diff options
author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
---|---|---|
committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
commit | 26a029d407be480d791972afb5975cf62c9360a6 (patch) | |
tree | f435a8308119effd964b339f76abb83a57c29483 /third_party/libwebrtc/test/peer_scenario/tests/remote_estimate_test.cc | |
parent | Initial commit. (diff) | |
download | firefox-26a029d407be480d791972afb5975cf62c9360a6.tar.xz firefox-26a029d407be480d791972afb5975cf62c9360a6.zip |
Adding upstream version 124.0.1.upstream/124.0.1
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/test/peer_scenario/tests/remote_estimate_test.cc')
-rw-r--r-- | third_party/libwebrtc/test/peer_scenario/tests/remote_estimate_test.cc | 113 |
1 files changed, 113 insertions, 0 deletions
diff --git a/third_party/libwebrtc/test/peer_scenario/tests/remote_estimate_test.cc b/third_party/libwebrtc/test/peer_scenario/tests/remote_estimate_test.cc new file mode 100644 index 0000000000..fa343a7fdf --- /dev/null +++ b/third_party/libwebrtc/test/peer_scenario/tests/remote_estimate_test.cc @@ -0,0 +1,113 @@ +/* + * 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 = + cricket::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", cricket::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 |