audioproc_f with simulated mic analog gain

webrtc/modules/audio_processing/test/fake_rec_device_linear.cc

Index: webrtc/modules/audio_processing/test/fake_rec_device_linear.cc
diff --git a/webrtc/modules/audio_processing/test/fake_rec_device_linear.cc b/webrtc/modules/audio_processing/test/fake_rec_device_linear.cc
new file mode 100644
index 0000000000000000000000000000000000000000..8cc649fb2fd3ad26521a340302f6b3f418bc4099
--- /dev/null
+++ b/webrtc/modules/audio_processing/test/fake_rec_device_linear.cc
@@ -0,0 +1,60 @@
+/*
+ *  Copyright (c) 2017 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 "webrtc/modules/audio_processing/test/fake_rec_device_linear.h"
+
+#include "webrtc/base/logging.h"
+
+namespace webrtc {
+namespace test {
+
+FakeRecordingDeviceLinear::FakeRecordingDeviceLinear(int initial_mic_level)
+    : FakeRecordingDevice(initial_mic_level) {}
+
+FakeRecordingDeviceLinear::~FakeRecordingDeviceLinear() = default;
+
+void FakeRecordingDeviceLinear::ModifySampleInt16(int16_t* sample) {
+  float sample_f = static_cast<float>(*sample);
+  rtc::Optional<int> undo_level = undo_mic_level();
+
+  // Virtually restore the unmodified microphone level.
+  if (undo_level) {
+    if (*undo_level == 0) {
+      LOG(LS_INFO) << "undo level is zero, cannot undo mic level";
+    } else {
+      sample_f = ClipSampleInt16(
+          sample_f * 255.0f / static_cast<float>(*undo_level));
+    }
+  }

[AleBzk, 2017/05/17 11:52:24]

Let's continue the discussion about clipping between the undo operation and the
gain simulation here.
To check if this way is correct, imagine the following:
- we have an AECdump and we want to simulate AGC, so we need to virtually
restore the unmodified mic level,
- at some point of the call, the mic gain is low (maybe because someone was
shouting or was too close to the mic)
- the low gain was probably selected by AGC (during the call) to avoid
saturation
- we virtually restore the unmodified mic level first, since we want to simulate
what we would have seen if the mic gain had been higher
- so, back to the shouting/too close case, a higher gain could have turned into
saturation

Am I elaborating too much? is this reasoning correct? And even if it makes sense
and it's correct, do we want this?

+
+  // Simulate the mic gain.
+  *sample = ClipSampleInt16(
+      sample_f * static_cast<float>(mic_level()) / 255.0f);
+}
+
+void FakeRecordingDeviceLinear::ModifySampleFloat(float* sample) {
+  rtc::Optional<int> undo_level = undo_mic_level();
+
+  // Virtually restore the unmodified microphone level.
+  if (undo_level) {
+    if (*undo_level == 0) {
+      LOG(LS_INFO) << "undo level is zero, cannot undo mic level";
+    } else {
+      *sample = ClipSampleFloat(
+          *sample * 255.0f / static_cast<float>(*undo_level));
+    }
+  }
+
+  // Simulate the mic gain.
+  *sample = ClipSampleFloat(*sample * static_cast<float>(mic_level()) / 255.0f);
+}
+
+}  // namespace test
+}  // namespace webrtc