and send a brief burst of signal into the machine. Listen carefully to the dying reverb tail: it should decay centrally, possibly even becoming narrower in width as it goes (although this depends on the particular algorithm). If the reverb tail appears to collapse towards one side or the other, your return channels have different gains and should be adjusted.



各位不好意思，文中有两段我实在不太明白其中确切的意思，所以没有完全翻译出来，但所涉及的问题不是太重要，而最后一段是请大家在调节混响的时候，小心的调节DECAY TIME，他能提高你声音的宽度。不好意思了，请大家见量，还有两段，我分开两次上传吧；
这是 来自WWW.SOUNDONSOUND.COM上的文章

好同志，您辛苦了

我认为混响只是营造了空间环境，而干湿比却控制了声源的远近。
例如当你在空旷的音乐厅里由远及近的走来（还不停的唱歌），混响的各个参数没变，但干湿比却在不断变大。所以，如果不希望随着加入混响，元素的位置后移。那就加大干湿比。
                                                                                           若认识有误，请老师们批评。

MONO AND STEREO这章很值得学习~
我在混音中就经常发现在MONO里两边的乐器混响声超大~比干声大声多~~

那么简单的说~解决的问题是不是把有很多混响的声音尽量往中间PAN?

难怪听外国的东西会发觉在极左极右的东西往往会很干~

稳重的意思是：不要极左极右就可以了，你的PAN设置是否正确，是否清晰听到所有乐器，最好的办法就是再MIXDOWN之前，点一下主输出的MONO按键，如果再mono的情况下都能听到所有乐器声音的话，那么基本上就正确了。文章的尽量靠前面不是叫你完全集合再终殃，这样就很容易导致所有乐器打架了，只是再适当的情况下尽量的靠前。

而混响其实不要靠中间，混响声应该和本身的干声PAN设置大致相同，我不知道其他软件是否有对反送回来的信号做PAN调节，但SAWSTUDIO就有这样设置，但没有精确的数字表现，而是靠听觉来恒定。如果你所有的混响都集中在中间，你会发现混响的声音都互相抵触了～就类似人声、GT、KICK、SNARE、STRING的PAN都设置在中间一样。

还有外国的东西应该比较少极左极右，他们有的时候叫极左极右叫DEAD LEFT /DEAD RIGHT.

我自己的看法是，混响最重要的元素不是WET/DRY，而是控制距离远近的PRE-DELAY和ER，WET/DRY只是反应干声和反射声的比例；当然也有一定的远近关系。但WET/DRY应该可以模拟一个空间的吸收（反射系数），前提条件是这个混响没有这方面的设置；空气吸收率越低，反射面反射率高，吸收率低，那么有可能提高空间的反射声的存在，不过这个只是参考。

我想PRE-DELAY和ER应该控制的是声源与反射面的距离，我说的距离，是在混音中的远近距离，或者是声源距拾音器的距离。而模拟反射系数的应该是FEEDBACK。YING兄觉得呢？

可以这样理解～～这个我也觉得是这样～

还有一个比较重要的是高频滚降，可以影响距离感，在空气中，高频会逐渐被吸收

简单的例子,提高湿干比的同时降低音量,可以明显感觉到在这个空间里声源变远