This week, I have gone back to using Audacity again. This morning, I had heard the crackling noise again while playing a little bit of a sound file that I was editing. I believe, in this instance, that this actually occurred when I had the Output Device set to the “HDA NVidia: ALC888 Analog (hw:0,0)” option. Seems to me that this crackling noise is more apt to happen whenever I do dissolve or crossfade type editing. It also seems more apt to happen whenever I play a small portion of a sound for a very short period, such as for no more than a couple of seconds.
Afterwords, I went to the link that you provided in your latest message, regarding the Pulse settings. I edited in the “PULSE_LATENCY_MSEC=30” and using the gedit program while logging in as a superuser. I am hoping that this remedies the situation.
The following is probably a different topic than what is discussed here, but I notice that, whenever I open up Audacity using a terminal program, I do see numerous errors listed. Please let me know if you want to see this text. I have it stored in the form of a text file.
If using the (hw:0,0) device you can try different settings in “Audio to buffer” in Audacity’s Recording Preferences (that setting affects playback too).
Make sure the project rate bottom left of Audacity is the same as the sample rate of the tracks, or there will be resampling, which might lead to break up. You can set faster real-time sample rate conversion in Quality Preferences if that is the issue.
Make sure there is no DC offset in the audio before editing, because DC offset can cause clicks. The Normalize effect can correct DC offset.
That will only help if using the “pulse” or “default” playback device.
If you see messages like these it is probably not a problem.
If you use the pulse playback device and change the “PULSE_LATENCY_MSEC” value, but see underruns noted in the terminal when you play audio, try a higher PULSE_LATENCY_MSEC value.
Under “Latency,” I see that “Audio to buffer” is currently set to 100 milliseconds, and Latency correction is set to -130 milliseconds.
The Project Rate (Hz) is currently set to 44100.
Some of the audio sources that I use are from video files. What would be your suggestion for using these sources? Maybe converting them to a certain audio format, perhaps?
I will also check out the link that you have provided regarding the DC offset. I think I know what you are talking about regarding the “Normalize,” as I have seen it listed in the “Effect” menu.
EDIT: I just went to the link that you provided. Once in a while, I do come across such audio, in which the sound is not right in the center, as shown in the image on that page that you have linked to. Until I had read the text on that page, I had no idea what causes certain audio files to have this “off-center” appearance when looking at the sound files this way with Audacity.
Only “Audio to buffer” might be relevant, not latency correction. 100 ms is the default, so generally that value should be OK on Linux. If it was relevant, increasing above 100 ms might help crackly/clicky playback.
In that case to avoid resampling during playback, the tracks also need to be 44100 Hz.
Converting will degrade them unless it’s to a lossless format like WAV or FLAC.
Audio in video files is often at 48000 Hz sample rate. In that case you should make sure the project rate is 48000 Hz.
DC Offset could well be the explanation of some of your issues, but that would tend to be a single click, not crackle.
FLAC is the format that I usually convert audio to when using Audacity. It is a nice compromise between file size and quality. I can make lossless files that are smaller in file size. And, unlike WAV, I can even add tag information.
That’s something that I can consider. I had never thought of doing that before. I was already aware of the 48000 Hz number for videos, however, particularly for the VOB format.
Among the video formats that some of the files are in that I use as a source, are FLV, MP4, AVI, sometimes MPG, and once in a while, VOB. A few of them are also the newer WEBM format. I’ve used a few MKV files as sources as well.