audioproc_f with simulated mic analog gain

webrtc/modules/audio_processing/test/fake_recording_device.cc

Index: webrtc/modules/audio_processing/test/fake_recording_device.cc
diff --git a/webrtc/modules/audio_processing/test/fake_recording_device.cc b/webrtc/modules/audio_processing/test/fake_recording_device.cc
new file mode 100644
index 0000000000000000000000000000000000000000..599ce6ab875ac1ed325758998d2448f66e279439
--- /dev/null
+++ b/webrtc/modules/audio_processing/test/fake_recording_device.cc
@@ -0,0 +1,142 @@
+/*
+ *  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_recording_device.h"
+
+#include <algorithm>
+
+#include "webrtc/base/logging.h"
+#include "webrtc/base/ptr_util.h"
+
+namespace webrtc {
+namespace test {
+
+namespace {
+
+constexpr int16_t kInt16SampleMin = -32768;
+constexpr int16_t kInt16SampleMax = 32767;
+constexpr float kFloatSampleMin = -1.0f;

[peah-webrtc, 2017/08/18 04:27:00]

The range for the float values inside APM is [-32768.f:32767.f]

+constexpr float kFloatSampleMax = 1.0f;
+
+}  // namespace
+
+// Abstract class for the different fake recording devices.
+class FakeRecordingDeviceWorker {
+ public:
+  FakeRecordingDeviceWorker(const int& mic_level,
+                            const rtc::Optional<int>& undo_mic_level)
+      : mic_level_(mic_level), undo_mic_level_(undo_mic_level) {}
+  virtual ~FakeRecordingDeviceWorker() = default;
+  virtual void ModifyBufferInt16(AudioFrame* buffer) = 0;
+  virtual void ModifyBufferFloat(ChannelBuffer<float>* buffer) = 0;
+
+ protected:
+  const int& mic_level_;
+  const rtc::Optional<int>& undo_mic_level_;

[peah-webrtc, 2017/08/18 04:27:00]

As discussed offline, please remove this as a referecence, and instead only use
the undo_mic_level_ located here.

+};
+
+namespace {
+
+// Identity fake recording device. The samples are not modified, which is
+// equivalent to a constant gain curve at 1.0 - only used for testing.
+class FakeRecordingDeviceIdentity final : public FakeRecordingDeviceWorker {
+ public:
+  FakeRecordingDeviceIdentity(const int& mic_level,
+                              const rtc::Optional<int>& undo_mic_level)
+      : FakeRecordingDeviceWorker(mic_level, undo_mic_level) {}
+  ~FakeRecordingDeviceIdentity() override = default;
+  void ModifyBufferInt16(AudioFrame* buffer) override {}
+  void ModifyBufferFloat(ChannelBuffer<float>* buffer) override {}
+};
+
+// Linear fake recording device. The gain curve is a linear function mapping the
+// mic levels range [0, 255] to [0.0, 1.0].

[peah-webrtc, 2017/08/18 04:27:00]

Since you specify the allowed range for the mic levels here, I think you should
add DCHECKS in the setter for that.

+class FakeRecordingDeviceLinear final : public FakeRecordingDeviceWorker {
+ public:
+  FakeRecordingDeviceLinear(const int& mic_level,
+                            const rtc::Optional<int>& undo_mic_level)
+      : FakeRecordingDeviceWorker(mic_level, undo_mic_level) {}
+  ~FakeRecordingDeviceLinear() override = default;
+  void ModifyBufferInt16(AudioFrame* buffer) override {

[peah-webrtc, 2017/08/18 04:27:00]

Since you add implementations within the class definition, I think the class
definition would benefit in terms of readability if you add an empty line
separating the methods that have implementations.

+    const size_t number_of_samples =
+        buffer->samples_per_channel_ * buffer->num_channels_;
+    RTC_DCHECK_LE(number_of_samples, AudioFrame::kMaxDataSizeSamples);
+    int16_t* data = buffer->mutable_data();
+    for (size_t i = 0; i < number_of_samples; ++i) {
+      const float sample_f = data[i];
+      if (undo_mic_level_ && *undo_mic_level_ > 0) {

[peah-webrtc, 2017/08/18 04:27:00]

Regarding *undo_mic_level_ > 0, please see comment below.

+        // Virtually restore the unmodified microphone level.
+        data[i] = std::max(kInt16SampleMin,
+                           std::min(kInt16SampleMax,
+                                    static_cast<int16_t>(sample_f * mic_level_ /
+                                                         *undo_mic_level_)));
+      } else {
+        // Simulate the mic gain only.
+        data[i] = std::max(
+            kInt16SampleMin,
+            std::min(kInt16SampleMax,
+                     static_cast<int16_t>(sample_f * mic_level_ / 255.0f)));
+      }
+    }
+  }
+  void ModifyBufferFloat(ChannelBuffer<float>* buffer) override {
+    for (size_t c = 0; c < buffer->num_channels(); ++c) {
+      for (size_t i = 0; i < buffer->num_frames(); ++i) {
+        if (undo_mic_level_ && *undo_mic_level_ > 0) {

[peah-webrtc, 2017/08/18 04:27:00]

why do you need to check for *undo_mic_level_ > 0 ?  If that is not a valid mic
level value, I'd rather DCHECK on it in the setter.

However, what I guess you want to do is to handle it in a specific manner. As it
is now, it is treated in the same manner as if undo_mic_level is not set.

+          // Virtually restore the unmodified microphone level.
+          buffer->channels()[c][i] = std::max(
+              kFloatSampleMin,
+              std::min(kFloatSampleMax, buffer->channels()[c][i] * mic_level_ /
+                                            *undo_mic_level_));
+        } else {
+          // Simulate the mic gain only.
+          buffer->channels()[c][i] =
+              std::max(kFloatSampleMin,
+                       std::min(kFloatSampleMax, buffer->channels()[c][i] *
+                                                     mic_level_ / 255.0f));
+        }
+      }
+    }
+  }
+};
+
+}  // namespace
+
+FakeRecordingDevice::FakeRecordingDevice(int initial_mic_level, int device_kind)
+    : mic_level_(initial_mic_level) {
+  switch (device_kind) {
+    case 0:
+      worker_ = rtc::MakeUnique<FakeRecordingDeviceIdentity>(mic_level_,
+                                                             undo_mic_level_);
+      break;
+    case 1:
+      worker_ = rtc::MakeUnique<FakeRecordingDeviceLinear>(mic_level_,
+                                                           undo_mic_level_);
+      break;
+    default:
+      RTC_NOTREACHED();
+      break;
+  }
+}
+
+FakeRecordingDevice::~FakeRecordingDevice() = default;
+
+void FakeRecordingDevice::SimulateAnalogGain(AudioFrame* buffer) {
+  RTC_DCHECK(worker_);
+  worker_->ModifyBufferInt16(buffer);
+}
+
+void FakeRecordingDevice::SimulateAnalogGain(ChannelBuffer<float>* buffer) {
+  RTC_DCHECK(worker_);
+  worker_->ModifyBufferFloat(buffer);
+}
+
+}  // namespace test
+}  // namespace webrtc