Moved the GainControlForNewAGC class to a separate file.

webrtc/modules/audio_processing/audio_processing_impl_locking_unittest.cc

 /*
  *  Copyright (c) 2015 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 582 matching lines @@
   EXPECT_EQ(apm_->echo_cancellation()->is_enabled(),
             ((test_config_->aec_type != AecType::AecTurnedOff) &&
              (test_config_->aec_type !=
               AecType::BasicWebRtcAecSettingsWithAecMobile)));
   apm_->echo_cancellation()->stream_drift_samples();
   EXPECT_EQ(apm_->echo_control_mobile()->is_enabled(),
             (test_config_->aec_type != AecType::AecTurnedOff) &&
                 (test_config_->aec_type ==
                  AecType::BasicWebRtcAecSettingsWithAecMobile));
   EXPECT_TRUE(apm_->gain_control()->is_enabled());
-  apm_->gain_control()->stream_analog_level();
   EXPECT_TRUE(apm_->noise_suppression()->is_enabled());
 
   // The below return values are not testable.
   apm_->noise_suppression()->speech_probability();
   apm_->voice_detection()->is_enabled();
 
   return true;
 }
 
 const float CaptureProcessor::kCaptureInputFloatLevel = 0.03125f;
@@ skipping 92 matching lines @@
       (frame_data_.output_number_of_channels == 1
            ? AudioProcessing::ChannelLayout::kMono
            : AudioProcessing::ChannelLayout::kStereo);
 }
 
 // Applies the capture side processing API call.
 void CaptureProcessor::CallApmCaptureSide() {
   // Prepare a proper capture side processing API call input.
   PrepareFrame();
 
-  // Set the stream delay
+  // Set the stream delay.
   apm_->set_stream_delay_ms(30);
 
+  // Set the analog level.
+  apm_->gain_control()->set_stream_analog_level(80);
+
   // Call the specified capture side API processing method.
   int result = AudioProcessing::kNoError;
   switch (test_config_->capture_api_function) {
     case CaptureApiImpl::ProcessStreamImpl1:
       result = apm_->ProcessStream(&frame_data_.frame);
       break;
     case CaptureApiImpl::ProcessStreamImpl2:
       result = apm_->ProcessStream(
           &frame_data_.input_frame[0], frame_data_.input_samples_per_channel,
           frame_data_.input_sample_rate_hz, frame_data_.input_channel_layout,
           frame_data_.output_sample_rate_hz, frame_data_.output_channel_layout,
           &frame_data_.output_frame[0]);
       break;
     case CaptureApiImpl::ProcessStreamImpl3:
       result = apm_->ProcessStream(
           &frame_data_.input_frame[0], frame_data_.input_stream_config,
           frame_data_.output_stream_config, &frame_data_.output_frame[0]);
       break;
     default:
       FAIL();
   }
 
+  // Retrieve the new analog level.
+  apm_->gain_control()->stream_analog_level();
+
   // Check the return code for error.
   ASSERT_EQ(AudioProcessing::kNoError, result);
 }
 
 // Applies any runtime capture APM API calls and audio stream characteristics
 // specified by the scheme for the test.
 void CaptureProcessor::ApplyRuntimeSettingScheme() {
   const int capture_count_local = frame_counters_->GetCaptureCounter();
 
   // Update the number of channels and sample rates for the input and output.
@@ skipping 377 matching lines @@
     DISABLED_AudioProcessingImplLockExtensive,
     AudioProcessingImplLockTest,
     ::testing::ValuesIn(TestConfig::GenerateExtensiveTestConfigs()));
 
 INSTANTIATE_TEST_CASE_P(
     AudioProcessingImplLockBrief,
     AudioProcessingImplLockTest,
     ::testing::ValuesIn(TestConfig::GenerateBriefTestConfigs()));
 
 }  // namespace webrtc