Add AnalogGainStatsReporter to compute and report analog gain statistics
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}
This commit is contained in:
Hanna Silen
WebRTC LUCI CQ
parent
3041eb21e9
commit
529131d3e4
@ -204,6 +204,7 @@ rtc_library("audio_processing") {
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"aec_dump:aec_dump",
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"aecm:aecm_core",
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"agc",
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"agc:analog_gain_stats_reporter",
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"agc:gain_control_interface",
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"agc:legacy_agc",
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"capture_levels_adjuster",
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@ -41,6 +41,20 @@ rtc_library("agc") {
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absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
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}
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rtc_library("analog_gain_stats_reporter") {
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sources = [
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"analog_gain_stats_reporter.cc",
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"analog_gain_stats_reporter.h",
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]
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deps = [
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"../../../rtc_base:gtest_prod",
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"../../../rtc_base:logging",
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"../../../rtc_base:safe_minmax",
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"../../../system_wrappers:metrics",
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]
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absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
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}
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rtc_library("clipping_predictor") {
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sources = [
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"clipping_predictor.cc",
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@ -142,6 +156,7 @@ if (rtc_include_tests) {
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testonly = true
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sources = [
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"agc_manager_direct_unittest.cc",
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"analog_gain_stats_reporter_unittest.cc",
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"clipping_predictor_evaluator_unittest.cc",
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"clipping_predictor_level_buffer_unittest.cc",
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"clipping_predictor_unittest.cc",
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@ -152,6 +167,7 @@ if (rtc_include_tests) {
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deps = [
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":agc",
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":analog_gain_stats_reporter",
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":clipping_predictor",
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":clipping_predictor_evaluator",
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":clipping_predictor_level_buffer",
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@ -161,6 +177,7 @@ if (rtc_include_tests) {
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"../../../rtc_base:checks",
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"../../../rtc_base:rtc_base_approved",
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"../../../rtc_base:safe_conversions",
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"../../../system_wrappers:metrics",
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"../../../test:field_trial",
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"../../../test:fileutils",
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"../../../test:test_support",
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130
modules/audio_processing/agc/analog_gain_stats_reporter.cc
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130
modules/audio_processing/agc/analog_gain_stats_reporter.cc
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@ -0,0 +1,130 @@
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/*
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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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#include "modules/audio_processing/agc/analog_gain_stats_reporter.h"
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#include <cmath>
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#include "rtc_base/logging.h"
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#include "rtc_base/numerics/safe_minmax.h"
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#include "system_wrappers/include/metrics.h"
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namespace webrtc {
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namespace {
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constexpr int kFramesIn60Seconds = 6000;
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constexpr int kMinGain = 0;
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constexpr int kMaxGain = 255;
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constexpr int kMaxUpdate = kMaxGain - kMinGain;
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float ComputeAverageUpdate(int sum_updates, int num_updates) {
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RTC_DCHECK_GE(sum_updates, 0);
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RTC_DCHECK_LE(sum_updates, kMaxUpdate * kFramesIn60Seconds);
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RTC_DCHECK_GE(num_updates, 0);
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RTC_DCHECK_LE(num_updates, kFramesIn60Seconds);
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if (num_updates == 0) {
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return 0.0f;
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}
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return std::round(static_cast<float>(sum_updates) /
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static_cast<float>(num_updates));
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}
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} // namespace
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AnalogGainStatsReporter::AnalogGainStatsReporter() = default;
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AnalogGainStatsReporter::~AnalogGainStatsReporter() = default;
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void AnalogGainStatsReporter::UpdateStatistics(int analog_mic_level) {
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RTC_DCHECK_GE(analog_mic_level, kMinGain);
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RTC_DCHECK_LE(analog_mic_level, kMaxGain);
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if (previous_analog_mic_level_.has_value() &&
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analog_mic_level != previous_analog_mic_level_.value()) {
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const int level_change =
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analog_mic_level - previous_analog_mic_level_.value();
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if (level_change < 0) {
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++level_update_stats_.num_decreases;
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level_update_stats_.sum_decreases -= level_change;
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} else {
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++level_update_stats_.num_increases;
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level_update_stats_.sum_increases += level_change;
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}
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}
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// Periodically log analog gain change metrics.
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if (++log_level_update_stats_counter_ >= kFramesIn60Seconds) {
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LogLevelUpdateStats();
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level_update_stats_ = {};
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log_level_update_stats_counter_ = 0;
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}
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previous_analog_mic_level_ = analog_mic_level;
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}
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void AnalogGainStatsReporter::LogLevelUpdateStats() const {
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const float average_decrease = ComputeAverageUpdate(
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level_update_stats_.sum_decreases, level_update_stats_.num_decreases);
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const float average_increase = ComputeAverageUpdate(
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level_update_stats_.sum_increases, level_update_stats_.num_increases);
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const int num_updates =
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level_update_stats_.num_decreases + level_update_stats_.num_increases;
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const float average_update = ComputeAverageUpdate(
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level_update_stats_.sum_decreases + level_update_stats_.sum_increases,
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num_updates);
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RTC_DLOG(LS_INFO) << "Analog gain update rate: "
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<< "num_updates=" << num_updates
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<< ", num_decreases=" << level_update_stats_.num_decreases
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<< ", num_increases=" << level_update_stats_.num_increases;
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RTC_DLOG(LS_INFO) << "Analog gain update average: "
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<< "average_update=" << average_update
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<< ", average_decrease=" << average_decrease
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<< ", average_increase=" << average_increase;
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RTC_HISTOGRAM_COUNTS_LINEAR(
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/*name=*/"WebRTC.Audio.ApmAnalogGainDecreaseRate",
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/*sample=*/level_update_stats_.num_decreases,
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/*min=*/1,
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/*max=*/kFramesIn60Seconds,
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/*bucket_count=*/50);
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if (level_update_stats_.num_decreases > 0) {
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RTC_HISTOGRAM_COUNTS_LINEAR(
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/*name=*/"WebRTC.Audio.ApmAnalogGainDecreaseAverage",
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/*sample=*/average_decrease,
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/*min=*/1,
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/*max=*/kMaxUpdate,
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/*bucket_count=*/50);
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}
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RTC_HISTOGRAM_COUNTS_LINEAR(
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/*name=*/"WebRTC.Audio.ApmAnalogGainIncreaseRate",
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/*sample=*/level_update_stats_.num_increases,
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/*min=*/1,
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/*max=*/kFramesIn60Seconds,
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/*bucket_count=*/50);
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if (level_update_stats_.num_increases > 0) {
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RTC_HISTOGRAM_COUNTS_LINEAR(
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/*name=*/"WebRTC.Audio.ApmAnalogGainIncreaseAverage",
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/*sample=*/average_increase,
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/*min=*/1,
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/*max=*/kMaxUpdate,
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/*bucket_count=*/50);
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}
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RTC_HISTOGRAM_COUNTS_LINEAR(
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/*name=*/"WebRTC.Audio.ApmAnalogGainUpdateRate",
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/*sample=*/num_updates,
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/*min=*/1,
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/*max=*/kFramesIn60Seconds,
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/*bucket_count=*/50);
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if (num_updates > 0) {
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RTC_HISTOGRAM_COUNTS_LINEAR(
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/*name=*/"WebRTC.Audio.ApmAnalogGainUpdateAverage",
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/*sample=*/average_update,
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/*min=*/1,
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/*max=*/kMaxUpdate,
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/*bucket_count=*/50);
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}
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}
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} // namespace webrtc
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67
modules/audio_processing/agc/analog_gain_stats_reporter.h
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67
modules/audio_processing/agc/analog_gain_stats_reporter.h
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@ -0,0 +1,67 @@
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/*
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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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@ -0,0 +1,161 @@
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/*
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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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#include "modules/audio_processing/agc/analog_gain_stats_reporter.h"
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#include "system_wrappers/include/metrics.h"
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#include "test/gmock.h"
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namespace webrtc {
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namespace {
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constexpr int kFramesIn60Seconds = 6000;
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TEST(AnalogGainStatsReporterTest, CheckLogLevelUpdateStatsEmpty) {
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AnalogGainStatsReporter stats_reporter;
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constexpr int kMicLevel = 10;
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stats_reporter.UpdateStatistics(kMicLevel);
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// Update almost until the periodic logging and reset.
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for (int i = 0; i < kFramesIn60Seconds - 2; i += 2) {
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stats_reporter.UpdateStatistics(kMicLevel + 2);
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stats_reporter.UpdateStatistics(kMicLevel);
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}
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EXPECT_METRIC_THAT(metrics::Samples("WebRTC.Audio.ApmAnalogGainUpdateRate"),
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::testing::ElementsAre());
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EXPECT_METRIC_THAT(metrics::Samples("WebRTC.Audio.ApmAnalogGainDecreaseRate"),
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::testing::ElementsAre());
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EXPECT_METRIC_THAT(metrics::Samples("WebRTC.Audio.ApmAnalogGainIncreaseRate"),
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::testing::ElementsAre());
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EXPECT_METRIC_THAT(
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metrics::Samples("WebRTC.Audio.ApmAnalogGainUpdateAverage"),
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::testing::ElementsAre());
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EXPECT_METRIC_THAT(
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metrics::Samples("WebRTC.Audio.ApmAnalogGainDecreaseAverage"),
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::testing::ElementsAre());
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EXPECT_METRIC_THAT(
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metrics::Samples("WebRTC.Audio.ApmAnalogGainIncreaseAverage"),
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::testing::ElementsAre());
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}
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TEST(AnalogGainStatsReporterTest, CheckLogLevelUpdateStatsNotEmpty) {
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AnalogGainStatsReporter stats_reporter;
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constexpr int kMicLevel = 10;
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stats_reporter.UpdateStatistics(kMicLevel);
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// Update until periodic logging.
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for (int i = 0; i < kFramesIn60Seconds; i += 2) {
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stats_reporter.UpdateStatistics(kMicLevel + 2);
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stats_reporter.UpdateStatistics(kMicLevel);
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}
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// Update until periodic logging.
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for (int i = 0; i < kFramesIn60Seconds; i += 2) {
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stats_reporter.UpdateStatistics(kMicLevel + 3);
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stats_reporter.UpdateStatistics(kMicLevel);
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}
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EXPECT_METRIC_THAT(
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metrics::Samples("WebRTC.Audio.ApmAnalogGainUpdateRate"),
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::testing::ElementsAre(::testing::Pair(kFramesIn60Seconds - 1, 1),
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::testing::Pair(kFramesIn60Seconds, 1)));
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EXPECT_METRIC_THAT(
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metrics::Samples("WebRTC.Audio.ApmAnalogGainDecreaseRate"),
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::testing::ElementsAre(::testing::Pair(kFramesIn60Seconds / 2 - 1, 1),
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::testing::Pair(kFramesIn60Seconds / 2, 1)));
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EXPECT_METRIC_THAT(
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metrics::Samples("WebRTC.Audio.ApmAnalogGainIncreaseRate"),
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::testing::ElementsAre(::testing::Pair(kFramesIn60Seconds / 2, 2)));
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EXPECT_METRIC_THAT(
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metrics::Samples("WebRTC.Audio.ApmAnalogGainUpdateAverage"),
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::testing::ElementsAre(::testing::Pair(2, 1), ::testing::Pair(3, 1)));
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EXPECT_METRIC_THAT(
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metrics::Samples("WebRTC.Audio.ApmAnalogGainDecreaseAverage"),
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::testing::ElementsAre(::testing::Pair(2, 1), ::testing::Pair(3, 1)));
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EXPECT_METRIC_THAT(
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metrics::Samples("WebRTC.Audio.ApmAnalogGainIncreaseAverage"),
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::testing::ElementsAre(::testing::Pair(2, 1), ::testing::Pair(3, 1)));
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}
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} // namespace
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TEST(AnalogGainStatsReporterTest, CheckLevelUpdateStatsForEmptyStats) {
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AnalogGainStatsReporter stats_reporter;
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const auto& update_stats = stats_reporter.level_update_stats();
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EXPECT_EQ(update_stats.num_decreases, 0);
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EXPECT_EQ(update_stats.sum_decreases, 0);
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EXPECT_EQ(update_stats.num_increases, 0);
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EXPECT_EQ(update_stats.sum_increases, 0);
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}
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TEST(AnalogGainStatsReporterTest, CheckLevelUpdateStatsAfterNoGainChange) {
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constexpr int kMicLevel = 10;
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AnalogGainStatsReporter stats_reporter;
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stats_reporter.UpdateStatistics(kMicLevel);
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stats_reporter.UpdateStatistics(kMicLevel);
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stats_reporter.UpdateStatistics(kMicLevel);
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const auto& update_stats = stats_reporter.level_update_stats();
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EXPECT_EQ(update_stats.num_decreases, 0);
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EXPECT_EQ(update_stats.sum_decreases, 0);
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EXPECT_EQ(update_stats.num_increases, 0);
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EXPECT_EQ(update_stats.sum_increases, 0);
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}
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TEST(AnalogGainStatsReporterTest, CheckLevelUpdateStatsAfterGainIncrease) {
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constexpr int kMicLevel = 10;
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AnalogGainStatsReporter stats_reporter;
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stats_reporter.UpdateStatistics(kMicLevel);
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stats_reporter.UpdateStatistics(kMicLevel + 4);
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stats_reporter.UpdateStatistics(kMicLevel + 5);
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const auto& update_stats = stats_reporter.level_update_stats();
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EXPECT_EQ(update_stats.num_decreases, 0);
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EXPECT_EQ(update_stats.sum_decreases, 0);
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EXPECT_EQ(update_stats.num_increases, 2);
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EXPECT_EQ(update_stats.sum_increases, 5);
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}
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TEST(AnalogGainStatsReporterTest, CheckLevelUpdateStatsAfterGainDecrease) {
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constexpr int kMicLevel = 10;
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AnalogGainStatsReporter stats_reporter;
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stats_reporter.UpdateStatistics(kMicLevel);
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stats_reporter.UpdateStatistics(kMicLevel - 4);
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stats_reporter.UpdateStatistics(kMicLevel - 5);
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const auto& stats_update = stats_reporter.level_update_stats();
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EXPECT_EQ(stats_update.num_decreases, 2);
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EXPECT_EQ(stats_update.sum_decreases, 5);
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EXPECT_EQ(stats_update.num_increases, 0);
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EXPECT_EQ(stats_update.sum_increases, 0);
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}
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TEST(AnalogGainStatsReporterTest, CheckLevelUpdateStatsAfterReset) {
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AnalogGainStatsReporter stats_reporter;
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constexpr int kMicLevel = 10;
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stats_reporter.UpdateStatistics(kMicLevel);
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// Update until the periodic reset.
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||||
for (int i = 0; i < kFramesIn60Seconds - 2; i += 2) {
|
||||
stats_reporter.UpdateStatistics(kMicLevel + 2);
|
||||
stats_reporter.UpdateStatistics(kMicLevel);
|
||||
}
|
||||
const auto& stats_before_reset = stats_reporter.level_update_stats();
|
||||
EXPECT_EQ(stats_before_reset.num_decreases, kFramesIn60Seconds / 2 - 1);
|
||||
EXPECT_EQ(stats_before_reset.sum_decreases, kFramesIn60Seconds - 2);
|
||||
EXPECT_EQ(stats_before_reset.num_increases, kFramesIn60Seconds / 2 - 1);
|
||||
EXPECT_EQ(stats_before_reset.sum_increases, kFramesIn60Seconds - 2);
|
||||
stats_reporter.UpdateStatistics(kMicLevel + 2);
|
||||
const auto& stats_during_reset = stats_reporter.level_update_stats();
|
||||
EXPECT_EQ(stats_during_reset.num_decreases, 0);
|
||||
EXPECT_EQ(stats_during_reset.sum_decreases, 0);
|
||||
EXPECT_EQ(stats_during_reset.num_increases, 0);
|
||||
EXPECT_EQ(stats_during_reset.sum_increases, 0);
|
||||
stats_reporter.UpdateStatistics(kMicLevel);
|
||||
stats_reporter.UpdateStatistics(kMicLevel + 3);
|
||||
const auto& stats_after_reset = stats_reporter.level_update_stats();
|
||||
EXPECT_EQ(stats_after_reset.num_decreases, 1);
|
||||
EXPECT_EQ(stats_after_reset.sum_decreases, 2);
|
||||
EXPECT_EQ(stats_after_reset.num_increases, 1);
|
||||
EXPECT_EQ(stats_after_reset.sum_increases, 3);
|
||||
}
|
||||
|
||||
} // namespace webrtc
|
||||
@ -1149,6 +1149,7 @@ int AudioProcessingImpl::ProcessCaptureStreamLocked() {
|
||||
capture_.prev_analog_mic_level != analog_mic_level &&
|
||||
capture_.prev_analog_mic_level != -1;
|
||||
capture_.prev_analog_mic_level = analog_mic_level;
|
||||
analog_gain_stats_reporter_.UpdateStatistics(analog_mic_level);
|
||||
|
||||
if (submodules_.echo_controller) {
|
||||
capture_.echo_path_gain_change = analog_mic_level_changed;
|
||||
|
||||
@ -21,6 +21,7 @@
|
||||
#include "api/function_view.h"
|
||||
#include "modules/audio_processing/aec3/echo_canceller3.h"
|
||||
#include "modules/audio_processing/agc/agc_manager_direct.h"
|
||||
#include "modules/audio_processing/agc/analog_gain_stats_reporter.h"
|
||||
#include "modules/audio_processing/agc/gain_control.h"
|
||||
#include "modules/audio_processing/audio_buffer.h"
|
||||
#include "modules/audio_processing/capture_levels_adjuster/capture_levels_adjuster.h"
|
||||
@ -531,6 +532,9 @@ class AudioProcessingImpl : public AudioProcessing {
|
||||
RmsLevel capture_output_rms_ RTC_GUARDED_BY(mutex_capture_);
|
||||
int capture_rms_interval_counter_ RTC_GUARDED_BY(mutex_capture_) = 0;
|
||||
|
||||
AnalogGainStatsReporter analog_gain_stats_reporter_
|
||||
RTC_GUARDED_BY(mutex_capture_);
|
||||
|
||||
// Lock protection not needed.
|
||||
std::unique_ptr<
|
||||
SwapQueue<std::vector<int16_t>, RenderQueueItemVerifier<int16_t>>>
|
||||
|
||||
Loading…
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Reference in New Issue
Block a user