Choosing ASIO drivers, where possible, should help you achieve the lowest latency, using the Control Panel window provided by your particular audio interface. Here you can see the Control Panels for the Echo (left) and Emu (right) ranges, as launched from the Cubase SX Device Setup window. If you're tempted to go and make a cup of tea in the gap between pressing a note on your keyboard and hearing it play on your soft synth, you need help, and quickly. The SOS Forums are still awash with queries from new PC musicians asking why they get a delay between pressing a key on their MIDI keyboard and hearing the output of a soft synth on their computer. Sometimes this delay may be as much as a second, making 'real time' performances almost impossible. Newcomers to computer music soon cotton on to the fact that this is because of 'latency' and 'buffer sizes', but are often left wondering just what to adjust and what the 'best' setting is. Setting the correct buffer size is crucial to achieving optimum performance from your audio interface: if it's too small you'll suffer audio clicks and pops, while if it's too large you'll encounter audible delays when performing in real time.
ASIO2KS is a generic ASIO driver for WDM kernel-streaming compliant soundcards on Windows 2000 and XP platforms. ASIO ( Audio Streaming Input Output ) was introduced by Steinberg to remedy the situation, that Windows had no means to stream audio-data from/to soundcards with the small amount of latency necessary for professional recording solutions. Cubase SX3, an ASIO-compatible host application, allows only one driver at a time to be selected in its Device Setup window, although it may be possible (using special drivers like ASIO4ALL, highlighted here) to persuade several interfaces to run with one driver.
The ideal setting can depend on quite a few different factors, including your particular PC and how you work with audio, while the parameters you're able to change, and how best to do it, can also vary considerably depending on which MIDI + Audio application you use. Let's start by briefly recapping on why software buffers are needed. Playing back digitised audio requires a continuous stream of data to be fed from your hard drive or RAM to the soundcard's D-A (digital to analogue) converter before you can listen to it on speakers or headphones, and recording an audio performance also requires a continuous stream of data, this time being converted by the soundcard's A-D (analogue to digital) converter from analogue waveform to digital data and then stored either in RAM or on your hard drive. No computer operating system can do everything at once, so a multitasking operating system such as Windows or Mac OS works by running lots of separate programs or tasks in turns, each one consuming a share of the available CPU (processor) and I/O (Input/Output) cycles.
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To maintain a continuous audio stream, small amounts of system RAM (buffers) are used to temporarily store a chunk of audio at a time. For playback, the soundcard continues accessing the data within these buffers while Windows goes off to perform its other tasks, and hopefully Windows will get back soon enough to drop the next chunk of audio data into the buffers before the existing data has been used up. Similarly, during audio recording the incoming data slowly fills up a second set of buffers, and Windows comes back every so often to grab a chunk of this and save it to your hard drive. If the buffers are too small and the data runs out before Windows can get back to top them up (playback) or empty them (recording) you'll get a gap in the audio stream that sounds like a click or pop in the waveform and is often referred to as a 'glitch'. If the buffers are far too small, these glitches occur more often, firstly giving rise to occasional crackles and eventually to almost continuous interruptions that sound like distortion as the audio starts to break up regularly. Making the buffers a lot bigger immediately solves the vast majority of problems with clicks and pops, but has an unfortunate side effect: any change that you make to the audio from your audio software doesn't take effect until the next buffer is accessed.
This is latency, and is most obvious in two situations: when playing a soft synth or soft sampler in 'real time', or when recording a performance. In the first case you may be pressing notes on your MIDI keyboard in real time, but the generated waveforms won't be heard until the next buffer is passed to the soundcard. You may not even be aware of a slight time lag at all (see 'Acceptable Latency Values' box), but as it gets longer it will eventually become noticeable, then annoying, and finally unmanageable. During recording, the situation is even worse, since you normally won't be able to hear the incoming signal until it has passed through the input buffers and reached the software application, then passed through a second set of output buffers on its way to the soundcard's D-A converter, and then to your speakers or headphones. 'Monitoring' latency is double that of playback latency. Some plug-ins add latency to the audio path, as revealed by this Plug-In Information window in Cubase SX. The window shows which plug-ins exhibit additional latency when used, and whether or not to automatically compensate for it.