I just bought a new LG Laptop running Windows 10, and I downloaded Audacity 2.4.1 onto it. When I import an audio track and push Play, the audio lags behind the playhead by about 0.17 seconds. I didn’t have that problem on previous computers, and I can’t figure out how to fix it. I hope it isn’t my computer because I don’t want to return it. Can someone advise me?
And as a side note, I have not updated Audacity on my other two computers to the latest version, and the audio lag on them is nil.
When I press play the cursor moves as if the audio is playing–but for a very brief fraction of a second no audio is playing. This is annoying because you cannot put the cursor on a position and press play and hear audio from that spot. I have to ‘back up’ a bit and then press play to make sure I hear the audio I need.
If I press play again from the same position then I can hear the audio from the start position.
Solved the issue for myself, updating for others to try:
My files were storing in my ‘onedrive/documents’ folder not the local folder on the computer itself. The lag was because the files were in the cloud and being accessed remotely.
I copied the audacity file and data folder from the cloud to the local drive and this is working without lag now.
Thank you. I moved an audio file from Dropbox (which is what I always use) to my desktop and then opened it up from there. But I still had the lag. Should I re-download Audacity, or is there another problem? Maybe it’s my computer since this is the first time it’s happened?
As SSEditing indicated, any form of “cloud” storage should be avoided when working with media editing as network delays and synchronization can cause a multitude of problems.
EverettVencel,
how long is the track that you are working with, and what is the sample rate? (see the “Hz” number in the info panel on the left hand end of the track).
You said you moved the audio file to your desktop–but where is the .aup file located? along with the data folder? Are those on the drive itself or cloud storage?
I appreciate you taking the time to help me work through this, Steve. Thank you.
I played around a little, and here are my results:
The shorter the buffer length, the shorter the lag (still only a minuscule difference though). That didn’t do the trick, so I played with the latency compensation, and that almost did it.
Right now, my buffer length is at 18 milliseconds and my latency compensation at -900 milliseconds. The lag is hardly noticeable, although because I have used Audacity for years I notice that something is still off.
I am a podcast engineer, and so I require speed (no lag) and quality. I think that changing the latency settings degraded the audio quality a bit-something I can’t have. I am at a loss for why this is happening on this particular computer.
Just checking if I am understanding the problem correctly:
If you generate a Rhythm Track and then press Play, do you hear each click before or after the play cursor reaches a peak? (We would expect the click to sound at the same time as the cursor is over the peak).
I hear the click after the play cursor passes the peak. When the buffer length is 100 milliseconds and the latency compensation is -150 milliseconds, the cursor is about 0.147 seconds past the peak by the time the sound comes out the speakers.
That suggests that the sound card has additional buffering, so that when Audacity sends audio to the sound card driver, it goes into a buffer, then works it’s way through the buffer, before eventually emerging through your speakers / headphones.
What happens if you set:
Buffer length: 0 milliseconds
Latency compensation: -130 milliseconds
The cursor moves very slowly (in 4 sec. it only moves 0.1 sec.), and the voice output is converted to loud static and nothing more. The lowest I could go before this started happening was 18 milliseconds buffer length.
Thanks for testing. That’s in line with what I was expecting.
This is weird. The “latency compensation” shouldn’t have any effect during playback
Do you see a difference in playback, between (for example)
buffer = 30ms, latency compensation = -900ms
and
buffer = 30ms, latency compensation = -90ms
Well, now I’m not so sure. If there was a difference in playback between the two settings, it was tiny. I think the -90ms resulted in slightly more lag. But perhaps my mind is just playing tricks on me.
One test I’ve been using is highlighting the first milliseconds of a recording and then listening to the playback. Any length under 0.017 seconds doesn’t come out the speakers. And if I generate a rhythm track and highlight and play just one click, I hear nothing, and those are 0.04 seconds long. I have to highlight a longer length if I want to hear something.
