| /* |
| * 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/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 kMaxMicLevel = 255; |
| static_assert(kGainMapSize > kMaxMicLevel, "gain map too small"); |
| constexpr int kMinMicLevel = 12; |
| |
| // Prevent very large microphone level changes. |
| constexpr int kMaxResidualGainChange = 15; |
| |
| 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; |
| } |
| |
| // 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 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. The value of `speech_level_dbfs` is |
| // required to be in the range [-90.0f, 30.0f]. Returns a positive value when |
| // the speech level is below the target range and a negative value when the |
| // speech level is above the target range. |
| int GetSpeechLevelErrorDb(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); |
| |
| // Ensure the speech level is in the range [-90.0f, 30.0f]. |
| speech_level_dbfs = rtc::SafeClamp<float>( |
| speech_level_dbfs, kMinSpeechLevelDbfs, kMaxSpeechLevelDbfs); |
| |
| // Compute the speech level distance to the target range |
| // [`target_range_min_dbfs`, `target_range_max_dbfs`]. |
| int rms_error_dbfs = 0; |
| if (speech_level_dbfs > target_range_max_dbfs) { |
| rms_error_dbfs = std::round(target_range_max_dbfs - speech_level_dbfs); |
| } else if (speech_level_dbfs < target_range_min_dbfs) { |
| rms_error_dbfs = std::round(target_range_min_dbfs - speech_level_dbfs); |
| } |
| |
| return rms_error_dbfs; |
| } |
| |
| } // namespace |
| |
| MonoInputVolumeController::MonoInputVolumeController( |
| int clipped_level_min, |
| int min_mic_level, |
| int update_input_volume_wait_frames, |
| float speech_probability_threshold, |
| float speech_ratio_threshold) |
| : min_mic_level_(min_mic_level), |
| max_level_(kMaxMicLevel), |
| clipped_level_min_(clipped_level_min), |
| 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(clipped_level_min_, 0); |
| RTC_DCHECK_LE(clipped_level_min_, 255); |
| RTC_DCHECK_GE(min_mic_level_, 0); |
| RTC_DCHECK_LE(min_mic_level_, 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_level_ = kMaxMicLevel; |
| 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_dbfs, |
| 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_) { |
| if (rms_error_dbfs.has_value()) { |
| UpdateInputVolume(*rms_error_dbfs); |
| } |
| } |
| } |
| |
| is_first_frame_ = false; |
| } |
| |
| void MonoInputVolumeController::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)); |
| frames_since_update_input_volume_ = 0; |
| speech_frames_since_update_input_volume_ = 0; |
| is_first_frame_ = false; |
| } |
| } |
| |
| void MonoInputVolumeController::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. |
| frames_since_update_input_volume_ = 0; |
| speech_frames_since_update_input_volume_ = 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 MonoInputVolumeController::SetMaxLevel(int level) { |
| RTC_DCHECK_GE(level, clipped_level_min_); |
| max_level_ = level; |
| RTC_DLOG(LS_INFO) << "[agc] max_level_=" << max_level_; |
| } |
| |
| 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 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; |
| |
| if (level < min_mic_level_) { |
| level = min_mic_level_; |
| RTC_DLOG(LS_INFO) << "[agc] Initial volume too low, raising to " << level; |
| recommended_input_volume_ = level; |
| } |
| |
| level_ = level; |
| 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_dbfs) { |
| const int residual_gain = rtc::SafeClamp( |
| rms_error_dbfs, -kMaxResidualGainChange, kMaxResidualGainChange); |
| |
| RTC_DLOG(LS_INFO) << "[agc] rms_error_dbfs=" << rms_error_dbfs |
| << ", residual_gain=" << residual_gain; |
| |
| if (residual_gain == 0) { |
| return; |
| } |
| |
| SetLevel(LevelFromGainError(residual_gain, level_, min_mic_level_)); |
| } |
| |
| InputVolumeController::InputVolumeController(int num_capture_channels, |
| const Config& config) |
| : num_capture_channels_(num_capture_channels), |
| min_mic_level_override_(GetMinMicLevelOverride()), |
| 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) { |
| const int min_mic_level = min_mic_level_override_.value_or(kMinMicLevel); |
| RTC_LOG(LS_INFO) << "[agc] Input volume controller enabled"; |
| RTC_LOG(LS_INFO) << "[agc] Min mic level: " << min_mic_level |
| << " (overridden: " |
| << (min_mic_level_override_.has_value() ? "yes" : "no") |
| << ")"; |
| |
| for (auto& controller : channel_controllers_) { |
| controller = std::make_unique<MonoInputVolumeController>( |
| config.clipped_level_min, min_mic_level, |
| 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() { |
| RTC_DLOG(LS_INFO) << "InputVolumeController::Initialize"; |
| for (auto& controller : channel_controllers_) { |
| controller->Initialize(); |
| } |
| capture_output_used_ = true; |
| |
| AggregateChannelLevels(); |
| clipping_rate_log_ = 0.0f; |
| clipping_rate_log_counter_ = 0; |
| } |
| |
| 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. 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. |
| 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]->clipped_level_min(), 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_controllers_) { |
| state_ch->HandleClipping(step); |
| } |
| frames_since_clipped_ = 0; |
| if (!!clipping_predictor_) { |
| clipping_predictor_->Reset(); |
| } |
| } |
| |
| AggregateChannelLevels(); |
| } |
| |
| void InputVolumeController::Process(float speech_probability, |
| absl::optional<float> speech_level_dbfs) { |
| AggregateChannelLevels(); |
| |
| if (!capture_output_used_) { |
| return; |
| } |
| |
| absl::optional<int> rms_error_dbfs; |
| if (speech_level_dbfs.has_value()) { |
| // Compute the error for all frames (both speech and non-speech frames). |
| rms_error_dbfs = GetSpeechLevelErrorDb( |
| *speech_level_dbfs, target_range_min_dbfs_, target_range_max_dbfs_); |
| } |
| |
| for (auto& controller : channel_controllers_) { |
| controller->Process(rms_error_dbfs, speech_probability); |
| } |
| |
| AggregateChannelLevels(); |
| } |
| |
| 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::set_stream_analog_level(int level) { |
| for (auto& controller : channel_controllers_) { |
| controller->set_stream_analog_level(level); |
| } |
| |
| 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 level = channel_controllers_[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_); |
| } |
| |
| recommended_input_volume_ = new_recommended_input_volume; |
| } |
| |
| } // namespace webrtc |