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..c9ca6bdaec9d505c254874f7ec1a02094d87e0b4 |
--- /dev/null |
+++ b/webrtc/modules/audio_processing/test/fake_recording_device.cc |
@@ -0,0 +1,92 @@ |
+/* |
+ * 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/modules/audio_processing/test/fake_rec_device_identity.h" |
+#include "webrtc/modules/audio_processing/test/fake_rec_device_linear.h" |
+ |
+#include "webrtc/base/logging.h" |
+#include "webrtc/base/ptr_util.h" |
+ |
+namespace webrtc { |
+namespace test { |
+ |
+namespace { |
+ |
+const int16_t kSampleMinInt16 = 32767; |
+const int16_t kSampleMaxInt16 = -32768; |
+const float kSampleMinFloat = -1.0f; |
+const float kSampleMaxFloat = 1.0f; |
+ |
+} // namespace |
+ |
+FakeRecordingDevice::FakeRecordingDevice(int initial_mic_level) |
+ : mic_level_(initial_mic_level) {} |
+ |
+std::unique_ptr<FakeRecordingDevice> FakeRecordingDevice::GetFakeRecDevice( |
+ FakeRecordingDevice::DeviceKind kind, int initial_mic_level) { |
+ switch (kind) { |
+ case FakeRecordingDevice::DeviceKind::IDENTITY: { |
+ return rtc::MakeUnique<FakeRecordingDeviceIdentity>(initial_mic_level); |
+ } |
+ case FakeRecordingDevice::DeviceKind::LINEAR: { |
+ return rtc::MakeUnique<FakeRecordingDeviceLinear>(initial_mic_level); |
+ } |
+ default: { |
+ RTC_NOTREACHED(); |
+ } |
+ } |
+} |
+ |
+FakeRecordingDevice::~FakeRecordingDevice() = default; |
+ |
+void FakeRecordingDevice::set_mic_level(int level) { |
+ mic_level_ = level; |
+} |
+ |
+int FakeRecordingDevice::mic_level() const { |
+ return mic_level_; |
+} |
+ |
+void FakeRecordingDevice::set_undo_mic_level(rtc::Optional<int> level) { |
+ undo_mic_level_ = level; |
+} |
+ |
+rtc::Optional<int> FakeRecordingDevice::undo_mic_level() const { |
+ return undo_mic_level_; |
+} |
+ |
+void FakeRecordingDevice::SimulateAnalogGain(ChannelBuffer<float>* buffer) { |
+ size_t number_of_samples = buffer->num_frames(); |
+ for (size_t i = 0; i < buffer->num_channels(); ++i) { |
+ std::for_each(buffer->channels()[i], |
+ buffer->channels()[i] + number_of_samples, |
+ [this](float& x) { ModifySampleFloat(&x); }); |
+ } |
+} |
+ |
+void FakeRecordingDevice::SimulateAnalogGain(AudioFrame* buffer) { |
+ const size_t number_of_samples = |
+ buffer->samples_per_channel_ * buffer->num_channels_; |
+ RTC_DCHECK_LE(number_of_samples, AudioFrame::kMaxDataSizeSamples); |
+ std::for_each(buffer->data_, buffer->data_ + number_of_samples, |
+ [this](int16_t& x) { ModifySampleInt16(&x); }); |
+} |
+ |
+int16_t FakeRecordingDevice::ClipSampleInt16(int16_t sample) { |
+ return std::max(std::min(sample, kSampleMaxInt16), kSampleMinInt16); |
+} |
+ |
+float FakeRecordingDevice::ClipSampleFloat(float sample) { |
+ return std::max(std::min(sample, kSampleMaxFloat), kSampleMinFloat); |
+} |
+ |
+} // namespace test |
+} // namespace webrtc |