| /* |
| * 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_ |