From 8bd3ae04a64573a9989027891b2518c7ca9c9fe0 Mon Sep 17 00:00:00 2001 From: Niklas Enbom Date: Mon, 24 Sep 2018 18:50:45 +0000 Subject: [PATCH] Revert "Bug in histogram metric reporting." This reverts commit 3a9731ff2f871bcc798ce9760fa51d81cfd5e755. Reason for revert: Seems to cause crashes in Chrome browser tests, see for example https://logs.chromium.org/logs/chromium/buildbucket/cr-buildbucket.appspot.com/8934487169011818016/+/steps/browser_tests__retry_with_patch_/0/logs/WebRtcBrowserTest.RunsAudioVideoWebRTCCallInTwoTabsOfferEcdsaAnswerEcdsa/0 Original change's description: > Bug in histogram metric reporting. > > A (actually several weeks) while ago, we noticed an error with the > WebRTC.Audio.Agc2.EstimatedNoiseLevel histogram. It always reported > the value 0. Here is why: > > The histogram bins go from 0 to 100. But the value logged is dBFS. It is > always less than or equal to 0. This CL changes the bins. > > Bug: webrtc:7494 > Change-Id: I45fd122e98f9396f9871bc965a708987bd1815f6 > Reviewed-on: https://webrtc-review.googlesource.com/101340 > Commit-Queue: Alex Loiko > Reviewed-by: Sam Zackrisson > Cr-Commit-Position: refs/heads/master@{#24800} TBR=saza@webrtc.org,aleloi@webrtc.org Change-Id: I84883f73710b7e13aa90ee29b140acfc417f109f No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:7494 Reviewed-on: https://webrtc-review.googlesource.com/101701 Reviewed-by: Niklas Enbom Commit-Queue: Niklas Enbom Cr-Commit-Position: refs/heads/master@{#24809} --- modules/audio_processing/agc2/adaptive_digital_gain_applier.cc | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/modules/audio_processing/agc2/adaptive_digital_gain_applier.cc b/modules/audio_processing/agc2/adaptive_digital_gain_applier.cc index 9d791ae234..f5342df829 100644 --- a/modules/audio_processing/agc2/adaptive_digital_gain_applier.cc +++ b/modules/audio_processing/agc2/adaptive_digital_gain_applier.cc @@ -83,7 +83,7 @@ void AdaptiveDigitalGainApplier::Process( RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.Agc2.DigitalGainApplied", last_gain_db_, 0, kMaxGainDb, kMaxGainDb + 1); RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.Agc2.EstimatedNoiseLevel", - input_noise_level_dbfs, -100, 0, 101); + input_noise_level_dbfs, 0, 100, 101); } input_level_dbfs = std::min(input_level_dbfs, 0.f);