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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 19:33:14 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 19:33:14 +0000 |
commit | 36d22d82aa202bb199967e9512281e9a53db42c9 (patch) | |
tree | 105e8c98ddea1c1e4784a60a5a6410fa416be2de /third_party/libwebrtc/test/testsupport/copy_to_file_audio_capturer.h | |
parent | Initial commit. (diff) | |
download | firefox-esr-36d22d82aa202bb199967e9512281e9a53db42c9.tar.xz firefox-esr-36d22d82aa202bb199967e9512281e9a53db42c9.zip |
Adding upstream version 115.7.0esr.upstream/115.7.0esrupstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/test/testsupport/copy_to_file_audio_capturer.h')
-rw-r--r-- | third_party/libwebrtc/test/testsupport/copy_to_file_audio_capturer.h | 49 |
1 files changed, 49 insertions, 0 deletions
diff --git a/third_party/libwebrtc/test/testsupport/copy_to_file_audio_capturer.h b/third_party/libwebrtc/test/testsupport/copy_to_file_audio_capturer.h new file mode 100644 index 0000000000..a410beeea8 --- /dev/null +++ b/third_party/libwebrtc/test/testsupport/copy_to_file_audio_capturer.h @@ -0,0 +1,49 @@ +/* + * 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_ |