m122 merge fixes

modules/audio_coding/neteq/packet_buffer.cc

@@ -104,7 +104,6 @@ void PacketBuffer::Flush() {
       << ", num_gaps_below_40ms=" << num_gaps_below_40ms
       << ", num_gaps_above_90ms=" << num_gaps_above_90ms
       << ", num_no_packet_info=" << num_no_packet_info;
-
   }
   buffer_.clear();
   stats_->FlushedPacketBuffer();
@@ -129,9 +128,12 @@ int PacketBuffer::InsertPacket(Packet&& packet) {
 
   if (buffer_.size() >= max_number_of_packets_) {
     // Buffer is full.
+    // RingRTC change to log more information around audio jitter buffer flushes
+    size_t buffer_size_before_flush = buffer_.size();
     Flush();
     return_val = kFlushed;
-    RTC_LOG(LS_WARNING) << "Packet buffer flushed.";
+    RTC_LOG(LS_WARNING) << "Packet buffer flushed"
+                        << ", packets discarded=" << buffer_size_before_flush;
   }
 
   // Get an iterator pointing to the place in the buffer where the new packet

p2p/client/basic_port_allocator.h

@@ -332,7 +332,8 @@ class TurnPort;
 // Performs the allocation of ports, in a sequenced (timed) manner, for a given
 // network and IP address.
 // This class is thread-compatible.
-class AllocationSequence {
+// RingRTC change to support ICE forking
+class AllocationSequence : public sigslot::has_slots<> {
  public:
   enum State {
     kInit,       // Initial state.

pc/peer_connection_factory.cc

@@ -248,7 +248,7 @@ PeerConnectionFactory::CreatePeerConnectionOrError(
     // RingRTC change to control whether PORTALLOCATOR_ENABLE_ANY_ADDRESS_PORTS
     // is enabled.
     uint32_t port_allocator_flags = configuration.port_allocator_config.flags;
-    if (trials->IsEnabled("RingRTC-AnyAddressPortsKillSwitch")) {
+    if (env.field_trials().IsEnabled("RingRTC-AnyAddressPortsKillSwitch")) {
       port_allocator_flags &= ~cricket::PORTALLOCATOR_ENABLE_ANY_ADDRESS_PORTS;
     }
     dependencies.allocator->set_flags(

ringrtc/rffi/src/injectable_network.cc

@@ -9,6 +9,7 @@
 #include "p2p/client/basic_port_allocator.h"
 #include "rffi/api/network.h"
 #include "rtc_base/ip_address.h"
+#include "rtc_base/network/received_packet.h"
 
 namespace webrtc {
 
@@ -67,8 +68,10 @@ class InjectableUdpSocket : public rtc::AsyncPacketSocket {
     RTC_LOG(LS_VERBOSE) << "InjectableUdpSocket::ReceiveFrom()"
                         << " from " << remote_address.ToString()
                         << " to " << local_address_.ToString();
-    auto now = rtc::TimeMicros();
-    SignalReadPacket(this, reinterpret_cast<const char*>(data), data_size, remote_address, now);
+    NotifyPacketReceived(rtc::ReceivedPacket::CreateFromLegacy(reinterpret_cast<const char*>(data),
+                                                               data_size,
+                                                               rtc::TimeMicros(),
+                                                               remote_address));
   }
 
   // As rtc::AsyncPacketSocket
@@ -191,10 +194,10 @@ class InjectableNetworkImpl : public InjectableNetwork, public rtc::NetworkManag
         [this, source, dest, data{std::vector<uint8_t>(data, data+size)}, size] { 
       auto local_address = IpPortToRtcSocketAddress(dest);
       auto remote_address = IpPortToRtcSocketAddress(source);
-     RTC_LOG(LS_VERBOSE) << "InjectableNetworkImpl::ReceiveUdp()"
-                         << " from " << remote_address.ToString()
-                         << " to " << local_address.ToString()
-                         << " size: " << size;
+      RTC_LOG(LS_VERBOSE) << "InjectableNetworkImpl::ReceiveUdp()"
+                          << " from " << remote_address.ToString()
+                          << " to " << local_address.ToString()
+                          << " size: " << size;
       auto udp_socket = udp_socket_by_local_address_.find(local_address);
       if (udp_socket == udp_socket_by_local_address_.end()) {
         RTC_LOG(LS_WARNING) << "Received packet for unknown local address.";
@@ -380,6 +383,3 @@ RUSTEXPORT void Rust_InjectableNetwork_ReceiveUdp(
 }  // namespace rffi
 
 }  // namespace webrtc
-
-
-

ringrtc/rffi/src/peer_connection_factory.cc

@@ -4,7 +4,7 @@
  */
 
 #include "api/create_peerconnection_factory.h"
-#include "api/call/call_factory_interface.h"
+#include "api/enable_media.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"
@@ -16,7 +16,6 @@
 #include "api/video_codecs/video_encoder_factory_template_libvpx_vp8_adapter.h"
 #include "api/video_codecs/video_encoder_factory_template_libvpx_vp9_adapter.h"
 #include "media/engine/simulcast_encoder_adapter.h"
-#include "media/engine/webrtc_media_engine.h"
 #include "modules/audio_mixer/audio_mixer_impl.h"
 #include "modules/audio_device/dummy/file_audio_device_factory.h"
 #include "modules/audio_processing/include/audio_processing.h"
@@ -118,10 +117,7 @@ class PeerConnectionFactoryWithOwnedThreads
     dependencies.worker_thread = worker_thread.get();
     dependencies.signaling_thread = signaling_thread.get();
     dependencies.task_queue_factory = CreateDefaultTaskQueueFactory();
-    dependencies.call_factory = CreateCallFactory();
-    dependencies.event_log_factory = std::make_unique<RtcEventLogFactory>(dependencies.task_queue_factory.get());
-    cricket::MediaEngineDependencies media_dependencies;
-    media_dependencies.task_queue_factory = dependencies.task_queue_factory.get();
+    dependencies.event_log_factory = std::make_unique<RtcEventLogFactory>();
 
 #if defined(WEBRTC_WIN)
     std::unique_ptr<ScopedCOMInitializer> com_initializer;
@@ -149,9 +145,10 @@ class PeerConnectionFactoryWithOwnedThreads
           AudioDeviceModule::kPlatformDefaultAudio, dependencies.task_queue_factory.get());
       }
     });
-    media_dependencies.adm = adm;
-    media_dependencies.audio_encoder_factory = CreateBuiltinAudioEncoderFactory();
-    media_dependencies.audio_decoder_factory = CreateBuiltinAudioDecoderFactory();
+
+    dependencies.adm = adm;
+    dependencies.audio_encoder_factory = CreateBuiltinAudioEncoderFactory();
+    dependencies.audio_decoder_factory = CreateBuiltinAudioDecoderFactory();
 
     AudioProcessing::Config config;
     config.high_pass_filter.enabled = audio_config.high_pass_filter_enabled;
@@ -159,17 +156,18 @@ class PeerConnectionFactoryWithOwnedThreads
     config.noise_suppression.enabled = audio_config.ns_enabled;
     config.gain_controller1.enabled = audio_config.agc_enabled;
 
-    media_dependencies.audio_processing = AudioProcessingBuilder()
+    dependencies.audio_processing = AudioProcessingBuilder()
       .SetConfig(config)
       .Create();
+    dependencies.audio_mixer = AudioMixerImpl::Create();
 
-    media_dependencies.audio_mixer = AudioMixerImpl::Create();
-    media_dependencies.video_encoder_factory =
+    dependencies.video_encoder_factory =
         std::make_unique<RingRTCVideoEncoderFactory>();
-    media_dependencies.video_decoder_factory =
+    dependencies.video_decoder_factory =
         std::make_unique<VideoDecoderFactoryTemplate<
             LibvpxVp8DecoderTemplateAdapter, LibvpxVp9DecoderTemplateAdapter>>();
-    dependencies.media_engine = cricket::CreateMediaEngine(std::move(media_dependencies));
+
+    EnableMedia(dependencies);
 
     auto factory = CreateModularPeerConnectionFactory(std::move(dependencies));
     return rtc::make_ref_counted<PeerConnectionFactoryWithOwnedThreads>(

ringrtc/rffi/src/stats_observer.cc

@@ -42,7 +42,7 @@ void StatsObserverRffi::OnStatsDelivered(const rtc::scoped_refptr<const RTCStats
       audio_sender.packets_sent = stat->packets_sent.ValueOrDefault(0);
       audio_sender.bytes_sent = stat->bytes_sent.ValueOrDefault(0);
 
-      if (stat->remote_id.is_defined()) {
+      if (stat->remote_id.has_value()) {
         auto remote_stat = report->GetAs<RTCRemoteInboundRtpStreamStats>(*stat->remote_id);
         if (remote_stat) {
           audio_sender.remote_packets_lost = remote_stat->packets_lost.ValueOrDefault(0);
@@ -51,7 +51,7 @@ void StatsObserverRffi::OnStatsDelivered(const rtc::scoped_refptr<const RTCStats
         }
       }
 
-      if (stat->media_source_id.is_defined()) {
+      if (stat->media_source_id.has_value()) {
         auto audio_source_stat = report->GetAs<RTCAudioSourceStats>(*stat->media_source_id);
         if (audio_source_stat) {
           audio_sender.total_audio_energy = audio_source_stat->total_audio_energy.ValueOrDefault(0.0);
@@ -75,7 +75,7 @@ void StatsObserverRffi::OnStatsDelivered(const rtc::scoped_refptr<const RTCStats
       video_sender.total_packet_send_delay = stat->total_packet_send_delay.ValueOrDefault(0.0);
       video_sender.nack_count = stat->nack_count.ValueOrDefault(0);
       video_sender.pli_count = stat->pli_count.ValueOrDefault(0);
-      if (stat->quality_limitation_reason.is_defined()) {
+      if (stat->quality_limitation_reason.has_value()) {
         // "none" = 0 (the default)
         if (*stat->quality_limitation_reason == "cpu") {
           video_sender.quality_limitation_reason = 1;
@@ -87,7 +87,7 @@ void StatsObserverRffi::OnStatsDelivered(const rtc::scoped_refptr<const RTCStats
       }
       video_sender.quality_limitation_resolution_changes = stat->quality_limitation_resolution_changes.ValueOrDefault(0);
 
-      if (stat->remote_id.is_defined()) {
+      if (stat->remote_id.has_value()) {
         auto remote_stat = report->GetAs<RTCRemoteInboundRtpStreamStats>(*stat->remote_id);
         if (remote_stat) {
           video_sender.remote_packets_lost = remote_stat->packets_lost.ValueOrDefault(0);

video/video_receive_stream2.cc

@@ -905,7 +905,7 @@ int VideoReceiveStream2::DecodeAndMaybeDispatchEncodedFrame(
   if (decode_result < WEBRTC_VIDEO_CODEC_OK) {
     // Asynchronous decoders may delay error reporting, potentially resulting in
     // error reports reflecting issues that occurred several frames back.
-    RTC_LOG(LS_WARNING)
+    RTC_LOG(LS_INFO)
         << "Failed to decode frame. Return code: " << decode_result
         << ", SSRC: " << remote_ssrc()
         << ", frame RTP timestamp: " << frame_ptr->RtpTimestamp()