if running cubase SX with a soundblaster live (yeah, I know it's crappy). using the built in directX full duplex driver setting in SX, the sync is lost after about 30 sec when recording audio live to harddrive, even with only a MIDI click track running. the multimedia driver works ok, but obviously has high latency. that's ok for just recording to midi clicks. however, the recorded audio is badly out of sync when played back - i thought Cubase adjusted for latencies for audio playback - shouldn't recorded audio be automatically lined up? have i set something up wrong? or do i just have to constantly go through and realign the audio segments?
odd latency problem?
- Thread starter jimmychan
- Start date
ActualTruth
New member
I'm using VST 3.7 with an SB live soundcard, and seem to be having a similar problem. Whenever I play something live, the program doesn't seem to pick it up until a few ms later. This can clearly be seen on the input bar in the EQ window.
I'm pretty new to this thing, and would greatly appreciate any help given.
I'm pretty new to this thing, and would greatly appreciate any help given.
Chris Tondreau
Member
AFAIK Soundblaster cards work strictly at a sample rate of 48khz. Cubase typically defaults to 44.1khz. This discrepancy can cause no end of pesky events. Change the settings in Cubase so that you are working at 48khz, and that should solve it.
Chris
Chris
Cubase SX is a big jump over the old VST's however latency is still a problem. With VST/32 I used to not only SEE the audio milliseconds later, but also HEAR it milliseconds later too. With SX though, I can hear the audio at the same time its being recorded. However, latency is all due to your computer. The soundcard is a big link in the latency issue. Upgrading your soundcard will improve your latency time, but will not make it go away. I too have this same problem, and Ive put up with having to line the audio up after every record. It IS a pain, but you get used to it. . . . . .eventually. . . . .and it becomes ever so slightly easier.
Chris Tondreau
Member
Latency is always a problem - depending on how you record. If you don't monitor through Cubase, then latency doesn't matter. If you DO monitor through Cubase, you'll be at the mercy of whatever latency your system can work with. Similarly, if you want to play VSTi's in real time, then you'll also be at the mercy of the same latency, but only inasmuch as what you hear played back will be delayed by the latency value. What is actually being recorded (ie. midi data) will line up perfectly to your click.
With audio, the same general idea applies. If you have a latency of, say, 100ms, then everything you hear back while monitoring through Cubase will be 100ms late. However, the audio signal goes INTO Cubase at the right time, so it should be in time. Latency is essentially a figure that describes how long it takes to process the in-coming signal and spit it back out again. You should not have to move any of your wavs so that they are in their proper time, unless they were not played as such in the first place - no matter how high your latency is. If you do, there is another problem somehwere.
As PsyCoNo said, latency is determined by a number of factors. To a small degree, the overall speed and efficiency of your system will affect it. The two biggest factors are the soundcard (and the accompanying drivers, of course...) you use, and where your buffer size is set to in Cubase. With my lowly Celeron 466 computer, and my good quality Delta44 soundcard, I can get as low as 8ms of latency, which is perfectly fine for monitoring through Cubase, or monitoring real time playing of VSTi's. This isn't a fully reliable figure, though, as adding more tracks into a project will require an adjustment to my buffer size, which means that I have to settle for higher latency. Once I've recorded about 20 tracks, I'm pretty much stuck with about 60ms of latency.
Chris
With audio, the same general idea applies. If you have a latency of, say, 100ms, then everything you hear back while monitoring through Cubase will be 100ms late. However, the audio signal goes INTO Cubase at the right time, so it should be in time. Latency is essentially a figure that describes how long it takes to process the in-coming signal and spit it back out again. You should not have to move any of your wavs so that they are in their proper time, unless they were not played as such in the first place - no matter how high your latency is. If you do, there is another problem somehwere.
As PsyCoNo said, latency is determined by a number of factors. To a small degree, the overall speed and efficiency of your system will affect it. The two biggest factors are the soundcard (and the accompanying drivers, of course...) you use, and where your buffer size is set to in Cubase. With my lowly Celeron 466 computer, and my good quality Delta44 soundcard, I can get as low as 8ms of latency, which is perfectly fine for monitoring through Cubase, or monitoring real time playing of VSTi's. This isn't a fully reliable figure, though, as adding more tracks into a project will require an adjustment to my buffer size, which means that I have to settle for higher latency. Once I've recorded about 20 tracks, I'm pretty much stuck with about 60ms of latency.
Chris
well yeah, I know that 441.khz has to be resampled to play properly at 48khz, presumably with some loss of quality (aliasing or something?). but if everything's recorded at 48khz when running cubase at 48khz it should be ok yeah? but will there be a significant loss of quality (by this i mean distortion or some such obvious thing) that would occur if you just use soundforge to reduce a final WAV from 48khz to 44.1khz to burn to CD?
Chris Tondreau
Member
You won't have any loss in quality when you convert down to 44.1khz., assuming you've done the conversion with a half decent converter. You also won't experience a lowering of pitch or a slowing down of the audio. That happens when you try to PLAY a 48khz file through a 44.1 setup - not when you convert the sample rate.
Even though you hear the soundcard's ouput mixed with the input, you still shouldn't have your audio all disaligned. If you're monitoring through Cubase, or if you've got global disable engaged and are monitoring through a mixer, it all amounts to a difference in time of what it takes to get from your soundcard to the hard drive. Not only inaudible, but I'd venture to say even barely measurable - even with your wave editor.
Chris
Even though you hear the soundcard's ouput mixed with the input, you still shouldn't have your audio all disaligned. If you're monitoring through Cubase, or if you've got global disable engaged and are monitoring through a mixer, it all amounts to a difference in time of what it takes to get from your soundcard to the hard drive. Not only inaudible, but I'd venture to say even barely measurable - even with your wave editor.
Chris