1 files changed, 90 insertions, 0 deletions
diff --git a/webrtc/modules/audio_processing/agc2/adaptive_agc.cc b/webrtc/modules/audio_processing/agc2/adaptive_agc.cc
new file mode 100644
index 0000000..0372ccf
--- /dev/null
+++ b/webrtc/modules/audio_processing/agc2/adaptive_agc.cc
@@ -0,0 +1,90 @@
+/*
+ *  Copyright (c) 2018 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/adaptive_agc.h"
+
+#include "common_audio/include/audio_util.h"
+#include "modules/audio_processing/agc2/vad_with_level.h"
+#include "modules/audio_processing/logging/apm_data_dumper.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+namespace {
+
+void DumpDebugData(const AdaptiveDigitalGainApplier::FrameInfo& info,
+                   ApmDataDumper& dumper) {
+  dumper.DumpRaw("agc2_vad_probability", info.vad_result.speech_probability);
+  dumper.DumpRaw("agc2_vad_rms_dbfs", info.vad_result.rms_dbfs);
+  dumper.DumpRaw("agc2_vad_peak_dbfs", info.vad_result.peak_dbfs);
+  dumper.DumpRaw("agc2_noise_estimate_dbfs", info.input_noise_level_dbfs);
+  dumper.DumpRaw("agc2_last_limiter_audio_level", info.limiter_envelope_dbfs);
+}
+
+constexpr int kGainApplierAdjacentSpeechFramesThreshold = 1;
+constexpr float kMaxGainChangePerSecondDb = 3.f;
+constexpr float kMaxOutputNoiseLevelDbfs = -50.f;
+
+}  // namespace
+
+AdaptiveAgc::AdaptiveAgc(ApmDataDumper* apm_data_dumper)
+    : speech_level_estimator_(apm_data_dumper),
+      gain_applier_(apm_data_dumper,
+                    kGainApplierAdjacentSpeechFramesThreshold,
+                    kMaxGainChangePerSecondDb,
+                    kMaxOutputNoiseLevelDbfs),
+      apm_data_dumper_(apm_data_dumper),
+      noise_level_estimator_(apm_data_dumper) {
+  RTC_DCHECK(apm_data_dumper);
+}
+
+AdaptiveAgc::AdaptiveAgc(ApmDataDumper* apm_data_dumper,
+                         const AudioProcessing::Config::GainController2& config)
+    : speech_level_estimator_(
+          apm_data_dumper,
+          config.adaptive_digital.level_estimator,
+          config.adaptive_digital
+              .level_estimator_adjacent_speech_frames_threshold,
+          config.adaptive_digital.initial_saturation_margin_db,
+          config.adaptive_digital.extra_saturation_margin_db),
+      vad_(config.adaptive_digital.vad_probability_attack),
+      gain_applier_(
+          apm_data_dumper,
+          config.adaptive_digital.gain_applier_adjacent_speech_frames_threshold,
+          config.adaptive_digital.max_gain_change_db_per_second,
+          config.adaptive_digital.max_output_noise_level_dbfs),
+      apm_data_dumper_(apm_data_dumper),
+      noise_level_estimator_(apm_data_dumper) {
+  RTC_DCHECK(apm_data_dumper);
+  if (!config.adaptive_digital.use_saturation_protector) {
+    RTC_LOG(LS_WARNING) << "The saturation protector cannot be disabled.";
+  }
+}
+
+AdaptiveAgc::~AdaptiveAgc() = default;
+
+void AdaptiveAgc::Process(AudioFrameView<float> frame, float limiter_envelope) {
+  AdaptiveDigitalGainApplier::FrameInfo info;
+  info.vad_result = vad_.AnalyzeFrame(frame);
+  speech_level_estimator_.Update(info.vad_result);
+  info.input_level_dbfs = speech_level_estimator_.level_dbfs();
+  info.input_noise_level_dbfs = noise_level_estimator_.Analyze(frame);
+  info.limiter_envelope_dbfs =
+      limiter_envelope > 0 ? FloatS16ToDbfs(limiter_envelope) : -90.f;
+  info.estimate_is_confident = speech_level_estimator_.IsConfident();
+  DumpDebugData(info, *apm_data_dumper_);
+  gain_applier_.Process(info, frame);
+}
+
+void AdaptiveAgc::Reset() {
+  speech_level_estimator_.Reset();
+}
+
+}  // namespace webrtc