By Martin WalkerChoosing 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. 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. Here are some thoughts on acceptable values for different recording purposes:. Vocals: This is the most difficult example, because anyone listening to their vocals in 'real time' will have headphones on, and therefore have the sounds 'inside their head'.
The only way they sound right is directly from Sonar.what gives? Perform-re sampling, if you turn this off, all of your audio gets slower than the original. About halfway down is a section with labelled 'File Bit Depths'.
A latency of even 3ms can be disconcerting in these conditions. Drums & Percussion: I suspect most drummers will prefer to work with latencies of 6ms or under, which should provide an 'immediate' response. Guitars: Electric guitarists generally play a few feet from their stacks, and since the speed of sound in air is roughly a thousand feet per second, each millisecond of delay is equivalent to listening to the sound from a point one foot further away. So if you can play an electric guitar 12 feet from your amp, you can easily cope with a 12ms latency. Keyboards: Even on acoustic pianos there's a delay between your hitting a key and the corresponding hammer hitting the string, so a smallish latency of 6ms ought to be perfectly acceptable to even the fussiest pianists. Famously, Donald Fagen and Walter Becker of Steely Dan claimed to be able to spot 5ms discrepancies in their performances, but the vast majority of musicians are unlikely to worry about 10ms, and many should find a latency of 23ms or more perfectly acceptable with most sounds, especially pads with longer attacks.Your application may support more than one driver format, in which case you've first got to decide on the best one.
You'll have the easiest time if you can choose ASIO drivers. Choosing A PC Audio Interface: SOS November 2004. DSP-assisted Audio Effects & Latency: SOS April 2004. Using Hardware Effects With Your PC Software Studio: SOS March 2004.
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PC Musician Jargon Buster: SOS February 2004. The Truth About Latency: SOS September 2002. The Truth About Latency Part 2: SOS October 2002. Hear No Evil: SOS August 1999. Mind The Gap: SOS April 1999So what's the best buffer size for your system? This isn't straightforward to answer. If you mainly play soft synths and soft samplers, or you're recording electric guitar, a 6ms ASIO or WDM/KS latency (256 samples at 44.1kHz) is probably low enough to be unnoticeable, and won't increase your CPU overhead too much.
However, if you're one of the many musicians who don't notice a 12ms latency, adopting a 512-sample buffer size at 44.1kHz will probably allow you more simultaneous notes and plug-ins.The best way to find out is to set a latency value of about 23ms (a buffer size of 1024 samples at 44.1kHz), and then choose a soft-synth sound with as fast an attack as possible (slow-attack pads can easily be played with latencies of over a second). See if you can detect any hesitation before each note starts, while you're playing in real time from a MIDI keyboard, and don't be embarassed if you can't! If you can detect a hesitation and you find this latency noticeable or even annoying, reduce the buffer to the next size down and try again, until you decide on a latency that works for you. This way you won't be wasting your processor's time by making it constantly fill unnecessarily tiny buffers.Whatever the latency value you choose, you may have to adopt a lower one when monitoring vocals during recording, if you want to add plug-in effects 'live'. Set the buffers inside your particular MIDI + Audio application to a high latency of about 23ms and then start playback of an existing song. Watch the application's CPU meter and note its approximate reading.
Now choose the next buffer size down and try again. If playback is still glitch-free, keep going, a setting at a time, until tell-tale clicks and pops start to appear. If you're lucky this won't happen until a 3ms or even lower latency, and may not even occur at the lowest available setting provided by your audio interface. But as soon as you hear even a single click, move back to the next buffer size up and try again, until you're sure that the current buffer setting is the lowest that is totally reliable. At the same time, note any increases in the CPU meter reading compared with its 23ms setting — if it's risen considerably, you'll probably find it preferable to only use this low latency during recording when you really need it for monitoring purposes, returning to your previously chosen playback setting at all other times.It's rare to run into any problems other than glitching when you're trying low ASIO or WDM buffer values. However, with MME and DirectSound drivers you may experience intense waveform breakup that can sometimes even crash your PC, so be very cautious when setting these driver buffer sizes below about 40ms. Change the value in small steps, and stop as soon as you start to experience glitching.Many musicians adopt a two-stage approach — a low latency value during the audio-recording phase and a more modest one during soft-synth recording, playback and mixdown, when they can add more plug-ins.
And while these procedures may sound complex, they should only take you an hour, at most, to complete, and you only need to perform them once with a particular combination of audio interface, driver version and PC. Once you've found the most suitable latency value for playback and the lowest possible latency supported by your particular PC, you'll know you're making the best use of your processor in all situations.
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