Added offline data logpoints and logging functionality to the gain controller

webrtc/modules/audio_processing/audio_processing_impl.cc

 /*
  *  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.
  */
 
@@ skipping 417 matching lines @@
     render_.render_audio.reset(nullptr);
     render_.render_converter.reset(nullptr);
   }
   capture_.capture_audio.reset(
       new AudioBuffer(formats_.api_format.input_stream().num_frames(),
                       formats_.api_format.input_stream().num_channels(),
                       capture_nonlocked_.capture_processing_format.num_frames(),
                       capture_audiobuffer_num_channels,
                       formats_.api_format.output_stream().num_frames()));
 
-  public_submodules_->gain_control->Initialize(num_proc_channels(),
-                                               proc_sample_rate_hz());
-
   public_submodules_->echo_cancellation->Initialize(
       proc_sample_rate_hz(), num_reverse_channels(), num_output_channels(),
       num_proc_channels());
   AllocateRenderQueue();
 
   public_submodules_->echo_control_mobile->Initialize(
       proc_split_sample_rate_hz(), num_reverse_channels(),
       num_output_channels());
+

[peah-webrtc, 2016/10/28 08:37:49]

I moved the initialization of the gain_control to be closer to the
initialization of other gain control functionality to make the code easier to
read. This has no functional impact though.

+  public_submodules_->gain_control->Initialize(num_proc_channels(),
+                                               proc_sample_rate_hz());
   if (constants_.use_experimental_agc) {
     if (!private_submodules_->agc_manager.get()) {
       private_submodules_->agc_manager.reset(new AgcManagerDirect(
           public_submodules_->gain_control.get(),
           public_submodules_->gain_control_for_experimental_agc.get(),
           constants_.agc_startup_min_volume));
     }
     private_submodules_->agc_manager->Initialize();
     private_submodules_->agc_manager->SetCaptureMuted(
         capture_.output_will_be_muted);
+    public_submodules_->gain_control_for_experimental_agc->Initialize();

[peah-webrtc, 2016/10/28 08:37:50]

This is needed to ensure that a new set of data dumps are created when the APM
is re-initialized. 

   }
   InitializeTransient();
   InitializeBeamformer();
 #if WEBRTC_INTELLIGIBILITY_ENHANCER
   InitializeIntelligibility();
 #endif
   public_submodules_->high_pass_filter->Initialize(num_proc_channels(),
                                                    proc_sample_rate_hz());
   public_submodules_->noise_suppression->Initialize(num_proc_channels(),
                                                     proc_sample_rate_hz());
@@ skipping 1264 matching lines @@
       capture_processing_format(kSampleRate16kHz),
       split_rate(kSampleRate16kHz) {}
 
 AudioProcessingImpl::ApmCaptureState::~ApmCaptureState() = default;
 
 AudioProcessingImpl::ApmRenderState::ApmRenderState() = default;
 
 AudioProcessingImpl::ApmRenderState::~ApmRenderState() = default;
 
 }  // namespace webrtc