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..240d07d03e6d9c0f456e02cf0aaa4242dde1045d |
--- /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; |
+constexpr float kFloatSampleMax = 1.0f; |
+ |
+int16_t ClipSampleFloatToInt16(float sample) { |
+ return std::max(std::min(sample, static_cast<float>(kInt16SampleMax)), |
+ static_cast<float>(kInt16SampleMin)); |
+} |
+ |
+float ClipSampleFloatToFloat(float sample) { |
+ return std::max(std::min(sample, kFloatSampleMax), kFloatSampleMin); |
+} |
AleBzk
2017/07/26 13:42:30
Removed since these functions may confuse the read
|
+ |
+} // 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_; |
+}; |
+ |
+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]. |
+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 { |
+ 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) { |
+ // Virtually restore the unmodified microphone level. |
+ data[i] = |
+ ClipSampleFloatToInt16(sample_f * mic_level_ / *undo_mic_level_); |
+ } else { |
+ // Simulate the mic gain only. |
+ data[i] = ClipSampleFloatToInt16(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) { |
+ // Virtually restore the unmodified microphone level. |
+ buffer->channels()[c][i] = ClipSampleFloatToFloat( |
+ buffer->channels()[c][i] * mic_level_ / *undo_mic_level_); |
+ } else { |
+ // Simulate the mic gain only. |
+ buffer->channels()[c][i] = ClipSampleFloatToFloat( |
+ buffer->channels()[c][i] * mic_level_ / 255.0f); |
+ } |
+ } |
+ } |
+ } |
+}; |
+ |
+} // namespace |
+ |
+FakeRecordingDevice::FakeRecordingDevice(int initial_mic_level, DeviceKind kind) |
peah-webrtc
2017/06/29 22:04:00
Having seen the usage of the constructor, I defini
|
+ : mic_level_(initial_mic_level) { |
+ switch (kind) { |
+ case FakeRecordingDevice::DeviceKind::IDENTITY: |
+ worker_ = rtc::MakeUnique<FakeRecordingDeviceIdentity>(mic_level_, |
+ undo_mic_level_); |
+ break; |
+ case FakeRecordingDevice::DeviceKind::LINEAR: |
+ 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 |