mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-15 14:50:39 +01:00

We add 2 metrics for measuring applied digital gain to AgcManagerDirect. We also add an applied gain and an estimated noise metric to Agc2. Chromium histogram CL is https://chromium-review.googlesource.com/c/chromium/src/+/1170833 Bug: webrtc:7494 Change-Id: Ie40873f9e43bc7d34d8f5473cd73bd47eb84e855 Reviewed-on: https://webrtc-review.googlesource.com/93468 Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Commit-Queue: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24290}
123 lines
4.6 KiB
C++
123 lines
4.6 KiB
C++
/*
|
|
* Copyright (c) 2018 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/agc2/adaptive_digital_gain_applier.h"
|
|
|
|
#include <algorithm>
|
|
|
|
#include "common_audio/include/audio_util.h"
|
|
#include "modules/audio_processing/agc2/agc2_common.h"
|
|
#include "modules/audio_processing/logging/apm_data_dumper.h"
|
|
#include "rtc_base/numerics/safe_minmax.h"
|
|
#include "system_wrappers/include/metrics.h"
|
|
|
|
namespace webrtc {
|
|
namespace {
|
|
|
|
// This function maps input level to desired applied gain. We want to
|
|
// boost the signal so that peaks are at -kHeadroomDbfs. We can't
|
|
// apply more than kMaxGainDb gain.
|
|
float ComputeGainDb(float input_level_dbfs) {
|
|
// If the level is very low, boost it as much as we can.
|
|
if (input_level_dbfs < -(kHeadroomDbfs + kMaxGainDb)) {
|
|
return kMaxGainDb;
|
|
}
|
|
|
|
// We expect to end up here most of the time: the level is below
|
|
// -headroom, but we can boost it to -headroom.
|
|
if (input_level_dbfs < -kHeadroomDbfs) {
|
|
return -kHeadroomDbfs - input_level_dbfs;
|
|
}
|
|
|
|
// Otherwise, the level is too high and we can't boost. The
|
|
// LevelEstimator is responsible for not reporting bogus gain
|
|
// values.
|
|
RTC_DCHECK_LE(input_level_dbfs, 0.f);
|
|
return 0.f;
|
|
}
|
|
|
|
// We require 'gain + noise_level <= kMaxNoiseLevelDbfs'.
|
|
float LimitGainByNoise(float target_gain,
|
|
float input_noise_level_dbfs,
|
|
ApmDataDumper* apm_data_dumper) {
|
|
const float noise_headroom_db = kMaxNoiseLevelDbfs - input_noise_level_dbfs;
|
|
apm_data_dumper->DumpRaw("agc2_noise_headroom_db", noise_headroom_db);
|
|
return std::min(target_gain, std::max(noise_headroom_db, 0.f));
|
|
}
|
|
|
|
// Computes how the gain should change during this frame.
|
|
// Return the gain difference in db to 'last_gain_db'.
|
|
float ComputeGainChangeThisFrameDb(float target_gain_db,
|
|
float last_gain_db,
|
|
bool gain_increase_allowed) {
|
|
float target_gain_difference_db = target_gain_db - last_gain_db;
|
|
if (!gain_increase_allowed) {
|
|
target_gain_difference_db = std::min(target_gain_difference_db, 0.f);
|
|
}
|
|
|
|
return rtc::SafeClamp(target_gain_difference_db, -kMaxGainChangePerFrameDb,
|
|
kMaxGainChangePerFrameDb);
|
|
}
|
|
} // namespace
|
|
|
|
AdaptiveDigitalGainApplier::AdaptiveDigitalGainApplier(
|
|
ApmDataDumper* apm_data_dumper)
|
|
: gain_applier_(false, DbToRatio(last_gain_db_)),
|
|
apm_data_dumper_(apm_data_dumper) {}
|
|
|
|
void AdaptiveDigitalGainApplier::Process(
|
|
float input_level_dbfs,
|
|
float input_noise_level_dbfs,
|
|
const VadWithLevel::LevelAndProbability vad_result,
|
|
AudioFrameView<float> float_frame) {
|
|
calls_since_last_gain_log_++;
|
|
if (calls_since_last_gain_log_ == 100) {
|
|
calls_since_last_gain_log_ = 0;
|
|
RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.Agc2.DigitalGainApplied",
|
|
last_gain_db_, 0, kMaxGainDb, kMaxGainDb + 1);
|
|
RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.Agc2.EstimatedNoiseLevel",
|
|
input_noise_level_dbfs, 0, 100, 101);
|
|
}
|
|
|
|
input_level_dbfs = std::min(input_level_dbfs, 0.f);
|
|
|
|
RTC_DCHECK_GE(input_level_dbfs, -150.f);
|
|
RTC_DCHECK_GE(float_frame.num_channels(), 1);
|
|
RTC_DCHECK_GE(float_frame.samples_per_channel(), 1);
|
|
|
|
const float target_gain_db =
|
|
LimitGainByNoise(ComputeGainDb(input_level_dbfs), input_noise_level_dbfs,
|
|
apm_data_dumper_);
|
|
|
|
// Forbid increasing the gain when there is no speech.
|
|
gain_increase_allowed_ =
|
|
vad_result.speech_probability > kVadConfidenceThreshold;
|
|
|
|
const float gain_change_this_frame_db = ComputeGainChangeThisFrameDb(
|
|
target_gain_db, last_gain_db_, gain_increase_allowed_);
|
|
|
|
apm_data_dumper_->DumpRaw("agc2_want_to_change_by_db",
|
|
target_gain_db - last_gain_db_);
|
|
apm_data_dumper_->DumpRaw("agc2_will_change_by_db",
|
|
gain_change_this_frame_db);
|
|
|
|
// Optimization: avoid calling math functions if gain does not
|
|
// change.
|
|
if (gain_change_this_frame_db != 0.f) {
|
|
gain_applier_.SetGainFactor(
|
|
DbToRatio(last_gain_db_ + gain_change_this_frame_db));
|
|
}
|
|
gain_applier_.ApplyGain(float_frame);
|
|
|
|
// Remember that the gain has changed for the next iteration.
|
|
last_gain_db_ = last_gain_db_ + gain_change_this_frame_db;
|
|
apm_data_dumper_->DumpRaw("agc2_applied_gain_db", last_gain_db_);
|
|
}
|
|
} // namespace webrtc
|