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libtgvoip/webrtc_dsp/modules/audio_processing/include/audio_processing_statistics.cc

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/*
* Copyright 2017 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 "modules/audio_processing/include/audio_processing_statistics.h"
namespace webrtc {
AudioProcessingStats::AudioProcessingStats() = default;
AudioProcessingStats::AudioProcessingStats(const AudioProcessingStats& other) =
default;
AudioProcessingStats::~AudioProcessingStats() = default;
} // namespace webrtc