audioproc_f with fake microphone.

webrtc/modules/audio_processing/test/fake_recording_device.cc

+/*
+ *  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(LevelToScalingMappingKind mapping_kind)
+    : mapping_kind_(mapping_kind) {}
+
+FakeRecordingDevice::~FakeRecordingDevice() = default;
+
+// |level| must be within [0, 255].
+void FakeRecordingDevice::set_analog_level(int level) {
+  RTC_DCHECK_LE(0, level);
+  RTC_DCHECK_LE(level, 255);
+  level_ = level;
+}
+
+int FakeRecordingDevice::analog_level() const {
+  return level_;
+}
+
+void FakeRecordingDevice::ProcessStream(
+    std::vector<rtc::ArrayView<const float>> src,
+    std::vector<rtc::ArrayView<float>> dest) {
+  RTC_DCHECK_EQ(src.size(), dest.size());
+
+  const float scaling_factor = ComputeCompoundScalingFactor();
+
+  for (size_t i = 0; i < src.size(); ++i) {
+    RTC_DCHECK_EQ(src[i].size(), dest[i].size());
+    std::transform(src[i].begin(), src[i].end(), dest[i].begin(),
+                   [scaling_factor](float x) { return scaling_factor * x; });
+  }
+  real_device_level_.reset();
+}
+
+void FakeRecordingDevice::ProcessStream(const AudioFrame* src,
+                                        AudioFrame* dest) {
+  RTC_DCHECK_EQ(src->num_channels_, dest->num_channels_);
+  RTC_DCHECK_EQ(src->samples_per_channel_, dest->samples_per_channel_);
+  const size_t number_of_samples =
+      src->samples_per_channel_ * src->num_channels_;
+  RTC_DCHECK_LE(number_of_samples, AudioFrame::kMaxDataSizeSamples);
+  const float scaling_factor = ComputeCompoundScalingFactor();
+
+  std::transform(src->data_, src->data_ + number_of_samples, dest->data_,
+                 [scaling_factor](int16_t x) {
+                   return rtc::saturated_cast<int16_t>(scaling_factor * x);
+                 });
+
+  real_device_level_.reset();
+}
+
+void FakeRecordingDevice::NotifyAudioDeviceLevel(int level) {
+  real_device_level_.emplace(level);
+}
+
+float FakeRecordingDevice::ComputeCompoundScalingFactor() {
+  const float level_scaling_factor = GetScalingFactor(level_);
+  const float undo_factor =
+      real_device_level_ ? GetScalingFactor(*real_device_level_) : 1.0f;
+
+  if (undo_factor == 0) {
+    LOG(LS_INFO) << "[fake rec device] Cannot undo analog level of 0.";
+    return level_scaling_factor;
+  } else {
+    return level_scaling_factor / undo_factor;
+  }
+}
+
+float FakeRecordingDevice::GetScalingFactor(int level) const {
+  switch (mapping_kind_) {
+    case LevelToScalingMappingKind::kLinear: {
+      return static_cast<float>(level) / 255.0f;
+    }
+    default: { RTC_NOTREACHED(); }
+  }
+}
+
+}  // namespace webrtc