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audio_processing: Changed kMinDiffDelayMs from 50 to 60 ms

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The UMA histograms WebRTC.Audio.AecSystemDelayJump and WebRTC.Audio.PlatformReportedStreamDelayJump triggers if the jump is larger than kMinDiffDelayMs.
Especially WebRTC.Audio.AecSystemDelayJump is sensitive around 50 ms differences, since the granularity is 4 ms and we can get a significant amount of hits at 52 ms.
Therefore, a change to 60 ms can make the logging more robust. The effect of not logging jumps in the interval 50-60 ms is of minor importance since they are not likely to affect the AEC performance. It's when we get values from ~100 ms and above that we should be worried.

Tested with a local ToT Chromium build where 52, 64 and 200 ms jumps were forced.

BUG=488124
TBR=henrik.lundin@webrtc.org

Review URL: https://codereview.webrtc.org/1208313003.

Cr-Commit-Position: refs/heads/master@{#9540}
This commit is contained in:
Bjorn Volcker 2015-07-05 10:46:01 +02:00
parent 72a8cee425
commit d92f2674d7
1 changed files with 3 additions and 1 deletions
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4
webrtc/modules/audio_processing/audio_processing_impl.cc
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@ -1000,7 +1000,7 @@ void AudioProcessingImpl::InitializeBeamformer() {
}
void AudioProcessingImpl::MaybeUpdateHistograms() {
static const int kMinDiffDelayMs = 50;
static const int kMinDiffDelayMs = 60;
if (echo_cancellation()->is_enabled()) {
// Detect a jump in platform reported system delay and log the difference.
@ -1024,6 +1024,8 @@ void AudioProcessingImpl::MaybeUpdateHistograms() {
100);
}
last_aec_system_delay_ms_ = aec_system_delay_ms;
// TODO(bjornv): Consider also logging amount of jumps. This gives a better
// indication of how frequent jumps are.
}
}