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 578 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();

[the sun, 2016/02/09 11:49:55]

Just for testing? It's a bit confusing that you say "Retrieve analog level" but
never actually do anything with it. EXPECT_EQ(80, apm->...); ?
I first thought you were relying on some side effect to be triggered.

[peah-webrtc, 2016/02/10 09:33:37]

The problem with that is that the purpose is not at all to test the
functionality but rather to hammer the api in order to find deadlocks and for
the thread sanitizers to be able to detect threading problems.. As I have no
idea of what the return value should be, I'd rather just not act on it. 

To test the actual functionality in the AGC I think it is better to test that in
another test that tests the AGC functionality.

Does that make sense?

+
   // 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