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/*
* Copyright (c) 2018 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 MODULES_AUDIO_PROCESSING_TEST_AUDIOPROC_FLOAT_IMPL_H_
#define MODULES_AUDIO_PROCESSING_TEST_AUDIOPROC_FLOAT_IMPL_H_
#include <memory>
#include "modules/audio_processing/include/audio_processing.h"
namespace webrtc {
namespace test {
// This function implements the audio processing simulation utility. Pass
// |input_aecdump| to provide the content of an AEC dump file as a string; if
// |input_aecdump| is not passed, a WAV or AEC input dump file must be specified
// via the |argv| argument. Pass |processed_capture_samples| to write in it the
// samples processed on the capture side; if |processed_capture_samples| is not
// passed, the output file can optionally be specified via the |argv| argument.
int AudioprocFloatImpl(std::unique_ptr<AudioProcessingBuilder> ap_builder,
int argc,
char* argv[],
absl::string_view input_aecdump,
std::vector<float>* processed_capture_samples);
} // namespace test
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_TEST_AUDIOPROC_FLOAT_IMPL_H_
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