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Attachment 359795 Details for
Bug 193693
: Resync libwebrtc with latest M72 branch
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[patch]
Patch
bug-193693-20190122155816.patch (text/plain), 62.37 KB, created by
youenn fablet
on 2019-01-22 15:58:17 PST
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Description:
Patch
Filename:
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Creator:
youenn fablet
Created:
2019-01-22 15:58:17 PST
Size:
62.37 KB
patch
obsolete
>Subversion Revision: 240159 >diff --git a/Source/ThirdParty/libwebrtc/ChangeLog b/Source/ThirdParty/libwebrtc/ChangeLog >index 98cb6212e187656f7b46b176ce63f5796f082079..fcc0b65ff3bbb8eae4bd06f515c303d40e99d725 100644 >--- a/Source/ThirdParty/libwebrtc/ChangeLog >+++ b/Source/ThirdParty/libwebrtc/ChangeLog >@@ -1,3 +1,35 @@ >+2019-01-22 Youenn Fablet <youenn@apple.com> >+ >+ Resync libwebrtc with latest M72 branch >+ https://bugs.webkit.org/show_bug.cgi?id=193693 >+ >+ Reviewed by NOBODY (OOPS!). >+ >+ Update libwebrtc up to latest M72 branch to fix some identified issues: >+ - Bad bandwidth estimation in case of multiple transceivers >+ - mid handling for legacy endpoints >+ - msid handling for updating mediastreams accordingly. >+ >+ * Source/webrtc/modules/congestion_controller/goog_cc/delay_based_bwe.cc: >+ * Source/webrtc/modules/congestion_controller/goog_cc/delay_based_bwe.h: >+ * Source/webrtc/modules/congestion_controller/goog_cc/goog_cc_network_control.cc: >+ * Source/webrtc/modules/congestion_controller/goog_cc/goog_cc_network_control_unittest.cc: >+ * Source/webrtc/modules/congestion_controller/send_side_congestion_controller_unittest.cc: >+ * Source/webrtc/pc/jsepsessiondescription_unittest.cc: >+ * Source/webrtc/pc/mediasession.cc: >+ * Source/webrtc/pc/mediasession_unittest.cc: >+ * Source/webrtc/pc/peerconnection.cc: >+ * Source/webrtc/pc/peerconnection.h: >+ * Source/webrtc/pc/peerconnection_jsep_unittest.cc: >+ * Source/webrtc/pc/peerconnection_media_unittest.cc: >+ * Source/webrtc/pc/peerconnection_rtp_unittest.cc: >+ * Source/webrtc/pc/sessiondescription.cc: >+ * Source/webrtc/pc/sessiondescription.h: >+ * Source/webrtc/pc/webrtcsdp.cc: >+ * Source/webrtc/pc/webrtcsdp_unittest.cc: >+ * Source/webrtc/system_wrappers/include/metrics.h: >+ * Source/webrtc/video/BUILD.gn: >+ > 2019-01-17 Truitt Savell <tsavell@apple.com> > > Unreviewed, rolling out r240124. >diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/delay_based_bwe.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/delay_based_bwe.cc >index 4894c9863b6836113c5e13bbac39d4892ce709e9..a76adb01c6997eca6db00f7b3306b604045e9413 100644 >--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/delay_based_bwe.cc >+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/delay_based_bwe.cc >@@ -46,8 +46,6 @@ constexpr size_t kDefaultTrendlineWindowSize = 20; > constexpr double kDefaultTrendlineSmoothingCoeff = 0.9; > constexpr double kDefaultTrendlineThresholdGain = 4.0; > >-constexpr int kMaxConsecutiveFailedLookups = 5; >- > const char kBweWindowSizeInPacketsExperiment[] = > "WebRTC-BweWindowSizeInPackets"; > >@@ -94,7 +92,6 @@ DelayBasedBwe::DelayBasedBwe(RtcEventLog* event_log) > : kDefaultTrendlineWindowSize), > trendline_smoothing_coeff_(kDefaultTrendlineSmoothingCoeff), > trendline_threshold_gain_(kDefaultTrendlineThresholdGain), >- consecutive_delayed_feedbacks_(0), > prev_bitrate_(DataRate::Zero()), > prev_state_(BandwidthUsage::kBwNormal) { > RTC_LOG(LS_INFO) >@@ -147,43 +144,12 @@ DelayBasedBwe::Result DelayBasedBwe::IncomingPacketFeedbackVector( > } > > if (delayed_feedback) { >- Timestamp arrival_time = Timestamp::PlusInfinity(); >- if (packet_feedback_vector.back().arrival_time_ms > 0) >- arrival_time = >- Timestamp::ms(packet_feedback_vector.back().arrival_time_ms); >- return OnDelayedFeedback(arrival_time); >- >- } else { >- consecutive_delayed_feedbacks_ = 0; >- return MaybeUpdateEstimate(acked_bitrate, probe_bitrate, >- recovered_from_overuse, at_time); >- } >- return Result(); >-} >- >-DelayBasedBwe::Result DelayBasedBwe::OnDelayedFeedback(Timestamp receive_time) { >- ++consecutive_delayed_feedbacks_; >- if (consecutive_delayed_feedbacks_ >= kMaxConsecutiveFailedLookups) { >- consecutive_delayed_feedbacks_ = 0; >- return OnLongFeedbackDelay(receive_time); >+ // TODO(bugs.webrtc.org/10125): Design a better mechanism to safe-guard >+ // against building very large network queues. >+ return Result(); > } >- return Result(); >-} >- >-DelayBasedBwe::Result DelayBasedBwe::OnLongFeedbackDelay( >- Timestamp arrival_time) { >- // Estimate should always be valid since a start bitrate always is set in the >- // Call constructor. An alternative would be to return an empty Result here, >- // or to estimate the throughput based on the feedback we received. >- RTC_DCHECK(rate_control_.ValidEstimate()); >- rate_control_.SetEstimate(rate_control_.LatestEstimate() / 2, arrival_time); >- Result result; >- result.updated = true; >- result.probe = false; >- result.target_bitrate = rate_control_.LatestEstimate(); >- RTC_LOG(LS_WARNING) << "Long feedback delay detected, reducing BWE to " >- << ToString(result.target_bitrate); >- return result; >+ return MaybeUpdateEstimate(acked_bitrate, probe_bitrate, >+ recovered_from_overuse, at_time); > } > > void DelayBasedBwe::IncomingPacketFeedback( >diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/delay_based_bwe.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/delay_based_bwe.h >index 616c5b0896bdf2fe03490b748ea0ce3419e8dac6..ead60f3012a399b674dea27c9c22103221533b14 100644 >--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/delay_based_bwe.h >+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/delay_based_bwe.h >@@ -49,7 +49,6 @@ class DelayBasedBwe { > absl::optional<DataRate> acked_bitrate, > absl::optional<DataRate> probe_bitrate, > Timestamp at_time); >- Result OnDelayedFeedback(Timestamp receive_time); > void OnRttUpdate(TimeDelta avg_rtt); > bool LatestEstimate(std::vector<uint32_t>* ssrcs, DataRate* bitrate) const; > void SetStartBitrate(DataRate start_bitrate); >@@ -60,7 +59,6 @@ class DelayBasedBwe { > friend class GoogCcStatePrinter; > void IncomingPacketFeedback(const PacketFeedback& packet_feedback, > Timestamp at_time); >- Result OnLongFeedbackDelay(Timestamp arrival_time); > Result MaybeUpdateEstimate(absl::optional<DataRate> acked_bitrate, > absl::optional<DataRate> probe_bitrate, > bool request_probe, >@@ -81,7 +79,6 @@ class DelayBasedBwe { > size_t trendline_window_size_; > double trendline_smoothing_coeff_; > double trendline_threshold_gain_; >- int consecutive_delayed_feedbacks_; > DataRate prev_bitrate_; > BandwidthUsage prev_state_; > >diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/goog_cc_network_control.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/goog_cc_network_control.cc >index 4376953b7cc1cf4935c22ab2a87de90efc0e9e36..982dd05d47c039233440e1d6e849285efee2f533 100644 >--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/goog_cc_network_control.cc >+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/goog_cc_network_control.cc >@@ -394,15 +394,9 @@ NetworkControlUpdate GoogCcNetworkController::OnTransportLossReport( > NetworkControlUpdate GoogCcNetworkController::OnTransportPacketsFeedback( > TransportPacketsFeedback report) { > if (report.packet_feedbacks.empty()) { >- DelayBasedBwe::Result result = delay_based_bwe_->OnDelayedFeedback( >- report.sendless_arrival_times.back()); >- NetworkControlUpdate update; >- if (result.updated) { >- bandwidth_estimation_->UpdateDelayBasedEstimate(report.feedback_time, >- result.target_bitrate); >- MaybeTriggerOnNetworkChanged(&update, report.feedback_time); >- } >- return update; >+ // TODO(bugs.webrtc.org/10125): Design a better mechanism to safe-guard >+ // against building very large network queues. >+ return NetworkControlUpdate(); > } > > if (congestion_window_pushback_controller_) { >diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/goog_cc_network_control_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/goog_cc_network_control_unittest.cc >index fecd1b32d464bcab04aaf37acbf0e49b40a13abc..ee2a6c22a7b6b2a9f00a30356a499efd8ad13c5e 100644 >--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/goog_cc_network_control_unittest.cc >+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/goog_cc_network_control_unittest.cc >@@ -193,74 +193,6 @@ TEST_F(GoogCcNetworkControllerTest, ProbeOnRouteChange) { > update = controller_->OnProcessInterval(DefaultInterval()); > } > >-// Estimated bitrate reduced when the feedbacks arrive with such a long delay, >-// that the send-time-history no longer holds the feedbacked packets. >-TEST_F(GoogCcNetworkControllerTest, LongFeedbackDelays) { >- TargetBitrateTrackingSetup(); >- const webrtc::PacedPacketInfo kPacingInfo0(0, 5, 2000); >- const webrtc::PacedPacketInfo kPacingInfo1(1, 8, 4000); >- const int64_t kFeedbackTimeoutMs = 60001; >- const int kMaxConsecutiveFailedLookups = 5; >- for (int i = 0; i < kMaxConsecutiveFailedLookups; ++i) { >- std::vector<PacketResult> packets; >- packets.push_back(CreateResult(i * 100, 2 * i * 100, 1500, kPacingInfo0)); >- packets.push_back( >- CreateResult(i * 100 + 10, 2 * i * 100 + 10, 1500, kPacingInfo0)); >- packets.push_back( >- CreateResult(i * 100 + 20, 2 * i * 100 + 20, 1500, kPacingInfo0)); >- packets.push_back( >- CreateResult(i * 100 + 30, 2 * i * 100 + 30, 1500, kPacingInfo1)); >- packets.push_back( >- CreateResult(i * 100 + 40, 2 * i * 100 + 40, 1500, kPacingInfo1)); >- >- TransportPacketsFeedback feedback; >- for (PacketResult& packet : packets) { >- controller_->OnSentPacket(packet.sent_packet); >- feedback.sendless_arrival_times.push_back(packet.receive_time); >- } >- >- feedback.feedback_time = packets[0].receive_time; >- AdvanceTimeMilliseconds(kFeedbackTimeoutMs); >- SentPacket later_packet; >- later_packet.send_time = Timestamp::ms(kFeedbackTimeoutMs + i * 200 + 40); >- later_packet.size = DataSize::bytes(1500); >- later_packet.pacing_info = kPacingInfo1; >- controller_->OnSentPacket(later_packet); >- >- OnUpdate(controller_->OnTransportPacketsFeedback(feedback)); >- } >- OnUpdate(controller_->OnProcessInterval(DefaultInterval())); >- >- EXPECT_EQ(kInitialBitrateKbps / 2, target_bitrate_->kbps()); >- >- // Test with feedback that isn't late enough to time out. >- { >- std::vector<PacketResult> packets; >- packets.push_back(CreateResult(100, 200, 1500, kPacingInfo0)); >- packets.push_back(CreateResult(110, 210, 1500, kPacingInfo0)); >- packets.push_back(CreateResult(120, 220, 1500, kPacingInfo0)); >- packets.push_back(CreateResult(130, 230, 1500, kPacingInfo1)); >- packets.push_back(CreateResult(140, 240, 1500, kPacingInfo1)); >- >- for (const PacketResult& packet : packets) >- controller_->OnSentPacket(packet.sent_packet); >- >- TransportPacketsFeedback feedback; >- feedback.feedback_time = packets[0].receive_time; >- feedback.packet_feedbacks = packets; >- >- AdvanceTimeMilliseconds(kFeedbackTimeoutMs - 1); >- >- SentPacket later_packet; >- later_packet.send_time = Timestamp::ms(kFeedbackTimeoutMs + 240); >- later_packet.size = DataSize::bytes(1500); >- later_packet.pacing_info = kPacingInfo1; >- controller_->OnSentPacket(later_packet); >- >- OnUpdate(controller_->OnTransportPacketsFeedback(feedback)); >- } >-} >- > // Bandwidth estimation is updated when feedbacks are received. > // Feedbacks which show an increasing delay cause the estimation to be reduced. > TEST_F(GoogCcNetworkControllerTest, UpdatesDelayBasedEstimate) { >diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/send_side_congestion_controller_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/send_side_congestion_controller_unittest.cc >index 5f58f21cdf8946c4ac8fa885454606c4b7563ef9..0f37e0eb3cedb5c995bf58677539981074e026b7 100644 >--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/send_side_congestion_controller_unittest.cc >+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/send_side_congestion_controller_unittest.cc >@@ -344,97 +344,6 @@ TEST_F(LegacySendSideCongestionControllerTest, ProbeOnRouteChange) { > 20 * kInitialBitrateBps); > } > >-// Estimated bitrate reduced when the feedbacks arrive with such a long delay, >-// that the send-time-history no longer holds the feedbacked packets. >-TEST_F(LegacySendSideCongestionControllerTest, LongFeedbackDelays) { >- TargetBitrateTrackingSetup(); >- >- const int64_t kFeedbackTimeoutMs = 60001; >- const int kMaxConsecutiveFailedLookups = 5; >- for (int i = 0; i < kMaxConsecutiveFailedLookups; ++i) { >- std::vector<PacketFeedback> packets; >- packets.push_back( >- PacketFeedback(i * 100, 2 * i * 100, 0, 1500, kPacingInfo0)); >- packets.push_back( >- PacketFeedback(i * 100 + 10, 2 * i * 100 + 10, 1, 1500, kPacingInfo0)); >- packets.push_back( >- PacketFeedback(i * 100 + 20, 2 * i * 100 + 20, 2, 1500, kPacingInfo0)); >- packets.push_back( >- PacketFeedback(i * 100 + 30, 2 * i * 100 + 30, 3, 1500, kPacingInfo1)); >- packets.push_back( >- PacketFeedback(i * 100 + 40, 2 * i * 100 + 40, 4, 1500, kPacingInfo1)); >- >- for (const PacketFeedback& packet : packets) >- OnSentPacket(packet); >- >- rtcp::TransportFeedback feedback; >- feedback.SetBase(packets[0].sequence_number, >- packets[0].arrival_time_ms * 1000); >- >- for (const PacketFeedback& packet : packets) { >- EXPECT_TRUE(feedback.AddReceivedPacket(packet.sequence_number, >- packet.arrival_time_ms * 1000)); >- } >- >- feedback.Build(); >- >- clock_.AdvanceTimeMilliseconds(kFeedbackTimeoutMs); >- PacketFeedback later_packet(kFeedbackTimeoutMs + i * 100 + 40, >- kFeedbackTimeoutMs + i * 200 + 40, 5, 1500, >- kPacingInfo1); >- OnSentPacket(later_packet); >- >- controller_->OnTransportFeedback(feedback); >- >- // Check that packets have timed out. >- for (PacketFeedback& packet : packets) { >- packet.send_time_ms = PacketFeedback::kNoSendTime; >- packet.payload_size = 0; >- packet.pacing_info = PacedPacketInfo(); >- } >- ComparePacketFeedbackVectors(packets, >- controller_->GetTransportFeedbackVector()); >- } >- >- controller_->Process(); >- >- EXPECT_EQ(kInitialBitrateBps / 2, target_bitrate_bps_); >- >- // Test with feedback that isn't late enough to time out. >- { >- std::vector<PacketFeedback> packets; >- packets.push_back(PacketFeedback(100, 200, 0, 1500, kPacingInfo0)); >- packets.push_back(PacketFeedback(110, 210, 1, 1500, kPacingInfo0)); >- packets.push_back(PacketFeedback(120, 220, 2, 1500, kPacingInfo0)); >- packets.push_back(PacketFeedback(130, 230, 3, 1500, kPacingInfo1)); >- packets.push_back(PacketFeedback(140, 240, 4, 1500, kPacingInfo1)); >- >- for (const PacketFeedback& packet : packets) >- OnSentPacket(packet); >- >- rtcp::TransportFeedback feedback; >- feedback.SetBase(packets[0].sequence_number, >- packets[0].arrival_time_ms * 1000); >- >- for (const PacketFeedback& packet : packets) { >- EXPECT_TRUE(feedback.AddReceivedPacket(packet.sequence_number, >- packet.arrival_time_ms * 1000)); >- } >- >- feedback.Build(); >- >- clock_.AdvanceTimeMilliseconds(kFeedbackTimeoutMs - 1); >- PacketFeedback later_packet(kFeedbackTimeoutMs + 140, >- kFeedbackTimeoutMs + 240, 5, 1500, >- kPacingInfo1); >- OnSentPacket(later_packet); >- >- controller_->OnTransportFeedback(feedback); >- ComparePacketFeedbackVectors(packets, >- controller_->GetTransportFeedbackVector()); >- } >-} >- > // Bandwidth estimation is updated when feedbacks are received. > // Feedbacks which show an increasing delay cause the estimation to be reduced. > TEST_F(LegacySendSideCongestionControllerTest, UpdatesDelayBasedEstimate) { >diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jsepsessiondescription_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jsepsessiondescription_unittest.cc >index d9e2588a55179f42a650c3c4d9031523e338e228..1c823265c37ec706a487c8225ac4c8634dcb81dd 100644 >--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jsepsessiondescription_unittest.cc >+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jsepsessiondescription_unittest.cc >@@ -60,16 +60,16 @@ static cricket::SessionDescription* CreateCricketSessionDescription() { > video->AddCodec(cricket::VideoCodec(120, "VP8")); > desc->AddContent(cricket::CN_VIDEO, MediaProtocolType::kRtp, video.release()); > >- EXPECT_TRUE(desc->AddTransportInfo(cricket::TransportInfo( >+ desc->AddTransportInfo(cricket::TransportInfo( > cricket::CN_AUDIO, > cricket::TransportDescription( > std::vector<std::string>(), kCandidateUfragVoice, kCandidatePwdVoice, >- cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_NONE, NULL)))); >- EXPECT_TRUE(desc->AddTransportInfo(cricket::TransportInfo( >+ cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_NONE, NULL))); >+ desc->AddTransportInfo(cricket::TransportInfo( > cricket::CN_VIDEO, > cricket::TransportDescription( > std::vector<std::string>(), kCandidateUfragVideo, kCandidatePwdVideo, >- cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_NONE, NULL)))); >+ cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_NONE, NULL))); > return desc; > } > >diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/mediasession.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/mediasession.cc >index 263f3693db57b95b95c9d1e23e68db14c3f9a8d9..cce7d22ddbf857db5e8508a5637d041a472c3b5b 100644 >--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/mediasession.cc >+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/mediasession.cc >@@ -1799,14 +1799,12 @@ bool MediaSessionDescriptionFactory::AddTransportOffer( > std::unique_ptr<TransportDescription> new_tdesc( > transport_desc_factory_->CreateOffer(transport_options, current_tdesc, > ice_credentials)); >- bool ret = >- (new_tdesc.get() != NULL && >- offer_desc->AddTransportInfo(TransportInfo(content_name, *new_tdesc))); >- if (!ret) { >+ if (!new_tdesc) { > RTC_LOG(LS_ERROR) << "Failed to AddTransportOffer, content name=" > << content_name; > } >- return ret; >+ offer_desc->AddTransportInfo(TransportInfo(content_name, *new_tdesc)); >+ return true; > } > > TransportDescription* MediaSessionDescriptionFactory::CreateTransportAnswer( >@@ -1831,12 +1829,7 @@ bool MediaSessionDescriptionFactory::AddTransportAnswer( > const std::string& content_name, > const TransportDescription& transport_desc, > SessionDescription* answer_desc) const { >- if (!answer_desc->AddTransportInfo( >- TransportInfo(content_name, transport_desc))) { >- RTC_LOG(LS_ERROR) << "Failed to AddTransportAnswer, content name=" >- << content_name; >- return false; >- } >+ answer_desc->AddTransportInfo(TransportInfo(content_name, transport_desc)); > return true; > } > >diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/mediasession_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/mediasession_unittest.cc >index 2098ec081943e8e4c26ee704497c9bd36f410bf5..fbc403539f7dd53a770f8159228160b76c64d2b2 100644 >--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/mediasession_unittest.cc >+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/mediasession_unittest.cc >@@ -426,14 +426,14 @@ class MediaSessionDescriptionFactoryTest : public testing::Test { > std::unique_ptr<SessionDescription> desc; > if (has_current_desc) { > current_desc.reset(new SessionDescription()); >- EXPECT_TRUE(current_desc->AddTransportInfo(TransportInfo( >+ current_desc->AddTransportInfo(TransportInfo( > "audio", >- TransportDescription(current_audio_ufrag, current_audio_pwd)))); >- EXPECT_TRUE(current_desc->AddTransportInfo(TransportInfo( >+ TransportDescription(current_audio_ufrag, current_audio_pwd))); >+ current_desc->AddTransportInfo(TransportInfo( > "video", >- TransportDescription(current_video_ufrag, current_video_pwd)))); >- EXPECT_TRUE(current_desc->AddTransportInfo(TransportInfo( >- "data", TransportDescription(current_data_ufrag, current_data_pwd)))); >+ TransportDescription(current_video_ufrag, current_video_pwd))); >+ current_desc->AddTransportInfo(TransportInfo( >+ "data", TransportDescription(current_data_ufrag, current_data_pwd))); > } > if (offer) { > desc.reset(f1_.CreateOffer(options, current_desc.get())); >diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection.cc >index ee8a909e03e86fdfc30b239ccb540123a14a3588..1307ef6eac1ddfbafffe9e69e78bb6432321673e 100644 >--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection.cc >+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection.cc >@@ -649,6 +649,25 @@ DataMessageType ToWebrtcDataMessageType(cricket::DataMessageType type) { > return DataMessageType::kControl; > } > >+// Find a new MID that is not already in |used_mids|, then add it to |used_mids| >+// and return a reference to it. >+// Generated MIDs should be no more than 3 bytes long to take up less space in >+// the RTP packet. >+const std::string& AllocateMid(std::set<std::string>* used_mids) { >+ RTC_DCHECK(used_mids); >+ // We're boring: just generate MIDs 0, 1, 2, ... >+ size_t i = 0; >+ std::set<std::string>::iterator it; >+ bool inserted; >+ do { >+ std::string mid = rtc::ToString(i++); >+ auto insert_result = used_mids->insert(mid); >+ it = insert_result.first; >+ inserted = insert_result.second; >+ } while (!inserted); >+ return *it; >+} >+ > } // namespace > > // Upon completion, posts a task to execute the callback of the >@@ -2231,6 +2250,56 @@ RTCError PeerConnection::ApplyLocalDescription( > return RTCError::OK(); > } > >+// The SDP parser used to populate these values by default for the 'content >+// name' if an a=mid line was absent. >+static absl::string_view GetDefaultMidForPlanB(cricket::MediaType media_type) { >+ switch (media_type) { >+ case cricket::MEDIA_TYPE_AUDIO: >+ return cricket::CN_AUDIO; >+ case cricket::MEDIA_TYPE_VIDEO: >+ return cricket::CN_VIDEO; >+ case cricket::MEDIA_TYPE_DATA: >+ return cricket::CN_DATA; >+ } >+ RTC_NOTREACHED(); >+ return ""; >+} >+ >+void PeerConnection::FillInMissingRemoteMids( >+ cricket::SessionDescription* remote_description) { >+ RTC_DCHECK(remote_description); >+ const cricket::ContentInfos& local_contents = >+ (local_description() ? local_description()->description()->contents() >+ : cricket::ContentInfos()); >+ std::set<std::string> used_mids = seen_mids_; >+ for (size_t i = 0; i < remote_description->contents().size(); ++i) { >+ cricket::ContentInfo& content = remote_description->contents()[i]; >+ if (!content.name.empty()) { >+ continue; >+ } >+ absl::string_view new_mid; >+ absl::string_view source_explanation; >+ if (IsUnifiedPlan()) { >+ if (i < local_contents.size()) { >+ new_mid = local_contents[i].name; >+ source_explanation = "from the matching local media section"; >+ } else { >+ new_mid = AllocateMid(&used_mids); >+ source_explanation = "generated just now"; >+ } >+ } else { >+ new_mid = GetDefaultMidForPlanB(content.media_description()->type()); >+ source_explanation = "to match pre-existing behavior"; >+ } >+ content.name = std::string(new_mid); >+ remote_description->transport_infos()[i].content_name = >+ std::string(new_mid); >+ RTC_LOG(LS_INFO) << "SetRemoteDescription: Remote media section at i=" << i >+ << " is missing an a=mid line. Filling in the value '" >+ << new_mid << "' " << source_explanation << "."; >+ } >+} >+ > void PeerConnection::SetRemoteDescription( > SetSessionDescriptionObserver* observer, > SessionDescriptionInterface* desc) { >@@ -2271,6 +2340,10 @@ void PeerConnection::SetRemoteDescription( > ReportSdpFormatReceived(*desc); > } > >+ // Handle remote descriptions missing a=mid lines for interop with legacy end >+ // points. >+ FillInMissingRemoteMids(desc->description()); >+ > RTCError error = ValidateSessionDescription(desc.get(), cricket::CS_REMOTE); > if (!error.ok()) { > std::string error_message = GetSetDescriptionErrorMessage( >@@ -2456,51 +2529,32 @@ RTCError PeerConnection::ApplyRemoteDescription( > const MediaContentDescription* media_desc = content->media_description(); > RtpTransceiverDirection local_direction = > RtpTransceiverDirectionReversed(media_desc->direction()); >- // From the WebRTC specification, steps 2.2.8.5/6 of section 4.4.1.6 "Set >- // the RTCSessionDescription: If direction is sendrecv or recvonly, and >- // transceiver's current direction is neither sendrecv nor recvonly, >- // process the addition of a remote track for the media description. >- std::vector<std::string> stream_ids; >- if (!media_desc->streams().empty()) { >- // The remote description has signaled the stream IDs. >- stream_ids = media_desc->streams()[0].stream_ids(); >- } >- if (RtpTransceiverDirectionHasRecv(local_direction) && >- (!transceiver->fired_direction() || >- !RtpTransceiverDirectionHasRecv(*transceiver->fired_direction()))) { >- RTC_LOG(LS_INFO) << "Processing the addition of a new track for MID=" >- << content->name << " (added to " >- << GetStreamIdsString(stream_ids) << ")."; >- >- std::vector<rtc::scoped_refptr<MediaStreamInterface>> media_streams; >- for (const std::string& stream_id : stream_ids) { >- rtc::scoped_refptr<MediaStreamInterface> stream = >- remote_streams_->find(stream_id); >- if (!stream) { >- stream = MediaStreamProxy::Create(rtc::Thread::Current(), >- MediaStream::Create(stream_id)); >- remote_streams_->AddStream(stream); >- added_streams.push_back(stream); >- } >- media_streams.push_back(stream); >+ // Roughly the same as steps 2.2.8.6 of section 4.4.1.6 "Set the >+ // RTCSessionDescription: Set the associated remote streams given >+ // transceiver.[[Receiver]], msids, addList, and removeList". >+ // https://w3c.github.io/webrtc-pc/#set-the-rtcsessiondescription >+ if (RtpTransceiverDirectionHasRecv(local_direction)) { >+ std::vector<std::string> stream_ids; >+ if (!media_desc->streams().empty()) { >+ // The remote description has signaled the stream IDs. >+ stream_ids = media_desc->streams()[0].stream_ids(); > } >- // Special case: "a=msid" missing, use random stream ID. >- if (media_streams.empty() && >- !(remote_description()->description()->msid_signaling() & >- cricket::kMsidSignalingMediaSection)) { >- if (!missing_msid_default_stream_) { >- missing_msid_default_stream_ = MediaStreamProxy::Create( >- rtc::Thread::Current(), >- MediaStream::Create(rtc::CreateRandomUuid())); >- added_streams.push_back(missing_msid_default_stream_); >- } >- media_streams.push_back(missing_msid_default_stream_); >+ RTC_LOG(LS_INFO) << "Processing the MSIDs for MID=" << content->name >+ << " (" << GetStreamIdsString(stream_ids) << ")."; >+ SetAssociatedRemoteStreams(transceiver->internal()->receiver_internal(), >+ stream_ids, &added_streams, >+ &removed_streams); >+ // From the WebRTC specification, steps 2.2.8.5/6 of section 4.4.1.6 >+ // "Set the RTCSessionDescription: If direction is sendrecv or recvonly, >+ // and transceiver's current direction is neither sendrecv nor recvonly, >+ // process the addition of a remote track for the media description. >+ if (!transceiver->fired_direction() || >+ !RtpTransceiverDirectionHasRecv(*transceiver->fired_direction())) { >+ RTC_LOG(LS_INFO) >+ << "Processing the addition of a remote track for MID=" >+ << content->name << "."; >+ now_receiving_transceivers.push_back(transceiver); > } >- // This will add the remote track to the streams. >- // TODO(hbos): When we remove remote_streams(), use set_stream_ids() >- // instead. https://crbug.com/webrtc/9480 >- transceiver->internal()->receiver_internal()->SetStreams(media_streams); >- now_receiving_transceivers.push_back(transceiver); > } > // 2.2.8.1.7: If direction is "sendonly" or "inactive", and transceiver's > // [[FiredDirection]] slot is either "sendrecv" or "recvonly", process the >@@ -2643,6 +2697,46 @@ RTCError PeerConnection::ApplyRemoteDescription( > return RTCError::OK(); > } > >+void PeerConnection::SetAssociatedRemoteStreams( >+ rtc::scoped_refptr<RtpReceiverInternal> receiver, >+ const std::vector<std::string>& stream_ids, >+ std::vector<rtc::scoped_refptr<MediaStreamInterface>>* added_streams, >+ std::vector<rtc::scoped_refptr<MediaStreamInterface>>* removed_streams) { >+ std::vector<rtc::scoped_refptr<MediaStreamInterface>> media_streams; >+ for (const std::string& stream_id : stream_ids) { >+ rtc::scoped_refptr<MediaStreamInterface> stream = >+ remote_streams_->find(stream_id); >+ if (!stream) { >+ stream = MediaStreamProxy::Create(rtc::Thread::Current(), >+ MediaStream::Create(stream_id)); >+ remote_streams_->AddStream(stream); >+ added_streams->push_back(stream); >+ } >+ media_streams.push_back(stream); >+ } >+ // Special case: "a=msid" missing, use random stream ID. >+ if (media_streams.empty() && >+ !(remote_description()->description()->msid_signaling() & >+ cricket::kMsidSignalingMediaSection)) { >+ if (!missing_msid_default_stream_) { >+ missing_msid_default_stream_ = MediaStreamProxy::Create( >+ rtc::Thread::Current(), MediaStream::Create(rtc::CreateRandomUuid())); >+ added_streams->push_back(missing_msid_default_stream_); >+ } >+ media_streams.push_back(missing_msid_default_stream_); >+ } >+ std::vector<rtc::scoped_refptr<MediaStreamInterface>> previous_streams = >+ receiver->streams(); >+ // SetStreams() will add/remove the receiver's track to/from the streams. This >+ // differs from the spec - the spec uses an "addList" and "removeList" to >+ // update the stream-track relationships in a later step. We do this earlier, >+ // changing the order of things, but the end-result is the same. >+ // TODO(hbos): When we remove remote_streams(), use set_stream_ids() >+ // instead. https://crbug.com/webrtc/9480 >+ receiver->SetStreams(media_streams); >+ RemoveRemoteStreamsIfEmpty(previous_streams, removed_streams); >+} >+ > void PeerConnection::ProcessRemovalOfRemoteTrack( > rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>> > transceiver, >@@ -2651,19 +2745,25 @@ void PeerConnection::ProcessRemovalOfRemoteTrack( > RTC_DCHECK(transceiver->mid()); > RTC_LOG(LS_INFO) << "Processing the removal of a track for MID=" > << *transceiver->mid(); >- std::vector<rtc::scoped_refptr<MediaStreamInterface>> media_streams = >+ std::vector<rtc::scoped_refptr<MediaStreamInterface>> previous_streams = > transceiver->internal()->receiver_internal()->streams(); > // This will remove the remote track from the streams. > transceiver->internal()->receiver_internal()->set_stream_ids({}); > remove_list->push_back(transceiver); >- // Remove any streams that no longer have tracks. >- // TODO(https://crbug.com/webrtc/9480): When we use stream IDs instead >- // of streams, see if the stream was removed by checking if this was the >- // last receiver with that stream ID. >- for (const auto& stream : media_streams) { >- if (stream->GetAudioTracks().empty() && stream->GetVideoTracks().empty()) { >- remote_streams_->RemoveStream(stream); >- removed_streams->push_back(stream); >+ RemoveRemoteStreamsIfEmpty(previous_streams, removed_streams); >+} >+ >+void PeerConnection::RemoveRemoteStreamsIfEmpty( >+ const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& remote_streams, >+ std::vector<rtc::scoped_refptr<MediaStreamInterface>>* removed_streams) { >+ // TODO(https://crbug.com/webrtc/9480): When we use stream IDs instead of >+ // streams, see if the stream was removed by checking if this was the last >+ // receiver with that stream ID. >+ for (auto remote_stream : remote_streams) { >+ if (remote_stream->GetAudioTracks().empty() && >+ remote_stream->GetVideoTracks().empty()) { >+ remote_streams_->RemoveStream(remote_stream); >+ removed_streams->push_back(remote_stream); > } > } > } >@@ -3908,25 +4008,6 @@ void PeerConnection::GetOptionsForPlanBOffer( > offer_answer_options.num_simulcast_layers); > } > >-// Find a new MID that is not already in |used_mids|, then add it to |used_mids| >-// and return a reference to it. >-// Generated MIDs should be no more than 3 bytes long to take up less space in >-// the RTP packet. >-static const std::string& AllocateMid(std::set<std::string>* used_mids) { >- RTC_DCHECK(used_mids); >- // We're boring: just generate MIDs 0, 1, 2, ... >- size_t i = 0; >- std::set<std::string>::iterator it; >- bool inserted; >- do { >- std::string mid = rtc::ToString(i++); >- auto insert_result = used_mids->insert(mid); >- it = insert_result.first; >- inserted = insert_result.second; >- } while (!inserted); >- return *it; >-} >- > static cricket::MediaDescriptionOptions > GetMediaDescriptionOptionsForTransceiver( > rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>> >@@ -6086,6 +6167,21 @@ bool PeerConnection::HasRtcpMuxEnabled(const cricket::ContentInfo* content) { > return content->media_description()->rtcp_mux(); > } > >+static RTCError ValidateMids(const cricket::SessionDescription& description) { >+ std::set<std::string> mids; >+ for (const cricket::ContentInfo& content : description.contents()) { >+ if (content.name.empty()) { >+ LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, >+ "A media section is missing a MID attribute."); >+ } >+ if (!mids.insert(content.name).second) { >+ LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, >+ "Duplicate a=mid value '" + content.name + "'."); >+ } >+ } >+ return RTCError::OK(); >+} >+ > RTCError PeerConnection::ValidateSessionDescription( > const SessionDescriptionInterface* sdesc, > cricket::ContentSource source) { >@@ -6105,6 +6201,11 @@ RTCError PeerConnection::ValidateSessionDescription( > "Called in wrong state: " + GetSignalingStateString(signaling_state())); > } > >+ RTCError error = ValidateMids(*sdesc->description()); >+ if (!error.ok()) { >+ return error; >+ } >+ > // Verify crypto settings. > std::string crypto_error; > if (webrtc_session_desc_factory_->SdesPolicy() == cricket::SEC_REQUIRED || >diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection.h >index 1810f1c2bcd26e328b6f85eabc78a38cf0df55c8..6d7584d4e5fd047185140a21506556bc02406e69 100644 >--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection.h >+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection.h >@@ -472,6 +472,14 @@ class PeerConnection : public PeerConnectionInternal, > transceiver, > const SessionDescriptionInterface* sdesc) const; > >+ // Runs the algorithm **set the associated remote streams** specified in >+ // https://w3c.github.io/webrtc-pc/#set-associated-remote-streams. >+ void SetAssociatedRemoteStreams( >+ rtc::scoped_refptr<RtpReceiverInternal> receiver, >+ const std::vector<std::string>& stream_ids, >+ std::vector<rtc::scoped_refptr<MediaStreamInterface>>* added_streams, >+ std::vector<rtc::scoped_refptr<MediaStreamInterface>>* removed_streams); >+ > // Runs the algorithm **process the removal of a remote track** specified in > // the WebRTC specification. > // This method will update the following lists: >@@ -486,6 +494,11 @@ class PeerConnection : public PeerConnectionInternal, > std::vector<rtc::scoped_refptr<RtpTransceiverInterface>>* remove_list, > std::vector<rtc::scoped_refptr<MediaStreamInterface>>* removed_streams); > >+ void RemoveRemoteStreamsIfEmpty( >+ const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& >+ remote_streams, >+ std::vector<rtc::scoped_refptr<MediaStreamInterface>>* removed_streams); >+ > void OnNegotiationNeeded(); > > bool IsClosed() const { >@@ -649,6 +662,12 @@ class PeerConnection : public PeerConnectionInternal, > return configuration_.sdp_semantics == SdpSemantics::kUnifiedPlan; > } > >+ // The offer/answer machinery assumes the media section MID is present and >+ // unique. To support legacy end points that do not supply a=mid lines, this >+ // method will modify the session description to add MIDs generated according >+ // to the SDP semantics. >+ void FillInMissingRemoteMids(cricket::SessionDescription* remote_description); >+ > // Is there an RtpSender of the given type? > bool HasRtpSender(cricket::MediaType type) const; > >diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_jsep_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_jsep_unittest.cc >index f0c9111885ca7b295771d6ab92702a244c462271..7a044747876a5edb1fa4601e5d534c249b71a764 100644 >--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_jsep_unittest.cc >+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_jsep_unittest.cc >@@ -1604,4 +1604,110 @@ TEST_F(PeerConnectionJsepTest, OneVideoUnifiedPlanToTwoVideoPlanBFails) { > EXPECT_EQ(RTCErrorType::INVALID_PARAMETER, error.type()); > } > >+// Removes the RTP header extension associated with the given URI from the media >+// description. >+static void RemoveRtpHeaderExtensionByUri( >+ MediaContentDescription* media_description, >+ absl::string_view uri) { >+ std::vector<RtpExtension> header_extensions = >+ media_description->rtp_header_extensions(); >+ header_extensions.erase(std::remove_if( >+ header_extensions.begin(), header_extensions.end(), >+ [uri](const RtpExtension& extension) { return extension.uri == uri; })); >+ media_description->set_rtp_header_extensions(header_extensions); >+} >+ >+// Transforms a session description to emulate a legacy endpoint which does not >+// support a=mid, BUNDLE, and the MID header extension. >+static void ClearMids(SessionDescriptionInterface* sdesc) { >+ cricket::SessionDescription* desc = sdesc->description(); >+ desc->RemoveGroupByName(cricket::GROUP_TYPE_BUNDLE); >+ cricket::ContentInfo* audio_content = cricket::GetFirstAudioContent(desc); >+ if (audio_content) { >+ desc->GetTransportInfoByName(audio_content->name)->content_name = ""; >+ audio_content->name = ""; >+ RemoveRtpHeaderExtensionByUri(audio_content->media_description(), >+ RtpExtension::kMidUri); >+ } >+ cricket::ContentInfo* video_content = cricket::GetFirstVideoContent(desc); >+ if (video_content) { >+ desc->GetTransportInfoByName(video_content->name)->content_name = ""; >+ video_content->name = ""; >+ RemoveRtpHeaderExtensionByUri(video_content->media_description(), >+ RtpExtension::kMidUri); >+ } >+} >+ >+// Test that negotiation works with legacy endpoints which do not support a=mid. >+TEST_F(PeerConnectionJsepTest, LegacyNoMidAudioOnlyOffer) { >+ auto caller = CreatePeerConnection(); >+ caller->AddAudioTrack("audio"); >+ auto callee = CreatePeerConnection(); >+ callee->AddAudioTrack("audio"); >+ >+ auto offer = caller->CreateOffer(); >+ ClearMids(offer.get()); >+ >+ ASSERT_TRUE(callee->SetRemoteDescription(std::move(offer))); >+ EXPECT_TRUE(callee->SetLocalDescription(callee->CreateAnswer())); >+} >+TEST_F(PeerConnectionJsepTest, LegacyNoMidAudioVideoOffer) { >+ auto caller = CreatePeerConnection(); >+ caller->AddAudioTrack("audio"); >+ caller->AddVideoTrack("video"); >+ auto callee = CreatePeerConnection(); >+ callee->AddAudioTrack("audio"); >+ callee->AddVideoTrack("video"); >+ >+ auto offer = caller->CreateOffer(); >+ ClearMids(offer.get()); >+ >+ ASSERT_TRUE(callee->SetRemoteDescription(std::move(offer))); >+ EXPECT_TRUE(callee->SetLocalDescription(callee->CreateAnswer())); >+} >+TEST_F(PeerConnectionJsepTest, LegacyNoMidAudioOnlyAnswer) { >+ auto caller = CreatePeerConnection(); >+ caller->AddAudioTrack("audio"); >+ auto callee = CreatePeerConnection(); >+ callee->AddAudioTrack("audio"); >+ >+ ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal())); >+ >+ auto answer = callee->CreateAnswer(); >+ ClearMids(answer.get()); >+ >+ EXPECT_TRUE(caller->SetRemoteDescription(std::move(answer))); >+} >+TEST_F(PeerConnectionJsepTest, LegacyNoMidAudioVideoAnswer) { >+ auto caller = CreatePeerConnection(); >+ caller->AddAudioTrack("audio"); >+ caller->AddVideoTrack("video"); >+ auto callee = CreatePeerConnection(); >+ callee->AddAudioTrack("audio"); >+ callee->AddVideoTrack("video"); >+ >+ ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal())); >+ >+ auto answer = callee->CreateAnswer(); >+ ClearMids(answer.get()); >+ >+ ASSERT_TRUE(caller->SetRemoteDescription(std::move(answer))); >+} >+ >+// Test that SetLocalDescription fails if a=mid lines are missing. >+TEST_F(PeerConnectionJsepTest, SetLocalDescriptionFailsMissingMid) { >+ auto caller = CreatePeerConnection(); >+ caller->AddAudioTrack("audio"); >+ >+ auto offer = caller->CreateOffer(); >+ ClearMids(offer.get()); >+ >+ std::string error; >+ ASSERT_FALSE(caller->SetLocalDescription(std::move(offer), &error)); >+ EXPECT_EQ( >+ "Failed to set local offer sdp: A media section is missing a MID " >+ "attribute.", >+ error); >+} >+ > } // namespace webrtc >diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_media_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_media_unittest.cc >index 6af0c9887e6e2dfe4a1c07cad220ebd120e56486..25aea9828497ec10e9c4256106c337a98b08e780 100644 >--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_media_unittest.cc >+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_media_unittest.cc >@@ -1084,6 +1084,22 @@ TEST_P(PeerConnectionMediaTest, ReOfferHasSameMidsAsFirstOffer) { > cricket::GetFirstVideoContent(reoffer->description())->name); > } > >+// Test that SetRemoteDescription returns an error if there are two m= sections >+// with the same MID value. >+TEST_P(PeerConnectionMediaTest, SetRemoteDescriptionFailsWithDuplicateMids) { >+ auto caller = CreatePeerConnectionWithAudioVideo(); >+ auto callee = CreatePeerConnectionWithAudioVideo(); >+ >+ auto offer = caller->CreateOffer(); >+ RenameContent(offer->description(), cricket::MEDIA_TYPE_AUDIO, "same"); >+ RenameContent(offer->description(), cricket::MEDIA_TYPE_VIDEO, "same"); >+ >+ std::string error; >+ EXPECT_FALSE(callee->SetRemoteDescription(std::move(offer), &error)); >+ EXPECT_EQ(error, >+ "Failed to set remote offer sdp: Duplicate a=mid value 'same'."); >+} >+ > TEST_P(PeerConnectionMediaTest, > CombinedAudioVideoBweConfigPropagatedToMediaEngine) { > RTCConfiguration config; >diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_rtp_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_rtp_unittest.cc >index 7098470baaeee98d0e74a6fc7074a87611f9596b..0b1859ab723c0d45017566ea01fcfe42186697c1 100644 >--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_rtp_unittest.cc >+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_rtp_unittest.cc >@@ -455,6 +455,33 @@ TEST_F(PeerConnectionRtpTestUnifiedPlan, > EXPECT_EQ(1u, callee->observer()->remove_track_events_.size()); > } > >+TEST_F(PeerConnectionRtpTestUnifiedPlan, ChangeMsidWhileReceiving) { >+ auto caller = CreatePeerConnection(); >+ caller->AddAudioTrack("audio_track", {"stream1"}); >+ auto callee = CreatePeerConnection(); >+ ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOffer())); >+ >+ ASSERT_EQ(1u, callee->observer()->on_track_transceivers_.size()); >+ auto transceiver = callee->observer()->on_track_transceivers_[0]; >+ ASSERT_EQ(1u, transceiver->receiver()->streams().size()); >+ EXPECT_EQ("stream1", transceiver->receiver()->streams()[0]->id()); >+ >+ ASSERT_TRUE(callee->SetLocalDescription(callee->CreateAnswer())); >+ >+ // Change the stream ID in the offer. >+ // TODO(https://crbug.com/webrtc/10129): When RtpSenderInterface::SetStreams >+ // is supported, this can use that instead of munging the SDP. >+ auto offer = caller->CreateOffer(); >+ auto contents = offer->description()->contents(); >+ ASSERT_EQ(1u, contents.size()); >+ auto& stream_params = contents[0].media_description()->mutable_streams(); >+ ASSERT_EQ(1u, stream_params.size()); >+ stream_params[0].set_stream_ids({"stream2"}); >+ ASSERT_TRUE(callee->SetRemoteDescription(std::move(offer))); >+ ASSERT_EQ(1u, transceiver->receiver()->streams().size()); >+ EXPECT_EQ("stream2", transceiver->receiver()->streams()[0]->id()); >+} >+ > // These tests examine the state of the peer connection as a result of > // performing SetRemoteDescription(). > >diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/sessiondescription.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/sessiondescription.cc >index 3a9b18ee6919b37ff5940ee6064eccb7b46a8d4d..3346012dbb033bf1a4181d92255fc3ff31e5f856 100644 >--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/sessiondescription.cc >+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/sessiondescription.cc >@@ -196,12 +196,8 @@ bool SessionDescription::RemoveContentByName(const std::string& name) { > return false; > } > >-bool SessionDescription::AddTransportInfo(const TransportInfo& transport_info) { >- if (GetTransportInfoByName(transport_info.content_name) != NULL) { >- return false; >- } >+void SessionDescription::AddTransportInfo(const TransportInfo& transport_info) { > transport_infos_.push_back(transport_info); >- return true; > } > > bool SessionDescription::RemoveTransportInfoByName(const std::string& name) { >diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/sessiondescription.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/sessiondescription.h >index 38291484cd8cb4e8888737a78d6b0f82154e80f4..3a98dceb29728ef08c0f20ac04664b0a895a6bec 100644 >--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/sessiondescription.h >+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/sessiondescription.h >@@ -447,8 +447,7 @@ class SessionDescription { > transport_infos_ = transport_infos; > } > // Adds a TransportInfo to this description. >- // Returns false if a TransportInfo with the same name already exists. >- bool AddTransportInfo(const TransportInfo& transport_info); >+ void AddTransportInfo(const TransportInfo& transport_info); > bool RemoveTransportInfoByName(const std::string& name); > > // Group accessors. >diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/webrtcsdp.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/webrtcsdp.cc >index 0cd6beceefa377dffe7f934b9e5b96635046c545..0caaa120baa2d338e9466c53715f28806aa57bdb 100644 >--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/webrtcsdp.cc >+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/webrtcsdp.cc >@@ -2363,20 +2363,6 @@ static C* ParseContentDescription(const std::string& message, > std::vector<JsepIceCandidate*>* candidates, > webrtc::SdpParseError* error) { > C* media_desc = new C(); >- switch (media_type) { >- case cricket::MEDIA_TYPE_AUDIO: >- *content_name = cricket::CN_AUDIO; >- break; >- case cricket::MEDIA_TYPE_VIDEO: >- *content_name = cricket::CN_VIDEO; >- break; >- case cricket::MEDIA_TYPE_DATA: >- *content_name = cricket::CN_DATA; >- break; >- default: >- RTC_NOTREACHED(); >- break; >- } > if (!ParseContent(message, media_type, mline_index, protocol, payload_types, > pos, content_name, bundle_only, msid_signaling, media_desc, > transport, candidates, error)) { >@@ -2559,14 +2545,7 @@ bool ParseMediaDescription(const std::string& message, > : MediaProtocolType::kRtp, > content_rejected, bundle_only, content.release()); > // Create TransportInfo with the media level "ice-pwd" and "ice-ufrag". >- TransportInfo transport_info(content_name, transport); >- >- if (!desc->AddTransportInfo(transport_info)) { >- rtc::StringBuilder description; >- description << "Failed to AddTransportInfo with content name: " >- << content_name; >- return ParseFailed("", description.str(), error); >- } >+ desc->AddTransportInfo(TransportInfo(content_name, transport)); > } > > desc->set_msid_signaling(msid_signaling); >diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/webrtcsdp_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/webrtcsdp_unittest.cc >index 6775957498fd9cf20d8142bfb219a267962e0c92..ff7f903aaba0026e6ebcc51488f76e947382fd79 100644 >--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/webrtcsdp_unittest.cc >+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/webrtcsdp_unittest.cc >@@ -25,6 +25,8 @@ > #include "rtc_base/messagedigest.h" > #include "rtc_base/stringencode.h" > #include "rtc_base/stringutils.h" >+#include "test/gmock.h" >+#include "test/gtest.h" > > #ifdef WEBRTC_ANDROID > #include "pc/test/androidtestinitializer.h" >@@ -34,24 +36,26 @@ > using cricket::AudioCodec; > using cricket::AudioContentDescription; > using cricket::Candidate; >+using cricket::ContentGroup; > using cricket::ContentInfo; > using cricket::CryptoParams; >-using cricket::ContentGroup; > using cricket::DataCodec; > using cricket::DataContentDescription; > using cricket::ICE_CANDIDATE_COMPONENT_RTCP; > using cricket::ICE_CANDIDATE_COMPONENT_RTP; > using cricket::kFecSsrcGroupSemantics; > using cricket::LOCAL_PORT_TYPE; >+using cricket::MediaProtocolType; > using cricket::RELAY_PORT_TYPE; > using cricket::SessionDescription; >-using cricket::MediaProtocolType; > using cricket::StreamParams; > using cricket::STUN_PORT_TYPE; > using cricket::TransportDescription; > using cricket::TransportInfo; > using cricket::VideoCodec; > using cricket::VideoContentDescription; >+using ::testing::ElementsAre; >+using ::testing::Field; > using webrtc::IceCandidateCollection; > using webrtc::IceCandidateInterface; > using webrtc::JsepIceCandidate; >@@ -915,10 +919,10 @@ class WebRtcSdpTest : public testing::Test { > desc_.AddContent(kVideoContentName, MediaProtocolType::kRtp, video_desc_); > > // TransportInfo >- EXPECT_TRUE(desc_.AddTransportInfo(TransportInfo( >- kAudioContentName, TransportDescription(kUfragVoice, kPwdVoice)))); >- EXPECT_TRUE(desc_.AddTransportInfo(TransportInfo( >- kVideoContentName, TransportDescription(kUfragVideo, kPwdVideo)))); >+ desc_.AddTransportInfo(TransportInfo( >+ kAudioContentName, TransportDescription(kUfragVoice, kPwdVoice))); >+ desc_.AddTransportInfo(TransportInfo( >+ kVideoContentName, TransportDescription(kUfragVideo, kPwdVideo))); > > // v4 host > int port = 1234; >@@ -1112,8 +1116,8 @@ class WebRtcSdpTest : public testing::Test { > audio_track_2.ssrcs.push_back(kAudioTrack2Ssrc); > audio_desc_2->AddStream(audio_track_2); > desc_.AddContent(kAudioContentName2, MediaProtocolType::kRtp, audio_desc_2); >- EXPECT_TRUE(desc_.AddTransportInfo(TransportInfo( >- kAudioContentName2, TransportDescription(kUfragVoice2, kPwdVoice2)))); >+ desc_.AddTransportInfo(TransportInfo( >+ kAudioContentName2, TransportDescription(kUfragVoice2, kPwdVoice2))); > // Video track 2, in stream 2. > VideoContentDescription* video_desc_2 = CreateVideoContentDescription(); > StreamParams video_track_2; >@@ -1123,8 +1127,8 @@ class WebRtcSdpTest : public testing::Test { > video_track_2.ssrcs.push_back(kVideoTrack2Ssrc); > video_desc_2->AddStream(video_track_2); > desc_.AddContent(kVideoContentName2, MediaProtocolType::kRtp, video_desc_2); >- EXPECT_TRUE(desc_.AddTransportInfo(TransportInfo( >- kVideoContentName2, TransportDescription(kUfragVideo2, kPwdVideo2)))); >+ desc_.AddTransportInfo(TransportInfo( >+ kVideoContentName2, TransportDescription(kUfragVideo2, kPwdVideo2))); > > // Video track 3, in stream 2. > VideoContentDescription* video_desc_3 = CreateVideoContentDescription(); >@@ -1135,8 +1139,8 @@ class WebRtcSdpTest : public testing::Test { > video_track_3.ssrcs.push_back(kVideoTrack3Ssrc); > video_desc_3->AddStream(video_track_3); > desc_.AddContent(kVideoContentName3, MediaProtocolType::kRtp, video_desc_3); >- EXPECT_TRUE(desc_.AddTransportInfo(TransportInfo( >- kVideoContentName3, TransportDescription(kUfragVideo3, kPwdVideo3)))); >+ desc_.AddTransportInfo(TransportInfo( >+ kVideoContentName3, TransportDescription(kUfragVideo3, kPwdVideo3))); > desc_.set_msid_signaling(cricket::kMsidSignalingMediaSection); > > ASSERT_TRUE(jdesc_.Initialize(desc_.Copy(), jdesc_.session_id(), >@@ -1179,8 +1183,8 @@ class WebRtcSdpTest : public testing::Test { > audio_track_2.ssrcs.push_back(kAudioTrack2Ssrc); > audio_desc_2->AddStream(audio_track_2); > desc_.AddContent(kAudioContentName2, MediaProtocolType::kRtp, audio_desc_2); >- EXPECT_TRUE(desc_.AddTransportInfo(TransportInfo( >- kAudioContentName2, TransportDescription(kUfragVoice2, kPwdVoice2)))); >+ desc_.AddTransportInfo(TransportInfo( >+ kAudioContentName2, TransportDescription(kUfragVoice2, kPwdVoice2))); > > // Audio track 3 has no stream ids. > AudioContentDescription* audio_desc_3 = CreateAudioContentDescription(); >@@ -1191,8 +1195,8 @@ class WebRtcSdpTest : public testing::Test { > audio_track_3.ssrcs.push_back(kAudioTrack3Ssrc); > audio_desc_3->AddStream(audio_track_3); > desc_.AddContent(kAudioContentName3, MediaProtocolType::kRtp, audio_desc_3); >- EXPECT_TRUE(desc_.AddTransportInfo(TransportInfo( >- kAudioContentName3, TransportDescription(kUfragVoice3, kPwdVoice3)))); >+ desc_.AddTransportInfo(TransportInfo( >+ kAudioContentName3, TransportDescription(kUfragVoice3, kPwdVoice3))); > desc_.set_msid_signaling(msid_signaling); > ASSERT_TRUE(jdesc_.Initialize(desc_.Copy(), jdesc_.session_id(), > jdesc_.session_version())); >@@ -1464,7 +1468,7 @@ class WebRtcSdpTest : public testing::Test { > SessionDescription* desc = > const_cast<SessionDescription*>(jdesc->description()); > desc->RemoveTransportInfoByName(content_name); >- EXPECT_TRUE(desc->AddTransportInfo(transport_info)); >+ desc->AddTransportInfo(transport_info); > for (size_t i = 0; i < jdesc_.number_of_mediasections(); ++i) { > const IceCandidateCollection* cc = jdesc_.candidates(i); > for (size_t j = 0; j < cc->count(); ++j) { >@@ -1503,16 +1507,16 @@ class WebRtcSdpTest : public testing::Test { > desc_.RemoveTransportInfoByName(kAudioContentName); > desc_.RemoveTransportInfoByName(kVideoContentName); > rtc::SSLFingerprint fingerprint(rtc::DIGEST_SHA_1, kIdentityDigest); >- EXPECT_TRUE(desc_.AddTransportInfo(TransportInfo( >+ desc_.AddTransportInfo(TransportInfo( > kAudioContentName, > TransportDescription(std::vector<std::string>(), kUfragVoice, kPwdVoice, > cricket::ICEMODE_FULL, >- cricket::CONNECTIONROLE_NONE, &fingerprint)))); >- EXPECT_TRUE(desc_.AddTransportInfo(TransportInfo( >+ cricket::CONNECTIONROLE_NONE, &fingerprint))); >+ desc_.AddTransportInfo(TransportInfo( > kVideoContentName, > TransportDescription(std::vector<std::string>(), kUfragVideo, kPwdVideo, > cricket::ICEMODE_FULL, >- cricket::CONNECTIONROLE_NONE, &fingerprint)))); >+ cricket::CONNECTIONROLE_NONE, &fingerprint))); > } > > void AddExtmap(bool encrypted) { >@@ -1629,8 +1633,8 @@ class WebRtcSdpTest : public testing::Test { > data_desc_->AddCodec(codec); > desc_.AddContent(kDataContentName, MediaProtocolType::kSctp, > data.release()); >- EXPECT_TRUE(desc_.AddTransportInfo(TransportInfo( >- kDataContentName, TransportDescription(kUfragData, kPwdData)))); >+ desc_.AddTransportInfo(TransportInfo( >+ kDataContentName, TransportDescription(kUfragData, kPwdData))); > } > > void AddRtpDataChannel() { >@@ -1649,8 +1653,8 @@ class WebRtcSdpTest : public testing::Test { > "inline:FvLcvU2P3ZWmQxgPAgcDu7Zl9vftYElFOjEzhWs5", "")); > data_desc_->set_protocol(cricket::kMediaProtocolSavpf); > desc_.AddContent(kDataContentName, MediaProtocolType::kRtp, data.release()); >- EXPECT_TRUE(desc_.AddTransportInfo(TransportInfo( >- kDataContentName, TransportDescription(kUfragData, kPwdData)))); >+ desc_.AddTransportInfo(TransportInfo( >+ kDataContentName, TransportDescription(kUfragData, kPwdData))); > } > > bool TestDeserializeDirection(RtpTransceiverDirection direction) { >@@ -2300,9 +2304,7 @@ TEST_F(WebRtcSdpTest, DeserializeSessionDescriptionWithoutRtpmap) { > JsepSessionDescription jdesc(kDummyType); > EXPECT_TRUE(SdpDeserialize(kSdpNoRtpmapString, &jdesc)); > cricket::AudioContentDescription* audio = >- jdesc.description() >- ->GetContentDescriptionByName(cricket::CN_AUDIO) >- ->as_audio(); >+ cricket::GetFirstAudioContentDescription(jdesc.description()); > AudioCodecs ref_codecs; > // The codecs in the AudioContentDescription should be in the same order as > // the payload types (<fmt>s) on the m= line. >@@ -2325,9 +2327,7 @@ TEST_F(WebRtcSdpTest, DeserializeSessionDescriptionWithoutRtpmapButWithFmtp) { > JsepSessionDescription jdesc(kDummyType); > EXPECT_TRUE(SdpDeserialize(kSdpNoRtpmapString, &jdesc)); > cricket::AudioContentDescription* audio = >- jdesc.description() >- ->GetContentDescriptionByName(cricket::CN_AUDIO) >- ->as_audio(); >+ cricket::GetFirstAudioContentDescription(jdesc.description()); > > cricket::AudioCodec g729 = audio->codecs()[0]; > EXPECT_EQ("G729", g729.name); >@@ -2924,15 +2924,11 @@ TEST_F(WebRtcSdpTest, DeserializeSdpWithConferenceFlag) { > > // Verify > cricket::AudioContentDescription* audio = >- jdesc.description() >- ->GetContentDescriptionByName(cricket::CN_AUDIO) >- ->as_audio(); >+ cricket::GetFirstAudioContentDescription(jdesc.description()); > EXPECT_TRUE(audio->conference_mode()); > > cricket::VideoContentDescription* video = >- jdesc.description() >- ->GetContentDescriptionByName(cricket::CN_VIDEO) >- ->as_video(); >+ cricket::GetFirstVideoContentDescription(jdesc.description()); > EXPECT_TRUE(video->conference_mode()); > } > >@@ -2947,15 +2943,11 @@ TEST_F(WebRtcSdpTest, SerializeSdpWithConferenceFlag) { > > // Verify. > cricket::AudioContentDescription* audio = >- jdesc.description() >- ->GetContentDescriptionByName(cricket::CN_AUDIO) >- ->as_audio(); >+ cricket::GetFirstAudioContentDescription(jdesc.description()); > EXPECT_TRUE(audio->conference_mode()); > > cricket::VideoContentDescription* video = >- jdesc.description() >- ->GetContentDescriptionByName(cricket::CN_VIDEO) >- ->as_video(); >+ cricket::GetFirstVideoContentDescription(jdesc.description()); > EXPECT_TRUE(video->conference_mode()); > } > >@@ -3909,3 +3901,19 @@ TEST_F(WebRtcSdpTest, DeserializeEmptySessionName) { > Replace("s=-\r\n", "s= \r\n", &sdp); > EXPECT_TRUE(SdpDeserialize(sdp, &jsep_desc)); > } >+ >+// Test that the content name is empty if the media section does not have an >+// a=mid line. >+TEST_F(WebRtcSdpTest, ParseNoMid) { >+ std::string sdp = kSdpString; >+ Replace("a=mid:audio_content_name\r\n", "", &sdp); >+ Replace("a=mid:video_content_name\r\n", "", &sdp); >+ >+ JsepSessionDescription output(kDummyType); >+ SdpParseError error; >+ ASSERT_TRUE(webrtc::SdpDeserialize(sdp, &output, &error)); >+ >+ EXPECT_THAT(output.description()->contents(), >+ ElementsAre(Field("name", &cricket::ContentInfo::name, ""), >+ Field("name", &cricket::ContentInfo::name, ""))); >+} >diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/include/metrics.h b/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/include/metrics.h >index f00ecf2622aefd817717086b63ab4eb4d9b33929..cbefa37bd1b5cded7689016e53c111e57ab6c6f7 100644 >--- a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/include/metrics.h >+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/include/metrics.h >@@ -133,7 +133,7 @@ > // TODO(qingsi): Refactor the default implementation given by RtcHistogram, > // which is already sparse, and remove the boundary argument from the macro. > #define RTC_HISTOGRAM_ENUMERATION_SPARSE(name, sample, boundary) \ >- RTC_HISTOGRAM_COMMON_BLOCK( \ >+ RTC_HISTOGRAM_COMMON_BLOCK_SLOW( \ > name, sample, \ > webrtc::metrics::SparseHistogramFactoryGetEnumeration(name, boundary)) > >@@ -148,7 +148,7 @@ > // Histogram for enumerators (evenly spaced buckets). > // |boundary| should be above the max enumerator sample. > #define RTC_HISTOGRAM_ENUMERATION(name, sample, boundary) \ >- RTC_HISTOGRAM_COMMON_BLOCK( \ >+ RTC_HISTOGRAM_COMMON_BLOCK_SLOW( \ > name, sample, \ > webrtc::metrics::HistogramFactoryGetEnumeration(name, boundary)) > >@@ -175,7 +175,6 @@ > } \ > } while (0) > >-// Deprecated. > // The histogram is constructed/found for each call. > // May be used for histograms with infrequent updates.` > #define RTC_HISTOGRAM_COMMON_BLOCK_SLOW(name, sample, factory_get_invocation) \ >diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/video/BUILD.gn >index b53d730ae07e06f8004e35536cae722a1a5a6ec6..28fb690cec5ec2f67a780326416a0aed4f78e67d 100644 >--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/BUILD.gn >+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/BUILD.gn >@@ -181,6 +181,7 @@ rtc_source_set("video_stream_encoder_impl") { > "../common_video:common_video", > "../modules/video_coding", > "../modules/video_coding:video_coding_utility", >+ "../modules/video_coding:webrtc_vp9_helpers", > "../rtc_base:checks", > "../rtc_base:criticalsection", > "../rtc_base:logging",
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