"Webrtc/ConnectionTest.Audio/0" is flaky |
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Issue description"Webrtc/ConnectionTest.Audio/0" is flaky. This issue was created automatically by the chromium-try-flakes app. Please find the right owner to fix the respective test/step and assign this issue to them. If the step/test is infrastructure-related, please add Infra-Troopers label and change issue status to Untriaged. When done, please remove the issue from Sheriff Bug Queue by removing the Sheriff-Chromium label. We have detected 12 recent flakes. List of all flakes can be found at https://chromium-try-flakes.appspot.com/all_flake_occurrences?key=ahVzfmNocm9taXVtLXRyeS1mbGFrZXNyKAsSBUZsYWtlIh1XZWJydGMvQ29ubmVjdGlvblRlc3QuQXVkaW8vMAw. Flaky tests should be disabled within 30 minutes unless culprit CL is found and reverted. Please see more details here: https://sites.google.com/a/chromium.org/dev/developers/tree-sheriffs/sheriffing-bug-queues#triaging-auto-filed-flakiness-bugs
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Feb 23 2017
Most of the flakes are on Linux and ChromeOS, so I'm disabling the test there, but leaving Win/MacOS enabled for now.
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Feb 23 2017
CL to disable on Linux/ChromeOS: https://codereview.chromium.org/2712693004/ Assigning to owner and taking out of the sheriff queue.
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Feb 23 2017
The following revision refers to this bug: https://chromium.googlesource.com/chromium/src.git/+/cb12cafde748de82eb55f491054eb2cd84fed63c commit cb12cafde748de82eb55f491054eb2cd84fed63c Author: treib <treib@chromium.org> Date: Thu Feb 23 14:09:48 2017 Disable flaky Webrtc/ConnectionTest.Audio on Linux/ChromeOS TBR=sergeyu@chromium.org BUG= 695384 Review-Url: https://codereview.chromium.org/2712693004 Cr-Commit-Position: refs/heads/master@{#452480} [modify] https://crrev.com/cb12cafde748de82eb55f491054eb2cd84fed63c/remoting/protocol/connection_unittest.cc
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Feb 24 2017
It appears that audio is sent at 64kbps, while we set target bitrate to 160kbps. Karl, is there a know issue with target bitrate being ignored by OPUS?
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Feb 24 2017
Looks like the problem is that x-google-min-bitrate parameter is ignored, and maxaveragebitrate should be used instead.
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Feb 25 2017
CC ossu@, who has been dealing a lot with send codecs lately.
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Feb 27 2017
The following revision refers to this bug: https://chromium.googlesource.com/chromium/src.git/+/6aaea61066c6df8a2bceead07fa433ae08ce365c commit 6aaea61066c6df8a2bceead07fa433ae08ce365c Author: sergeyu <sergeyu@chromium.org> Date: Mon Feb 27 19:29:44 2017 Deflake ConnectionTest.Audio test WebRTC now ignores x-google-min-bitrate parameter for opus codec, which was used in remoting. As result the audio was sent at 64kbps, instead of desired 160kbps. Updated webrtc_transport to use maxaveragebitrate to specify target bitrate, which should make the test non-flaky. BUG= 695384 Review-Url: https://codereview.chromium.org/2711053005 Cr-Commit-Position: refs/heads/master@{#453295} [modify] https://crrev.com/6aaea61066c6df8a2bceead07fa433ae08ce365c/remoting/protocol/connection_unittest.cc [modify] https://crrev.com/6aaea61066c6df8a2bceead07fa433ae08ce365c/remoting/protocol/webrtc_transport.cc
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Mar 17 2017
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Comment 1 by treib@chromium.org
, Feb 23 2017