| /* |
| * 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 <vector> |
| |
| #include "absl/strings/str_cat.h" |
| #include "api/media_types.h" |
| #include "api/peer_connection_interface.h" |
| #include "api/rtp_parameters.h" |
| #include "api/rtp_transceiver_direction.h" |
| #include "api/test/rtc_error_matchers.h" |
| #include "pc/test/integration_test_helpers.h" |
| #include "test/gmock.h" |
| #include "test/gtest.h" |
| #include "test/wait_until.h" |
| |
| namespace webrtc { |
| |
| using ::testing::Eq; |
| using ::testing::Field; |
| using ::testing::Gt; |
| using ::testing::HasSubstr; |
| using ::testing::IsTrue; |
| using ::testing::Not; |
| |
| class PeerConnectionCongestionControlTest |
| : public PeerConnectionIntegrationBaseTest { |
| public: |
| PeerConnectionCongestionControlTest() |
| : PeerConnectionIntegrationBaseTest(SdpSemantics::kUnifiedPlan) {} |
| }; |
| |
| TEST_F(PeerConnectionCongestionControlTest, OfferContainsCcfbIfEnabled) { |
| SetFieldTrials("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) { |
| SetFieldTrials("WebRTC-RFC8888CongestionControlFeedback/Enabled/"); |
| ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| ConnectFakeSignalingForSdpOnly(); |
| caller()->AddAudioVideoTracks(); |
| caller()->CreateAndSetAndSignalOffer(); |
| ASSERT_THAT(WaitUntil([&] { return SignalingStateStable(); }, IsTrue()), |
| IsRtcOk()); |
| { |
| // 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. |
| auto 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()); |
| } |
| } |
| // Check that the answer does not contain transport-cc |
| std::string answer_str = absl::StrCat(*caller()->pc()->remote_description()); |
| EXPECT_THAT(answer_str, Not(HasSubstr("transport-cc"))); |
| } |
| |
| TEST_F(PeerConnectionCongestionControlTest, NegotiatingCcfbRemovesTsn) { |
| SetFieldTrials("WebRTC-RFC8888CongestionControlFeedback/Enabled/"); |
| ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| ConnectFakeSignalingForSdpOnly(); |
| callee()->AddVideoTrack(); |
| // Add transceivers to caller in order to accomodate reception |
| caller()->pc()->AddTransceiver(MediaType::VIDEO); |
| auto parameters = caller()->pc()->GetSenders()[0]->GetParameters(); |
| caller()->CreateAndSetAndSignalOffer(); |
| ASSERT_THAT(WaitUntil([&] { return SignalingStateStable(); }, IsTrue()), |
| IsRtcOk()); |
| |
| std::vector<RtpHeaderExtensionCapability> negotiated_header_extensions = |
| caller()->pc()->GetTransceivers()[0]->GetNegotiatedHeaderExtensions(); |
| EXPECT_THAT( |
| negotiated_header_extensions, |
| Not(Contains( |
| AllOf(Field("uri", &RtpHeaderExtensionCapability::uri, |
| RtpExtension::kTransportSequenceNumberUri), |
| Not(Field("direction", &RtpHeaderExtensionCapability::direction, |
| RtpTransceiverDirection::kStopped)))))) |
| << " in caller negotiated header extensions"; |
| |
| parameters = caller()->pc()->GetSenders()[0]->GetParameters(); |
| EXPECT_THAT(parameters.header_extensions, |
| Not(Contains(Field("uri", &RtpExtension::uri, |
| RtpExtension::kTransportSequenceNumberUri)))) |
| << " in caller sender parameters"; |
| parameters = caller()->pc()->GetReceivers()[0]->GetParameters(); |
| EXPECT_THAT(parameters.header_extensions, |
| Not(Contains(Field("uri", &RtpExtension::uri, |
| RtpExtension::kTransportSequenceNumberUri)))) |
| << " in caller receiver parameters"; |
| |
| parameters = callee()->pc()->GetSenders()[0]->GetParameters(); |
| EXPECT_THAT(parameters.header_extensions, |
| Not(Contains(Field("uri", &RtpExtension::uri, |
| RtpExtension::kTransportSequenceNumberUri)))) |
| << " in callee sender parameters"; |
| |
| parameters = callee()->pc()->GetReceivers()[0]->GetParameters(); |
| EXPECT_THAT(parameters.header_extensions, |
| Not(Contains(Field("uri", &RtpExtension::uri, |
| RtpExtension::kTransportSequenceNumberUri)))) |
| << " in callee receiver parameters"; |
| } |
| |
| TEST_F(PeerConnectionCongestionControlTest, CcfbGetsUsed) { |
| SetFieldTrials("WebRTC-RFC8888CongestionControlFeedback/Enabled/"); |
| ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| ConnectFakeSignaling(); |
| caller()->AddAudioVideoTracks(); |
| caller()->CreateAndSetAndSignalOffer(); |
| ASSERT_THAT(WaitUntil([&] { return SignalingStateStable(); }, IsTrue()), |
| IsRtcOk()); |
| MediaExpectations media_expectations; |
| media_expectations.CalleeExpectsSomeAudio(); |
| media_expectations.CalleeExpectsSomeVideo(); |
| ASSERT_TRUE(ExpectNewFrames(media_expectations)); |
| auto pc_internal = caller()->pc_internal(); |
| EXPECT_THAT( |
| WaitUntil( |
| [&] { |
| return pc_internal->FeedbackAccordingToRfc8888CountForTesting(); |
| }, |
| Gt(0)), |
| IsRtcOk()); |
| // There should be no transport-cc generated. |
| EXPECT_THAT(pc_internal->FeedbackAccordingToTransportCcCountForTesting(), |
| Eq(0)); |
| } |
| |
| TEST_F(PeerConnectionCongestionControlTest, TransportCcGetsUsed) { |
| SetFieldTrials("WebRTC-RFC8888CongestionControlFeedback/Disabled/"); |
| ASSERT_TRUE(CreatePeerConnectionWrappers()); |
| ConnectFakeSignaling(); |
| caller()->AddAudioVideoTracks(); |
| caller()->CreateAndSetAndSignalOffer(); |
| ASSERT_THAT(WaitUntil([&] { return SignalingStateStable(); }, IsTrue()), |
| IsRtcOk()); |
| MediaExpectations media_expectations; |
| media_expectations.CalleeExpectsSomeAudio(); |
| media_expectations.CalleeExpectsSomeVideo(); |
| ASSERT_TRUE(ExpectNewFrames(media_expectations)); |
| auto pc_internal = caller()->pc_internal(); |
| EXPECT_THAT( |
| WaitUntil( |
| [&] { |
| return pc_internal->FeedbackAccordingToTransportCcCountForTesting(); |
| }, |
| Gt(0)), |
| IsRtcOk()); |
| // Test that RFC 8888 feedback is NOT generated when field trial disabled. |
| EXPECT_THAT(pc_internal->FeedbackAccordingToRfc8888CountForTesting(), Eq(0)); |
| } |
| |
| } // namespace webrtc |