[supermoo]

Hello, I've been using otachan's ASIO plug-in for WinAmp for a while now in conjunction with my X-Fi XM. There's a very annoying problem that I encounter all too often.

Some MP3s I try to play do not play at all when I use the ASIO plug-in (as in the elapsed play time stays at 00:00 and doesn't change) even though another song from the EXACT same album will play fine. This is getting really annoying as I have to constantly switch to the default K-Mixer to acutally play the song. I've only experienced this problem with MP3s so far and haven't noticed .flac files giving me any trouble.

Does anyone know of a solution to this?

Thanks in advance.

 

[Rempert]

If you did not make these mp3s yourself, it is possible that the mp3s which don't work are using some sample rate other than 44.1 khz (mp3 encoders are capable of downsampling to a few different sample rates to optimize sound quality at very low bitrates). With ASIO streams, you must have the right sample rate, because kmixer and the soundcard drivers will not do any resampling for you. If this is the problem, the solution would be to add a resampler to your DSP chain, and set it to 44.1 khz.

 

[Tabi]

The otachan ASIO output plugin has a built-in resampler.
Just set Resampling to "Enable" and use the correct sample rate for your soundcard.
You may also check the "convert 1 channel to 2 channels", otherwise mono files might not work.

 

[rb67]

Thanks for the info tabi, I have been having the same problem with my foobar2000 and some mp3's.

 

[supermoo]

This is odd because I ripped my CDs into flac and then converted a couple to mp3 (using EAC + Lame 3.97) with a sample rate of 44.1KHz and it still does it. Meanwhile adding a resampler (such as the built-in one in the ASIO plugin) would mean that some of my flac that came in 48KHz would also get resampled. I've tried using the resampler built-in to otachan's ASIO plugin and setting it to 44.1KHz (the same as the mp3s original sampling rate) but it still doesn't play the mp3s that didn't play before (but setting it to anything else, like 96KHz, will let it play).

I also noticed that regardless of the sampling rate of the mp3/flac file I played the master sampling rate (in Creative's Audio Creation mode) automatically detects and adjusts to it so wouldn't that eliminate the need to add the resampler into my chain?

I would love a solution where I can play the files that I currently can't play (they are encoded at 44.1KHz but won't play at that sampling rate for some reason) and not have to resample everything else in my playlist that plays fine. I hope that all made sense.

Thanks.

 

[supermoo]

Anyone? I really want to avoid resampling and definitely want to avoid having to use WinAmp's DirectSound. There must be a way to fix this without messing with the resampling rate isn't there?

 

[Joshatdot]

foobar

 

[supermoo]

From the sound of rb67's post it seems foobar has the same problem?

 

[EnOYiN]

What do you have against resampling if I may ask?

 

[supermoo]

Well I mean if I didn't mind resampling I would just stick with the kmixer/directsound in Windows. Since I'm using ASIO I would expect that it be used for its intended purpose which is to bypass the OS' mixer and listen to my music the way they were recorded (however "hot" they may be) and not have it altered. Even though the difference is somewhat small with my current setup I can't see why I should have to resample if I don't have to.

 

[EnOYiN]

Quote:

Originally Posted by supermoo /img/forum/go_quote.gif Well I mean if I didn't mind resampling I would just stick with the kmixer/directsound in Windows. Since I'm using ASIO I would expect that it be used for its intended purpose which is to bypass the OS' mixer and listen to my music the way they were recorded (however "hot" they may be) and not have it altered. Even though the difference is somewhat small with my current setup I can't see why I should have to resample if I don't have to.

I think you are confusing decrease of bit-depth with resampling. Resampling itself it not something that is harmfull for the SQ in any way. (when done right) The kmixer of windows decrease the bit-depth of a file when you lower the volume for instance. This will indeed decrease the dynamic range of the file. Well programmed resamplers will not do such things. They will only change the sampling rate from 44.1 to 48 kHz and will be able to do so without changing the SQ.

When recording music a lot of people resample the files to higher bitdepths and sampling rates than the original files just to use DSPs. The DSP will have a better effect on the music this way. This is not harmfull to the SQ at all.

So my advice would be to just use resamplers if you have the need for them. Unless you really want bit-perfect output. I can imagine that having bit-perfect output is beneficial when using an external receiver to play DTS files or maybe something else that would really benefit from getting the original file bit for bit. Otherwise you should just try to get the best sound out of your setup with as much DSPs as necessary.

This is just my opinion though. You can do as you like ofcourse and I am not forcing you to do anything.

 

[supermoo]

If using the resampling DSP in the ASIO plugin has no detrimental effects on the SQ then I won't mind using it. However what I have noticed is an increase in CPU usage and even as I type this now there are constant pauses before what I type shows up. Should this be happening when my cpu is a AMD 3700+? :\

 

[EnOYiN]

Quote:

Originally Posted by supermoo /img/forum/go_quote.gif If using the resampling DSP in the ASIO plugin has no detrimental effects on the SQ then I won't mind using it. However what I have noticed is an increase in CPU usage and even as I type this now there are constant pauses before what I type shows up. Should this be happening when my cpu is a AMD 3700+? :\

There will be no difference in SQ when using a well made resampler. There are however a lot of standard resamplers which do not really work that well and will use a lot of CPU time.

I am not sure about what program you are using. ( I am assuming winamp?) There are better resamplers than the standard one. You should try to look for another one with which you will be able to change the kind of resampling used.
i.e. one with a good interpolator, a medium interpolator and a fast interpolator.

Is the 3700 a dual or a single core? Afaik the 3700 should be running at about 2.4 GHz? ( not sure) It should be well enough to be able to run a resampler and play a computergame without any problems. Most likely the resampler you are using does not work well enough and once again I would like to advice you to try another encoder.

A medium or fast interpolator should be working well enough to use it for playback. The best interpolator is fun, but it does not provide any audible (to my ears - btw I can discern between FLAC and lossy) difference. The best sinc interpolator is mainly interesting for recording in studios to apply DSPs in my opinion. You can try it and how this will affect the CPU usage. If you can use it without a problem then by all means do.

If you can tell me what program you are using I am willing to look up a good resampler for you if you are not able to find one yourself.

 

[supermoo]

I'm using WinAmp 5.24 (kinda old but does the job) and am using otachan's ASIO plug in (http://otachan.com/). The settings of the resampler were Thread Priority: Highest, Sample rate: 48000Hz, and Quality: Ultra. Also the 3700+ is a single-core 2.2GHz cpu. If you know of a better resampler please share as I'd like to try them out. Thanks. :]

 

[EnOYiN]

I'll try to look something up. Might take a while though.