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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/aec3/mock | |
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/aec3/mock')
8 files changed, 314 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/mock/mock_block_processor.cc b/third_party/libwebrtc/modules/audio_processing/aec3/mock/mock_block_processor.cc new file mode 100644 index 0000000000..c5c33dbd68 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/aec3/mock/mock_block_processor.cc @@ -0,0 +1,20 @@ +/* + * Copyright (c) 2018 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/aec3/mock/mock_block_processor.h" + +namespace webrtc { +namespace test { + +MockBlockProcessor::MockBlockProcessor() = default; +MockBlockProcessor::~MockBlockProcessor() = default; + +} // namespace test +} // namespace webrtc diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/mock/mock_block_processor.h b/third_party/libwebrtc/modules/audio_processing/aec3/mock/mock_block_processor.h new file mode 100644 index 0000000000..c9ae38c4aa --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/aec3/mock/mock_block_processor.h @@ -0,0 +1,53 @@ +/* + * Copyright (c) 2016 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. + */ + +#ifndef MODULES_AUDIO_PROCESSING_AEC3_MOCK_MOCK_BLOCK_PROCESSOR_H_ +#define MODULES_AUDIO_PROCESSING_AEC3_MOCK_MOCK_BLOCK_PROCESSOR_H_ + +#include <vector> + +#include "modules/audio_processing/aec3/block_processor.h" +#include "test/gmock.h" + +namespace webrtc { +namespace test { + +class MockBlockProcessor : public BlockProcessor { + public: + MockBlockProcessor(); + virtual ~MockBlockProcessor(); + + MOCK_METHOD(void, + ProcessCapture, + (bool level_change, + bool saturated_microphone_signal, + Block* linear_output, + Block* capture_block), + (override)); + MOCK_METHOD(void, BufferRender, (const Block& block), (override)); + MOCK_METHOD(void, + UpdateEchoLeakageStatus, + (bool leakage_detected), + (override)); + MOCK_METHOD(void, + GetMetrics, + (EchoControl::Metrics * metrics), + (const, override)); + MOCK_METHOD(void, SetAudioBufferDelay, (int delay_ms), (override)); + MOCK_METHOD(void, + SetCaptureOutputUsage, + (bool capture_output_used), + (override)); +}; + +} // namespace test +} // namespace webrtc + +#endif // MODULES_AUDIO_PROCESSING_AEC3_MOCK_MOCK_BLOCK_PROCESSOR_H_ diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/mock/mock_echo_remover.cc b/third_party/libwebrtc/modules/audio_processing/aec3/mock/mock_echo_remover.cc new file mode 100644 index 0000000000..b903bf0785 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/aec3/mock/mock_echo_remover.cc @@ -0,0 +1,20 @@ +/* + * Copyright (c) 2018 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/aec3/mock/mock_echo_remover.h" + +namespace webrtc { +namespace test { + +MockEchoRemover::MockEchoRemover() = default; +MockEchoRemover::~MockEchoRemover() = default; + +} // namespace test +} // namespace webrtc diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/mock/mock_echo_remover.h b/third_party/libwebrtc/modules/audio_processing/aec3/mock/mock_echo_remover.h new file mode 100644 index 0000000000..31f075ef0a --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/aec3/mock/mock_echo_remover.h @@ -0,0 +1,56 @@ +/* + * 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. + */ + +#ifndef MODULES_AUDIO_PROCESSING_AEC3_MOCK_MOCK_ECHO_REMOVER_H_ +#define MODULES_AUDIO_PROCESSING_AEC3_MOCK_MOCK_ECHO_REMOVER_H_ + +#include <vector> + +#include "absl/types/optional.h" +#include "modules/audio_processing/aec3/echo_path_variability.h" +#include "modules/audio_processing/aec3/echo_remover.h" +#include "modules/audio_processing/aec3/render_buffer.h" +#include "test/gmock.h" + +namespace webrtc { +namespace test { + +class MockEchoRemover : public EchoRemover { + public: + MockEchoRemover(); + virtual ~MockEchoRemover(); + + MOCK_METHOD(void, + ProcessCapture, + (EchoPathVariability echo_path_variability, + bool capture_signal_saturation, + const absl::optional<DelayEstimate>& delay_estimate, + RenderBuffer* render_buffer, + Block* linear_output, + Block* capture), + (override)); + MOCK_METHOD(void, + UpdateEchoLeakageStatus, + (bool leakage_detected), + (override)); + MOCK_METHOD(void, + GetMetrics, + (EchoControl::Metrics * metrics), + (const, override)); + MOCK_METHOD(void, + SetCaptureOutputUsage, + (bool capture_output_used), + (override)); +}; + +} // namespace test +} // namespace webrtc + +#endif // MODULES_AUDIO_PROCESSING_AEC3_MOCK_MOCK_ECHO_REMOVER_H_ diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/mock/mock_render_delay_buffer.cc b/third_party/libwebrtc/modules/audio_processing/aec3/mock/mock_render_delay_buffer.cc new file mode 100644 index 0000000000..d4ad09b4bc --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/aec3/mock/mock_render_delay_buffer.cc @@ -0,0 +1,36 @@ +/* + * Copyright (c) 2018 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/aec3/mock/mock_render_delay_buffer.h" + +namespace webrtc { +namespace test { + +MockRenderDelayBuffer::MockRenderDelayBuffer(int sample_rate_hz, + size_t num_channels) + : block_buffer_(GetRenderDelayBufferSize(4, 4, 12), + NumBandsForRate(sample_rate_hz), + num_channels), + spectrum_buffer_(block_buffer_.buffer.size(), num_channels), + fft_buffer_(block_buffer_.buffer.size(), num_channels), + render_buffer_(&block_buffer_, &spectrum_buffer_, &fft_buffer_), + downsampled_render_buffer_(GetDownSampledBufferSize(4, 4)) { + ON_CALL(*this, GetRenderBuffer()) + .WillByDefault( + ::testing::Invoke(this, &MockRenderDelayBuffer::FakeGetRenderBuffer)); + ON_CALL(*this, GetDownsampledRenderBuffer()) + .WillByDefault(::testing::Invoke( + this, &MockRenderDelayBuffer::FakeGetDownsampledRenderBuffer)); +} + +MockRenderDelayBuffer::~MockRenderDelayBuffer() = default; + +} // namespace test +} // namespace webrtc diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/mock/mock_render_delay_buffer.h b/third_party/libwebrtc/modules/audio_processing/aec3/mock/mock_render_delay_buffer.h new file mode 100644 index 0000000000..c17fd62caa --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/aec3/mock/mock_render_delay_buffer.h @@ -0,0 +1,67 @@ +/* + * 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. + */ + +#ifndef MODULES_AUDIO_PROCESSING_AEC3_MOCK_MOCK_RENDER_DELAY_BUFFER_H_ +#define MODULES_AUDIO_PROCESSING_AEC3_MOCK_MOCK_RENDER_DELAY_BUFFER_H_ + +#include <vector> + +#include "modules/audio_processing/aec3/aec3_common.h" +#include "modules/audio_processing/aec3/downsampled_render_buffer.h" +#include "modules/audio_processing/aec3/render_buffer.h" +#include "modules/audio_processing/aec3/render_delay_buffer.h" +#include "test/gmock.h" + +namespace webrtc { +namespace test { + +class MockRenderDelayBuffer : public RenderDelayBuffer { + public: + MockRenderDelayBuffer(int sample_rate_hz, size_t num_channels); + virtual ~MockRenderDelayBuffer(); + + MOCK_METHOD(void, Reset, (), (override)); + MOCK_METHOD(RenderDelayBuffer::BufferingEvent, + Insert, + (const Block& block), + (override)); + MOCK_METHOD(void, HandleSkippedCaptureProcessing, (), (override)); + MOCK_METHOD(RenderDelayBuffer::BufferingEvent, + PrepareCaptureProcessing, + (), + (override)); + MOCK_METHOD(bool, AlignFromDelay, (size_t delay), (override)); + MOCK_METHOD(void, AlignFromExternalDelay, (), (override)); + MOCK_METHOD(size_t, Delay, (), (const, override)); + MOCK_METHOD(size_t, MaxDelay, (), (const, override)); + MOCK_METHOD(RenderBuffer*, GetRenderBuffer, (), (override)); + MOCK_METHOD(const DownsampledRenderBuffer&, + GetDownsampledRenderBuffer, + (), + (const, override)); + MOCK_METHOD(void, SetAudioBufferDelay, (int delay_ms), (override)); + MOCK_METHOD(bool, HasReceivedBufferDelay, (), (override)); + + private: + RenderBuffer* FakeGetRenderBuffer() { return &render_buffer_; } + const DownsampledRenderBuffer& FakeGetDownsampledRenderBuffer() const { + return downsampled_render_buffer_; + } + BlockBuffer block_buffer_; + SpectrumBuffer spectrum_buffer_; + FftBuffer fft_buffer_; + RenderBuffer render_buffer_; + DownsampledRenderBuffer downsampled_render_buffer_; +}; + +} // namespace test +} // namespace webrtc + +#endif // MODULES_AUDIO_PROCESSING_AEC3_MOCK_MOCK_RENDER_DELAY_BUFFER_H_ diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/mock/mock_render_delay_controller.cc b/third_party/libwebrtc/modules/audio_processing/aec3/mock/mock_render_delay_controller.cc new file mode 100644 index 0000000000..4ae2af96bf --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/aec3/mock/mock_render_delay_controller.cc @@ -0,0 +1,20 @@ +/* + * Copyright (c) 2018 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/aec3/mock/mock_render_delay_controller.h" + +namespace webrtc { +namespace test { + +MockRenderDelayController::MockRenderDelayController() = default; +MockRenderDelayController::~MockRenderDelayController() = default; + +} // namespace test +} // namespace webrtc diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/mock/mock_render_delay_controller.h b/third_party/libwebrtc/modules/audio_processing/aec3/mock/mock_render_delay_controller.h new file mode 100644 index 0000000000..14d499dd28 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/aec3/mock/mock_render_delay_controller.h @@ -0,0 +1,42 @@ +/* + * 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. + */ + +#ifndef MODULES_AUDIO_PROCESSING_AEC3_MOCK_MOCK_RENDER_DELAY_CONTROLLER_H_ +#define MODULES_AUDIO_PROCESSING_AEC3_MOCK_MOCK_RENDER_DELAY_CONTROLLER_H_ + +#include "absl/types/optional.h" +#include "api/array_view.h" +#include "modules/audio_processing/aec3/downsampled_render_buffer.h" +#include "modules/audio_processing/aec3/render_delay_controller.h" +#include "test/gmock.h" + +namespace webrtc { +namespace test { + +class MockRenderDelayController : public RenderDelayController { + public: + MockRenderDelayController(); + virtual ~MockRenderDelayController(); + + MOCK_METHOD(void, Reset, (bool reset_delay_statistics), (override)); + MOCK_METHOD(void, LogRenderCall, (), (override)); + MOCK_METHOD(absl::optional<DelayEstimate>, + GetDelay, + (const DownsampledRenderBuffer& render_buffer, + size_t render_delay_buffer_delay, + const Block& capture), + (override)); + MOCK_METHOD(bool, HasClockdrift, (), (const, override)); +}; + +} // namespace test +} // namespace webrtc + +#endif // MODULES_AUDIO_PROCESSING_AEC3_MOCK_MOCK_RENDER_DELAY_CONTROLLER_H_ |