modules/audio_processing/agc/agc_manager_direct.cc - src - Git at Google

 /*
  *  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/agc/agc_manager_direct.h"

 #include <algorithm>
 #include <cmath>

 #include "common_audio/include/audio_util.h"
 #include "modules/audio_processing/agc/gain_control.h"
 #include "modules/audio_processing/agc/gain_map_internal.h"
 #include "rtc_base/atomic_ops.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 the microphone level is lowered with every clipping event.
 const int kClippedLevelStep = 15;
 // Proportion of clipped samples required to declare a clipping event.
 const float kClippedRatioThreshold = 0.1f;
 // Time in frames to wait after a clipping event before checking again.
 const int kClippedWaitFrames = 300;

 // Amount of error we tolerate in the microphone level (presumably due to OS
 // quantization) before we assume the user has manually adjusted the microphone.
 const int kLevelQuantizationSlack = 25;

 const int kDefaultCompressionGain = 7;
 const int kMaxCompressionGain = 12;
 const int kMinCompressionGain = 2;
 // Controls the rate of compression changes towards the target.
 const float kCompressionGainStep = 0.05f;

 const int kMaxMicLevel = 255;
 static_assert(kGainMapSize > kMaxMicLevel, "gain map too small");
 const int kMinMicLevel = 12;

 // Prevent very large microphone level changes.
 const int kMaxResidualGainChange = 15;

 // Maximum additional gain allowed to compensate for microphone level
 // restrictions from clipping events.
 const int kSurplusCompressionGain = 6;

 // Returns whether a fall-back solution to choose the maximum level should be
 // chosen.
 bool UseMaxAnalogChannelLevel() {
   return field_trial::IsEnabled("WebRTC-UseMaxAnalogAgcChannelLevel");
 }

 // Returns kMinMicLevel if no field trial exists or if it has been disabled.
 // Returns a value between 0 and 255 depending on the field-trial string.
 // Example: 'WebRTC-Audio-AgcMinMicLevelExperiment/Enabled-80' => returns 80.
 int GetMinMicLevel() {
   RTC_LOG(LS_INFO) << "[agc] GetMinMicLevel";
   constexpr char kMinMicLevelFieldTrial[] =
       "WebRTC-Audio-AgcMinMicLevelExperiment";
   if (!webrtc::field_trial::IsEnabled(kMinMicLevelFieldTrial)) {
     RTC_LOG(LS_INFO) << "[agc] Using default min mic level: " << kMinMicLevel;
     return kMinMicLevel;
   }
   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) {
     RTC_LOG(LS_INFO) << "[agc] Experimental min mic level: " << min_mic_level;
     return min_mic_level;
   } else {
     RTC_LOG(LS_WARNING) << "[agc] Invalid parameter for "
                         << kMinMicLevelFieldTrial << ", ignored.";
     return kMinMicLevel;
   }
 }

 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.f || audio[ch][i] <= -32768.f) {
         ++num_clipped_in_ch;
       }
     }
     num_clipped = std::max(num_clipped, num_clipped_in_ch);
   }
   return static_cast<float>(num_clipped) / (samples_per_channel);
 }

 }  // namespace

 MonoAgc::MonoAgc(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) {}

 MonoAgc::~MonoAgc() = default;

 void MonoAgc::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;
 }

 void MonoAgc::Process(const int16_t* audio,
                       size_t samples_per_channel,
                       int sample_rate_hz) {
   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, samples_per_channel, sample_rate_hz);

   UpdateGain();
   if (!disable_digital_adaptive_) {
     UpdateCompressor();
   }
 }

 void MonoAgc::HandleClipping() {
   // Always decrease the maximum level, even if the current level is below
   // threshold.
   SetMaxLevel(std::max(clipped_level_min_, max_level_ - kClippedLevelStep));
   if (log_to_histograms_) {
     RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.AgcClippingAdjustmentAllowed",
                           level_ - kClippedLevelStep >= 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_ - kClippedLevelStep));
     // Reset the AGCs for all channels since the level has changed.
     agc_->Reset();
   }
 }

 void MonoAgc::SetLevel(int new_level) {
   int voe_level = stream_analog_level_;
   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;
   }

   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();

     return;
   }

   new_level = std::min(new_level, max_level_);
   if (new_level == level_) {
     return;
   }

   stream_analog_level_ = new_level;
   RTC_DLOG(LS_INFO) << "[agc] voe_level=" << voe_level << ", level_=" << level_
                     << ", new_level=" << new_level;
   level_ = new_level;
 }

 void MonoAgc::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 MonoAgc::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 MonoAgc::CheckVolumeAndReset() {
   int level = stream_analog_level_;
   // 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;
     stream_analog_level_ = level;
   }
   agc_->Reset();
   level_ = level;
   startup_ = false;
   return 0;
 }

 // Requests the RMS error from AGC and 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 MonoAgc::UpdateGain() {
   int rms_error = 0;
   if (!agc_->GetRmsErrorDb(&rms_error)) {
     // No error update ready.
     return;
   }
   // 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 MonoAgc::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_;
   }
 }

 int AgcManagerDirect::instance_counter_ = 0;

 AgcManagerDirect::AgcManagerDirect(Agc* agc,
                                    int startup_min_level,
                                    int clipped_level_min,
                                    int sample_rate_hz)
     : AgcManagerDirect(/*num_capture_channels*/ 1,
                        startup_min_level,
                        clipped_level_min,
                        /*disable_digital_adaptive*/ false,
                        sample_rate_hz) {
   RTC_DCHECK(channel_agcs_[0]);
   RTC_DCHECK(agc);
   channel_agcs_[0]->set_agc(agc);
 }

 AgcManagerDirect::AgcManagerDirect(int num_capture_channels,
                                    int startup_min_level,
                                    int clipped_level_min,
                                    bool disable_digital_adaptive,
                                    int sample_rate_hz)
     : data_dumper_(
           new ApmDataDumper(rtc::AtomicOps::Increment(&instance_counter_))),
       use_min_channel_level_(!UseMaxAnalogChannelLevel()),
       sample_rate_hz_(sample_rate_hz),
       num_capture_channels_(num_capture_channels),
       disable_digital_adaptive_(disable_digital_adaptive),
       frames_since_clipped_(kClippedWaitFrames),
       capture_output_used_(true),
       channel_agcs_(num_capture_channels),
       new_compressions_to_set_(num_capture_channels) {
   const int min_mic_level = GetMinMicLevel();
   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<MonoAgc>(
         data_dumper_ch, startup_min_level, clipped_level_min,
         disable_digital_adaptive_, min_mic_level);
   }
   RTC_DCHECK_LT(0, channel_agcs_.size());
   channel_agcs_[0]->ActivateLogging();
 }

 AgcManagerDirect::~AgcManagerDirect() {}

 void AgcManagerDirect::Initialize() {
   RTC_DLOG(LS_INFO) << "AgcManagerDirect::Initialize";
   data_dumper_->InitiateNewSetOfRecordings();
   for (size_t ch = 0; ch < channel_agcs_.size(); ++ch) {
     channel_agcs_[ch]->Initialize();
   }
   capture_output_used_ = true;

   AggregateChannelLevels();
 }

 void AgcManagerDirect::SetupDigitalGainControl(
     GainControl* gain_control) const {
   RTC_DCHECK(gain_control);
   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 AgcManagerDirect::AnalyzePreProcess(const AudioBuffer* audio) {
   RTC_DCHECK(audio);
   AnalyzePreProcess(audio->channels_const(), audio->num_frames());
 }

 void AgcManagerDirect::AnalyzePreProcess(const float* const* audio,
                                          size_t samples_per_channel) {
   RTC_DCHECK(audio);
   AggregateChannelLevels();
   if (!capture_output_used_) {
     return;
   }

   if (frames_since_clipped_ < kClippedWaitFrames) {
     ++frames_since_clipped_;
     return;
   }

   // 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);

   if (clipped_ratio > kClippedRatioThreshold) {
     RTC_DLOG(LS_INFO) << "[agc] Clipping detected. clipped_ratio="
                       << clipped_ratio;
     for (auto& state_ch : channel_agcs_) {
       state_ch->HandleClipping();
     }
     frames_since_clipped_ = 0;
   }
   AggregateChannelLevels();
 }

 void AgcManagerDirect::Process(const AudioBuffer* audio) {
   AggregateChannelLevels();

   if (!capture_output_used_) {
     return;
   }

   for (size_t ch = 0; ch < channel_agcs_.size(); ++ch) {
     int16_t* audio_use = nullptr;
     std::array<int16_t, AudioBuffer::kMaxSampleRate / 100> audio_data;
     int num_frames_per_band;
     if (audio) {
       FloatS16ToS16(audio->split_bands_const_f(ch)[0],
                     audio->num_frames_per_band(), audio_data.data());
       audio_use = audio_data.data();
       num_frames_per_band = audio->num_frames_per_band();
     } else {
       // Only used for testing.
       // TODO(peah): Change unittests to only allow on non-null audio input.
       num_frames_per_band = 320;
     }
     channel_agcs_[ch]->Process(audio_use, num_frames_per_band, sample_rate_hz_);
     new_compressions_to_set_[ch] = channel_agcs_[ch]->new_compression();
   }

   AggregateChannelLevels();
 }

 absl::optional<int> AgcManagerDirect::GetDigitalComressionGain() {
   return new_compressions_to_set_[channel_controlling_gain_];
 }

 void AgcManagerDirect::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 AgcManagerDirect::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 AgcManagerDirect::set_stream_analog_level(int level) {
   for (size_t ch = 0; ch < channel_agcs_.size(); ++ch) {
     channel_agcs_[ch]->set_stream_analog_level(level);
   }

   AggregateChannelLevels();
 }

 void AgcManagerDirect::AggregateChannelLevels() {
   stream_analog_level_ = channel_agcs_[0]->stream_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]->stream_analog_level();
       if (level < stream_analog_level_) {
         stream_analog_level_ = level;
         channel_controlling_gain_ = static_cast<int>(ch);
       }
     }
   } else {
     for (size_t ch = 1; ch < channel_agcs_.size(); ++ch) {
       int level = channel_agcs_[ch]->stream_analog_level();
       if (level > stream_analog_level_) {
         stream_analog_level_ = level;
         channel_controlling_gain_ = static_cast<int>(ch);
       }
     }
   }
 }

 }  // namespace webrtc
	/*
	* 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/agc/agc_manager_direct.h"

	#include <algorithm>
	#include <cmath>

	#include "common_audio/include/audio_util.h"
	#include "modules/audio_processing/agc/gain_control.h"
	#include "modules/audio_processing/agc/gain_map_internal.h"
	#include "rtc_base/atomic_ops.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 the microphone level is lowered with every clipping event.
	const int kClippedLevelStep = 15;
	// Proportion of clipped samples required to declare a clipping event.
	const float kClippedRatioThreshold = 0.1f;
	// Time in frames to wait after a clipping event before checking again.
	const int kClippedWaitFrames = 300;

	// Amount of error we tolerate in the microphone level (presumably due to OS
	// quantization) before we assume the user has manually adjusted the microphone.
	const int kLevelQuantizationSlack = 25;

	const int kDefaultCompressionGain = 7;
	const int kMaxCompressionGain = 12;
	const int kMinCompressionGain = 2;
	// Controls the rate of compression changes towards the target.
	const float kCompressionGainStep = 0.05f;

	const int kMaxMicLevel = 255;
	static_assert(kGainMapSize > kMaxMicLevel, "gain map too small");
	const int kMinMicLevel = 12;

	// Prevent very large microphone level changes.
	const int kMaxResidualGainChange = 15;

	// Maximum additional gain allowed to compensate for microphone level
	// restrictions from clipping events.
	const int kSurplusCompressionGain = 6;

	// Returns whether a fall-back solution to choose the maximum level should be
	// chosen.
	bool UseMaxAnalogChannelLevel() {
	return field_trial::IsEnabled("WebRTC-UseMaxAnalogAgcChannelLevel");
	}

	// Returns kMinMicLevel if no field trial exists or if it has been disabled.
	// Returns a value between 0 and 255 depending on the field-trial string.
	// Example: 'WebRTC-Audio-AgcMinMicLevelExperiment/Enabled-80' => returns 80.
	int GetMinMicLevel() {
	RTC_LOG(LS_INFO) << "[agc] GetMinMicLevel";
	constexpr char kMinMicLevelFieldTrial[] =
	"WebRTC-Audio-AgcMinMicLevelExperiment";
	if (!webrtc::field_trial::IsEnabled(kMinMicLevelFieldTrial)) {
	RTC_LOG(LS_INFO) << "[agc] Using default min mic level: " << kMinMicLevel;
	return kMinMicLevel;
	}
	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) {
	RTC_LOG(LS_INFO) << "[agc] Experimental min mic level: " << min_mic_level;
	return min_mic_level;
	} else {
	RTC_LOG(LS_WARNING) << "[agc] Invalid parameter for "
	<< kMinMicLevelFieldTrial << ", ignored.";
	return kMinMicLevel;
	}
	}

	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.f \|\| audio[ch][i] <= -32768.f) {
	++num_clipped_in_ch;
	}
	}
	num_clipped = std::max(num_clipped, num_clipped_in_ch);
	}
	return static_cast<float>(num_clipped) / (samples_per_channel);
	}

	} // namespace

	MonoAgc::MonoAgc(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) {}

	MonoAgc::~MonoAgc() = default;

	void MonoAgc::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;
	}

	void MonoAgc::Process(const int16_t* audio,
	size_t samples_per_channel,
	int sample_rate_hz) {
	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, samples_per_channel, sample_rate_hz);

	UpdateGain();
	if (!disable_digital_adaptive_) {
	UpdateCompressor();
	}
	}

	void MonoAgc::HandleClipping() {
	// Always decrease the maximum level, even if the current level is below
	// threshold.
	SetMaxLevel(std::max(clipped_level_min_, max_level_ - kClippedLevelStep));
	if (log_to_histograms_) {
	RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.AgcClippingAdjustmentAllowed",
	level_ - kClippedLevelStep >= 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_ - kClippedLevelStep));
	// Reset the AGCs for all channels since the level has changed.
	agc_->Reset();
	}
	}

	void MonoAgc::SetLevel(int new_level) {
	int voe_level = stream_analog_level_;
	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;
	}

	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();

	return;
	}

	new_level = std::min(new_level, max_level_);
	if (new_level == level_) {
	return;
	}

	stream_analog_level_ = new_level;
	RTC_DLOG(LS_INFO) << "[agc] voe_level=" << voe_level << ", level_=" << level_
	<< ", new_level=" << new_level;
	level_ = new_level;
	}

	void MonoAgc::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 MonoAgc::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 MonoAgc::CheckVolumeAndReset() {
	int level = stream_analog_level_;
	// 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;
	stream_analog_level_ = level;
	}
	agc_->Reset();
	level_ = level;
	startup_ = false;
	return 0;
	}

	// Requests the RMS error from AGC and 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 MonoAgc::UpdateGain() {
	int rms_error = 0;
	if (!agc_->GetRmsErrorDb(&rms_error)) {
	// No error update ready.
	return;
	}
	// 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 MonoAgc::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_;
	}
	}

	int AgcManagerDirect::instance_counter_ = 0;

	AgcManagerDirect::AgcManagerDirect(Agc* agc,
	int startup_min_level,
	int clipped_level_min,
	int sample_rate_hz)
	: AgcManagerDirect(/num_capture_channels/ 1,
	startup_min_level,
	clipped_level_min,
	/disable_digital_adaptive/ false,
	sample_rate_hz) {
	RTC_DCHECK(channel_agcs_[0]);
	RTC_DCHECK(agc);
	channel_agcs_[0]->set_agc(agc);
	}

	AgcManagerDirect::AgcManagerDirect(int num_capture_channels,
	int startup_min_level,
	int clipped_level_min,
	bool disable_digital_adaptive,
	int sample_rate_hz)
	: data_dumper_(
	new ApmDataDumper(rtc::AtomicOps::Increment(&instance_counter_))),
	use_min_channel_level_(!UseMaxAnalogChannelLevel()),
	sample_rate_hz_(sample_rate_hz),
	num_capture_channels_(num_capture_channels),
	disable_digital_adaptive_(disable_digital_adaptive),
	frames_since_clipped_(kClippedWaitFrames),
	capture_output_used_(true),
	channel_agcs_(num_capture_channels),
	new_compressions_to_set_(num_capture_channels) {
	const int min_mic_level = GetMinMicLevel();
	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<MonoAgc>(
	data_dumper_ch, startup_min_level, clipped_level_min,
	disable_digital_adaptive_, min_mic_level);
	}
	RTC_DCHECK_LT(0, channel_agcs_.size());
	channel_agcs_[0]->ActivateLogging();
	}

	AgcManagerDirect::~AgcManagerDirect() {}

	void AgcManagerDirect::Initialize() {
	RTC_DLOG(LS_INFO) << "AgcManagerDirect::Initialize";
	data_dumper_->InitiateNewSetOfRecordings();
	for (size_t ch = 0; ch < channel_agcs_.size(); ++ch) {
	channel_agcs_[ch]->Initialize();
	}
	capture_output_used_ = true;

	AggregateChannelLevels();
	}

	void AgcManagerDirect::SetupDigitalGainControl(
	GainControl* gain_control) const {
	RTC_DCHECK(gain_control);
	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 AgcManagerDirect::AnalyzePreProcess(const AudioBuffer* audio) {
	RTC_DCHECK(audio);
	AnalyzePreProcess(audio->channels_const(), audio->num_frames());
	}

	void AgcManagerDirect::AnalyzePreProcess(const float* const* audio,
	size_t samples_per_channel) {
	RTC_DCHECK(audio);
	AggregateChannelLevels();
	if (!capture_output_used_) {
	return;
	}

	if (frames_since_clipped_ < kClippedWaitFrames) {
	++frames_since_clipped_;
	return;
	}

	// 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);

	if (clipped_ratio > kClippedRatioThreshold) {
	RTC_DLOG(LS_INFO) << "[agc] Clipping detected. clipped_ratio="
	<< clipped_ratio;
	for (auto& state_ch : channel_agcs_) {
	state_ch->HandleClipping();
	}
	frames_since_clipped_ = 0;
	}
	AggregateChannelLevels();
	}

	void AgcManagerDirect::Process(const AudioBuffer* audio) {
	AggregateChannelLevels();

	if (!capture_output_used_) {
	return;
	}

	for (size_t ch = 0; ch < channel_agcs_.size(); ++ch) {
	int16_t* audio_use = nullptr;
	std::array<int16_t, AudioBuffer::kMaxSampleRate / 100> audio_data;
	int num_frames_per_band;
	if (audio) {
	FloatS16ToS16(audio->split_bands_const_f(ch)[0],
	audio->num_frames_per_band(), audio_data.data());
	audio_use = audio_data.data();
	num_frames_per_band = audio->num_frames_per_band();
	} else {
	// Only used for testing.
	// TODO(peah): Change unittests to only allow on non-null audio input.
	num_frames_per_band = 320;
	}
	channel_agcs_[ch]->Process(audio_use, num_frames_per_band, sample_rate_hz_);
	new_compressions_to_set_[ch] = channel_agcs_[ch]->new_compression();
	}

	AggregateChannelLevels();
	}

	absl::optional<int> AgcManagerDirect::GetDigitalComressionGain() {
	return new_compressions_to_set_[channel_controlling_gain_];
	}

	void AgcManagerDirect::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 AgcManagerDirect::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 AgcManagerDirect::set_stream_analog_level(int level) {
	for (size_t ch = 0; ch < channel_agcs_.size(); ++ch) {
	channel_agcs_[ch]->set_stream_analog_level(level);
	}

	AggregateChannelLevels();
	}

	void AgcManagerDirect::AggregateChannelLevels() {
	stream_analog_level_ = channel_agcs_[0]->stream_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]->stream_analog_level();
	if (level < stream_analog_level_) {
	stream_analog_level_ = level;
	channel_controlling_gain_ = static_cast<int>(ch);
	}
	}
	} else {
	for (size_t ch = 1; ch < channel_agcs_.size(); ++ch) {
	int level = channel_agcs_[ch]->stream_analog_level();
	if (level > stream_analog_level_) {
	stream_analog_level_ = level;
	channel_controlling_gain_ = static_cast<int>(ch);
	}
	}
	}
	}

	} // namespace webrtc