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Copy AgcManagerDirect files to agc2 and rename the classes

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Copy AgcManagerDirect files from agc to agc2. Rename the newly
created files and classes ahead of refactoring. Add a build
target.

This change is done to enable creating a class
InputVolumeController based on AgcManagerDirect. The added
temporary dependency on files in agc will be removed
in https://webrtc-review.googlesource.com/c/src/+/278625.

The exact copy of the files happened in the 1st patchset and it
has been verified as follows:

Checksum check:
```
$ git checkout main && git pull
# Go back to the tree state before [1] landed
$ git new-branch tmp
$ git reset --hard 2235776597e2f47ec353ac911428eb9a54d64a10
$ cd modules/audio_processing/agc/
$ md5 agc_manager_direct*
MD5 (agc_manager_direct.cc) = e661481a85f72596cae4599b62907f5b
MD5 (agc_manager_direct.h) = bf68280e2d0f689b4ebcd665b5db6052
MD5 (agc_manager_direct_unittest.cc) = 6bf0bf45ff5e940b1a3bb37154f09269
```

Patchset 1 (see [2])
```
$ cd modules/audio_processing/agc2/
$ md5 input_volume_controlle*
MD5 (input_volume_controller.cc) = e661481a85f72596cae4599b62907f5b
MD5 (input_volume_controller.h) = bf68280e2d0f689b4ebcd665b5db6052
MD5 (input_volume_controller_unittest.cc) = 6bf0bf45ff5e940b1a3bb37154f09269
```

[1] https://webrtc-review.googlesource.com/c/src/+/278781
[2] https://webrtc-review.googlesource.com/c/src/+/278624/1

Bug: webrtc:7494
Change-Id: I7804da899d18adf556b089c76a567ce27c299a62
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278624
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38512}
This commit is contained in:
Hanna Silen 2022-10-18 16:57:36 +02:00 committed by WebRTC LUCI CQ
parent b24ebc535b
commit 7587755d29
4 changed files with 3217 additions and 1 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

34
modules/audio_processing/agc2/BUILD.gn
View File

@ -184,6 +184,38 @@ rtc_source_set("gain_map") {
}
rtc_library("input_volume_controller") {
sources = [
"input_volume_controller.cc",
"input_volume_controller.h",
]
configs += [ "..:apm_debug_dump" ]
deps = [
":clipping_predictor",
":gain_map",
"..:api",
"..:apm_logging",
"..:audio_buffer",
"..:audio_frame_view",
"../../../api:array_view",
"../../../common_audio",
"../../../common_audio:common_audio_c",
"../../../rtc_base:checks",
"../../../rtc_base:gtest_prod",
"../../../rtc_base:logging",
"../../../rtc_base:safe_minmax",
"../../../system_wrappers:field_trial",
"../../../system_wrappers:metrics",
"../agc:gain_control_interface",
"../agc:level_estimation",
"../vad",
]
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
}
rtc_library("speech_probability_buffer") {
sources = [
"speech_probability_buffer.cc",
"speech_probability_buffer.h",
@ -357,7 +389,7 @@ rtc_library("input_volume_controller_unittests") {
deps = [
":clipping_predictor",
":gain_map",
":input_volume_controller",
":speech_probability_buffer",
"../../../rtc_base:checks",
"../../../rtc_base:random",
"../../../rtc_base:safe_conversions",

750
modules/audio_processing/agc2/input_volume_controller.cc Normal file
View File

@ -0,0 +1,750 @@
/*
* 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 "common_audio/include/audio_util.h"
#include "modules/audio_processing/agc/gain_control.h"
#include "modules/audio_processing/agc2/gain_map_internal.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 level (presumably due to OS
// quantization) before we assume the user has manually adjusted the microphone.
constexpr int kLevelQuantizationSlack = 25;
constexpr int kDefaultCompressionGain = 7;
constexpr int kMaxCompressionGain = 12;
constexpr int kMinCompressionGain = 2;
// Controls the rate of compression changes towards the target.
constexpr float kCompressionGainStep = 0.05f;
constexpr int kMaxMicLevel = 255;
static_assert(kGainMapSize > kMaxMicLevel, "gain map too small");
constexpr int kMinMicLevel = 12;
// Prevent very large microphone level changes.
constexpr int kMaxResidualGainChange = 15;
// Maximum additional gain allowed to compensate for microphone level
// restrictions from clipping events.
constexpr int kSurplusCompressionGain = 6;
// Target speech level (dBFs) and speech probability threshold used to compute
// the RMS error override in `GetSpeechLevelErrorDb()`. These are only used for
// computing the error override and they are not passed to `agc_`.
// TODO(webrtc:7494): Move these to a config and pass in the ctor.
constexpr float kOverrideTargetSpeechLevelDbfs = -18.0f;
constexpr float kOverrideSpeechProbabilitySilenceThreshold = 0.5f;
// The minimum number of frames between `UpdateGain()` calls.
// TODO(webrtc:7494): Move this to a config and pass in the ctor with
// kOverrideWaitFrames = 100. Default value zero needed for the unit tests.
constexpr int kOverrideWaitFrames = 0;
using AnalogAgcConfig =
AudioProcessing::Config::GainController1::AnalogGainController;
// Returns whether a fall-back solution to choose the maximum level should be
// chosen.
bool UseMaxAnalogChannelLevel() {
return field_trial::IsEnabled("WebRTC-UseMaxAnalogAgcChannelLevel");
}
// If the "WebRTC-Audio-2ndAgcMinMicLevelExperiment" field trial is specified,
// parses it and returns a value between 0 and 255 depending on the field-trial
// string. Returns an unspecified value if the field trial is not specified, if
// disabled or if it cannot be parsed. Example:
// 'WebRTC-Audio-2ndAgcMinMicLevelExperiment/Enabled-80' => returns 80.
absl::optional<int> GetMinMicLevelOverride() {
constexpr char kMinMicLevelFieldTrial[] =
"WebRTC-Audio-2ndAgcMinMicLevelExperiment";
if (!webrtc::field_trial::IsEnabled(kMinMicLevelFieldTrial)) {
return absl::nullopt;
}
const auto field_trial_string =
webrtc::field_trial::FindFullName(kMinMicLevelFieldTrial);
int min_mic_level = -1;
sscanf(field_trial_string.c_str(), "Enabled-%d", &min_mic_level);
if (min_mic_level >= 0 && min_mic_level <= 255) {
return min_mic_level;
} else {
RTC_LOG(LS_WARNING) << "[agc] Invalid parameter for "
<< kMinMicLevelFieldTrial << ", ignored.";
return absl::nullopt;
}
}
int ClampLevel(int mic_level, int min_mic_level) {
return rtc::SafeClamp(mic_level, min_mic_level, kMaxMicLevel);
}
int LevelFromGainError(int gain_error, int level, int min_mic_level) {
RTC_DCHECK_GE(level, 0);
RTC_DCHECK_LE(level, kMaxMicLevel);
if (gain_error == 0) {
return level;
}
int new_level = level;
if (gain_error > 0) {
while (kGainMap[new_level] - kGainMap[level] < gain_error &&
new_level < kMaxMicLevel) {
++new_level;
}
} else {
while (kGainMap[new_level] - kGainMap[level] > gain_error &&
new_level > min_mic_level) {
--new_level;
}
}
return new_level;
}
// 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) << "Input clipping rate: " << clipping_rate << "%";
RTC_HISTOGRAM_COUNTS_LINEAR(/*name=*/"WebRTC.Audio.Agc.InputClippingRate",
/*sample=*/clipping_rate, /*min=*/0, /*max=*/100,
/*bucket_count=*/50);
}
// Computes the speech level error in dB. `speech_level_dbfs` is required to be
// in the range [-90.0f, 30.0f] and `speech_probability` in the range
// [0.0f, 1.0f].
int GetSpeechLevelErrorDb(float speech_level_dbfs, float speech_probability) {
constexpr float kMinSpeechLevelDbfs = -90.0f;
constexpr float kMaxSpeechLevelDbfs = 30.0f;
RTC_DCHECK_GE(speech_level_dbfs, kMinSpeechLevelDbfs);
RTC_DCHECK_LE(speech_level_dbfs, kMaxSpeechLevelDbfs);
RTC_DCHECK_GE(speech_probability, 0.0f);
RTC_DCHECK_LE(speech_probability, 1.0f);
if (speech_probability < kOverrideSpeechProbabilitySilenceThreshold) {
return 0;
}
const float speech_level = rtc::SafeClamp<float>(
speech_level_dbfs, kMinSpeechLevelDbfs, kMaxSpeechLevelDbfs);
return std::round(kOverrideTargetSpeechLevelDbfs - speech_level);
}
} // namespace
RecommendedInputVolumeEstimator::RecommendedInputVolumeEstimator(
ApmDataDumper* data_dumper,
int startup_min_level,
int clipped_level_min,
bool disable_digital_adaptive,
int min_mic_level)
: min_mic_level_(min_mic_level),
disable_digital_adaptive_(disable_digital_adaptive),
agc_(std::make_unique<Agc>()),
max_level_(kMaxMicLevel),
max_compression_gain_(kMaxCompressionGain),
target_compression_(kDefaultCompressionGain),
compression_(target_compression_),
compression_accumulator_(compression_),
startup_min_level_(ClampLevel(startup_min_level, min_mic_level_)),
clipped_level_min_(clipped_level_min) {}
RecommendedInputVolumeEstimator::~RecommendedInputVolumeEstimator() = default;
void RecommendedInputVolumeEstimator::Initialize() {
max_level_ = kMaxMicLevel;
max_compression_gain_ = kMaxCompressionGain;
target_compression_ = disable_digital_adaptive_ ? 0 : kDefaultCompressionGain;
compression_ = disable_digital_adaptive_ ? 0 : target_compression_;
compression_accumulator_ = compression_;
capture_output_used_ = true;
check_volume_on_next_process_ = true;
frames_since_update_gain_ = 0;
is_first_frame_ = true;
}
void RecommendedInputVolumeEstimator::Process(
rtc::ArrayView<const int16_t> audio,
absl::optional<int> rms_error_override) {
new_compression_to_set_ = absl::nullopt;
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();
}
agc_->Process(audio);
// Always check if `agc_` has a new error available. If yes, `agc_` gets
// reset.
// TODO(webrtc:7494) Replace the `agc_` call `GetRmsErrorDb()` with `Reset()`
// if an error override is used.
int rms_error = 0;
bool update_gain = agc_->GetRmsErrorDb(&rms_error);
if (rms_error_override.has_value()) {
if (is_first_frame_ || frames_since_update_gain_ < kOverrideWaitFrames) {
update_gain = false;
} else {
rms_error = *rms_error_override;
update_gain = true;
}
}
if (update_gain) {
UpdateGain(rms_error);
}
if (!disable_digital_adaptive_) {
UpdateCompressor();
}
is_first_frame_ = false;
if (frames_since_update_gain_ < kOverrideWaitFrames) {
++frames_since_update_gain_;
}
}
void RecommendedInputVolumeEstimator::HandleClipping(int clipped_level_step) {
RTC_DCHECK_GT(clipped_level_step, 0);
// Always decrease the maximum level, even if the current level is below
// threshold.
SetMaxLevel(std::max(clipped_level_min_, max_level_ - clipped_level_step));
if (log_to_histograms_) {
RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.AgcClippingAdjustmentAllowed",
level_ - clipped_level_step >= clipped_level_min_);
}
if (level_ > clipped_level_min_) {
// Don't try to adjust the level if we're already below the limit. As
// a consequence, if the user has brought the level above the limit, we
// will still not react until the postproc updates the level.
SetLevel(std::max(clipped_level_min_, level_ - clipped_level_step));
// Reset the AGCs for all channels since the level has changed.
agc_->Reset();
frames_since_update_gain_ = 0;
is_first_frame_ = false;
}
}
void RecommendedInputVolumeEstimator::SetLevel(int new_level) {
int voe_level = recommended_input_volume_;
if (voe_level == 0) {
RTC_DLOG(LS_INFO)
<< "[agc] VolumeCallbacks returned level=0, taking no action.";
return;
}
if (voe_level < 0 || voe_level > kMaxMicLevel) {
RTC_LOG(LS_ERROR) << "VolumeCallbacks returned an invalid level="
<< voe_level;
return;
}
// Detect manual input volume adjustments by checking if the current level
// `voe_level` is outside of the `[level_ - kLevelQuantizationSlack, level_ +
// kLevelQuantizationSlack]` range where `level_` is the last input volume
// known by this gain controller.
if (voe_level > level_ + kLevelQuantizationSlack ||
voe_level < level_ - kLevelQuantizationSlack) {
RTC_DLOG(LS_INFO) << "[agc] Mic volume was manually adjusted. Updating "
"stored level from "
<< level_ << " to " << voe_level;
level_ = voe_level;
// Always allow the user to increase the volume.
if (level_ > max_level_) {
SetMaxLevel(level_);
}
// Take no action in this case, since we can't be sure when the volume
// was manually adjusted. The compressor will still provide some of the
// desired gain change.
agc_->Reset();
frames_since_update_gain_ = 0;
is_first_frame_ = false;
return;
}
new_level = std::min(new_level, max_level_);
if (new_level == level_) {
return;
}
recommended_input_volume_ = new_level;
RTC_DLOG(LS_INFO) << "[agc] voe_level=" << voe_level << ", level_=" << level_
<< ", new_level=" << new_level;
level_ = new_level;
}
void RecommendedInputVolumeEstimator::SetMaxLevel(int level) {
RTC_DCHECK_GE(level, clipped_level_min_);
max_level_ = level;
// Scale the `kSurplusCompressionGain` linearly across the restricted
// level range.
max_compression_gain_ =
kMaxCompressionGain + std::floor((1.f * kMaxMicLevel - max_level_) /
(kMaxMicLevel - clipped_level_min_) *
kSurplusCompressionGain +
0.5f);
RTC_DLOG(LS_INFO) << "[agc] max_level_=" << max_level_
<< ", max_compression_gain_=" << max_compression_gain_;
}
void RecommendedInputVolumeEstimator::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 RecommendedInputVolumeEstimator::CheckVolumeAndReset() {
int level = 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 `level` == 0 we should raise it so the
// AGC can do its job properly.
if (level == 0 && !startup_) {
RTC_DLOG(LS_INFO)
<< "[agc] VolumeCallbacks returned level=0, taking no action.";
return 0;
}
if (level < 0 || level > kMaxMicLevel) {
RTC_LOG(LS_ERROR) << "[agc] VolumeCallbacks returned an invalid level="
<< level;
return -1;
}
RTC_DLOG(LS_INFO) << "[agc] Initial GetMicVolume()=" << level;
int minLevel = startup_ ? startup_min_level_ : min_mic_level_;
if (level < minLevel) {
level = minLevel;
RTC_DLOG(LS_INFO) << "[agc] Initial volume too low, raising to " << level;
recommended_input_volume_ = level;
}
agc_->Reset();
level_ = level;
startup_ = false;
frames_since_update_gain_ = 0;
is_first_frame_ = true;
return 0;
}
// Distributes the required gain change between the digital compression stage
// and volume slider. We use the compressor first, providing a slack region
// around the current slider position to reduce movement.
//
// If the slider needs to be moved, we check first if the user has adjusted
// it, in which case we take no action and cache the updated level.
void RecommendedInputVolumeEstimator::UpdateGain(int rms_error_db) {
int rms_error = rms_error_db;
// Always reset the counter regardless of whether the gain is changed
// or not. This matches with the bahvior of `agc_` where the histogram is
// reset every time an RMS error is successfully read.
frames_since_update_gain_ = 0;
// The compressor will always add at least kMinCompressionGain. In effect,
// this adjusts our target gain upward by the same amount and rms_error
// needs to reflect that.
rms_error += kMinCompressionGain;
// Handle as much error as possible with the compressor first.
int raw_compression =
rtc::SafeClamp(rms_error, kMinCompressionGain, max_compression_gain_);
// Deemphasize the compression gain error. Move halfway between the current
// target and the newly received target. This serves to soften perceptible
// intra-talkspurt adjustments, at the cost of some adaptation speed.
if ((raw_compression == max_compression_gain_ &&
target_compression_ == max_compression_gain_ - 1) ||
(raw_compression == kMinCompressionGain &&
target_compression_ == kMinCompressionGain + 1)) {
// Special case to allow the target to reach the endpoints of the
// compression range. The deemphasis would otherwise halt it at 1 dB shy.
target_compression_ = raw_compression;
} else {
target_compression_ =
(raw_compression - target_compression_) / 2 + target_compression_;
}
// Residual error will be handled by adjusting the volume slider. Use the
// raw rather than deemphasized compression here as we would otherwise
// shrink the amount of slack the compressor provides.
const int residual_gain =
rtc::SafeClamp(rms_error - raw_compression, -kMaxResidualGainChange,
kMaxResidualGainChange);
RTC_DLOG(LS_INFO) << "[agc] rms_error=" << rms_error
<< ", target_compression=" << target_compression_
<< ", residual_gain=" << residual_gain;
if (residual_gain == 0)
return;
int old_level = level_;
SetLevel(LevelFromGainError(residual_gain, level_, min_mic_level_));
if (old_level != level_) {
// level_ was updated by SetLevel; log the new value.
RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.AgcSetLevel", level_, 1,
kMaxMicLevel, 50);
// Reset the AGC since the level has changed.
agc_->Reset();
}
}
void RecommendedInputVolumeEstimator::UpdateCompressor() {
calls_since_last_gain_log_++;
if (calls_since_last_gain_log_ == 100) {
calls_since_last_gain_log_ = 0;
RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.Agc.DigitalGainApplied",
compression_, 0, kMaxCompressionGain,
kMaxCompressionGain + 1);
}
if (compression_ == target_compression_) {
return;
}
// Adapt the compression gain slowly towards the target, in order to avoid
// highly perceptible changes.
if (target_compression_ > compression_) {
compression_accumulator_ += kCompressionGainStep;
} else {
compression_accumulator_ -= kCompressionGainStep;
}
// The compressor accepts integer gains in dB. Adjust the gain when
// we've come within half a stepsize of the nearest integer. (We don't
// check for equality due to potential floating point imprecision).
int new_compression = compression_;
int nearest_neighbor = std::floor(compression_accumulator_ + 0.5);
if (std::fabs(compression_accumulator_ - nearest_neighbor) <
kCompressionGainStep / 2) {
new_compression = nearest_neighbor;
}
// Set the new compression gain.
if (new_compression != compression_) {
RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.Agc.DigitalGainUpdated",
new_compression, 0, kMaxCompressionGain,
kMaxCompressionGain + 1);
compression_ = new_compression;
compression_accumulator_ = new_compression;
new_compression_to_set_ = compression_;
}
}
std::atomic<int> InputVolumeController::instance_counter_(0);
InputVolumeController::InputVolumeController(
const AudioProcessing::Config::GainController1::AnalogGainController&
analog_config,
Agc* agc)
: InputVolumeController(/*num_capture_channels=*/1, analog_config) {
RTC_DCHECK(channel_agcs_[0]);
RTC_DCHECK(agc);
channel_agcs_[0]->set_agc(agc);
}
InputVolumeController::InputVolumeController(
int num_capture_channels,
const AnalogAgcConfig& analog_config)
: analog_controller_enabled_(analog_config.enabled),
min_mic_level_override_(GetMinMicLevelOverride()),
data_dumper_(new ApmDataDumper(instance_counter_.fetch_add(1) + 1)),
use_min_channel_level_(!UseMaxAnalogChannelLevel()),
num_capture_channels_(num_capture_channels),
disable_digital_adaptive_(!analog_config.enable_digital_adaptive),
frames_since_clipped_(analog_config.clipped_wait_frames),
capture_output_used_(true),
clipped_level_step_(analog_config.clipped_level_step),
clipped_ratio_threshold_(analog_config.clipped_ratio_threshold),
clipped_wait_frames_(analog_config.clipped_wait_frames),
channel_agcs_(num_capture_channels),
new_compressions_to_set_(num_capture_channels),
clipping_predictor_(
CreateClippingPredictor(num_capture_channels,
analog_config.clipping_predictor)),
use_clipping_predictor_step_(
!!clipping_predictor_ &&
analog_config.clipping_predictor.use_predicted_step),
clipping_rate_log_(0.0f),
clipping_rate_log_counter_(0) {
RTC_LOG(LS_INFO) << "[agc] analog controller enabled: "
<< (analog_controller_enabled_ ? "yes" : "no");
const int min_mic_level = min_mic_level_override_.value_or(kMinMicLevel);
RTC_LOG(LS_INFO) << "[agc] Min mic level: " << min_mic_level
<< " (overridden: "
<< (min_mic_level_override_.has_value() ? "yes" : "no")
<< ")";
RTC_LOG(LS_INFO) << "[agc] Startup min volume: "
<< analog_config.startup_min_volume;
for (size_t ch = 0; ch < channel_agcs_.size(); ++ch) {
ApmDataDumper* data_dumper_ch = ch == 0 ? data_dumper_.get() : nullptr;
channel_agcs_[ch] = std::make_unique<RecommendedInputVolumeEstimator>(
data_dumper_ch, analog_config.startup_min_volume,
analog_config.clipped_level_min, disable_digital_adaptive_,
min_mic_level);
}
RTC_DCHECK(!channel_agcs_.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_agcs_[0]->ActivateLogging();
}
InputVolumeController::~InputVolumeController() {}
void InputVolumeController::Initialize() {
RTC_DLOG(LS_INFO) << "InputVolumeController::Initialize";
data_dumper_->InitiateNewSetOfRecordings();
for (size_t ch = 0; ch < channel_agcs_.size(); ++ch) {
channel_agcs_[ch]->Initialize();
}
capture_output_used_ = true;
AggregateChannelLevels();
clipping_rate_log_ = 0.0f;
clipping_rate_log_counter_ = 0;
}
void InputVolumeController::SetupDigitalGainControl(
GainControl& gain_control) const {
if (gain_control.set_mode(GainControl::kFixedDigital) != 0) {
RTC_LOG(LS_ERROR) << "set_mode(GainControl::kFixedDigital) failed.";
}
const int target_level_dbfs = disable_digital_adaptive_ ? 0 : 2;
if (gain_control.set_target_level_dbfs(target_level_dbfs) != 0) {
RTC_LOG(LS_ERROR) << "set_target_level_dbfs() failed.";
}
const int compression_gain_db =
disable_digital_adaptive_ ? 0 : kDefaultCompressionGain;
if (gain_control.set_compression_gain_db(compression_gain_db) != 0) {
RTC_LOG(LS_ERROR) << "set_compression_gain_db() failed.";
}
const bool enable_limiter = !disable_digital_adaptive_;
if (gain_control.enable_limiter(enable_limiter) != 0) {
RTC_LOG(LS_ERROR) << "enable_limiter() failed.";
}
}
void InputVolumeController::AnalyzePreProcess(const AudioBuffer& audio_buffer) {
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, as the AGC has difficulty detecting pitch
// under clipping distortion. 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 level
// and enforce a new maximum level, dropped the same amount from the current
// maximum. This harsh treatment is an effort to avoid repeated clipped echo
// events. As compensation for this restriction, the maximum compression
// gain is increased, through SetMaxLevel().
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_agcs_[channel]->min_mic_level(), kMaxMicLevel);
if (step.has_value()) {
predicted_step = std::max(predicted_step, step.value());
clipping_predicted = true;
}
}
}
if (clipping_detected) {
RTC_DLOG(LS_INFO) << "[agc] Clipping detected. clipped_ratio="
<< clipped_ratio;
}
int step = clipped_level_step_;
if (clipping_predicted) {
predicted_step = std::max(predicted_step, clipped_level_step_);
RTC_DLOG(LS_INFO) << "[agc] Clipping predicted. 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_agcs_) {
state_ch->HandleClipping(step);
}
frames_since_clipped_ = 0;
if (!!clipping_predictor_) {
clipping_predictor_->Reset();
}
}
AggregateChannelLevels();
}
void InputVolumeController::Process(const AudioBuffer& audio_buffer) {
Process(audio_buffer, /*speech_probability=*/absl::nullopt,
/*speech_level_dbfs=*/absl::nullopt);
}
void InputVolumeController::Process(const AudioBuffer& audio_buffer,
absl::optional<float> speech_probability,
absl::optional<float> speech_level_dbfs) {
AggregateChannelLevels();
if (!capture_output_used_) {
return;
}
const size_t num_frames_per_band = audio_buffer.num_frames_per_band();
absl::optional<int> rms_error_override = absl::nullopt;
if (speech_probability.has_value() && speech_level_dbfs.has_value()) {
rms_error_override =
GetSpeechLevelErrorDb(*speech_level_dbfs, *speech_probability);
}
for (size_t ch = 0; ch < channel_agcs_.size(); ++ch) {
std::array<int16_t, AudioBuffer::kMaxSampleRate / 100> audio_data;
int16_t* audio_use = audio_data.data();
FloatS16ToS16(audio_buffer.split_bands_const_f(ch)[0], num_frames_per_band,
audio_use);
channel_agcs_[ch]->Process({audio_use, num_frames_per_band},
rms_error_override);
new_compressions_to_set_[ch] = channel_agcs_[ch]->new_compression();
}
AggregateChannelLevels();
}
absl::optional<int> InputVolumeController::GetDigitalComressionGain() {
return new_compressions_to_set_[channel_controlling_gain_];
}
void InputVolumeController::HandleCaptureOutputUsedChange(
bool capture_output_used) {
for (size_t ch = 0; ch < channel_agcs_.size(); ++ch) {
channel_agcs_[ch]->HandleCaptureOutputUsedChange(capture_output_used);
}
capture_output_used_ = capture_output_used;
}
float InputVolumeController::voice_probability() const {
float max_prob = 0.f;
for (const auto& state_ch : channel_agcs_) {
max_prob = std::max(max_prob, state_ch->voice_probability());
}
return max_prob;
}
void InputVolumeController::set_stream_analog_level(int level) {
if (!analog_controller_enabled_) {
recommended_input_volume_ = level;
}
for (size_t ch = 0; ch < channel_agcs_.size(); ++ch) {
channel_agcs_[ch]->set_stream_analog_level(level);
}
AggregateChannelLevels();
}
void InputVolumeController::AggregateChannelLevels() {
int new_recommended_input_volume =
channel_agcs_[0]->recommended_analog_level();
channel_controlling_gain_ = 0;
if (use_min_channel_level_) {
for (size_t ch = 1; ch < channel_agcs_.size(); ++ch) {
int level = channel_agcs_[ch]->recommended_analog_level();
if (level < new_recommended_input_volume) {
new_recommended_input_volume = level;
channel_controlling_gain_ = static_cast<int>(ch);
}
}
} else {
for (size_t ch = 1; ch < channel_agcs_.size(); ++ch) {
int level = channel_agcs_[ch]->recommended_analog_level();
if (level > new_recommended_input_volume) {
new_recommended_input_volume = level;
channel_controlling_gain_ = static_cast<int>(ch);
}
}
}
if (min_mic_level_override_.has_value() && new_recommended_input_volume > 0) {
new_recommended_input_volume =
std::max(new_recommended_input_volume, *min_mic_level_override_);
}
if (analog_controller_enabled_) {
recommended_input_volume_ = new_recommended_input_volume;
}
}
} // namespace webrtc

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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.
*/
#ifndef MODULES_AUDIO_PROCESSING_AGC2_INPUT_VOLUME_CONTROLLER_H_
#define MODULES_AUDIO_PROCESSING_AGC2_INPUT_VOLUME_CONTROLLER_H_
#include <atomic>
#include <memory>
#include <vector>
#include "absl/types/optional.h"
#include "api/array_view.h"
#include "modules/audio_processing/agc/agc.h"
#include "modules/audio_processing/agc2/clipping_predictor.h"
#include "modules/audio_processing/audio_buffer.h"
#include "modules/audio_processing/include/audio_processing.h"
#include "modules/audio_processing/logging/apm_data_dumper.h"
#include "rtc_base/gtest_prod_util.h"
namespace webrtc {
class RecommendedInputVolumeEstimator;
class GainControl;
// Adaptive Gain Controller (AGC) that controls the input volume and a digital
// gain. The input volume controller recommends what volume to use, handles
// volume changes and clipping. In particular, it handles changes triggered by
// the user (e.g., volume set to zero by a HW mute button). The digital
// controller chooses and applies the digital compression gain.
// This class is not thread-safe.
// TODO(bugs.webrtc.org/7494): Use applied/recommended input volume naming
// convention.
class InputVolumeController final {
public:
// Ctor. `num_capture_channels` specifies the number of channels for the audio
// passed to `AnalyzePreProcess()` and `Process()`. Clamps
// `analog_config.startup_min_level` in the [12, 255] range.
InputVolumeController(
int num_capture_channels,
const AudioProcessing::Config::GainController1::AnalogGainController&
analog_config);
~InputVolumeController();
InputVolumeController(const InputVolumeController&) = delete;
InputVolumeController& operator=(const InputVolumeController&) = delete;
void Initialize();
// Configures `gain_control` to work as a fixed digital controller so that the
// adaptive part is only handled by this gain controller. Must be called if
// `gain_control` is also used to avoid the side-effects of running two AGCs.
void SetupDigitalGainControl(GainControl& gain_control) const;
// Sets the applied input volume.
void set_stream_analog_level(int level);
// TODO(bugs.webrtc.org/7494): Add argument for the applied input volume and
// remove `set_stream_analog_level()`.
// Analyzes `audio` before `Process()` is called so that the analysis can be
// performed before external digital processing operations take place (e.g.,
// echo cancellation). The analysis consists of input clipping detection and
// prediction (if enabled). Must be called after `set_stream_analog_level()`.
void AnalyzePreProcess(const AudioBuffer& audio_buffer);
// Processes `audio_buffer`. Chooses a digital compression gain and the new
// input volume to recommend. Must be called after `AnalyzePreProcess()`. If
// `speech_probability` (range [0.0f, 1.0f]) and `speech_level_dbfs` (range
// [-90.f, 30.0f]) are given, uses them to override the estimated RMS error.
// TODO(webrtc:7494): This signature is needed for testing purposes, unify
// the signatures when the clean-up is done.
void Process(const AudioBuffer& audio_buffer,
absl::optional<float> speech_probability,
absl::optional<float> speech_level_dbfs);
// Processes `audio_buffer`. Chooses a digital compression gain and the new
// input volume to recommend. Must be called after `AnalyzePreProcess()`.
void Process(const AudioBuffer& audio_buffer);
// TODO(bugs.webrtc.org/7494): Return recommended input volume and remove
// `recommended_analog_level()`.
// Returns the recommended input volume. If the input volume contoller is
// disabled, returns the input volume set via the latest
// `set_stream_analog_level()` call. Must be called after
// `AnalyzePreProcess()` and `Process()`.
int recommended_analog_level() const { return recommended_input_volume_; }
// Call when the capture stream output has been flagged to be used/not-used.
// If unused, the manager disregards all incoming audio.
void HandleCaptureOutputUsedChange(bool capture_output_used);
float voice_probability() const;
int num_channels() const { return num_capture_channels_; }
// If available, returns the latest digital compression gain that has been
// chosen.
absl::optional<int> GetDigitalComressionGain();
// Returns true if clipping prediction is enabled.
bool clipping_predictor_enabled() const { return !!clipping_predictor_; }
// Returns true if clipping prediction is used to adjust the input volume.
bool use_clipping_predictor_step() const {
return use_clipping_predictor_step_;
}
private:
friend class InputVolumeControllerTestHelper;
FRIEND_TEST_ALL_PREFIXES(InputVolumeControllerTest,
DisableDigitalDisablesDigital);
FRIEND_TEST_ALL_PREFIXES(InputVolumeControllerTest,
AgcMinMicLevelExperimentDefault);
FRIEND_TEST_ALL_PREFIXES(InputVolumeControllerTest,
AgcMinMicLevelExperimentDisabled);
FRIEND_TEST_ALL_PREFIXES(InputVolumeControllerTest,
AgcMinMicLevelExperimentOutOfRangeAbove);
FRIEND_TEST_ALL_PREFIXES(InputVolumeControllerTest,
AgcMinMicLevelExperimentOutOfRangeBelow);
FRIEND_TEST_ALL_PREFIXES(InputVolumeControllerTest,
AgcMinMicLevelExperimentEnabled50);
FRIEND_TEST_ALL_PREFIXES(InputVolumeControllerTest,
AgcMinMicLevelExperimentEnabledAboveStartupLevel);
FRIEND_TEST_ALL_PREFIXES(InputVolumeControllerParametrizedTest,
ClippingParametersVerified);
FRIEND_TEST_ALL_PREFIXES(InputVolumeControllerParametrizedTest,
DisableClippingPredictorDoesNotLowerVolume);
FRIEND_TEST_ALL_PREFIXES(InputVolumeControllerParametrizedTest,
UsedClippingPredictionsProduceLowerAnalogLevels);
FRIEND_TEST_ALL_PREFIXES(InputVolumeControllerParametrizedTest,
UnusedClippingPredictionsProduceEqualAnalogLevels);
FRIEND_TEST_ALL_PREFIXES(InputVolumeControllerParametrizedTest,
EmptyRmsErrorOverrideHasNoEffect);
FRIEND_TEST_ALL_PREFIXES(InputVolumeControllerParametrizedTest,
NonEmptyRmsErrorOverrideHasEffect);
// Ctor that creates a single channel AGC and by injecting `agc`.
// `agc` will be owned by this class; hence, do not delete it.
InputVolumeController(
const AudioProcessing::Config::GainController1::AnalogGainController&
analog_config,
Agc* agc);
void AggregateChannelLevels();
const bool analog_controller_enabled_;
const absl::optional<int> min_mic_level_override_;
std::unique_ptr<ApmDataDumper> data_dumper_;
static std::atomic<int> instance_counter_;
const bool use_min_channel_level_;
const int num_capture_channels_;
const bool disable_digital_adaptive_;
int frames_since_clipped_;
// TODO(bugs.webrtc.org/7494): Create a separate member for the applied input
// volume.
// TODO(bugs.webrtc.org/7494): Once
// `AudioProcessingImpl::recommended_stream_analog_level()` becomes a trivial
// getter, leave uninitialized.
// Recommended input volume. After `set_stream_analog_level()` is called it
// holds the observed input volume. Possibly updated by `AnalyzePreProcess()`
// and `Process()`; after these calls, holds the recommended input volume.
int recommended_input_volume_ = 0;
bool capture_output_used_;
int channel_controlling_gain_ = 0;
const int clipped_level_step_;
const float clipped_ratio_threshold_;
const int clipped_wait_frames_;
std::vector<std::unique_ptr<RecommendedInputVolumeEstimator>> channel_agcs_;
std::vector<absl::optional<int>> new_compressions_to_set_;
const std::unique_ptr<ClippingPredictor> clipping_predictor_;
const bool use_clipping_predictor_step_;
float clipping_rate_log_;
int clipping_rate_log_counter_;
};
// TODO(bugs.webrtc.org/7494): Use applied/recommended input volume naming
// convention.
class RecommendedInputVolumeEstimator {
public:
RecommendedInputVolumeEstimator(ApmDataDumper* data_dumper,
int startup_min_level,
int clipped_level_min,
bool disable_digital_adaptive,
int min_mic_level);
~RecommendedInputVolumeEstimator();
RecommendedInputVolumeEstimator(const RecommendedInputVolumeEstimator&) =
delete;
RecommendedInputVolumeEstimator& operator=(
const RecommendedInputVolumeEstimator&) = delete;
void Initialize();
void HandleCaptureOutputUsedChange(bool capture_output_used);
// Sets the current input volume.
void set_stream_analog_level(int level) { recommended_input_volume_ = level; }
// Lowers the recommended input volume in response to clipping based on the
// suggested reduction `clipped_level_step`. Must be called after
// `set_stream_analog_level()`.
void HandleClipping(int clipped_level_step);
// Analyzes `audio`, requests the RMS error from AGC, updates the recommended
// input volume based on the estimated speech level and, if enabled, updates
// the (digital) compression gain to be applied by `agc_`. Must be called
// after `HandleClipping()`. If `rms_error_override` has a value, RMS error
// from AGC is overridden by it.
void Process(rtc::ArrayView<const int16_t> audio,
absl::optional<int> rms_error_override);
// Returns the recommended input volume. Must be called after `Process()`.
int recommended_analog_level() const { return recommended_input_volume_; }
float voice_probability() const { return agc_->voice_probability(); }
void ActivateLogging() { log_to_histograms_ = true; }
absl::optional<int> new_compression() const {
return new_compression_to_set_;
}
// Only used for testing.
void set_agc(Agc* agc) { agc_.reset(agc); }
int min_mic_level() const { return min_mic_level_; }
int startup_min_level() const { return startup_min_level_; }
private:
// Sets a new input volume, after first checking that it hasn't been updated
// by the user, in which case no action is taken.
void SetLevel(int new_level);
// Set the maximum input volume the AGC is allowed to apply. Also updates the
// maximum compression gain to compensate. The volume must be at least
// `kClippedLevelMin`.
void SetMaxLevel(int level);
int CheckVolumeAndReset();
void UpdateGain(int rms_error_db);
void UpdateCompressor();
const int min_mic_level_;
const bool disable_digital_adaptive_;
std::unique_ptr<Agc> agc_;
int level_ = 0;
int max_level_;
int max_compression_gain_;
int target_compression_;
int compression_;
float compression_accumulator_;
bool capture_output_used_ = true;
bool check_volume_on_next_process_ = true;
bool startup_ = true;
int startup_min_level_;
int calls_since_last_gain_log_ = 0;
// TODO(bugs.webrtc.org/7494): Create a separate member for the applied
// input volume.
// Recommended input volume. After `set_stream_analog_level()` is
// called, it holds the observed applied input volume. Possibly updated by
// `HandleClipping()` and `Process()`; after these calls, holds the
// recommended input volume.
int recommended_input_volume_ = 0;
absl::optional<int> new_compression_to_set_;
bool log_to_histograms_ = false;
const int clipped_level_min_;
// Frames since the last `UpdateGain()` call.
int frames_since_update_gain_ = 0;
// Set to true for the first frame after startup and reset, otherwise false.
bool is_first_frame_ = true;
};
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_AGC2_INPUT_VOLUME_CONTROLLER_H_

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