Recently I came across a plugin for winamp that uses ASIO for music playback, apparently there are some benefits to using ASIO for this purpose in terms of improved sound quality as it uses a different method of playback to stop the windows sound management messing with it), but I am having problems getting it to work at all This is the plugin...(compatible with winamp 2 and 5 apparently) http://www3.cypress.ne.jp/otachan/out_asio(dll)_040asse.zip I copied the files into the winamp plugins folder, and configured it to use asio for DMX6Fire LT (my sound card, which does support ASIO), but whenever I'd go to play an mp3 it would just skip from one track straight the next without any sound at all. I am a bit of a n00b when it comes to things like ASIO, since I don't do any recording/music creation on my pc, so i was wondering if anyone had any advice on what I need to do to check asio is working, what settings I need to set up for the sound card in the control panel to make sure its using asio properly (disable sensaura3d?), or whether there are any more plugins that will do a similar thing (on the chance that this one is no good..).
First download Asio caps here.... http://www3.cypress.ne.jp/otachan/ASIOcaps.html Click on the caps output button, if your ASIO driver is working or installed you should get something that looks like this: If it brings up an error (which it should do in your case) try finding a ASIO driver on the DMX website. If that fails download and install the CuBase demo this will install a generic ASIO driver which will work with the WinAmp plugin.
You could also try the EXE version of the ASIO plugin. http://www3.cypress.ne.jp/otachan/out_asio(exe).html Its never worked for me but you never know
Cheers I tried the ASIO caps thing and it tells me I can only use 48000Hz for ASIO output (should I be able to use more sampling rates?) then I set the plugin to resample to 48000Hz (since my mp3s are 44kHz), and it works, but its very laggy (can't drag the winamp slider smoothly without the sound taking two seconds to start again from where I've effectively fast forwarded to)...very odd really, resampling is not good though surely? It does seem to sound slightly better but I'll have to have a good listen to see whether its really any different, but the lagginess is annoying me to say the least
Dunno might be a ASIO Latency problem. Look in your sound card driver control panel and see if you have a ASIO buffer setting, lower the better I think. IMO Its really not worth the hassle messing with all that ASIO stuff, you really wont hear the difference Heres some more info on ASIO Latency: [RIP] ASIO is a standard for audio device drivers created by Steinberg. As much as possible, ASIO bypasses the Windows or Mac operating system, creating a more efficient communication between the audio device and the software. Currently, all of the Steinberg programs use ASIO (of course), while other programs (including software synthesizers) have also adapted the standard. The question, "what's the latency" in this instance is only relevant if the program is ASIO-compliant and if the audio card's device drivers also contain ASIO drivers. Then, this question will need to be answered by the sound card company. Different sound cards will have different latencies at different sampling rates -- the higher the sampling rate, the lower the latency. In that sense, the latency occurs in numbers of samples, dependent on the number of samples that need to be put into a buffer before monitoring begins. Because the latency in samples is fixed or defined by the card, then the faster the sampling rate, the quicker a fixed number of samples will pass through the buffer. Hence, faster sampling rates = lower latencies. Often, a buffer size can be set in the sound card's control panel, and a lower buffer size = fewer samples that need to be buffered. As long as your system can handle the lower buffer size, lowest is best. The latency in this case comes into play when we are monitoring in a "tape type" fashion, which is essentially monitoring through the program. We'll discuss Windows latency later in this article, which also affects the user while monitoring, but only with the efficiency of ASIO are we able to achieve this type of monitoring. It goes something like this: While I have the program in 'input,' I hear my instrument from the inputs of the program much like a pro tape deck. If I 'roll tape,' or rather put the program into play, I no longer hear the instrument until I punch in, again like a pro tape deck. All of this, unlike a pro tape deck, occurs with a bit of latency between what you're playing and what you're hearing through the program. Steinberg says that 11 or 12 milliseconds of latency is acceptable. You can be the judge. At higher sampling rates, 3 ms latency might be possible. If you desire this type of monitoring, which is a fairly normal and accepted way of recording, then this may be the best that hard disk recording has to offer. [/RIP]
