audio_processing: Changed kMinDiffDelayMs from 50 to 60 ms
audio_processing: Changed kMinDiffDelayMs from 50 to 60 ms

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-Original-Commit-Position: 972c6d2dc6dd5efdad1377c0d224e03eb8f276f7@{#9540}
Cr-Mirrored-From: https://chromium.googlesource.com/external/webrtc
Cr-Mirrored-Commit: d92f2674d7e99ae69b26d434b47ea0adf14aff0c

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