audioproc_f with fake microphone.

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 141 matching lines @@
   if (!settings_.use_ts) {
     if (msg.has_keypress()) {
       ap_->set_stream_key_pressed(msg.keypress());
     }
   } else {
     ap_->set_stream_key_pressed(*settings_.use_ts);
   }
 
   // TODO(peah): Add support for controlling the analog level via the
   // command-line.
-  RTC_CHECK(msg.has_level());  // Level is always logged in AEC dumps.
-  // When the analog gain is simulated, use the gain suggested by AGC instead of
-  // that stored in the AEC dump.
-  RTC_CHECK_EQ(AudioProcessing::kNoError,
-               ap_->gain_control()->set_stream_analog_level(
-                   settings_.simulate_mic_gain ?
-                      last_specified_microphone_level_ : msg.level()));
-  // TODO(aleloi): If settings_.simulate_mic_gain, set undo level to
-  // |msg.level()| via FakeRecordingDevice::NotifyAudioDeviceLevel().
+
+  // Level is always logged in AEC dumps.
+  RTC_CHECK(msg.has_level());
+  // When the analog gain is simulated, use the gain suggested by AGC
+  // instead of that stored in the AEC dump.
+  RTC_CHECK_EQ(
+      AudioProcessing::kNoError,
+      ap_->gain_control()->set_stream_analog_level(
+          settings_.simulate_mic_gain ? fake_recording_device_->analog_level()
+                                      : msg.level()));
+  if (settings_.simulate_mic_gain) {
+    fake_recording_device_->NotifyAudioDeviceLevel(msg.level());
+  }
 }
 
 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 379 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);
   ProcessStream(interface_used_ == InterfaceType::kFixedInterface);
+
+  // Call stream analog level even if !simulate_mic_gain to ensure
+  // that any side-effects are triggered.
+  const int level = ap_->gain_control()->stream_analog_level();
   if (settings_.simulate_mic_gain) {
-    // Store analog level for the next analyzed frame.
-    last_specified_microphone_level_ =
-        ap_->gain_control()->stream_analog_level();
-    // TODO(aleloi): Set the returned value into a FakeRecordingDevice instance
-    // via FakeRecordingDevice::set_analog_level() instead of using
-    // last_specified_microphone_level_.
-  } else {
-    // Call stream analog level to ensure that any side-effects are triggered.
-    (void)ap_->gain_control()->stream_analog_level();
+    fake_recording_device_->set_analog_level(level);
   }
   VerifyProcessStreamBitExactness(msg);
 }
 
 void AecDumpBasedSimulator::HandleMessage(
     const webrtc::audioproc::ReverseStream& msg) {
   PrepareReverseProcessStreamCall(msg);
   ProcessReverseStream(interface_used_ == InterfaceType::kFixedInterface);
 }
 
 }  // namespace test
 }  // namespace webrtc