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/*
* 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_controller.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/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/safe_minmax.h"
#include "system_wrappers/include/metrics.h"
namespace webrtc {
namespace {
using AdaptiveDigitalConfig =
AudioProcessing::Config::GainController2::AdaptiveDigital;
constexpr int kHeadroomHistogramMin = 0;
constexpr int kHeadroomHistogramMax = 50;
constexpr int kGainDbHistogramMax = 30;
// Computes the gain for `input_level_dbfs` to reach `-config.headroom_db`.
// Clamps the gain in [0, `config.max_gain_db`]. `config.headroom_db` is a
// safety margin to allow transient peaks to exceed the target peak level
// without clipping.
float ComputeGainDb(float input_level_dbfs,
const AdaptiveDigitalConfig& config) {
// If the level is very low, apply the maximum gain.
if (input_level_dbfs < -(config.headroom_db + config.max_gain_db)) {
return config.max_gain_db;
}
// 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 < -config.headroom_db) {
return -config.headroom_db - input_level_dbfs;
}
// The level is too high and we can't boost.
RTC_DCHECK_GE(input_level_dbfs, -config.headroom_db);
return 0.0f;
}
// Returns `target_gain_db` if applying such a gain to `input_noise_level_dbfs`
// does not exceed `max_output_noise_level_dbfs`. Otherwise lowers and returns
// `target_gain_db` so that the output noise level equals
// `max_output_noise_level_dbfs`.
float LimitGainByNoise(float target_gain_db,
float input_noise_level_dbfs,
float max_output_noise_level_dbfs,
ApmDataDumper& apm_data_dumper) {
const float max_allowed_gain_db =
max_output_noise_level_dbfs - input_noise_level_dbfs;
apm_data_dumper.DumpRaw("agc2_adaptive_gain_applier_max_allowed_gain_db",
max_allowed_gain_db);
return std::min(target_gain_db, std::max(max_allowed_gain_db, 0.0f));
}
float LimitGainByLowConfidence(float target_gain_db,
float last_gain_db,
float limiter_audio_level_dbfs,
bool estimate_is_confident) {
if (estimate_is_confident ||
limiter_audio_level_dbfs <= kLimiterThresholdForAgcGainDbfs) {
return target_gain_db;
}
const float limiter_level_dbfs_before_gain =
limiter_audio_level_dbfs - last_gain_db;
// Compute a new gain so that `limiter_level_dbfs_before_gain` +
// `new_target_gain_db` is not great than `kLimiterThresholdForAgcGainDbfs`.
const float new_target_gain_db = std::max(
kLimiterThresholdForAgcGainDbfs - limiter_level_dbfs_before_gain, 0.0f);
return std::min(new_target_gain_db, target_gain_db);
}
// 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 max_gain_decrease_db,
float max_gain_increase_db) {
RTC_DCHECK_GT(max_gain_decrease_db, 0);
RTC_DCHECK_GT(max_gain_increase_db, 0);
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, 0.0f);
}
return SafeClamp(target_gain_difference_db, -max_gain_decrease_db,
max_gain_increase_db);
}
} // namespace
AdaptiveDigitalGainController::AdaptiveDigitalGainController(
ApmDataDumper* apm_data_dumper,
const AudioProcessing::Config::GainController2::AdaptiveDigital& config,
int adjacent_speech_frames_threshold)
: apm_data_dumper_(apm_data_dumper),
gain_applier_(
/*hard_clip_samples=*/false,
/*initial_gain_factor=*/DbToRatio(config.initial_gain_db)),
config_(config),
adjacent_speech_frames_threshold_(adjacent_speech_frames_threshold),
max_gain_change_db_per_10ms_(config_.max_gain_change_db_per_second *
kFrameDurationMs / 1000.0f),
calls_since_last_gain_log_(0),
frames_to_gain_increase_allowed_(adjacent_speech_frames_threshold),
last_gain_db_(config_.initial_gain_db) {
RTC_DCHECK_GT(max_gain_change_db_per_10ms_, 0.0f);
RTC_DCHECK_GE(frames_to_gain_increase_allowed_, 1);
RTC_DCHECK_GE(config_.max_output_noise_level_dbfs, -90.0f);
RTC_DCHECK_LE(config_.max_output_noise_level_dbfs, 0.0f);
}
void AdaptiveDigitalGainController::Process(const FrameInfo& info,
DeinterleavedView<float> frame) {
RTC_DCHECK_GE(info.speech_level_dbfs, -150.0f);
RTC_DCHECK_GE(frame.num_channels(), 1);
RTC_DCHECK(
frame.samples_per_channel() == 80 || frame.samples_per_channel() == 160 ||
frame.samples_per_channel() == 320 || frame.samples_per_channel() == 480)
<< "`frame` does not look like a 10 ms frame for an APM supported sample "
"rate";
// Compute the input level used to select the desired gain.
RTC_DCHECK_GT(info.headroom_db, 0.0f);
const float input_level_dbfs = info.speech_level_dbfs + info.headroom_db;
const float target_gain_db = LimitGainByLowConfidence(
LimitGainByNoise(ComputeGainDb(input_level_dbfs, config_),
info.noise_rms_dbfs, config_.max_output_noise_level_dbfs,
*apm_data_dumper_),
last_gain_db_, info.limiter_envelope_dbfs, info.speech_level_reliable);
// Forbid increasing the gain until enough adjacent speech frames are
// observed.
bool first_confident_speech_frame = false;
if (info.speech_probability < kVadConfidenceThreshold) {
frames_to_gain_increase_allowed_ = adjacent_speech_frames_threshold_;
} else if (frames_to_gain_increase_allowed_ > 0) {
frames_to_gain_increase_allowed_--;
first_confident_speech_frame = frames_to_gain_increase_allowed_ == 0;
}
apm_data_dumper_->DumpRaw(
"agc2_adaptive_gain_applier_frames_to_gain_increase_allowed",
frames_to_gain_increase_allowed_);
const bool gain_increase_allowed = frames_to_gain_increase_allowed_ == 0;
float max_gain_increase_db = max_gain_change_db_per_10ms_;
if (first_confident_speech_frame) {
// No gain increase happened while waiting for a long enough speech
// sequence. Therefore, temporarily allow a faster gain increase.
RTC_DCHECK(gain_increase_allowed);
max_gain_increase_db *= adjacent_speech_frames_threshold_;
}
const float gain_change_this_frame_db = ComputeGainChangeThisFrameDb(
target_gain_db, last_gain_db_, gain_increase_allowed,
/*max_gain_decrease_db=*/max_gain_change_db_per_10ms_,
max_gain_increase_db);
apm_data_dumper_->DumpRaw("agc2_adaptive_gain_applier_want_to_change_by_db",
target_gain_db - last_gain_db_);
apm_data_dumper_->DumpRaw("agc2_adaptive_gain_applier_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(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_adaptive_gain_applier_applied_gain_db",
last_gain_db_);
// Log every 10 seconds.
calls_since_last_gain_log_++;
if (calls_since_last_gain_log_ == 1000) {
calls_since_last_gain_log_ = 0;
RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.Agc2.EstimatedSpeechLevel",
-info.speech_level_dbfs, 0, 100, 101);
RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.Agc2.EstimatedNoiseLevel",
-info.noise_rms_dbfs, 0, 100, 101);
RTC_HISTOGRAM_COUNTS_LINEAR(
"WebRTC.Audio.Agc2.Headroom", info.headroom_db, kHeadroomHistogramMin,
kHeadroomHistogramMax,
kHeadroomHistogramMax - kHeadroomHistogramMin + 1);
RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.Agc2.DigitalGainApplied",
last_gain_db_, 0, kGainDbHistogramMax,
kGainDbHistogramMax + 1);
RTC_LOG(LS_INFO) << "AGC2 adaptive digital"
<< " | speech_dbfs: " << info.speech_level_dbfs
<< " | noise_dbfs: " << info.noise_rms_dbfs
<< " | headroom_db: " << info.headroom_db
<< " | gain_db: " << last_gain_db_;
}
}
} // namespace webrtc
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