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1 /* | 1 /* |
2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2014 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 * This file includes unit tests for the RtcpPacket. | 10 * This file includes unit tests for the RtcpPacket. |
11 */ | 11 */ |
12 | 12 |
13 #include "testing/gmock/include/gmock/gmock.h" | 13 #include "testing/gmock/include/gmock/gmock.h" |
14 #include "testing/gtest/include/gtest/gtest.h" | 14 #include "testing/gtest/include/gtest/gtest.h" |
15 | 15 |
16 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h" | 16 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h" |
17 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/app.h" | 17 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/app.h" |
18 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h" | 18 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h" |
19 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h" | 19 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h" |
20 #include "webrtc/test/rtcp_packet_parser.h" | 20 #include "webrtc/test/rtcp_packet_parser.h" |
21 | 21 |
22 using ::testing::ElementsAre; | 22 using ::testing::ElementsAre; |
23 | 23 |
24 using webrtc::rtcp::App; | 24 using webrtc::rtcp::App; |
25 using webrtc::rtcp::Bye; | 25 using webrtc::rtcp::Bye; |
26 using webrtc::rtcp::Dlrr; | 26 using webrtc::rtcp::Dlrr; |
27 using webrtc::rtcp::Fir; | 27 using webrtc::rtcp::Fir; |
28 using webrtc::rtcp::RawPacket; | 28 using webrtc::rtcp::RawPacket; |
29 using webrtc::rtcp::ReceiverReport; | 29 using webrtc::rtcp::ReceiverReport; |
30 using webrtc::rtcp::Remb; | |
31 using webrtc::rtcp::ReportBlock; | 30 using webrtc::rtcp::ReportBlock; |
32 using webrtc::rtcp::Rpsi; | 31 using webrtc::rtcp::Rpsi; |
33 using webrtc::rtcp::Rrtr; | 32 using webrtc::rtcp::Rrtr; |
34 using webrtc::rtcp::Sdes; | 33 using webrtc::rtcp::Sdes; |
35 using webrtc::rtcp::SenderReport; | 34 using webrtc::rtcp::SenderReport; |
36 using webrtc::rtcp::VoipMetric; | 35 using webrtc::rtcp::VoipMetric; |
37 using webrtc::rtcp::Xr; | 36 using webrtc::rtcp::Xr; |
38 using webrtc::test::RtcpPacketParser; | 37 using webrtc::test::RtcpPacketParser; |
39 | 38 |
40 namespace webrtc { | 39 namespace webrtc { |
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317 class Verifier : public rtcp::RtcpPacket::PacketReadyCallback { | 316 class Verifier : public rtcp::RtcpPacket::PacketReadyCallback { |
318 void OnPacketReady(uint8_t* data, size_t length) override { | 317 void OnPacketReady(uint8_t* data, size_t length) override { |
319 ADD_FAILURE() << "Packet should not fit within max size."; | 318 ADD_FAILURE() << "Packet should not fit within max size."; |
320 } | 319 } |
321 } verifier; | 320 } verifier; |
322 const size_t kBufferSize = kRrLength + kReportBlockLength - 1; | 321 const size_t kBufferSize = kRrLength + kReportBlockLength - 1; |
323 uint8_t buffer[kBufferSize]; | 322 uint8_t buffer[kBufferSize]; |
324 EXPECT_FALSE(rr.BuildExternalBuffer(buffer, kBufferSize, &verifier)); | 323 EXPECT_FALSE(rr.BuildExternalBuffer(buffer, kBufferSize, &verifier)); |
325 } | 324 } |
326 | 325 |
327 TEST(RtcpPacketTest, Remb) { | |
328 Remb remb; | |
329 remb.From(kSenderSsrc); | |
330 remb.AppliesTo(kRemoteSsrc); | |
331 remb.AppliesTo(kRemoteSsrc + 1); | |
332 remb.AppliesTo(kRemoteSsrc + 2); | |
333 remb.WithBitrateBps(261011); | |
334 | |
335 rtc::scoped_ptr<RawPacket> packet(remb.Build()); | |
336 RtcpPacketParser parser; | |
337 parser.Parse(packet->Buffer(), packet->Length()); | |
338 EXPECT_EQ(1, parser.psfb_app()->num_packets()); | |
339 EXPECT_EQ(kSenderSsrc, parser.psfb_app()->Ssrc()); | |
340 EXPECT_EQ(1, parser.remb_item()->num_packets()); | |
341 EXPECT_EQ(261011, parser.remb_item()->last_bitrate_bps()); | |
342 std::vector<uint32_t> ssrcs = parser.remb_item()->last_ssrc_list(); | |
343 EXPECT_EQ(kRemoteSsrc, ssrcs[0]); | |
344 EXPECT_EQ(kRemoteSsrc + 1, ssrcs[1]); | |
345 EXPECT_EQ(kRemoteSsrc + 2, ssrcs[2]); | |
346 } | |
347 | 326 |
348 TEST(RtcpPacketTest, XrWithNoReportBlocks) { | 327 TEST(RtcpPacketTest, XrWithNoReportBlocks) { |
349 Xr xr; | 328 Xr xr; |
350 xr.From(kSenderSsrc); | 329 xr.From(kSenderSsrc); |
351 | 330 |
352 rtc::scoped_ptr<RawPacket> packet(xr.Build()); | 331 rtc::scoped_ptr<RawPacket> packet(xr.Build()); |
353 RtcpPacketParser parser; | 332 RtcpPacketParser parser; |
354 parser.Parse(packet->Buffer(), packet->Length()); | 333 parser.Parse(packet->Buffer(), packet->Length()); |
355 EXPECT_EQ(1, parser.xr_header()->num_packets()); | 334 EXPECT_EQ(1, parser.xr_header()->num_packets()); |
356 EXPECT_EQ(kSenderSsrc, parser.xr_header()->Ssrc()); | 335 EXPECT_EQ(kSenderSsrc, parser.xr_header()->Ssrc()); |
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582 EXPECT_TRUE(xr.WithDlrr(&dlrr)); | 561 EXPECT_TRUE(xr.WithDlrr(&dlrr)); |
583 EXPECT_FALSE(xr.WithDlrr(&dlrr)); | 562 EXPECT_FALSE(xr.WithDlrr(&dlrr)); |
584 | 563 |
585 VoipMetric voip_metric; | 564 VoipMetric voip_metric; |
586 for (int i = 0; i < kMaxBlocks; ++i) | 565 for (int i = 0; i < kMaxBlocks; ++i) |
587 EXPECT_TRUE(xr.WithVoipMetric(&voip_metric)); | 566 EXPECT_TRUE(xr.WithVoipMetric(&voip_metric)); |
588 EXPECT_FALSE(xr.WithVoipMetric(&voip_metric)); | 567 EXPECT_FALSE(xr.WithVoipMetric(&voip_metric)); |
589 } | 568 } |
590 | 569 |
591 } // namespace webrtc | 570 } // namespace webrtc |
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