All graphs for this bug:
  https://chromeperf.appspot.com/group_report?bug_id=625599

Original alerts at time of bug-filing:
  https://chromeperf.appspot.com/group_report?keys=agxzfmNocm9tZXBlcmZyFAsSB0Fub21hbHkYgICgkrmqsgsM,agxzfmNocm9tZXBlcmZyFAsSB0Fub21hbHkYgICgorfW6QoM,agxzfmNocm9tZXBlcmZyFAsSB0Fub21hbHkYgICg4ruVrAkM,agxzfmNocm9tZXBlcmZyFAsSB0Fub21hbHkYgICg4qfApwoM,agxzfmNocm9tZXBlcmZyFAsSB0Fub21hbHkYgICgkr2VpAoM,agxzfmNocm9tZXBlcmZyFAsSB0Fub21hbHkYgICg4qPZvAoM,agxzfmNocm9tZXBlcmZyFAsSB0Fub21hbHkYgICgkuzwswoM,agxzfmNocm9tZXBlcmZyFAsSB0Fub21hbHkYgICgkoX5ogoM,agxzfmNocm9tZXBlcmZyFAsSB0Fub21hbHkYgICg4uO65wkM,agxzfmNocm9tZXBlcmZyFAsSB0Fub21hbHkYgICgkta5vAoM,agxzfmNocm9tZXBlcmZyFAsSB0Fub21hbHkYgICgkrKVtAoM,agxzfmNocm9tZXBlcmZyFAsSB0Fub21hbHkYgICg0tjlowoM


Bot(s) for this bug's original alert(s):

webrtc-android-tests-nexus5

webrtc changelog:

772e322 Add config to prune low-priority TURN ports for creating connections When the flag prune_turn_ports is set, When a high-priority turn port becomes available, it will prune low-priority ones. The pruned port will not be used for creating connections locally and its candidates will not be sent over to the remove side (unless they have been sent before being pruned).
c1f28db Compensate for the LevelController gain in the IntelligibilityEnhancer
939d863 Keep track of the user-facing number of channels in a ChannelBuffer
176655c This helps recognize more network types and even if the "unknown" network type is not helpful for identifying the network type, it helps bind sockets to the network.
348cd33 This CL provides improved parameter tuning for the level controller as well as some further minor changes.
e44301a FrameBuffer2 now has Start/Stop methods.
e906969 Dispose audio source correctly in AppRTC Demo on Android.
048b9b9 Respect VP8.automaticResizeOn for MediaCodec.
d8f5d43 Disabling the performance unittests on Android for the level controller.
be9301f Added empty directory with myself as owner for new mixer.
9d2bc45 Revert of Workaround for clang bug http://llvm.org/PR28348. (patchset #1 id:1 of https://codereview.webrtc.org/2110043003/ )
dac4d77 Added an empty member to the union of rtc::Optional, so that it is always initializable.
7cd77e7 Revert of Add config to prune TURN ports (patchset #12 id:360001 of https://codereview.webrtc.org/2093623004/ )
7f4673c AndroidVideoCapturer: Remove unused member variable
822f09e Revert of Move RtcEventLog object from inside VoiceEngine to Call. (patchset #16 id:420001 of https://codereview.webrtc.org/1748403002/ )
9aa2093 Revert of Fix to make the start/stop functions for the Rtc Eventlog non-virtual. (patchset #2 id:40001 of https://codereview.webrtc.org/2107253002/ )
f268422 Revert of Fix for RtcEventLog ObjC interface (patchset #3 id:40001 of https://codereview.webrtc.org/2106103003/ )
79069b8 Fix for RtcEventLog ObjC interface
3ead781 This CL adds activation logic of the new APM level control functionality and exposes the functionality using the MediaConstraints.
1dd53e2 Add config to prune low-priority TURN ports for creating connections When the flag prune_turn_ports is set, When a high-priority turn port becomes available, it will prune low-priority ones. The pruned port will not be used for creating connections locally and its candidates will not be sent over to the remove side (unless they have been sent before being pruned).
fda88f4 Separate the JNI function that controls logging levels into two.
638fa16 New module for the adaptive level controlling functionality in the audio processing module
dd9f455 Add back P2PTransportChannel.Connect, for Chromium.
e612603 Start ICE connectivity checks as soon as the first pair is pingable.
ae02bea Workaround for clang bug http://llvm.org/PR28348.
f23efdc Fixing flakiness of CreateDataChannelAfterNegotiate.
2a2ce29 Add RTX codecs for codecs only supported by external encoder.
4d75f9e Fixing P2PTransportChannelTest.TestIceConfigWillPassDownToPort.
f39e10a Fixing flakiness of TestDetectUnresolvedProxy.
98c692d Remove unused constructor from VCMPacket.
aedfb00 Fix to make the start/stop functions for the Rtc Eventlog non-virtual.
7097718 Expanded error message for unexpected packet in the flaky test EndToEndTest.DecodesRetransmittedFrame*
1b868e3 Add GN targets for AppRTC Demo on Android.
1e2f1e5 Move RtcEventLog object from inside VoiceEngine to Call.
d5232ff Android: Camera2 implementation and tests for it.
4a7493f VideoCapturerAndroid: Remove unused function getCameraThreadHandler
7790e22 Reland: Remove global list of SRTP sessions.
b665ea9 Fixing race condition in IncomingVideoStream.
fe5afaa Revert "Pull out the PostFilter to its own NonlinearBeamformer API"
5f736d0 Constructor in Camera1Enumerator should be public.
88570c1 Pull out the PostFilter to its own NonlinearBeamformer API
7df0f95 Fix the turn and udp port type. The port type was not set if it was created on a shared socket.
327e348 Add override annotation to appropriate methods in Camera1Enumerator.
6bf7b98 Create a null decoder for h264 when not supported, instead of crashing.
603b40b Remove unused output parameter in VCMReceiver::FrameForDecoding().
8489874 Remove unused parameters from VCMReceiver::InsertPacket().
ee99359 Logging and tracing of audio devices on Andriod.
d6e7474 Fixing bug where Connection drops packets when presumed writable.
7dc4f29 Add virtual Initialize methods to PortAllocator and NetworkManager.
bc1f6bd Reland 2 of: Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver.
8d451be Using fake clock for TURN port tests and un-disabling some tests.
b2a3fa7 Disabling EndToEndTest.RestartingSendStreamPreservesRtpStatesWithRtx.
b9bcc84 Decrease the amount of maximum outstanding frames for Android HW H.264 decoder.
fd27336 Delete a method that was not used. This was a mistake from code merging.
10a5fd9 Replace OS_ANDROID define to disable webrtc test.
a3886cd Build webrtc_nonparallel_tests under GN.
6609904 Disable flaky ProcessNoLossChangeFrameRateFrameDropVP9 on Android
ea1d5f9 Fixing a comment on AEC divergence metric.
69e722e Make mixing test die earlier on failure instead of spamming errors
d5bebe4 Allow disabling capture to texture on Camera1Enumerator using constructor parameter.
105bb15 Remove a thread checker for a decoder thread from IncomingVideoStream. The specific decoder thread may vary when using VideoToolbox.
c2c83d8 Return both IPv6 and IPv4 address from the lookup. We currently only return IPv4 address, which may cause issues in IPv6 networks if we provide host name as the turn servers.
cb507c1 Fix stats hud in AppRTC Demo for iOS.
00dbf61 Revert of Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. (patchset #3 id:40001 of https://codereview.webrtc.org/2046173002/ )
ac8472e Cleanups in cricket::VideoFrame and cricket::WebRtcVideoFrame.
23492b9 Reland of: Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver.
a7ba978 Revert of Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. (patchset #3 id:40001 of https://codereview.webrtc.org/2046173002/ )
f29b3f1 Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver.
6811fb4 Revert of Add virtual Initialize methods to PortAllocator and NetworkManager. (patchset #4 id:60001 of https://codereview.webrtc.org/2097653002/ )
af5a5ef Add virtual Initialize methods to PortAllocator and NetworkManager.
0e15799 Reland of "Revert of Update the BWE when the network route changes. (patchset #5 id:180001 of https://… (patchset #1 id:1 of https://codereview.webrtc.org/2098703004/ )
632995d Revert of Revert "Revert of Update the BWE when the network route changes. (patchset #5 id:180001 of https://… (patchset #5 id:120001 of https://codereview.webrtc.org/2041593002/ )
cb624fd Revert "Revert of Update the BWE when the network route changes. (patchset #5 id:180001 of https://codereview.webrtc.org/2000063003/ )"
9ce8095 [Chromoting] DirectX based capturer should always return a temporary error
57972e7 Allow receiving a packet on a TURN port from an unknown address.
dce7cd8 Revert of Cleanups in cricket::VideoFrame and cricket::WebRtcVideoFrame. (patchset #5 id:110001 of https://codereview.webrtc.org/2075983003/ )

Nothing jumps out at me, but I don't know much about webrtc.

Nothing jumped out for me as well. I have talked to phoglund@ and apparently we can expect quite big changes when changing bots, changing to wont-fix.