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This CL creates a new stream interface and uses it to replace most of the usage of AudioFrame in the non-test code. The CL changes some of the test code as well, as the other changes required that. The CL will be followed by 2 more related CLs. Bug: webrtc:5298 Change-Id: I5cfbe6079f30fc3fbf35b35fd077b6fb49c7def0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170040 Commit-Queue: Per Åhgren <peah@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30799}
64 lines
2.1 KiB
C++
64 lines
2.1 KiB
C++
/*
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* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef MODULES_AUDIO_PROCESSING_AEC_DUMP_MOCK_AEC_DUMP_H_
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#define MODULES_AUDIO_PROCESSING_AEC_DUMP_MOCK_AEC_DUMP_H_
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#include <memory>
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#include "modules/audio_processing/include/aec_dump.h"
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#include "test/gmock.h"
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namespace webrtc {
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namespace test {
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class MockAecDump : public AecDump {
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public:
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MockAecDump();
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virtual ~MockAecDump();
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MOCK_METHOD2(WriteInitMessage,
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void(const ProcessingConfig& api_format, int64_t time_now_ms));
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MOCK_METHOD1(AddCaptureStreamInput,
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void(const AudioFrameView<const float>& src));
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MOCK_METHOD1(AddCaptureStreamOutput,
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void(const AudioFrameView<const float>& src));
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MOCK_METHOD3(AddCaptureStreamInput,
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void(const int16_t* const data,
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int num_channels,
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int samples_per_channel));
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MOCK_METHOD3(AddCaptureStreamOutput,
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void(const int16_t* const data,
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int num_channels,
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int samples_per_channel));
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MOCK_METHOD1(AddAudioProcessingState,
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void(const AudioProcessingState& state));
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MOCK_METHOD0(WriteCaptureStreamMessage, void());
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MOCK_METHOD3(WriteRenderStreamMessage,
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void(const int16_t* const data,
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int num_channels,
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int samples_per_channel));
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MOCK_METHOD1(WriteRenderStreamMessage,
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void(const AudioFrameView<const float>& src));
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MOCK_METHOD1(WriteConfig, void(const InternalAPMConfig& config));
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MOCK_METHOD1(WriteRuntimeSetting,
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void(const AudioProcessing::RuntimeSetting& config));
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};
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} // namespace test
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_AEC_DUMP_MOCK_AEC_DUMP_H_
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