| OLD | NEW |
| 1 /* | 1 /* |
| 2 * Copyright 2016 The WebRTC project authors. All Rights Reserved. | 2 * Copyright 2016 The WebRTC project authors. All Rights Reserved. |
| 3 * | 3 * |
| 4 * Use of this source code is governed by a BSD-style license | 4 * Use of this source code is governed by a BSD-style license |
| 5 * that can be found in the LICENSE file in the root of the source | 5 * that can be found in the LICENSE file in the root of the source |
| 6 * tree. An additional intellectual property rights grant can be found | 6 * tree. An additional intellectual property rights grant can be found |
| 7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
| 8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
| 9 */ | 9 */ |
| 10 | 10 |
| 11 #include "webrtc/stats/rtcstatscollector.h" | 11 #include "webrtc/stats/rtcstatscollector.h" |
| 12 | 12 |
| 13 #include <memory> | 13 #include <memory> |
| 14 #include <string> | 14 #include <string> |
| 15 #include <vector> | 15 #include <vector> |
| 16 | 16 |
| 17 #include "webrtc/api/jsepsessiondescription.h" | 17 #include "webrtc/api/jsepsessiondescription.h" |
| 18 #include "webrtc/api/rtcstats_objects.h" | 18 #include "webrtc/api/rtcstats_objects.h" |
| 19 #include "webrtc/api/rtcstatsreport.h" | 19 #include "webrtc/api/rtcstatsreport.h" |
| 20 #include "webrtc/api/test/mock_datachannel.h" | 20 #include "webrtc/api/test/mock_datachannel.h" |
| 21 #include "webrtc/api/test/mock_peerconnection.h" | 21 #include "webrtc/api/test/mock_peerconnection.h" |
| 22 #include "webrtc/api/test/mock_webrtcsession.h" | 22 #include "webrtc/api/test/mock_webrtcsession.h" |
| 23 #include "webrtc/base/checks.h" | 23 #include "webrtc/base/checks.h" |
| 24 #include "webrtc/base/fakeclock.h" | 24 #include "webrtc/base/fakeclock.h" |
| 25 #include "webrtc/base/gunit.h" | 25 #include "webrtc/base/gunit.h" |
| 26 #include "webrtc/base/logging.h" | 26 #include "webrtc/base/logging.h" |
| 27 #include "webrtc/base/timedelta.h" | 27 #include "webrtc/base/timedelta.h" |
| 28 #include "webrtc/base/timeutils.h" | 28 #include "webrtc/base/timeutils.h" |
| 29 #include "webrtc/base/timing.h" | |
| 30 #include "webrtc/media/base/fakemediaengine.h" | 29 #include "webrtc/media/base/fakemediaengine.h" |
| 31 | 30 |
| 32 using testing::Return; | 31 using testing::Return; |
| 33 using testing::ReturnRef; | 32 using testing::ReturnRef; |
| 34 | 33 |
| 35 namespace webrtc { | 34 namespace webrtc { |
| 36 | 35 |
| 37 class RTCStatsCollectorTester : public SetSessionDescriptionObserver { | 36 class RTCStatsCollectorTester : public SetSessionDescriptionObserver { |
| 38 public: | 37 public: |
| 39 RTCStatsCollectorTester() | 38 RTCStatsCollectorTester() |
| (...skipping 62 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
| 102 EXPECT_TRUE(c); | 101 EXPECT_TRUE(c); |
| 103 EXPECT_NE(b.get(), c.get()); | 102 EXPECT_NE(b.get(), c.get()); |
| 104 // Invalidate cache by advancing time. | 103 // Invalidate cache by advancing time. |
| 105 fake_clock.AdvanceTime(rtc::TimeDelta::FromMilliseconds(51)); | 104 fake_clock.AdvanceTime(rtc::TimeDelta::FromMilliseconds(51)); |
| 106 rtc::scoped_refptr<const RTCStatsReport> d = collector_.GetStatsReport(); | 105 rtc::scoped_refptr<const RTCStatsReport> d = collector_.GetStatsReport(); |
| 107 EXPECT_TRUE(d); | 106 EXPECT_TRUE(d); |
| 108 EXPECT_NE(c.get(), d.get()); | 107 EXPECT_NE(c.get(), d.get()); |
| 109 } | 108 } |
| 110 | 109 |
| 111 TEST_F(RTCStatsCollectorTest, CollectRTCPeerConnectionStats) { | 110 TEST_F(RTCStatsCollectorTest, CollectRTCPeerConnectionStats) { |
| 112 int64_t before = static_cast<int64_t>( | 111 int64_t before = rtc::TimeUTCMicros(); |
| 113 rtc::Timing::WallTimeNow() * rtc::kNumMicrosecsPerSec); | |
| 114 rtc::scoped_refptr<const RTCStatsReport> report = collector_.GetStatsReport(); | 112 rtc::scoped_refptr<const RTCStatsReport> report = collector_.GetStatsReport(); |
| 115 int64_t after = static_cast<int64_t>( | 113 int64_t after = rtc::TimeUTCMicros(); |
| 116 rtc::Timing::WallTimeNow() * rtc::kNumMicrosecsPerSec); | |
| 117 EXPECT_EQ(report->GetStatsOfType<RTCPeerConnectionStats>().size(), | 114 EXPECT_EQ(report->GetStatsOfType<RTCPeerConnectionStats>().size(), |
| 118 static_cast<size_t>(1)) << "Expecting 1 RTCPeerConnectionStats."; | 115 static_cast<size_t>(1)) << "Expecting 1 RTCPeerConnectionStats."; |
| 119 const RTCStats* stats = report->Get("RTCPeerConnection"); | 116 const RTCStats* stats = report->Get("RTCPeerConnection"); |
| 120 EXPECT_TRUE(stats); | 117 EXPECT_TRUE(stats); |
| 121 EXPECT_LE(before, stats->timestamp_us()); | 118 EXPECT_LE(before, stats->timestamp_us()); |
| 122 EXPECT_LE(stats->timestamp_us(), after); | 119 EXPECT_LE(stats->timestamp_us(), after); |
| 123 { | 120 { |
| 124 // Expected stats with no data channels | 121 // Expected stats with no data channels |
| 125 const RTCPeerConnectionStats& pcstats = | 122 const RTCPeerConnectionStats& pcstats = |
| 126 stats->cast_to<RTCPeerConnectionStats>(); | 123 stats->cast_to<RTCPeerConnectionStats>(); |
| (...skipping 23 matching lines...) Expand all Loading... |
| 150 // open/closed, we would expect opened >= closed and (opened - closed) to be | 147 // open/closed, we would expect opened >= closed and (opened - closed) to be |
| 151 // the number currently open. crbug.com/636818. | 148 // the number currently open. crbug.com/636818. |
| 152 const RTCPeerConnectionStats& pcstats = | 149 const RTCPeerConnectionStats& pcstats = |
| 153 stats->cast_to<RTCPeerConnectionStats>(); | 150 stats->cast_to<RTCPeerConnectionStats>(); |
| 154 EXPECT_EQ(*pcstats.data_channels_opened, static_cast<uint32_t>(1)); | 151 EXPECT_EQ(*pcstats.data_channels_opened, static_cast<uint32_t>(1)); |
| 155 EXPECT_EQ(*pcstats.data_channels_closed, static_cast<uint32_t>(3)); | 152 EXPECT_EQ(*pcstats.data_channels_closed, static_cast<uint32_t>(3)); |
| 156 } | 153 } |
| 157 } | 154 } |
| 158 | 155 |
| 159 } // namespace webrtc | 156 } // namespace webrtc |
| OLD | NEW |