1 /*
2 * Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
3 *
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
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
9 */
10
11 #include "modules/rtp_rtcp/source/rtp_utility.h"
12 #include "pc/media_session.h"
13 #include "pc/session_description.h"
14 #include "test/field_trial.h"
15 #include "test/gtest.h"
16 #include "test/peer_scenario/peer_scenario.h"
17
18 namespace webrtc {
19 namespace test {
20 namespace {
AudioExtensions(const SessionDescriptionInterface & session)21 RtpHeaderExtensionMap AudioExtensions(
22 const SessionDescriptionInterface& session) {
23 auto* audio_desc =
24 cricket::GetFirstAudioContentDescription(session.description());
25 return RtpHeaderExtensionMap(audio_desc->rtp_header_extensions());
26 }
27
GetRtpPacketExtensions(const rtc::ArrayView<const uint8_t> packet,const RtpHeaderExtensionMap & extension_map)28 absl::optional<RTPHeaderExtension> GetRtpPacketExtensions(
29 const rtc::ArrayView<const uint8_t> packet,
30 const RtpHeaderExtensionMap& extension_map) {
31 RtpUtility::RtpHeaderParser rtp_parser(packet.data(), packet.size());
32 if (!rtp_parser.RTCP()) {
33 RTPHeader header;
34 if (rtp_parser.Parse(&header, &extension_map, true)) {
35 return header.extension;
36 }
37 }
38 return absl::nullopt;
39 }
40
41 } // namespace
42
TEST(RemoteEstimateEndToEnd,OfferedCapabilityIsInAnswer)43 TEST(RemoteEstimateEndToEnd, OfferedCapabilityIsInAnswer) {
44 PeerScenario s(*test_info_);
45
46 auto* caller = s.CreateClient(PeerScenarioClient::Config());
47 auto* callee = s.CreateClient(PeerScenarioClient::Config());
48
49 auto send_link = {s.net()->NodeBuilder().Build().node};
50 auto ret_link = {s.net()->NodeBuilder().Build().node};
51
52 s.net()->CreateRoute(caller->endpoint(), send_link, callee->endpoint());
53 s.net()->CreateRoute(callee->endpoint(), ret_link, caller->endpoint());
54
55 auto signaling = s.ConnectSignaling(caller, callee, send_link, ret_link);
56 caller->CreateVideo("VIDEO", PeerScenarioClient::VideoSendTrackConfig());
57 std::atomic<bool> offer_exchange_done(false);
58 signaling.NegotiateSdp(
59 [](SessionDescriptionInterface* offer) {
60 for (auto& cont : offer->description()->contents()) {
61 cont.media_description()->set_remote_estimate(true);
62 }
63 },
64 [&](const SessionDescriptionInterface& answer) {
65 for (auto& cont : answer.description()->contents()) {
66 EXPECT_TRUE(cont.media_description()->remote_estimate());
67 }
68 offer_exchange_done = true;
69 });
70 RTC_CHECK(s.WaitAndProcess(&offer_exchange_done));
71 }
72
TEST(RemoteEstimateEndToEnd,AudioUsesAbsSendTimeExtension)73 TEST(RemoteEstimateEndToEnd, AudioUsesAbsSendTimeExtension) {
74 // Defined before PeerScenario so it gets destructed after, to avoid use after free.
75 std::atomic<bool> received_abs_send_time(false);
76 PeerScenario s(*test_info_);
77
78 auto* caller = s.CreateClient(PeerScenarioClient::Config());
79 auto* callee = s.CreateClient(PeerScenarioClient::Config());
80
81 auto send_node = s.net()->NodeBuilder().Build().node;
82 auto ret_node = s.net()->NodeBuilder().Build().node;
83
84 s.net()->CreateRoute(caller->endpoint(), {send_node}, callee->endpoint());
85 s.net()->CreateRoute(callee->endpoint(), {ret_node}, caller->endpoint());
86
87 auto signaling = s.ConnectSignaling(caller, callee, {send_node}, {ret_node});
88 caller->CreateAudio("AUDIO", cricket::AudioOptions());
89 signaling.StartIceSignaling();
90 RtpHeaderExtensionMap extension_map;
91 std::atomic<bool> offer_exchange_done(false);
92 signaling.NegotiateSdp(
93 [&extension_map](SessionDescriptionInterface* offer) {
94 extension_map = AudioExtensions(*offer);
95 EXPECT_TRUE(extension_map.IsRegistered(kRtpExtensionAbsoluteSendTime));
96 },
97 [&](const SessionDescriptionInterface& answer) {
98 EXPECT_TRUE(AudioExtensions(answer).IsRegistered(
99 kRtpExtensionAbsoluteSendTime));
100 offer_exchange_done = true;
101 });
102 RTC_CHECK(s.WaitAndProcess(&offer_exchange_done));
103 send_node->router()->SetWatcher(
104 [extension_map, &received_abs_send_time](const EmulatedIpPacket& packet) {
105 // The dummy packets used by the fake signaling are filled with 0. We
106 // want to ignore those and we can do that on the basis that the first
107 // byte of RTP packets are guaranteed to not be 0.
108 // TODO(srte): Find a more elegant way to check for RTP traffic.
109 if (packet.size() > 1 && packet.cdata()[0] != 0) {
110 auto extensions = GetRtpPacketExtensions(packet.data, extension_map);
111 if (extensions) {
112 EXPECT_TRUE(extensions->hasAbsoluteSendTime);
113 received_abs_send_time = true;
114 }
115 }
116 });
117 RTC_CHECK(s.WaitAndProcess(&received_abs_send_time));
118 }
119 } // namespace test
120 } // namespace webrtc
121