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A Snap-Crackle-Pop sound when I playback in Sonar


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I just installed Sonar 3 on my new laptop and when I playback audio or midi with VSTi samples applied I hear a popping kind of sound. Kind of like an old record player but really loud and in regular intervals. It doesn't happen while playing through a silent section, only when there are notes. The sounds themselves sound just like always. This popping sound is kind of riding on top of it. According to the volume gauge, it isn't too loud, so I don't think this is clipping. Any idea what could be causing it?

Thanks in advance for your help.

Edit: I just exported a file to audio and when playing it in windows media player it doesn't have the noise. Interesting. It only happens to sound being played in Sonar.

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Assuming you're using PC, install and use ASIO drivers (Google ASIO4ALL and install it). I'll direct you further once you have installed the drivers.

The popping noise is associated with the buffer length when using a low sample rate (increasing gets rid of it, but you'll get a major delay in latency) at least in FL Studio it does that as well. ASIO is the way to go.

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Considering you're an OCRemixer, I would assume you know about under-runs, but that's what it sounds like. If you don't know, then go into your soundcard settings (preferably with some sort of ASIO interface) and increase the buffer size until it stops.

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Assuming you're using PC, install and use ASIO drivers (Google ASIO4ALL and install it). I'll direct you further once you have installed the drivers.

The popping noise is associated with the buffer length when using a low sample rate (increasing gets rid of it, but you'll get a major delay in latency) at least in FL Studio it does that as well. ASIO is the way to go.

Thanks a whole lot Brian. I haven't tried it yet, but thanks anyway just for giving me such detailed advice.

Go into your soundcard settings (preferably with some sort of ASIO interface) and increase the buffer size until it stops.

Thanks to you too, Zephyr.

Considering you're an OCRemixer, I would assume you know about under-runs, but that's what it sounds like.

Hahaha. Actually, I only have one mix up and I only wrote it. Another guy collaborated with me and did all the production part. So, the title of OCRemixer is a little deceptive in my case. I'm really only half an OCRemixer. Now I know what under-runs are. ^_^

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FYI: In SONAR, if you don't want to go with the ASIO drivers, the buffer settings for the default WDM drivers are in Options > Audio > General Tab. Just increase the number of "buffers in playback queue" and the "buffer size" and *poof* no more popping! As Brian said, there's no free lunch here so your latency will increase. I'd say that if you can keep the latency less than 10ms without popping, you and your system are doing well :)

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A note about using ASIO4ALL or ASIO in general. You can only run one instance at a time per device or channel. You cannot play audio in two or more applications at the same time. That means an application will take "ahold" of the asio driver during use and another application will not play (it will either not play, is muted, or gives you an error). You must release the driver by stopping the application from playing or opening the asio control panel and disabling it. Once you've freed up a slot, you can enable the other application. You'll need to do this back and forth between applications. Midi drivers will prevent ASIO from taking ahold of the program should they be active on startup. Disable the midi driver within the program, and reset the ASIO by disabling then enabling it again in the asio control panel. Disable Spdifout from the control panel as well if you're not using spdifout.

There is a workaround if you don't want to go back and forth between applications. http://www.cubase.net/phpbb2/viewtopic.php?t=101066

This will treat each ASIO instances independent of each other thus allowing you to play multiple applications at the same time (multiclient).

One last thing, ASIO doesn't have volume control unlike the Kmixer. You'll need to adjust the volume externally or within the program, not windows volume mixer.

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Any idea what could be causing it?

Hey Chipp,

Try going into 'Option>Global>Audio-data' and verify that the file bit-depth match the import bit-depth.

If you got an import depth of 24bit and you play a 16bit sample for example, audio artifact will likely happen.

The ideal would be to switch the import depth to 'original', so it would match any bit-rate of the sample you insert.

That's it for audio samples.

VST is another thing, but Harmony spoke the truth about that. ;-)

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Hey everybody, thanks for the help. Let me tell you what's been going on with this. Pardon my noobediness but it really is a mystery to me. On my old computer which is not NEARLY as powerful, the noise never occurred AND there was never any latency in the playback that I could notice. So it makes me wonder why, when I open up the audio options section, the playback buffer says that it's set to 105 milliseconds of latency. That should be TOTALLY noticeable right? But I don't notice any latency, just tons of popping. So I first tried out Harmony's method with the number and size of the buffers but no combination that I tried reduced the popping enough and the latency did increase a lot. Then I tried the ASIO method which DID reduce the noise by like 90% but still not completely no matter how high I set it and the latency doesn't change on any setting as well. What's going on? I want it yo be flawless like my old computer was. thanks again.

EDIT: FIXED? I set the driver mode to the only other remaining option of MME 32-bit which seems to have solved the problem. the latency says like 40ms but I don't notice any.

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Latency becomes an issue only if you're recording in real time. Everything looks very fluid when latency is very low. To a newbie if you're only sticking one note at a time then latency is probably undetectable and unimportant. Again, latency is only important when you need the computer to respond to you synchronously in real time when you input something, to provide real time feedback. With 40ms (close to 2048 samples in ASIO) it's undetectable, but during recording it's not. I usually keep it around one figure 1-9ms (around 128 samples or less) to minimize latency as much as possible.

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OK, so i fixed that first problem but now I have ONE more. I wonder if this one is fixable. when I line out from my sampler to the mic jack in my computer, it sounds like heck. but there are no other jacks to use. It's difficult for me to explain the exact difference in sound I'm hearing, but to put it simply it's just a major reduction in quality. One of the things that happens is that if the sound is too loud continuously, it gets compressed or something, but not just some. It gets reduced a LOT. but even if I turn it down really low, there is still a weird difference in the sound. It sounds perhaps more trebly and less bassy.

I read several articles on the net regarding line out to mic recording, but they all just talked about how much hotter mic inputs are. I don't think my problem is one of volume though. Any ideas?

1000 thanks for all your help.

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What OS are you using? It could be a sound card thing, but before that try this. Open the recording devices options (control panel -> Sound). Open Microphone / Line In properties. Click the Levels tab and move the slider in the mic / line in section to about 15 or so. Mic Boost to 0.0dB.

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