98bb6640d2
- First audio RTP packet sent / received - First RTP packet of the first video frame sent / received - Last RTP packet of the first video frame sent / received These timestamps should make it easier to measure how fast the call becomes established from the user's perspective. Review URL: https://codereview.webrtc.org/1765443002 Cr-Commit-Position: refs/heads/master@{#12287}
34 lines
864 B
C++
34 lines
864 B
C++
/*
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* Copyright 2016 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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#include "webrtc/base/gunit.h"
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#include "webrtc/base/onetimeevent.h"
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namespace webrtc {
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TEST(OneTimeEventTest, ThreadSafe) {
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OneTimeEvent ot;
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// The one time event is expected to evaluate to true only the first time.
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EXPECT_TRUE(ot());
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EXPECT_FALSE(ot());
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EXPECT_FALSE(ot());
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}
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TEST(OneTimeEventTest, ThreadUnsafe) {
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ThreadUnsafeOneTimeEvent ot;
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EXPECT_TRUE(ot());
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EXPECT_FALSE(ot());
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EXPECT_FALSE(ot());
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}
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} // namespace webrtc
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