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>
 #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/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 30 matching lines @@
         }
       }
     }
   }
   return true;
 }
 
 }  // namespace
 
 AecDumpBasedSimulator::AecDumpBasedSimulator(const SimulationSettings& settings)
-    : AudioProcessingSimulator(settings) {}
+    : AudioProcessingSimulator(settings) {
+  if (settings_.simulate_mic_gain) {
+    LOG(LS_VERBOSE) << "Setting mic gain undo level from AEC dump";

[peah-webrtc, 2017/05/05 20:25:21]

I think that the log line is a bit unclear. 
What about 
Simulating analog microphone gain

[AleBzk, 2017/05/16 08:53:02]

I see your point, it comes out of the blue as it is.
Just wanted to notify the user that the mic level undo op will take place
because an AEC dump is given and the analog gain mic simulation mode is enabled.
So either we remove it or we make it better. I'll try the second.

+  }
+}
 
 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_[0]) * fwd_frame_.samples_per_channel_ *
                      fwd_frame_.num_channels_,
@@ skipping 66 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;
+  // Level is always logged in AEC dumps.
+  RTC_CHECK(msg.has_level());
+
+  if (settings_.simulate_mic_gain) {
+    // When the analog gain is simulated, set the undo level to |msg.level()| to
+    // virtually restore the unmodified microphone signal level.
+    real_recording_device_level_ = msg.level();
   } else {
-    *set_stream_analog_level_called = false;
+    // When the analog gain is not simulated, the AEC dump level has to be used
+    // in AudioProcessingSimulator::ProcessStream().
+    if (last_specified_microphone_level_ != msg.level()) {
+      LOG(LS_VERBOSE) << "AEC dump overriding AGC suggested level to "

[peah-webrtc, 2017/05/05 20:25:21]

(last_specified_microphone_level_ != msg.level()) will be true all the time if
you are simulating the microphone behavior. And according to previous
statements, I thing the value to have permanent log lines for this is very
limited. I therefore think you should remove lines 173-178.

[AleBzk, 2017/05/16 08:53:03]

At that point of code, there's no simulation of the mic (else branch of "if
(settings_.simulate_mic_gain) {").
Log removed.

+          << msg.level();
+    }
+    last_specified_microphone_level_ = 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 378 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