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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 API_TEST_AUDIOPROC_FLOAT_H_
#define API_TEST_AUDIOPROC_FLOAT_H_
#include <memory>
#include <vector>
#include "modules/audio_processing/include/audio_processing.h"
namespace webrtc {
namespace test {
// This is an interface for the audio processing simulation utility. This
// utility can be used to simulate the audioprocessing module using a recording
// (either an AEC dump or wav files), and generate the output as a wav file.
// The |ap_builder| object will be used to create the AudioProcessing instance
// that is used during the simulation. The |ap_builder| supports setting of
// injectable components, which will be passed on to the created AudioProcessing
// instance. It is needed to pass the command line flags as |argc| and |argv|,
// so these can be interpreted properly by the utility.
// To get a fully-working audioproc_f utility, all that is needed is to write a
// main function, create an AudioProcessingBuilder, optionally set custom
// processing components on it, and pass the builder together with the command
// line arguments into this function.
// To see a list of all supported command line flags, run the executable with
// the '--help' flag.
int AudioprocFloat(std::unique_ptr<AudioProcessingBuilder> ap_builder,
int argc,
char* argv[]);
// Interface for the audio processing simulation utility, which is similar to
// the one above, but which adds the option of receiving the input as a string
// and returning the output as an array. The first three arguments fulfill the
// same purpose as above. Pass the |input_aecdump| to provide the content of an
// AEC dump file as a string. After the simulation is completed,
// |processed_capture_samples| will contain the the samples processed on the
// capture side.
int AudioprocFloat(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 // API_TEST_AUDIOPROC_FLOAT_H_