m126 merge fixes

api/audio/audio_device.h

@@ -151,6 +151,9 @@ class AudioDeviceModule : public webrtc::RefCountInterface {
   virtual bool BuiltInAECIsAvailable() const = 0;
   virtual bool BuiltInAGCIsAvailable() const = 0;
   virtual bool BuiltInNSIsAvailable() const = 0;
+  // RingRTC change to allow control of AEC3 vs AECM
+  // When using software AEC, use AECM instead of AEC3.
+  virtual bool UseAecm() const { return false; }
 
   // Enables the built-in audio effects. Only supported on Android.
   virtual int32_t EnableBuiltInAEC(bool enable) = 0;

api/audio/audio_processing.h

@@ -536,7 +536,17 @@ class RTC_EXPORT AudioProcessing : public RefCountInterface {
   // but some components may change behavior based on this information.
   // Default false. This method takes a lock. To achieve this in a lock-less
   // manner the PostRuntimeSetting can instead be used.
-  virtual void set_output_will_be_muted(bool muted) = 0;
+  //
+  // RingRTC change to make it possible to share an APM (and thus, WebRtcVoiceEngine)
+  // between PeerConnections created from the same PeerConnectionFactory.
+  // Without this change, the code has a "last user wins" behavior,
+  // which breaks when creating multiple PeerConnections from the
+  // same PeerConnectionFactory.  In particular, this happens when using
+  // ICE forking and the unused PeerConnections signal they are not using
+  // captured output after the used PeerConnection signals that it is using
+  // captured output.  In that case, the call will end up
+  // without APM processing, which means it will have noise and echo.
+  virtual void set_capture_output_used(void* user, bool muted) = 0;
 
   // Enqueues a runtime setting.
   virtual void SetRuntimeSetting(RuntimeSetting setting) = 0;

audio/channel_receive.cc

@@ -703,7 +703,7 @@ void ChannelReceive::ReceivePacket(const uint8_t* packet,
     const std::vector<uint32_t> csrcs(header.arrOfCSRCs,
                                       header.arrOfCSRCs + header.numCSRCs);
     // RingRTC change to determine if TOC is encrypted
-    bool has_encrypted_toc = header.extension.hasAudioLevel && ((header.extension.audioLevel & 0x1) != 0);
+    bool has_encrypted_toc = header.extension.audio_level() && ((header.extension.audio_level()->level() & 0x1) != 0);
     const FrameDecryptorInterface::Result decrypt_result =
         frame_decryptor_->Decrypt(
             cricket::MEDIA_TYPE_AUDIO, csrcs,

modules/audio_coding/codecs/opus/opus_interface.cc

@@ -373,6 +373,11 @@ int16_t WebRtcOpus_EnableDtx(OpusEncInst* inst) {
   // RingRTC change to log opus setters
   RTC_LOG(LS_INFO) << "WebRtcOpus_EnableDtx";
   if (inst) {
+    // RingRTC change to force the Opus signal type to voice
+    int ret = ENCODER_CTL(inst, OPUS_SET_SIGNAL(OPUS_SIGNAL_VOICE));
+    if (ret != OPUS_OK) {
+      return ret;
+    }
     return ENCODER_CTL(inst, OPUS_SET_DTX(1));
   } else {
     return -1;

p2p/base/p2p_transport_channel.cc

@@ -953,7 +953,7 @@ void P2PTransportChannel::StartGatheringWithSharedGatherer(
   RTC_DCHECK_RUN_ON(network_thread_);
   if (gathering_state_ != kIceGatheringGathering) {
     gathering_state_ = kIceGatheringGathering;
-    SignalGatheringState(this);
+    SendGatheringStateEvent();
   }
 
   // Note: we must set this before calling OnPortReady below to make sure

p2p/base/p2p_transport_channel_unittest.cc

@@ -6187,7 +6187,7 @@ TEST_F(P2PTransportChannelPingTest, Forking) {
       kShortTimeout, clock);
   transport1->SetRemoteIceParameters(kIceParams[1]);
   transport1->AddRemoteCandidate(
-      CreateUdpCandidate(LOCAL_PORT_TYPE, "1.1.1.1", 1, 1));
+      CreateUdpCandidate(IceCandidateType::kHost, "1.1.1.1", 1, 1));
   ASSERT_EQ(1u, transport1->connections().size());
   Connection* transport1_conn =
       WaitForConnectionTo(transport1.get(), "1.1.1.1", 1, &clock);
@@ -6208,7 +6208,7 @@ TEST_F(P2PTransportChannelPingTest, Forking) {
       kShortTimeout, clock);
   transport2->SetRemoteIceParameters(kIceParams[2]);
   transport2->AddRemoteCandidate(
-      CreateUdpCandidate(LOCAL_PORT_TYPE, "2.2.2.2", 2, 2));
+      CreateUdpCandidate(IceCandidateType::kHost, "2.2.2.2", 2, 2));
   ASSERT_EQ(1u, transport2->connections().size());
   Connection* transport2_conn =
       WaitForConnectionTo(transport2.get(), "2.2.2.2", 2, &clock);

p2p/client/basic_port_allocator.cc

@@ -282,7 +282,6 @@ BasicPortAllocator::CreateIceGatherer(const std::string& name) {
 
   // 4. User setters to set flags and other settings
   new_allocator->set_flags(flags());
-  new_allocator->set_proxy(user_agent(), proxy());
   new_allocator->SetPortRange(min_port(), max_port());
   new_allocator->set_max_ipv6_networks(max_ipv6_networks());
   new_allocator->set_step_delay(step_delay());

pc/channel_unittest.cc

@@ -341,6 +341,8 @@ class ChannelTest : public ::testing::Test, public sigslot::has_slots<> {
       rtc::PacketTransportInternal* rtcp_packet_transport) {
     auto rtp_transport = std::make_unique<webrtc::RtpTransport>(
         rtcp_packet_transport == nullptr);
+    // RingRTC change to drop all incoming packets until explicitly allowed
+    rtp_transport->SetIncomingRtpEnabled(true);
 
     SendTask(network_thread_,
              [&rtp_transport, rtp_packet_transport, rtcp_packet_transport] {
@@ -357,6 +359,8 @@ class ChannelTest : public ::testing::Test, public sigslot::has_slots<> {
       cricket::DtlsTransportInternal* rtcp_dtls_transport) {
     auto dtls_srtp_transport = std::make_unique<webrtc::DtlsSrtpTransport>(
         rtcp_dtls_transport == nullptr, field_trials_);
+    // RingRTC change to drop all incoming packets until explicitly allowed
+    dtls_srtp_transport->SetIncomingRtpEnabled(true);
 
     SendTask(network_thread_,
              [&dtls_srtp_transport, rtp_dtls_transport, rtcp_dtls_transport] {

pc/dtls_srtp_transport_unittest.cc

@@ -62,6 +62,9 @@ class DtlsSrtpTransportTest : public ::testing::Test,
     auto dtls_srtp_transport =
         std::make_unique<DtlsSrtpTransport>(rtcp_mux_enabled, field_trials_);
 
+    // RingRTC change to drop all incoming packets until explicitly allowed
+    dtls_srtp_transport->SetIncomingRtpEnabled(true);
+
     dtls_srtp_transport->SetDtlsTransports(rtp_dtls, rtcp_dtls);
 
     return dtls_srtp_transport;

pc/rtp_transport_unittest.cc

@@ -83,6 +83,8 @@ class SignalObserver : public sigslot::has_slots<> {
 
 TEST(RtpTransportTest, SettingRtcpAndRtpSignalsReady) {
   RtpTransport transport(kMuxDisabled);
+  // RingRTC change to drop all incoming packets until explicitly allowed
+  transport.SetIncomingRtpEnabled(true);
   SignalObserver observer(&transport);
   rtc::FakePacketTransport fake_rtcp("fake_rtcp");
   fake_rtcp.SetWritable(true);
@@ -97,6 +99,8 @@ TEST(RtpTransportTest, SettingRtcpAndRtpSignalsReady) {
 
 TEST(RtpTransportTest, SettingRtpAndRtcpSignalsReady) {
   RtpTransport transport(kMuxDisabled);
+  // RingRTC change to drop all incoming packets until explicitly allowed
+  transport.SetIncomingRtpEnabled(true);
   SignalObserver observer(&transport);
   rtc::FakePacketTransport fake_rtcp("fake_rtcp");
   fake_rtcp.SetWritable(true);
@@ -111,6 +115,8 @@ TEST(RtpTransportTest, SettingRtpAndRtcpSignalsReady) {
 
 TEST(RtpTransportTest, SettingRtpWithRtcpMuxEnabledSignalsReady) {
   RtpTransport transport(kMuxEnabled);
+  // RingRTC change to drop all incoming packets until explicitly allowed
+  transport.SetIncomingRtpEnabled(true);
   SignalObserver observer(&transport);
   rtc::FakePacketTransport fake_rtp("fake_rtp");
   fake_rtp.SetWritable(true);
@@ -121,6 +127,8 @@ TEST(RtpTransportTest, SettingRtpWithRtcpMuxEnabledSignalsReady) {
 
 TEST(RtpTransportTest, DisablingRtcpMuxSignalsNotReady) {
   RtpTransport transport(kMuxEnabled);
+  // RingRTC change to drop all incoming packets until explicitly allowed
+  transport.SetIncomingRtpEnabled(true);
   SignalObserver observer(&transport);
   rtc::FakePacketTransport fake_rtp("fake_rtp");
   fake_rtp.SetWritable(true);
@@ -134,6 +142,8 @@ TEST(RtpTransportTest, DisablingRtcpMuxSignalsNotReady) {
 
 TEST(RtpTransportTest, EnablingRtcpMuxSignalsReady) {
   RtpTransport transport(kMuxDisabled);
+  // RingRTC change to drop all incoming packets until explicitly allowed
+  transport.SetIncomingRtpEnabled(true);
   SignalObserver observer(&transport);
   rtc::FakePacketTransport fake_rtp("fake_rtp");
   fake_rtp.SetWritable(true);
@@ -148,6 +158,8 @@ TEST(RtpTransportTest, EnablingRtcpMuxSignalsReady) {
 // Tests the SignalNetworkRoute is fired when setting a packet transport.
 TEST(RtpTransportTest, SetRtpTransportWithNetworkRouteChanged) {
   RtpTransport transport(kMuxDisabled);
+  // RingRTC change to drop all incoming packets until explicitly allowed
+  transport.SetIncomingRtpEnabled(true);
   SignalObserver observer(&transport);
   rtc::FakePacketTransport fake_rtp("fake_rtp");
 
@@ -177,6 +189,8 @@ TEST(RtpTransportTest, SetRtpTransportWithNetworkRouteChanged) {
 
 TEST(RtpTransportTest, SetRtcpTransportWithNetworkRouteChanged) {
   RtpTransport transport(kMuxDisabled);
+  // RingRTC change to drop all incoming packets until explicitly allowed
+  transport.SetIncomingRtpEnabled(true);
   SignalObserver observer(&transport);
   rtc::FakePacketTransport fake_rtcp("fake_rtcp");
 
@@ -210,6 +224,8 @@ TEST(RtpTransportTest, RtcpPacketSentOverCorrectTransport) {
   // If the RTCP-mux is not enabled, RTCP packets are expected to be sent over
   // the RtcpPacketTransport.
   RtpTransport transport(kMuxDisabled);
+  // RingRTC change to drop all incoming packets until explicitly allowed
+  transport.SetIncomingRtpEnabled(true);
   rtc::FakePacketTransport fake_rtcp("fake_rtcp");
   rtc::FakePacketTransport fake_rtp("fake_rtp");
   transport.SetRtcpPacketTransport(&fake_rtcp);  // rtcp ready
@@ -232,6 +248,8 @@ TEST(RtpTransportTest, RtcpPacketSentOverCorrectTransport) {
 
 TEST(RtpTransportTest, ChangingReadyToSendStateOnlySignalsWhenChanged) {
   RtpTransport transport(kMuxEnabled);
+  // RingRTC change to drop all incoming packets until explicitly allowed
+  transport.SetIncomingRtpEnabled(true);
   TransportObserver observer(&transport);
   rtc::FakePacketTransport fake_rtp("fake_rtp");
   fake_rtp.SetWritable(true);
@@ -257,6 +275,8 @@ TEST(RtpTransportTest, ChangingReadyToSendStateOnlySignalsWhenChanged) {
 // received.
 TEST(RtpTransportTest, SignalDemuxedRtcp) {
   RtpTransport transport(kMuxDisabled);
+  // RingRTC change to drop all incoming packets until explicitly allowed
+  transport.SetIncomingRtpEnabled(true);
   rtc::FakePacketTransport fake_rtp("fake_rtp");
   fake_rtp.SetDestination(&fake_rtp, true);
   transport.SetRtpPacketTransport(&fake_rtp);
@@ -280,6 +300,8 @@ static const int kRtpLen = 12;
 // handled payload type is received.
 TEST(RtpTransportTest, SignalHandledRtpPayloadType) {
   RtpTransport transport(kMuxDisabled);
+  // RingRTC change to drop all incoming packets until explicitly allowed
+  transport.SetIncomingRtpEnabled(true);
   rtc::FakePacketTransport fake_rtp("fake_rtp");
   fake_rtp.SetDestination(&fake_rtp, true);
   transport.SetRtpPacketTransport(&fake_rtp);
@@ -303,6 +325,8 @@ TEST(RtpTransportTest, SignalHandledRtpPayloadType) {
 
 TEST(RtpTransportTest, ReceivedPacketEcnMarkingPropagatedToDemuxedPacket) {
   RtpTransport transport(kMuxDisabled);
+  // RingRTC change to drop all incoming packets until explicitly allowed
+  transport.SetIncomingRtpEnabled(true);
   // Setup FakePacketTransport to send packets to itself.
   rtc::FakePacketTransport fake_rtp("fake_rtp");
   fake_rtp.SetDestination(&fake_rtp, true);
@@ -328,6 +352,8 @@ TEST(RtpTransportTest, ReceivedPacketEcnMarkingPropagatedToDemuxedPacket) {
 // unhandled payload type is received.
 TEST(RtpTransportTest, DontSignalUnhandledRtpPayloadType) {
   RtpTransport transport(kMuxDisabled);
+  // RingRTC change to drop all incoming packets until explicitly allowed
+  transport.SetIncomingRtpEnabled(true);
   rtc::FakePacketTransport fake_rtp("fake_rtp");
   fake_rtp.SetDestination(&fake_rtp, true);
   transport.SetRtpPacketTransport(&fake_rtp);
@@ -352,6 +378,8 @@ TEST(RtpTransportTest, RecursiveSetSendDoesNotCrash) {
   const int kShortTimeout = 100;
   test::RunLoop loop;
   RtpTransport transport(kMuxEnabled);
+  // RingRTC change to drop all incoming packets until explicitly allowed
+  transport.SetIncomingRtpEnabled(true);
   rtc::FakePacketTransport fake_rtp("fake_rtp");
   transport.SetRtpPacketTransport(&fake_rtp);
   TransportObserver observer(&transport);
@@ -376,6 +404,8 @@ TEST(RtpTransportTest, RecursiveOnSentPacketDoesNotCrash) {
   const int kShortTimeout = 100;
   test::RunLoop loop;
   RtpTransport transport(kMuxEnabled);
+  // RingRTC change to drop all incoming packets until explicitly allowed
+  transport.SetIncomingRtpEnabled(true);
   rtc::FakePacketTransport fake_rtp("fake_rtp");
   transport.SetRtpPacketTransport(&fake_rtp);
   fake_rtp.SetDestination(&fake_rtp, true);

pc/srtp_transport_unittest.cc

@@ -63,6 +63,10 @@ class SrtpTransportTest : public ::testing::Test, public sigslot::has_slots<> {
     srtp_transport2_ =
         std::make_unique<SrtpTransport>(rtcp_mux_enabled, field_trials_);
 
+    // RingRTC change to drop all incoming packets until explicitly allowed
+    srtp_transport1_->SetIncomingRtpEnabled(true);
+    srtp_transport2_->SetIncomingRtpEnabled(true);
+
     srtp_transport1_->SetRtpPacketTransport(rtp_packet_transport1_.get());
     srtp_transport2_->SetRtpPacketTransport(rtp_packet_transport2_.get());
 
@@ -433,6 +437,9 @@ TEST_F(SrtpTransportTest, RemoveSrtpReceiveStream) {
   auto rtp_packet_transport = std::make_unique<rtc::FakePacketTransport>(
       "fake_packet_transport_loopback");
 
+  // RingRTC change to drop all incoming packets until explicitly allowed
+  srtp_transport->SetIncomingRtpEnabled(true);
+
   bool asymmetric = false;
   rtp_packet_transport->SetDestination(rtp_packet_transport.get(), asymmetric);
   srtp_transport->SetRtpPacketTransport(rtp_packet_transport.get());

ringrtc/rffi/src/injectable_network.cc

@@ -320,8 +320,6 @@ class InjectableNetworkImpl : public InjectableNetwork, public rtc::NetworkManag
   rtc::AsyncPacketSocket* CreateClientTcpSocket(
       const rtc::SocketAddress& local_address,
       const rtc::SocketAddress& remote_address,
-      const rtc::ProxyInfo& proxy_info,
-      const std::string& user_agent,
       const rtc::PacketSocketTcpOptions& tcp_options) override {
     // TODO: Support TCP for TURN
     return nullptr;

ringrtc/rffi/src/peer_connection.cc

@@ -393,7 +393,7 @@ Rust_sessionDescriptionFromV4(bool offer,
     audio->AddCodec(opus_red);
   }
 
-  auto add_video_feedback_params = [] (cricket::VideoCodec* video_codec) {
+  auto add_video_feedback_params = [] (cricket::Codec* video_codec) {
     video_codec->AddFeedbackParam(cricket::FeedbackParam(cricket::kRtcpFbParamTransportCc, cricket::kParamValueEmpty));
     video_codec->AddFeedbackParam(cricket::FeedbackParam(cricket::kRtcpFbParamCcm, cricket::kRtcpFbCcmParamFir));
     video_codec->AddFeedbackParam(cricket::FeedbackParam(cricket::kRtcpFbParamNack, cricket::kParamValueEmpty));
@@ -618,7 +618,7 @@ CreateSessionDescriptionForGroupCall(bool local,
     remote_audio->AddCodec(opus_red);
   }
 
-  auto add_video_feedback_params = [] (cricket::VideoCodec* video_codec) {
+  auto add_video_feedback_params = [] (cricket::Codec* video_codec) {
     video_codec->AddFeedbackParam(cricket::FeedbackParam(cricket::kRtcpFbParamTransportCc, cricket::kParamValueEmpty));
     video_codec->AddFeedbackParam(cricket::FeedbackParam(cricket::kRtcpFbParamCcm, cricket::kRtcpFbCcmParamFir));
     video_codec->AddFeedbackParam(cricket::FeedbackParam(cricket::kRtcpFbParamNack, cricket::kParamValueEmpty));
@@ -649,7 +649,7 @@ CreateSessionDescriptionForGroupCall(bool local,
     remote_video->AddCodec(red_rtx);
   }
 
-  auto audio_level = webrtc::RtpExtension(webrtc::AudioLevel::Uri(), AUDIO_LEVEL_EXT_ID);
+  auto audio_level = webrtc::RtpExtension(webrtc::AudioLevelExtension::Uri(), AUDIO_LEVEL_EXT_ID);
   // Note: Do not add transport-cc for audio.  Using transport-cc with audio is still experimental in WebRTC.
   // And don't add abs_send_time because it's only used for video.
   local_audio->AddRtpHeaderExtension(audio_level);
@@ -973,7 +973,7 @@ Rust_addIceCandidateFromServer(PeerConnectionInterface* pc_borrowed_rc,
   // WebRTC generally prefers IPv6 over IPv4 for local candidates (see rtc_base::IPAddressPrecedence).
   // So we leave the priority unset to allow the local candidate preference to break the tie.
   candidate.set_component(cricket::ICE_CANDIDATE_COMPONENT_RTP);
-  candidate.set_type(cricket::LOCAL_PORT_TYPE);  // AKA "host"
+  candidate.set_type(webrtc::IceCandidateType::kHost);
   candidate.set_address(rtc::SocketAddress(IpToRtcIp(ip), port));
   candidate.set_protocol(tcp ? cricket::TCP_PROTOCOL_NAME : cricket::UDP_PROTOCOL_NAME);
 

ringrtc/rffi/src/peer_connection_factory.cc

@@ -5,6 +5,7 @@
 
 #include "api/create_peerconnection_factory.h"
 #include "api/enable_media.h"
+#include "api/environment/environment.h"
 #include "api/task_queue/default_task_queue_factory.h"
 #include "api/rtc_event_log/rtc_event_log_factory.h"
 #include "api/audio_codecs/builtin_audio_decoder_factory.h"
@@ -49,7 +50,8 @@ class RingRTCVideoEncoderFactory : public VideoEncoderFactory {
     return factory_.GetSupportedFormats();
   }
 
-  std::unique_ptr<VideoEncoder> CreateVideoEncoder(
+  std::unique_ptr<VideoEncoder> Create(
+      const Environment& env,
       const SdpVideoFormat& format) override {
     if (format.IsCodecInList(
         factory_.GetSupportedFormats())) {
@@ -60,7 +62,7 @@ class RingRTCVideoEncoderFactory : public VideoEncoderFactory {
         // The adapter has a passthrough mode for the case that simulcast is not
         // used, so all responsibility can be delegated to it.
         return std::make_unique<SimulcastEncoderAdapter>(
-            &factory_, *original_format);
+            env, &factory_, nullptr, *original_format);
       }
     }
     return nullptr;

ringrtc/rffi/src/peer_connection_observer.cc

@@ -35,7 +35,7 @@ void PeerConnectionObserverRffi::OnIceCandidate(const IceCandidateInterface* can
   candidate->ToString(&sdp);
   rust_candidate.sdp_borrowed = sdp.c_str();
 
-  rust_candidate.is_relayed = (candidate->candidate().type() == cricket::RELAY_PORT_TYPE);
+  rust_candidate.is_relayed = (candidate->candidate().type() == IceCandidateType::kRelay);
   rust_candidate.relay_protocol = TransportProtocol::kUnknown;
   if (candidate->candidate().relay_protocol() == cricket::UDP_PROTOCOL_NAME) {
     rust_candidate.relay_protocol = TransportProtocol::kUdp;
@@ -95,7 +95,7 @@ void PeerConnectionObserverRffi::OnIceSelectedCandidatePairChanged(
   auto& remote = event.selected_candidate_pair.remote_candidate();
   auto local_adapter_type = local.network_type();
   auto local_adapter_type_under_vpn = local.underlying_type_for_vpn();
-  bool local_relayed = (local.type() == cricket::RELAY_PORT_TYPE) || !local.relay_protocol().empty();
+  bool local_relayed = (local.type() == IceCandidateType::kRelay) || !local.relay_protocol().empty();
   TransportProtocol local_relay_protocol = TransportProtocol::kUnknown;
   if (local.relay_protocol() == cricket::UDP_PROTOCOL_NAME) {
     local_relay_protocol = TransportProtocol::kUdp;
@@ -104,7 +104,7 @@ void PeerConnectionObserverRffi::OnIceSelectedCandidatePairChanged(
   } else if (local.relay_protocol() == cricket::TLS_PROTOCOL_NAME) {
     local_relay_protocol = TransportProtocol::kTls;
   }
-  bool remote_relayed = (remote.type() == cricket::RELAY_PORT_TYPE);
+  bool remote_relayed = (remote.type() == IceCandidateType::kRelay);
   auto network_route = webrtc::rffi::NetworkRoute{ local_adapter_type, local_adapter_type_under_vpn, local_relayed, local_relay_protocol, remote_relayed};
 
   callbacks_.onIceNetworkRouteChange(observer_, network_route, SdpSerializeCandidate(local).c_str(), SdpSerializeCandidate(remote).c_str());

sdk/android/api/org/webrtc/PeerConnection.java

@@ -843,11 +843,12 @@ public class PeerConnection {
       return offerExtmapAllowMixed;
     }
 
-    @CalledByNative("RTCConfiguration")
-    @PortAllocatorFlags
-    int getPortAllocatorFlags() {
-      return portAllocatorFlags;
-    }
+    // RingRTC change to use default port allocator flags
+    // @CalledByNative("RTCConfiguration")
+    // @PortAllocatorFlags
+    // int getPortAllocatorFlags() {
+    //   return portAllocatorFlags;
+    // }
   };
 
   private final List<MediaStream> localStreams = new ArrayList<>();

sdk/android/src/jni/pc/peer_connection.cc

@@ -280,8 +280,9 @@ void JavaToNativeRTCConfiguration(
     rtc_config->turn_logging_id = JavaToNativeString(jni, j_turn_logging_id);
   }
 
-  rtc_config->port_allocator_config.flags =
-    Java_RTCConfiguration_getPortAllocatorFlags(jni, j_rtc_config);
+  // RingRTC change to use default port allocator flags
+  // rtc_config->port_allocator_config.flags =
+  //   Java_RTCConfiguration_getPortAllocatorFlags(jni, j_rtc_config);
 }
 
 rtc::KeyType GetRtcConfigKeyType(JNIEnv* env,

video/encoder_rtcp_feedback.cc

@@ -37,7 +37,8 @@ EncoderRtcpFeedback::EncoderRtcpFeedback(
       per_layer_keyframes_(per_layer_keyframes),
       get_packet_infos_(std::move(get_packet_infos)),
       video_stream_encoder_(encoder),
-      time_last_packet_delivery_queue_(per_layer_keyframes ? ssrcs.size() : 1,
+      // RingRTC change to enable per-layer PLI for screen sharing
+      time_last_packet_delivery_queue_(ssrcs.size(),
                                        Timestamp::Zero()),
       min_keyframe_send_interval_(
           TimeDelta::Millis(KeyframeIntervalSettings::ParseFromFieldTrials()