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529131d3e4
Implement AnalogGainStatsReporter and add it in AudioProcessingImpl. This class computes statistics for analog gain updates and periodically reports them into a histogram. The added histograms for analog gain update statistics: - WebRTC.Audio.ApmAnalogGainDecreaseRate - WebRTC.Audio.ApmAnalogGainIncreaseRate - WebRTC.Audio.ApmAnalogGainUpdateRate - WebRTC.Audio.ApmAnalogGainDecreaseAverage - WebRTC.Audio.ApmAnalogGainIncreaseAverage - WebRTC.Audio.ApmAnalogGainUpdateAverage The histograms are defined in https://chromium-review.googlesource.com/c/chromium/src/+/3207987 Bug: webrtc:12774 Change-Id: I3c58d4bb3eb034a11c3f39ab8edb2bc67c5fd5e4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234140 Commit-Queue: Hanna Silen <silen@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35301}
67 lines
2.6 KiB
C++
67 lines
2.6 KiB
C++
/*
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* Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef MODULES_AUDIO_PROCESSING_AGC_ANALOG_GAIN_STATS_REPORTER_H_
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#define MODULES_AUDIO_PROCESSING_AGC_ANALOG_GAIN_STATS_REPORTER_H_
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#include "absl/types/optional.h"
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#include "rtc_base/gtest_prod_util.h"
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namespace webrtc {
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// Analog gain statistics calculator. Computes aggregate stats based on the
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// framewise mic levels processed in `UpdateStatistics()`. Periodically logs the
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// statistics into a histogram.
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class AnalogGainStatsReporter {
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public:
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AnalogGainStatsReporter();
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AnalogGainStatsReporter(const AnalogGainStatsReporter&) = delete;
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AnalogGainStatsReporter operator=(const AnalogGainStatsReporter&) = delete;
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~AnalogGainStatsReporter();
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// Updates the stats based on the `analog_mic_level`. Periodically logs the
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// stats into a histogram.
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void UpdateStatistics(int analog_mic_level);
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private:
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FRIEND_TEST_ALL_PREFIXES(AnalogGainStatsReporterTest,
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CheckLevelUpdateStatsForEmptyStats);
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FRIEND_TEST_ALL_PREFIXES(AnalogGainStatsReporterTest,
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CheckLevelUpdateStatsAfterNoGainChange);
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FRIEND_TEST_ALL_PREFIXES(AnalogGainStatsReporterTest,
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CheckLevelUpdateStatsAfterGainIncrease);
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FRIEND_TEST_ALL_PREFIXES(AnalogGainStatsReporterTest,
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CheckLevelUpdateStatsAfterGainDecrease);
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FRIEND_TEST_ALL_PREFIXES(AnalogGainStatsReporterTest,
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CheckLevelUpdateStatsAfterReset);
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// Stores analog gain update stats to enable calculation of update rate and
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// average update separately for gain increases and decreases.
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struct LevelUpdateStats {
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int num_decreases = 0;
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int num_increases = 0;
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int sum_decreases = 0;
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int sum_increases = 0;
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} level_update_stats_;
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// Returns a copy of the stored statistics. Use only for testing.
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const LevelUpdateStats level_update_stats() const {
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return level_update_stats_;
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}
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// Computes aggregate stat and logs them into a histogram.
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void LogLevelUpdateStats() const;
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int log_level_update_stats_counter_ = 0;
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absl::optional<int> previous_analog_mic_level_ = absl::nullopt;
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_AGC_ANALOG_GAIN_STATS_REPORTER_H_
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