Hi all, first post and looking for some advice please.

I have just purchased my first MIDI controller (Edirol PCR-M50) and was having problems with a noticeable delay from striking the key to the sample being played on my PC. I took some advice on another forum to change my sample rate and buffer size which helped out a lot. However, I do not know what the effects of this will be on my sound quality. What does changing these settings actually do?

Secondly, I currently only have on-board sound and need advice on a decent sound card to slot into my PC. The processor and memory are fine, but the sound is letting me down.

I'm on a tight budget so what do you recommend?

From what i understand the latency doesnt lower the quality in your sound quality. the sound card i have( sb audigy ) is excellent onboard effects, remote, external box, midi everything and the asio drive for that is amazing....i would go asio if i were you...smooth streaming


oh hows that edirol controller...i was looking into one of them?

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2 MINDS ARE BETTER THAN 1
http://www.keesimo.com

www.asio4all.com

this is a universal asio driver try it out its good.

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An ASIO driver also takes the stress of work off your computer's processor.

Buffer size sounds like your enemy for sure. The lower the buffer size=less latency in your midi keyboard. Adjust it to lowest buffer size, where you do not hear noticeable pops and clicks.

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The Edirolr is really good. Well worth the money and a good safe buy for my first ever controller.

I've managed to get over my problem by buying an M-Audio Audiophile sound card. What a beast! Thanks for the help everyone.

So I need a sound card for good quality? I'm playing with a korgx50 for controller via usb chord only. I've got terrible sound quality and I'm also experiencing the lag Mulrooney mentioned. Noobalert! Noobalert! lol

Sound cards that natively support ASIO drivers are the way forward when experiencing latency (a delay between striking your note and hearing the sound).

For those with inbuilt cards, REFILL kings post should help, but for those wishing to have an ultra low latency, should consider getting a decent soundcard.

Not only do decent soundcards provide low latency, but you can mix in higher sample rates and better bit depths. I'd truly reccommend it if you are taking a step forward.

Check here for some ideas:

http://www.dv247.com/sound-cards/

I personally am using an M-Audio card with latency as low as 12ms @96khz....

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Some more pointers...

Sample Rate:

The amount of 'snapshots' the computer records ever second. You all probably know that CD's are recorded at 44,100 samples per second. Simply put the higher the sample rate, the better the quality in general.

The sampling frequency must be at least twice as high as the highest frequency that you wish to reproduce because you must have at least 1 data point for each half cycle of the audio waveform, which is WHY CD's are recorded at 44.1k - the audible range of human being is commonly quoted as from 20Hz to 20,000 Hz, which in fact will vary with age, health, past exposure to noises, and so forth.

Bit depth:

If the Sample rate is seen as samples taken a horizontal line (time), then Bit Depth is basically the vertical equivalent (amplitude). This is extremely important in terms of dynamic range.

Using a higher Bit Depth with a higher sampling rate yeilds better quality audio.

Latency / Buffer Sizes:

Let's start by going over why software buffers are needed:

Playing back digitised audio requires a continuous and uninterrupted 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 your speakers or headphones.

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.

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.

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', like we do in Reason, or when recording a performance. In the first case you may be pressing notes on your MIDI keyboard in real time, but the generated sound 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 but as buffer sizes are increased to give you stable audio, the latency gets longer - it will eventually become noticeable, then annoying, and finally unmanageable, especially on built in soundcards that are not designed to have low buffer sizes.

Plug-in effects can also add their own processing latency, particularly compressors, limiters and de-essers that look ahead in the waveform to spot peaks (specifically the "4ms lookahead" on the mastering compressor within Reason).

My advice then, is if your sound card gives you the option of changing buffer sizes, don't be afraid to tweak these a little. The lower the better, but remember, place it too low and your audio will click pop and glitch, or just become completely unstable - especially the more you throw at your computer - so the more powerful your computer, the better.

Try and get as much going on as you can to max out your CPU, and get your buffer to match it at it's highest workload. There's no point in placing an obscenely low buffer size, just for you to add another instrument or track, and for it to start glitching when your CPU starts working harder and fails to catch up. Remember - The higher the buffer, the more latency you will get, so try and find the sweet spot for your sound card and computer.

My personal settings have a buffer size of 128-256, with around 6-12ms latency.

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