blob: a410beeea89dcc2ef4cdcae2543a48dc4efe4aa9 [file] [log] [blame]
* 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.
#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 {
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;
std::unique_ptr<TestAudioDeviceModule::Capturer> delegate_;
std::unique_ptr<WavWriter> wav_writer_;
} // namespace test
} // namespace webrtc