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3) is most certainly at play here, as Discord allows clients to set their preferred bitrate (RX&TX), which would not be possible in multi-party calls without re
by franga2000 7y ago
3) is most certainly at play here, as Discord allows clients to set their preferred bitrate (RX&TX), which would not be possible in multi-party calls without re-encoding.
- FractalParadigm 7y agoCould they not just drop the quality of the whole call down to the lowest bandwidth allowed by a user? I feel that would reduce a computational burden on Discord's end, while allowing the lowest client-to-client latency
- PudgePacket 7y agoThey could, but if you have 4 people in a call and 3 can receive high bandwidth audio, lowering it just for the 1 person with low bandwidth is the best user experience. Otherwise people with good networks who have their call quality dragged down will just think Discords voice chat is bad.
- duskwuff 7y agoKeep in mind that a major use case for Discord is open voice chats (e.g, for gaming groups), not just organized person-to-person calls. Having the quality for a whole chat drop just because someone joined from a mobile phone would be a really disappointing user experience.
- ollien 7y agoThis is absolutely something that Discord does, though. I've had friends just drop the bitrate slider as low as possible in Discord just to make the whole channel sound awful
- caffeinewriter 7y agoThat's a channel-specific option though. You can set per-channel bitrate, but it's not something Discord does automatically to accommodate lower throughput clients.
- deleted 7y ago[deleted]
- mmis1000 7y agoThe channel bitrate is just a client hint as far as i know (at least for now). Bots are able to send whatever bitrate they want to the channel, and other clients received as is. The server simply relay opus data without re-encode it.
- heydabop 7y ago> Discord allows clients to set their preferred bitrate (RX&TX) Where? All I see is setting bitrates on channels, but not on the client/app as a whole.
- jagger27 7y agoWouldn’t it be possible to send a stream cipher encrypted packet of audio to a hub server where the codec has a “progressive” decoding mode? If I’m not mistaken, Opus can already do something like this. That way a client could set their desired bitrate and the server would truncate the packets before passing them off and doesn’t need to be re-encoded. This only works with a stream cipher though. Maybe an audio engineer or cryptographer could chime in?
- caffeinewriter 7y agoIt looks like Ogg Vorbis has theoretical support in the spec for something called "Bitrate Peeling"[1], but there is no functional implementation for this yet, and there's been an open bounty on it since 2004.[2] This is a really neat idea though. Truncating the packet to change the bitrate per client without re-encoding. [1]: https://en.wikipedia.org/wiki/Bitrate_peeling https://en.wikipedia.org/wiki/Bitrate_peeling [2]: https://wiki.xiph.org/index.php?title=Bounties&diff=196&oldid=195 https://wiki.xiph.org/index.php?title=Bounties&diff=196&oldi...
- duskwuff 7y agoPeeling has been a goal for many audio/video codecs in the past. Nobody's been able to make it work acceptably, though -- either the low-bitrate version sounds awful, or the high-bitrate version increases in size to the point that it might as well just have a low-bitrate version alongside it.
- jagger27 7y agoSo in that case you could peel off whichever stream you don’t need, but you waste some upload bandwidth over simply negotiating a bitrate in advance.