Bump to WebRTC M120 release

Some API deprecation -- ExperimentalAgc and ExperimentalNs are gone.
We're continuing to carry iSAC even though it's gone upstream, but maybe
we'll want to drop that soon.

webrtc/modules/audio_processing/agc2/adaptive_digital_gain_controller.h

@@ -0,0 +1,66 @@
+/*
+ *  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.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_DIGITAL_GAIN_CONTROLLER_H_
+#define MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_DIGITAL_GAIN_CONTROLLER_H_
+
+#include <vector>
+
+#include "modules/audio_processing/agc2/gain_applier.h"
+#include "modules/audio_processing/include/audio_frame_view.h"
+#include "modules/audio_processing/include/audio_processing.h"
+
+namespace webrtc {
+
+class ApmDataDumper;
+
+// Selects the target digital gain, decides when and how quickly to adapt to the
+// target and applies the current gain to 10 ms frames.
+class AdaptiveDigitalGainController {
+ public:
+  // Information about a frame to process.
+  struct FrameInfo {
+    float speech_probability;    // Probability of speech in the [0, 1] range.
+    float speech_level_dbfs;     // Estimated speech level (dBFS).
+    bool speech_level_reliable;  // True with reliable speech level estimation.
+    float noise_rms_dbfs;        // Estimated noise RMS level (dBFS).
+    float headroom_db;           // Headroom (dB).
+    // TODO(bugs.webrtc.org/7494): Remove `limiter_envelope_dbfs`.
+    float limiter_envelope_dbfs;  // Envelope level from the limiter (dBFS).
+  };
+
+  AdaptiveDigitalGainController(
+      ApmDataDumper* apm_data_dumper,
+      const AudioProcessing::Config::GainController2::AdaptiveDigital& config,
+      int adjacent_speech_frames_threshold);
+  AdaptiveDigitalGainController(const AdaptiveDigitalGainController&) = delete;
+  AdaptiveDigitalGainController& operator=(
+      const AdaptiveDigitalGainController&) = delete;
+
+  // Analyzes `info`, updates the digital gain and applies it to a 10 ms
+  // `frame`. Supports any sample rate supported by APM.
+  void Process(const FrameInfo& info, AudioFrameView<float> frame);
+
+ private:
+  ApmDataDumper* const apm_data_dumper_;
+  GainApplier gain_applier_;
+
+  const AudioProcessing::Config::GainController2::AdaptiveDigital config_;
+  const int adjacent_speech_frames_threshold_;
+  const float max_gain_change_db_per_10ms_;
+
+  int calls_since_last_gain_log_;
+  int frames_to_gain_increase_allowed_;
+  float last_gain_db_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_DIGITAL_GAIN_CONTROLLER_H_