Roll src/third_party/webrtc 94079f84523b..1ff16c87aa63 (2 commits)

git log 94079f84523b..1ff16c87aa63 --date=short --no-merges --format='%ad %ae %s'
2019-05-20 Add RtpSenderInterface.SetStreams
2019-05-20 AudioEncoderOpus: Don't mix up sample rate and RTP timestamp rate

Created with:
  gclient setdep -r src/third_party/webrtc@1ff16c87aa63

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Change-Id: I791dcd9ce8345f471155eb0da8bfbe55b0d718e0
Reviewed-by: chromium-autoroll <>
Commit-Queue: chromium-autoroll <>
Cr-Commit-Position: refs/heads/master@{#661484}
1 file changed