Added calling of the stream_analog_level api in audioproc_f

webrtc/modules/audio_processing/test/aec_dump_based_simulator.cc

 /*
  *  Copyright (c) 2016 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 45 matching lines @@
         }
       }
     }
   }
   return true;
 }
 
 }  // namespace
 
 void AecDumpBasedSimulator::PrepareProcessStreamCall(
-    const webrtc::audioproc::Stream& msg) {
+    const webrtc::audioproc::Stream& msg,
+    bool* set_stream_analog_level_called) {
   if (msg.has_input_data()) {
     // Fixed interface processing.
     // Verify interface invariance.
     RTC_CHECK(interface_used_ == InterfaceType::kFixedInterface ||
               interface_used_ == InterfaceType::kNotSpecified);
     interface_used_ = InterfaceType::kFixedInterface;
 
     // Populate input buffer.
     RTC_CHECK_EQ(sizeof(fwd_frame_.data_[0]) * fwd_frame_.samples_per_channel_ *
                      fwd_frame_.num_channels_,
@@ skipping 43 matching lines @@
     }
   } else {
     ap_->set_stream_key_pressed(*settings_.use_ts);
   }
 
   // TODO(peah): Add support for controlling the analog level via the
   // command-line.
   if (msg.has_level()) {
     RTC_CHECK_EQ(AudioProcessing::kNoError,
                  ap_->gain_control()->set_stream_analog_level(msg.level()));
+    *set_stream_analog_level_called = true;
+  } else {
+    *set_stream_analog_level_called = false;
   }
 }
 
 void AecDumpBasedSimulator::VerifyProcessStreamBitExactness(
     const webrtc::audioproc::Stream& msg) {
   if (bitexact_output_) {
     if (interface_used_ == InterfaceType::kFixedInterface) {
       bitexact_output_ = VerifyFixedBitExactness(msg, fwd_frame_);
     } else {
       bitexact_output_ = VerifyFloatBitExactness(msg, out_config_, *out_buf_);
@@ skipping 360 matching lines @@
   }
 
   SetupBuffersConfigsOutputs(
       msg.sample_rate(), output_sample_rate, msg.reverse_sample_rate(),
       reverse_output_sample_rate, msg.num_input_channels(), num_output_channels,
       msg.num_reverse_channels(), num_reverse_output_channels);
 }
 
 void AecDumpBasedSimulator::HandleMessage(
     const webrtc::audioproc::Stream& msg) {
-  PrepareProcessStreamCall(msg);
+  bool set_stream_analog_level_called = false;
+  PrepareProcessStreamCall(msg, &set_stream_analog_level_called);
   ProcessStream(interface_used_ == InterfaceType::kFixedInterface);
+  if (set_stream_analog_level_called) {
+    // Call stream analog level to ensure that any side-effects are triggered.
+    (void)ap_->gain_control()->stream_analog_level();
+  }
+
   VerifyProcessStreamBitExactness(msg);
 }
 
 void AecDumpBasedSimulator::HandleMessage(
     const webrtc::audioproc::ReverseStream& msg) {
   PrepareReverseProcessStreamCall(msg);
   ProcessReverseStream(interface_used_ == InterfaceType::kFixedInterface);
 }
 
 }  // namespace test
 }  // namespace webrtc