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..2b5b13b71f32cfcff34cf5d753d989522f785b0b
--- /dev/null
+++ b/webrtc/modules/audio_processing/test/fake_recording_device.cc
@@ -0,0 +1,128 @@
+/*
+ *  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 "webrtc/base/logging.h"
+
+namespace webrtc {
+
+FakeRecordingDevice::FakeRecordingDevice(MicrophoneKind mapping_kind)
+    : mapping_kind_(mapping_kind) {
+  // Create the analog gain simulatory callback depending on the wanted
+  // microphone kind.
+  switch (mapping_kind_) {
+    case MicrophoneKind::kIdentity: {
+      simulator_callback_int16_ = AnalogGainSimulatorIdentityInt16();

[peah-webrtc, 2017/05/16 12:19:36]

These are not used as callbacks, so I don't think that should be part of the
name.

[AleBzk, 2017/05/17 11:52:23]

Acknowledged.

+      simulator_callback_float_ = AnalogGainSimulatorIdentityFloat();
+      break;
+    }
+    case MicrophoneKind::kLinear: {
+      simulator_callback_int16_ = AnalogGainSimulatorLinearInt16();

[peah-webrtc, 2017/05/16 12:19:36]

I would rather have an interface with different implementations of the analog
gains than doing it like this with method pointers (I know it sounds like the
opposite to what I stated before but this is a bit different). For this to work
the callbacks need to have access to the full scope of the FakeRecordingDevice
class and I think the resulting code becomes a bit more complicated than it has
to.

For instance,
-you need to declare SimulatorCallbackInt16 in the header, as well as the
variables of SimulatorCallbackInt16. 
-The std::transform calls contain lambdas with full object scope that call
methods using method pointers.

[AleBzk, 2017/05/17 11:52:23]

Perfect. Let's then go for interface + implementations :)

+      simulator_callback_float_ = AnalogGainSimulatorLinearFloat();
+      break;
+    }
+    default: { RTC_NOTREACHED(); }
+  }
+}
+
+FakeRecordingDevice::~FakeRecordingDevice() = default;
+
+void FakeRecordingDevice::SimulateAnalogGain(
+    int level, rtc::Optional<int> real_device_level,
+    std::vector<rtc::ArrayView<float>> buffer) {
+  for (size_t i = 0; i < buffer.size(); ++i) {
+    std::transform(

[peah-webrtc, 2017/05/16 12:19:35]

I think std::for_each is better suited here.

[AleBzk, 2017/05/17 11:52:23]

Acknowledged.

+        buffer[i].begin(), buffer[i].end(), buffer[i].begin(),
+        [this, level, real_device_level](float x) {
+          return simulator_callback_float_(x, level, real_device_level);

[peah-webrtc, 2017/05/16 12:19:36]

As you stated offline, this will not scale with memory in the microphone. But
lets keep it for now until we know if we need memory.

[AleBzk, 2017/05/17 11:52:23]

Acknowledged.

+        });
+  }
+}
+
+void FakeRecordingDevice::SimulateAnalogGain(

[peah-webrtc, 2017/05/16 12:19:36]

Why is this needed as a separate method?

[AleBzk, 2017/05/17 11:52:23]

Please, see my answer to a related question of yours at line 63 in
https://codereview.webrtc.org/2834643002/diff/140001/webrtc/modules/audio_pro...

+    int level, rtc::Optional<int> real_device_level,
+    ChannelBuffer<float>* buffer) {
+  std::vector<rtc::ArrayView<float>> buffer_view;
+  for (size_t i = 0; i < buffer->num_channels(); ++i) {
+    buffer_view.emplace_back(buffer->channels()[i], buffer->num_frames());
+  }
+  SimulateAnalogGain(level, real_device_level, buffer_view);
+}
+
+void FakeRecordingDevice::SimulateAnalogGain(
+    int level, rtc::Optional<int> real_device_level, AudioFrame* buffer) {
+  const size_t number_of_samples =
+      buffer->samples_per_channel_ * buffer->num_channels_;
+  RTC_DCHECK_LE(number_of_samples, AudioFrame::kMaxDataSizeSamples);
+  std::transform(

[peah-webrtc, 2017/05/16 12:19:36]

std::for_each

[AleBzk, 2017/05/17 11:52:23]

Acknowledged.

+      buffer->data_, buffer->data_ + number_of_samples, buffer->data_,
+      [this, level, real_device_level](int16_t x) {
+        return simulator_callback_int16_(x, level, real_device_level);
+      });
+}
+
+SimulatorCallbackInt16 FakeRecordingDevice::AnalogGainSimulatorIdentityInt16() {
+  return [this](int16_t sample, int level,
+                rtc::Optional<int> real_device_level) {
+    return sample;
+  };
+}
+
+SimulatorCallbackFloat FakeRecordingDevice::AnalogGainSimulatorIdentityFloat() {
+  return [this](float sample, int level, rtc::Optional<int> real_device_level) {
+    return sample;
+  };
+}
+
+SimulatorCallbackInt16 FakeRecordingDevice::AnalogGainSimulatorLinearInt16() {
+  return [this](int16_t sample, int level,
+                rtc::Optional<int> real_device_level) {
+    float sample_f = static_cast<float>(sample);
+
+    // Virtually restore the unmodified microphone level.
+    if (real_device_level) {
+      RTC_DCHECK_GT(*real_device_level, 0)
+          << "zero real device level, cannot undo mic level";
+      sample_f = ClipSample(
+          sample_f * 255.0f / static_cast<float>(*real_device_level));
+    }
+
+    // Simulate the mic gain.
+    return ClipSample(sample_f * static_cast<float>(level) / 255.0f);
+  };
+}
+
+SimulatorCallbackFloat FakeRecordingDevice::AnalogGainSimulatorLinearFloat() {
+  return [this](float sample, int level, rtc::Optional<int> real_device_level) {
+    // Virtually restore the unmodified microphone level.
+    if (real_device_level) {
+      RTC_DCHECK_GT(*real_device_level, 0)
+          << "zero real device level, cannot undo mic level";
+      sample = ClipSample(
+          sample * 255.0f / static_cast<float>(*real_device_level));
+    }
+
+    // Simulate the mic gain.
+    return ClipSample(sample * static_cast<float>(level) / 255.0f);
+  };
+}
+
+int16_t FakeRecordingDevice::ClipSample(int16_t sample) {
+  return sample < kSampleMinInt16 ?
+      kSampleMinInt16 : (sample > kSampleMaxInt16 ? kSampleMaxInt16 : sample);

[peah-webrtc, 2017/05/16 12:19:35]

Use std::min/std::max instead

[peah-webrtc, 2017/05/16 12:19:36]

Why do you clip the sample value to +-255?

[AleBzk, 2017/05/17 11:52:23]

Done.

[AleBzk, 2017/05/17 11:52:23]

Right, I got confused with the mic level.

+}
+
+float FakeRecordingDevice::ClipSample(float sample) {
+  return sample < kSampleMinFloat ?

[peah-webrtc, 2017/05/16 12:19:36]

Use std::min/std::max instead

[AleBzk, 2017/05/17 11:52:23]

Done.

+      kSampleMinFloat : (sample > kSampleMaxFloat ? kSampleMaxFloat : sample);
+}
+
+}  // namespace webrtc