diff --git a/logging/BUILD.gn b/logging/BUILD.gn
index 571ddf8..d3f821a 100644
--- a/logging/BUILD.gn
+++ b/logging/BUILD.gn
@@ -160,25 +160,11 @@
     "rtc_event_log/encoder/blob_encoding.h",
     "rtc_event_log/encoder/delta_encoding.cc",
     "rtc_event_log/encoder/delta_encoding.h",
-    "rtc_event_log/encoder/rtc_event_log_encoder_legacy.cc",
-    "rtc_event_log/encoder/rtc_event_log_encoder_legacy.h",
   ]
 
   defines = []
 
   deps = [
-    ":ice_log",
-    ":rtc_event_audio",
-    ":rtc_event_bwe",
-    ":rtc_event_log_api",
-    ":rtc_event_log_impl_output",
-    ":rtc_event_pacing",
-    ":rtc_event_rtp_rtcp",
-    ":rtc_event_video",
-    ":rtc_stream_config",
-    "../modules/audio_coding:audio_network_adaptor",
-    "../modules/remote_bitrate_estimator:remote_bitrate_estimator",
-    "../modules/rtp_rtcp:rtp_rtcp_format",
     "../rtc_base:checks",
     "../rtc_base:rtc_base_approved",
     "//third_party/abseil-cpp/absl/memory",
@@ -188,7 +174,29 @@
 
   if (rtc_enable_protobuf) {
     defines += [ "ENABLE_RTC_EVENT_LOG" ]
-    deps += [ ":rtc_event_log_proto" ]
+    deps += [
+      ":ice_log",
+      ":rtc_event_audio",
+      ":rtc_event_bwe",
+      ":rtc_event_log2_proto",
+      ":rtc_event_log_api",
+      ":rtc_event_log_impl_output",
+      ":rtc_event_log_proto",
+      ":rtc_event_pacing",
+      ":rtc_event_rtp_rtcp",
+      ":rtc_event_video",
+      ":rtc_stream_config",
+      "../api:array_view",
+      "../modules/audio_coding:audio_network_adaptor",
+      "../modules/remote_bitrate_estimator:remote_bitrate_estimator",
+      "../modules/rtp_rtcp:rtp_rtcp_format",
+    ]
+    sources += [
+      "rtc_event_log/encoder/rtc_event_log_encoder_legacy.cc",
+      "rtc_event_log/encoder/rtc_event_log_encoder_legacy.h",
+      "rtc_event_log/encoder/rtc_event_log_encoder_new_format.cc",
+      "rtc_event_log/encoder/rtc_event_log_encoder_new_format.h",
+    ]
   }
 }
 
@@ -219,8 +227,7 @@
   deps = [
     ":ice_log",
     ":rtc_event_log_api",
-    ":rtc_event_log_impl_encoder",
-    ":rtc_event_log_impl_output",
+    "../api:libjingle_logging_api",
     "../rtc_base:checks",
     "../rtc_base:rtc_base_approved",
     "../rtc_base:rtc_task_queue",
@@ -231,7 +238,7 @@
 
   if (rtc_enable_protobuf) {
     defines += [ "ENABLE_RTC_EVENT_LOG" ]
-    deps += [ ":rtc_event_log_proto" ]
+    deps += [ ":rtc_event_log_impl_encoder" ]
   }
 }
 
@@ -319,6 +326,7 @@
         ":ice_log",
         ":rtc_event_audio",
         ":rtc_event_bwe",
+        ":rtc_event_log2_proto",
         ":rtc_event_log_api",
         ":rtc_event_log_impl_base",
         ":rtc_event_log_impl_encoder",
diff --git a/logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.cc b/logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.cc
index 1db2bf8..618c861 100644
--- a/logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.cc
+++ b/logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.cc
@@ -47,7 +47,6 @@
 #include "rtc_base/ignore_wundef.h"
 #include "rtc_base/logging.h"
 
-#ifdef ENABLE_RTC_EVENT_LOG
 
 // *.pb.h files are generated at build-time by the protobuf compiler.
 RTC_PUSH_IGNORING_WUNDEF()
@@ -756,5 +755,3 @@
 }
 
 }  // namespace webrtc
-
-#endif  // ENABLE_RTC_EVENT_LOG
diff --git a/logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.h b/logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.h
index 87db039..fd109f4 100644
--- a/logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.h
+++ b/logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.h
@@ -18,8 +18,6 @@
 #include "logging/rtc_event_log/encoder/rtc_event_log_encoder.h"
 #include "rtc_base/buffer.h"
 
-#if defined(ENABLE_RTC_EVENT_LOG)
-
 namespace webrtc {
 
 namespace rtclog {
@@ -105,6 +103,4 @@
 
 }  // namespace webrtc
 
-#endif  // ENABLE_RTC_EVENT_LOG
-
 #endif  // LOGGING_RTC_EVENT_LOG_ENCODER_RTC_EVENT_LOG_ENCODER_LEGACY_H_
diff --git a/logging/rtc_event_log/encoder/rtc_event_log_encoder_new_format.cc b/logging/rtc_event_log/encoder/rtc_event_log_encoder_new_format.cc
new file mode 100644
index 0000000..8b2199b
--- /dev/null
+++ b/logging/rtc_event_log/encoder/rtc_event_log_encoder_new_format.cc
@@ -0,0 +1,875 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "logging/rtc_event_log/encoder/rtc_event_log_encoder_new_format.h"
+
+#include <vector>
+
+#include "api/array_view.h"
+#include "logging/rtc_event_log/events/rtc_event_alr_state.h"
+#include "logging/rtc_event_log/events/rtc_event_audio_network_adaptation.h"
+#include "logging/rtc_event_log/events/rtc_event_audio_playout.h"
+#include "logging/rtc_event_log/events/rtc_event_audio_receive_stream_config.h"
+#include "logging/rtc_event_log/events/rtc_event_audio_send_stream_config.h"
+#include "logging/rtc_event_log/events/rtc_event_bwe_update_delay_based.h"
+#include "logging/rtc_event_log/events/rtc_event_bwe_update_loss_based.h"
+#include "logging/rtc_event_log/events/rtc_event_ice_candidate_pair.h"
+#include "logging/rtc_event_log/events/rtc_event_ice_candidate_pair_config.h"
+#include "logging/rtc_event_log/events/rtc_event_probe_cluster_created.h"
+#include "logging/rtc_event_log/events/rtc_event_probe_result_failure.h"
+#include "logging/rtc_event_log/events/rtc_event_probe_result_success.h"
+#include "logging/rtc_event_log/events/rtc_event_rtcp_packet_incoming.h"
+#include "logging/rtc_event_log/events/rtc_event_rtcp_packet_outgoing.h"
+#include "logging/rtc_event_log/events/rtc_event_rtp_packet_incoming.h"
+#include "logging/rtc_event_log/events/rtc_event_rtp_packet_outgoing.h"
+#include "logging/rtc_event_log/events/rtc_event_video_receive_stream_config.h"
+#include "logging/rtc_event_log/events/rtc_event_video_send_stream_config.h"
+#include "logging/rtc_event_log/rtc_stream_config.h"
+#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
+#include "modules/remote_bitrate_estimator/include/bwe_defines.h"
+#include "modules/rtp_rtcp/include/rtp_cvo.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/app.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/bye.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/common_header.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/extended_reports.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/psfb.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/rtpfb.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/sdes.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
+#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
+#include "modules/rtp_rtcp/source/rtp_packet.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/ignore_wundef.h"
+#include "rtc_base/logging.h"
+
+// *.pb.h files are generated at build-time by the protobuf compiler.
+RTC_PUSH_IGNORING_WUNDEF()
+#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
+#include "external/webrtc/webrtc/logging/rtc_event_log/rtc_event_log2.pb.h"
+#else
+#include "logging/rtc_event_log/rtc_event_log2.pb.h"
+#endif
+RTC_POP_IGNORING_WUNDEF()
+
+namespace webrtc {
+
+namespace {
+rtclog2::DelayBasedBweUpdates::DetectorState ConvertToProtoFormat(
+    BandwidthUsage state) {
+  switch (state) {
+    case BandwidthUsage::kBwNormal:
+      return rtclog2::DelayBasedBweUpdates::BWE_NORMAL;
+    case BandwidthUsage::kBwUnderusing:
+      return rtclog2::DelayBasedBweUpdates::BWE_UNDERUSING;
+    case BandwidthUsage::kBwOverusing:
+      return rtclog2::DelayBasedBweUpdates::BWE_OVERUSING;
+    case BandwidthUsage::kLast:
+      RTC_NOTREACHED();
+  }
+  RTC_NOTREACHED();
+  return rtclog2::DelayBasedBweUpdates::BWE_UNKNOWN_STATE;
+}
+
+rtclog2::BweProbeResultFailure::FailureReason ConvertToProtoFormat(
+    ProbeFailureReason failure_reason) {
+  switch (failure_reason) {
+    case ProbeFailureReason::kInvalidSendReceiveInterval:
+      return rtclog2::BweProbeResultFailure::INVALID_SEND_RECEIVE_INTERVAL;
+    case ProbeFailureReason::kInvalidSendReceiveRatio:
+      return rtclog2::BweProbeResultFailure::INVALID_SEND_RECEIVE_RATIO;
+    case ProbeFailureReason::kTimeout:
+      return rtclog2::BweProbeResultFailure::TIMEOUT;
+    case ProbeFailureReason::kLast:
+      RTC_NOTREACHED();
+  }
+  RTC_NOTREACHED();
+  return rtclog2::BweProbeResultFailure::UNKNOWN;
+}
+
+// Returns true if there are recognized extensions that we should log
+// and false if there are no extensions or all extensions are types we don't
+// log. The protobuf representation of the header configs is written to
+// |proto_config|.
+bool ConvertToProtoFormat(const std::vector<RtpExtension>& extensions,
+                          rtclog2::RtpHeaderExtensionConfig* proto_config) {
+  size_t unknown_extensions = 0;
+  for (auto& extension : extensions) {
+    if (extension.uri == RtpExtension::kAudioLevelUri) {
+      proto_config->set_audio_level_id(extension.id);
+    } else if (extension.uri == RtpExtension::kTimestampOffsetUri) {
+      proto_config->set_transmission_time_offset_id(extension.id);
+    } else if (extension.uri == RtpExtension::kAbsSendTimeUri) {
+      proto_config->set_absolute_send_time_id(extension.id);
+    } else if (extension.uri == RtpExtension::kTransportSequenceNumberUri) {
+      proto_config->set_transport_sequence_number_id(extension.id);
+    } else if (extension.uri == RtpExtension::kVideoRotationUri) {
+      proto_config->set_video_rotation_id(extension.id);
+    } else {
+      ++unknown_extensions;
+    }
+  }
+  return unknown_extensions < extensions.size();
+}
+
+rtclog2::IceCandidatePairConfig::IceCandidatePairConfigType
+ConvertToProtoFormat(IceCandidatePairConfigType type) {
+  switch (type) {
+    case IceCandidatePairConfigType::kAdded:
+      return rtclog2::IceCandidatePairConfig::ADDED;
+    case IceCandidatePairConfigType::kUpdated:
+      return rtclog2::IceCandidatePairConfig::UPDATED;
+    case IceCandidatePairConfigType::kDestroyed:
+      return rtclog2::IceCandidatePairConfig::DESTROYED;
+    case IceCandidatePairConfigType::kSelected:
+      return rtclog2::IceCandidatePairConfig::SELECTED;
+    case IceCandidatePairConfigType::kNumValues:
+      RTC_NOTREACHED();
+  }
+  RTC_NOTREACHED();
+  return rtclog2::IceCandidatePairConfig::UNKNOWN_CONFIG_TYPE;
+}
+
+rtclog2::IceCandidatePairConfig::IceCandidateType ConvertToProtoFormat(
+    IceCandidateType type) {
+  switch (type) {
+    case IceCandidateType::kUnknown:
+      return rtclog2::IceCandidatePairConfig::UNKNOWN_CANDIDATE_TYPE;
+    case IceCandidateType::kLocal:
+      return rtclog2::IceCandidatePairConfig::LOCAL;
+    case IceCandidateType::kStun:
+      return rtclog2::IceCandidatePairConfig::STUN;
+    case IceCandidateType::kPrflx:
+      return rtclog2::IceCandidatePairConfig::PRFLX;
+    case IceCandidateType::kRelay:
+      return rtclog2::IceCandidatePairConfig::RELAY;
+    case IceCandidateType::kNumValues:
+      RTC_NOTREACHED();
+  }
+  RTC_NOTREACHED();
+  return rtclog2::IceCandidatePairConfig::UNKNOWN_CANDIDATE_TYPE;
+}
+
+rtclog2::IceCandidatePairConfig::Protocol ConvertToProtoFormat(
+    IceCandidatePairProtocol protocol) {
+  switch (protocol) {
+    case IceCandidatePairProtocol::kUnknown:
+      return rtclog2::IceCandidatePairConfig::UNKNOWN_PROTOCOL;
+    case IceCandidatePairProtocol::kUdp:
+      return rtclog2::IceCandidatePairConfig::UDP;
+    case IceCandidatePairProtocol::kTcp:
+      return rtclog2::IceCandidatePairConfig::TCP;
+    case IceCandidatePairProtocol::kSsltcp:
+      return rtclog2::IceCandidatePairConfig::SSLTCP;
+    case IceCandidatePairProtocol::kTls:
+      return rtclog2::IceCandidatePairConfig::TLS;
+    case IceCandidatePairProtocol::kNumValues:
+      RTC_NOTREACHED();
+  }
+  RTC_NOTREACHED();
+  return rtclog2::IceCandidatePairConfig::UNKNOWN_PROTOCOL;
+}
+
+rtclog2::IceCandidatePairConfig::AddressFamily ConvertToProtoFormat(
+    IceCandidatePairAddressFamily address_family) {
+  switch (address_family) {
+    case IceCandidatePairAddressFamily::kUnknown:
+      return rtclog2::IceCandidatePairConfig::UNKNOWN_ADDRESS_FAMILY;
+    case IceCandidatePairAddressFamily::kIpv4:
+      return rtclog2::IceCandidatePairConfig::IPV4;
+    case IceCandidatePairAddressFamily::kIpv6:
+      return rtclog2::IceCandidatePairConfig::IPV6;
+    case IceCandidatePairAddressFamily::kNumValues:
+      RTC_NOTREACHED();
+  }
+  RTC_NOTREACHED();
+  return rtclog2::IceCandidatePairConfig::UNKNOWN_ADDRESS_FAMILY;
+}
+
+rtclog2::IceCandidatePairConfig::NetworkType ConvertToProtoFormat(
+    IceCandidateNetworkType network_type) {
+  switch (network_type) {
+    case IceCandidateNetworkType::kUnknown:
+      return rtclog2::IceCandidatePairConfig::UNKNOWN_NETWORK_TYPE;
+    case IceCandidateNetworkType::kEthernet:
+      return rtclog2::IceCandidatePairConfig::ETHERNET;
+    case IceCandidateNetworkType::kLoopback:
+      return rtclog2::IceCandidatePairConfig::LOOPBACK;
+    case IceCandidateNetworkType::kWifi:
+      return rtclog2::IceCandidatePairConfig::WIFI;
+    case IceCandidateNetworkType::kVpn:
+      return rtclog2::IceCandidatePairConfig::VPN;
+    case IceCandidateNetworkType::kCellular:
+      return rtclog2::IceCandidatePairConfig::CELLULAR;
+    case IceCandidateNetworkType::kNumValues:
+      RTC_NOTREACHED();
+  }
+  RTC_NOTREACHED();
+  return rtclog2::IceCandidatePairConfig::UNKNOWN_NETWORK_TYPE;
+}
+
+rtclog2::IceCandidatePairEvent::IceCandidatePairEventType ConvertToProtoFormat(
+    IceCandidatePairEventType type) {
+  switch (type) {
+    case IceCandidatePairEventType::kCheckSent:
+      return rtclog2::IceCandidatePairEvent::CHECK_SENT;
+    case IceCandidatePairEventType::kCheckReceived:
+      return rtclog2::IceCandidatePairEvent::CHECK_RECEIVED;
+    case IceCandidatePairEventType::kCheckResponseSent:
+      return rtclog2::IceCandidatePairEvent::CHECK_RESPONSE_SENT;
+    case IceCandidatePairEventType::kCheckResponseReceived:
+      return rtclog2::IceCandidatePairEvent::CHECK_RESPONSE_RECEIVED;
+    case IceCandidatePairEventType::kNumValues:
+      RTC_NOTREACHED();
+  }
+  RTC_NOTREACHED();
+  return rtclog2::IceCandidatePairEvent::UNKNOWN_CHECK_TYPE;
+}
+
+// Copies all RTCP blocks except APP, SDES and unknown from |packet| to
+// |buffer|. |buffer| must have space for |IP_PACKET_SIZE| bytes. |packet| must
+// be at most |IP_PACKET_SIZE| bytes long.
+size_t RemoveNonWhitelistedRtcpBlocks(const rtc::Buffer& packet,
+                                      uint8_t* buffer) {
+  RTC_DCHECK(packet.size() <= IP_PACKET_SIZE);
+  RTC_DCHECK(buffer != nullptr);
+  rtcp::CommonHeader header;
+  const uint8_t* block_begin = packet.data();
+  const uint8_t* packet_end = packet.data() + packet.size();
+  size_t buffer_length = 0;
+  while (block_begin < packet_end) {
+    if (!header.Parse(block_begin, packet_end - block_begin)) {
+      break;  // Incorrect message header.
+    }
+    const uint8_t* next_block = header.NextPacket();
+    RTC_DCHECK_GT(next_block, block_begin);
+    RTC_DCHECK_LE(next_block, packet_end);
+    size_t block_size = next_block - block_begin;
+    switch (header.type()) {
+      case rtcp::Bye::kPacketType:
+      case rtcp::ExtendedJitterReport::kPacketType:
+      case rtcp::ExtendedReports::kPacketType:
+      case rtcp::Psfb::kPacketType:
+      case rtcp::ReceiverReport::kPacketType:
+      case rtcp::Rtpfb::kPacketType:
+      case rtcp::SenderReport::kPacketType:
+        // We log sender reports, receiver reports, bye messages
+        // inter-arrival jitter, third-party loss reports, payload-specific
+        // feedback and extended reports.
+        // TODO(terelius): As an optimization, don't copy anything if all blocks
+        // in the packet are whitelisted types.
+        memcpy(buffer + buffer_length, block_begin, block_size);
+        buffer_length += block_size;
+        break;
+      case rtcp::App::kPacketType:
+      case rtcp::Sdes::kPacketType:
+      default:
+        // We don't log sender descriptions, application defined messages
+        // or message blocks of unknown type.
+        break;
+    }
+
+    block_begin += block_size;
+  }
+  return buffer_length;
+}
+}  // namespace
+
+std::string RtcEventLogEncoderNewFormat::EncodeLogStart(int64_t timestamp_us) {
+  rtclog2::EventStream event_stream;
+  rtclog2::BeginLogEvent* proto_batch = event_stream.add_begin_log_events();
+  proto_batch->set_timestamp_ms(timestamp_us / 1000);
+  return event_stream.SerializeAsString();
+}
+
+std::string RtcEventLogEncoderNewFormat::EncodeLogEnd(int64_t timestamp_us) {
+  rtclog2::EventStream event_stream;
+  rtclog2::EndLogEvent* proto_batch = event_stream.add_end_log_events();
+  proto_batch->set_timestamp_ms(timestamp_us / 1000);
+  return event_stream.SerializeAsString();
+}
+
+std::string RtcEventLogEncoderNewFormat::EncodeBatch(
+    std::deque<std::unique_ptr<RtcEvent>>::const_iterator begin,
+    std::deque<std::unique_ptr<RtcEvent>>::const_iterator end) {
+  rtclog2::EventStream event_stream;
+  std::string encoded_output;
+
+  {
+    std::vector<const RtcEventAlrState*> alr_state_events;
+    std::vector<const RtcEventAudioNetworkAdaptation*>
+        audio_network_adaptation_events;
+    std::vector<const RtcEventAudioPlayout*> audio_playout_events;
+    std::vector<const RtcEventAudioReceiveStreamConfig*>
+        audio_recv_stream_configs;
+    std::vector<const RtcEventAudioSendStreamConfig*> audio_send_stream_configs;
+    std::vector<const RtcEventBweUpdateDelayBased*> bwe_delay_based_updates;
+    std::vector<const RtcEventBweUpdateLossBased*> bwe_loss_based_updates;
+    std::vector<const RtcEventProbeClusterCreated*>
+        probe_cluster_created_events;
+    std::vector<const RtcEventProbeResultFailure*> probe_result_failure_events;
+    std::vector<const RtcEventProbeResultSuccess*> probe_result_success_events;
+    std::vector<const RtcEventRtcpPacketIncoming*> incoming_rtcp_packets;
+    std::vector<const RtcEventRtcpPacketOutgoing*> outgoing_rtcp_packets;
+    std::vector<const RtcEventRtpPacketIncoming*> incoming_rtp_packets;
+    std::vector<const RtcEventRtpPacketOutgoing*> outgoing_rtp_packets;
+    std::vector<const RtcEventVideoReceiveStreamConfig*>
+        video_recv_stream_configs;
+    std::vector<const RtcEventVideoSendStreamConfig*> video_send_stream_configs;
+    std::vector<const RtcEventIceCandidatePairConfig*> ice_candidate_configs;
+    std::vector<const RtcEventIceCandidatePair*> ice_candidate_events;
+
+    for (auto it = begin; it != end; ++it) {
+      switch ((*it)->GetType()) {
+        case RtcEvent::Type::AlrStateEvent: {
+          auto* rtc_event =
+              static_cast<const RtcEventAlrState* const>(it->get());
+          alr_state_events.push_back(rtc_event);
+          break;
+        }
+        case RtcEvent::Type::AudioNetworkAdaptation: {
+          auto* rtc_event =
+              static_cast<const RtcEventAudioNetworkAdaptation* const>(
+                  it->get());
+          audio_network_adaptation_events.push_back(rtc_event);
+          break;
+        }
+        case RtcEvent::Type::AudioPlayout: {
+          auto* rtc_event =
+              static_cast<const RtcEventAudioPlayout* const>(it->get());
+          audio_playout_events.push_back(rtc_event);
+          break;
+        }
+        case RtcEvent::Type::AudioReceiveStreamConfig: {
+          auto* rtc_event =
+              static_cast<const RtcEventAudioReceiveStreamConfig* const>(
+                  it->get());
+          audio_recv_stream_configs.push_back(rtc_event);
+          break;
+        }
+        case RtcEvent::Type::AudioSendStreamConfig: {
+          auto* rtc_event =
+              static_cast<const RtcEventAudioSendStreamConfig* const>(
+                  it->get());
+          audio_send_stream_configs.push_back(rtc_event);
+          break;
+        }
+        case RtcEvent::Type::BweUpdateDelayBased: {
+          auto* rtc_event =
+              static_cast<const RtcEventBweUpdateDelayBased* const>(it->get());
+          bwe_delay_based_updates.push_back(rtc_event);
+          break;
+        }
+        case RtcEvent::Type::BweUpdateLossBased: {
+          auto* rtc_event =
+              static_cast<const RtcEventBweUpdateLossBased* const>(it->get());
+          bwe_loss_based_updates.push_back(rtc_event);
+          break;
+        }
+        case RtcEvent::Type::ProbeClusterCreated: {
+          auto* rtc_event =
+              static_cast<const RtcEventProbeClusterCreated* const>(it->get());
+          probe_cluster_created_events.push_back(rtc_event);
+          break;
+        }
+        case RtcEvent::Type::ProbeResultFailure: {
+          auto* rtc_event =
+              static_cast<const RtcEventProbeResultFailure* const>(it->get());
+          probe_result_failure_events.push_back(rtc_event);
+          break;
+        }
+        case RtcEvent::Type::ProbeResultSuccess: {
+          auto* rtc_event =
+              static_cast<const RtcEventProbeResultSuccess* const>(it->get());
+          probe_result_success_events.push_back(rtc_event);
+          break;
+        }
+        case RtcEvent::Type::RtcpPacketIncoming: {
+          auto* rtc_event =
+              static_cast<const RtcEventRtcpPacketIncoming* const>(it->get());
+          incoming_rtcp_packets.push_back(rtc_event);
+          break;
+        }
+        case RtcEvent::Type::RtcpPacketOutgoing: {
+          auto* rtc_event =
+              static_cast<const RtcEventRtcpPacketOutgoing* const>(it->get());
+          outgoing_rtcp_packets.push_back(rtc_event);
+          break;
+        }
+        case RtcEvent::Type::RtpPacketIncoming: {
+          auto* rtc_event =
+              static_cast<const RtcEventRtpPacketIncoming* const>(it->get());
+          incoming_rtp_packets.push_back(rtc_event);
+          break;
+        }
+        case RtcEvent::Type::RtpPacketOutgoing: {
+          auto* rtc_event =
+              static_cast<const RtcEventRtpPacketOutgoing* const>(it->get());
+          outgoing_rtp_packets.push_back(rtc_event);
+          break;
+        }
+        case RtcEvent::Type::VideoReceiveStreamConfig: {
+          auto* rtc_event =
+              static_cast<const RtcEventVideoReceiveStreamConfig* const>(
+                  it->get());
+          video_recv_stream_configs.push_back(rtc_event);
+          break;
+        }
+        case RtcEvent::Type::VideoSendStreamConfig: {
+          auto* rtc_event =
+              static_cast<const RtcEventVideoSendStreamConfig* const>(
+                  it->get());
+          video_send_stream_configs.push_back(rtc_event);
+          break;
+        }
+        case RtcEvent::Type::IceCandidatePairConfig: {
+          auto* rtc_event =
+              static_cast<const RtcEventIceCandidatePairConfig* const>(
+                  it->get());
+          ice_candidate_configs.push_back(rtc_event);
+          break;
+        }
+        case RtcEvent::Type::IceCandidatePairEvent: {
+          auto* rtc_event =
+              static_cast<const RtcEventIceCandidatePair* const>(it->get());
+          ice_candidate_events.push_back(rtc_event);
+          break;
+        }
+      }
+    }
+
+    EncodeAlrState(alr_state_events, &event_stream);
+    EncodeAudioNetworkAdaptation(audio_network_adaptation_events,
+                                 &event_stream);
+    EncodeAudioPlayout(audio_playout_events, &event_stream);
+    EncodeAudioRecvStreamConfig(audio_recv_stream_configs, &event_stream);
+    EncodeAudioSendStreamConfig(audio_send_stream_configs, &event_stream);
+    EncodeBweUpdateDelayBased(bwe_delay_based_updates, &event_stream);
+    EncodeBweUpdateLossBased(bwe_loss_based_updates, &event_stream);
+    EncodeProbeClusterCreated(probe_cluster_created_events, &event_stream);
+    EncodeProbeResultFailure(probe_result_failure_events, &event_stream);
+    EncodeProbeResultSuccess(probe_result_success_events, &event_stream);
+    EncodeRtcpPacketIncoming(incoming_rtcp_packets, &event_stream);
+    EncodeRtcpPacketOutgoing(outgoing_rtcp_packets, &event_stream);
+    EncodeRtpPacketIncoming(incoming_rtp_packets, &event_stream);
+    EncodeRtpPacketOutgoing(outgoing_rtp_packets, &event_stream);
+    EncodeVideoRecvStreamConfig(video_recv_stream_configs, &event_stream);
+    EncodeVideoSendStreamConfig(video_send_stream_configs, &event_stream);
+    EncodeIceCandidatePairConfig(ice_candidate_configs, &event_stream);
+    EncodeIceCandidatePairEvent(ice_candidate_events, &event_stream);
+  }  // Deallocate the temporary vectors.
+
+  return event_stream.SerializeAsString();
+}
+
+void RtcEventLogEncoderNewFormat::EncodeAlrState(
+    rtc::ArrayView<const RtcEventAlrState*> batch,
+    rtclog2::EventStream* event_stream) {
+  for (const RtcEventAlrState* base_event : batch) {
+    rtclog2::AlrState* proto_batch = event_stream->add_alr_states();
+    proto_batch->set_timestamp_ms(base_event->timestamp_us_ / 1000);
+    proto_batch->set_in_alr(base_event->in_alr_);
+  }
+  // TODO(terelius): Should we delta-compress this event type?
+}
+
+void RtcEventLogEncoderNewFormat::EncodeAudioNetworkAdaptation(
+    rtc::ArrayView<const RtcEventAudioNetworkAdaptation*> batch,
+    rtclog2::EventStream* event_stream) {
+  if (batch.size() == 0)
+    return;
+  for (const RtcEventAudioNetworkAdaptation* base_event : batch) {
+    rtclog2::AudioNetworkAdaptations* proto_batch =
+        event_stream->add_audio_network_adaptations();
+    proto_batch->set_timestamp_ms(base_event->timestamp_us_ / 1000);
+    if (base_event->config_->bitrate_bps.has_value())
+      proto_batch->set_bitrate_bps(base_event->config_->bitrate_bps.value());
+    if (base_event->config_->frame_length_ms.has_value()) {
+      proto_batch->set_frame_length_ms(
+          base_event->config_->frame_length_ms.value());
+    }
+    if (base_event->config_->uplink_packet_loss_fraction.has_value()) {
+      proto_batch->set_uplink_packet_loss_fraction(
+          base_event->config_->uplink_packet_loss_fraction.value());
+    }
+    if (base_event->config_->enable_fec.has_value())
+      proto_batch->set_enable_fec(base_event->config_->enable_fec.value());
+    if (base_event->config_->enable_dtx.has_value())
+      proto_batch->set_enable_dtx(base_event->config_->enable_dtx.value());
+    if (base_event->config_->num_channels.has_value())
+      proto_batch->set_num_channels(base_event->config_->num_channels.value());
+  }
+  // TODO(terelius): Delta-compress rest of batch.
+}
+
+void RtcEventLogEncoderNewFormat::EncodeAudioPlayout(
+    rtc::ArrayView<const RtcEventAudioPlayout*> batch,
+    rtclog2::EventStream* event_stream) {
+  if (batch.size() == 0)
+    return;
+  for (const RtcEventAudioPlayout* base_event : batch) {
+    rtclog2::AudioPlayoutEvents* proto_batch =
+        event_stream->add_audio_playout_events();
+    proto_batch->set_timestamp_ms(base_event->timestamp_us_ / 1000);
+    proto_batch->set_local_ssrc(base_event->ssrc_);
+  }
+  // TODO(terelius): Delta-compress rest of batch.
+}
+
+void RtcEventLogEncoderNewFormat::EncodeAudioRecvStreamConfig(
+    rtc::ArrayView<const RtcEventAudioReceiveStreamConfig*> batch,
+    rtclog2::EventStream* event_stream) {
+  for (const RtcEventAudioReceiveStreamConfig* base_event : batch) {
+    rtclog2::AudioRecvStreamConfig* proto_batch =
+        event_stream->add_audio_recv_stream_configs();
+    proto_batch->set_timestamp_ms(base_event->timestamp_us_ / 1000);
+    proto_batch->set_remote_ssrc(base_event->config_->remote_ssrc);
+    proto_batch->set_local_ssrc(base_event->config_->local_ssrc);
+    if (!base_event->config_->rsid.empty())
+      proto_batch->set_rsid(base_event->config_->rsid);
+
+    rtclog2::RtpHeaderExtensionConfig* proto_config =
+        proto_batch->mutable_header_extensions();
+    bool has_recognized_extensions =
+        ConvertToProtoFormat(base_event->config_->rtp_extensions, proto_config);
+    if (!has_recognized_extensions)
+      proto_batch->clear_header_extensions();
+  }
+}
+
+void RtcEventLogEncoderNewFormat::EncodeAudioSendStreamConfig(
+    rtc::ArrayView<const RtcEventAudioSendStreamConfig*> batch,
+    rtclog2::EventStream* event_stream) {
+  for (const RtcEventAudioSendStreamConfig* base_event : batch) {
+    rtclog2::AudioSendStreamConfig* proto_batch =
+        event_stream->add_audio_send_stream_configs();
+    proto_batch->set_timestamp_ms(base_event->timestamp_us_ / 1000);
+    proto_batch->set_ssrc(base_event->config_->local_ssrc);
+    if (!base_event->config_->rsid.empty())
+      proto_batch->set_rsid(base_event->config_->rsid);
+
+    rtclog2::RtpHeaderExtensionConfig* proto_config =
+        proto_batch->mutable_header_extensions();
+    bool has_recognized_extensions =
+        ConvertToProtoFormat(base_event->config_->rtp_extensions, proto_config);
+    if (!has_recognized_extensions)
+      proto_batch->clear_header_extensions();
+  }
+}
+
+void RtcEventLogEncoderNewFormat::EncodeBweUpdateDelayBased(
+    rtc::ArrayView<const RtcEventBweUpdateDelayBased*> batch,
+    rtclog2::EventStream* event_stream) {
+  if (batch.size() == 0)
+    return;
+  for (const RtcEventBweUpdateDelayBased* base_event : batch) {
+    rtclog2::DelayBasedBweUpdates* proto_batch =
+        event_stream->add_delay_based_bwe_updates();
+    proto_batch->set_timestamp_ms(base_event->timestamp_us_ / 1000);
+    proto_batch->set_bitrate_bps(base_event->bitrate_bps_);
+    proto_batch->set_detector_state(
+        ConvertToProtoFormat(base_event->detector_state_));
+  }
+  // TODO(terelius): Delta-compress rest of batch.
+}
+
+void RtcEventLogEncoderNewFormat::EncodeBweUpdateLossBased(
+    rtc::ArrayView<const RtcEventBweUpdateLossBased*> batch,
+    rtclog2::EventStream* event_stream) {
+  if (batch.size() == 0)
+    return;
+  for (const RtcEventBweUpdateLossBased* base_event : batch) {
+    rtclog2::LossBasedBweUpdates* proto_batch =
+        event_stream->add_loss_based_bwe_updates();
+    proto_batch->set_timestamp_ms(base_event->timestamp_us_ / 1000);
+    proto_batch->set_bitrate_bps(base_event->bitrate_bps_);
+    proto_batch->set_fraction_loss(base_event->fraction_loss_);
+    proto_batch->set_total_packets(base_event->total_packets_);
+  }
+  // TODO(terelius): Delta-compress rest of batch.
+}
+
+void RtcEventLogEncoderNewFormat::EncodeProbeClusterCreated(
+    rtc::ArrayView<const RtcEventProbeClusterCreated*> batch,
+    rtclog2::EventStream* event_stream) {
+  for (const RtcEventProbeClusterCreated* base_event : batch) {
+    rtclog2::BweProbeCluster* proto_batch = event_stream->add_probe_clusters();
+    proto_batch->set_timestamp_ms(base_event->timestamp_us_ / 1000);
+    proto_batch->set_id(base_event->id_);
+    proto_batch->set_bitrate_bps(base_event->bitrate_bps_);
+    proto_batch->set_min_packets(base_event->min_probes_);
+    proto_batch->set_min_bytes(base_event->min_bytes_);
+  }
+}
+
+void RtcEventLogEncoderNewFormat::EncodeProbeResultFailure(
+    rtc::ArrayView<const RtcEventProbeResultFailure*> batch,
+    rtclog2::EventStream* event_stream) {
+  for (const RtcEventProbeResultFailure* base_event : batch) {
+    rtclog2::BweProbeResultFailure* proto_batch =
+        event_stream->add_probe_failure();
+    proto_batch->set_timestamp_ms(base_event->timestamp_us_ / 1000);
+    proto_batch->set_id(base_event->id_);
+    proto_batch->set_failure(ConvertToProtoFormat(base_event->failure_reason_));
+  }
+  // TODO(terelius): Should we delta-compress this event type?
+}
+
+void RtcEventLogEncoderNewFormat::EncodeProbeResultSuccess(
+    rtc::ArrayView<const RtcEventProbeResultSuccess*> batch,
+    rtclog2::EventStream* event_stream) {
+  for (const RtcEventProbeResultSuccess* base_event : batch) {
+    rtclog2::BweProbeResultSuccess* proto_batch =
+        event_stream->add_probe_success();
+    proto_batch->set_timestamp_ms(base_event->timestamp_us_ / 1000);
+    proto_batch->set_id(base_event->id_);
+    proto_batch->set_bitrate_bps(base_event->bitrate_bps_);
+  }
+  // TODO(terelius): Should we delta-compress this event type?
+}
+
+void RtcEventLogEncoderNewFormat::EncodeRtcpPacketIncoming(
+    rtc::ArrayView<const RtcEventRtcpPacketIncoming*> batch,
+    rtclog2::EventStream* event_stream) {
+  if (batch.size() == 0)
+    return;
+  for (const RtcEventRtcpPacketIncoming* base_event : batch) {
+    rtclog2::IncomingRtcpPackets* proto_batch =
+        event_stream->add_incoming_rtcp_packets();
+    proto_batch->set_timestamp_ms(base_event->timestamp_us_ / 1000);
+
+    uint8_t buffer[IP_PACKET_SIZE];
+    size_t buffer_length =
+        RemoveNonWhitelistedRtcpBlocks(base_event->packet_, buffer);
+    proto_batch->set_raw_packet(buffer, buffer_length);
+  }
+  // TODO(terelius): Delta-compress rest of batch.
+}
+
+void RtcEventLogEncoderNewFormat::EncodeRtcpPacketOutgoing(
+    rtc::ArrayView<const RtcEventRtcpPacketOutgoing*> batch,
+    rtclog2::EventStream* event_stream) {
+  if (batch.size() == 0)
+    return;
+  for (const RtcEventRtcpPacketOutgoing* base_event : batch) {
+    rtclog2::OutgoingRtcpPackets* proto_batch =
+        event_stream->add_outgoing_rtcp_packets();
+    proto_batch->set_timestamp_ms(base_event->timestamp_us_ / 1000);
+
+    uint8_t buffer[IP_PACKET_SIZE];
+    size_t buffer_length =
+        RemoveNonWhitelistedRtcpBlocks(base_event->packet_, buffer);
+    proto_batch->set_raw_packet(buffer, buffer_length);
+  }
+  // TODO(terelius): Delta-compress rest of batch.
+}
+
+void RtcEventLogEncoderNewFormat::EncodeRtpPacketIncoming(
+    rtc::ArrayView<const RtcEventRtpPacketIncoming*> batch,
+    rtclog2::EventStream* event_stream) {
+  if (batch.size() == 0)
+    return;
+  for (const RtcEventRtpPacketIncoming* base_event : batch) {
+    rtclog2::IncomingRtpPackets* proto_batch =
+        event_stream->add_incoming_rtp_packets();
+    proto_batch->set_timestamp_ms(base_event->timestamp_us_ / 1000);
+    proto_batch->set_marker(base_event->header_.Marker());
+    // TODO(terelius): Is payload type needed?
+    proto_batch->set_payload_type(base_event->header_.PayloadType());
+    proto_batch->set_sequence_number(base_event->header_.SequenceNumber());
+    proto_batch->set_rtp_timestamp(base_event->header_.Timestamp());
+    proto_batch->set_ssrc(base_event->header_.Ssrc());
+    proto_batch->set_payload_size(base_event->payload_length_);
+    proto_batch->set_header_size(base_event->header_length_);
+    proto_batch->set_padding_size(base_event->padding_length_);
+
+    // Add header extensions.
+    if (base_event->header_.HasExtension<TransmissionOffset>()) {
+      int32_t offset;
+      base_event->header_.GetExtension<TransmissionOffset>(&offset);
+      proto_batch->set_transmission_time_offset(offset);
+    }
+    if (base_event->header_.HasExtension<AbsoluteSendTime>()) {
+      uint32_t sendtime;
+      base_event->header_.GetExtension<AbsoluteSendTime>(&sendtime);
+      proto_batch->set_absolute_send_time(sendtime);
+    }
+    if (base_event->header_.HasExtension<TransportSequenceNumber>()) {
+      uint16_t seqnum;
+      base_event->header_.GetExtension<TransportSequenceNumber>(&seqnum);
+      proto_batch->set_transport_sequence_number(seqnum);
+    }
+    if (base_event->header_.HasExtension<AudioLevel>()) {
+      bool voice_activity;
+      uint8_t audio_level;
+      base_event->header_.GetExtension<AudioLevel>(&voice_activity,
+                                                   &audio_level);
+      RTC_DCHECK(audio_level < 128);
+      if (voice_activity) {
+        audio_level += 128;  // Most significant bit indicates voice activity.
+      }
+      proto_batch->set_audio_level(audio_level);
+    }
+    if (base_event->header_.HasExtension<VideoOrientation>()) {
+      VideoRotation video_rotation;
+      base_event->header_.GetExtension<VideoOrientation>(&video_rotation);
+      proto_batch->set_video_rotation(
+          ConvertVideoRotationToCVOByte(video_rotation));
+    }
+  }
+  // TODO(terelius): Delta-compress rest of batch.
+}
+
+void RtcEventLogEncoderNewFormat::EncodeRtpPacketOutgoing(
+    rtc::ArrayView<const RtcEventRtpPacketOutgoing*> batch,
+    rtclog2::EventStream* event_stream) {
+  if (batch.size() == 0)
+    return;
+  for (const RtcEventRtpPacketOutgoing* base_event : batch) {
+    rtclog2::OutgoingRtpPackets* proto_batch =
+        event_stream->add_outgoing_rtp_packets();
+    proto_batch->set_timestamp_ms(base_event->timestamp_us_ / 1000);
+    proto_batch->set_marker(base_event->header_.Marker());
+    // TODO(terelius): Is payload type needed?
+    proto_batch->set_payload_type(base_event->header_.PayloadType());
+    proto_batch->set_sequence_number(base_event->header_.SequenceNumber());
+    proto_batch->set_rtp_timestamp(base_event->header_.Timestamp());
+    proto_batch->set_ssrc(base_event->header_.Ssrc());
+    proto_batch->set_payload_size(base_event->payload_length_);
+    proto_batch->set_header_size(base_event->header_length_);
+    proto_batch->set_padding_size(base_event->padding_length_);
+
+    // Add header extensions.
+    if (base_event->header_.HasExtension<TransmissionOffset>()) {
+      int32_t offset;
+      base_event->header_.GetExtension<TransmissionOffset>(&offset);
+      proto_batch->set_transmission_time_offset(offset);
+    }
+    if (base_event->header_.HasExtension<AbsoluteSendTime>()) {
+      uint32_t sendtime;
+      base_event->header_.GetExtension<AbsoluteSendTime>(&sendtime);
+      proto_batch->set_absolute_send_time(sendtime);
+    }
+    if (base_event->header_.HasExtension<TransportSequenceNumber>()) {
+      uint16_t seqnum;
+      base_event->header_.GetExtension<TransportSequenceNumber>(&seqnum);
+      proto_batch->set_transport_sequence_number(seqnum);
+    }
+    if (base_event->header_.HasExtension<AudioLevel>()) {
+      bool voice_activity;
+      uint8_t audio_level;
+      base_event->header_.GetExtension<AudioLevel>(&voice_activity,
+                                                   &audio_level);
+      RTC_DCHECK(audio_level < 128);
+      if (voice_activity) {
+        audio_level += 128;  // Most significant bit indicates voice activity.
+      }
+      proto_batch->set_audio_level(audio_level);
+    }
+    if (base_event->header_.HasExtension<VideoOrientation>()) {
+      VideoRotation video_rotation;
+      base_event->header_.GetExtension<VideoOrientation>(&video_rotation);
+      proto_batch->set_video_rotation(
+          ConvertVideoRotationToCVOByte(video_rotation));
+    }
+  }
+  // TODO(terelius): Delta-compress rest of batch.
+}
+
+void RtcEventLogEncoderNewFormat::EncodeVideoRecvStreamConfig(
+    rtc::ArrayView<const RtcEventVideoReceiveStreamConfig*> batch,
+    rtclog2::EventStream* event_stream) {
+  for (const RtcEventVideoReceiveStreamConfig* base_event : batch) {
+    rtclog2::VideoRecvStreamConfig* proto_batch =
+        event_stream->add_video_recv_stream_configs();
+    proto_batch->set_timestamp_ms(base_event->timestamp_us_ / 1000);
+    proto_batch->set_remote_ssrc(base_event->config_->remote_ssrc);
+    proto_batch->set_local_ssrc(base_event->config_->local_ssrc);
+    proto_batch->set_rtx_ssrc(base_event->config_->rtx_ssrc);
+    if (!base_event->config_->rsid.empty())
+      proto_batch->set_rsid(base_event->config_->rsid);
+
+    rtclog2::RtpHeaderExtensionConfig* proto_config =
+        proto_batch->mutable_header_extensions();
+    bool has_recognized_extensions =
+        ConvertToProtoFormat(base_event->config_->rtp_extensions, proto_config);
+    if (!has_recognized_extensions)
+      proto_batch->clear_header_extensions();
+  }
+}
+
+void RtcEventLogEncoderNewFormat::EncodeVideoSendStreamConfig(
+    rtc::ArrayView<const RtcEventVideoSendStreamConfig*> batch,
+    rtclog2::EventStream* event_stream) {
+  for (const RtcEventVideoSendStreamConfig* base_event : batch) {
+    rtclog2::VideoSendStreamConfig* proto_batch =
+        event_stream->add_video_send_stream_configs();
+    proto_batch->set_timestamp_ms(base_event->timestamp_us_ / 1000);
+    proto_batch->set_ssrc(base_event->config_->local_ssrc);
+    proto_batch->set_rtx_ssrc(base_event->config_->rtx_ssrc);
+    if (!base_event->config_->rsid.empty())
+      proto_batch->set_rsid(base_event->config_->rsid);
+
+    rtclog2::RtpHeaderExtensionConfig* proto_config =
+        proto_batch->mutable_header_extensions();
+    bool has_recognized_extensions =
+        ConvertToProtoFormat(base_event->config_->rtp_extensions, proto_config);
+    if (!has_recognized_extensions)
+      proto_batch->clear_header_extensions();
+  }
+}
+
+void RtcEventLogEncoderNewFormat::EncodeIceCandidatePairConfig(
+    rtc::ArrayView<const RtcEventIceCandidatePairConfig*> batch,
+    rtclog2::EventStream* event_stream) {
+  for (const RtcEventIceCandidatePairConfig* base_event : batch) {
+    rtclog2::IceCandidatePairConfig* proto_batch =
+        event_stream->add_ice_candidate_configs();
+
+    proto_batch->set_timestamp_ms(base_event->timestamp_us_ / 1000);
+    proto_batch->set_config_type(ConvertToProtoFormat(base_event->type_));
+    proto_batch->set_candidate_pair_id(base_event->candidate_pair_id_);
+    const auto& desc = base_event->candidate_pair_desc_;
+    proto_batch->set_local_candidate_type(
+        ConvertToProtoFormat(desc.local_candidate_type));
+    proto_batch->set_local_relay_protocol(
+        ConvertToProtoFormat(desc.local_relay_protocol));
+    proto_batch->set_local_network_type(
+        ConvertToProtoFormat(desc.local_network_type));
+    proto_batch->set_local_address_family(
+        ConvertToProtoFormat(desc.local_address_family));
+    proto_batch->set_remote_candidate_type(
+        ConvertToProtoFormat(desc.remote_candidate_type));
+    proto_batch->set_remote_address_family(
+        ConvertToProtoFormat(desc.remote_address_family));
+    proto_batch->set_candidate_pair_protocol(
+        ConvertToProtoFormat(desc.candidate_pair_protocol));
+  }
+  // TODO(terelius): Should we delta-compress this event type?
+}
+
+void RtcEventLogEncoderNewFormat::EncodeIceCandidatePairEvent(
+    rtc::ArrayView<const RtcEventIceCandidatePair*> batch,
+    rtclog2::EventStream* event_stream) {
+  for (const RtcEventIceCandidatePair* base_event : batch) {
+    rtclog2::IceCandidatePairEvent* proto_batch =
+        event_stream->add_ice_candidate_events();
+
+    proto_batch->set_timestamp_ms(base_event->timestamp_us_ / 1000);
+
+    proto_batch->set_event_type(ConvertToProtoFormat(base_event->type_));
+    proto_batch->set_candidate_pair_id(base_event->candidate_pair_id_);
+  }
+  // TODO(terelius): Should we delta-compress this event type?
+}
+
+}  // namespace webrtc
diff --git a/logging/rtc_event_log/encoder/rtc_event_log_encoder_new_format.h b/logging/rtc_event_log/encoder/rtc_event_log_encoder_new_format.h
new file mode 100644
index 0000000..b49286d
--- /dev/null
+++ b/logging/rtc_event_log/encoder/rtc_event_log_encoder_new_format.h
@@ -0,0 +1,122 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef LOGGING_RTC_EVENT_LOG_ENCODER_RTC_EVENT_LOG_ENCODER_NEW_FORMAT_H_
+#define LOGGING_RTC_EVENT_LOG_ENCODER_RTC_EVENT_LOG_ENCODER_NEW_FORMAT_H_
+
+#include <deque>
+#include <memory>
+#include <string>
+
+#include "api/array_view.h"
+#include "logging/rtc_event_log/encoder/rtc_event_log_encoder.h"
+
+namespace webrtc {
+
+namespace rtclog2 {
+class EventStream;  // Auto-generated from protobuf.
+}  // namespace rtclog2
+
+class RtcEventAlrState;
+class RtcEventAudioNetworkAdaptation;
+class RtcEventAudioPlayout;
+class RtcEventAudioReceiveStreamConfig;
+class RtcEventAudioSendStreamConfig;
+class RtcEventBweUpdateDelayBased;
+class RtcEventBweUpdateLossBased;
+class RtcEventLoggingStarted;
+class RtcEventLoggingStopped;
+class RtcEventProbeClusterCreated;
+class RtcEventProbeResultFailure;
+class RtcEventProbeResultSuccess;
+class RtcEventRtcpPacketIncoming;
+class RtcEventRtcpPacketOutgoing;
+class RtcEventRtpPacketIncoming;
+class RtcEventRtpPacketOutgoing;
+class RtcEventVideoReceiveStreamConfig;
+class RtcEventVideoSendStreamConfig;
+class RtcEventIceCandidatePairConfig;
+class RtcEventIceCandidatePair;
+class RtpPacket;
+
+class RtcEventLogEncoderNewFormat final : public RtcEventLogEncoder {
+ public:
+  ~RtcEventLogEncoderNewFormat() override = default;
+
+  std::string EncodeBatch(
+      std::deque<std::unique_ptr<RtcEvent>>::const_iterator begin,
+      std::deque<std::unique_ptr<RtcEvent>>::const_iterator end) override;
+
+  std::string EncodeLogStart(int64_t timestamp_us) override;
+  std::string EncodeLogEnd(int64_t timestamp_us) override;
+
+ private:
+  // Encoding entry-point for the various RtcEvent subclasses.
+  void EncodeAlrState(rtc::ArrayView<const RtcEventAlrState*> batch,
+                      rtclog2::EventStream* event_stream);
+  void EncodeAudioNetworkAdaptation(
+      rtc::ArrayView<const RtcEventAudioNetworkAdaptation*> batch,
+      rtclog2::EventStream* event_stream);
+  void EncodeAudioPlayout(rtc::ArrayView<const RtcEventAudioPlayout*> batch,
+                          rtclog2::EventStream* event_stream);
+  void EncodeAudioRecvStreamConfig(
+      rtc::ArrayView<const RtcEventAudioReceiveStreamConfig*> batch,
+      rtclog2::EventStream* event_stream);
+  void EncodeAudioSendStreamConfig(
+      rtc::ArrayView<const RtcEventAudioSendStreamConfig*> batch,
+      rtclog2::EventStream* event_stream);
+  void EncodeBweUpdateDelayBased(
+      rtc::ArrayView<const RtcEventBweUpdateDelayBased*> batch,
+      rtclog2::EventStream* event_stream);
+  void EncodeBweUpdateLossBased(
+      rtc::ArrayView<const RtcEventBweUpdateLossBased*> batch,
+      rtclog2::EventStream* event_stream);
+  void EncodeLoggingStarted(rtc::ArrayView<const RtcEventLoggingStarted*> batch,
+                            rtclog2::EventStream* event_stream);
+  void EncodeLoggingStopped(rtc::ArrayView<const RtcEventLoggingStopped*> batch,
+                            rtclog2::EventStream* event_stream);
+  void EncodeProbeClusterCreated(
+      rtc::ArrayView<const RtcEventProbeClusterCreated*> batch,
+      rtclog2::EventStream* event_stream);
+  void EncodeProbeResultFailure(
+      rtc::ArrayView<const RtcEventProbeResultFailure*> batch,
+      rtclog2::EventStream* event_stream);
+  void EncodeProbeResultSuccess(
+      rtc::ArrayView<const RtcEventProbeResultSuccess*> batch,
+      rtclog2::EventStream* event_stream);
+  void EncodeRtcpPacketIncoming(
+      rtc::ArrayView<const RtcEventRtcpPacketIncoming*> batch,
+      rtclog2::EventStream* event_stream);
+  void EncodeRtcpPacketOutgoing(
+      rtc::ArrayView<const RtcEventRtcpPacketOutgoing*> batch,
+      rtclog2::EventStream* event_stream);
+  void EncodeRtpPacketIncoming(
+      rtc::ArrayView<const RtcEventRtpPacketIncoming*> batch,
+      rtclog2::EventStream* event_stream);
+  void EncodeRtpPacketOutgoing(
+      rtc::ArrayView<const RtcEventRtpPacketOutgoing*> batch,
+      rtclog2::EventStream* event_stream);
+  void EncodeVideoRecvStreamConfig(
+      rtc::ArrayView<const RtcEventVideoReceiveStreamConfig*> batch,
+      rtclog2::EventStream* event_stream);
+  void EncodeVideoSendStreamConfig(
+      rtc::ArrayView<const RtcEventVideoSendStreamConfig*> batch,
+      rtclog2::EventStream* event_stream);
+  void EncodeIceCandidatePairConfig(
+      rtc::ArrayView<const RtcEventIceCandidatePairConfig*> batch,
+      rtclog2::EventStream* event_stream);
+  void EncodeIceCandidatePairEvent(
+      rtc::ArrayView<const RtcEventIceCandidatePair*> batch,
+      rtclog2::EventStream* event_stream);
+};
+
+}  // namespace webrtc
+
+#endif  // LOGGING_RTC_EVENT_LOG_ENCODER_RTC_EVENT_LOG_ENCODER_NEW_FORMAT_H_
diff --git a/logging/rtc_event_log/rtc_event_log.h b/logging/rtc_event_log/rtc_event_log.h
index e09d184..7d900c0 100644
--- a/logging/rtc_event_log/rtc_event_log.h
+++ b/logging/rtc_event_log/rtc_event_log.h
@@ -28,10 +28,9 @@
   enum : size_t { kUnlimitedOutput = 0 };
   enum : int64_t { kImmediateOutput = 0 };
 
-  // TODO(eladalon): Two stages are upcoming.
-  // 1. Extend this to actually support the new encoding.
-  // 2. Get rid of the legacy encoding, allowing us to get rid of this enum.
-  enum class EncodingType { Legacy };
+  // TODO(eladalon):  Get rid of the legacy encoding and this enum once all
+  // clients have migrated to the new format.
+  enum class EncodingType { Legacy, NewFormat };
 
   virtual ~RtcEventLog() {}
 
diff --git a/logging/rtc_event_log/rtc_event_log2.proto b/logging/rtc_event_log/rtc_event_log2.proto
index 213aee7..fb8f760 100644
--- a/logging/rtc_event_log/rtc_event_log2.proto
+++ b/logging/rtc_event_log/rtc_event_log2.proto
@@ -263,9 +263,10 @@
   optional uint32 bitrate_bps = 2;
 
   enum DetectorState {
-    BWE_NORMAL = 0;
-    BWE_UNDERUSING = 1;
-    BWE_OVERUSING = 2;
+    BWE_UNKNOWN_STATE = 0;
+    BWE_NORMAL = 1;
+    BWE_UNDERUSING = 2;
+    BWE_OVERUSING = 3;
   }
   optional DetectorState detector_state = 3;
 
@@ -474,10 +475,11 @@
 
 message IceCandidatePairConfig {
   enum IceCandidatePairConfigType {
-    ADDED = 0;
-    UPDATED = 1;
-    DESTROYED = 2;
-    SELECTED = 3;
+    UNKNOWN_CONFIG_TYPE = 0;
+    ADDED = 1;
+    UPDATED = 2;
+    DESTROYED = 3;
+    SELECTED = 4;
   }
 
   enum IceCandidateType {
@@ -544,10 +546,11 @@
 
 message IceCandidatePairEvent {
   enum IceCandidatePairEventType {
-    CHECK_SENT = 0;
-    CHECK_RECEIVED = 1;
-    CHECK_RESPONSE_SENT = 2;
-    CHECK_RESPONSE_RECEIVED = 3;
+    UNKNOWN_CHECK_TYPE = 0;
+    CHECK_SENT = 1;
+    CHECK_RECEIVED = 2;
+    CHECK_RESPONSE_SENT = 3;
+    CHECK_RESPONSE_RECEIVED = 4;
   }
 
   // required
diff --git a/logging/rtc_event_log/rtc_event_log_impl.cc b/logging/rtc_event_log/rtc_event_log_impl.cc
index b6b0428..2f46e3e 100644
--- a/logging/rtc_event_log/rtc_event_log_impl.cc
+++ b/logging/rtc_event_log/rtc_event_log_impl.cc
@@ -18,8 +18,9 @@
 #include <vector>
 
 #include "absl/memory/memory.h"
+#include "api/rtceventlogoutput.h"
 #include "logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.h"
-#include "logging/rtc_event_log/output/rtc_event_log_output_file.h"
+#include "logging/rtc_event_log/encoder/rtc_event_log_encoder_new_format.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/constructormagic.h"
 #include "rtc_base/event.h"
@@ -65,6 +66,8 @@
   switch (type) {
     case RtcEventLog::EncodingType::Legacy:
       return absl::make_unique<RtcEventLogEncoderLegacy>();
+    case RtcEventLog::EncodingType::NewFormat:
+      return absl::make_unique<RtcEventLogEncoderNewFormat>();
     default:
       RTC_LOG(LS_ERROR) << "Unknown RtcEventLog encoder type (" << int(type)
                         << ")";