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1 /* | 1 /* |
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2012 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 <algorithm> | 11 #include <algorithm> |
12 #include <vector> | 12 #include <vector> |
13 | 13 |
14 #include "webrtc/test/gtest.h" | |
15 | |
16 #include "webrtc/call/mock/mock_rtc_event_log.h" | 14 #include "webrtc/call/mock/mock_rtc_event_log.h" |
17 #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h" | 15 #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h" |
18 #include "webrtc/modules/pacing/mock/mock_paced_sender.h" | 16 #include "webrtc/modules/pacing/mock/mock_paced_sender.h" |
19 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" | 17 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" |
20 #include "webrtc/test/field_trial.h" | 18 #include "webrtc/test/field_trial.h" |
| 19 #include "webrtc/test/gtest.h" |
21 | 20 |
22 using ::testing::Exactly; | 21 using ::testing::Exactly; |
23 using ::testing::Return; | 22 using ::testing::Return; |
24 | 23 |
25 using webrtc::BitrateController; | 24 using webrtc::BitrateController; |
26 using webrtc::BitrateObserver; | 25 using webrtc::BitrateObserver; |
27 using webrtc::PacedSender; | 26 using webrtc::PacedSender; |
28 using webrtc::RtcpBandwidthObserver; | 27 using webrtc::RtcpBandwidthObserver; |
29 | 28 |
30 uint8_t WeightedLoss(int num_packets1, uint8_t fraction_loss1, | 29 uint8_t WeightedLoss(int num_packets1, uint8_t fraction_loss1, |
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506 clock_.AdvanceTimeMilliseconds(100); | 505 clock_.AdvanceTimeMilliseconds(100); |
507 | 506 |
508 // 15 packets reported as received since last, enough to increase. | 507 // 15 packets reported as received since last, enough to increase. |
509 report_blocks.push_back(CreateReportBlock(1, 2, 0, 41)); | 508 report_blocks.push_back(CreateReportBlock(1, 2, 0, 41)); |
510 bandwidth_observer_->OnReceivedRtcpReceiverReport( | 509 bandwidth_observer_->OnReceivedRtcpReceiverReport( |
511 report_blocks, 50, clock_.TimeInMilliseconds()); | 510 report_blocks, 50, clock_.TimeInMilliseconds()); |
512 expected_bitrate_bps = expected_bitrate_bps * 1.08 + 1000; | 511 expected_bitrate_bps = expected_bitrate_bps * 1.08 + 1000; |
513 EXPECT_EQ(expected_bitrate_bps, bitrate_observer_.last_bitrate_); | 512 EXPECT_EQ(expected_bitrate_bps, bitrate_observer_.last_bitrate_); |
514 clock_.AdvanceTimeMilliseconds(1000); | 513 clock_.AdvanceTimeMilliseconds(1000); |
515 } | 514 } |
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