audioproc_f with simulated mic analog gain

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.
  */
 
+#include <algorithm>

[peah-webrtc, 2017/06/29 05:45:26]

Why was algorithm and utility added to the includes? Are those required by the
former code?

[AleBzk, 2017/06/29 11:43:35]

Good catch! Thanks!

 #include <iostream>
+#include <utility>
 
 #include "webrtc/modules/audio_processing/test/aec_dump_based_simulator.h"
 
 #include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/modules/audio_processing/test/fake_recording_device.h"

[peah-webrtc, 2017/06/29 05:45:26]

This include is not needed, right?

[AleBzk, 2017/06/29 11:43:35]

Yet another good catch :)

 #include "webrtc/modules/audio_processing/test/protobuf_utils.h"
 #include "webrtc/test/testsupport/trace_to_stderr.h"
 
 namespace webrtc {
 namespace test {
 namespace {
 
 // Verify output bitexactness for the fixed interface.
 // TODO(peah): Check whether it would make sense to add a threshold
 // to use for checking the bitexactness in a soft manner.
@@ skipping 31 matching lines @@
         }
       }
     }
   }
   return true;
 }
 
 }  // namespace
 
 AecDumpBasedSimulator::AecDumpBasedSimulator(const SimulationSettings& settings)
-    : AudioProcessingSimulator(settings) {}
+    : AudioProcessingSimulator(settings) {
+  if (settings_.simulate_mic_gain) {
+    LOG(LS_VERBOSE) << "Simulating analog mic gain using AEC dump as input "
+        << "(the unmodified mic gain level will be virtually restored)";
+  }
+}
 
 AecDumpBasedSimulator::~AecDumpBasedSimulator() = default;
 
 void AecDumpBasedSimulator::PrepareProcessStreamCall(
-    const webrtc::audioproc::Stream& msg,
-    bool* set_stream_analog_level_called) {
+    const webrtc::audioproc::Stream& msg) {
   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()) * fwd_frame_.samples_per_channel_ *
                      fwd_frame_.num_channels_,
@@ skipping 68 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.
-  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;
-  }
+  // Level is always logged in AEC dumps.
+  RTC_CHECK(msg.has_level());
+  aec_dump_mic_level_ = rtc::Optional<int>(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 377 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) {
-  bool set_stream_analog_level_called = false;
-  PrepareProcessStreamCall(msg, &set_stream_analog_level_called);
+  PrepareProcessStreamCall(msg);
   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