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/*
* Copyright (c) 2013 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/input_volume_controller.h"
#include <algorithm>
#include <cmath>
#include "api/array_view.h"
#include "modules/audio_processing/agc2/gain_map_internal.h"
#include "modules/audio_processing/agc2/input_volume_stats_reporter.h"
#include "modules/audio_processing/include/audio_frame_view.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/safe_minmax.h"
#include "system_wrappers/include/field_trial.h"
#include "system_wrappers/include/metrics.h"
namespace webrtc {
namespace {
// Amount of error we tolerate in the microphone input volume (presumably due to
// OS quantization) before we assume the user has manually adjusted the volume.
constexpr int kVolumeQuantizationSlack = 25;
constexpr int kMaxInputVolume = 255;
static_assert(kGainMapSize > kMaxInputVolume, "gain map too small");
// Maximum absolute RMS error.
constexpr int KMaxAbsRmsErrorDbfs = 15;
static_assert(KMaxAbsRmsErrorDbfs > 0, "");
using Agc1ClippingPredictorConfig = AudioProcessing::Config::GainController1::
AnalogGainController::ClippingPredictor;
// TODO(webrtc:7494): Hardcode clipping predictor parameters and remove this
// function after no longer needed in the ctor.
Agc1ClippingPredictorConfig CreateClippingPredictorConfig(bool enabled) {
Agc1ClippingPredictorConfig config;
config.enabled = enabled;
return config;
}
// Returns an input volume in the [`min_input_volume`, `kMaxInputVolume`] range
// that reduces `gain_error_db`, which is a gain error estimated when
// `input_volume` was applied, according to a fixed gain map.
int ComputeVolumeUpdate(int gain_error_db,
int input_volume,
int min_input_volume) {
RTC_DCHECK_GE(input_volume, 0);
RTC_DCHECK_LE(input_volume, kMaxInputVolume);
if (gain_error_db == 0) {
return input_volume;
}
int new_volume = input_volume;
if (gain_error_db > 0) {
while (kGainMap[new_volume] - kGainMap[input_volume] < gain_error_db &&
new_volume < kMaxInputVolume) {
++new_volume;
}
} else {
while (kGainMap[new_volume] - kGainMap[input_volume] > gain_error_db &&
new_volume > min_input_volume) {
--new_volume;
}
}
return new_volume;
}
// Returns the proportion of samples in the buffer which are at full-scale
// (and presumably clipped).
float ComputeClippedRatio(const float* const* audio,
size_t num_channels,
size_t samples_per_channel) {
RTC_DCHECK_GT(samples_per_channel, 0);
int num_clipped = 0;
for (size_t ch = 0; ch < num_channels; ++ch) {
int num_clipped_in_ch = 0;
for (size_t i = 0; i < samples_per_channel; ++i) {
RTC_DCHECK(audio[ch]);
if (audio[ch][i] >= 32767.0f || audio[ch][i] <= -32768.0f) {
++num_clipped_in_ch;
}
}
num_clipped = std::max(num_clipped, num_clipped_in_ch);
}
return static_cast<float>(num_clipped) / (samples_per_channel);
}
void LogClippingMetrics(int clipping_rate) {
RTC_LOG(LS_INFO) << "[AGC2] Input clipping rate: " << clipping_rate << "%";
RTC_HISTOGRAM_COUNTS_LINEAR(/*name=*/"WebRTC.Audio.Agc.InputClippingRate",
/*sample=*/clipping_rate, /*min=*/0, /*max=*/100,
/*bucket_count=*/50);
}
// Compares `speech_level_dbfs` to the [`target_range_min_dbfs`,
// `target_range_max_dbfs`] range and returns the error to be compensated via
// input volume adjustment. Returns a positive value when the level is below
// the range, a negative value when the level is above the range, zero
// otherwise.
int GetSpeechLevelRmsErrorDb(float speech_level_dbfs,
int target_range_min_dbfs,
int target_range_max_dbfs) {
constexpr float kMinSpeechLevelDbfs = -90.0f;
constexpr float kMaxSpeechLevelDbfs = 30.0f;
RTC_DCHECK_GE(speech_level_dbfs, kMinSpeechLevelDbfs);
RTC_DCHECK_LE(speech_level_dbfs, kMaxSpeechLevelDbfs);
speech_level_dbfs = rtc::SafeClamp<float>(
speech_level_dbfs, kMinSpeechLevelDbfs, kMaxSpeechLevelDbfs);
int rms_error_db = 0;
if (speech_level_dbfs > target_range_max_dbfs) {
rms_error_db = std::round(target_range_max_dbfs - speech_level_dbfs);
} else if (speech_level_dbfs < target_range_min_dbfs) {
rms_error_db = std::round(target_range_min_dbfs - speech_level_dbfs);
}
return rms_error_db;
}
} // namespace
MonoInputVolumeController::MonoInputVolumeController(
int min_input_volume_after_clipping,
int min_input_volume,
int update_input_volume_wait_frames,
float speech_probability_threshold,
float speech_ratio_threshold)
: min_input_volume_(min_input_volume),
min_input_volume_after_clipping_(min_input_volume_after_clipping),
max_input_volume_(kMaxInputVolume),
update_input_volume_wait_frames_(
std::max(update_input_volume_wait_frames, 1)),
speech_probability_threshold_(speech_probability_threshold),
speech_ratio_threshold_(speech_ratio_threshold) {
RTC_DCHECK_GE(min_input_volume_, 0);
RTC_DCHECK_LE(min_input_volume_, 255);
RTC_DCHECK_GE(min_input_volume_after_clipping_, 0);
RTC_DCHECK_LE(min_input_volume_after_clipping_, 255);
RTC_DCHECK_GE(max_input_volume_, 0);
RTC_DCHECK_LE(max_input_volume_, 255);
RTC_DCHECK_GE(update_input_volume_wait_frames_, 0);
RTC_DCHECK_GE(speech_probability_threshold_, 0.0f);
RTC_DCHECK_LE(speech_probability_threshold_, 1.0f);
RTC_DCHECK_GE(speech_ratio_threshold_, 0.0f);
RTC_DCHECK_LE(speech_ratio_threshold_, 1.0f);
}
MonoInputVolumeController::~MonoInputVolumeController() = default;
void MonoInputVolumeController::Initialize() {
max_input_volume_ = kMaxInputVolume;
capture_output_used_ = true;
check_volume_on_next_process_ = true;
frames_since_update_input_volume_ = 0;
speech_frames_since_update_input_volume_ = 0;
is_first_frame_ = true;
}
// A speeh segment is considered active if at least
// `update_input_volume_wait_frames_` new frames have been processed since the
// previous update and the ratio of non-silence frames (i.e., frames with a
// `speech_probability` higher than `speech_probability_threshold_`) is at least
// `speech_ratio_threshold_`.
void MonoInputVolumeController::Process(absl::optional<int> rms_error_db,
float speech_probability) {
if (check_volume_on_next_process_) {
check_volume_on_next_process_ = false;
// We have to wait until the first process call to check the volume,
// because Chromium doesn't guarantee it to be valid any earlier.
CheckVolumeAndReset();
}
// Count frames with a high speech probability as speech.
if (speech_probability >= speech_probability_threshold_) {
++speech_frames_since_update_input_volume_;
}
// Reset the counters and maybe update the input volume.
if (++frames_since_update_input_volume_ >= update_input_volume_wait_frames_) {
const float speech_ratio =
static_cast<float>(speech_frames_since_update_input_volume_) /
static_cast<float>(update_input_volume_wait_frames_);
// Always reset the counters regardless of whether the volume changes or
// not.
frames_since_update_input_volume_ = 0;
speech_frames_since_update_input_volume_ = 0;
// Update the input volume if allowed.
if (!is_first_frame_ && speech_ratio >= speech_ratio_threshold_ &&
rms_error_db.has_value()) {
UpdateInputVolume(*rms_error_db);
}
}
is_first_frame_ = false;
}
void MonoInputVolumeController::HandleClipping(int clipped_level_step) {
RTC_DCHECK_GT(clipped_level_step, 0);
// Always decrease the maximum input volume, even if the current input volume
// is below threshold.
SetMaxLevel(std::max(min_input_volume_after_clipping_,
max_input_volume_ - clipped_level_step));
if (log_to_histograms_) {
RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.AgcClippingAdjustmentAllowed",
last_recommended_input_volume_ - clipped_level_step >=
min_input_volume_after_clipping_);
}
if (last_recommended_input_volume_ > min_input_volume_after_clipping_) {
// Don't try to adjust the input volume if we're already below the limit. As
// a consequence, if the user has brought the input volume above the limit,
// we will still not react until the postproc updates the input volume.
SetInputVolume(
std::max(min_input_volume_after_clipping_,
last_recommended_input_volume_ - clipped_level_step));
frames_since_update_input_volume_ = 0;
speech_frames_since_update_input_volume_ = 0;
is_first_frame_ = false;
}
}
void MonoInputVolumeController::SetInputVolume(int new_volume) {
int applied_input_volume = recommended_input_volume_;
if (applied_input_volume == 0) {
RTC_DLOG(LS_INFO)
<< "[AGC2] The applied input volume is zero, taking no action.";
return;
}
if (applied_input_volume < 0 || applied_input_volume > kMaxInputVolume) {
RTC_LOG(LS_ERROR) << "[AGC2] Invalid value for the applied input volume: "
<< applied_input_volume;
return;
}
// Detect manual input volume adjustments by checking if the
// `applied_input_volume` is outside of the `[last_recommended_input_volume_ -
// kVolumeQuantizationSlack, last_recommended_input_volume_ +
// kVolumeQuantizationSlack]` range.
if (applied_input_volume >
last_recommended_input_volume_ + kVolumeQuantizationSlack ||
applied_input_volume <
last_recommended_input_volume_ - kVolumeQuantizationSlack) {
RTC_DLOG(LS_INFO)
<< "[AGC2] The input volume was manually adjusted. Updating "
"stored input volume from "
<< last_recommended_input_volume_ << " to " << applied_input_volume;
last_recommended_input_volume_ = applied_input_volume;
// Always allow the user to increase the volume.
if (last_recommended_input_volume_ > max_input_volume_) {
SetMaxLevel(last_recommended_input_volume_);
}
// Take no action in this case, since we can't be sure when the volume
// was manually adjusted.
frames_since_update_input_volume_ = 0;
speech_frames_since_update_input_volume_ = 0;
is_first_frame_ = false;
return;
}
new_volume = std::min(new_volume, max_input_volume_);
if (new_volume == last_recommended_input_volume_) {
return;
}
recommended_input_volume_ = new_volume;
RTC_DLOG(LS_INFO) << "[AGC2] Applied input volume: " << applied_input_volume
<< " | last recommended input volume: "
<< last_recommended_input_volume_
<< " | newly recommended input volume: " << new_volume;
last_recommended_input_volume_ = new_volume;
}
void MonoInputVolumeController::SetMaxLevel(int input_volume) {
RTC_DCHECK_GE(input_volume, min_input_volume_after_clipping_);
max_input_volume_ = input_volume;
RTC_DLOG(LS_INFO) << "[AGC2] Maximum input volume updated: "
<< max_input_volume_;
}
void MonoInputVolumeController::HandleCaptureOutputUsedChange(
bool capture_output_used) {
if (capture_output_used_ == capture_output_used) {
return;
}
capture_output_used_ = capture_output_used;
if (capture_output_used) {
// When we start using the output, we should reset things to be safe.
check_volume_on_next_process_ = true;
}
}
int MonoInputVolumeController::CheckVolumeAndReset() {
int input_volume = recommended_input_volume_;
// Reasons for taking action at startup:
// 1) A person starting a call is expected to be heard.
// 2) Independent of interpretation of `input_volume` == 0 we should raise it
// so the AGC can do its job properly.
if (input_volume == 0 && !startup_) {
RTC_DLOG(LS_INFO)
<< "[AGC2] The applied input volume is zero, taking no action.";
return 0;
}
if (input_volume < 0 || input_volume > kMaxInputVolume) {
RTC_LOG(LS_ERROR) << "[AGC2] Invalid value for the applied input volume: "
<< input_volume;
return -1;
}
RTC_DLOG(LS_INFO) << "[AGC2] Initial input volume: " << input_volume;
if (input_volume < min_input_volume_) {
input_volume = min_input_volume_;
RTC_DLOG(LS_INFO)
<< "[AGC2] The initial input volume is too low, raising to "
<< input_volume;
recommended_input_volume_ = input_volume;
}
last_recommended_input_volume_ = input_volume;
startup_ = false;
frames_since_update_input_volume_ = 0;
speech_frames_since_update_input_volume_ = 0;
is_first_frame_ = true;
return 0;
}
void MonoInputVolumeController::UpdateInputVolume(int rms_error_db) {
RTC_DLOG(LS_INFO) << "[AGC2] RMS error: " << rms_error_db << " dB";
// Prevent too large microphone input volume changes by clamping the RMS
// error.
rms_error_db =
rtc::SafeClamp(rms_error_db, -KMaxAbsRmsErrorDbfs, KMaxAbsRmsErrorDbfs);
if (rms_error_db == 0) {
return;
}
SetInputVolume(ComputeVolumeUpdate(
rms_error_db, last_recommended_input_volume_, min_input_volume_));
}
InputVolumeController::InputVolumeController(int num_capture_channels,
const Config& config)
: num_capture_channels_(num_capture_channels),
min_input_volume_(config.min_input_volume),
capture_output_used_(true),
clipped_level_step_(config.clipped_level_step),
clipped_ratio_threshold_(config.clipped_ratio_threshold),
clipped_wait_frames_(config.clipped_wait_frames),
clipping_predictor_(CreateClippingPredictor(
num_capture_channels,
CreateClippingPredictorConfig(config.enable_clipping_predictor))),
use_clipping_predictor_step_(
!!clipping_predictor_ &&
CreateClippingPredictorConfig(config.enable_clipping_predictor)
.use_predicted_step),
frames_since_clipped_(config.clipped_wait_frames),
clipping_rate_log_counter_(0),
clipping_rate_log_(0.0f),
target_range_max_dbfs_(config.target_range_max_dbfs),
target_range_min_dbfs_(config.target_range_min_dbfs),
channel_controllers_(num_capture_channels) {
RTC_LOG(LS_INFO)
<< "[AGC2] Input volume controller enabled. Minimum input volume: "
<< min_input_volume_;
for (auto& controller : channel_controllers_) {
controller = std::make_unique<MonoInputVolumeController>(
config.clipped_level_min, min_input_volume_,
config.update_input_volume_wait_frames,
config.speech_probability_threshold, config.speech_ratio_threshold);
}
RTC_DCHECK(!channel_controllers_.empty());
RTC_DCHECK_GT(clipped_level_step_, 0);
RTC_DCHECK_LE(clipped_level_step_, 255);
RTC_DCHECK_GT(clipped_ratio_threshold_, 0.0f);
RTC_DCHECK_LT(clipped_ratio_threshold_, 1.0f);
RTC_DCHECK_GT(clipped_wait_frames_, 0);
channel_controllers_[0]->ActivateLogging();
}
InputVolumeController::~InputVolumeController() {}
void InputVolumeController::Initialize() {
for (auto& controller : channel_controllers_) {
controller->Initialize();
}
capture_output_used_ = true;
AggregateChannelLevels();
clipping_rate_log_ = 0.0f;
clipping_rate_log_counter_ = 0;
applied_input_volume_ = absl::nullopt;
}
void InputVolumeController::AnalyzeInputAudio(int applied_input_volume,
const AudioBuffer& audio_buffer) {
RTC_DCHECK_GE(applied_input_volume, 0);
RTC_DCHECK_LE(applied_input_volume, 255);
SetAppliedInputVolume(applied_input_volume);
RTC_DCHECK_EQ(audio_buffer.num_channels(), channel_controllers_.size());
const float* const* audio = audio_buffer.channels_const();
size_t samples_per_channel = audio_buffer.num_frames();
RTC_DCHECK(audio);
AggregateChannelLevels();
if (!capture_output_used_) {
return;
}
if (!!clipping_predictor_) {
AudioFrameView<const float> frame = AudioFrameView<const float>(
audio, num_capture_channels_, static_cast<int>(samples_per_channel));
clipping_predictor_->Analyze(frame);
}
// Check for clipped samples. We do this in the preprocessing phase in order
// to catch clipped echo as well.
//
// If we find a sufficiently clipped frame, drop the current microphone
// input volume and enforce a new maximum input volume, dropped the same
// amount from the current maximum. This harsh treatment is an effort to avoid
// repeated clipped echo events.
float clipped_ratio =
ComputeClippedRatio(audio, num_capture_channels_, samples_per_channel);
clipping_rate_log_ = std::max(clipped_ratio, clipping_rate_log_);
clipping_rate_log_counter_++;
constexpr int kNumFramesIn30Seconds = 3000;
if (clipping_rate_log_counter_ == kNumFramesIn30Seconds) {
LogClippingMetrics(std::round(100.0f * clipping_rate_log_));
clipping_rate_log_ = 0.0f;
clipping_rate_log_counter_ = 0;
}
if (frames_since_clipped_ < clipped_wait_frames_) {
++frames_since_clipped_;
return;
}
const bool clipping_detected = clipped_ratio > clipped_ratio_threshold_;
bool clipping_predicted = false;
int predicted_step = 0;
if (!!clipping_predictor_) {
for (int channel = 0; channel < num_capture_channels_; ++channel) {
const auto step = clipping_predictor_->EstimateClippedLevelStep(
channel, recommended_input_volume_, clipped_level_step_,
channel_controllers_[channel]->min_input_volume_after_clipping(),
kMaxInputVolume);
if (step.has_value()) {
predicted_step = std::max(predicted_step, step.value());
clipping_predicted = true;
}
}
}
if (clipping_detected) {
RTC_DLOG(LS_INFO) << "[AGC2] Clipping detected (ratio: " << clipped_ratio
<< ")";
}
int step = clipped_level_step_;
if (clipping_predicted) {
predicted_step = std::max(predicted_step, clipped_level_step_);
RTC_DLOG(LS_INFO) << "[AGC2] Clipping predicted (volume down step: "
<< predicted_step << ")";
if (use_clipping_predictor_step_) {
step = predicted_step;
}
}
if (clipping_detected ||
(clipping_predicted && use_clipping_predictor_step_)) {
for (auto& state_ch : channel_controllers_) {
state_ch->HandleClipping(step);
}
frames_since_clipped_ = 0;
if (!!clipping_predictor_) {
clipping_predictor_->Reset();
}
}
AggregateChannelLevels();
}
absl::optional<int> InputVolumeController::RecommendInputVolume(
float speech_probability,
absl::optional<float> speech_level_dbfs) {
// Only process if applied input volume is set.
if (!applied_input_volume_.has_value()) {
RTC_LOG(LS_ERROR) << "[AGC2] Applied input volume not set.";
return absl::nullopt;
}
AggregateChannelLevels();
const int volume_after_clipping_handling = recommended_input_volume_;
if (!capture_output_used_) {
return applied_input_volume_;
}
absl::optional<int> rms_error_db;
if (speech_level_dbfs.has_value()) {
// Compute the error for all frames (both speech and non-speech frames).
rms_error_db = GetSpeechLevelRmsErrorDb(
*speech_level_dbfs, target_range_min_dbfs_, target_range_max_dbfs_);
}
for (auto& controller : channel_controllers_) {
controller->Process(rms_error_db, speech_probability);
}
AggregateChannelLevels();
if (volume_after_clipping_handling != recommended_input_volume_) {
// The recommended input volume was adjusted in order to match the target
// level.
UpdateHistogramOnRecommendedInputVolumeChangeToMatchTarget(
recommended_input_volume_);
}
applied_input_volume_ = absl::nullopt;
return recommended_input_volume();
}
void InputVolumeController::HandleCaptureOutputUsedChange(
bool capture_output_used) {
for (auto& controller : channel_controllers_) {
controller->HandleCaptureOutputUsedChange(capture_output_used);
}
capture_output_used_ = capture_output_used;
}
void InputVolumeController::SetAppliedInputVolume(int input_volume) {
applied_input_volume_ = input_volume;
for (auto& controller : channel_controllers_) {
controller->set_stream_analog_level(input_volume);
}
AggregateChannelLevels();
}
void InputVolumeController::AggregateChannelLevels() {
int new_recommended_input_volume =
channel_controllers_[0]->recommended_analog_level();
channel_controlling_gain_ = 0;
for (size_t ch = 1; ch < channel_controllers_.size(); ++ch) {
int input_volume = channel_controllers_[ch]->recommended_analog_level();
if (input_volume < new_recommended_input_volume) {
new_recommended_input_volume = input_volume;
channel_controlling_gain_ = static_cast<int>(ch);
}
}
// Enforce the minimum input volume when a recommendation is made.
if (applied_input_volume_.has_value() && *applied_input_volume_ > 0) {
new_recommended_input_volume =
std::max(new_recommended_input_volume, min_input_volume_);
}
recommended_input_volume_ = new_recommended_input_volume;
}
} // namespace webrtc