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7220ee97aa
git ls-files | grep -e "\(\.h\|\.cc\)$" | grep -vE "^(rtc_base|sdk|modules|api|call|common_audio|examples|media|net|p2p|pc)/" | xargs clang-format -i ; git cl format after landing: add to .git-blame-ignore-revs Bug: webrtc:15082 Change-Id: I9c7fc4e6fbb023809fb22a89a78be713de6990d3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302063 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39978}
113 lines
4.5 KiB
C++
113 lines
4.5 KiB
C++
/*
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* Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
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#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
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#include "modules/rtp_rtcp/source/rtp_packet.h"
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#include "modules/rtp_rtcp/source/rtp_util.h"
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#include "pc/media_session.h"
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#include "pc/session_description.h"
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#include "test/field_trial.h"
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#include "test/gtest.h"
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#include "test/peer_scenario/peer_scenario.h"
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namespace webrtc {
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namespace test {
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namespace {
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RtpHeaderExtensionMap AudioExtensions(
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const SessionDescriptionInterface& session) {
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auto* audio_desc =
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cricket::GetFirstAudioContentDescription(session.description());
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return RtpHeaderExtensionMap(audio_desc->rtp_header_extensions());
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}
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} // namespace
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TEST(RemoteEstimateEndToEnd, OfferedCapabilityIsInAnswer) {
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PeerScenario s(*test_info_);
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auto* caller = s.CreateClient(PeerScenarioClient::Config());
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auto* callee = s.CreateClient(PeerScenarioClient::Config());
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auto send_link = {s.net()->NodeBuilder().Build().node};
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auto ret_link = {s.net()->NodeBuilder().Build().node};
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s.net()->CreateRoute(caller->endpoint(), send_link, callee->endpoint());
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s.net()->CreateRoute(callee->endpoint(), ret_link, caller->endpoint());
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auto signaling = s.ConnectSignaling(caller, callee, send_link, ret_link);
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caller->CreateVideo("VIDEO", PeerScenarioClient::VideoSendTrackConfig());
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std::atomic<bool> offer_exchange_done(false);
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signaling.NegotiateSdp(
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[](SessionDescriptionInterface* offer) {
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for (auto& cont : offer->description()->contents()) {
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cont.media_description()->set_remote_estimate(true);
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}
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},
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[&](const SessionDescriptionInterface& answer) {
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for (auto& cont : answer.description()->contents()) {
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EXPECT_TRUE(cont.media_description()->remote_estimate());
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}
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offer_exchange_done = true;
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});
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RTC_CHECK(s.WaitAndProcess(&offer_exchange_done));
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}
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TEST(RemoteEstimateEndToEnd, AudioUsesAbsSendTimeExtension) {
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// Defined before PeerScenario so it gets destructed after, to avoid use after
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// free.
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std::atomic<bool> received_abs_send_time(false);
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PeerScenario s(*test_info_);
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auto* caller = s.CreateClient(PeerScenarioClient::Config());
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auto* callee = s.CreateClient(PeerScenarioClient::Config());
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auto send_node = s.net()->NodeBuilder().Build().node;
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auto ret_node = s.net()->NodeBuilder().Build().node;
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s.net()->CreateRoute(caller->endpoint(), {send_node}, callee->endpoint());
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s.net()->CreateRoute(callee->endpoint(), {ret_node}, caller->endpoint());
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auto signaling = s.ConnectSignaling(caller, callee, {send_node}, {ret_node});
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caller->CreateAudio("AUDIO", cricket::AudioOptions());
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signaling.StartIceSignaling();
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RtpHeaderExtensionMap extension_map;
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std::atomic<bool> offer_exchange_done(false);
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signaling.NegotiateSdp(
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[&extension_map](SessionDescriptionInterface* offer) {
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extension_map = AudioExtensions(*offer);
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EXPECT_TRUE(extension_map.IsRegistered(kRtpExtensionAbsoluteSendTime));
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},
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[&](const SessionDescriptionInterface& answer) {
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EXPECT_TRUE(AudioExtensions(answer).IsRegistered(
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kRtpExtensionAbsoluteSendTime));
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offer_exchange_done = true;
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});
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RTC_CHECK(s.WaitAndProcess(&offer_exchange_done));
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send_node->router()->SetWatcher(
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[extension_map, &received_abs_send_time](const EmulatedIpPacket& packet) {
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// The dummy packets used by the fake signaling are filled with 0. We
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// want to ignore those and we can do that on the basis that the first
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// byte of RTP packets are guaranteed to not be 0.
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RtpPacket rtp_packet(&extension_map);
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// TODO(bugs.webrtc.org/14525): Look why there are RTP packets with
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// payload 72 or 73 (these don't have the RTP AbsoluteSendTime
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// Extension).
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if (rtp_packet.Parse(packet.data) && rtp_packet.PayloadType() == 111) {
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EXPECT_TRUE(rtp_packet.HasExtension<AbsoluteSendTime>());
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received_abs_send_time = true;
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}
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});
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RTC_CHECK(s.WaitAndProcess(&received_abs_send_time));
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caller->pc()->Close();
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callee->pc()->Close();
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}
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} // namespace test
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} // namespace webrtc
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