Recording is slower and lower pitch than original

[whogburn]

I’ve used Audacity for years to record myself singing along with a track. I use wired ear buds from my Bose computer speaker and record with a Fifine usb microphone. I’m using Audacity version 3.4.2. In the last few weeks I’ve had an occasional problem that my voice is recording at a slower rate and lower pitch than the track I’m singing with. I’ve closed all other apps, rebooted my computer, and even uninstalled and reinstalled Audacity and I still have the same issue.

[DVDdoug]

In the last few weeks

Same microphone and computer as always?

If it’s just little off, there are a couple of possibilities -

1- There could be a mix-up between 44.kHz and 48kHz. That’s around 10% or around 2 notes/semitones. That’s rare but you can try changing from whatever your settings are to the other one.

2- The USB mic and your soundcard each have their own clock/oscillator (again to generate the 44.1kHz, or whatever sample rate). The clocks will never match perfectly and a “consumer soundcard” can be off by enough to cause problems.

If your microphone has a headphone jack, try playing/monitoring your backing track through it.

If not, the best (but not inexpensive) solution is a audio interface which can be used for recording and monitoring.

If you get one, it’s good to get one with zero-latency direct-hardware monitoring so you can monitor yourself without going through the computer, along with an optional backing track from the computer.

But if your microphone doesn’t have an XLR connection you’d need a different microphone.

Or there are USB “podcast mics” with headphone jacks.

[whogburn]

Thanks. I thing you solved the problem. I was able to find the settings for both my speakers and mic and the speakers were at 24 bit 48000 Hz and the mic was at 16 bit 48000 Hz. When I opened Audacity and found the settings both the Project Sample Rate and Default Sample rate were set at 44100 HZ I changed them both to 48000 Hz. The Default Sample Format was 16 bit and I changed that to 24 bit. I left the Buffer length at 100 milliseconds and the Latency compensation at -130. I tried recording the same song I was working on yesterday and it sounded perfectly synched (at least to my brain and ears).