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1 /* | 1 /* |
2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2013 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 "testing/gtest/include/gtest/gtest.h" | 11 #include "testing/gtest/include/gtest/gtest.h" |
12 #include "webrtc/base/checks.h" | 12 #include "webrtc/base/checks.h" |
13 #include "webrtc/base/common.h" | 13 #include "webrtc/base/common.h" |
14 #include "webrtc/base/event.h" | 14 #include "webrtc/base/event.h" |
| 15 #include "webrtc/base/platform_thread.h" |
15 #include "webrtc/modules/pacing/packet_router.h" | 16 #include "webrtc/modules/pacing/packet_router.h" |
16 #include "webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_s
end_time.h" | 17 #include "webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_s
end_time.h" |
17 #include "webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_singl
e_stream.h" | 18 #include "webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_singl
e_stream.h" |
18 #include "webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.h" | 19 #include "webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.h" |
19 #include "webrtc/modules/rtp_rtcp/include/receive_statistics.h" | 20 #include "webrtc/modules/rtp_rtcp/include/receive_statistics.h" |
20 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" | 21 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" |
21 #include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h" | 22 #include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h" |
22 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" | 23 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" |
23 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h" | 24 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h" |
24 #include "webrtc/system_wrappers/include/critical_section_wrapper.h" | 25 #include "webrtc/system_wrappers/include/critical_section_wrapper.h" |
25 #include "webrtc/system_wrappers/include/thread_wrapper.h" | |
26 #include "webrtc/test/testsupport/perf_test.h" | 26 #include "webrtc/test/testsupport/perf_test.h" |
27 #include "webrtc/video/rampup_tests.h" | 27 #include "webrtc/video/rampup_tests.h" |
28 | 28 |
29 namespace webrtc { | 29 namespace webrtc { |
30 namespace { | 30 namespace { |
31 | 31 |
32 static const int64_t kPollIntervalMs = 20; | 32 static const int64_t kPollIntervalMs = 20; |
33 | 33 |
34 std::vector<uint32_t> GenerateSsrcs(size_t num_streams, | 34 std::vector<uint32_t> GenerateSsrcs(size_t num_streams, |
35 uint32_t ssrc_offset) { | 35 uint32_t ssrc_offset) { |
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53 red_(red), | 53 red_(red), |
54 send_stream_(nullptr), | 54 send_stream_(nullptr), |
55 start_bitrate_bps_(start_bitrate_bps), | 55 start_bitrate_bps_(start_bitrate_bps), |
56 start_bitrate_verified_(false), | 56 start_bitrate_verified_(false), |
57 expected_bitrate_bps_(0), | 57 expected_bitrate_bps_(0), |
58 test_start_ms_(-1), | 58 test_start_ms_(-1), |
59 ramp_up_finished_ms_(-1), | 59 ramp_up_finished_ms_(-1), |
60 extension_type_(extension_type), | 60 extension_type_(extension_type), |
61 ssrcs_(GenerateSsrcs(num_streams, 100)), | 61 ssrcs_(GenerateSsrcs(num_streams, 100)), |
62 rtx_ssrcs_(GenerateSsrcs(num_streams, 200)), | 62 rtx_ssrcs_(GenerateSsrcs(num_streams, 200)), |
63 poller_thread_(ThreadWrapper::CreateThread(&BitrateStatsPollingThread, | 63 poller_thread_(PlatformThread::CreateThread(&BitrateStatsPollingThread, |
64 this, | 64 this, |
65 "BitrateStatsPollingThread")), | 65 "BitrateStatsPollingThread")), |
66 sender_call_(nullptr) { | 66 sender_call_(nullptr) { |
67 if (rtx_) { | 67 if (rtx_) { |
68 for (size_t i = 0; i < ssrcs_.size(); ++i) | 68 for (size_t i = 0; i < ssrcs_.size(); ++i) |
69 rtx_ssrc_map_[rtx_ssrcs_[i]] = ssrcs_[i]; | 69 rtx_ssrc_map_[rtx_ssrcs_[i]] = ssrcs_[i]; |
70 } | 70 } |
71 } | 71 } |
72 | 72 |
73 RampUpTester::~RampUpTester() { | 73 RampUpTester::~RampUpTester() { |
74 event_.Set(); | 74 event_.Set(); |
75 } | 75 } |
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503 RampUpTester test(3, 0, RtpExtension::kTransportSequenceNumber, true, true); | 503 RampUpTester test(3, 0, RtpExtension::kTransportSequenceNumber, true, true); |
504 RunBaseTest(&test, FakeNetworkPipe::Config()); | 504 RunBaseTest(&test, FakeNetworkPipe::Config()); |
505 } | 505 } |
506 | 506 |
507 TEST_F(RampUpTest, TransportSequenceNumberSingleStreamWithHighStartBitrate) { | 507 TEST_F(RampUpTest, TransportSequenceNumberSingleStreamWithHighStartBitrate) { |
508 RampUpTester test(1, 0.9 * kSingleStreamTargetBps, | 508 RampUpTester test(1, 0.9 * kSingleStreamTargetBps, |
509 RtpExtension::kTransportSequenceNumber, false, false); | 509 RtpExtension::kTransportSequenceNumber, false, false); |
510 RunBaseTest(&test, FakeNetworkPipe::Config()); | 510 RunBaseTest(&test, FakeNetworkPipe::Config()); |
511 } | 511 } |
512 } // namespace webrtc | 512 } // namespace webrtc |
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