Closed Bug 1406941 Opened 4 years ago Closed 3 years ago

Write unittest for configuring AudioConduit


(Core :: WebRTC: Audio/Video, enhancement, P3)

58 Branch



Tracking Status
firefox58 --- affected
firefox65 --- fixed


(Reporter: pehrsons, Assigned: dminor)


(Blocks 1 open bug)



(7 files)

Configure AudioConduit similar to how our signaling layer does it, and see that we call into a mocked encoder stack with the right values.

Repeat for different signaling properties, like fec and dtmf. See AudioCodecConfig.
Depends on: 1425039
Now that AudioConduit is using the Call interface this is much more tractable.
Assignee: nobody → dminor
With the branch 64 update we no longer configure packet size and rate
ourselves. Instead, we use the defaults provided in
This removes the unused fields from AudioCodecConfig, the next commit does the
same thing for JsepAudioCodecDescription.
Packet size and rate are no longer configured inside AudioConduit, so there is
no reason to continue to define them here. We now take the defaults provided

Depends on D12012
This was regressed by the branch 64 update. The parameter is used in

Depends on D12014
Rather than returning an error, the channel proxies have asserts that the
underlying calls to the channel objects succeeded.

Depends on D12015
This makes Init and DeleteChannels virtual and mRecvChannelProxy and
mSendChannelProxy protected. This will allow unit test code to override
the creation of channels so that we can use a mocked ChannelProxy instead.

Depends on D12016
Pushed by
Remove unused fields from AudioCodecConfig; r=padenot
Remove unused fields from JsepAudioCodecDescription; r=bwc
Fix typo in videoconduit_unittests.cpp; r=padenot
Set opus maxplaybackrate in AudioConduit; r=padenot
Remove unused error handling code from SetLocalRTPExtensions; r=padenot
Make AudioConduit more easily unit testable; r=padenot
Add unittests for configuring AudioConduit; r=padenot
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