| Index: webrtc/modules/audio_processing/level_controller/level_controller.cc
|
| diff --git a/webrtc/modules/audio_processing/level_controller/level_controller.cc b/webrtc/modules/audio_processing/level_controller/level_controller.cc
|
| index a9fed9bf95ab04a9a0a6843e0616eb40e9034204..2294396607505f86511e83f12f110a8a636601bf 100644
|
| --- a/webrtc/modules/audio_processing/level_controller/level_controller.cc
|
| +++ b/webrtc/modules/audio_processing/level_controller/level_controller.cc
|
| @@ -25,6 +25,7 @@
|
| #include "webrtc/modules/audio_processing/level_controller/saturating_gain_estimator.h"
|
| #include "webrtc/modules/audio_processing/level_controller/signal_classifier.h"
|
| #include "webrtc/modules/audio_processing/logging/apm_data_dumper.h"
|
| +#include "webrtc/system_wrappers/include/logging.h"
|
| #include "webrtc/system_wrappers/include/metrics.h"
|
|
|
| namespace webrtc {
|
| @@ -93,54 +94,86 @@ void LevelController::Metrics::Reset() {
|
| max_noise_energy_ = 0.f;
|
| }
|
|
|
| -void LevelController::Metrics::Update(float peak_level,
|
| +void LevelController::Metrics::Update(float long_term_peak_level,
|
| float noise_energy,
|
| - float gain) {
|
| + float gain,
|
| + float frame_peak_level) {
|
| const float kdBFSOffset = 90.3090f;
|
| gain_sum_ += gain;
|
| - peak_level_sum_ += peak_level;
|
| + peak_level_sum_ += long_term_peak_level;
|
| noise_energy_sum_ += noise_energy;
|
| max_gain_ = std::max(max_gain_, gain);
|
| - max_peak_level_ = std::max(max_peak_level_, peak_level);
|
| + max_peak_level_ = std::max(max_peak_level_, long_term_peak_level);
|
| max_noise_energy_ = std::max(max_noise_energy_, noise_energy);
|
|
|
| ++metrics_frame_counter_;
|
| if (metrics_frame_counter_ == kMetricsFrameInterval) {
|
| - RTC_HISTOGRAM_COUNTS(
|
| - "WebRTC.Audio.LevelControl.MaxNoisePower",
|
| - static_cast<int>(10 * log10(max_noise_energy_ / frame_length_ + 1e-10f)
|
| - - kdBFSOffset),
|
| - -90, 0, 50);
|
| - RTC_HISTOGRAM_COUNTS(
|
| - "WebRTC.Audio.LevelControl.AverageNoisePower",
|
| - static_cast<int>(10 * log10(noise_energy_sum_ /
|
| - (frame_length_ * kMetricsFrameInterval) +
|
| - 1e-10f) - kdBFSOffset),
|
| - -90, 0, 50);
|
| -
|
| - RTC_HISTOGRAM_COUNTS(
|
| - "WebRTC.Audio.LevelControl.MaxPeakLevel",
|
| - static_cast<int>(10 * log10(max_peak_level_ * max_peak_level_ + 1e-10f)
|
| - - kdBFSOffset),
|
| - -90, 0, 50);
|
| - RTC_HISTOGRAM_COUNTS(
|
| - "WebRTC.Audio.LevelControl.AveragePeakLevel",
|
| - static_cast<int>(10 * log10(peak_level_sum_ * peak_level_sum_ /
|
| - (kMetricsFrameInterval *
|
| - kMetricsFrameInterval) +
|
| - 1e-10f) - kdBFSOffset),
|
| - -90, 0, 50);
|
| + RTC_DCHECK_LT(0, frame_length_);
|
| + RTC_DCHECK_LT(0, kMetricsFrameInterval);
|
| +
|
| + const int max_noise_power_dbfs = static_cast<int>(
|
| + 10 * log10(max_noise_energy_ / frame_length_ + 1e-10f) - kdBFSOffset);
|
| + RTC_HISTOGRAM_COUNTS("WebRTC.Audio.LevelControl.MaxNoisePower",
|
| + max_noise_power_dbfs, -90, 0, 50);
|
| +
|
| + const int average_noise_power_dbfs = static_cast<int>(
|
| + 10 * log10(noise_energy_sum_ / (frame_length_ * kMetricsFrameInterval) +
|
| + 1e-10f) -
|
| + kdBFSOffset);
|
| + RTC_HISTOGRAM_COUNTS("WebRTC.Audio.LevelControl.AverageNoisePower",
|
| + average_noise_power_dbfs, -90, 0, 50);
|
| +
|
| + const int max_peak_level_dbfs = static_cast<int>(
|
| + 10 * log10(max_peak_level_ * max_peak_level_ + 1e-10f) - kdBFSOffset);
|
| + RTC_HISTOGRAM_COUNTS("WebRTC.Audio.LevelControl.MaxPeakLevel",
|
| + max_peak_level_dbfs, -90, 0, 50);
|
| +
|
| + const int average_peak_level_dbfs = static_cast<int>(
|
| + 10 * log10(peak_level_sum_ * peak_level_sum_ /
|
| + (kMetricsFrameInterval * kMetricsFrameInterval) +
|
| + 1e-10f) -
|
| + kdBFSOffset);
|
| + RTC_HISTOGRAM_COUNTS("WebRTC.Audio.LevelControl.AveragePeakLevel",
|
| + average_peak_level_dbfs, -90, 0, 50);
|
|
|
| RTC_DCHECK_LE(1.f, max_gain_);
|
| RTC_DCHECK_LE(1.f, gain_sum_ / kMetricsFrameInterval);
|
| - RTC_HISTOGRAM_COUNTS("WebRTC.Audio.LevelControl.MaxGain",
|
| - static_cast<int>(10 * log10(max_gain_ * max_gain_)),
|
| - 0, 33, 30);
|
| +
|
| + const int max_gain_db = static_cast<int>(10 * log10(max_gain_ * max_gain_));
|
| + RTC_HISTOGRAM_COUNTS("WebRTC.Audio.LevelControl.MaxGain", max_gain_db, 0,
|
| + 33, 30);
|
| +
|
| + const int average_gain_db = static_cast<int>(
|
| + 10 * log10(gain_sum_ * gain_sum_ /
|
| + (kMetricsFrameInterval * kMetricsFrameInterval)));
|
| RTC_HISTOGRAM_COUNTS("WebRTC.Audio.LevelControl.AverageGain",
|
| - static_cast<int>(10 * log10(gain_sum_ * gain_sum_ /
|
| - (kMetricsFrameInterval *
|
| - kMetricsFrameInterval))),
|
| - 0, 33, 30);
|
| + average_gain_db, 0, 33, 30);
|
| +
|
| + const int long_term_peak_level_dbfs = static_cast<int>(
|
| + 10 * log10(long_term_peak_level * long_term_peak_level + 1e-10f) -
|
| + kdBFSOffset);
|
| +
|
| + const int frame_peak_level_dbfs = static_cast<int>(
|
| + 10 * log10(frame_peak_level * frame_peak_level + 1e-10f) - kdBFSOffset);
|
| +
|
| + LOG(LS_INFO) << "Level Controller metrics: " << std::endl
|
| + << "Max noise power: " << max_noise_power_dbfs
|
| + << " dBFS" << std::endl
|
| + << "Average noise power: " << average_noise_power_dbfs
|
| + << " dBFS" << std::endl
|
| + << "Max long term peak level: " << max_peak_level_dbfs
|
| + << " dBFS" << std::endl
|
| + << "Average long term peak level: " << average_peak_level_dbfs
|
| + << " dBFS" << std::endl
|
| + << "Max gain: " << max_gain_db << " dB"
|
| + << std::endl
|
| + << "Average gain: " << average_gain_db << " dB"
|
| + << std::endl
|
| + << "Long term peak level: "
|
| + << long_term_peak_level_dbfs << " dBFS" << std::endl
|
| + << "Last frame peak level: " << frame_peak_level_dbfs
|
| + << " dBFS";
|
| +
|
| Reset();
|
| }
|
| }
|
| @@ -200,13 +233,14 @@ void LevelController::Process(AudioBuffer* audio) {
|
| noise_level_estimator_.Analyze(signal_type, FrameEnergy(*audio));
|
|
|
| // Estimate the overall signal peak level.
|
| - float peak_level =
|
| - peak_level_estimator_.Analyze(signal_type, PeakLevel(*audio));
|
| + const float frame_peak_level = PeakLevel(*audio);
|
| + const float long_term_peak_level =
|
| + peak_level_estimator_.Analyze(signal_type, frame_peak_level);
|
|
|
| float saturating_gain = saturating_gain_estimator_.GetGain();
|
|
|
| // Compute the new gain to apply.
|
| - last_gain_ = gain_selector_.GetNewGain(peak_level, noise_energy,
|
| + last_gain_ = gain_selector_.GetNewGain(long_term_peak_level, noise_energy,
|
| saturating_gain, signal_type);
|
|
|
| // Apply the gain to the signal.
|
| @@ -216,11 +250,12 @@ void LevelController::Process(AudioBuffer* audio) {
|
| saturating_gain_estimator_.Update(last_gain_, num_saturations);
|
|
|
| // Update the metrics.
|
| - metrics_.Update(peak_level, noise_energy, last_gain_);
|
| + metrics_.Update(long_term_peak_level, noise_energy, last_gain_,
|
| + frame_peak_level);
|
|
|
| data_dumper_->DumpRaw("lc_selected_gain", 1, &last_gain_);
|
| data_dumper_->DumpRaw("lc_noise_energy", 1, &noise_energy);
|
| - data_dumper_->DumpRaw("lc_peak_level", 1, &peak_level);
|
| + data_dumper_->DumpRaw("lc_peak_level", 1, &long_term_peak_level);
|
| data_dumper_->DumpRaw("lc_saturating_gain", 1, &saturating_gain);
|
|
|
| data_dumper_->DumpWav("lc_output", audio->num_frames(),
|
|
|