43 lines
1.8 KiB
C
43 lines
1.8 KiB
C
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/*
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* Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef API_TEST_AUDIOPROC_FLOAT_H_
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#define API_TEST_AUDIOPROC_FLOAT_H_
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#include <memory>
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#include "modules/audio_processing/include/audio_processing.h"
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namespace webrtc {
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namespace test {
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// This is an interface for the audio processing simulation utility. This
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// utility can be used to simulate the audioprocessing module using a recording
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// (either an AEC dump or wav files), and generate the output as a wav file.
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// The |ap_builder| object will be used to create the AudioProcessing instance
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// that is used during the simulation. The |ap_builder| supports setting of
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// injectable components, which will be passed on to the created AudioProcessing
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// instance. It is needed to pass the command line flags as |argc| and |argv|,
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// so these can be interpreted properly by the utility.
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// To get a fully-working audioproc_f utility, all that is needed is to write a
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// main function, create an AudioProcessingBuilder, optionally set custom
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// processing components on it, and pass the builder together with the command
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// line arguments into this function.
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// To see a list of all supported command line flags, run the executable with
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// the '--help' flag.
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int AudioprocFloat(std::unique_ptr<AudioProcessingBuilder> ap_builder,
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int argc,
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char* argv[]);
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} // namespace test
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} // namespace webrtc
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#endif // API_TEST_AUDIOPROC_FLOAT_H_
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