Add AudioProcessingCaptureStats and a level estimator replacement

This adds an interface for accessing stats on the capture stream, and
adds a level estimator to report one of the stats.

Bug: webrtc:9947
Change-Id: Id472534fa2e04d46c9ab700671f620584a246afb
Reviewed-on: https://webrtc-review.googlesource.com/c/109587
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25786}
This commit is contained in:
Sam Zackrisson
2018-11-26 16:18:25 +01:00
committed by Commit Bot
parent 2918d4e309
commit b24c00f02d
5 changed files with 109 additions and 40 deletions

View File

@ -283,6 +283,11 @@ class AudioProcessing : public rtc::RefCountInterface {
} adaptive_digital;
} gain_controller2;
// Enables reporting of |output_rms_dbfs| in webrtc::AudioProcessingStats.
struct LevelEstimation {
bool enabled = false;
} level_estimation;
// Explicit copy assignment implementation to avoid issues with memory
// sanitizer complaints in case of self-assignment.
// TODO(peah): Add buildflag to ensure that this is only included for memory