blob: 998853293c4fd30d2a39b610a7f9a7acf412522d [file] [log] [blame]
/*
* 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/source/rtp_utility.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 =
cricket::GetFirstAudioContentDescription(session.description());
return RtpHeaderExtensionMap(audio_desc->rtp_header_extensions());
}
absl::optional<RTPHeaderExtension> GetRtpPacketExtensions(
const rtc::ArrayView<const uint8_t> packet,
const RtpHeaderExtensionMap& extension_map) {
RtpUtility::RtpHeaderParser rtp_parser(packet.data(), packet.size());
if (!rtp_parser.RTCP()) {
RTPHeader header;
if (rtp_parser.Parse(&header, &extension_map, true)) {
return header.extension;
}
}
return absl::nullopt;
}
} // 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());
rtc::Event offer_exchange_done;
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.Set();
});
RTC_CHECK(s.WaitAndProcess(&offer_exchange_done));
}
TEST(RemoteEstimateEndToEnd, AudioUsesAbsSendTimeExtension) {
ScopedFieldTrials trials("WebRTC-KeepAbsSendTimeExtension/Enabled/");
// Defined before PeerScenario so it gets destructed after, to avoid use after free.
rtc::Event received_abs_send_time;
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", cricket::AudioOptions());
signaling.StartIceSignaling();
RtpHeaderExtensionMap extension_map;
rtc::Event offer_exchange_done;
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.Set();
});
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.
// TODO(srte): Find a more elegant way to check for RTP traffic.
if (packet.size() > 1 && packet.cdata()[0] != 0) {
auto extensions = GetRtpPacketExtensions(packet.data, extension_map);
if (extensions) {
EXPECT_TRUE(extensions->hasAbsoluteSendTime);
received_abs_send_time.Set();
}
}
});
RTC_CHECK(s.WaitAndProcess(&received_abs_send_time));
}
} // namespace test
} // namespace webrtc