Closed
Bug 1425039
Opened 7 years ago
Closed 7 years ago
AudioConduit should use webrtc.org call interface
Categories
(Core :: WebRTC: Audio/Video, enhancement, P2)
Core
WebRTC: Audio/Video
Tracking
()
RESOLVED
WONTFIX
People
(Reporter: dminor, Assigned: dminor)
References
(Blocks 3 open bugs)
Details
The old interfaces on which we rely are going away and updating to the call interface will clean up the code and make it easier to write unit tests.
Assignee | ||
Updated•7 years ago
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Rank: 15
Assignee | ||
Updated•7 years ago
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Blocks: webrtc-call-quality
Assignee | ||
Comment 1•7 years ago
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The VoE interface is still present in branch 64 but has been removed from upstream, so this will need to be fixed after the branch 64 update at the latest.
Assignee | ||
Comment 2•7 years ago
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It is much easier to mock out the call interface than the the VoE* ones, so we need to fix this prior to or concurrently with writing gtests for the AudioConduit.
Assignee | ||
Comment 3•7 years ago
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It looks like we'll have to do this as part of the branch 64 update rather than after.
Status: NEW → RESOLVED
Closed: 7 years ago
Resolution: --- → WONTFIX
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Description
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