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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
commit | 26a029d407be480d791972afb5975cf62c9360a6 (patch) | |
tree | f435a8308119effd964b339f76abb83a57c29483 /third_party/libwebrtc/modules/audio_processing/test/fake_recording_device.cc | |
parent | Initial commit. (diff) | |
download | firefox-26a029d407be480d791972afb5975cf62c9360a6.tar.xz firefox-26a029d407be480d791972afb5975cf62c9360a6.zip |
Adding upstream version 124.0.1.upstream/124.0.1
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/modules/audio_processing/test/fake_recording_device.cc')
-rw-r--r-- | third_party/libwebrtc/modules/audio_processing/test/fake_recording_device.cc | 190 |
1 files changed, 190 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_processing/test/fake_recording_device.cc b/third_party/libwebrtc/modules/audio_processing/test/fake_recording_device.cc new file mode 100644 index 0000000000..3fd80b2ede --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/test/fake_recording_device.cc @@ -0,0 +1,190 @@ +/* + * 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 "modules/audio_processing/test/fake_recording_device.h" + +#include <algorithm> +#include <memory> + +#include "absl/types/optional.h" +#include "modules/audio_processing/agc2/gain_map_internal.h" +#include "rtc_base/logging.h" +#include "rtc_base/numerics/safe_conversions.h" +#include "rtc_base/numerics/safe_minmax.h" + +namespace webrtc { +namespace test { + +namespace { + +constexpr float kFloatSampleMin = -32768.f; +constexpr float kFloatSampleMax = 32767.0f; + +} // namespace + +// Abstract class for the different fake recording devices. +class FakeRecordingDeviceWorker { + public: + explicit FakeRecordingDeviceWorker(const int initial_mic_level) + : mic_level_(initial_mic_level) {} + int mic_level() const { return mic_level_; } + void set_mic_level(const int level) { mic_level_ = level; } + void set_undo_mic_level(const int level) { undo_mic_level_ = level; } + virtual ~FakeRecordingDeviceWorker() = default; + virtual void ModifyBufferInt16(rtc::ArrayView<int16_t> buffer) = 0; + virtual void ModifyBufferFloat(ChannelBuffer<float>* buffer) = 0; + + protected: + // Mic level to simulate. + int mic_level_; + // Optional mic level to undo. + absl::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: + explicit FakeRecordingDeviceIdentity(const int initial_mic_level) + : FakeRecordingDeviceWorker(initial_mic_level) {} + ~FakeRecordingDeviceIdentity() override = default; + void ModifyBufferInt16(rtc::ArrayView<int16_t> 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: + explicit FakeRecordingDeviceLinear(const int initial_mic_level) + : FakeRecordingDeviceWorker(initial_mic_level) {} + ~FakeRecordingDeviceLinear() override = default; + void ModifyBufferInt16(rtc::ArrayView<int16_t> buffer) override { + const size_t number_of_samples = buffer.size(); + int16_t* data = buffer.data(); + // If an undo level is specified, virtually restore the unmodified + // microphone level; otherwise simulate the mic gain only. + const float divisor = + (undo_mic_level_ && *undo_mic_level_ > 0) ? *undo_mic_level_ : 255.f; + for (size_t i = 0; i < number_of_samples; ++i) { + data[i] = rtc::saturated_cast<int16_t>(data[i] * mic_level_ / divisor); + } + } + void ModifyBufferFloat(ChannelBuffer<float>* buffer) override { + // If an undo level is specified, virtually restore the unmodified + // microphone level; otherwise simulate the mic gain only. + const float divisor = + (undo_mic_level_ && *undo_mic_level_ > 0) ? *undo_mic_level_ : 255.f; + for (size_t c = 0; c < buffer->num_channels(); ++c) { + for (size_t i = 0; i < buffer->num_frames(); ++i) { + buffer->channels()[c][i] = + rtc::SafeClamp(buffer->channels()[c][i] * mic_level_ / divisor, + kFloatSampleMin, kFloatSampleMax); + } + } + } +}; + +float ComputeAgcLinearFactor(const absl::optional<int>& undo_mic_level, + int mic_level) { + // If an undo level is specified, virtually restore the unmodified + // microphone level; otherwise simulate the mic gain only. + const int undo_level = + (undo_mic_level && *undo_mic_level > 0) ? *undo_mic_level : 100; + return DbToRatio(kGainMap[mic_level] - kGainMap[undo_level]); +} + +// Roughly dB-scale fake recording device. Valid levels are [0, 255]. The mic +// applies a gain from kGainMap in agc/gain_map_internal.h. +class FakeRecordingDeviceAgc final : public FakeRecordingDeviceWorker { + public: + explicit FakeRecordingDeviceAgc(const int initial_mic_level) + : FakeRecordingDeviceWorker(initial_mic_level) {} + ~FakeRecordingDeviceAgc() override = default; + void ModifyBufferInt16(rtc::ArrayView<int16_t> buffer) override { + const float scaling_factor = + ComputeAgcLinearFactor(undo_mic_level_, mic_level_); + const size_t number_of_samples = buffer.size(); + int16_t* data = buffer.data(); + for (size_t i = 0; i < number_of_samples; ++i) { + data[i] = rtc::saturated_cast<int16_t>(data[i] * scaling_factor); + } + } + void ModifyBufferFloat(ChannelBuffer<float>* buffer) override { + const float scaling_factor = + ComputeAgcLinearFactor(undo_mic_level_, mic_level_); + for (size_t c = 0; c < buffer->num_channels(); ++c) { + for (size_t i = 0; i < buffer->num_frames(); ++i) { + buffer->channels()[c][i] = + rtc::SafeClamp(buffer->channels()[c][i] * scaling_factor, + kFloatSampleMin, kFloatSampleMax); + } + } + } +}; + +} // namespace + +FakeRecordingDevice::FakeRecordingDevice(int initial_mic_level, + int device_kind) { + switch (device_kind) { + case 0: + worker_ = + std::make_unique<FakeRecordingDeviceIdentity>(initial_mic_level); + break; + case 1: + worker_ = std::make_unique<FakeRecordingDeviceLinear>(initial_mic_level); + break; + case 2: + worker_ = std::make_unique<FakeRecordingDeviceAgc>(initial_mic_level); + break; + default: + RTC_DCHECK_NOTREACHED(); + break; + } +} + +FakeRecordingDevice::~FakeRecordingDevice() = default; + +int FakeRecordingDevice::MicLevel() const { + RTC_CHECK(worker_); + return worker_->mic_level(); +} + +void FakeRecordingDevice::SetMicLevel(const int level) { + RTC_CHECK(worker_); + if (level != worker_->mic_level()) + RTC_LOG(LS_INFO) << "Simulate mic level update: " << level; + worker_->set_mic_level(level); +} + +void FakeRecordingDevice::SetUndoMicLevel(const int level) { + RTC_DCHECK(worker_); + // TODO(alessiob): The behavior with undo level equal to zero is not clear yet + // and will be defined in future CLs once more FakeRecordingDeviceWorker + // implementations need to be added. + RTC_CHECK(level > 0) << "Zero undo mic level is unsupported"; + worker_->set_undo_mic_level(level); +} + +void FakeRecordingDevice::SimulateAnalogGain(rtc::ArrayView<int16_t> buffer) { + RTC_DCHECK(worker_); + worker_->ModifyBufferInt16(buffer); +} + +void FakeRecordingDevice::SimulateAnalogGain(ChannelBuffer<float>* buffer) { + RTC_DCHECK(worker_); + worker_->ModifyBufferFloat(buffer); +} + +} // namespace test +} // namespace webrtc |