/* * 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 #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. // Any audio_processing object specified in the input is used for the // simulation. Note that when the audio_processing object is specified all // functionality that relies on using the internal builder is deactivated, // since the AudioProcessing object is already created and the builder is not // used in the simulation. int AudioprocFloatImpl(rtc::scoped_refptr audio_processing, int argc, char* argv[]); // 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 ap_builder, int argc, char* argv[], absl::string_view input_aecdump, std::vector* processed_capture_samples); } // namespace test } // namespace webrtc #endif // MODULES_AUDIO_PROCESSING_TEST_AUDIOPROC_FLOAT_IMPL_H_