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|    1 /* |    1 /* | 
|    2  *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. |    2  *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. | 
|    3  * |    3  * | 
|    4  *  Use of this source code is governed by a BSD-style license |    4  *  Use of this source code is governed by a BSD-style license | 
|    5  *  that can be found in the LICENSE file in the root of the source |    5  *  that can be found in the LICENSE file in the root of the source | 
|    6  *  tree. An additional intellectual property rights grant can be found |    6  *  tree. An additional intellectual property rights grant can be found | 
|    7  *  in the file PATENTS.  All contributing project authors may |    7  *  in the file PATENTS.  All contributing project authors may | 
|    8  *  be found in the AUTHORS file in the root of the source tree. |    8  *  be found in the AUTHORS file in the root of the source tree. | 
|    9  */ |    9  */ | 
|   10  |   10  | 
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|  201   saturating_gain_estimator_.Initialize(); |  201   saturating_gain_estimator_.Initialize(); | 
|  202   metrics_.Initialize(sample_rate_hz); |  202   metrics_.Initialize(sample_rate_hz); | 
|  203  |  203  | 
|  204   last_gain_ = 1.0f; |  204   last_gain_ = 1.0f; | 
|  205   sample_rate_hz_ = rtc::Optional<int>(sample_rate_hz); |  205   sample_rate_hz_ = rtc::Optional<int>(sample_rate_hz); | 
|  206   dc_forgetting_factor_ = 0.01f * sample_rate_hz / 48000.f; |  206   dc_forgetting_factor_ = 0.01f * sample_rate_hz / 48000.f; | 
|  207   std::fill(dc_level_, dc_level_ + arraysize(dc_level_), 0.f); |  207   std::fill(dc_level_, dc_level_ + arraysize(dc_level_), 0.f); | 
|  208 } |  208 } | 
|  209  |  209  | 
|  210 void LevelController::Process(AudioBuffer* audio) { |  210 void LevelController::Process(AudioBuffer* audio) { | 
|  211   RTC_DCHECK_LT(0u, audio->num_channels()); |  211   RTC_DCHECK_LT(0, audio->num_channels()); | 
|  212   RTC_DCHECK_GE(2u, audio->num_channels()); |  212   RTC_DCHECK_GE(2, audio->num_channels()); | 
|  213   RTC_DCHECK_NE(0.f, dc_forgetting_factor_); |  213   RTC_DCHECK_NE(0.f, dc_forgetting_factor_); | 
|  214   RTC_DCHECK(sample_rate_hz_); |  214   RTC_DCHECK(sample_rate_hz_); | 
|  215   data_dumper_->DumpWav("lc_input", audio->num_frames(), |  215   data_dumper_->DumpWav("lc_input", audio->num_frames(), | 
|  216                         audio->channels_const_f()[0], *sample_rate_hz_, 1); |  216                         audio->channels_const_f()[0], *sample_rate_hz_, 1); | 
|  217  |  217  | 
|  218   // Remove DC level. |  218   // Remove DC level. | 
|  219   for (size_t k = 0; k < audio->num_channels(); ++k) { |  219   for (size_t k = 0; k < audio->num_channels(); ++k) { | 
|  220     UpdateAndRemoveDcLevel( |  220     UpdateAndRemoveDcLevel( | 
|  221         dc_forgetting_factor_, &dc_level_[k], |  221         dc_forgetting_factor_, &dc_level_[k], | 
|  222         rtc::ArrayView<float>(audio->channels_f()[k], audio->num_frames())); |  222         rtc::ArrayView<float>(audio->channels_f()[k], audio->num_frames())); | 
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|  284  |  284  | 
|  285 bool LevelController::Validate( |  285 bool LevelController::Validate( | 
|  286     const AudioProcessing::Config::LevelController& config) { |  286     const AudioProcessing::Config::LevelController& config) { | 
|  287   return (config.initial_peak_level_dbfs < |  287   return (config.initial_peak_level_dbfs < | 
|  288               std::numeric_limits<float>::epsilon() && |  288               std::numeric_limits<float>::epsilon() && | 
|  289           config.initial_peak_level_dbfs > |  289           config.initial_peak_level_dbfs > | 
|  290               -(100.f + std::numeric_limits<float>::epsilon())); |  290               -(100.f + std::numeric_limits<float>::epsilon())); | 
|  291 } |  291 } | 
|  292  |  292  | 
|  293 }  // namespace webrtc |  293 }  // namespace webrtc | 
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