Today in "ffmpeg's AAC encoder is still a embarrassment and nobody should ever use it": OBS on Windows using it, at 320kbps, makes music sound like a low bitrate mp3 (metallic hi-hats, etc). No problem with a custom OBS build on Gentoo with libfdk_aac enabled, at *160* kbps.
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OBS doesn't let you pick a codec for streaming, but it *does* let you configure a "recording" output that actually goes to an RTMP stream, and *that* lets you pick codecs, so you can pick MP3 and get way better audio quality than with the ffmpeg AAC encoder (at a lower bitrate).
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Honestly it's kind of funny how the open source world has best-in-class h.264 encoding (x264) and worst-in-class AAC encoding (ffmpeg aac) and those two are most often used together.
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oh no, but that means the video game footage I upload to Twitter is doubly, possibly even triply bad-AAC-encoded
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would it be easy to fix?
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I don't know, but I do know that the reason it isn't getting fixed is that the ffmpeg dev who wrote that code no longer cares.
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What’s worse is that the videos always start streaming at even lower bitrates and then out of a sudden it jumps to a “better” quality. I still can’t understand why they can’t just buffer the thing for a couple more seconds. Like you said, it’s 2020.
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