1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
|
/*
* Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
#include "modules/rtp_rtcp/source/rtp_packet.h"
#include "modules/rtp_rtcp/source/rtp_util.h"
#include "pc/media_session.h"
#include "pc/session_description.h"
#include "test/field_trial.h"
#include "test/gtest.h"
#include "test/peer_scenario/peer_scenario.h"
namespace webrtc {
namespace test {
namespace {
RtpHeaderExtensionMap AudioExtensions(
const SessionDescriptionInterface& session) {
auto* audio_desc = GetFirstAudioContentDescription(session.description());
return RtpHeaderExtensionMap(audio_desc->rtp_header_extensions());
}
} // namespace
TEST(RemoteEstimateEndToEnd, OfferedCapabilityIsInAnswer) {
PeerScenario s(*test_info_);
auto* caller = s.CreateClient(PeerScenarioClient::Config());
auto* callee = s.CreateClient(PeerScenarioClient::Config());
auto send_link = {s.net()->NodeBuilder().Build().node};
auto ret_link = {s.net()->NodeBuilder().Build().node};
s.net()->CreateRoute(caller->endpoint(), send_link, callee->endpoint());
s.net()->CreateRoute(callee->endpoint(), ret_link, caller->endpoint());
auto signaling = s.ConnectSignaling(caller, callee, send_link, ret_link);
caller->CreateVideo("VIDEO", PeerScenarioClient::VideoSendTrackConfig());
std::atomic<bool> offer_exchange_done(false);
signaling.NegotiateSdp(
[](SessionDescriptionInterface* offer) {
for (auto& cont : offer->description()->contents()) {
cont.media_description()->set_remote_estimate(true);
}
},
[&](const SessionDescriptionInterface& answer) {
for (auto& cont : answer.description()->contents()) {
EXPECT_TRUE(cont.media_description()->remote_estimate());
}
offer_exchange_done = true;
});
RTC_CHECK(s.WaitAndProcess(&offer_exchange_done));
}
TEST(RemoteEstimateEndToEnd, AudioUsesAbsSendTimeExtension) {
// Defined before PeerScenario so it gets destructed after, to avoid use after
// free.
std::atomic<bool> received_abs_send_time(false);
PeerScenario s(*test_info_);
auto* caller = s.CreateClient(PeerScenarioClient::Config());
auto* callee = s.CreateClient(PeerScenarioClient::Config());
auto send_node = s.net()->NodeBuilder().Build().node;
auto ret_node = s.net()->NodeBuilder().Build().node;
s.net()->CreateRoute(caller->endpoint(), {send_node}, callee->endpoint());
s.net()->CreateRoute(callee->endpoint(), {ret_node}, caller->endpoint());
auto signaling = s.ConnectSignaling(caller, callee, {send_node}, {ret_node});
caller->CreateAudio("AUDIO", AudioOptions());
signaling.StartIceSignaling();
RtpHeaderExtensionMap extension_map;
std::atomic<bool> offer_exchange_done(false);
signaling.NegotiateSdp(
[&extension_map](SessionDescriptionInterface* offer) {
extension_map = AudioExtensions(*offer);
EXPECT_TRUE(extension_map.IsRegistered(kRtpExtensionAbsoluteSendTime));
},
[&](const SessionDescriptionInterface& answer) {
EXPECT_TRUE(AudioExtensions(answer).IsRegistered(
kRtpExtensionAbsoluteSendTime));
offer_exchange_done = true;
});
RTC_CHECK(s.WaitAndProcess(&offer_exchange_done));
send_node->router()->SetWatcher(
[extension_map, &received_abs_send_time](const EmulatedIpPacket& packet) {
// The dummy packets used by the fake signaling are filled with 0. We
// want to ignore those and we can do that on the basis that the first
// byte of RTP packets are guaranteed to not be 0.
RtpPacket rtp_packet(&extension_map);
// TODO(bugs.webrtc.org/14525): Look why there are RTP packets with
// payload 72 or 73 (these don't have the RTP AbsoluteSendTime
// Extension).
if (rtp_packet.Parse(packet.data) && rtp_packet.PayloadType() == 111) {
EXPECT_TRUE(rtp_packet.HasExtension<AbsoluteSendTime>());
received_abs_send_time = true;
}
});
RTC_CHECK(s.WaitAndProcess(&received_abs_send_time));
caller->pc()->Close();
callee->pc()->Close();
}
} // namespace test
} // namespace webrtc
|
