Marth64 changed the topic of #ffmpeg to: Welcome to the FFmpeg USER support channel | Development channel: #ffmpeg-devel | Bug reports: https://ffmpeg.org/bugreports.html | Wiki: https://trac.ffmpeg.org/ | This channel is publically logged | FFmpeg 7.1 is released
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<lemondevxyz> Hello, I wanna use ffmpeg to send raw opus audio to Discord but I keep hearing subtle yet noticable audio artifacts with my audio input. I have tried using a pipe as an input, using a file as an input but to no avail
<lemondevxyz> I have worked before on using the libopus directly and it worked wonderfully but I want to use ffmpeg because importing C libraries in the language I am working in is a hassle.
<lemondevxyz> Here's the ffmpeg command I currently use wrapped in backquotes: `ffmpeg -i - -ar 48000 -ac 2 -acodec libopus -b:a 96K -map 0:a:0 -f data -application audio -frame_duration 20 -vbr off pipe:1`
<lemondevxyz> Another quirk I noticed is that if I read anymore than 240 bytes, the audio becomes completely unrecognizeable. I've tried 241 and 239 and both sound like records being scratched. I tinkered with the amount by using the original byte size I used, it originally was 960*2 (2 audio channels) and I kept halfing the value until it worked...
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<lemondevxyz2> Update: I haven't figured out the issue with ffmpeg but I found a solution. I have resorted to using ffmpeg as a universal decoder and encoding the result as ogg and using a native Golang ogg decoder and reading the opus frames directly from the Golang ogg decoder.
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<noobaroo> I'm wondering something, is there a way to specify areas of a to target high quality and specify which areas to disregard quality? Like a dynamic QP (CRF) setting? I've seen options before (although never used them) to set path to a film grain file or something like that, it would be cool if maybe something like this could be used to preserve specific areas as more clear than the rest?
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<noobaroo> lemondevxyz I think I'm behind your level so no help from me, but could you share the opus binary command you use that you feel is better?
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<grib> noobaroo: (assuming i understand your question) the two-pass variable bitrate encoding already does something a lot like that -- pass 1 decides where to target/disregard quality, pass 2 varies the Q based on pass 1
<grib> you could presumably edit or create the pass 1 results yourself
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