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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 <memory> | 11 #include <memory> |
12 #include <vector> | 12 #include <vector> |
13 | 13 |
14 #include "webrtc/base/buffer.h" | 14 #include "webrtc/base/buffer.h" |
15 #include "webrtc/base/rate_limiter.h" | 15 #include "webrtc/base/rate_limiter.h" |
16 #include "webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.h" | 16 #include "webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.h" |
17 #include "webrtc/modules/rtp_rtcp/include/rtp_cvo.h" | 17 #include "webrtc/modules/rtp_rtcp/include/rtp_cvo.h" |
18 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" | 18 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" |
19 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" | 19 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" |
20 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h" | 20 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h" |
21 #include "webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h" | 21 #include "webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h" |
22 #include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h" | 22 #include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h" |
23 #include "webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h" | 23 #include "webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h" |
24 #include "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h" | 24 #include "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h" |
25 #include "webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h" | 25 #include "webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h" |
26 #include "webrtc/modules/rtp_rtcp/source/rtp_sender.h" | 26 #include "webrtc/modules/rtp_rtcp/source/rtp_sender.h" |
27 #include "webrtc/modules/rtp_rtcp/source/rtp_sender_video.h" | 27 #include "webrtc/modules/rtp_rtcp/source/rtp_sender_video.h" |
28 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" | 28 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" |
29 #include "webrtc/system_wrappers/include/stl_util.h" | 29 #include "webrtc/system_wrappers/include/stl_util.h" |
30 #include "webrtc/test/field_trial.h" | |
30 #include "webrtc/test/gmock.h" | 31 #include "webrtc/test/gmock.h" |
31 #include "webrtc/test/gtest.h" | 32 #include "webrtc/test/gtest.h" |
32 #include "webrtc/test/mock_transport.h" | 33 #include "webrtc/test/mock_transport.h" |
33 #include "webrtc/typedefs.h" | 34 #include "webrtc/typedefs.h" |
34 | 35 |
35 namespace webrtc { | 36 namespace webrtc { |
36 | 37 |
37 namespace { | 38 namespace { |
38 const int kTransmissionTimeOffsetExtensionId = 1; | 39 const int kTransmissionTimeOffsetExtensionId = 1; |
39 const int kAbsoluteSendTimeExtensionId = 14; | 40 const int kAbsoluteSendTimeExtensionId = 14; |
(...skipping 1483 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... | |
1523 | 1524 |
1524 EXPECT_CALL(mock_overhead_observer, OnOverheadChanged(_)).Times(1); | 1525 EXPECT_CALL(mock_overhead_observer, OnOverheadChanged(_)).Times(1); |
1525 rtp_sender_->SetTransportOverhead(28); | 1526 rtp_sender_->SetTransportOverhead(28); |
1526 rtp_sender_->SetTransportOverhead(28); | 1527 rtp_sender_->SetTransportOverhead(28); |
1527 | 1528 |
1528 EXPECT_CALL(mock_overhead_observer, OnOverheadChanged(_)).Times(1); | 1529 EXPECT_CALL(mock_overhead_observer, OnOverheadChanged(_)).Times(1); |
1529 SendGenericPayload(); | 1530 SendGenericPayload(); |
1530 SendGenericPayload(); | 1531 SendGenericPayload(); |
1531 } | 1532 } |
1532 | 1533 |
1534 TEST_F(RtpSenderTest, AddOverheadToTransportFeedbackObserver) { | |
1535 constexpr int kTransportOverheadBytesPerPacket = 28; | |
1536 constexpr int kRtpOverheadBytesPerPacket = 12 + 8; | |
1537 test::ScopedFieldTrials override_field_trials( | |
1538 "WebRTC-SendSideBwe-WithOverhead/Enabled/"); | |
1539 testing::NiceMock<MockOverheadObserver> mock_overhead_observer; | |
1540 rtp_sender_.reset(new RTPSender( | |
1541 false, &fake_clock_, &transport_, nullptr, nullptr, &seq_num_allocator_, | |
1542 &feedback_observer_, nullptr, nullptr, nullptr, &mock_rtc_event_log_, | |
1543 nullptr, &retransmission_rate_limiter_, &mock_overhead_observer)); | |
1544 rtp_sender_->SetTransportOverhead(kTransportOverheadBytesPerPacket); | |
1545 EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension( | |
1546 kRtpExtensionTransportSequenceNumber, | |
1547 kTransportSequenceNumberExtensionId)); | |
1548 EXPECT_CALL(seq_num_allocator_, AllocateSequenceNumber()) | |
1549 .WillOnce(testing::Return(kTransportSequenceNumber)); | |
1550 EXPECT_CALL(feedback_observer_, | |
1551 AddPacket(kTransportSequenceNumber, | |
1552 sizeof(kPayloadData) + kGenericHeaderLength + | |
1553 kRtpOverheadBytesPerPacket + | |
1554 kTransportOverheadBytesPerPacket, | |
1555 PacketInfo::kNotAProbe)) | |
1556 .Times(1); | |
minyue-webrtc
2016/11/29 09:24:35
I am not a fun of Times(1)
According to this
http
michaelt
2016/11/29 09:55:53
If some one don't know this rule its may be a bit
minyue-webrtc
2016/11/29 09:59:49
Ok. I am not very strong on not using Times(1) eit
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1557 EXPECT_CALL(mock_overhead_observer, | |
1558 OnOverheadChanged(kTransportOverheadBytesPerPacket + | |
1559 kRtpOverheadBytesPerPacket)) | |
1560 .Times(1); | |
1561 SendGenericPayload(); | |
1562 } | |
1563 | |
1533 } // namespace webrtc | 1564 } // namespace webrtc |
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