That’s not unusual. It’s because by the time playback has stopped, the audio samples have not worked their way through the sound buffers.
Does you sound card work if you change the “host” to “WASAPI”? If it does, then that may allow you to play very short selections.
The funny thing is, my other computers will play a 0.17sec. audio clip just fine. It’s this computer that doesn’t do it. I rarely need to listen to such short samples, but it comes in handy when I am making precise edits in a song.
When I change the host to WASAPI and then push play, I get an message saying “Error opening sound device. Try changing audio host, playback device and the project sample rate.”
Just wondering – and as a disclaimer, I am not a computer expert, so this might be a silly question – would a solid state drive effect the playback differently than a HDD? I think my other computers have HDD, and this new one has only a SSD.
Another quick question: Does changing the buffer length and latency compensation affect the quality of the audio? I thought I noticed a difference when I changed those settings so drastically, but perhaps I heard wrong. Thanks
Different sound card with different drivers. The sound card “should” flush it’s playback buffer on stop (continue playing until the playback buffer is empty), but some sound cards don’t do that.
No, that’s not going to affect it. SSDs are preferable to HDDs because they access data much faster. A slow HDD can cause play/record problems when using very high sample rates and/or a lot of tracks in the project, but that is not going to be a problem with a SSD.
Normally no, but if the buffer length is too short, it can cause anything from stuttering / glitching to completely garbled audio. I generally leave the buffer size at 100, but if you need to make it smaller, check that recording and playback are smooth, find the smallest size that you can go with smooth recording and playback, then set the buffer a bit bigger than that (as a safety margin). For example, if the sound starts breaking up below 17 ms, try setting it to 25 ms.
Latency compensation is necessary if you are doing multi-track recordings, or using “punch and roll”. If you are not using either of them, then you can leave latency compensation at the default (-130). If you do multi-track recording, or use “punch and roll”, then you need to follow these instructions to set up latency compensation: Latency Test - Audacity Manual
Thanks, Steve, for all your help. I decided to return my LG laptop and get a different one that hopefully will have less latency. Thankfully Costco has a good return policy!