Which model of Scarlett are you using? Is the problem exactly the same with the two different microphones? Which mic were you using to record your 27sec test that you posted? Presumably you turned the phantom power off when using the V69 (which has its own power supply)?

Hey Audiokid and Boswell! It's so good to see some familiar faces! Thanks for taking the time to reply.

I woke up on Sunday morning and while I was still laying in bed all of a sudden it came back to me. This sounds like a bit rate problem. I just tried changing the DAW to 16 bit instead of 24 and it works perfectly. It's funny how those cobwebs get shaken off at the strangest time. 

Now I'm wondering if 16 bit is worse than 24 bit? Should I be recording at 16 or 24 or 32? I can't really remember what that's all about. But at least I know what the problem is.

ClarkJaman wrote:
Hey Audiokid and Boswell! It's so good to see some familiar faces! Thanks for taking the time to reply.

I woke up on Sunday morning and while I was still laying in bed all of a sudden it came back to me. This sounds like a bit rate problem. I just tried changing the DAW to 16 bit instead of 24 and it works perfectly. It's funny how those cobwebs get shaken off at the strangest time. 

Now I'm wondering if 16 bit is worse than 24 bit? Should I be recording at 16 or 24 or 32? I can't really remember what that's all about. But at least I know what the problem is.

Generally 24 bit is where you want to be at today.  Set your mic pre amp gain so averages around -18dbfs (usually around 1/2 way up the meter or so).  

There's no reason I can think of to record at 16bit. If you need to make a cd you can always just convert your file to 16bit.

Cheers!

Thanks Kmetal. It sure is nice to be back and see so many of the same stalwarts posting on here.

Ever since I started fooling around the bit rate I'm not having the problem any more, even when I change it back to 24 bits. So I guess for now I'll record at 24 bit 96khz. ¯\_(ツ)_/¯

My interface only has two options: 32 bit float and 64 bit float. If my interface is set to 32 bit float and my DAW project is set to 24bit will that potentionally cause problems?

If you are noticing this due to moving the sample rates I suspect it is still related to your buffer setting. Raise your buffer setting and I suspect it may go away.

I've never heard 16, 24, 32 or 64 bit sample rate create the sound you are experiencing but the sample rate is related to buffering. The sound you are hearing can happen when a higher sample rate is increased or you increase your processing requirements.

Thus, increasing the buffering setting improve the processing workflow.

What is your buffer setting?

ClarkJaman wrote:
Thanks Kmetal. It sure is nice to be back and see so many of the same stalwarts posting on here.

Ever since I started fooling around the bit rate I'm not having the problem any more, even when I change it back to 24 bits. So I guess for now I'll record at 24 bit 96khz. ¯\_(ツ)_/¯

Excellent!

ClarkJaman wrote:
Thanks Kmetal. It sure is nice to be back and see so many of the same stalwarts posting on here.

Ever since I started fooling around the bit rate I'm not having the problem any more, even when I change it back to 24 bits. So I guess for now I'll record at 24 bit 96khz. ¯\_(ツ)_/¯

Very cool, welcome back!!

I came to the party late, but would also have jumped into the buffer emptying! For me, 32 bit 48k is now my starting point. I cannot hear 96k differences and it makes the files twice as big. I’ve also never tried 64 bit. If you’ve got the latest driver running, then tweaking as suggested, the buffer sizes to suit your system seems a sensible, and often frustrating thing. The critical thing though is to do it in each of your common record and replay modes. Getting it right for recording two tracks at the same time while sending a headphone send might be different to the best one for mixing with loads of plugins and lots of tracks, so test both. You want to get it set, then forget all about it. One guy a few weeks back had multiple buffer settings for different things which must be hell! Welcome back.

paulears wrote:
I came to the party late, but would also have jumped into the buffer emptying! For me, 32 bit 48k is now my starting point. I cannot hear 96k differences and it makes the files twice as big. I’ve also never tried 64 bit. If you’ve got the latest driver running, then tweaking as suggested, the buffer sizes to suit your system seems a sensible, and often frustrating thing. The critical thing though is to do it in each of your common record and replay modes. Getting it right for recording two tracks at the same time while sending a headphone send might be different to the best one for mixing with loads of plugins and lots of tracks, so test both. You want to get it set, then forget all about it. One guy a few weeks back had multiple buffer settings for different things which must be hell! Welcome back.

Thanks Paulears! And good to see you again! :)

A couple of points:

1. The sampling rate for professional audio is measured in samples per second, and is independent of how many bits are in each sample. It is not a "bit-rate", which is a property of serial interfaces such as Ethernet and SPI.
2. Most modern audio interfaces have A-D converters that convert 24 bits in each sample (the wordlength). This is a fixed-point (integer) format, and the data is usually sent to the computer in that format. It is up to the DAW (with its drivers) whether it uses and stores the samples as 24-bit integers or converts them to 32-bit or 64-bit floating-point format prior to processing and storage. For D-A conversion, the data goes through a similar process in reverse. Some audio interfaces that have built-in processing (filtering, effects, etc) use a floating-point DSP internally, and it's possible they may have an option for sending the processed data to the computer in that format. However, all DAWs will transfer 24-bit integer data to and from an interface.