Added offline data logpoints and logging functionality to the gain controller

webrtc/modules/audio_processing/gain_control_impl.h

 /*
  *  Copyright (c) 2012 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 WEBRTC_MODULES_AUDIO_PROCESSING_GAIN_CONTROL_IMPL_H_
 #define WEBRTC_MODULES_AUDIO_PROCESSING_GAIN_CONTROL_IMPL_H_
 
 #include <memory>
 #include <vector>
 
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/swap_queue.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/modules/audio_processing/render_queue_item_verifier.h"
 
 namespace webrtc {
 
+class ApmDataDumper;
 class AudioBuffer;
 
 class GainControlImpl : public GainControl {
  public:
   GainControlImpl(rtc::CriticalSection* crit_render,
                   rtc::CriticalSection* crit_capture);
   ~GainControlImpl() override;
 
   void ProcessRenderAudio(rtc::ArrayView<const int16_t> packed_render_audio);
   int AnalyzeCaptureAudio(AudioBuffer* audio);
@@ skipping 26 matching lines @@
   int set_analog_level_limits(int minimum, int maximum) override;
   int analog_level_minimum() const override;
   int analog_level_maximum() const override;
   bool stream_is_saturated() const override;
 
   int Configure();
 
   rtc::CriticalSection* const crit_render_ ACQUIRED_BEFORE(crit_capture_);
   rtc::CriticalSection* const crit_capture_;
 
+  std::unique_ptr<ApmDataDumper> data_dumper_;
+
   bool enabled_ = false;
 
   Mode mode_ GUARDED_BY(crit_capture_);
   int minimum_capture_level_ GUARDED_BY(crit_capture_);
   int maximum_capture_level_ GUARDED_BY(crit_capture_);
   bool limiter_enabled_ GUARDED_BY(crit_capture_);
   int target_level_dbfs_ GUARDED_BY(crit_capture_);
   int compression_gain_db_ GUARDED_BY(crit_capture_);
   int analog_capture_level_ GUARDED_BY(crit_capture_);
   bool was_analog_level_set_ GUARDED_BY(crit_capture_);
   bool stream_is_saturated_ GUARDED_BY(crit_capture_);
 
   std::vector<std::unique_ptr<GainController>> gain_controllers_;
 
   rtc::Optional<size_t> num_proc_channels_ GUARDED_BY(crit_capture_);
   rtc::Optional<int> sample_rate_hz_ GUARDED_BY(crit_capture_);
 
+  static int instance_counter_;
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(GainControlImpl);
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_GAIN_CONTROL_IMPL_H_