| /* |
| * Copyright 2024 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. |
| */ |
| |
| // This file contains tests that verify that congestion control options |
| // are correctly negotiated in the SDP offer/answer. |
| |
| #include <string> |
| |
| #include "absl/strings/str_cat.h" |
| #include "api/peer_connection_interface.h" |
| #include "pc/test/integration_test_helpers.h" |
| #include "rtc_base/gunit.h" |
| #include "test/field_trial.h" |
| #include "test/gmock.h" |
| #include "test/gtest.h" |
| |
| namespace webrtc { |
| |
| using testing::HasSubstr; |
| |
| class PeerConnectionCongestionControlTest |
| : public PeerConnectionIntegrationBaseTest { |
| public: |
| PeerConnectionCongestionControlTest() |
| : PeerConnectionIntegrationBaseTest(SdpSemantics::kUnifiedPlan) {} |
| }; |
| |
| TEST_F(PeerConnectionCongestionControlTest, OfferContainsCcfbIfEnabled) { |
| test::ScopedFieldTrials trials( |
| "WebRTC-RFC8888CongestionControlFeedback/Enabled/"); |
| ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| caller()->AddAudioVideoTracks(); |
| auto offer = caller()->CreateOfferAndWait(); |
| std::string offer_str = absl::StrCat(*offer); |
| EXPECT_THAT(offer_str, HasSubstr("a=rtcp-fb:* ack ccfb\r\n")); |
| } |
| |
| TEST_F(PeerConnectionCongestionControlTest, ReceiveOfferSetsCcfbFlag) { |
| test::ScopedFieldTrials trials( |
| "WebRTC-RFC8888CongestionControlFeedback/Enabled/"); |
| ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| ConnectFakeSignalingForSdpOnly(); |
| caller()->AddAudioVideoTracks(); |
| caller()->CreateAndSetAndSignalOffer(); |
| ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| // Check that the callee parsed it. |
| auto parsed_contents = |
| callee()->pc()->remote_description()->description()->contents(); |
| EXPECT_FALSE(parsed_contents.empty()); |
| for (const auto& content : parsed_contents) { |
| EXPECT_TRUE(content.media_description()->rtcp_fb_ack_ccfb()); |
| } |
| // Check that the caller also parsed it. |
| parsed_contents = |
| caller()->pc()->remote_description()->description()->contents(); |
| EXPECT_FALSE(parsed_contents.empty()); |
| for (const auto& content : parsed_contents) { |
| EXPECT_TRUE(content.media_description()->rtcp_fb_ack_ccfb()); |
| } |
| } |
| |
| TEST_F(PeerConnectionCongestionControlTest, CcfbGetsUsed) { |
| test::ScopedFieldTrials trials( |
| "WebRTC-RFC8888CongestionControlFeedback/Enabled/"); |
| ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| ConnectFakeSignaling(); |
| caller()->AddAudioVideoTracks(); |
| caller()->CreateAndSetAndSignalOffer(); |
| ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
| MediaExpectations media_expectations; |
| media_expectations.CalleeExpectsSomeAudio(); |
| media_expectations.CalleeExpectsSomeVideo(); |
| ASSERT_TRUE(ExpectNewFrames(media_expectations)); |
| auto pc_internal = caller()->pc_internal(); |
| EXPECT_TRUE_WAIT(pc_internal->FeedbackAccordingToRfc8888CountForTesting() > 0, |
| kDefaultTimeout); |
| } |
| |
| } // namespace webrtc |