i am currently trying to unterstand how screen capturing with ffmpet works, i found some commands to use but i do not understand what most of the details actually do, but at the moment i have a problem getting the sound capturing to work.
I tried this command:
So it seems pulse is not recognized, so i tried another option i found, replacing pulse by hw:0
this kind of works, but i only get the video and no audio!
Whats the problem here, what audio option can i use, is there a way to find a list of them?
What i am trying to achieve is to capture the system audio (browser, music player and such) not my microphone!
Nominally pulse audio should be your default record device.
After launching ffmpeg with that, you may also need to go to pavucontrol and in the “recording tab” next to "ALSA plug-in [ffmpeg] : ALSA capture from " select your recording device. Else you may only get static.
" -i default " works for me, albeit the remainder of my ffmpeg line is different from yours, and I did not check out the rest of your ffmpeg line.
Yes this kind of worked, i had to change the device in pavucontrol though as you said.
I am not sure but it could be that there are some slowdown or slow motion effects, i need to investigate this further.
I do not now what it means, thats it.
Additionally i am sure now that there is a problem with the capturing, i am not sure if this effects the video two but the sound is kind of wobbly!
I thought that this could be related to the thread size message (which i was not able to change btw. i added it to the command but that did not had any effect), or could this be audio codec related?
I tried 15 fps but the wobbly (not stuttering!) audio is still there, and btw i have i pretty good cpu i think, it is a Intep Core i7-980X (6 cores (12 with HT)) and running ffmpeg barely shows any cpus usage.
But instead of the “pcm_s16le” acodec i tried “libvorbis” (with 30 fps) this time the audio was perfect i think, but there way one irritating warning at the beginning:
[libvorbis @ 0x176d900] Queue input is backward in time
and a lot of warnings of this type (about 50):
[matroska @ 0x176b3c0] Non-monotonous DTS in output stream 0:1; previous: 0, current: -1822; changing to 0. This may result in incorrect timestamps in the output file.
I am not sure if there is any delay or if the video/audio is asynchronous but these warnings irritate me.
Your questions are very ffmpeg specific. May I suggest you follow up on a communications channel that is specific to ffmpeg ? Possibly this link will offer some venues to follow up this question on : https://ffmpeg.org/contact.html