mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-13 13:50:40 +01:00
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}
130 lines
4.6 KiB
C++
130 lines
4.6 KiB
C++
/*
|
|
* Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
|
|
*
|
|
* Use of this source code is governed by a BSD-style license
|
|
* that can be found in the LICENSE file in the root of the source
|
|
* tree. An additional intellectual property rights grant can be found
|
|
* in the file PATENTS. All contributing project authors may
|
|
* be found in the AUTHORS file in the root of the source tree.
|
|
*/
|
|
|
|
#include "modules/audio_processing/agc/analog_gain_stats_reporter.h"
|
|
|
|
#include <cmath>
|
|
|
|
#include "rtc_base/logging.h"
|
|
#include "rtc_base/numerics/safe_minmax.h"
|
|
#include "system_wrappers/include/metrics.h"
|
|
|
|
namespace webrtc {
|
|
namespace {
|
|
|
|
constexpr int kFramesIn60Seconds = 6000;
|
|
constexpr int kMinGain = 0;
|
|
constexpr int kMaxGain = 255;
|
|
constexpr int kMaxUpdate = kMaxGain - kMinGain;
|
|
|
|
float ComputeAverageUpdate(int sum_updates, int num_updates) {
|
|
RTC_DCHECK_GE(sum_updates, 0);
|
|
RTC_DCHECK_LE(sum_updates, kMaxUpdate * kFramesIn60Seconds);
|
|
RTC_DCHECK_GE(num_updates, 0);
|
|
RTC_DCHECK_LE(num_updates, kFramesIn60Seconds);
|
|
if (num_updates == 0) {
|
|
return 0.0f;
|
|
}
|
|
return std::round(static_cast<float>(sum_updates) /
|
|
static_cast<float>(num_updates));
|
|
}
|
|
} // namespace
|
|
|
|
AnalogGainStatsReporter::AnalogGainStatsReporter() = default;
|
|
|
|
AnalogGainStatsReporter::~AnalogGainStatsReporter() = default;
|
|
|
|
void AnalogGainStatsReporter::UpdateStatistics(int analog_mic_level) {
|
|
RTC_DCHECK_GE(analog_mic_level, kMinGain);
|
|
RTC_DCHECK_LE(analog_mic_level, kMaxGain);
|
|
if (previous_analog_mic_level_.has_value() &&
|
|
analog_mic_level != previous_analog_mic_level_.value()) {
|
|
const int level_change =
|
|
analog_mic_level - previous_analog_mic_level_.value();
|
|
if (level_change < 0) {
|
|
++level_update_stats_.num_decreases;
|
|
level_update_stats_.sum_decreases -= level_change;
|
|
} else {
|
|
++level_update_stats_.num_increases;
|
|
level_update_stats_.sum_increases += level_change;
|
|
}
|
|
}
|
|
// Periodically log analog gain change metrics.
|
|
if (++log_level_update_stats_counter_ >= kFramesIn60Seconds) {
|
|
LogLevelUpdateStats();
|
|
level_update_stats_ = {};
|
|
log_level_update_stats_counter_ = 0;
|
|
}
|
|
previous_analog_mic_level_ = analog_mic_level;
|
|
}
|
|
|
|
void AnalogGainStatsReporter::LogLevelUpdateStats() const {
|
|
const float average_decrease = ComputeAverageUpdate(
|
|
level_update_stats_.sum_decreases, level_update_stats_.num_decreases);
|
|
const float average_increase = ComputeAverageUpdate(
|
|
level_update_stats_.sum_increases, level_update_stats_.num_increases);
|
|
const int num_updates =
|
|
level_update_stats_.num_decreases + level_update_stats_.num_increases;
|
|
const float average_update = ComputeAverageUpdate(
|
|
level_update_stats_.sum_decreases + level_update_stats_.sum_increases,
|
|
num_updates);
|
|
RTC_DLOG(LS_INFO) << "Analog gain update rate: "
|
|
<< "num_updates=" << num_updates
|
|
<< ", num_decreases=" << level_update_stats_.num_decreases
|
|
<< ", num_increases=" << level_update_stats_.num_increases;
|
|
RTC_DLOG(LS_INFO) << "Analog gain update average: "
|
|
<< "average_update=" << average_update
|
|
<< ", average_decrease=" << average_decrease
|
|
<< ", average_increase=" << average_increase;
|
|
RTC_HISTOGRAM_COUNTS_LINEAR(
|
|
/*name=*/"WebRTC.Audio.ApmAnalogGainDecreaseRate",
|
|
/*sample=*/level_update_stats_.num_decreases,
|
|
/*min=*/1,
|
|
/*max=*/kFramesIn60Seconds,
|
|
/*bucket_count=*/50);
|
|
if (level_update_stats_.num_decreases > 0) {
|
|
RTC_HISTOGRAM_COUNTS_LINEAR(
|
|
/*name=*/"WebRTC.Audio.ApmAnalogGainDecreaseAverage",
|
|
/*sample=*/average_decrease,
|
|
/*min=*/1,
|
|
/*max=*/kMaxUpdate,
|
|
/*bucket_count=*/50);
|
|
}
|
|
RTC_HISTOGRAM_COUNTS_LINEAR(
|
|
/*name=*/"WebRTC.Audio.ApmAnalogGainIncreaseRate",
|
|
/*sample=*/level_update_stats_.num_increases,
|
|
/*min=*/1,
|
|
/*max=*/kFramesIn60Seconds,
|
|
/*bucket_count=*/50);
|
|
if (level_update_stats_.num_increases > 0) {
|
|
RTC_HISTOGRAM_COUNTS_LINEAR(
|
|
/*name=*/"WebRTC.Audio.ApmAnalogGainIncreaseAverage",
|
|
/*sample=*/average_increase,
|
|
/*min=*/1,
|
|
/*max=*/kMaxUpdate,
|
|
/*bucket_count=*/50);
|
|
}
|
|
RTC_HISTOGRAM_COUNTS_LINEAR(
|
|
/*name=*/"WebRTC.Audio.ApmAnalogGainUpdateRate",
|
|
/*sample=*/num_updates,
|
|
/*min=*/1,
|
|
/*max=*/kFramesIn60Seconds,
|
|
/*bucket_count=*/50);
|
|
if (num_updates > 0) {
|
|
RTC_HISTOGRAM_COUNTS_LINEAR(
|
|
/*name=*/"WebRTC.Audio.ApmAnalogGainUpdateAverage",
|
|
/*sample=*/average_update,
|
|
/*min=*/1,
|
|
/*max=*/kMaxUpdate,
|
|
/*bucket_count=*/50);
|
|
}
|
|
}
|
|
|
|
} // namespace webrtc
|