22.4 音频配置和设置——REAPER用户手册中文版

琴韵晓波

楼主 琴韵晓波说：
22.4 Audio Configuration and Settings
22.4 音频配置和设置

We’ve already mentioned (Chapter 1) the Audio Device screen (in the Preferences window), but it’s worth discussing this topic a little more. Two important issues in the digital audio world concern buffers and latency. Understanding what these terms mean will help you in obtaining the optimum settings for your system.
我们已经提到(第1章)音频设备接口(在“首选项”窗口中),但这个主题值得进一步讨论。数字音频世界中的两个重要问题,涉及缓冲区和延迟。了解这些术语的含义将有助于您获得系统的最佳设置。
Getting buffer size right is essential if you are to get the best performance from your audio interface. Make them too small and you'll get audio clicks and pops; make them too large you'll notice delays in your headphones when you’re input monitoring, especially with software synthesizers.
如果要从音频接口(声卡)获得最佳性能,正确的缓冲区大小至关重要。把它们调得太小,你会听到音频卡嗒声和爆裂声;把它们调得太大,你会注意到在进行输入监听时,耳机出现延迟,特别是使用软件合成器时。
The goal is to achieve the lowest possible latency without experiencing dropouts. The optimum setting depends on a number of things, not least of which are the audio interface (PCI sound card, firewire or USB device) that you are using, your PC’s characteristics, as well as which software you’re running. The use of plug-ins can also increase latency, some more than others.
我们的目标是实现尽可能低的延迟,而不会出现丢失。最佳设置取决于许多因素,尤其是您正在使用的音频接口(PCI声卡、火线或USB设备)、PC的特性以及您正在运行的软件。插件的使用也会增加延迟,有些比其他的更多。
Consider what happens when you are working at your DAW. As you play your tracks, a constant stream of data passes from your hard disk and/or RAM to your sound card. Amongst other things, this converts the digital data to an analog form, so that it can be sent to, and heard thru, your speakers or headphones. When you record the reverse happens - the incoming analog audio stream has to be converted to a digital format.
想想当你在你的DAW工作时会发生什么。播放轨道时,从硬盘和RAM到声卡的数据源源不断。其中,它将数字数据转换为模拟格式,以便可以将其发送到您的扬声器或耳机,并通过扬声器或耳机收听。当您录音时,是相反的情况,传入的模拟音频流必须转换为数字格式。
When you are overdubbing, both of these activities are going on at the same time. Indeed, with input monitoring, the track or tracks being recorded are actually being fed into the computer in analog form, converted to digital, processed, then converted back to analog again as they are fed back to your headphones. Latency is a measure of this delay. Incidentally, you can make this activity easier if your audio device itself supports input monitoring. In this case, the incoming audio stream that is being recorded is fed straight back into your headphones before it even gets into the computer.
当你做叠加的时候,这两个活动是同时进行的。事实上,通过输入监听,录制的一条或多条轨道,实际上是以模拟形式输入计算机,转换为数字,进行处理,然后在反馈到您的耳机时,再次转换回模拟形式。“Latency(延迟)”是这种延迟的单位。顺便说一句,如果您的音频设备本身支持输入监听,您可以使此活动变得更容易。在这种情况下,正在录制的传入音频流甚至在进入计算机之前就会直接反馈到您的耳机中。
As if this wasn’t complicated enough, remember that your Mac or Windows computer is performing all sorts of other tasks at the same time as you are recording. Windows users, for example, can view their Task Manager and check the Processes page to have a look. That’s what we mean by multitasking.
如果这还不够复杂,请记住,您的Mac或Windows计算机在录制的同时还在执行各种其他任务。例如,Windows用户可以查看他们的任务管理器,并查看进程页面。这就是我们所说的多任务处理。
So how is Windows able to keep up with its other 99 or so tasks at the same time as handling your recording? It does so by dividing its resources between all the tasks at hand. In order to maintain a constant audio stream, small amounts of memory are allocated to storing this audio stream. These areas are called buffers.
那么,Windows如何才能在处理录音的同时,完成其他99项左右的任务呢?它通过在手头的所有任务之间分配资源来做到这一点。为了保持恒定的音频流,分配少量的存储器来存储该音频流。这些区域称为缓冲区。
Thus, as you play back your audio, Windows hands over to your sound card a block of audio that the sound card releases at a steady rate to your external amplifier. If it empties these buffers before Windows gets round to handing it some more, then you’ve got a problem. The same principle applies, but in the reverse direction, when you’re recording. And again, when you’re layering or overdubbing, or using REAPER’s input monitoring, both activities are going on at the same time.
因此,当您播放音频时,Windows会将一个音频块交给您的声卡,声卡会以稳定的速率将其释放到外部放大器。如果它在Windows抽出时间处理更多缓冲区之前清空这些缓冲区,那么您就有问题了。同样的原理也适用,但是当你在录音的时候,方向正好相反。再说一次,当你在分层或叠加录音,或者使用REAPER的输入监听时,这两种活动是同时进行的。
If the buffers are too small, you’ll get gaps in the audio stream. These can lead to pops, crackles and in some cases even dropouts (that’s when playback and/or recording just stops suddenly and unexpectedly). Making buffers larger is usually enough to fix this problem, but it brings with it another. Make them too large, and the data at the end of the buffer has a long wait before it is processed. That’s when you experience unacceptably large latency, such as the delay between pressing a keyboard and hearing the note in your headphones.
如果缓冲区太小,音频流中会出现间隙。这可能会导致爆裂、咔哒,在某些情况下甚至会中断(即播放或录制突然意外停止)。增加缓冲区通常足以解决这个问题,但它也带来了另一个问题。如果将它们设置得太大,则缓冲区末尾的数据在处理之前需要等待很长时间。这就是当你体验到不可接受的长时间延迟时,比如按下键盘和通过耳机听到音符之间的延迟。
The macOS uses CoreAudio to manage these issues. PC users, on the other hand, are faced with a possibly confusing choice as to which drivers to use.
MacOS使用CoreAudio来管理这些问题。另一方面,PC用户在使用哪种驱动程序方面面临着可能令人困惑的选择。
ASIO drivers generally offer better latency than others. As a rule, the preferred order is ASIO, WDM, DirectX and MME, in that order. This is where the ASIO Configuration button on the Audio Device screen is so important (assuming, of course that you are using ASIO drivers). It opens the audio settings screen for your soundcard. Shown below is an example of the settings page for one particular external soundcard. It is only an example - don’t expect yours to look identical, although it is likely to contain similar settings (sample rate, buffer size, etc.)
ASIO驱动程序通常比其他驱动程序提供更好的延迟。通常,首选顺序是ASIO、WDM、DirectX和MME。这就是音频设备界面上的ASIO配置按钮非常重要的地方(当然,假设您使用的是ASIO驱动程序)。它会打开声卡的音频设置界面。下面显示的是一个特定外部声卡的设置页面示例。这只是一个示例-不要期望您的设置看起来完全相同,尽管它可能包含相似的设置(采样率、缓冲区大小等)。
The smaller the buffer size, the greater the load on your computer’s CPU. So after making changes to your buffer size, check the CPU usage on REAPER’s Performance Meter. You can use this fact to your advantage when mixing.
缓冲区越小,计算机CPU上的负载就越大。因此,在对缓冲区大小进行更改后,请检查REAPER的性能仪表上的CPU使用率。在混音时,你可以利用这一点对你有利。
As a rule, low levels of latency are only really needed for recording, not when you are only playing audio back. Therefore, if you find that you are pushing your CPU close to its limits, you will often be able to fix this by accessing your audio card’s control software and increasing the buffer size.
通常,只有在录音时才真正需要低延迟,而不是在只播放音频时。因此,如果您发现您的CPU接近极限,您通常可以通过访问声卡的控制软件并增加缓冲区大小来修复此问题。
Sample Rate also affects CPU usage. Going from 44100 to 88200 doubles CPU usage. This is an area about which there is much debate, but in reality few of us have ears that can really detect whether a track has been recorded at 44100 or at 88200. Test this out for yourself and see what you think.
采样率也会影响CPU使用率。从44100到88200,CPU使用量翻了一番。这是一个有很大争议的领域,但事实上,我们中很少有人有耳朵能真正检测到一条轨道是在44100还是在88200录制的。自己测试一下,看看你的想法。
Check also your MIDI Devices settings preferences. This has been covered in Chapter 1.
还要检查您的MIDI设备设置首选项。这已经在第一章中讨论过了。

琴韵晓波

沙发 琴韵晓波说：
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