| /* | 
 |  *  Copyright (c) 2019 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 TEST_TESTSUPPORT_COPY_TO_FILE_AUDIO_CAPTURER_H_ | 
 | #define TEST_TESTSUPPORT_COPY_TO_FILE_AUDIO_CAPTURER_H_ | 
 |  | 
 | #include <memory> | 
 | #include <string> | 
 |  | 
 | #include "absl/types/optional.h" | 
 | #include "api/array_view.h" | 
 | #include "common_audio/wav_file.h" | 
 | #include "modules/audio_device/include/test_audio_device.h" | 
 | #include "rtc_base/buffer.h" | 
 |  | 
 | namespace webrtc { | 
 | namespace test { | 
 |  | 
 | // TestAudioDeviceModule::Capturer that will store audio data, captured by | 
 | // delegate to the specified output file. Can be used to create a copy of | 
 | // generated audio data to be able then to compare it as a reference with | 
 | // audio on the TestAudioDeviceModule::Renderer side. | 
 | class CopyToFileAudioCapturer : public TestAudioDeviceModule::Capturer { | 
 |  public: | 
 |   CopyToFileAudioCapturer( | 
 |       std::unique_ptr<TestAudioDeviceModule::Capturer> delegate, | 
 |       std::string stream_dump_file_name); | 
 |   ~CopyToFileAudioCapturer() override; | 
 |  | 
 |   int SamplingFrequency() const override; | 
 |   int NumChannels() const override; | 
 |   bool Capture(rtc::BufferT<int16_t>* buffer) override; | 
 |  | 
 |  private: | 
 |   std::unique_ptr<TestAudioDeviceModule::Capturer> delegate_; | 
 |   std::unique_ptr<WavWriter> wav_writer_; | 
 | }; | 
 |  | 
 | }  // namespace test | 
 | }  // namespace webrtc | 
 |  | 
 | #endif  // TEST_TESTSUPPORT_COPY_TO_FILE_AUDIO_CAPTURER_H_ |