diff options
Diffstat (limited to 'test/peer_scenario/tests/unsignaled_stream_test.cc')
| -rw-r--r-- | test/peer_scenario/tests/unsignaled_stream_test.cc | 148 |
1 files changed, 132 insertions, 16 deletions
diff --git a/test/peer_scenario/tests/unsignaled_stream_test.cc b/test/peer_scenario/tests/unsignaled_stream_test.cc index 95510a24bd..edcfb36ea3 100644 --- a/test/peer_scenario/tests/unsignaled_stream_test.cc +++ b/test/peer_scenario/tests/unsignaled_stream_test.cc @@ -24,6 +24,31 @@ namespace webrtc { namespace test { namespace { +enum class MidTestConfiguration { + // Legacy endpoint setup where PT demuxing is used. + kMidNotNegotiated, + // MID is negotiated but missing from packets. PT demuxing is disabled, so + // SSRCs have to be added to the SDP for WebRTC to forward packets correctly. + // Happens when client is spec compliant but the SFU isn't. Popular legacy. + kMidNegotiatedButMissingFromPackets, + // Fully spec-compliant: MID is present so we can safely drop packets with + // unknown MIDs. + kMidNegotiatedAndPresentInPackets, +}; + +// Gives the parameterized test a readable suffix. +std::string TestParametersMidTestConfigurationToString( + testing::TestParamInfo<MidTestConfiguration> info) { + switch (info.param) { + case MidTestConfiguration::kMidNotNegotiated: + return "MidNotNegotiated"; + case MidTestConfiguration::kMidNegotiatedButMissingFromPackets: + return "MidNegotiatedButMissingFromPackets"; + case MidTestConfiguration::kMidNegotiatedAndPresentInPackets: + return "MidNegotiatedAndPresentInPackets"; + } +} + class FrameObserver : public rtc::VideoSinkInterface<VideoFrame> { public: FrameObserver() : frame_observed_(false) {} @@ -53,19 +78,24 @@ void set_ssrc(SessionDescriptionInterface* offer, size_t index, uint32_t ssrc) { } // namespace -TEST(UnsignaledStreamTest, ReplacesUnsignaledStreamOnCompletedSignaling) { +class UnsignaledStreamTest + : public ::testing::Test, + public ::testing::WithParamInterface<MidTestConfiguration> {}; + +TEST_P(UnsignaledStreamTest, ReplacesUnsignaledStreamOnCompletedSignaling) { // This test covers a scenario that might occur if a remote client starts - // sending media packets before negotiation has completed. These packets will - // trigger an unsignalled default stream to be created, and connects that to - // a default video sink. - // In some edge cases using unified plan, the default stream is create in a - // different transceiver to where the media SSRC will actually be used. - // This test verifies that the default stream is removed properly, and that - // packets are demuxed and video frames reach the desired sink. + // sending media packets before negotiation has completed. Depending on setup, + // these packets either get dropped or trigger an unsignalled default stream + // to be created, and connects that to a default video sink. + // In some edge cases using Unified Plan and PT demuxing, the default stream + // is create in a different transceiver to where the media SSRC will actually + // be used. This test verifies that the default stream is removed properly, + // and that packets are demuxed and video frames reach the desired sink. + const MidTestConfiguration kMidTestConfiguration = GetParam(); // Defined before PeerScenario so it gets destructed after, to avoid use after // free. - PeerScenario s(*test_info_); + PeerScenario s(*::testing::UnitTest::GetInstance()->current_test_info()); PeerScenarioClient::Config config = PeerScenarioClient::Config(); // Disable encryption so that we can inject a fake early media packet without @@ -93,14 +123,61 @@ TEST(UnsignaledStreamTest, ReplacesUnsignaledStreamOnCompletedSignaling) { std::atomic<bool> got_unsignaled_packet(false); // We will capture the media ssrc of the first added stream, and preemptively - // inject a new media packet using a different ssrc. - // This will create "default stream" for the second ssrc and connected it to - // the default video sink (not set in this test). + // inject a new media packet using a different ssrc. What happens depends on + // the test configuration. + // + // MidTestConfiguration::kMidNotNegotiated: + // - MID is not negotiated which means PT-based demuxing is enabled. Because + // the packets have no MID, the second ssrc packet gets forwarded to the + // first m= section. This will create a "default stream" for the second ssrc + // and connect it to the default video sink (not set in this test). The test + // verifies we can recover from this when we later get packets for the first + // ssrc. + // + // MidTestConfiguration::kMidNegotiatedButMissingFromPackets: + // - MID is negotiated wich means PT-based demuxing is disabled. Because we + // modify the packets not to contain the MID anyway (simulating a legacy SFU + // that does not negotiate properly) unknown SSRCs are dropped but do not + // otherwise cause any issues. + // + // MidTestConfiguration::kMidNegotiatedAndPresentInPackets: + // - MID is negotiated which means PT-based demuxing is enabled. In this case + // the packets have the MID so they either get forwarded or dropped + // depending on if the MID is known. The spec-compliant way is also the most + // straight-forward one. + uint32_t first_ssrc = 0; uint32_t second_ssrc = 0; + absl::optional<int> mid_header_extension_id = absl::nullopt; signaling.NegotiateSdp( - /* munge_sdp = */ {}, + /* munge_sdp = */ + [&](SessionDescriptionInterface* offer) { + // Obtain the MID header extension ID and if we want the + // MidTestConfiguration::kMidNotNegotiated setup then we remove the MID + // header extension through SDP munging (otherwise SDP is not modified). + for (cricket::ContentInfo& content_info : + offer->description()->contents()) { + std::vector<RtpExtension> header_extensions = + content_info.media_description()->rtp_header_extensions(); + for (auto it = header_extensions.begin(); + it != header_extensions.end(); ++it) { + if (it->uri == RtpExtension::kMidUri) { + // MID header extension found! + mid_header_extension_id = it->id; + if (kMidTestConfiguration == + MidTestConfiguration::kMidNotNegotiated) { + // Munge away the extension. + header_extensions.erase(it); + } + break; + } + } + content_info.media_description()->set_rtp_header_extensions( + std::move(header_extensions)); + } + ASSERT_TRUE(mid_header_extension_id.has_value()); + }, /* modify_sdp = */ [&](SessionDescriptionInterface* offer) { first_ssrc = get_ssrc(offer, 0); @@ -113,9 +190,40 @@ TEST(UnsignaledStreamTest, ReplacesUnsignaledStreamOnCompletedSignaling) { if (ByteReader<uint32_t>::ReadBigEndian(&(packet.cdata()[8])) == first_ssrc && !got_unsignaled_packet) { - rtc::CopyOnWriteBuffer updated_buffer = packet.data; - ByteWriter<uint32_t>::WriteBigEndian( - updated_buffer.MutableData() + 8, second_ssrc); + // Parse packet and modify the SSRC to simulate a second m= + // section that has not been negotiated yet. + std::vector<RtpExtension> extensions; + extensions.emplace_back(RtpExtension::kMidUri, + mid_header_extension_id.value()); + RtpHeaderExtensionMap extensions_map(extensions); + RtpPacket parsed_packet; + parsed_packet.IdentifyExtensions(extensions_map); + ASSERT_TRUE(parsed_packet.Parse(packet.data)); + parsed_packet.SetSsrc(second_ssrc); + // The MID extension is present if and only if it was negotiated. + // If present, we either want to remove it or modify it depending + // on setup. + switch (kMidTestConfiguration) { + case MidTestConfiguration::kMidNotNegotiated: + EXPECT_FALSE(parsed_packet.HasExtension<RtpMid>()); + break; + case MidTestConfiguration::kMidNegotiatedButMissingFromPackets: + EXPECT_TRUE(parsed_packet.HasExtension<RtpMid>()); + ASSERT_TRUE(parsed_packet.RemoveExtension(RtpMid::kId)); + break; + case MidTestConfiguration::kMidNegotiatedAndPresentInPackets: + EXPECT_TRUE(parsed_packet.HasExtension<RtpMid>()); + // The simulated second m= section would have a different MID. + // If we don't modify it here then |second_ssrc| would end up + // being mapped to the first m= section which would cause SSRC + // conflicts if we later add the same SSRC to a second m= + // section. Hidden assumption: first m= section does not use + // MID:1. + ASSERT_TRUE(parsed_packet.SetExtension<RtpMid>("1")); + break; + } + // Inject the modified packet. + rtc::CopyOnWriteBuffer updated_buffer = parsed_packet.Buffer(); EmulatedIpPacket updated_packet( packet.from, packet.to, updated_buffer, packet.arrival_time); send_node->OnPacketReceived(std::move(updated_packet)); @@ -153,5 +261,13 @@ TEST(UnsignaledStreamTest, ReplacesUnsignaledStreamOnCompletedSignaling) { EXPECT_TRUE(s.WaitAndProcess(&second_sink.frame_observed_)); } +INSTANTIATE_TEST_SUITE_P( + All, + UnsignaledStreamTest, + ::testing::Values(MidTestConfiguration::kMidNotNegotiated, + MidTestConfiguration::kMidNegotiatedButMissingFromPackets, + MidTestConfiguration::kMidNegotiatedAndPresentInPackets), + TestParametersMidTestConfigurationToString); + } // namespace test } // namespace webrtc |