f83a94a41e
Reason for revert: There seems to be a TSan warning that wasn't caught by the trybot: https://build.chromium.org/p/client.webrtc/builders/Linux%20Tsan%20v2/builds/6732/steps/peerconnection_unittests/logs/stdio Apparently a thread is still alive even after destroying WebRTCSession. Need to think of a way to fix this, without adding a critical section around g_clock (since that would hurt performance). Original issue's description: > Improving the fake clock and using it to fix a flaky STUN timeout test. > > When the fake clock's time is advanced, it now ensures all pending > queued messages have been dispatched. This allows us to write a > "SIMULATED_WAIT" macro that ticks the simulated clock by milliseconds up > until the target time. > > Useful in this case, where we know the STUN timeout should take a total > of 9500ms, but it would be overly complex to write test code that waits > for each individual timeout, ensures a STUN packet has been > retransmited, etc. > > (The test described above *should* be written, but it belongs in > p2ptransportchannel_unittest.cc, not webrtcsession_unittest.cc). > > Committed: https://crrev.com/ffbe0e17e2c9b7fe101023acf40574dc0c95631a > Cr-Commit-Position: refs/heads/master@{#13043} TBR=pthatcher@webrtc.org,tommi@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true Review-Url: https://codereview.webrtc.org/2038213002 Cr-Commit-Position: refs/heads/master@{#13045}
89 lines
2.9 KiB
C
89 lines
2.9 KiB
C
/*
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* Copyright 2004 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 WEBRTC_BASE_GUNIT_H_
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#define WEBRTC_BASE_GUNIT_H_
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#include "webrtc/base/logging.h"
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#include "webrtc/base/thread.h"
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#if defined(GTEST_RELATIVE_PATH)
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#include "testing/gtest/include/gtest/gtest.h"
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#else
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#include "testing/base/public/gunit.h"
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#endif
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// Wait until "ex" is true, or "timeout" expires.
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#define WAIT(ex, timeout) \
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for (int64_t start = rtc::TimeMillis(); \
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!(ex) && rtc::TimeMillis() < start + timeout;) \
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rtc::Thread::Current()->ProcessMessages(1);
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// This returns the result of the test in res, so that we don't re-evaluate
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// the expression in the XXXX_WAIT macros below, since that causes problems
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// when the expression is only true the first time you check it.
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#define WAIT_(ex, timeout, res) \
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do { \
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int64_t start = rtc::TimeMillis(); \
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res = (ex); \
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while (!res && rtc::TimeMillis() < start + timeout) { \
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rtc::Thread::Current()->ProcessMessages(1); \
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res = (ex); \
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} \
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} while (0)
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// The typical EXPECT_XXXX and ASSERT_XXXXs, but done until true or a timeout.
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#define EXPECT_TRUE_WAIT(ex, timeout) \
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do { \
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bool res; \
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WAIT_(ex, timeout, res); \
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if (!res) EXPECT_TRUE(ex); \
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} while (0)
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#define EXPECT_EQ_WAIT(v1, v2, timeout) \
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do { \
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bool res; \
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WAIT_(v1 == v2, timeout, res); \
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if (!res) EXPECT_EQ(v1, v2); \
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} while (0)
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#define ASSERT_TRUE_WAIT(ex, timeout) \
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do { \
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bool res; \
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WAIT_(ex, timeout, res); \
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if (!res) ASSERT_TRUE(ex); \
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} while (0)
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#define ASSERT_EQ_WAIT(v1, v2, timeout) \
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do { \
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bool res; \
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WAIT_(v1 == v2, timeout, res); \
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if (!res) ASSERT_EQ(v1, v2); \
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} while (0)
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// Version with a "soft" timeout and a margin. This logs if the timeout is
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// exceeded, but it only fails if the expression still isn't true after the
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// margin time passes.
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#define EXPECT_TRUE_WAIT_MARGIN(ex, timeout, margin) \
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do { \
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bool res; \
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WAIT_(ex, timeout, res); \
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if (res) { \
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break; \
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} \
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LOG(LS_WARNING) << "Expression " << #ex << " still not true after " << \
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timeout << "ms; waiting an additional " << margin << "ms"; \
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WAIT_(ex, margin, res); \
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if (!res) { \
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EXPECT_TRUE(ex); \
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} \
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} while (0)
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#endif // WEBRTC_BASE_GUNIT_H_
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