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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/aec_dump/mock_aec_dump.h | |
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/aec_dump/mock_aec_dump.h')
-rw-r--r-- | third_party/libwebrtc/modules/audio_processing/aec_dump/mock_aec_dump.h | 82 |
1 files changed, 82 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_processing/aec_dump/mock_aec_dump.h b/third_party/libwebrtc/modules/audio_processing/aec_dump/mock_aec_dump.h new file mode 100644 index 0000000000..b396739de4 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/aec_dump/mock_aec_dump.h @@ -0,0 +1,82 @@ +/* + * 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_AEC_DUMP_MOCK_AEC_DUMP_H_ +#define MODULES_AUDIO_PROCESSING_AEC_DUMP_MOCK_AEC_DUMP_H_ + +#include <memory> + +#include "modules/audio_processing/include/aec_dump.h" +#include "test/gmock.h" + +namespace webrtc { + +namespace test { + +class MockAecDump : public AecDump { + public: + MockAecDump(); + virtual ~MockAecDump(); + + MOCK_METHOD(void, + WriteInitMessage, + (const ProcessingConfig& api_format, int64_t time_now_ms), + (override)); + + MOCK_METHOD(void, + AddCaptureStreamInput, + (const AudioFrameView<const float>& src), + (override)); + MOCK_METHOD(void, + AddCaptureStreamOutput, + (const AudioFrameView<const float>& src), + (override)); + MOCK_METHOD(void, + AddCaptureStreamInput, + (const int16_t* const data, + int num_channels, + int samples_per_channel), + (override)); + MOCK_METHOD(void, + AddCaptureStreamOutput, + (const int16_t* const data, + int num_channels, + int samples_per_channel), + (override)); + MOCK_METHOD(void, + AddAudioProcessingState, + (const AudioProcessingState& state), + (override)); + MOCK_METHOD(void, WriteCaptureStreamMessage, (), (override)); + + MOCK_METHOD(void, + WriteRenderStreamMessage, + (const int16_t* const data, + int num_channels, + int samples_per_channel), + (override)); + MOCK_METHOD(void, + WriteRenderStreamMessage, + (const AudioFrameView<const float>& src), + (override)); + + MOCK_METHOD(void, WriteConfig, (const InternalAPMConfig& config), (override)); + + MOCK_METHOD(void, + WriteRuntimeSetting, + (const AudioProcessing::RuntimeSetting& config), + (override)); +}; + +} // namespace test + +} // namespace webrtc + +#endif // MODULES_AUDIO_PROCESSING_AEC_DUMP_MOCK_AEC_DUMP_H_ |