feat(blink): refactor blink and improve call quality #376
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dariusc93
approved these changes
Dec 1, 2023
use Mutex instead of RwLock in host_media controller.
use futures::future::ready instead of async({}) in simple_webrtc
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What this PR does 📖
eliminates the problem where audio cuts out by doing the following
HeapRbused to pass data to/from CPALcreated a submodule to aid in testing
blink-replcan usewarp-blink-wrtcwith theloopbackfeature to make it send any received audio back to all connected peers. This program can be run on a cloud device, allowing one person to test a call between two separate locations.refactored warp-blink-wrtc
host_mediamodule. This was the next logical step after a previous PR improved IPFS signaling. The abstraction over the codec type was removed because it seems that we will always just use opus. The audio sink was split into a decoder task and a receiver task. The audio source was split into an encoder task and a sender task. Opus encoding/decoding is done on dedicated threads. The decoder task decodes in parallel using rayon.misc fixes
AudioDeviceConfigtrait to use aoneshot::channelto signal whentest_speakerandtest_microphoneare done.Cpalstreams can't be sent safely between threads so these functions, which create said streams, can't use async.test_speakerwhere the sound continues to play after the test if the program doesn't drop theAudioDeviceConfigImplstruct.Which issue(s) this PR fixes 🔨
Special notes for reviewers 🗒️
Additional comments 🎤
The stuff about datachannels isn't used yet and was left in for later. Eventually i'd like to use them to determine round trip time between peers, for detecting poor network conditions. That's what I was thinking of doing when I set out on this refactor.