
Review URL: http://webrtc-codereview.appspot.com/38002 git-svn-id: http://webrtc.googlecode.com/svn/trunk@90 4adac7df-926f-26a2-2b94-8c16560cd09d
37 lines
1013 B
C++
37 lines
1013 B
C++
/*
|
|
* Copyright (c) 2011 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 WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_TEST_UNIT_TEST_UNIT_TEST_H_
|
|
#define WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_TEST_UNIT_TEST_UNIT_TEST_H_
|
|
|
|
#include <gtest/gtest.h>
|
|
|
|
namespace webrtc {
|
|
|
|
class AudioProcessing;
|
|
class AudioFrame;
|
|
|
|
class ApmTest : public ::testing::Test {
|
|
protected:
|
|
ApmTest();
|
|
virtual void SetUp();
|
|
virtual void TearDown();
|
|
|
|
webrtc::AudioProcessing* apm_;
|
|
FILE* far_file_;
|
|
FILE* near_file_;
|
|
FILE* stat_file_;
|
|
AudioFrame* frame_;
|
|
AudioFrame* reverse_frame_;
|
|
};
|
|
} // namespace webrtc
|
|
|
|
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_TEST_UNIT_TEST_UNIT_TEST_H_
|