I have been trying to solve a problem with my Audigy 2 ZS PCI card. If I record a sine wave at 16bit-48k and set the recording level so that the peaks just reach full amplitude (100% or 0VU), then without changing the recording level if I change to 16bit-44.1k and record, the peaks are about 5% lower. Normally, this wouldn't be much of an issue, but if I try to increase the recording level, the peaks clip at that level instead of the 0VU level one would expect. As I understand it, the Audigy initially digitizes at 48k. If you want to record at a lower rate, such as 44.1k, the card has a recording sample rate converter (SRC) that resamples the 48k signal down to 44.1k. In that process the audio level should remain constant, and it has in many other SoundBlaster cards I have worked with. But in my Audigy the level drops 5%. What I am seeing when I see clipping at 95% of full scale at 44.1k is the actual 0VU digital clipping that had occurred at the initial 48k sampling. The important thing to note is that this only happens with 16-bit recordings. 24-bit is okay. This becomes a problem because I cannot use the peak indicator on the VU meter to warn of clipping when recording at 16bit-44.1k. No matter the level, the peak indicator will never light because the audio level never reaches 0VU. I had tried other versions of Audigy 2 ZS drivers and one version actually made the problem worse, making 44.1k peaks clip at 90% of full scale. So this seems to be a driver issue related to the SRC process during recording. I have been communicating with Creative's Advanced Support Group on this problem, but they cannot duplicate it, so they can't offer any help. Here's how you can help: 1. There are about 25 files involved in the Audigy driver set. If you know the specific driver file(s) that affect recording SRC in the Audigy, please let me know. If I can isolate this to a file or two, perhaps Creative can solve my problem. 2. It would be very helpful if you would run a simple test if you have an Audigy, or any other SoundBlaster for that matter, and report back here. All you have to do is feed some audio in your line-in connector and begin recording at 16-bit, 48K (make sure of that setting). Increase the audio level until it clips heavily. Stop recording and look at your recorded waveform to verify that it is clipped at 0VU (100% of the possible amplitude). Now, without changing the recording level, change your recording setting to 16-bit, 44.1k and record again. Stop recording and look at the waveform. Does it clip at the same level, 0VU, as the 48k recording, or does it clip at a lower level, about 5 to 10% lower than before? I am using percentages because many recording software doesn't have a dB scale on the recorded waveform, but it is very easy to estimate by eye if a waveform is 5% or 10% lower. This proplem happens for me regardless of if I record from line-in or record a wave file being played back by Windows Media Player, so it has nothing to do with the analog audio level at the line-in connector.