Diff
Modified: trunk/LayoutTests/imported/w3c/ChangeLog (240319 => 240320)
--- trunk/LayoutTests/imported/w3c/ChangeLog 2019-01-23 04:28:57 UTC (rev 240319)
+++ trunk/LayoutTests/imported/w3c/ChangeLog 2019-01-23 04:42:54 UTC (rev 240320)
@@ -1,3 +1,13 @@
+2019-01-22 Youenn Fablet <[email protected]>
+
+ Resync libwebrtc with latest M72 branch
+ https://bugs.webkit.org/show_bug.cgi?id=193693
+ <rdar://problem/47463803>
+
+ Reviewed by Eric Carlson.
+
+ * web-platform-tests/webrtc/RTCRtpTransceiver.https-expected.txt:
+
2019-01-22 Oriol Brufau <[email protected]>
[css-logical] Implement flow-relative margin, padding and border shorthands
Modified: trunk/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCRtpTransceiver.https-expected.txt (240319 => 240320)
--- trunk/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCRtpTransceiver.https-expected.txt 2019-01-23 04:28:57 UTC (rev 240319)
+++ trunk/LayoutTests/imported/w3c/web-platform-tests/webrtc/RTCRtpTransceiver.https-expected.txt 2019-01-23 04:42:54 UTC (rev 240320)
@@ -8,7 +8,7 @@
PASS checkAddTransceiverWithSetRemoteOfferSending
PASS checkAddTransceiverWithSetRemoteOfferNoSend
PASS checkAddTransceiverBadKind
-FAIL checkNoMidOffer promise_test: Unhandled rejection with value: object "InvalidAccessError: Failed to set remote offer sdp: The BUNDLE group contains MID:0 matching no m= section."
+PASS checkNoMidOffer
PASS checkSetDirection
PASS checkCurrentDirection
PASS checkSendrecvWithNoSendTrack
Modified: trunk/Source/ThirdParty/libwebrtc/ChangeLog (240319 => 240320)
--- trunk/Source/ThirdParty/libwebrtc/ChangeLog 2019-01-23 04:28:57 UTC (rev 240319)
+++ trunk/Source/ThirdParty/libwebrtc/ChangeLog 2019-01-23 04:42:54 UTC (rev 240320)
@@ -1,3 +1,35 @@
+2019-01-22 Youenn Fablet <[email protected]>
+
+ Resync libwebrtc with latest M72 branch
+ https://bugs.webkit.org/show_bug.cgi?id=193693
+
+ Reviewed by Eric Carlson.
+
+ 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:
+
2019-01-18 Jer Noble <[email protected]>
SDK_VARIANT build destinations should be separate from non-SDK_VARIANT builds
Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/delay_based_bwe.cc (240319 => 240320)
--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/delay_based_bwe.cc 2019-01-23 04:28:57 UTC (rev 240319)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/delay_based_bwe.cc 2019-01-23 04:42:54 UTC (rev 240320)
@@ -46,8 +46,6 @@
constexpr double kDefaultTrendlineSmoothingCoeff = 0.9;
constexpr double kDefaultTrendlineThresholdGain = 4.0;
-constexpr int kMaxConsecutiveFailedLookups = 5;
-
const char kBweWindowSizeInPacketsExperiment[] =
"WebRTC-BweWindowSizeInPackets";
@@ -94,7 +92,6 @@
: 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)
@@ -147,45 +144,14 @@
}
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);
+ // TODO(bugs.webrtc.org/10125): Design a better mechanism to safe-guard
+ // against building very large network queues.
+ return Result();
}
- return Result();
+ return MaybeUpdateEstimate(acked_bitrate, probe_bitrate,
+ recovered_from_overuse, at_time);
}
-DelayBasedBwe::Result DelayBasedBwe::OnDelayedFeedback(Timestamp receive_time) {
- ++consecutive_delayed_feedbacks_;
- if (consecutive_delayed_feedbacks_ >= kMaxConsecutiveFailedLookups) {
- consecutive_delayed_feedbacks_ = 0;
- return OnLongFeedbackDelay(receive_time);
- }
- 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;
-}
-
void DelayBasedBwe::IncomingPacketFeedback(
const PacketFeedback& packet_feedback,
Timestamp at_time) {
Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/delay_based_bwe.h (240319 => 240320)
--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/delay_based_bwe.h 2019-01-23 04:28:57 UTC (rev 240319)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/delay_based_bwe.h 2019-01-23 04:42:54 UTC (rev 240320)
@@ -49,7 +49,6 @@
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 @@
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 @@
size_t trendline_window_size_;
double trendline_smoothing_coeff_;
double trendline_threshold_gain_;
- int consecutive_delayed_feedbacks_;
DataRate prev_bitrate_;
BandwidthUsage prev_state_;
Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/goog_cc_network_control.cc (240319 => 240320)
--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/goog_cc_network_control.cc 2019-01-23 04:28:57 UTC (rev 240319)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/goog_cc_network_control.cc 2019-01-23 04:42:54 UTC (rev 240320)
@@ -394,15 +394,9 @@
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_) {
Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/goog_cc_network_control_unittest.cc (240319 => 240320)
--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/goog_cc_network_control_unittest.cc 2019-01-23 04:28:57 UTC (rev 240319)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/goog_cc_network_control_unittest.cc 2019-01-23 04:42:54 UTC (rev 240320)
@@ -193,74 +193,6 @@
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) {
Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/send_side_congestion_controller_unittest.cc (240319 => 240320)
--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/send_side_congestion_controller_unittest.cc 2019-01-23 04:28:57 UTC (rev 240319)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/send_side_congestion_controller_unittest.cc 2019-01-23 04:42:54 UTC (rev 240320)
@@ -344,97 +344,6 @@
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) {
Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jsepsessiondescription_unittest.cc (240319 => 240320)
--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jsepsessiondescription_unittest.cc 2019-01-23 04:28:57 UTC (rev 240319)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jsepsessiondescription_unittest.cc 2019-01-23 04:42:54 UTC (rev 240320)
@@ -60,16 +60,16 @@
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;
}
Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/pc/mediasession.cc (240319 => 240320)
--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/pc/mediasession.cc 2019-01-23 04:28:57 UTC (rev 240319)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/pc/mediasession.cc 2019-01-23 04:42:54 UTC (rev 240320)
@@ -1799,14 +1799,12 @@
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 @@
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;
}
Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/pc/mediasession_unittest.cc (240319 => 240320)
--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/pc/mediasession_unittest.cc 2019-01-23 04:28:57 UTC (rev 240319)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/pc/mediasession_unittest.cc 2019-01-23 04:42:54 UTC (rev 240320)
@@ -426,14 +426,14 @@
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()));
Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection.cc (240319 => 240320)
--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection.cc 2019-01-23 04:28:57 UTC (rev 240319)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection.cc 2019-01-23 04:42:54 UTC (rev 240320)
@@ -649,6 +649,25 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
"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 ||
Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection.h (240319 => 240320)
--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection.h 2019-01-23 04:28:57 UTC (rev 240319)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection.h 2019-01-23 04:42:54 UTC (rev 240320)
@@ -472,6 +472,14 @@
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 @@
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 @@
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;
Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_jsep_unittest.cc (240319 => 240320)
--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_jsep_unittest.cc 2019-01-23 04:28:57 UTC (rev 240319)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_jsep_unittest.cc 2019-01-23 04:42:54 UTC (rev 240320)
@@ -1604,4 +1604,110 @@
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
Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_media_unittest.cc (240319 => 240320)
--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_media_unittest.cc 2019-01-23 04:28:57 UTC (rev 240319)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_media_unittest.cc 2019-01-23 04:42:54 UTC (rev 240320)
@@ -1084,6 +1084,22 @@
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;
Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_rtp_unittest.cc (240319 => 240320)
--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_rtp_unittest.cc 2019-01-23 04:28:57 UTC (rev 240319)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_rtp_unittest.cc 2019-01-23 04:42:54 UTC (rev 240320)
@@ -455,6 +455,33 @@
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().
Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/pc/sessiondescription.cc (240319 => 240320)
--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/pc/sessiondescription.cc 2019-01-23 04:28:57 UTC (rev 240319)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/pc/sessiondescription.cc 2019-01-23 04:42:54 UTC (rev 240320)
@@ -196,12 +196,8 @@
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) {
Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/pc/sessiondescription.h (240319 => 240320)
--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/pc/sessiondescription.h 2019-01-23 04:28:57 UTC (rev 240319)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/pc/sessiondescription.h 2019-01-23 04:42:54 UTC (rev 240320)
@@ -447,8 +447,7 @@
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.
Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/pc/webrtcsdp.cc (240319 => 240320)
--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/pc/webrtcsdp.cc 2019-01-23 04:28:57 UTC (rev 240319)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/pc/webrtcsdp.cc 2019-01-23 04:42:54 UTC (rev 240320)
@@ -2363,20 +2363,6 @@
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 @@
: 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);
Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/pc/webrtcsdp_unittest.cc (240319 => 240320)
--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/pc/webrtcsdp_unittest.cc 2019-01-23 04:28:57 UTC (rev 240319)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/pc/webrtcsdp_unittest.cc 2019-01-23 04:42:54 UTC (rev 240320)
@@ -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"
@@ -34,9 +36,9 @@
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;
@@ -43,9 +45,9 @@
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;
@@ -52,6 +54,8 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
"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 @@
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 @@
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 @@
// 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 @@
// 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 @@
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, "")));
+}
Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/include/metrics.h (240319 => 240320)
--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/include/metrics.h 2019-01-23 04:28:57 UTC (rev 240319)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/include/metrics.h 2019-01-23 04:42:54 UTC (rev 240320)
@@ -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) \
Modified: trunk/Source/ThirdParty/libwebrtc/Source/webrtc/video/BUILD.gn (240319 => 240320)
--- trunk/Source/ThirdParty/libwebrtc/Source/webrtc/video/BUILD.gn 2019-01-23 04:28:57 UTC (rev 240319)
+++ trunk/Source/ThirdParty/libwebrtc/Source/webrtc/video/BUILD.gn 2019-01-23 04:42:54 UTC (rev 240320)
@@ -181,6 +181,7 @@
"../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",
