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https://github.com/danog/libtgvoip.git
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5caaaafa42
I'm now using the entire audio processing module from WebRTC as opposed to individual DSP algorithms pulled from there before. Seems to work better this way.
39 lines
1.5 KiB
C
39 lines
1.5 KiB
C
/*
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* Copyright (c) 2012 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 RTC_BASE_GTEST_PROD_UTIL_H_
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#define RTC_BASE_GTEST_PROD_UTIL_H_
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// Define our own version of FRIEND_TEST here rather than including
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// gtest_prod.h to avoid depending on any part of GTest in production code.
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#define FRIEND_TEST_WEBRTC(test_case_name, test_name) \
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friend class test_case_name##_##test_name##_Test
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// This file is a plain copy of Chromium's base/gtest_prod_util.h.
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//
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// This is a wrapper for gtest's FRIEND_TEST macro that friends
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// test with all possible prefixes. This is very helpful when changing the test
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// prefix, because the friend declarations don't need to be updated.
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//
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// Example usage:
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//
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// class MyClass {
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// private:
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// void MyMethod();
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// FRIEND_TEST_ALL_PREFIXES(MyClassTest, MyMethod);
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// };
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#define FRIEND_TEST_ALL_PREFIXES(test_case_name, test_name) \
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FRIEND_TEST_WEBRTC(test_case_name, test_name); \
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FRIEND_TEST_WEBRTC(test_case_name, DISABLED_##test_name); \
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FRIEND_TEST_WEBRTC(test_case_name, FLAKY_##test_name); \
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FRIEND_TEST_WEBRTC(test_case_name, FAILS_##test_name)
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#endif // RTC_BASE_GTEST_PROD_UTIL_H_
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