Adding upstream version 125.0.1.upstream/125.0.1

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

92 files changed, 2813 insertions, 2869 deletions

diff --git a/third_party/libwebrtc/pc/BUILD.gn b/third_party/libwebrtc/pc/BUILD.gn
index 7c22a26d12..e9549cdfd8 100644
--- a/third_party/libwebrtc/pc/BUILD.gn
+++ b/third_party/libwebrtc/pc/BUILD.gn
@@ -314,6 +314,15 @@ rtc_source_set("jsep_transport_controller") {
   ]
 }
 
+rtc_source_set("media_factory") {
+  sources = [ "media_factory.h" ]
+  deps = [
+    "../api/environment",
+    "../call:call_interfaces",
+    "../media:rtc_media_base",
+  ]
+}
+
 rtc_source_set("media_session") {
   visibility = [ "*" ]  # Used by Chrome
   sources = [
@@ -730,6 +739,7 @@ rtc_library("media_protocol_names") {
 rtc_source_set("peerconnection") {
   # TODO(bugs.webrtc.org/13661): Reduce visibility if possible
   visibility = [ "*" ]  # Used by Chromium and others
+  allow_poison = [ "environment_construction" ]
   cflags = []
   sources = []
 
@@ -919,6 +929,7 @@ rtc_library("connection_context") {
     "connection_context.h",
   ]
   deps = [
+    ":media_factory",
     "../api:callfactory_api",
     "../api:field_trials_view",
     "../api:libjingle_peerconnection_api",
@@ -926,6 +937,7 @@ rtc_library("connection_context") {
     "../api:refcountedbase",
     "../api:scoped_refptr",
     "../api:sequence_checker",
+    "../api/environment",
     "../api/neteq:neteq_api",
     "../api/transport:field_trial_based_config",
     "../api/transport:sctp_transport_factory_interface",
@@ -1223,6 +1235,7 @@ rtc_source_set("peer_connection") {
     "../api:turn_customizer",
     "../api/adaptation:resource_adaptation_api",
     "../api/crypto:options",
+    "../api/environment",
     "../api/rtc_event_log",
     "../api/task_queue:pending_task_safety_flag",
     "../api/transport:bitrate_settings",
@@ -1486,6 +1499,7 @@ rtc_library("media_stream_observer") {
 rtc_source_set("peer_connection_factory") {
   # TODO(bugs.webrtc.org/13661): Reduce visibility if possible
   visibility = [ "*" ]  # Known to be used externally
+  allow_poison = [ "environment_construction" ]
   sources = [
     "peer_connection_factory.cc",
     "peer_connection_factory.h",
@@ -1510,10 +1524,11 @@ rtc_source_set("peer_connection_factory") {
     "../api:rtp_parameters",
     "../api:scoped_refptr",
     "../api:sequence_checker",
+    "../api/environment",
+    "../api/environment:environment_factory",
     "../api/metronome",
     "../api/neteq:neteq_api",
     "../api/rtc_event_log:rtc_event_log",
-    "../api/task_queue:task_queue",
     "../api/transport:bitrate_settings",
     "../api/transport:network_control",
     "../api/transport:sctp_transport_factory_interface",
@@ -1525,6 +1540,7 @@ rtc_source_set("peer_connection_factory") {
     "../p2p:rtc_p2p",
     "../pc:audio_track",
     "../pc:connection_context",
+    "../pc:media_factory",
     "../pc:media_stream",
     "../pc:rtp_parameters_conversion",
     "../pc:session_description",
@@ -2051,6 +2067,7 @@ rtc_source_set("legacy_stats_collector_interface") {
 rtc_source_set("libjingle_peerconnection") {
   # TODO(bugs.webrtc.org/13661): Reduce visibility if possible
   visibility = [ "*" ]  # Used by Chrome and others
+  allow_poison = [ "environment_construction" ]
 
   deps = [
     ":peerconnection",
@@ -2362,6 +2379,7 @@ if (rtc_include_tests && !build_with_chromium) {
       ":dtls_srtp_transport",
       ":dtls_transport",
       ":dtmf_sender",
+      ":enable_fake_media",
       ":ice_server_parsing",
       ":integration_test_helpers",
       ":jitter_buffer_delay",
@@ -2401,10 +2419,13 @@ if (rtc_include_tests && !build_with_chromium) {
       ":webrtc_sdp",
       "../api:array_view",
       "../api:audio_options_api",
+      "../api:callfactory_api",
       "../api:candidate",
       "../api:create_peerconnection_factory",
       "../api:dtls_transport_interface",
       "../api:dtmf_sender_interface",
+      "../api:enable_media",
+      "../api:enable_media_with_defaults",
       "../api:fake_frame_decryptor",
       "../api:fake_frame_encryptor",
       "../api:field_trials_view",
@@ -2429,6 +2450,7 @@ if (rtc_include_tests && !build_with_chromium) {
       "../api/crypto:frame_decryptor_interface",
       "../api/crypto:frame_encryptor_interface",
       "../api/crypto:options",
+      "../api/environment:environment_factory",
       "../api/rtc_event_log",
       "../api/rtc_event_log:rtc_event_log_factory",
       "../api/task_queue",
@@ -2457,7 +2479,6 @@ if (rtc_include_tests && !build_with_chromium) {
       "../media:rid_description",
       "../media:rtc_data_sctp_transport_internal",
       "../media:rtc_media_config",
-      "../media:rtc_media_engine_defaults",
       "../media:stream_params",
       "../modules/audio_device:audio_device_api",
       "../modules/audio_processing:audio_processing_statistics",
@@ -2520,7 +2541,6 @@ if (rtc_include_tests && !build_with_chromium) {
       ":libjingle_peerconnection",
       ":pc_test_utils",
       ":rtc_pc",
-      "../api:callfactory_api",
       "../api:rtc_event_log_output_file",
       "../api:rtc_stats_api",
       "../api:rtp_parameters",
@@ -2627,9 +2647,9 @@ if (rtc_include_tests && !build_with_chromium) {
       ":video_track_source",
       "../api:array_view",
       "../api:audio_options_api",
-      "../api:callfactory_api",
       "../api:candidate",
       "../api:create_peerconnection_factory",
+      "../api:enable_media_with_defaults",
       "../api:fake_frame_decryptor",
       "../api:fake_frame_encryptor",
       "../api:field_trials_view",
@@ -2665,10 +2685,8 @@ if (rtc_include_tests && !build_with_chromium) {
       "../call:call_interfaces",
       "../call/adaptation:resource_adaptation_test_utilities",
       "../logging:fake_rtc_event_log",
-      "../media:rtc_audio_video",
       "../media:rtc_media_base",
       "../media:rtc_media_config",
-      "../media:rtc_media_engine_defaults",
       "../media:rtc_media_tests_utils",
       "../media:stream_params",
       "../modules/audio_device:audio_device_api",
@@ -2719,6 +2737,24 @@ if (rtc_include_tests && !build_with_chromium) {
     ]
   }
 
+  rtc_library("enable_fake_media") {
+    testonly = true
+    visibility = [ ":*" ]
+    sources = [
+      "test/enable_fake_media.cc",
+      "test/enable_fake_media.h",
+    ]
+    deps = [
+      ":media_factory",
+      "../api:libjingle_peerconnection_api",
+      "../api/environment",
+      "../call:call_interfaces",
+      "../media:rtc_media_tests_utils",
+      "../rtc_base:checks",
+    ]
+    absl_deps = [ "//third_party/abseil-cpp/absl/base:nullability" ]
+  }
+
   rtc_library("pc_test_utils") {
     testonly = true
     sources = [
@@ -2751,6 +2787,7 @@ if (rtc_include_tests && !build_with_chromium) {
     deps = [
       ":channel",
       ":channel_interface",
+      ":enable_fake_media",
       ":jitter_buffer_delay",
       ":libjingle_peerconnection",
       ":peer_connection_internal",
@@ -2778,6 +2815,7 @@ if (rtc_include_tests && !build_with_chromium) {
       "../api:sequence_checker",
       "../api/audio:audio_mixer_api",
       "../api/audio_codecs:audio_codecs_api",
+      "../api/environment:environment_factory",
       "../api/task_queue",
       "../api/task_queue:default_task_queue_factory",
       "../api/units:time_delta",
diff --git a/third_party/libwebrtc/pc/audio_rtp_receiver.cc b/third_party/libwebrtc/pc/audio_rtp_receiver.cc
index a8659de5f9..6e7ca6d0b5 100644
--- a/third_party/libwebrtc/pc/audio_rtp_receiver.cc
+++ b/third_party/libwebrtc/pc/audio_rtp_receiver.cc
@@ -278,7 +278,7 @@ std::vector<RtpSource> AudioRtpReceiver::GetSources() const {
 }
 
 void AudioRtpReceiver::SetDepacketizerToDecoderFrameTransformer(
-    rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer) {
+    rtc::scoped_refptr<FrameTransformerInterface> frame_transformer) {
   RTC_DCHECK_RUN_ON(worker_thread_);
   if (media_channel_) {
     media_channel_->SetDepacketizerToDecoderFrameTransformer(
diff --git a/third_party/libwebrtc/pc/audio_rtp_receiver.h b/third_party/libwebrtc/pc/audio_rtp_receiver.h
index 86c42d532a..36cbdffc35 100644
--- a/third_party/libwebrtc/pc/audio_rtp_receiver.h
+++ b/third_party/libwebrtc/pc/audio_rtp_receiver.h
@@ -118,8 +118,7 @@ class AudioRtpReceiver : public ObserverInterface,
   std::vector<RtpSource> GetSources() const override;
   int AttachmentId() const override { return attachment_id_; }
   void SetDepacketizerToDecoderFrameTransformer(
-      rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer)
-      override;
+      rtc::scoped_refptr<FrameTransformerInterface> frame_transformer) override;
 
  private:
   void RestartMediaChannel(absl::optional<uint32_t> ssrc)
diff --git a/third_party/libwebrtc/pc/audio_rtp_receiver_unittest.cc b/third_party/libwebrtc/pc/audio_rtp_receiver_unittest.cc
index 9eb20c982f..e031f90359 100644
--- a/third_party/libwebrtc/pc/audio_rtp_receiver_unittest.cc
+++ b/third_party/libwebrtc/pc/audio_rtp_receiver_unittest.cc
@@ -98,7 +98,7 @@ TEST_F(AudioRtpReceiverTest, VolumesSetBeforeStartingAreRespected) {
 // thread when a media channel pointer is passed to the receiver via the
 // constructor.
 TEST(AudioRtpReceiver, OnChangedNotificationsAfterConstruction) {
-  webrtc::test::RunLoop loop;
+  test::RunLoop loop;
   auto* thread = rtc::Thread::Current();  // Points to loop's thread.
   cricket::MockVoiceMediaReceiveChannelInterface receive_channel;
   auto receiver = rtc::make_ref_counted<AudioRtpReceiver>(
diff --git a/third_party/libwebrtc/pc/audio_track.h b/third_party/libwebrtc/pc/audio_track.h
index ae326b304b..92c3141d8a 100644
--- a/third_party/libwebrtc/pc/audio_track.h
+++ b/third_party/libwebrtc/pc/audio_track.h
@@ -58,7 +58,7 @@ class AudioTrack : public MediaStreamTrack<AudioTrackInterface>,
 
  private:
   const rtc::scoped_refptr<AudioSourceInterface> audio_source_;
-  RTC_NO_UNIQUE_ADDRESS webrtc::SequenceChecker signaling_thread_checker_;
+  RTC_NO_UNIQUE_ADDRESS SequenceChecker signaling_thread_checker_;
 };
 
 }  // namespace webrtc
diff --git a/third_party/libwebrtc/pc/channel.cc b/third_party/libwebrtc/pc/channel.cc
index 0024ba0e35..7a1f55d9e9 100644
--- a/third_party/libwebrtc/pc/channel.cc
+++ b/third_party/libwebrtc/pc/channel.cc
@@ -79,7 +79,7 @@ struct StreamFinder {
 }  // namespace
 
 void MediaChannelParametersFromMediaDescription(
-    const RtpMediaContentDescription* desc,
+    const MediaContentDescription* desc,
     const RtpHeaderExtensions& extensions,
     bool is_stream_active,
     MediaChannelParameters* params) {
@@ -97,7 +97,7 @@ void MediaChannelParametersFromMediaDescription(
 }
 
 void RtpSendParametersFromMediaDescription(
-    const RtpMediaContentDescription* desc,
+    const MediaContentDescription* desc,
     webrtc::RtpExtension::Filter extensions_filter,
     SenderParameters* send_params) {
   RtpHeaderExtensions extensions =
@@ -875,7 +875,7 @@ bool VoiceChannel::SetLocalContent_w(const MediaContentDescription* content,
 
   AudioReceiverParameters recv_params = last_recv_params_;
   MediaChannelParametersFromMediaDescription(
-      content->as_audio(), header_extensions,
+      content, header_extensions,
       webrtc::RtpTransceiverDirectionHasRecv(content->direction()),
       &recv_params);
 
@@ -927,8 +927,8 @@ bool VoiceChannel::SetRemoteContent_w(const MediaContentDescription* content,
   RTC_LOG(LS_INFO) << "Setting remote voice description for " << ToString();
 
   AudioSenderParameter send_params = last_send_params_;
-  RtpSendParametersFromMediaDescription(content->as_audio(),
-                                        extensions_filter(), &send_params);
+  RtpSendParametersFromMediaDescription(content, extensions_filter(),
+                                        &send_params);
   send_params.mid = mid();
 
   bool parameters_applied =
@@ -1016,7 +1016,7 @@ bool VideoChannel::SetLocalContent_w(const MediaContentDescription* content,
   VideoReceiverParameters recv_params = last_recv_params_;
 
   MediaChannelParametersFromMediaDescription(
-      content->as_video(), header_extensions,
+      content, header_extensions,
       webrtc::RtpTransceiverDirectionHasRecv(content->direction()),
       &recv_params);
 
@@ -1100,7 +1100,7 @@ bool VideoChannel::SetLocalContent_w(const MediaContentDescription* content,
     last_send_params_ = send_params;
   }
 
-  if (!UpdateLocalStreams_w(content->as_video()->streams(), type, error_desc)) {
+  if (!UpdateLocalStreams_w(content->streams(), type, error_desc)) {
     RTC_DCHECK(!error_desc.empty());
     return false;
   }
@@ -1128,13 +1128,11 @@ bool VideoChannel::SetRemoteContent_w(const MediaContentDescription* content,
   TRACE_EVENT0("webrtc", "VideoChannel::SetRemoteContent_w");
   RTC_LOG(LS_INFO) << "Setting remote video description for " << ToString();
 
-  const VideoContentDescription* video = content->as_video();
-
   VideoSenderParameters send_params = last_send_params_;
-  RtpSendParametersFromMediaDescription(video, extensions_filter(),
+  RtpSendParametersFromMediaDescription(content, extensions_filter(),
                                         &send_params);
   send_params.mid = mid();
-  send_params.conference_mode = video->conference_mode();
+  send_params.conference_mode = content->conference_mode();
 
   VideoReceiverParameters recv_params = last_recv_params_;
 
@@ -1145,15 +1143,15 @@ bool VideoChannel::SetRemoteContent_w(const MediaContentDescription* content,
   // instead.
   bool needs_recv_params_update = false;
   if (type == SdpType::kAnswer || type == SdpType::kPrAnswer) {
-    webrtc::flat_set<const VideoCodec*> matched_codecs;
-    for (VideoCodec& recv_codec : recv_params.codecs) {
-      if (absl::c_any_of(matched_codecs, [&](const VideoCodec* c) {
+    webrtc::flat_set<const Codec*> matched_codecs;
+    for (Codec& recv_codec : recv_params.codecs) {
+      if (absl::c_any_of(matched_codecs, [&](const Codec* c) {
             return recv_codec.Matches(*c);
           })) {
         continue;
       }
 
-      std::vector<const VideoCodec*> send_codecs =
+      std::vector<const Codec*> send_codecs =
           FindAllMatchingCodecs(send_params.codecs, recv_codec);
       if (send_codecs.empty()) {
         continue;
@@ -1161,7 +1159,7 @@ bool VideoChannel::SetRemoteContent_w(const MediaContentDescription* content,
 
       bool may_ignore_packetization = false;
       bool has_matching_packetization = false;
-      for (const VideoCodec* send_codec : send_codecs) {
+      for (const Codec* send_codec : send_codecs) {
         if (!send_codec->packetization.has_value() &&
             recv_codec.packetization.has_value()) {
           may_ignore_packetization = true;
diff --git a/third_party/libwebrtc/pc/channel_unittest.cc b/third_party/libwebrtc/pc/channel_unittest.cc
index c675cd0446..98a61ea673 100644
--- a/third_party/libwebrtc/pc/channel_unittest.cc
+++ b/third_party/libwebrtc/pc/channel_unittest.cc
@@ -2094,12 +2094,12 @@ TEST_F(VideoChannelSingleThreadTest, TestSetLocalOfferWithPacketization) {
   EXPECT_TRUE(channel1_->SetLocalContent(&video, SdpType::kOffer, err));
   EXPECT_THAT(media_send_channel1_impl()->send_codecs(), testing::IsEmpty());
   ASSERT_THAT(media_receive_channel1_impl()->recv_codecs(), testing::SizeIs(2));
-  EXPECT_TRUE(media_receive_channel1_impl()->recv_codecs()[0].Matches(
-      kVp8Codec, &field_trials_));
+  EXPECT_TRUE(
+      media_receive_channel1_impl()->recv_codecs()[0].Matches(kVp8Codec));
   EXPECT_EQ(media_receive_channel1_impl()->recv_codecs()[0].packetization,
             absl::nullopt);
-  EXPECT_TRUE(media_receive_channel1_impl()->recv_codecs()[1].Matches(
-      vp9_codec, &field_trials_));
+  EXPECT_TRUE(
+      media_receive_channel1_impl()->recv_codecs()[1].Matches(vp9_codec));
   EXPECT_EQ(media_receive_channel1_impl()->recv_codecs()[1].packetization,
             cricket::kPacketizationParamRaw);
 }
@@ -2118,12 +2118,10 @@ TEST_F(VideoChannelSingleThreadTest, TestSetRemoteOfferWithPacketization) {
   EXPECT_TRUE(err.empty());
   EXPECT_THAT(media_receive_channel1_impl()->recv_codecs(), testing::IsEmpty());
   ASSERT_THAT(media_send_channel1_impl()->send_codecs(), testing::SizeIs(2));
-  EXPECT_TRUE(media_send_channel1_impl()->send_codecs()[0].Matches(
-      kVp8Codec, &field_trials_));
+  EXPECT_TRUE(media_send_channel1_impl()->send_codecs()[0].Matches(kVp8Codec));
   EXPECT_EQ(media_send_channel1_impl()->send_codecs()[0].packetization,
             absl::nullopt);
-  EXPECT_TRUE(media_send_channel1_impl()->send_codecs()[1].Matches(
-      vp9_codec, &field_trials_));
+  EXPECT_TRUE(media_send_channel1_impl()->send_codecs()[1].Matches(vp9_codec));
   EXPECT_EQ(media_send_channel1_impl()->send_codecs()[1].packetization,
             cricket::kPacketizationParamRaw);
 }
@@ -2143,21 +2141,19 @@ TEST_F(VideoChannelSingleThreadTest, TestSetAnswerWithPacketization) {
   EXPECT_TRUE(channel1_->SetRemoteContent(&video, SdpType::kAnswer, err));
   EXPECT_TRUE(err.empty());
   ASSERT_THAT(media_receive_channel1_impl()->recv_codecs(), testing::SizeIs(2));
-  EXPECT_TRUE(media_receive_channel1_impl()->recv_codecs()[0].Matches(
-      kVp8Codec, &field_trials_));
+  EXPECT_TRUE(
+      media_receive_channel1_impl()->recv_codecs()[0].Matches(kVp8Codec));
   EXPECT_EQ(media_receive_channel1_impl()->recv_codecs()[0].packetization,
             absl::nullopt);
-  EXPECT_TRUE(media_receive_channel1_impl()->recv_codecs()[1].Matches(
-      vp9_codec, &field_trials_));
+  EXPECT_TRUE(
+      media_receive_channel1_impl()->recv_codecs()[1].Matches(vp9_codec));
   EXPECT_EQ(media_receive_channel1_impl()->recv_codecs()[1].packetization,
             cricket::kPacketizationParamRaw);
   EXPECT_THAT(media_send_channel1_impl()->send_codecs(), testing::SizeIs(2));
-  EXPECT_TRUE(media_send_channel1_impl()->send_codecs()[0].Matches(
-      kVp8Codec, &field_trials_));
+  EXPECT_TRUE(media_send_channel1_impl()->send_codecs()[0].Matches(kVp8Codec));
   EXPECT_EQ(media_send_channel1_impl()->send_codecs()[0].packetization,
             absl::nullopt);
-  EXPECT_TRUE(media_send_channel1_impl()->send_codecs()[1].Matches(
-      vp9_codec, &field_trials_));
+  EXPECT_TRUE(media_send_channel1_impl()->send_codecs()[1].Matches(vp9_codec));
   EXPECT_EQ(media_send_channel1_impl()->send_codecs()[1].packetization,
             cricket::kPacketizationParamRaw);
 }
diff --git a/third_party/libwebrtc/pc/connection_context.cc b/third_party/libwebrtc/pc/connection_context.cc
index f436e27c0a..df4522bf13 100644
--- a/third_party/libwebrtc/pc/connection_context.cc
+++ b/third_party/libwebrtc/pc/connection_context.cc
@@ -14,9 +14,11 @@
 #include <utility>
 #include <vector>
 
+#include "api/environment/environment.h"
 #include "api/transport/field_trial_based_config.h"
 #include "media/base/media_engine.h"
 #include "media/sctp/sctp_transport_factory.h"
+#include "pc/media_factory.h"
 #include "rtc_base/helpers.h"
 #include "rtc_base/internal/default_socket_server.h"
 #include "rtc_base/socket_server.h"
@@ -77,12 +79,14 @@ std::unique_ptr<SctpTransportFactoryInterface> MaybeCreateSctpFactory(
 
 // Static
 rtc::scoped_refptr<ConnectionContext> ConnectionContext::Create(
+    const Environment& env,
     PeerConnectionFactoryDependencies* dependencies) {
   return rtc::scoped_refptr<ConnectionContext>(
-      new ConnectionContext(dependencies));
+      new ConnectionContext(env, dependencies));
 }
 
 ConnectionContext::ConnectionContext(
+    const Environment& env,
     PeerConnectionFactoryDependencies* dependencies)
     : network_thread_(MaybeStartNetworkThread(dependencies->network_thread,
                                               owned_socket_factory_,
@@ -96,18 +100,21 @@ ConnectionContext::ConnectionContext(
                      }),
       signaling_thread_(MaybeWrapThread(dependencies->signaling_thread,
                                         wraps_current_thread_)),
-      trials_(dependencies->trials ? std::move(dependencies->trials)
-                                   : std::make_unique<FieldTrialBasedConfig>()),
-      media_engine_(std::move(dependencies->media_engine)),
+      env_(env),
+      media_engine_(
+          dependencies->media_factory != nullptr
+              ? dependencies->media_factory->CreateMediaEngine(env_,
+                                                               *dependencies)
+              : nullptr),
       network_monitor_factory_(
           std::move(dependencies->network_monitor_factory)),
       default_network_manager_(std::move(dependencies->network_manager)),
-      call_factory_(std::move(dependencies->call_factory)),
+      call_factory_(std::move(dependencies->media_factory)),
       default_socket_factory_(std::move(dependencies->packet_socket_factory)),
       sctp_factory_(
           MaybeCreateSctpFactory(std::move(dependencies->sctp_factory),
                                  network_thread(),
-                                 *trials_.get())),
+                                 env_.field_trials())),
       use_rtx_(true) {
   RTC_DCHECK_RUN_ON(signaling_thread_);
   RTC_DCHECK(!(default_network_manager_ && network_monitor_factory_))
@@ -150,7 +157,7 @@ ConnectionContext::ConnectionContext(
     // If network_monitor_factory_ is non-null, it will be used to create a
     // network monitor while on the network thread.
     default_network_manager_ = std::make_unique<rtc::BasicNetworkManager>(
-        network_monitor_factory_.get(), socket_factory, &field_trials());
+        network_monitor_factory_.get(), socket_factory, &env_.field_trials());
   }
   if (!default_socket_factory_) {
     default_socket_factory_ =
diff --git a/third_party/libwebrtc/pc/connection_context.h b/third_party/libwebrtc/pc/connection_context.h
index 399e7c2b45..893a3b0e52 100644
--- a/third_party/libwebrtc/pc/connection_context.h
+++ b/third_party/libwebrtc/pc/connection_context.h
@@ -15,6 +15,7 @@
 #include <string>
 
 #include "api/call/call_factory_interface.h"
+#include "api/environment/environment.h"
 #include "api/field_trials_view.h"
 #include "api/media_stream_interface.h"
 #include "api/peer_connection_interface.h"
@@ -39,8 +40,6 @@ class UniqueRandomIdGenerator;
 
 namespace webrtc {
 
-class RtcEventLog;
-
 // This class contains resources needed by PeerConnection and associated
 // objects. A reference to this object is passed to each PeerConnection. The
 // methods on this object are assumed not to change the state in any way that
@@ -54,6 +53,7 @@ class ConnectionContext final
   // The Dependencies class allows simple management of all new dependencies
   // being added to the ConnectionContext.
   static rtc::scoped_refptr<ConnectionContext> Create(
+      const Environment& env,
       PeerConnectionFactoryDependencies* dependencies);
 
   // This class is not copyable or movable.
@@ -76,11 +76,16 @@ class ConnectionContext final
   rtc::Thread* network_thread() { return network_thread_; }
   const rtc::Thread* network_thread() const { return network_thread_; }
 
+  // Environment associated with the PeerConnectionFactory.
+  // Note: environments are different for different PeerConnections,
+  // but they are not supposed to change after creating the PeerConnection.
+  const Environment& env() const { return env_; }
+
   // Field trials associated with the PeerConnectionFactory.
   // Note: that there can be different field trials for different
   // PeerConnections (but they are not supposed change after creating the
   // PeerConnection).
-  const FieldTrialsView& field_trials() const { return *trials_.get(); }
+  const FieldTrialsView& field_trials() const { return env_.field_trials(); }
 
   // Accessors only used from the PeerConnectionFactory class
   rtc::NetworkManager* default_network_manager() {
@@ -91,7 +96,7 @@ class ConnectionContext final
     RTC_DCHECK_RUN_ON(signaling_thread_);
     return default_socket_factory_.get();
   }
-  CallFactoryInterface* call_factory() {
+  MediaFactory* call_factory() {
     RTC_DCHECK_RUN_ON(worker_thread());
     return call_factory_.get();
   }
@@ -106,7 +111,8 @@ class ConnectionContext final
   void set_use_rtx(bool use_rtx) { use_rtx_ = use_rtx; }
 
  protected:
-  explicit ConnectionContext(PeerConnectionFactoryDependencies* dependencies);
+  ConnectionContext(const Environment& env,
+                    PeerConnectionFactoryDependencies* dependencies);
 
   friend class rtc::RefCountedNonVirtual<ConnectionContext>;
   ~ConnectionContext();
@@ -122,8 +128,7 @@ class ConnectionContext final
   AlwaysValidPointer<rtc::Thread> const worker_thread_;
   rtc::Thread* const signaling_thread_;
 
-  // Accessed both on signaling thread and worker thread.
-  std::unique_ptr<FieldTrialsView> const trials_;
+  const Environment env_;
 
   // This object is const over the lifetime of the ConnectionContext, and is
   // only altered in the destructor.
@@ -138,7 +143,7 @@ class ConnectionContext final
       RTC_GUARDED_BY(signaling_thread_);
   std::unique_ptr<rtc::NetworkManager> default_network_manager_
       RTC_GUARDED_BY(signaling_thread_);
-  std::unique_ptr<webrtc::CallFactoryInterface> const call_factory_
+  std::unique_ptr<MediaFactory> const call_factory_
       RTC_GUARDED_BY(worker_thread());
 
   std::unique_ptr<rtc::PacketSocketFactory> default_socket_factory_
diff --git a/third_party/libwebrtc/pc/data_channel_controller_unittest.cc b/third_party/libwebrtc/pc/data_channel_controller_unittest.cc
index 3b8adb6819..7d4e60467e 100644
--- a/third_party/libwebrtc/pc/data_channel_controller_unittest.cc
+++ b/third_party/libwebrtc/pc/data_channel_controller_unittest.cc
@@ -27,7 +27,7 @@ namespace {
 using ::testing::NiceMock;
 using ::testing::Return;
 
-class MockDataChannelTransport : public webrtc::DataChannelTransportInterface {
+class MockDataChannelTransport : public DataChannelTransportInterface {
  public:
   ~MockDataChannelTransport() override {}
 
diff --git a/third_party/libwebrtc/pc/data_channel_integrationtest.cc b/third_party/libwebrtc/pc/data_channel_integrationtest.cc
index faec76d03e..5a8004c72a 100644
--- a/third_party/libwebrtc/pc/data_channel_integrationtest.cc
+++ b/third_party/libwebrtc/pc/data_channel_integrationtest.cc
@@ -90,7 +90,7 @@ class FakeClockForTest : public rtc::ScopedFakeClock {
     // Some things use a time of "0" as a special value, so we need to start out
     // the fake clock at a nonzero time.
     // TODO(deadbeef): Fix this.
-    AdvanceTime(webrtc::TimeDelta::Seconds(1));
+    AdvanceTime(TimeDelta::Seconds(1));
   }
 
   // Explicit handle.
@@ -422,7 +422,7 @@ TEST_P(DataChannelIntegrationTest, CalleeClosesSctpDataChannel) {
 TEST_P(DataChannelIntegrationTest, SctpDataChannelConfigSentToOtherSide) {
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();
-  webrtc::DataChannelInit init;
+  DataChannelInit init;
   init.id = 53;
   init.maxRetransmits = 52;
   caller()->CreateDataChannel("data-channel", &init);
@@ -453,7 +453,7 @@ TEST_P(DataChannelIntegrationTest, StressTestUnorderedSctpDataChannel) {
   // Normal procedure, but with unordered data channel config.
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();
-  webrtc::DataChannelInit init;
+  DataChannelInit init;
   init.ordered = false;
   caller()->CreateDataChannel(&init);
   caller()->CreateAndSetAndSignalOffer();
@@ -515,7 +515,7 @@ TEST_P(DataChannelIntegrationTest, StressTestOpenCloseChannelNoDelay) {
   const size_t kIterations = 10;
   bool has_negotiated = false;
 
-  webrtc::DataChannelInit init;
+  DataChannelInit init;
   for (size_t repeats = 0; repeats < kIterations; ++repeats) {
     RTC_LOG(LS_INFO) << "Iteration " << (repeats + 1) << "/" << kIterations;
 
@@ -592,7 +592,7 @@ TEST_P(DataChannelIntegrationTest, StressTestOpenCloseChannelWithDelay) {
   const size_t kIterations = 10;
   bool has_negotiated = false;
 
-  webrtc::DataChannelInit init;
+  DataChannelInit init;
   for (size_t repeats = 0; repeats < kIterations; ++repeats) {
     RTC_LOG(LS_INFO) << "Iteration " << (repeats + 1) << "/" << kIterations;
 
diff --git a/third_party/libwebrtc/pc/data_channel_unittest.cc b/third_party/libwebrtc/pc/data_channel_unittest.cc
index 9b84a1be61..a27a66c3de 100644
--- a/third_party/libwebrtc/pc/data_channel_unittest.cc
+++ b/third_party/libwebrtc/pc/data_channel_unittest.cc
@@ -81,8 +81,7 @@ class SctpDataChannelTest : public ::testing::Test {
         controller_(new FakeDataChannelController(&network_thread_)) {
     network_thread_.Start();
     inner_channel_ = controller_->CreateDataChannel("test", init_);
-    channel_ =
-        webrtc::SctpDataChannel::CreateProxy(inner_channel_, signaling_safety_);
+    channel_ = SctpDataChannel::CreateProxy(inner_channel_, signaling_safety_);
   }
   ~SctpDataChannelTest() override {
     run_loop_.Flush();
@@ -510,7 +509,7 @@ TEST_F(SctpDataChannelTest, LateCreatedChannelTransitionToOpen) {
   SetChannelReady();
   InternalDataChannelInit init;
   init.id = 1;
-  auto dc = webrtc::SctpDataChannel::CreateProxy(
+  auto dc = SctpDataChannel::CreateProxy(
       controller_->CreateDataChannel("test1", init), signaling_safety_);
   EXPECT_EQ(DataChannelInterface::kOpen, dc->state());
 }
@@ -524,7 +523,7 @@ TEST_F(SctpDataChannelTest, SendUnorderedAfterReceivesOpenAck) {
   init.ordered = false;
   rtc::scoped_refptr<SctpDataChannel> dc =
       controller_->CreateDataChannel("test1", init);
-  auto proxy = webrtc::SctpDataChannel::CreateProxy(dc, signaling_safety_);
+  auto proxy = SctpDataChannel::CreateProxy(dc, signaling_safety_);
 
   EXPECT_EQ_WAIT(DataChannelInterface::kOpen, proxy->state(), 1000);
 
@@ -553,7 +552,7 @@ TEST_F(SctpDataChannelTest, DeprecatedSendUnorderedAfterReceivesOpenAck) {
   init.ordered = false;
   rtc::scoped_refptr<SctpDataChannel> dc =
       controller_->CreateDataChannel("test1", init);
-  auto proxy = webrtc::SctpDataChannel::CreateProxy(dc, signaling_safety_);
+  auto proxy = SctpDataChannel::CreateProxy(dc, signaling_safety_);
 
   EXPECT_EQ_WAIT(DataChannelInterface::kOpen, proxy->state(), 1000);
 
@@ -582,7 +581,7 @@ TEST_F(SctpDataChannelTest, SendUnorderedAfterReceiveData) {
   init.ordered = false;
   rtc::scoped_refptr<SctpDataChannel> dc =
       controller_->CreateDataChannel("test1", init);
-  auto proxy = webrtc::SctpDataChannel::CreateProxy(dc, signaling_safety_);
+  auto proxy = SctpDataChannel::CreateProxy(dc, signaling_safety_);
 
   EXPECT_EQ_WAIT(DataChannelInterface::kOpen, proxy->state(), 1000);
 
@@ -605,7 +604,7 @@ TEST_F(SctpDataChannelTest, DeprecatedSendUnorderedAfterReceiveData) {
   init.ordered = false;
   rtc::scoped_refptr<SctpDataChannel> dc =
       controller_->CreateDataChannel("test1", init);
-  auto proxy = webrtc::SctpDataChannel::CreateProxy(dc, signaling_safety_);
+  auto proxy = SctpDataChannel::CreateProxy(dc, signaling_safety_);
 
   EXPECT_EQ_WAIT(DataChannelInterface::kOpen, proxy->state(), 1000);
 
@@ -714,7 +713,7 @@ TEST_F(SctpDataChannelTest, NoMsgSentIfNegotiatedAndNotFromOpenMsg) {
   SetChannelReady();
   rtc::scoped_refptr<SctpDataChannel> dc =
       controller_->CreateDataChannel("test1", config);
-  auto proxy = webrtc::SctpDataChannel::CreateProxy(dc, signaling_safety_);
+  auto proxy = SctpDataChannel::CreateProxy(dc, signaling_safety_);
 
   EXPECT_EQ_WAIT(DataChannelInterface::kOpen, proxy->state(), 1000);
   EXPECT_EQ(0, controller_->last_sid());
@@ -779,7 +778,7 @@ TEST_F(SctpDataChannelTest, OpenAckSentIfCreatedFromOpenMessage) {
   SetChannelReady();
   rtc::scoped_refptr<SctpDataChannel> dc =
       controller_->CreateDataChannel("test1", config);
-  auto proxy = webrtc::SctpDataChannel::CreateProxy(dc, signaling_safety_);
+  auto proxy = SctpDataChannel::CreateProxy(dc, signaling_safety_);
 
   EXPECT_EQ_WAIT(DataChannelInterface::kOpen, proxy->state(), 1000);
 
diff --git a/third_party/libwebrtc/pc/dtls_srtp_transport.h b/third_party/libwebrtc/pc/dtls_srtp_transport.h
index 0f8338ca0d..995809ed4b 100644
--- a/third_party/libwebrtc/pc/dtls_srtp_transport.h
+++ b/third_party/libwebrtc/pc/dtls_srtp_transport.h
@@ -49,15 +49,6 @@ class DtlsSrtpTransport : public SrtpTransport {
 
   void SetOnDtlsStateChange(std::function<void(void)> callback);
 
-  RTCError SetSrtpSendKey(const cricket::CryptoParams& params) override {
-    return RTCError(RTCErrorType::UNSUPPORTED_OPERATION,
-                    "Set SRTP keys for DTLS-SRTP is not supported.");
-  }
-  RTCError SetSrtpReceiveKey(const cricket::CryptoParams& params) override {
-    return RTCError(RTCErrorType::UNSUPPORTED_OPERATION,
-                    "Set SRTP keys for DTLS-SRTP is not supported.");
-  }
-
   // If `active_reset_srtp_params_` is set to be true, the SRTP parameters will
   // be reset whenever the DtlsTransports are reset.
   void SetActiveResetSrtpParams(bool active_reset_srtp_params) {
diff --git a/third_party/libwebrtc/pc/ice_server_parsing_unittest.cc b/third_party/libwebrtc/pc/ice_server_parsing_unittest.cc
index 4356b1efb0..a38638e507 100644
--- a/third_party/libwebrtc/pc/ice_server_parsing_unittest.cc
+++ b/third_party/libwebrtc/pc/ice_server_parsing_unittest.cc
@@ -62,9 +62,7 @@ class IceServerParsingTest : public ::testing::Test {
     server.tls_cert_policy = tls_certificate_policy;
     server.hostname = hostname;
     servers.push_back(server);
-    return webrtc::ParseIceServersOrError(servers, &stun_servers_,
-                                          &turn_servers_)
-        .ok();
+    return ParseIceServersOrError(servers, &stun_servers_, &turn_servers_).ok();
   }
 
  protected:
@@ -233,8 +231,7 @@ TEST_F(IceServerParsingTest, ParseMultipleUrls) {
   server.password = "bar";
   servers.push_back(server);
   EXPECT_TRUE(
-      webrtc::ParseIceServersOrError(servers, &stun_servers_, &turn_servers_)
-          .ok());
+      ParseIceServersOrError(servers, &stun_servers_, &turn_servers_).ok());
   EXPECT_EQ(1U, stun_servers_.size());
   EXPECT_EQ(1U, turn_servers_.size());
 }
diff --git a/third_party/libwebrtc/pc/ice_transport_unittest.cc b/third_party/libwebrtc/pc/ice_transport_unittest.cc
index aaf9f2e57a..a42c107072 100644
--- a/third_party/libwebrtc/pc/ice_transport_unittest.cc
+++ b/third_party/libwebrtc/pc/ice_transport_unittest.cc
@@ -32,7 +32,7 @@ class IceTransportTest : public ::testing::Test {
 
   rtc::SocketServer* socket_server() const { return socket_server_.get(); }
 
-  webrtc::test::ScopedKeyValueConfig field_trials_;
+  test::ScopedKeyValueConfig field_trials_;
 
  private:
   std::unique_ptr<rtc::SocketServer> socket_server_;
diff --git a/third_party/libwebrtc/pc/jsep_transport_controller.cc b/third_party/libwebrtc/pc/jsep_transport_controller.cc
index 7c669a5ae3..d5d1cd24a9 100644
--- a/third_party/libwebrtc/pc/jsep_transport_controller.cc
+++ b/third_party/libwebrtc/pc/jsep_transport_controller.cc
@@ -76,14 +76,18 @@ JsepTransportController::~JsepTransportController() {
 
 RTCError JsepTransportController::SetLocalDescription(
     SdpType type,
-    const cricket::SessionDescription* description) {
+    const cricket::SessionDescription* local_desc,
+    const cricket::SessionDescription* remote_desc) {
+  RTC_DCHECK(local_desc);
   TRACE_EVENT0("webrtc", "JsepTransportController::SetLocalDescription");
+
   if (!network_thread_->IsCurrent()) {
     return network_thread_->BlockingCall(
-        [=] { return SetLocalDescription(type, description); });
+        [=] { return SetLocalDescription(type, local_desc, remote_desc); });
   }
 
   RTC_DCHECK_RUN_ON(network_thread_);
+
   if (!initial_offerer_.has_value()) {
     initial_offerer_.emplace(type == SdpType::kOffer);
     if (*initial_offerer_) {
@@ -92,20 +96,22 @@ RTCError JsepTransportController::SetLocalDescription(
       SetIceRole_n(cricket::ICEROLE_CONTROLLED);
     }
   }
-  return ApplyDescription_n(/*local=*/true, type, description);
+  return ApplyDescription_n(/*local=*/true, type, local_desc, remote_desc);
 }
 
 RTCError JsepTransportController::SetRemoteDescription(
     SdpType type,
-    const cricket::SessionDescription* description) {
+    const cricket::SessionDescription* local_desc,
+    const cricket::SessionDescription* remote_desc) {
+  RTC_DCHECK(remote_desc);
   TRACE_EVENT0("webrtc", "JsepTransportController::SetRemoteDescription");
   if (!network_thread_->IsCurrent()) {
     return network_thread_->BlockingCall(
-        [=] { return SetRemoteDescription(type, description); });
+        [=] { return SetRemoteDescription(type, local_desc, remote_desc); });
   }
 
   RTC_DCHECK_RUN_ON(network_thread_);
-  return ApplyDescription_n(/*local=*/false, type, description);
+  return ApplyDescription_n(/*local=*/false, type, local_desc, remote_desc);
 }
 
 RtpTransportInternal* JsepTransportController::GetRtpTransport(
@@ -148,7 +154,7 @@ JsepTransportController::GetRtcpDtlsTransport(const std::string& mid) const {
   return jsep_transport->rtcp_dtls_transport();
 }
 
-rtc::scoped_refptr<webrtc::DtlsTransport>
+rtc::scoped_refptr<DtlsTransport>
 JsepTransportController::LookupDtlsTransportByMid(const std::string& mid) {
   RTC_DCHECK_RUN_ON(network_thread_);
   auto jsep_transport = GetJsepTransportForMid(mid);
@@ -383,7 +389,7 @@ RTCError JsepTransportController::RollbackTransports() {
   return RTCError::OK();
 }
 
-rtc::scoped_refptr<webrtc::IceTransportInterface>
+rtc::scoped_refptr<IceTransportInterface>
 JsepTransportController::CreateIceTransport(const std::string& transport_name,
                                             bool rtcp) {
   int component = rtcp ? cricket::ICE_CANDIDATE_COMPONENT_RTCP
@@ -433,29 +439,43 @@ JsepTransportController::CreateDtlsTransport(
       this, &JsepTransportController::OnTransportWritableState_n);
   dtls->SignalReceivingState.connect(
       this, &JsepTransportController::OnTransportReceivingState_n);
-  dtls->ice_transport()->SignalGatheringState.connect(
-      this, &JsepTransportController::OnTransportGatheringState_n);
+  dtls->ice_transport()->SetGatheringStateCallback(
+      [this](cricket::IceTransportInternal* transport) {
+        RTC_DCHECK_RUN_ON(network_thread_);
+        OnTransportGatheringState_n(transport);
+      });
   dtls->ice_transport()->SignalCandidateGathered.connect(
       this, &JsepTransportController::OnTransportCandidateGathered_n);
-  dtls->ice_transport()->SignalCandidateError.connect(
-      this, &JsepTransportController::OnTransportCandidateError_n);
-  dtls->ice_transport()->SignalCandidatesRemoved.connect(
-      this, &JsepTransportController::OnTransportCandidatesRemoved_n);
+  dtls->ice_transport()->SetCandidateErrorCallback(
+      [this](cricket::IceTransportInternal* transport,
+             const cricket::IceCandidateErrorEvent& error) {
+        RTC_DCHECK_RUN_ON(network_thread_);
+        OnTransportCandidateError_n(transport, error);
+      });
+  dtls->ice_transport()->SetCandidatesRemovedCallback(
+      [this](cricket::IceTransportInternal* transport,
+             const cricket::Candidates& candidates) {
+        RTC_DCHECK_RUN_ON(network_thread_);
+        OnTransportCandidatesRemoved_n(transport, candidates);
+      });
   dtls->ice_transport()->SignalRoleConflict.connect(
       this, &JsepTransportController::OnTransportRoleConflict_n);
   dtls->ice_transport()->SignalStateChanged.connect(
       this, &JsepTransportController::OnTransportStateChanged_n);
   dtls->ice_transport()->SignalIceTransportStateChanged.connect(
       this, &JsepTransportController::OnTransportStateChanged_n);
-  dtls->ice_transport()->SignalCandidatePairChanged.connect(
-      this, &JsepTransportController::OnTransportCandidatePairChanged_n);
+  dtls->ice_transport()->SetCandidatePairChangeCallback(
+      [this](const cricket::CandidatePairChangeEvent& event) {
+        RTC_DCHECK_RUN_ON(network_thread_);
+        OnTransportCandidatePairChanged_n(event);
+      });
 
   dtls->SubscribeDtlsHandshakeError(
       [this](rtc::SSLHandshakeError error) { OnDtlsHandshakeError(error); });
   return dtls;
 }
 
-std::unique_ptr<webrtc::RtpTransport>
+std::unique_ptr<RtpTransport>
 JsepTransportController::CreateUnencryptedRtpTransport(
     const std::string& transport_name,
     rtc::PacketTransportInternal* rtp_packet_transport,
@@ -470,13 +490,12 @@ JsepTransportController::CreateUnencryptedRtpTransport(
   return unencrypted_rtp_transport;
 }
 
-std::unique_ptr<webrtc::SrtpTransport>
-JsepTransportController::CreateSdesTransport(
+std::unique_ptr<SrtpTransport> JsepTransportController::CreateSdesTransport(
     const std::string& transport_name,
     cricket::DtlsTransportInternal* rtp_dtls_transport,
     cricket::DtlsTransportInternal* rtcp_dtls_transport) {
   RTC_DCHECK_RUN_ON(network_thread_);
-  auto srtp_transport = std::make_unique<webrtc::SrtpTransport>(
+  auto srtp_transport = std::make_unique<SrtpTransport>(
       rtcp_dtls_transport == nullptr, *config_.field_trials);
   RTC_DCHECK(rtp_dtls_transport);
   srtp_transport->SetRtpPacketTransport(rtp_dtls_transport);
@@ -489,13 +508,13 @@ JsepTransportController::CreateSdesTransport(
   return srtp_transport;
 }
 
-std::unique_ptr<webrtc::DtlsSrtpTransport>
+std::unique_ptr<DtlsSrtpTransport>
 JsepTransportController::CreateDtlsSrtpTransport(
     const std::string& transport_name,
     cricket::DtlsTransportInternal* rtp_dtls_transport,
     cricket::DtlsTransportInternal* rtcp_dtls_transport) {
   RTC_DCHECK_RUN_ON(network_thread_);
-  auto dtls_srtp_transport = std::make_unique<webrtc::DtlsSrtpTransport>(
+  auto dtls_srtp_transport = std::make_unique<DtlsSrtpTransport>(
       rtcp_dtls_transport == nullptr, *config_.field_trials);
   if (config_.enable_external_auth) {
     dtls_srtp_transport->EnableExternalAuth();
@@ -550,18 +569,20 @@ JsepTransportController::GetActiveDtlsTransports() {
 RTCError JsepTransportController::ApplyDescription_n(
     bool local,
     SdpType type,
-    const cricket::SessionDescription* description) {
+    const cricket::SessionDescription* local_desc,
+    const cricket::SessionDescription* remote_desc) {
   TRACE_EVENT0("webrtc", "JsepTransportController::ApplyDescription_n");
-  RTC_DCHECK(description);
 
-  if (local) {
-    local_desc_ = description;
-  } else {
-    remote_desc_ = description;
-  }
+  // Stash away the description object that we'll be applying (since this
+  // function is used for both local and remote).
+  const cricket::SessionDescription* description =
+      local ? local_desc : remote_desc;
+
+  RTC_DCHECK(description);
 
   RTCError error;
-  error = ValidateAndMaybeUpdateBundleGroups(local, type, description);
+  error =
+      ValidateAndMaybeUpdateBundleGroups(local, type, local_desc, remote_desc);
   if (!error.ok()) {
     return error;
   }
@@ -673,7 +694,11 @@ RTCError JsepTransportController::ApplyDescription_n(
 RTCError JsepTransportController::ValidateAndMaybeUpdateBundleGroups(
     bool local,
     SdpType type,
-    const cricket::SessionDescription* description) {
+    const cricket::SessionDescription* local_desc,
+    const cricket::SessionDescription* remote_desc) {
+  const cricket::SessionDescription* description =
+      local ? local_desc : remote_desc;
+
   RTC_DCHECK(description);
 
   std::vector<const cricket::ContentGroup*> new_bundle_groups =
@@ -739,72 +764,74 @@ RTCError JsepTransportController::ValidateAndMaybeUpdateBundleGroups(
       }
     }
   } else if (type == SdpType::kAnswer) {
-    std::vector<const cricket::ContentGroup*> offered_bundle_groups =
-        local ? remote_desc_->GetGroupsByName(cricket::GROUP_TYPE_BUNDLE)
-              : local_desc_->GetGroupsByName(cricket::GROUP_TYPE_BUNDLE);
-
-    std::map<std::string, const cricket::ContentGroup*>
-        offered_bundle_groups_by_mid;
-    for (const cricket::ContentGroup* offered_bundle_group :
-         offered_bundle_groups) {
-      for (const std::string& content_name :
-           offered_bundle_group->content_names()) {
-        offered_bundle_groups_by_mid[content_name] = offered_bundle_group;
+    if ((local && remote_desc) || (!local && local_desc)) {
+      std::vector<const cricket::ContentGroup*> offered_bundle_groups =
+          local ? remote_desc->GetGroupsByName(cricket::GROUP_TYPE_BUNDLE)
+                : local_desc->GetGroupsByName(cricket::GROUP_TYPE_BUNDLE);
+
+      std::map<std::string, const cricket::ContentGroup*>
+          offered_bundle_groups_by_mid;
+      for (const cricket::ContentGroup* offered_bundle_group :
+           offered_bundle_groups) {
+        for (const std::string& content_name :
+             offered_bundle_group->content_names()) {
+          offered_bundle_groups_by_mid[content_name] = offered_bundle_group;
+        }
       }
-    }
 
-    std::map<const cricket::ContentGroup*, const cricket::ContentGroup*>
-        new_bundle_groups_by_offered_bundle_groups;
-    for (const cricket::ContentGroup* new_bundle_group : new_bundle_groups) {
-      if (!new_bundle_group->FirstContentName()) {
-        // Empty groups could be a subset of any group.
-        continue;
-      }
-      // The group in the answer (new_bundle_group) must have a corresponding
-      // group in the offer (original_group), because the answer groups may only
-      // be subsets of the offer groups.
-      auto it = offered_bundle_groups_by_mid.find(
-          *new_bundle_group->FirstContentName());
-      if (it == offered_bundle_groups_by_mid.end()) {
-        return RTCError(RTCErrorType::INVALID_PARAMETER,
-                        "A BUNDLE group was added in the answer that did not "
-                        "exist in the offer.");
-      }
-      const cricket::ContentGroup* offered_bundle_group = it->second;
-      if (new_bundle_groups_by_offered_bundle_groups.find(
-              offered_bundle_group) !=
-          new_bundle_groups_by_offered_bundle_groups.end()) {
-        return RTCError(RTCErrorType::INVALID_PARAMETER,
-                        "A MID in the answer has changed group.");
-      }
-      new_bundle_groups_by_offered_bundle_groups.insert(
-          std::make_pair(offered_bundle_group, new_bundle_group));
-      for (const std::string& content_name :
-           new_bundle_group->content_names()) {
-        it = offered_bundle_groups_by_mid.find(content_name);
-        // The BUNDLE group in answer should be a subset of offered group.
-        if (it == offered_bundle_groups_by_mid.end() ||
-            it->second != offered_bundle_group) {
+      std::map<const cricket::ContentGroup*, const cricket::ContentGroup*>
+          new_bundle_groups_by_offered_bundle_groups;
+      for (const cricket::ContentGroup* new_bundle_group : new_bundle_groups) {
+        if (!new_bundle_group->FirstContentName()) {
+          // Empty groups could be a subset of any group.
+          continue;
+        }
+        // The group in the answer (new_bundle_group) must have a corresponding
+        // group in the offer (original_group), because the answer groups may
+        // only be subsets of the offer groups.
+        auto it = offered_bundle_groups_by_mid.find(
+            *new_bundle_group->FirstContentName());
+        if (it == offered_bundle_groups_by_mid.end()) {
           return RTCError(RTCErrorType::INVALID_PARAMETER,
-                          "A BUNDLE group in answer contains a MID='" +
-                              content_name +
-                              "' that was not in the offered group.");
+                          "A BUNDLE group was added in the answer that did not "
+                          "exist in the offer.");
         }
-      }
-    }
-
-    for (const auto& bundle_group : bundles_.bundle_groups()) {
-      for (const std::string& content_name : bundle_group->content_names()) {
-        // An answer that removes m= sections from pre-negotiated BUNDLE group
-        // without rejecting it, is invalid.
-        auto it = new_bundle_groups_by_mid.find(content_name);
-        if (it == new_bundle_groups_by_mid.end()) {
-          auto* content_info = description->GetContentByName(content_name);
-          if (!content_info || !content_info->rejected) {
+        const cricket::ContentGroup* offered_bundle_group = it->second;
+        if (new_bundle_groups_by_offered_bundle_groups.find(
+                offered_bundle_group) !=
+            new_bundle_groups_by_offered_bundle_groups.end()) {
+          return RTCError(RTCErrorType::INVALID_PARAMETER,
+                          "A MID in the answer has changed group.");
+        }
+        new_bundle_groups_by_offered_bundle_groups.insert(
+            std::make_pair(offered_bundle_group, new_bundle_group));
+        for (const std::string& content_name :
+             new_bundle_group->content_names()) {
+          it = offered_bundle_groups_by_mid.find(content_name);
+          // The BUNDLE group in answer should be a subset of offered group.
+          if (it == offered_bundle_groups_by_mid.end() ||
+              it->second != offered_bundle_group) {
             return RTCError(RTCErrorType::INVALID_PARAMETER,
-                            "Answer cannot remove m= section with mid='" +
+                            "A BUNDLE group in answer contains a MID='" +
                                 content_name +
-                                "' from already-established BUNDLE group.");
+                                "' that was not in the offered group.");
+          }
+        }
+      }
+
+      for (const auto& bundle_group : bundles_.bundle_groups()) {
+        for (const std::string& content_name : bundle_group->content_names()) {
+          // An answer that removes m= sections from pre-negotiated BUNDLE group
+          // without rejecting it, is invalid.
+          auto it = new_bundle_groups_by_mid.find(content_name);
+          if (it == new_bundle_groups_by_mid.end()) {
+            auto* content_info = description->GetContentByName(content_name);
+            if (!content_info || !content_info->rejected) {
+              return RTCError(RTCErrorType::INVALID_PARAMETER,
+                              "Answer cannot remove m= section with mid='" +
+                                  content_name +
+                                  "' from already-established BUNDLE group.");
+            }
           }
         }
       }
@@ -985,13 +1012,12 @@ int JsepTransportController::GetRtpAbsSendTimeHeaderExtensionId(
   const cricket::MediaContentDescription* content_desc =
       content_info.media_description();
 
-  const webrtc::RtpExtension* send_time_extension =
-      webrtc::RtpExtension::FindHeaderExtensionByUri(
-          content_desc->rtp_header_extensions(),
-          webrtc::RtpExtension::kAbsSendTimeUri,
+  const RtpExtension* send_time_extension =
+      RtpExtension::FindHeaderExtensionByUri(
+          content_desc->rtp_header_extensions(), RtpExtension::kAbsSendTimeUri,
           config_.crypto_options.srtp.enable_encrypted_rtp_header_extensions
-              ? webrtc::RtpExtension::kPreferEncryptedExtension
-              : webrtc::RtpExtension::kDiscardEncryptedExtension);
+              ? RtpExtension::kPreferEncryptedExtension
+              : RtpExtension::kDiscardEncryptedExtension);
   return send_time_extension ? send_time_extension->id : -1;
 }
 
@@ -1039,7 +1065,7 @@ RTCError JsepTransportController::MaybeCreateJsepTransport(
                     "SDES and DTLS-SRTP cannot be enabled at the same time.");
   }
 
-  rtc::scoped_refptr<webrtc::IceTransportInterface> ice =
+  rtc::scoped_refptr<IceTransportInterface> ice =
       CreateIceTransport(content_info.name, /*rtcp=*/false);
 
   std::unique_ptr<cricket::DtlsTransportInternal> rtp_dtls_transport =
@@ -1050,7 +1076,7 @@ RTCError JsepTransportController::MaybeCreateJsepTransport(
   std::unique_ptr<SrtpTransport> sdes_transport;
   std::unique_ptr<DtlsSrtpTransport> dtls_srtp_transport;
 
-  rtc::scoped_refptr<webrtc::IceTransportInterface> rtcp_ice;
+  rtc::scoped_refptr<IceTransportInterface> rtcp_ice;
   if (config_.rtcp_mux_policy !=
           PeerConnectionInterface::kRtcpMuxPolicyRequire &&
       content_info.type == cricket::MediaProtocolType::kRtp) {
@@ -1096,7 +1122,7 @@ RTCError JsepTransportController::MaybeCreateJsepTransport(
         OnRtcpPacketReceived_n(buffer, packet_time_ms);
       });
   jsep_transport->rtp_transport()->SetUnDemuxableRtpPacketReceivedHandler(
-      [this](webrtc::RtpPacketReceived& packet) {
+      [this](RtpPacketReceived& packet) {
         RTC_DCHECK_RUN_ON(network_thread_);
         OnUnDemuxableRtpPacketReceived_n(packet);
       });
@@ -1421,7 +1447,7 @@ void JsepTransportController::OnRtcpPacketReceived_n(
 }
 
 void JsepTransportController::OnUnDemuxableRtpPacketReceived_n(
-    const webrtc::RtpPacketReceived& packet) {
+    const RtpPacketReceived& packet) {
   RTC_DCHECK(config_.un_demuxable_packet_handler);
   config_.un_demuxable_packet_handler(packet);
 }
diff --git a/third_party/libwebrtc/pc/jsep_transport_controller.h b/third_party/libwebrtc/pc/jsep_transport_controller.h
index 5880e346cd..448844ac79 100644
--- a/third_party/libwebrtc/pc/jsep_transport_controller.h
+++ b/third_party/libwebrtc/pc/jsep_transport_controller.h
@@ -112,7 +112,7 @@ class JsepTransportController : public sigslot::has_slots<> {
     rtc::SSLProtocolVersion ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12;
     // `crypto_options` is used to determine if created DTLS transports
     // negotiate GCM crypto suites or not.
-    webrtc::CryptoOptions crypto_options;
+    CryptoOptions crypto_options;
     PeerConnectionInterface::BundlePolicy bundle_policy =
         PeerConnectionInterface::kBundlePolicyBalanced;
     PeerConnectionInterface::RtcpMuxPolicy rtcp_mux_policy =
@@ -120,7 +120,7 @@ class JsepTransportController : public sigslot::has_slots<> {
     bool disable_encryption = false;
     bool enable_external_auth = false;
     // Used to inject the ICE/DTLS transports created externally.
-    webrtc::IceTransportFactory* ice_transport_factory = nullptr;
+    IceTransportFactory* ice_transport_factory = nullptr;
     cricket::DtlsTransportFactory* dtls_transport_factory = nullptr;
     Observer* transport_observer = nullptr;
     // Must be provided and valid for the lifetime of the
@@ -140,7 +140,7 @@ class JsepTransportController : public sigslot::has_slots<> {
     std::function<void(rtc::SSLHandshakeError)> on_dtls_handshake_error_;
 
     // Field trials.
-    const webrtc::FieldTrialsView* field_trials;
+    const FieldTrialsView* field_trials;
   };
 
   // The ICE related events are fired on the `network_thread`.
@@ -161,11 +161,24 @@ class JsepTransportController : public sigslot::has_slots<> {
   // level, creating/destroying transport objects as needed and updating their
   // properties. This includes RTP, DTLS, and ICE (but not SCTP). At least not
   // yet? May make sense to in the future.
+  //
+  // `local_desc` must always be valid. If a remote description has previously
+  // been set via a call to `SetRemoteDescription()` then `remote_desc` should
+  // point to that description object in order to keep the current local and
+  // remote session descriptions in sync.
   RTCError SetLocalDescription(SdpType type,
-                               const cricket::SessionDescription* description);
-
+                               const cricket::SessionDescription* local_desc,
+                               const cricket::SessionDescription* remote_desc);
+
+  // Call to apply a remote description (See `SetLocalDescription()` for local).
+  //
+  // `remote_desc` must always be valid. If a local description has previously
+  // been set via a call to `SetLocalDescription()` then `local_desc` should
+  // point to that description object in order to keep the current local and
+  // remote session descriptions in sync.
   RTCError SetRemoteDescription(SdpType type,
-                                const cricket::SessionDescription* description);
+                                const cricket::SessionDescription* local_desc,
+                                const cricket::SessionDescription* remote_desc);
 
   // Get transports to be used for the provided `mid`. If bundling is enabled,
   // calling GetRtpTransport for multiple MIDs may yield the same object.
@@ -174,7 +187,7 @@ class JsepTransportController : public sigslot::has_slots<> {
   const cricket::DtlsTransportInternal* GetRtcpDtlsTransport(
       const std::string& mid) const;
   // Gets the externally sharable version of the DtlsTransport.
-  rtc::scoped_refptr<webrtc::DtlsTransport> LookupDtlsTransportByMid(
+  rtc::scoped_refptr<DtlsTransport> LookupDtlsTransportByMid(
       const std::string& mid);
   rtc::scoped_refptr<SctpTransport> GetSctpTransport(
       const std::string& mid) const;
@@ -325,14 +338,23 @@ class JsepTransportController : public sigslot::has_slots<> {
   CallbackList<const cricket::CandidatePairChangeEvent&>
       signal_ice_candidate_pair_changed_ RTC_GUARDED_BY(network_thread_);
 
+  // Called from SetLocalDescription and SetRemoteDescription.
+  // When `local` is true, local_desc must be valid. Similarly when
+  // `local` is false, remote_desc must be valid. The description counterpart
+  // to the one that's being applied, may be nullptr but when it's supplied
+  // the counterpart description's content groups will  be kept up to date for
+  // `type == SdpType::kAnswer`.
   RTCError ApplyDescription_n(bool local,
                               SdpType type,
-                              const cricket::SessionDescription* description)
+                              const cricket::SessionDescription* local_desc,
+                              const cricket::SessionDescription* remote_desc)
       RTC_RUN_ON(network_thread_);
   RTCError ValidateAndMaybeUpdateBundleGroups(
       bool local,
       SdpType type,
-      const cricket::SessionDescription* description);
+      const cricket::SessionDescription* local_desc,
+      const cricket::SessionDescription* remote_desc)
+      RTC_RUN_ON(network_thread_);
   RTCError ValidateContent(const cricket::ContentInfo& content_info);
 
   void HandleRejectedContent(const cricket::ContentInfo& content_info)
@@ -399,19 +421,19 @@ class JsepTransportController : public sigslot::has_slots<> {
   std::unique_ptr<cricket::DtlsTransportInternal> CreateDtlsTransport(
       const cricket::ContentInfo& content_info,
       cricket::IceTransportInternal* ice);
-  rtc::scoped_refptr<webrtc::IceTransportInterface> CreateIceTransport(
+  rtc::scoped_refptr<IceTransportInterface> CreateIceTransport(
       const std::string& transport_name,
       bool rtcp);
 
-  std::unique_ptr<webrtc::RtpTransport> CreateUnencryptedRtpTransport(
+  std::unique_ptr<RtpTransport> CreateUnencryptedRtpTransport(
       const std::string& transport_name,
       rtc::PacketTransportInternal* rtp_packet_transport,
       rtc::PacketTransportInternal* rtcp_packet_transport);
-  std::unique_ptr<webrtc::SrtpTransport> CreateSdesTransport(
+  std::unique_ptr<SrtpTransport> CreateSdesTransport(
       const std::string& transport_name,
       cricket::DtlsTransportInternal* rtp_dtls_transport,
       cricket::DtlsTransportInternal* rtcp_dtls_transport);
-  std::unique_ptr<webrtc::DtlsSrtpTransport> CreateDtlsSrtpTransport(
+  std::unique_ptr<DtlsSrtpTransport> CreateDtlsSrtpTransport(
       const std::string& transport_name,
       cricket::DtlsTransportInternal* rtp_dtls_transport,
       cricket::DtlsTransportInternal* rtcp_dtls_transport);
@@ -453,7 +475,7 @@ class JsepTransportController : public sigslot::has_slots<> {
   void OnRtcpPacketReceived_n(rtc::CopyOnWriteBuffer* packet,
                               int64_t packet_time_us)
       RTC_RUN_ON(network_thread_);
-  void OnUnDemuxableRtpPacketReceived_n(const webrtc::RtpPacketReceived& packet)
+  void OnUnDemuxableRtpPacketReceived_n(const RtpPacketReceived& packet)
       RTC_RUN_ON(network_thread_);
 
   void OnDtlsHandshakeError(rtc::SSLHandshakeError error);
@@ -481,8 +503,6 @@ class JsepTransportController : public sigslot::has_slots<> {
   const Config config_;
   bool active_reset_srtp_params_ RTC_GUARDED_BY(network_thread_);
 
-  const cricket::SessionDescription* local_desc_ = nullptr;
-  const cricket::SessionDescription* remote_desc_ = nullptr;
   absl::optional<bool> initial_offerer_;
 
   cricket::IceConfig ice_config_;
diff --git a/third_party/libwebrtc/pc/jsep_transport_controller_unittest.cc b/third_party/libwebrtc/pc/jsep_transport_controller_unittest.cc
index faa8842e35..7696d82be8 100644
--- a/third_party/libwebrtc/pc/jsep_transport_controller_unittest.cc
+++ b/third_party/libwebrtc/pc/jsep_transport_controller_unittest.cc
@@ -56,7 +56,7 @@ static const char kDataMid1[] = "data1";
 
 namespace webrtc {
 
-class FakeIceTransportFactory : public webrtc::IceTransportFactory {
+class FakeIceTransportFactory : public IceTransportFactory {
  public:
   ~FakeIceTransportFactory() override = default;
   rtc::scoped_refptr<IceTransportInterface> CreateIceTransport(
@@ -72,7 +72,7 @@ class FakeDtlsTransportFactory : public cricket::DtlsTransportFactory {
  public:
   std::unique_ptr<cricket::DtlsTransportInternal> CreateDtlsTransport(
       cricket::IceTransportInternal* ice,
-      const webrtc::CryptoOptions& crypto_options,
+      const CryptoOptions& crypto_options,
       rtc::SSLProtocolVersion max_version) override {
     return std::make_unique<FakeDtlsTransport>(
         static_cast<cricket::FakeIceTransport*>(ice));
@@ -265,9 +265,10 @@ class JsepTransportControllerTest : public JsepTransportController::Observer,
     }
 
     auto description = CreateSessionDescriptionWithBundleGroup();
-    EXPECT_TRUE(transport_controller_
-                    ->SetLocalDescription(SdpType::kOffer, description.get())
-                    .ok());
+    EXPECT_TRUE(
+        transport_controller_
+            ->SetLocalDescription(SdpType::kOffer, description.get(), nullptr)
+            .ok());
 
     transport_controller_->MaybeStartGathering();
     auto fake_audio_dtls = static_cast<FakeDtlsTransport*>(
@@ -379,15 +380,16 @@ class JsepTransportControllerTest : public JsepTransportController::Observer,
   // Transport controller needs to be destroyed first, because it may issue
   // callbacks that modify the changed_*_by_mid in the destructor.
   std::unique_ptr<JsepTransportController> transport_controller_;
-  webrtc::test::ScopedKeyValueConfig field_trials_;
+  test::ScopedKeyValueConfig field_trials_;
 };
 
 TEST_F(JsepTransportControllerTest, GetRtpTransport) {
   CreateJsepTransportController(JsepTransportController::Config());
   auto description = CreateSessionDescriptionWithoutBundle();
-  EXPECT_TRUE(transport_controller_
-                  ->SetLocalDescription(SdpType::kOffer, description.get())
-                  .ok());
+  EXPECT_TRUE(
+      transport_controller_
+          ->SetLocalDescription(SdpType::kOffer, description.get(), nullptr)
+          .ok());
   auto audio_rtp_transport = transport_controller_->GetRtpTransport(kAudioMid1);
   auto video_rtp_transport = transport_controller_->GetRtpTransport(kVideoMid1);
   EXPECT_NE(nullptr, audio_rtp_transport);
@@ -402,9 +404,10 @@ TEST_F(JsepTransportControllerTest, GetDtlsTransport) {
   config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyNegotiate;
   CreateJsepTransportController(std::move(config));
   auto description = CreateSessionDescriptionWithoutBundle();
-  EXPECT_TRUE(transport_controller_
-                  ->SetLocalDescription(SdpType::kOffer, description.get())
-                  .ok());
+  EXPECT_TRUE(
+      transport_controller_
+          ->SetLocalDescription(SdpType::kOffer, description.get(), nullptr)
+          .ok());
   EXPECT_NE(nullptr, transport_controller_->GetDtlsTransport(kAudioMid1));
   EXPECT_NE(nullptr, transport_controller_->GetRtcpDtlsTransport(kAudioMid1));
   EXPECT_NE(nullptr,
@@ -425,7 +428,7 @@ TEST_F(JsepTransportControllerTest, GetDtlsTransport) {
   // and verify that the resulting container is empty.
   auto dtls_transport =
       transport_controller_->LookupDtlsTransportByMid(kVideoMid1);
-  webrtc::DtlsTransport* my_transport =
+  DtlsTransport* my_transport =
       static_cast<DtlsTransport*>(dtls_transport.get());
   EXPECT_NE(nullptr, my_transport->internal());
   transport_controller_.reset();
@@ -437,9 +440,10 @@ TEST_F(JsepTransportControllerTest, GetDtlsTransportWithRtcpMux) {
   config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyRequire;
   CreateJsepTransportController(std::move(config));
   auto description = CreateSessionDescriptionWithoutBundle();
-  EXPECT_TRUE(transport_controller_
-                  ->SetLocalDescription(SdpType::kOffer, description.get())
-                  .ok());
+  EXPECT_TRUE(
+      transport_controller_
+          ->SetLocalDescription(SdpType::kOffer, description.get(), nullptr)
+          .ok());
   EXPECT_NE(nullptr, transport_controller_->GetDtlsTransport(kAudioMid1));
   EXPECT_EQ(nullptr, transport_controller_->GetRtcpDtlsTransport(kAudioMid1));
   EXPECT_NE(nullptr, transport_controller_->GetDtlsTransport(kVideoMid1));
@@ -449,9 +453,10 @@ TEST_F(JsepTransportControllerTest, GetDtlsTransportWithRtcpMux) {
 TEST_F(JsepTransportControllerTest, SetIceConfig) {
   CreateJsepTransportController(JsepTransportController::Config());
   auto description = CreateSessionDescriptionWithoutBundle();
-  EXPECT_TRUE(transport_controller_
-                  ->SetLocalDescription(SdpType::kOffer, description.get())
-                  .ok());
+  EXPECT_TRUE(
+      transport_controller_
+          ->SetLocalDescription(SdpType::kOffer, description.get(), nullptr)
+          .ok());
 
   transport_controller_->SetIceConfig(
       CreateIceConfig(kTimeout, cricket::GATHER_CONTINUALLY));
@@ -467,9 +472,10 @@ TEST_F(JsepTransportControllerTest, SetIceConfig) {
                   cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
                   nullptr);
 
-  EXPECT_TRUE(transport_controller_
-                  ->SetLocalDescription(SdpType::kOffer, description.get())
-                  .ok());
+  EXPECT_TRUE(
+      transport_controller_
+          ->SetLocalDescription(SdpType::kOffer, description.get(), nullptr)
+          .ok());
   fake_audio_dtls = static_cast<FakeDtlsTransport*>(
       transport_controller_->GetDtlsTransport(kAudioMid2));
   ASSERT_NE(nullptr, fake_audio_dtls);
@@ -482,11 +488,14 @@ TEST_F(JsepTransportControllerTest, SetIceConfig) {
 TEST_F(JsepTransportControllerTest, NeedIceRestart) {
   CreateJsepTransportController(JsepTransportController::Config());
   auto description = CreateSessionDescriptionWithoutBundle();
+  EXPECT_TRUE(
+      transport_controller_
+          ->SetLocalDescription(SdpType::kOffer, description.get(), nullptr)
+          .ok());
+  // TODO(tommi): Note that _now_ we set `remote`. (was not set before).
   EXPECT_TRUE(transport_controller_
-                  ->SetLocalDescription(SdpType::kOffer, description.get())
-                  .ok());
-  EXPECT_TRUE(transport_controller_
-                  ->SetRemoteDescription(SdpType::kAnswer, description.get())
+                  ->SetRemoteDescription(SdpType::kAnswer, description.get(),
+                                         description.get())
                   .ok());
 
   // Initially NeedsIceRestart should return false.
@@ -505,7 +514,8 @@ TEST_F(JsepTransportControllerTest, NeedIceRestart) {
   audio_transport_info->description.ice_ufrag = kIceUfrag2;
   audio_transport_info->description.ice_pwd = kIcePwd2;
   EXPECT_TRUE(transport_controller_
-                  ->SetLocalDescription(SdpType::kOffer, description.get())
+                  ->SetLocalDescription(SdpType::kOffer, description.get(),
+                                        description.get())
                   .ok());
   // Because the ICE is only restarted for audio, NeedsIceRestart is expected to
   // return false for audio and true for video.
@@ -516,9 +526,10 @@ TEST_F(JsepTransportControllerTest, NeedIceRestart) {
 TEST_F(JsepTransportControllerTest, MaybeStartGathering) {
   CreateJsepTransportController(JsepTransportController::Config());
   auto description = CreateSessionDescriptionWithBundleGroup();
-  EXPECT_TRUE(transport_controller_
-                  ->SetLocalDescription(SdpType::kOffer, description.get())
-                  .ok());
+  EXPECT_TRUE(
+      transport_controller_
+          ->SetLocalDescription(SdpType::kOffer, description.get(), nullptr)
+          .ok());
   // After setting the local description, we should be able to start gathering
   // candidates.
   transport_controller_->MaybeStartGathering();
@@ -529,10 +540,10 @@ TEST_F(JsepTransportControllerTest, MaybeStartGathering) {
 TEST_F(JsepTransportControllerTest, AddRemoveRemoteCandidates) {
   CreateJsepTransportController(JsepTransportController::Config());
   auto description = CreateSessionDescriptionWithoutBundle();
-  transport_controller_->SetLocalDescription(SdpType::kOffer,
-                                             description.get());
-  transport_controller_->SetRemoteDescription(SdpType::kAnswer,
-                                              description.get());
+  transport_controller_->SetLocalDescription(SdpType::kOffer, description.get(),
+                                             nullptr);
+  transport_controller_->SetRemoteDescription(
+      SdpType::kAnswer, description.get(), description.get());
   auto fake_audio_dtls = static_cast<FakeDtlsTransport*>(
       transport_controller_->GetDtlsTransport(kAudioMid1));
   ASSERT_NE(nullptr, fake_audio_dtls);
@@ -565,9 +576,10 @@ TEST_F(JsepTransportControllerTest, SetAndGetLocalCertificate) {
   // Apply the local certificate.
   EXPECT_TRUE(transport_controller_->SetLocalCertificate(certificate1));
   // Apply the local description.
-  EXPECT_TRUE(transport_controller_
-                  ->SetLocalDescription(SdpType::kOffer, description.get())
-                  .ok());
+  EXPECT_TRUE(
+      transport_controller_
+          ->SetLocalDescription(SdpType::kOffer, description.get(), nullptr)
+          .ok());
   returned_certificate = transport_controller_->GetLocalCertificate(kAudioMid1);
   EXPECT_TRUE(returned_certificate);
   EXPECT_EQ(certificate1->identity()->certificate().ToPEMString(),
@@ -586,9 +598,10 @@ TEST_F(JsepTransportControllerTest, SetAndGetLocalCertificate) {
 TEST_F(JsepTransportControllerTest, GetRemoteSSLCertChain) {
   CreateJsepTransportController(JsepTransportController::Config());
   auto description = CreateSessionDescriptionWithBundleGroup();
-  EXPECT_TRUE(transport_controller_
-                  ->SetLocalDescription(SdpType::kOffer, description.get())
-                  .ok());
+  EXPECT_TRUE(
+      transport_controller_
+          ->SetLocalDescription(SdpType::kOffer, description.get(), nullptr)
+          .ok());
   rtc::FakeSSLCertificate fake_certificate("fake_data");
 
   auto fake_audio_dtls = static_cast<FakeDtlsTransport*>(
@@ -622,16 +635,18 @@ TEST_F(JsepTransportControllerTest, GetDtlsRole) {
                   cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_PASSIVE,
                   answer_certificate);
 
-  EXPECT_TRUE(transport_controller_
-                  ->SetLocalDescription(SdpType::kOffer, offer_desc.get())
-                  .ok());
+  EXPECT_TRUE(
+      transport_controller_
+          ->SetLocalDescription(SdpType::kOffer, offer_desc.get(), nullptr)
+          .ok());
 
   absl::optional<rtc::SSLRole> role =
       transport_controller_->GetDtlsRole(kAudioMid1);
   // The DTLS role is not decided yet.
   EXPECT_FALSE(role);
   EXPECT_TRUE(transport_controller_
-                  ->SetRemoteDescription(SdpType::kAnswer, answer_desc.get())
+                  ->SetRemoteDescription(SdpType::kAnswer, offer_desc.get(),
+                                         answer_desc.get())
                   .ok());
   role = transport_controller_->GetDtlsRole(kAudioMid1);
 
@@ -642,9 +657,10 @@ TEST_F(JsepTransportControllerTest, GetDtlsRole) {
 TEST_F(JsepTransportControllerTest, GetStats) {
   CreateJsepTransportController(JsepTransportController::Config());
   auto description = CreateSessionDescriptionWithBundleGroup();
-  EXPECT_TRUE(transport_controller_
-                  ->SetLocalDescription(SdpType::kOffer, description.get())
-                  .ok());
+  EXPECT_TRUE(
+      transport_controller_
+          ->SetLocalDescription(SdpType::kOffer, description.get(), nullptr)
+          .ok());
 
   cricket::TransportStats stats;
   EXPECT_TRUE(transport_controller_->GetStats(kAudioMid1, &stats));
@@ -657,9 +673,10 @@ TEST_F(JsepTransportControllerTest, GetStats) {
 TEST_F(JsepTransportControllerTest, SignalConnectionStateFailed) {
   CreateJsepTransportController(JsepTransportController::Config());
   auto description = CreateSessionDescriptionWithoutBundle();
-  EXPECT_TRUE(transport_controller_
-                  ->SetLocalDescription(SdpType::kOffer, description.get())
-                  .ok());
+  EXPECT_TRUE(
+      transport_controller_
+          ->SetLocalDescription(SdpType::kOffer, description.get(), nullptr)
+          .ok());
 
   auto fake_ice = static_cast<cricket::FakeIceTransport*>(
       transport_controller_->GetDtlsTransport(kAudioMid1)->ice_transport());
@@ -681,9 +698,10 @@ TEST_F(JsepTransportControllerTest,
        SignalConnectionStateConnectedNoMediaTransport) {
   CreateJsepTransportController(JsepTransportController::Config());
   auto description = CreateSessionDescriptionWithoutBundle();
-  EXPECT_TRUE(transport_controller_
-                  ->SetLocalDescription(SdpType::kOffer, description.get())
-                  .ok());
+  EXPECT_TRUE(
+      transport_controller_
+          ->SetLocalDescription(SdpType::kOffer, description.get(), nullptr)
+          .ok());
 
   auto fake_audio_dtls = static_cast<FakeDtlsTransport*>(
       transport_controller_->GetDtlsTransport(kAudioMid1));
@@ -729,9 +747,10 @@ TEST_F(JsepTransportControllerTest,
 TEST_F(JsepTransportControllerTest, SignalConnectionStateComplete) {
   CreateJsepTransportController(JsepTransportController::Config());
   auto description = CreateSessionDescriptionWithoutBundle();
-  EXPECT_TRUE(transport_controller_
-                  ->SetLocalDescription(SdpType::kOffer, description.get())
-                  .ok());
+  EXPECT_TRUE(
+      transport_controller_
+          ->SetLocalDescription(SdpType::kOffer, description.get(), nullptr)
+          .ok());
 
   auto fake_audio_dtls = static_cast<FakeDtlsTransport*>(
       transport_controller_->GetDtlsTransport(kAudioMid1));
@@ -788,9 +807,10 @@ TEST_F(JsepTransportControllerTest, SignalConnectionStateComplete) {
 TEST_F(JsepTransportControllerTest, SignalIceGatheringStateGathering) {
   CreateJsepTransportController(JsepTransportController::Config());
   auto description = CreateSessionDescriptionWithoutBundle();
-  EXPECT_TRUE(transport_controller_
-                  ->SetLocalDescription(SdpType::kOffer, description.get())
-                  .ok());
+  EXPECT_TRUE(
+      transport_controller_
+          ->SetLocalDescription(SdpType::kOffer, description.get(), nullptr)
+          .ok());
 
   auto fake_audio_dtls = static_cast<FakeDtlsTransport*>(
       transport_controller_->GetDtlsTransport(kAudioMid1));
@@ -803,9 +823,10 @@ TEST_F(JsepTransportControllerTest, SignalIceGatheringStateGathering) {
 TEST_F(JsepTransportControllerTest, SignalIceGatheringStateComplete) {
   CreateJsepTransportController(JsepTransportController::Config());
   auto description = CreateSessionDescriptionWithoutBundle();
-  EXPECT_TRUE(transport_controller_
-                  ->SetLocalDescription(SdpType::kOffer, description.get())
-                  .ok());
+  EXPECT_TRUE(
+      transport_controller_
+          ->SetLocalDescription(SdpType::kOffer, description.get(), nullptr)
+          .ok());
 
   auto fake_audio_dtls = static_cast<FakeDtlsTransport*>(
       transport_controller_->GetDtlsTransport(kAudioMid1));
@@ -838,9 +859,10 @@ TEST_F(JsepTransportControllerTest,
        SignalingWhenLastIncompleteTransportDestroyed) {
   CreateJsepTransportController(JsepTransportController::Config());
   auto description = CreateSessionDescriptionWithBundleGroup();
-  EXPECT_TRUE(transport_controller_
-                  ->SetLocalDescription(SdpType::kOffer, description.get())
-                  .ok());
+  EXPECT_TRUE(
+      transport_controller_
+          ->SetLocalDescription(SdpType::kOffer, description.get(), nullptr)
+          .ok());
 
   auto fake_audio_dtls = static_cast<FakeDtlsTransport*>(
       transport_controller_->GetDtlsTransport(kAudioMid1));
@@ -861,7 +883,8 @@ TEST_F(JsepTransportControllerTest,
 
   // Set the remote description and enable the bundle.
   EXPECT_TRUE(transport_controller_
-                  ->SetRemoteDescription(SdpType::kAnswer, description.get())
+                  ->SetRemoteDescription(SdpType::kAnswer, description.get(),
+                                         description.get())
                   .ok());
   // The BUNDLE should be enabled, the incomplete video transport should be
   // deleted and the states should be updated.
@@ -887,11 +910,13 @@ TEST_F(JsepTransportControllerTest,
   AddAudioSection(description.get(), kAudioMid1, kIceUfrag1, kIcePwd1,
                   cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
                   nullptr);
+  EXPECT_TRUE(
+      transport_controller_
+          ->SetLocalDescription(SdpType::kOffer, description.get(), nullptr)
+          .ok());
   EXPECT_TRUE(transport_controller_
-                  ->SetLocalDescription(SdpType::kOffer, description.get())
-                  .ok());
-  EXPECT_TRUE(transport_controller_
-                  ->SetRemoteDescription(SdpType::kAnswer, description.get())
+                  ->SetRemoteDescription(SdpType::kAnswer, description.get(),
+                                         description.get())
                   .ok());
 
   // Trigger and verify initial non-new states.
@@ -899,7 +924,7 @@ TEST_F(JsepTransportControllerTest,
       transport_controller_->GetDtlsTransport(kAudioMid1));
   fake_audio_dtls->fake_ice_transport()->MaybeStartGathering();
   fake_audio_dtls->fake_ice_transport()->SetTransportState(
-      webrtc::IceTransportState::kChecking,
+      IceTransportState::kChecking,
       cricket::IceTransportState::STATE_CONNECTING);
   EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionChecking,
                  ice_connection_state_, kTimeout);
@@ -914,7 +939,8 @@ TEST_F(JsepTransportControllerTest,
   // to "new".
   description->contents()[0].rejected = true;
   EXPECT_TRUE(transport_controller_
-                  ->SetRemoteDescription(SdpType::kOffer, description.get())
+                  ->SetRemoteDescription(SdpType::kOffer, description.get(),
+                                         description.get())
                   .ok());
   EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionNew,
                  ice_connection_state_, kTimeout);
@@ -941,9 +967,10 @@ TEST_F(JsepTransportControllerTest,
 TEST_F(JsepTransportControllerTest, SignalCandidatesGathered) {
   CreateJsepTransportController(JsepTransportController::Config());
   auto description = CreateSessionDescriptionWithBundleGroup();
-  EXPECT_TRUE(transport_controller_
-                  ->SetLocalDescription(SdpType::kOffer, description.get())
-                  .ok());
+  EXPECT_TRUE(
+      transport_controller_
+          ->SetLocalDescription(SdpType::kOffer, description.get(), nullptr)
+          .ok());
   transport_controller_->MaybeStartGathering();
 
   auto fake_audio_dtls = static_cast<FakeDtlsTransport*>(
@@ -998,11 +1025,13 @@ TEST_F(JsepTransportControllerTest, IceRoleNotRedetermined) {
                   cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_PASSIVE,
                   nullptr);
 
+  EXPECT_TRUE(
+      transport_controller_
+          ->SetRemoteDescription(SdpType::kOffer, nullptr, remote_offer.get())
+          .ok());
   EXPECT_TRUE(transport_controller_
-                  ->SetRemoteDescription(SdpType::kOffer, remote_offer.get())
-                  .ok());
-  EXPECT_TRUE(transport_controller_
-                  ->SetLocalDescription(SdpType::kAnswer, local_answer.get())
+                  ->SetLocalDescription(SdpType::kAnswer, local_answer.get(),
+                                        remote_offer.get())
                   .ok());
 
   auto fake_dtls = static_cast<FakeDtlsTransport*>(
@@ -1015,10 +1044,11 @@ TEST_F(JsepTransportControllerTest, IceRoleNotRedetermined) {
   AddAudioSection(restart_local_offer.get(), kAudioMid1, kIceUfrag3, kIcePwd3,
                   cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
                   nullptr);
-  EXPECT_TRUE(
-      transport_controller_
-          ->SetLocalDescription(SdpType::kOffer, restart_local_offer.get())
-          .ok());
+  EXPECT_TRUE(transport_controller_
+                  ->SetLocalDescription(SdpType::kOffer,
+                                        restart_local_offer.get(),
+                                        remote_offer.get())
+                  .ok());
   EXPECT_EQ(cricket::ICEROLE_CONTROLLED,
             fake_dtls->fake_ice_transport()->GetIceRole());
 }
@@ -1030,9 +1060,10 @@ TEST_F(JsepTransportControllerTest, SetIceRoleWhenIceLiteInRemoteAnswer) {
   AddAudioSection(local_offer.get(), kAudioMid1, kIceUfrag1, kIcePwd1,
                   cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
                   nullptr);
-  EXPECT_TRUE(transport_controller_
-                  ->SetLocalDescription(SdpType::kOffer, local_offer.get())
-                  .ok());
+  EXPECT_TRUE(
+      transport_controller_
+          ->SetLocalDescription(SdpType::kOffer, local_offer.get(), nullptr)
+          .ok());
   auto fake_dtls = static_cast<FakeDtlsTransport*>(
       transport_controller_->GetDtlsTransport(kAudioMid1));
   EXPECT_EQ(cricket::ICEROLE_CONTROLLING,
@@ -1045,7 +1076,8 @@ TEST_F(JsepTransportControllerTest, SetIceRoleWhenIceLiteInRemoteAnswer) {
                   cricket::ICEMODE_LITE, cricket::CONNECTIONROLE_PASSIVE,
                   nullptr);
   EXPECT_TRUE(transport_controller_
-                  ->SetRemoteDescription(SdpType::kAnswer, remote_answer.get())
+                  ->SetRemoteDescription(SdpType::kAnswer, local_offer.get(),
+                                         remote_answer.get())
                   .ok());
   EXPECT_EQ(cricket::ICEROLE_CONTROLLING,
             fake_dtls->fake_ice_transport()->GetIceRole());
@@ -1069,11 +1101,13 @@ TEST_F(JsepTransportControllerTest,
                   nullptr);
   // Initial Offer/Answer exchange. If the remote offerer is ICE-Lite, then the
   // local side is the controlling.
+  EXPECT_TRUE(
+      transport_controller_
+          ->SetRemoteDescription(SdpType::kOffer, nullptr, remote_offer.get())
+          .ok());
   EXPECT_TRUE(transport_controller_
-                  ->SetRemoteDescription(SdpType::kOffer, remote_offer.get())
-                  .ok());
-  EXPECT_TRUE(transport_controller_
-                  ->SetLocalDescription(SdpType::kAnswer, local_answer.get())
+                  ->SetLocalDescription(SdpType::kAnswer, local_answer.get(),
+                                        remote_offer.get())
                   .ok());
   auto fake_dtls = static_cast<FakeDtlsTransport*>(
       transport_controller_->GetDtlsTransport(kAudioMid1));
@@ -1085,15 +1119,17 @@ TEST_F(JsepTransportControllerTest,
   AddAudioSection(remote_offer2.get(), kAudioMid1, kIceUfrag2, kIcePwd2,
                   cricket::ICEMODE_LITE, cricket::CONNECTIONROLE_ACTPASS,
                   nullptr);
+  EXPECT_TRUE(transport_controller_
+                  ->SetRemoteDescription(SdpType::kOffer, local_answer.get(),
+                                         remote_offer2.get())
+                  .ok());
   auto local_answer2 = std::make_unique<cricket::SessionDescription>();
   AddAudioSection(local_answer2.get(), kAudioMid1, kIceUfrag2, kIcePwd2,
                   cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_PASSIVE,
                   nullptr);
   EXPECT_TRUE(transport_controller_
-                  ->SetRemoteDescription(SdpType::kOffer, remote_offer2.get())
-                  .ok());
-  EXPECT_TRUE(transport_controller_
-                  ->SetLocalDescription(SdpType::kAnswer, local_answer2.get())
+                  ->SetLocalDescription(SdpType::kAnswer, local_answer2.get(),
+                                        remote_offer2.get())
                   .ok());
   fake_dtls = static_cast<FakeDtlsTransport*>(
       transport_controller_->GetDtlsTransport(kAudioMid1));
@@ -1145,11 +1181,13 @@ TEST_F(JsepTransportControllerTest, MultipleMediaSectionsOfSameTypeWithBundle) {
   local_offer->AddGroup(bundle_group);
   remote_answer->AddGroup(bundle_group);
 
+  EXPECT_TRUE(
+      transport_controller_
+          ->SetLocalDescription(SdpType::kOffer, local_offer.get(), nullptr)
+          .ok());
   EXPECT_TRUE(transport_controller_
-                  ->SetLocalDescription(SdpType::kOffer, local_offer.get())
-                  .ok());
-  EXPECT_TRUE(transport_controller_
-                  ->SetRemoteDescription(SdpType::kAnswer, remote_answer.get())
+                  ->SetRemoteDescription(SdpType::kAnswer, local_offer.get(),
+                                         remote_answer.get())
                   .ok());
   // Verify that all the sections are bundled on kAudio1.
   auto transport1 = transport_controller_->GetRtpTransport(kAudioMid1);
@@ -1224,11 +1262,13 @@ TEST_F(JsepTransportControllerTest, MultipleBundleGroups) {
   remote_answer->AddGroup(bundle_group1);
   remote_answer->AddGroup(bundle_group2);
 
+  EXPECT_TRUE(
+      transport_controller_
+          ->SetLocalDescription(SdpType::kOffer, local_offer.get(), nullptr)
+          .ok());
   EXPECT_TRUE(transport_controller_
-                  ->SetLocalDescription(SdpType::kOffer, local_offer.get())
-                  .ok());
-  EXPECT_TRUE(transport_controller_
-                  ->SetRemoteDescription(SdpType::kAnswer, remote_answer.get())
+                  ->SetRemoteDescription(SdpType::kAnswer, local_offer.get(),
+                                         remote_answer.get())
                   .ok());
 
   // Verify that (kMid1Audio,kMid2Video) and (kMid3Audio,kMid4Video) form two
@@ -1307,11 +1347,13 @@ TEST_F(JsepTransportControllerTest,
   // endpoint that does not have support for multiple BUNDLE groups.
   remote_answer->AddGroup(bundle_group1);
 
+  EXPECT_TRUE(
+      transport_controller_
+          ->SetLocalDescription(SdpType::kOffer, local_offer.get(), nullptr)
+          .ok());
   EXPECT_TRUE(transport_controller_
-                  ->SetLocalDescription(SdpType::kOffer, local_offer.get())
-                  .ok());
-  EXPECT_TRUE(transport_controller_
-                  ->SetRemoteDescription(SdpType::kAnswer, remote_answer.get())
+                  ->SetRemoteDescription(SdpType::kAnswer, local_offer.get(),
+                                         remote_answer.get())
                   .ok());
 
   // Verify that (kMid1Audio,kMid2Video) form a bundle group, but that
@@ -1382,12 +1424,14 @@ TEST_F(JsepTransportControllerTest, MultipleBundleGroupsIllegallyChangeGroup) {
   remote_answer->AddGroup(answer_bundle_group2);
 
   // Accept offer.
-  EXPECT_TRUE(transport_controller_
-                  ->SetLocalDescription(SdpType::kOffer, local_offer.get())
-                  .ok());
+  EXPECT_TRUE(
+      transport_controller_
+          ->SetLocalDescription(SdpType::kOffer, local_offer.get(), nullptr)
+          .ok());
   // Reject answer!
   EXPECT_FALSE(transport_controller_
-                   ->SetRemoteDescription(SdpType::kAnswer, remote_answer.get())
+                   ->SetRemoteDescription(SdpType::kAnswer, local_offer.get(),
+                                          remote_answer.get())
                    .ok());
 }
 
@@ -1445,12 +1489,14 @@ TEST_F(JsepTransportControllerTest, MultipleBundleGroupsInvalidSubsets) {
   remote_answer->AddGroup(answer_bundle_group2);
 
   // Accept offer.
-  EXPECT_TRUE(transport_controller_
-                  ->SetLocalDescription(SdpType::kOffer, local_offer.get())
-                  .ok());
+  EXPECT_TRUE(
+      transport_controller_
+          ->SetLocalDescription(SdpType::kOffer, local_offer.get(), nullptr)
+          .ok());
   // Reject answer!
   EXPECT_FALSE(transport_controller_
-                   ->SetRemoteDescription(SdpType::kAnswer, remote_answer.get())
+                   ->SetRemoteDescription(SdpType::kAnswer, local_offer.get(),
+                                          remote_answer.get())
                    .ok());
 }
 
@@ -1483,11 +1529,12 @@ TEST_F(JsepTransportControllerTest, MultipleBundleGroupsInvalidOverlap) {
   offer->AddGroup(offer_bundle_group2);
 
   // Reject offer, both if set as local or remote.
+  EXPECT_FALSE(transport_controller_
+                   ->SetLocalDescription(SdpType::kOffer, offer.get(), nullptr)
+                   .ok());
   EXPECT_FALSE(
-      transport_controller_->SetLocalDescription(SdpType::kOffer, offer.get())
-          .ok());
-  EXPECT_FALSE(
-      transport_controller_->SetRemoteDescription(SdpType::kOffer, offer.get())
+      transport_controller_
+          ->SetRemoteDescription(SdpType::kOffer, offer.get(), offer.get())
           .ok());
 }
 
@@ -1563,11 +1610,13 @@ TEST_F(JsepTransportControllerTest, MultipleBundleGroupsUnbundleFirstMid) {
   remote_answer->AddGroup(answer_bundle_group1);
   remote_answer->AddGroup(answer_bundle_group2);
 
+  EXPECT_TRUE(
+      transport_controller_
+          ->SetLocalDescription(SdpType::kOffer, local_offer.get(), nullptr)
+          .ok());
   EXPECT_TRUE(transport_controller_
-                  ->SetLocalDescription(SdpType::kOffer, local_offer.get())
-                  .ok());
-  EXPECT_TRUE(transport_controller_
-                  ->SetRemoteDescription(SdpType::kAnswer, remote_answer.get())
+                  ->SetRemoteDescription(SdpType::kAnswer, local_offer.get(),
+                                         remote_answer.get())
                   .ok());
 
   auto mid1_transport = transport_controller_->GetRtpTransport(kMid1Audio);
@@ -1659,9 +1708,10 @@ TEST_F(JsepTransportControllerTest, MultipleBundleGroupsChangeFirstMid) {
   remote_answer->AddGroup(answer_bundle_group1);
   remote_answer->AddGroup(answer_bundle_group2);
 
-  EXPECT_TRUE(transport_controller_
-                  ->SetLocalDescription(SdpType::kOffer, local_offer.get())
-                  .ok());
+  EXPECT_TRUE(
+      transport_controller_
+          ->SetLocalDescription(SdpType::kOffer, local_offer.get(), nullptr)
+          .ok());
 
   // The fact that we accept this answer is actually a bug. If we accept the
   // first MID to be in the group, we should also accept that it is the tagged
@@ -1669,7 +1719,8 @@ TEST_F(JsepTransportControllerTest, MultipleBundleGroupsChangeFirstMid) {
   // TODO(https://crbug.com/webrtc/12699): When this issue is fixed, change this
   // to EXPECT_FALSE and remove the below expectations about transports.
   EXPECT_TRUE(transport_controller_
-                  ->SetRemoteDescription(SdpType::kAnswer, remote_answer.get())
+                  ->SetRemoteDescription(SdpType::kAnswer, local_offer.get(),
+                                         remote_answer.get())
                   .ok());
   auto mid1_transport = transport_controller_->GetRtpTransport(kMid1Audio);
   auto mid2_transport = transport_controller_->GetRtpTransport(kMid2Audio);
@@ -1734,11 +1785,13 @@ TEST_F(JsepTransportControllerTest,
   remote_answer->AddGroup(bundle_group1);
   remote_answer->AddGroup(bundle_group2);
 
+  EXPECT_TRUE(
+      transport_controller_
+          ->SetLocalDescription(SdpType::kOffer, local_offer.get(), nullptr)
+          .ok());
   EXPECT_TRUE(transport_controller_
-                  ->SetLocalDescription(SdpType::kOffer, local_offer.get())
-                  .ok());
-  EXPECT_TRUE(transport_controller_
-                  ->SetRemoteDescription(SdpType::kAnswer, remote_answer.get())
+                  ->SetRemoteDescription(SdpType::kAnswer, local_offer.get(),
+                                         remote_answer.get())
                   .ok());
 
   // Add kMid3Audio and kMid6Video to the respective audio/video bundle groups.
@@ -1769,7 +1822,8 @@ TEST_F(JsepTransportControllerTest,
   subsequent_offer->AddGroup(bundle_group1);
   subsequent_offer->AddGroup(bundle_group2);
   EXPECT_TRUE(transport_controller_
-                  ->SetLocalDescription(SdpType::kOffer, subsequent_offer.get())
+                  ->SetLocalDescription(SdpType::kOffer, subsequent_offer.get(),
+                                        remote_answer.get())
                   .ok());
   auto mid1_transport = transport_controller_->GetRtpTransport(kMid1Audio);
   auto mid2_transport = transport_controller_->GetRtpTransport(kMid2Audio);
@@ -1832,11 +1886,13 @@ TEST_F(JsepTransportControllerTest,
   remote_answer->AddGroup(bundle_group1);
   remote_answer->AddGroup(bundle_group2);
 
+  EXPECT_TRUE(
+      transport_controller_
+          ->SetLocalDescription(SdpType::kOffer, local_offer.get(), nullptr)
+          .ok());
   EXPECT_TRUE(transport_controller_
-                  ->SetLocalDescription(SdpType::kOffer, local_offer.get())
-                  .ok());
-  EXPECT_TRUE(transport_controller_
-                  ->SetRemoteDescription(SdpType::kAnswer, remote_answer.get())
+                  ->SetRemoteDescription(SdpType::kAnswer, local_offer.get(),
+                                         remote_answer.get())
                   .ok());
 
   // Switch to grouping (kMid1Audio,kMid2Audio,kMid3Video,kMid4Video).
@@ -1861,10 +1917,11 @@ TEST_F(JsepTransportControllerTest,
                   cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
                   nullptr);
   subsequent_offer->AddGroup(new_bundle_group);
-  EXPECT_FALSE(
-      transport_controller_
-          ->SetLocalDescription(SdpType::kOffer, subsequent_offer.get())
-          .ok());
+  EXPECT_FALSE(transport_controller_
+                   ->SetLocalDescription(SdpType::kOffer,
+                                         subsequent_offer.get(),
+                                         remote_answer.get())
+                   .ok());
 }
 
 TEST_F(JsepTransportControllerTest,
@@ -1912,11 +1969,13 @@ TEST_F(JsepTransportControllerTest,
                   nullptr);
   remote_answer->AddGroup(bundle_group);
 
+  EXPECT_TRUE(
+      transport_controller_
+          ->SetLocalDescription(SdpType::kOffer, local_offer.get(), nullptr)
+          .ok());
   EXPECT_TRUE(transport_controller_
-                  ->SetLocalDescription(SdpType::kOffer, local_offer.get())
-                  .ok());
-  EXPECT_TRUE(transport_controller_
-                  ->SetRemoteDescription(SdpType::kAnswer, remote_answer.get())
+                  ->SetRemoteDescription(SdpType::kAnswer, local_offer.get(),
+                                         remote_answer.get())
                   .ok());
 
   // Switch to grouping (kMid1Audio,kMid2Audio) and (kMid3Video,kMid4Video).
@@ -1943,10 +2002,11 @@ TEST_F(JsepTransportControllerTest,
                   nullptr);
   subsequent_offer->AddGroup(new_bundle_group1);
   subsequent_offer->AddGroup(new_bundle_group2);
-  EXPECT_FALSE(
-      transport_controller_
-          ->SetLocalDescription(SdpType::kOffer, subsequent_offer.get())
-          .ok());
+  EXPECT_FALSE(transport_controller_
+                   ->SetLocalDescription(SdpType::kOffer,
+                                         subsequent_offer.get(),
+                                         remote_answer.get())
+                   .ok());
 }
 
 TEST_F(JsepTransportControllerTest,
@@ -1997,11 +2057,13 @@ TEST_F(JsepTransportControllerTest,
   remote_answer->AddGroup(bundle_group1);
   remote_answer->AddGroup(bundle_group2);
 
+  EXPECT_TRUE(
+      transport_controller_
+          ->SetLocalDescription(SdpType::kOffer, local_offer.get(), nullptr)
+          .ok());
   EXPECT_TRUE(transport_controller_
-                  ->SetLocalDescription(SdpType::kOffer, local_offer.get())
-                  .ok());
-  EXPECT_TRUE(transport_controller_
-                  ->SetRemoteDescription(SdpType::kAnswer, remote_answer.get())
+                  ->SetRemoteDescription(SdpType::kAnswer, local_offer.get(),
+                                         remote_answer.get())
                   .ok());
 
   // Switch to grouping (kMid1Audio,kMid3Video) and (kMid2Audio,kMid3Video).
@@ -2028,10 +2090,11 @@ TEST_F(JsepTransportControllerTest,
                   nullptr);
   subsequent_offer->AddGroup(new_bundle_group1);
   subsequent_offer->AddGroup(new_bundle_group2);
-  EXPECT_FALSE(
-      transport_controller_
-          ->SetLocalDescription(SdpType::kOffer, subsequent_offer.get())
-          .ok());
+  EXPECT_FALSE(transport_controller_
+                   ->SetLocalDescription(SdpType::kOffer,
+                                         subsequent_offer.get(),
+                                         remote_answer.get())
+                   .ok());
 }
 
 // Tests that only a subset of all the m= sections are bundled.
@@ -2065,11 +2128,13 @@ TEST_F(JsepTransportControllerTest, BundleSubsetOfMediaSections) {
 
   local_offer->AddGroup(bundle_group);
   remote_answer->AddGroup(bundle_group);
+  EXPECT_TRUE(
+      transport_controller_
+          ->SetLocalDescription(SdpType::kOffer, local_offer.get(), nullptr)
+          .ok());
   EXPECT_TRUE(transport_controller_
-                  ->SetLocalDescription(SdpType::kOffer, local_offer.get())
-                  .ok());
-  EXPECT_TRUE(transport_controller_
-                  ->SetRemoteDescription(SdpType::kAnswer, remote_answer.get())
+                  ->SetRemoteDescription(SdpType::kAnswer, local_offer.get(),
+                                         remote_answer.get())
                   .ok());
 
   // Verifiy that only `kAudio1` and `kVideo1` are bundled.
@@ -2106,11 +2171,13 @@ TEST_F(JsepTransportControllerTest, BundleOnDataSectionInSubsequentOffer) {
   local_offer->AddGroup(bundle_group);
   remote_answer->AddGroup(bundle_group);
 
+  EXPECT_TRUE(
+      transport_controller_
+          ->SetLocalDescription(SdpType::kOffer, local_offer.get(), nullptr)
+          .ok());
   EXPECT_TRUE(transport_controller_
-                  ->SetLocalDescription(SdpType::kOffer, local_offer.get())
-                  .ok());
-  EXPECT_TRUE(transport_controller_
-                  ->SetRemoteDescription(SdpType::kAnswer, remote_answer.get())
+                  ->SetRemoteDescription(SdpType::kAnswer, local_offer.get(),
+                                         remote_answer.get())
                   .ok());
   auto data_transport = transport_controller_->GetRtpTransport(kDataMid1);
 
@@ -2132,15 +2199,17 @@ TEST_F(JsepTransportControllerTest, BundleOnDataSectionInSubsequentOffer) {
   bundle_group.AddContentName(kAudioMid1);
   bundle_group.AddContentName(kVideoMid1);
   local_offer->RemoveGroupByName(cricket::GROUP_TYPE_BUNDLE);
-  remote_answer->RemoveGroupByName(cricket::GROUP_TYPE_BUNDLE);
   local_offer->AddGroup(bundle_group);
-  remote_answer->AddGroup(bundle_group);
 
   EXPECT_TRUE(transport_controller_
-                  ->SetLocalDescription(SdpType::kOffer, local_offer.get())
+                  ->SetLocalDescription(SdpType::kOffer, local_offer.get(),
+                                        remote_answer.get())
                   .ok());
+  remote_answer->RemoveGroupByName(cricket::GROUP_TYPE_BUNDLE);
+  remote_answer->AddGroup(bundle_group);
   EXPECT_TRUE(transport_controller_
-                  ->SetRemoteDescription(SdpType::kAnswer, remote_answer.get())
+                  ->SetRemoteDescription(SdpType::kAnswer, local_offer.get(),
+                                         remote_answer.get())
                   .ok());
 
   auto audio_transport = transport_controller_->GetRtpTransport(kAudioMid1);
@@ -2186,11 +2255,13 @@ TEST_F(JsepTransportControllerTest, VideoDataRejectedInAnswer) {
   local_offer->AddGroup(bundle_group);
   remote_answer->AddGroup(bundle_group);
 
+  EXPECT_TRUE(
+      transport_controller_
+          ->SetLocalDescription(SdpType::kOffer, local_offer.get(), nullptr)
+          .ok());
   EXPECT_TRUE(transport_controller_
-                  ->SetLocalDescription(SdpType::kOffer, local_offer.get())
-                  .ok());
-  EXPECT_TRUE(transport_controller_
-                  ->SetRemoteDescription(SdpType::kAnswer, remote_answer.get())
+                  ->SetRemoteDescription(SdpType::kAnswer, local_offer.get(),
+                                         remote_answer.get())
                   .ok());
 
   // Verify the RtpTransport/DtlsTransport is destroyed correctly.
@@ -2233,11 +2304,13 @@ TEST_F(JsepTransportControllerTest, ChangeBundledMidNotSupported) {
 
   local_offer->AddGroup(bundle_group);
   remote_answer->AddGroup(bundle_group);
+  EXPECT_TRUE(
+      transport_controller_
+          ->SetLocalDescription(SdpType::kOffer, local_offer.get(), nullptr)
+          .ok());
   EXPECT_TRUE(transport_controller_
-                  ->SetLocalDescription(SdpType::kOffer, local_offer.get())
-                  .ok());
-  EXPECT_TRUE(transport_controller_
-                  ->SetRemoteDescription(SdpType::kAnswer, remote_answer.get())
+                  ->SetRemoteDescription(SdpType::kAnswer, local_offer.get(),
+                                         remote_answer.get())
                   .ok());
   EXPECT_EQ(transport_controller_->GetRtpTransport(kAudioMid1),
             transport_controller_->GetRtpTransport(kVideoMid1));
@@ -2245,15 +2318,17 @@ TEST_F(JsepTransportControllerTest, ChangeBundledMidNotSupported) {
   // Reorder the bundle group.
   EXPECT_TRUE(bundle_group.RemoveContentName(kAudioMid1));
   bundle_group.AddContentName(kAudioMid1);
+  EXPECT_TRUE(transport_controller_
+                  ->SetLocalDescription(SdpType::kOffer, local_offer.get(),
+                                        remote_answer.get())
+                  .ok());
   // The answerer uses the new bundle group and now the bundle mid is changed to
   // `kVideo1`.
   remote_answer->RemoveGroupByName(cricket::GROUP_TYPE_BUNDLE);
   remote_answer->AddGroup(bundle_group);
-  EXPECT_TRUE(transport_controller_
-                  ->SetLocalDescription(SdpType::kOffer, local_offer.get())
-                  .ok());
   EXPECT_FALSE(transport_controller_
-                   ->SetRemoteDescription(SdpType::kAnswer, remote_answer.get())
+                   ->SetRemoteDescription(SdpType::kAnswer, local_offer.get(),
+                                          remote_answer.get())
                    .ok());
 }
 // Test that rejecting only the first m= section of a BUNDLE group is treated as
@@ -2294,18 +2369,21 @@ TEST_F(JsepTransportControllerTest, RejectFirstContentInBundleGroup) {
   local_offer->AddGroup(bundle_group);
   remote_answer->AddGroup(bundle_group);
 
-  EXPECT_TRUE(transport_controller_
-                  ->SetLocalDescription(SdpType::kOffer, local_offer.get())
-                  .ok());
+  EXPECT_TRUE(
+      transport_controller_
+          ->SetLocalDescription(SdpType::kOffer, local_offer.get(), nullptr)
+          .ok());
   EXPECT_FALSE(transport_controller_
-                   ->SetRemoteDescription(SdpType::kAnswer, remote_answer.get())
+                   ->SetRemoteDescription(SdpType::kAnswer, local_offer.get(),
+                                          remote_answer.get())
                    .ok());
 
   // Reject all the contents.
   remote_answer->contents()[1].rejected = true;
   remote_answer->contents()[2].rejected = true;
   EXPECT_TRUE(transport_controller_
-                  ->SetRemoteDescription(SdpType::kAnswer, remote_answer.get())
+                  ->SetRemoteDescription(SdpType::kAnswer, local_offer.get(),
+                                         remote_answer.get())
                   .ok());
   EXPECT_EQ(nullptr, transport_controller_->GetRtpTransport(kAudioMid1));
   EXPECT_EQ(nullptr, transport_controller_->GetRtpTransport(kVideoMid1));
@@ -2325,9 +2403,10 @@ TEST_F(JsepTransportControllerTest, ApplyNonRtcpMuxOfferWhenMuxingRequired) {
 
   local_offer->contents()[0].media_description()->set_rtcp_mux(false);
   // Applying a non-RTCP-mux offer is expected to fail.
-  EXPECT_FALSE(transport_controller_
-                   ->SetLocalDescription(SdpType::kOffer, local_offer.get())
-                   .ok());
+  EXPECT_FALSE(
+      transport_controller_
+          ->SetLocalDescription(SdpType::kOffer, local_offer.get(), nullptr)
+          .ok());
 }
 
 // Tests that applying non-RTCP-mux answer would fail when kRtcpMuxPolicyRequire
@@ -2340,9 +2419,10 @@ TEST_F(JsepTransportControllerTest, ApplyNonRtcpMuxAnswerWhenMuxingRequired) {
   AddAudioSection(local_offer.get(), kAudioMid1, kIceUfrag1, kIcePwd1,
                   cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
                   nullptr);
-  EXPECT_TRUE(transport_controller_
-                  ->SetLocalDescription(SdpType::kOffer, local_offer.get())
-                  .ok());
+  EXPECT_TRUE(
+      transport_controller_
+          ->SetLocalDescription(SdpType::kOffer, local_offer.get(), nullptr)
+          .ok());
 
   auto remote_answer = std::make_unique<cricket::SessionDescription>();
   AddAudioSection(remote_answer.get(), kAudioMid1, kIceUfrag1, kIcePwd1,
@@ -2351,7 +2431,8 @@ TEST_F(JsepTransportControllerTest, ApplyNonRtcpMuxAnswerWhenMuxingRequired) {
   // Applying a non-RTCP-mux answer is expected to fail.
   remote_answer->contents()[0].media_description()->set_rtcp_mux(false);
   EXPECT_FALSE(transport_controller_
-                   ->SetRemoteDescription(SdpType::kAnswer, remote_answer.get())
+                   ->SetRemoteDescription(SdpType::kAnswer, local_offer.get(),
+                                          remote_answer.get())
                    .ok());
 }
 
@@ -2371,11 +2452,13 @@ TEST_F(JsepTransportControllerTest,
   answer_bundle_group.AddContentName(kAudioMid1);
   answer_bundle_group.AddContentName(kVideoMid1);
   remote_answer->AddGroup(answer_bundle_group);
-  EXPECT_TRUE(transport_controller_
-                  ->SetLocalDescription(SdpType::kOffer, local_offer.get())
-                  .ok());
+  EXPECT_TRUE(
+      transport_controller_
+          ->SetLocalDescription(SdpType::kOffer, local_offer.get(), nullptr)
+          .ok());
   EXPECT_FALSE(transport_controller_
-                   ->SetRemoteDescription(SdpType::kAnswer, remote_answer.get())
+                   ->SetRemoteDescription(SdpType::kAnswer, local_offer.get(),
+                                          remote_answer.get())
                    .ok());
 }
 
@@ -2392,11 +2475,13 @@ TEST_F(JsepTransportControllerTest, RejectBundleGroupWithNonExistingMid) {
   local_offer->AddGroup(invalid_bundle_group);
   remote_answer->AddGroup(invalid_bundle_group);
 
+  EXPECT_FALSE(
+      transport_controller_
+          ->SetLocalDescription(SdpType::kOffer, local_offer.get(), nullptr)
+          .ok());
   EXPECT_FALSE(transport_controller_
-                   ->SetLocalDescription(SdpType::kOffer, local_offer.get())
-                   .ok());
-  EXPECT_FALSE(transport_controller_
-                   ->SetRemoteDescription(SdpType::kAnswer, remote_answer.get())
+                   ->SetRemoteDescription(SdpType::kAnswer, local_offer.get(),
+                                          remote_answer.get())
                    .ok());
 }
 
@@ -2407,16 +2492,19 @@ TEST_F(JsepTransportControllerTest, RemoveContentFromBundleGroup) {
 
   auto local_offer = CreateSessionDescriptionWithBundleGroup();
   auto remote_answer = CreateSessionDescriptionWithBundleGroup();
+  EXPECT_TRUE(
+      transport_controller_
+          ->SetLocalDescription(SdpType::kOffer, local_offer.get(), nullptr)
+          .ok());
   EXPECT_TRUE(transport_controller_
-                  ->SetLocalDescription(SdpType::kOffer, local_offer.get())
-                  .ok());
-  EXPECT_TRUE(transport_controller_
-                  ->SetRemoteDescription(SdpType::kAnswer, remote_answer.get())
+                  ->SetRemoteDescription(SdpType::kAnswer, local_offer.get(),
+                                         remote_answer.get())
                   .ok());
 
   // Do an re-offer/answer.
   EXPECT_TRUE(transport_controller_
-                  ->SetLocalDescription(SdpType::kOffer, local_offer.get())
+                  ->SetLocalDescription(SdpType::kOffer, local_offer.get(),
+                                        remote_answer.get())
                   .ok());
   auto new_answer = CreateSessionDescriptionWithoutBundle();
   cricket::ContentGroup new_bundle_group(cricket::GROUP_TYPE_BUNDLE);
@@ -2427,7 +2515,8 @@ TEST_F(JsepTransportControllerTest, RemoveContentFromBundleGroup) {
 
   // Applying invalid answer is expected to fail.
   EXPECT_FALSE(transport_controller_
-                   ->SetRemoteDescription(SdpType::kAnswer, new_answer.get())
+                   ->SetRemoteDescription(SdpType::kAnswer, local_offer.get(),
+                                          new_answer.get())
                    .ok());
 
   // Rejected the video content.
@@ -2435,7 +2524,8 @@ TEST_F(JsepTransportControllerTest, RemoveContentFromBundleGroup) {
   ASSERT_TRUE(video_content);
   video_content->rejected = true;
   EXPECT_TRUE(transport_controller_
-                  ->SetRemoteDescription(SdpType::kAnswer, new_answer.get())
+                  ->SetRemoteDescription(SdpType::kAnswer, local_offer.get(),
+                                         new_answer.get())
                   .ok());
 }
 
@@ -2453,14 +2543,16 @@ TEST_F(JsepTransportControllerTest, ChangeTaggedMediaSectionMaxBundle) {
   cricket::ContentGroup bundle_group(cricket::GROUP_TYPE_BUNDLE);
   bundle_group.AddContentName(kAudioMid1);
   local_offer->AddGroup(bundle_group);
-  EXPECT_TRUE(transport_controller_
-                  ->SetLocalDescription(SdpType::kOffer, local_offer.get())
-                  .ok());
+  EXPECT_TRUE(
+      transport_controller_
+          ->SetLocalDescription(SdpType::kOffer, local_offer.get(), nullptr)
+          .ok());
 
   std::unique_ptr<cricket::SessionDescription> remote_answer(
       local_offer->Clone());
   EXPECT_TRUE(transport_controller_
-                  ->SetRemoteDescription(SdpType::kAnswer, remote_answer.get())
+                  ->SetRemoteDescription(SdpType::kAnswer, local_offer.get(),
+                                         remote_answer.get())
                   .ok());
 
   std::unique_ptr<cricket::SessionDescription> local_reoffer(
@@ -2475,14 +2567,15 @@ TEST_F(JsepTransportControllerTest, ChangeTaggedMediaSectionMaxBundle) {
   local_reoffer->AddGroup(new_bundle_group);
 
   EXPECT_TRUE(transport_controller_
-                  ->SetLocalDescription(SdpType::kOffer, local_reoffer.get())
+                  ->SetLocalDescription(SdpType::kOffer, local_reoffer.get(),
+                                        remote_answer.get())
                   .ok());
   std::unique_ptr<cricket::SessionDescription> remote_reanswer(
       local_reoffer->Clone());
-  EXPECT_TRUE(
-      transport_controller_
-          ->SetRemoteDescription(SdpType::kAnswer, remote_reanswer.get())
-          .ok());
+  EXPECT_TRUE(transport_controller_
+                  ->SetRemoteDescription(SdpType::kAnswer, local_reoffer.get(),
+                                         remote_reanswer.get())
+                  .ok());
 }
 
 TEST_F(JsepTransportControllerTest, RollbackRestoresRejectedTransport) {
@@ -2496,11 +2589,13 @@ TEST_F(JsepTransportControllerTest, RollbackRestoresRejectedTransport) {
                   nullptr);
   std::unique_ptr<cricket::SessionDescription> remote_answer(
       local_offer->Clone());
+  EXPECT_TRUE(
+      transport_controller_
+          ->SetLocalDescription(SdpType::kOffer, local_offer.get(), nullptr)
+          .ok());
   EXPECT_TRUE(transport_controller_
-                  ->SetLocalDescription(SdpType::kOffer, local_offer.get())
-                  .ok());
-  EXPECT_TRUE(transport_controller_
-                  ->SetRemoteDescription(SdpType::kAnswer, remote_answer.get())
+                  ->SetRemoteDescription(SdpType::kAnswer, local_offer.get(),
+                                         remote_answer.get())
                   .ok());
 
   auto mid1_transport = transport_controller_->GetRtpTransport(kMid1Audio);
@@ -2514,7 +2609,8 @@ TEST_F(JsepTransportControllerTest, RollbackRestoresRejectedTransport) {
   local_reoffer->contents()[0].rejected = true;
 
   EXPECT_TRUE(transport_controller_
-                  ->SetLocalDescription(SdpType::kOffer, local_reoffer.get())
+                  ->SetLocalDescription(SdpType::kOffer, local_reoffer.get(),
+                                        remote_answer.get())
                   .ok());
   auto old_mid1_transport = mid1_transport;
   mid1_transport = transport_controller_->GetRtpTransport(kMid1Audio);
@@ -2556,11 +2652,13 @@ TEST_F(JsepTransportControllerTest, RollbackRestoresPreviousTransportMapping) {
   std::unique_ptr<cricket::SessionDescription> remote_answer(
       local_offer->Clone());
 
+  EXPECT_TRUE(
+      transport_controller_
+          ->SetLocalDescription(SdpType::kOffer, local_offer.get(), nullptr)
+          .ok());
   EXPECT_TRUE(transport_controller_
-                  ->SetLocalDescription(SdpType::kOffer, local_offer.get())
-                  .ok());
-  EXPECT_TRUE(transport_controller_
-                  ->SetRemoteDescription(SdpType::kAnswer, remote_answer.get())
+                  ->SetRemoteDescription(SdpType::kAnswer, local_offer.get(),
+                                         remote_answer.get())
                   .ok());
 
   auto mid1_transport = transport_controller_->GetRtpTransport(kMid1Audio);
@@ -2585,7 +2683,8 @@ TEST_F(JsepTransportControllerTest, RollbackRestoresPreviousTransportMapping) {
   local_reoffer->AddGroup(bundle_group);
 
   EXPECT_TRUE(transport_controller_
-                  ->SetLocalDescription(SdpType::kOffer, local_reoffer.get())
+                  ->SetLocalDescription(SdpType::kOffer, local_reoffer.get(),
+                                        remote_answer.get())
                   .ok());
 
   // Store the old transport pointer and verify that the offer actually changed
@@ -2633,11 +2732,13 @@ TEST_F(JsepTransportControllerTest, RollbackAndAddToDifferentBundleGroup) {
   std::unique_ptr<cricket::SessionDescription> remote_answer(
       local_offer->Clone());
 
+  EXPECT_TRUE(
+      transport_controller_
+          ->SetLocalDescription(SdpType::kOffer, local_offer.get(), nullptr)
+          .ok());
   EXPECT_TRUE(transport_controller_
-                  ->SetLocalDescription(SdpType::kOffer, local_offer.get())
-                  .ok());
-  EXPECT_TRUE(transport_controller_
-                  ->SetRemoteDescription(SdpType::kAnswer, remote_answer.get())
+                  ->SetRemoteDescription(SdpType::kAnswer, local_offer.get(),
+                                         remote_answer.get())
                   .ok());
 
   // Apply an offer that adds kMid3Audio to the first BUNDLE group.,
@@ -2657,10 +2758,11 @@ TEST_F(JsepTransportControllerTest, RollbackAndAddToDifferentBundleGroup) {
   subsequent_offer_1->AddGroup(modified_bundle_group1);
   subsequent_offer_1->AddGroup(bundle_group2);
 
-  EXPECT_TRUE(
-      transport_controller_
-          ->SetLocalDescription(SdpType::kOffer, subsequent_offer_1.get())
-          .ok());
+  EXPECT_TRUE(transport_controller_
+                  ->SetLocalDescription(SdpType::kOffer,
+                                        subsequent_offer_1.get(),
+                                        remote_answer.get())
+                  .ok());
 
   auto mid1_transport = transport_controller_->GetRtpTransport(kMid1Audio);
   auto mid2_transport = transport_controller_->GetRtpTransport(kMid2Audio);
@@ -2689,10 +2791,11 @@ TEST_F(JsepTransportControllerTest, RollbackAndAddToDifferentBundleGroup) {
   subsequent_offer_2->AddGroup(bundle_group1);
   subsequent_offer_2->AddGroup(modified_bundle_group2);
 
-  EXPECT_TRUE(
-      transport_controller_
-          ->SetLocalDescription(SdpType::kOffer, subsequent_offer_2.get())
-          .ok());
+  EXPECT_TRUE(transport_controller_
+                  ->SetLocalDescription(SdpType::kOffer,
+                                        subsequent_offer_2.get(),
+                                        remote_answer.get())
+                  .ok());
 
   mid1_transport = transport_controller_->GetRtpTransport(kMid1Audio);
   mid2_transport = transport_controller_->GetRtpTransport(kMid2Audio);
@@ -2722,9 +2825,9 @@ TEST_F(JsepTransportControllerTest, BundleOnlySectionDoesNotNeedRtcpMux) {
   offer->contents()[1].media_description()->set_rtcp_mux(false);
   offer->contents()[1].bundle_only = true;
 
-  EXPECT_TRUE(
-      transport_controller_->SetRemoteDescription(SdpType::kOffer, offer.get())
-          .ok());
+  EXPECT_TRUE(transport_controller_
+                  ->SetRemoteDescription(SdpType::kOffer, nullptr, offer.get())
+                  .ok());
 }
 
 // Test that with max-bundle a single unbundled m-line is accepted.
@@ -2738,9 +2841,9 @@ TEST_F(JsepTransportControllerTest,
   AddAudioSection(offer.get(), kAudioMid1, kIceUfrag1, kIcePwd1,
                   cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
                   nullptr);
-  EXPECT_TRUE(
-      transport_controller_->SetRemoteDescription(SdpType::kOffer, offer.get())
-          .ok());
+  EXPECT_TRUE(transport_controller_
+                  ->SetRemoteDescription(SdpType::kOffer, nullptr, offer.get())
+                  .ok());
 }
 
 }  // namespace webrtc
diff --git a/third_party/libwebrtc/pc/legacy_stats_collector.cc b/third_party/libwebrtc/pc/legacy_stats_collector.cc
index 3bc65ee3ee..98b7cb9677 100644
--- a/third_party/libwebrtc/pc/legacy_stats_collector.cc
+++ b/third_party/libwebrtc/pc/legacy_stats_collector.cc
@@ -355,9 +355,8 @@ void ExtractStats(const cricket::VideoReceiverInfo& info,
   report->AddInt64(StatsReport::kStatsValueNameInterframeDelayMaxMs,
                    info.interframe_delay_max_ms);
 
-  report->AddString(
-      StatsReport::kStatsValueNameContentType,
-      webrtc::videocontenttypehelpers::ToString(info.content_type));
+  report->AddString(StatsReport::kStatsValueNameContentType,
+                    videocontenttypehelpers::ToString(info.content_type));
 }
 
 void ExtractStats(const cricket::VideoSenderInfo& info,
@@ -398,9 +397,8 @@ void ExtractStats(const cricket::VideoSenderInfo& info,
   for (const auto& i : ints)
     report->AddInt(i.name, i.value);
   report->AddString(StatsReport::kStatsValueNameMediaType, "video");
-  report->AddString(
-      StatsReport::kStatsValueNameContentType,
-      webrtc::videocontenttypehelpers::ToString(info.content_type));
+  report->AddString(StatsReport::kStatsValueNameContentType,
+                    videocontenttypehelpers::ToString(info.content_type));
 }
 
 void ExtractStats(const cricket::BandwidthEstimationInfo& info,
@@ -1033,7 +1031,7 @@ void LegacyStatsCollector::ExtractBweInfo() {
   if (pc_->signaling_state() == PeerConnectionInterface::kClosed)
     return;
 
-  webrtc::Call::Stats call_stats = pc_->GetCallStats();
+  Call::Stats call_stats = pc_->GetCallStats();
   cricket::BandwidthEstimationInfo bwe_info;
   bwe_info.available_send_bandwidth = call_stats.send_bandwidth_bps;
   bwe_info.available_recv_bandwidth = call_stats.recv_bandwidth_bps;
diff --git a/third_party/libwebrtc/pc/legacy_stats_collector.h b/third_party/libwebrtc/pc/legacy_stats_collector.h
index e905b39d48..1c7aad0636 100644
--- a/third_party/libwebrtc/pc/legacy_stats_collector.h
+++ b/third_party/libwebrtc/pc/legacy_stats_collector.h
@@ -177,9 +177,9 @@ class LegacyStatsCollector : public LegacyStatsCollectorInterface {
   void ExtractMediaInfo(
       const std::map<std::string, std::string>& transport_names_by_mid);
   void ExtractSenderInfo();
-  webrtc::StatsReport* GetReport(const StatsReport::StatsType& type,
-                                 const std::string& id,
-                                 StatsReport::Direction direction);
+  StatsReport* GetReport(const StatsReport::StatsType& type,
+                         const std::string& id,
+                         StatsReport::Direction direction);
 
   // Helper method to get stats from the local audio tracks.
   void UpdateStatsFromExistingLocalAudioTracks(bool has_remote_tracks);
diff --git a/third_party/libwebrtc/pc/media_factory.h b/third_party/libwebrtc/pc/media_factory.h
new file mode 100644
index 0000000000..c86784632e
--- /dev/null
+++ b/third_party/libwebrtc/pc/media_factory.h
@@ -0,0 +1,45 @@
+/*
+ *  Copyright 2023 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 PC_MEDIA_FACTORY_H_
+#define PC_MEDIA_FACTORY_H_
+
+#include <memory>
+
+#include "api/environment/environment.h"
+#include "call/call.h"
+#include "call/call_config.h"
+#include "media/base/media_engine.h"
+
+namespace webrtc {
+
+// PeerConnectionFactoryDependencies is forward declared because of circular
+// dependency between MediaFactory and PeerConnectionFactoryDependencies:
+// PeerConnectionFactoryDependencies keeps an instance of MediaFactory and thus
+// needs to know how to destroy it.
+// MediaFactory mentions PeerConnectionFactoryDependencies in api, but does not
+// need its full definition.
+struct PeerConnectionFactoryDependencies;
+
+// Interface repsponsible for constructing media specific classes for
+// PeerConnectionFactory and PeerConnection.
+class MediaFactory {
+ public:
+  virtual ~MediaFactory() = default;
+
+  virtual std::unique_ptr<Call> CreateCall(const CallConfig& config) = 0;
+  virtual std::unique_ptr<cricket::MediaEngineInterface> CreateMediaEngine(
+      const Environment& env,
+      PeerConnectionFactoryDependencies& dependencies) = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // PC_MEDIA_FACTORY_H_
diff --git a/third_party/libwebrtc/pc/media_session.cc b/third_party/libwebrtc/pc/media_session.cc
index a763919c16..573e35225e 100644
--- a/third_party/libwebrtc/pc/media_session.cc
+++ b/third_party/libwebrtc/pc/media_session.cc
@@ -117,52 +117,38 @@ namespace cricket {
 
 namespace {
 
-bool IsRtxCodec(const Codec& codec) {
-  return absl::EqualsIgnoreCase(codec.name, kRtxCodecName);
-}
-
 bool IsRtxCodec(const webrtc::RtpCodecCapability& capability) {
   return absl::EqualsIgnoreCase(capability.name, kRtxCodecName);
 }
 
 bool ContainsRtxCodec(const std::vector<Codec>& codecs) {
-  for (const auto& codec : codecs) {
-    if (IsRtxCodec(codec)) {
-      return true;
-    }
-  }
-  return false;
-}
-
-bool IsRedCodec(const Codec& codec) {
-  return absl::EqualsIgnoreCase(codec.name, kRedCodecName);
+  return absl::c_find_if(codecs, [](const Codec& c) {
+           return c.GetResiliencyType() == Codec::ResiliencyType::kRtx;
+         }) != codecs.end();
 }
 
 bool IsRedCodec(const webrtc::RtpCodecCapability& capability) {
   return absl::EqualsIgnoreCase(capability.name, kRedCodecName);
 }
 
-bool IsFlexfecCodec(const Codec& codec) {
-  return absl::EqualsIgnoreCase(codec.name, kFlexfecCodecName);
-}
-
 bool ContainsFlexfecCodec(const std::vector<Codec>& codecs) {
-  for (const auto& codec : codecs) {
-    if (IsFlexfecCodec(codec)) {
-      return true;
-    }
-  }
-  return false;
-}
-
-bool IsUlpfecCodec(const Codec& codec) {
-  return absl::EqualsIgnoreCase(codec.name, kUlpfecCodecName);
+  return absl::c_find_if(codecs, [](const Codec& c) {
+           return c.GetResiliencyType() == Codec::ResiliencyType::kFlexfec;
+         }) != codecs.end();
 }
 
 bool IsComfortNoiseCodec(const Codec& codec) {
   return absl::EqualsIgnoreCase(codec.name, kComfortNoiseCodecName);
 }
 
+void StripCNCodecs(AudioCodecs* audio_codecs) {
+  audio_codecs->erase(std::remove_if(audio_codecs->begin(), audio_codecs->end(),
+                                     [](const AudioCodec& codec) {
+                                       return IsComfortNoiseCodec(codec);
+                                     }),
+                      audio_codecs->end());
+}
+
 RtpTransceiverDirection NegotiateRtpTransceiverDirection(
     RtpTransceiverDirection offer,
     RtpTransceiverDirection wants) {
@@ -727,12 +713,10 @@ RTCError CreateMediaContentOffer(
 bool ReferencedCodecsMatch(const std::vector<Codec>& codecs1,
                            const int codec1_id,
                            const std::vector<Codec>& codecs2,
-                           const int codec2_id,
-                           const webrtc::FieldTrialsView* field_trials) {
+                           const int codec2_id) {
   const Codec* codec1 = FindCodecById(codecs1, codec1_id);
   const Codec* codec2 = FindCodecById(codecs2, codec2_id);
-  return codec1 != nullptr && codec2 != nullptr &&
-         codec1->Matches(*codec2, field_trials);
+  return codec1 != nullptr && codec2 != nullptr && codec1->Matches(*codec2);
 }
 
 void NegotiatePacketization(const Codec& local_codec,
@@ -747,11 +731,9 @@ void NegotiatePacketization(const Codec& local_codec,
 // Finds a codec in `codecs2` that matches `codec_to_match`, which is
 // a member of `codecs1`. If `codec_to_match` is an RED or RTX codec, both
 // the codecs themselves and their associated codecs must match.
-absl::optional<Codec> FindMatchingCodec(
-    const std::vector<Codec>& codecs1,
-    const std::vector<Codec>& codecs2,
-    const Codec& codec_to_match,
-    const webrtc::FieldTrialsView* field_trials) {
+absl::optional<Codec> FindMatchingCodec(const std::vector<Codec>& codecs1,
+                                        const std::vector<Codec>& codecs2,
+                                        const Codec& codec_to_match) {
   // `codec_to_match` should be a member of `codecs1`, in order to look up
   // RED/RTX codecs' associated codecs correctly. If not, that's a programming
   // error.
@@ -759,8 +741,8 @@ absl::optional<Codec> FindMatchingCodec(
     return &codec == &codec_to_match;
   }));
   for (const Codec& potential_match : codecs2) {
-    if (potential_match.Matches(codec_to_match, field_trials)) {
-      if (IsRtxCodec(codec_to_match)) {
+    if (potential_match.Matches(codec_to_match)) {
+      if (codec_to_match.GetResiliencyType() == Codec::ResiliencyType::kRtx) {
         int apt_value_1 = 0;
         int apt_value_2 = 0;
         if (!codec_to_match.GetParam(kCodecParamAssociatedPayloadType,
@@ -770,11 +752,12 @@ absl::optional<Codec> FindMatchingCodec(
           RTC_LOG(LS_WARNING) << "RTX missing associated payload type.";
           continue;
         }
-        if (!ReferencedCodecsMatch(codecs1, apt_value_1, codecs2, apt_value_2,
-                                   field_trials)) {
+        if (!ReferencedCodecsMatch(codecs1, apt_value_1, codecs2,
+                                   apt_value_2)) {
           continue;
         }
-      } else if (IsRedCodec(codec_to_match)) {
+      } else if (codec_to_match.GetResiliencyType() ==
+                 Codec::ResiliencyType::kRed) {
         auto red_parameters_1 =
             codec_to_match.params.find(kCodecParamNotInNameValueFormat);
         auto red_parameters_2 =
@@ -814,7 +797,7 @@ absl::optional<Codec> FindMatchingCodec(
             if (rtc::FromString(redundant_payloads_1[0], &red_value_1) &&
                 rtc::FromString(redundant_payloads_2[0], &red_value_2)) {
               if (!ReferencedCodecsMatch(codecs1, red_value_1, codecs2,
-                                         red_value_2, field_trials)) {
+                                         red_value_2)) {
                 continue;
               }
             }
@@ -832,18 +815,17 @@ absl::optional<Codec> FindMatchingCodec(
 void NegotiateCodecs(const std::vector<Codec>& local_codecs,
                      const std::vector<Codec>& offered_codecs,
                      std::vector<Codec>* negotiated_codecs,
-                     bool keep_offer_order,
-                     const webrtc::FieldTrialsView* field_trials) {
+                     bool keep_offer_order) {
   for (const Codec& ours : local_codecs) {
     absl::optional<Codec> theirs =
-        FindMatchingCodec(local_codecs, offered_codecs, ours, field_trials);
+        FindMatchingCodec(local_codecs, offered_codecs, ours);
     // Note that we intentionally only find one matching codec for each of our
     // local codecs, in case the remote offer contains duplicate codecs.
     if (theirs) {
       Codec negotiated = ours;
       NegotiatePacketization(ours, *theirs, &negotiated);
       negotiated.IntersectFeedbackParams(*theirs);
-      if (IsRtxCodec(negotiated)) {
+      if (negotiated.GetResiliencyType() == Codec::ResiliencyType::kRtx) {
         const auto apt_it =
             theirs->params.find(kCodecParamAssociatedPayloadType);
         // FindMatchingCodec shouldn't return something with no apt value.
@@ -855,7 +837,8 @@ void NegotiateCodecs(const std::vector<Codec>& local_codecs,
         if (rtx_time_it != theirs->params.end()) {
           negotiated.SetParam(kCodecParamRtxTime, rtx_time_it->second);
         }
-      } else if (IsRedCodec(negotiated)) {
+      } else if (negotiated.GetResiliencyType() ==
+                 Codec::ResiliencyType::kRed) {
         const auto red_it =
             theirs->params.find(kCodecParamNotInNameValueFormat);
         if (red_it != theirs->params.end()) {
@@ -960,15 +943,15 @@ const Codec* GetAssociatedCodecForRed(const std::vector<Codec>& codec_list,
 // collide.
 void MergeCodecs(const std::vector<Codec>& reference_codecs,
                  std::vector<Codec>* offered_codecs,
-                 UsedPayloadTypes* used_pltypes,
-                 const webrtc::FieldTrialsView* field_trials) {
+                 UsedPayloadTypes* used_pltypes) {
   // Add all new codecs that are not RTX/RED codecs.
   // The two-pass splitting of the loops means preferring payload types
   // of actual codecs with respect to collisions.
   for (const Codec& reference_codec : reference_codecs) {
-    if (!IsRtxCodec(reference_codec) && !IsRedCodec(reference_codec) &&
-        !FindMatchingCodec(reference_codecs, *offered_codecs, reference_codec,
-                           field_trials)) {
+    if (reference_codec.GetResiliencyType() != Codec::ResiliencyType::kRtx &&
+        reference_codec.GetResiliencyType() != Codec::ResiliencyType::kRed &&
+        !FindMatchingCodec(reference_codecs, *offered_codecs,
+                           reference_codec)) {
       Codec codec = reference_codec;
       used_pltypes->FindAndSetIdUsed(&codec);
       offered_codecs->push_back(codec);
@@ -977,9 +960,9 @@ void MergeCodecs(const std::vector<Codec>& reference_codecs,
 
   // Add all new RTX or RED codecs.
   for (const Codec& reference_codec : reference_codecs) {
-    if (IsRtxCodec(reference_codec) &&
-        !FindMatchingCodec(reference_codecs, *offered_codecs, reference_codec,
-                           field_trials)) {
+    if (reference_codec.GetResiliencyType() == Codec::ResiliencyType::kRtx &&
+        !FindMatchingCodec(reference_codecs, *offered_codecs,
+                           reference_codec)) {
       Codec rtx_codec = reference_codec;
       const Codec* associated_codec =
           GetAssociatedCodecForRtx(reference_codecs, rtx_codec);
@@ -989,7 +972,7 @@ void MergeCodecs(const std::vector<Codec>& reference_codecs,
       // Find a codec in the offered list that matches the reference codec.
       // Its payload type may be different than the reference codec.
       absl::optional<Codec> matching_codec = FindMatchingCodec(
-          reference_codecs, *offered_codecs, *associated_codec, field_trials);
+          reference_codecs, *offered_codecs, *associated_codec);
       if (!matching_codec) {
         RTC_LOG(LS_WARNING)
             << "Couldn't find matching " << associated_codec->name << " codec.";
@@ -1000,15 +983,16 @@ void MergeCodecs(const std::vector<Codec>& reference_codecs,
           rtc::ToString(matching_codec->id);
       used_pltypes->FindAndSetIdUsed(&rtx_codec);
       offered_codecs->push_back(rtx_codec);
-    } else if (IsRedCodec(reference_codec) &&
+    } else if (reference_codec.GetResiliencyType() ==
+                   Codec::ResiliencyType::kRed &&
                !FindMatchingCodec(reference_codecs, *offered_codecs,
-                                  reference_codec, field_trials)) {
+                                  reference_codec)) {
       Codec red_codec = reference_codec;
       const Codec* associated_codec =
           GetAssociatedCodecForRed(reference_codecs, red_codec);
       if (associated_codec) {
         absl::optional<Codec> matching_codec = FindMatchingCodec(
-            reference_codecs, *offered_codecs, *associated_codec, field_trials);
+            reference_codecs, *offered_codecs, *associated_codec);
         if (!matching_codec) {
           RTC_LOG(LS_WARNING) << "Couldn't find matching "
                               << associated_codec->name << " codec.";
@@ -1032,8 +1016,7 @@ void MergeCodecs(const std::vector<Codec>& reference_codecs,
 std::vector<Codec> MatchCodecPreference(
     const std::vector<webrtc::RtpCodecCapability>& codec_preferences,
     const std::vector<Codec>& codecs,
-    const std::vector<Codec>& supported_codecs,
-    const webrtc::FieldTrialsView* field_trials) {
+    const std::vector<Codec>& supported_codecs) {
   std::vector<Codec> filtered_codecs;
   bool want_rtx = false;
   bool want_red = false;
@@ -1060,12 +1043,13 @@ std::vector<Codec> MatchCodecPreference(
         });
 
     if (found_codec != supported_codecs.end()) {
-      absl::optional<Codec> found_codec_with_correct_pt = FindMatchingCodec(
-          supported_codecs, codecs, *found_codec, field_trials);
+      absl::optional<Codec> found_codec_with_correct_pt =
+          FindMatchingCodec(supported_codecs, codecs, *found_codec);
       if (found_codec_with_correct_pt) {
         // RED may already have been added if its primary codec is before RED
         // in the codec list.
-        bool is_red_codec = IsRedCodec(*found_codec_with_correct_pt);
+        bool is_red_codec = found_codec_with_correct_pt->GetResiliencyType() ==
+                            Codec::ResiliencyType::kRed;
         if (!is_red_codec || !red_was_added) {
           filtered_codecs.push_back(*found_codec_with_correct_pt);
           red_was_added = is_red_codec ? true : red_was_added;
@@ -1074,14 +1058,15 @@ std::vector<Codec> MatchCodecPreference(
         // Search for the matching rtx or red codec.
         if (want_red || want_rtx) {
           for (const auto& codec : codecs) {
-            if (IsRtxCodec(codec)) {
+            if (codec.GetResiliencyType() == Codec::ResiliencyType::kRtx) {
               const auto apt =
                   codec.params.find(cricket::kCodecParamAssociatedPayloadType);
               if (apt != codec.params.end() && apt->second == id) {
                 filtered_codecs.push_back(codec);
                 break;
               }
-            } else if (IsRedCodec(codec)) {
+            } else if (codec.GetResiliencyType() ==
+                       Codec::ResiliencyType::kRed) {
               // For RED, do not insert the codec again if it was already
               // inserted. audio/red for opus gets enabled by having RED before
               // the primary codec.
@@ -1110,10 +1095,8 @@ std::vector<Codec> MatchCodecPreference(
 }
 
 // Compute the union of `codecs1` and `codecs2`.
-std::vector<Codec> ComputeCodecsUnion(
-    const std::vector<Codec>& codecs1,
-    const std::vector<Codec>& codecs2,
-    const webrtc::FieldTrialsView* field_trials) {
+std::vector<Codec> ComputeCodecsUnion(const std::vector<Codec>& codecs1,
+                                      const std::vector<Codec>& codecs2) {
   std::vector<Codec> all_codecs;
   UsedPayloadTypes used_payload_types;
   for (const Codec& codec : codecs1) {
@@ -1124,7 +1107,7 @@ std::vector<Codec> ComputeCodecsUnion(
 
   // Use MergeCodecs to merge the second half of our list as it already checks
   // and fixes problems with duplicate payload types.
-  MergeCodecs(codecs2, &all_codecs, &used_payload_types, field_trials);
+  MergeCodecs(codecs2, &all_codecs, &used_payload_types);
 
   return all_codecs;
 }
@@ -1333,14 +1316,6 @@ void NegotiateRtpHeaderExtensions(const RtpHeaderExtensions& local_extensions,
   }
 }
 
-void StripCNCodecs(AudioCodecs* audio_codecs) {
-  audio_codecs->erase(std::remove_if(audio_codecs->begin(), audio_codecs->end(),
-                                     [](const AudioCodec& codec) {
-                                       return IsComfortNoiseCodec(codec);
-                                     }),
-                      audio_codecs->end());
-}
-
 bool SetCodecsInAnswer(const MediaContentDescription* offer,
                        const std::vector<Codec>& local_codecs,
                        const MediaDescriptionOptions& media_description_options,
@@ -1353,8 +1328,7 @@ bool SetCodecsInAnswer(const MediaContentDescription* offer,
              offer->type() == MEDIA_TYPE_VIDEO);
   std::vector<Codec> negotiated_codecs;
   NegotiateCodecs(local_codecs, offer->codecs(), &negotiated_codecs,
-                  media_description_options.codec_preferences.empty(),
-                  &field_trials);
+                  media_description_options.codec_preferences.empty());
   answer->AddCodecs(negotiated_codecs);
   answer->set_protocol(offer->protocol());
   if (!AddStreamParams(media_description_options.sender_options,
@@ -1512,6 +1486,145 @@ bool IsDtlsActive(const ContentInfo* content,
       .description.secure();
 }
 
+webrtc::RTCErrorOr<AudioCodecs> GetNegotiatedCodecsForOffer(
+    const MediaDescriptionOptions& media_description_options,
+    const MediaSessionOptions& session_options,
+    const ContentInfo* current_content,
+    const std::vector<Codec>& codecs,
+    const std::vector<Codec>& supported_codecs) {
+  std::vector<Codec> filtered_codecs;
+  if (!media_description_options.codec_preferences.empty()) {
+    // Add the codecs from the current transceiver's codec preferences.
+    // They override any existing codecs from previous negotiations.
+    filtered_codecs = MatchCodecPreference(
+        media_description_options.codec_preferences, codecs, supported_codecs);
+  } else {
+    // Add the codecs from current content if it exists and is not rejected nor
+    // recycled.
+    if (current_content && !current_content->rejected &&
+        current_content->name == media_description_options.mid) {
+      if (!IsMediaContentOfType(current_content,
+                                media_description_options.type)) {
+        // Can happen if the remote side re-uses a MID while recycling.
+        LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
+                             "Media type for content with mid='" +
+                                 current_content->name +
+                                 "' does not match previous type.");
+      }
+      const MediaContentDescription* mcd = current_content->media_description();
+      for (const Codec& codec : mcd->codecs()) {
+        if (FindMatchingCodec(mcd->codecs(), codecs, codec)) {
+          filtered_codecs.push_back(codec);
+        }
+      }
+    }
+    // Add other supported codecs.
+    for (const Codec& codec : supported_codecs) {
+      absl::optional<Codec> found_codec =
+          FindMatchingCodec(supported_codecs, codecs, codec);
+      if (found_codec &&
+          !FindMatchingCodec(supported_codecs, filtered_codecs, codec)) {
+        // Use the `found_codec` from `codecs` because it has the
+        // correctly mapped payload type.
+        // This is only done for video since we do not yet have rtx for audio.
+        if (media_description_options.type == MEDIA_TYPE_VIDEO &&
+            found_codec->GetResiliencyType() == Codec::ResiliencyType::kRtx) {
+          // For RTX we might need to adjust the apt parameter if we got a
+          // remote offer without RTX for a codec for which we support RTX.
+          auto referenced_codec =
+              GetAssociatedCodecForRtx(supported_codecs, codec);
+          RTC_DCHECK(referenced_codec);
+
+          // Find the codec we should be referencing and point to it.
+          absl::optional<Codec> changed_referenced_codec = FindMatchingCodec(
+              supported_codecs, filtered_codecs, *referenced_codec);
+          if (changed_referenced_codec) {
+            found_codec->SetParam(kCodecParamAssociatedPayloadType,
+                                  changed_referenced_codec->id);
+          }
+        }
+        filtered_codecs.push_back(*found_codec);
+      }
+    }
+  }
+
+  if (media_description_options.type == MEDIA_TYPE_AUDIO &&
+      !session_options.vad_enabled) {
+    // If application doesn't want CN codecs in offer.
+    StripCNCodecs(&filtered_codecs);
+  } else if (media_description_options.type == MEDIA_TYPE_VIDEO &&
+             session_options.raw_packetization_for_video) {
+    for (Codec& codec : filtered_codecs) {
+      if (codec.IsMediaCodec()) {
+        codec.packetization = kPacketizationParamRaw;
+      }
+    }
+  }
+  return filtered_codecs;
+}
+
+webrtc::RTCErrorOr<AudioCodecs> GetNegotiatedCodecsForAnswer(
+    const MediaDescriptionOptions& media_description_options,
+    const MediaSessionOptions& session_options,
+    const ContentInfo* current_content,
+    const std::vector<Codec>& codecs,
+    const std::vector<Codec>& supported_codecs) {
+  std::vector<Codec> filtered_codecs;
+
+  if (!media_description_options.codec_preferences.empty()) {
+    filtered_codecs = MatchCodecPreference(
+        media_description_options.codec_preferences, codecs, supported_codecs);
+  } else {
+    // Add the codecs from current content if it exists and is not rejected nor
+    // recycled.
+    if (current_content && !current_content->rejected &&
+        current_content->name == media_description_options.mid) {
+      if (!IsMediaContentOfType(current_content,
+                                media_description_options.type)) {
+        // Can happen if the remote side re-uses a MID while recycling.
+        LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
+                             "Media type for content with mid='" +
+                                 current_content->name +
+                                 "' does not match previous type.");
+      }
+      const MediaContentDescription* mcd = current_content->media_description();
+      for (const Codec& codec : mcd->codecs()) {
+        if (FindMatchingCodec(mcd->codecs(), codecs, codec)) {
+          filtered_codecs.push_back(codec);
+        }
+      }
+    }
+    // Add other supported video codecs.
+    std::vector<Codec> other_codecs;
+    for (const Codec& codec : supported_codecs) {
+      if (FindMatchingCodec(supported_codecs, codecs, codec) &&
+          !FindMatchingCodec(supported_codecs, filtered_codecs, codec)) {
+        // We should use the local codec with local parameters and the codec id
+        // would be correctly mapped in `NegotiateCodecs`.
+        other_codecs.push_back(codec);
+      }
+    }
+
+    // Use ComputeCodecsUnion to avoid having duplicate payload IDs.
+    // This is a no-op for audio until RTX is added.
+    filtered_codecs = ComputeCodecsUnion(filtered_codecs, other_codecs);
+  }
+
+  if (media_description_options.type == MEDIA_TYPE_AUDIO &&
+      !session_options.vad_enabled) {
+    // If application doesn't want CN codecs in offer.
+    StripCNCodecs(&filtered_codecs);
+  } else if (media_description_options.type == MEDIA_TYPE_VIDEO &&
+             session_options.raw_packetization_for_video) {
+    for (Codec& codec : filtered_codecs) {
+      if (codec.IsMediaCodec()) {
+        codec.packetization = kPacketizationParamRaw;
+      }
+    }
+  }
+  return filtered_codecs;
+}
+
 }  // namespace
 
 void MediaDescriptionOptions::AddAudioSender(
@@ -1683,15 +1796,16 @@ MediaSessionDescriptionFactory::CreateOfferOrError(
     RTCError error;
     switch (media_description_options.type) {
       case MEDIA_TYPE_AUDIO:
-        error = AddAudioContentForOffer(
-            media_description_options, session_options, current_content,
-            current_description, extensions_with_ids.audio, offer_audio_codecs,
-            &current_streams, offer.get(), &ice_credentials);
-        break;
       case MEDIA_TYPE_VIDEO:
-        error = AddVideoContentForOffer(
+        error = AddRtpContentForOffer(
             media_description_options, session_options, current_content,
-            current_description, extensions_with_ids.video, offer_video_codecs,
+            current_description,
+            media_description_options.type == MEDIA_TYPE_AUDIO
+                ? extensions_with_ids.audio
+                : extensions_with_ids.video,
+            media_description_options.type == MEDIA_TYPE_AUDIO
+                ? offer_audio_codecs
+                : offer_video_codecs,
             &current_streams, offer.get(), &ice_credentials);
         break;
       case MEDIA_TYPE_DATA:
@@ -1852,18 +1966,15 @@ MediaSessionDescriptionFactory::CreateAnswerOrError(
     RTCError error;
     switch (media_description_options.type) {
       case MEDIA_TYPE_AUDIO:
-        error = AddAudioContentForAnswer(
-            media_description_options, session_options, offer_content, offer,
-            current_content, current_description, bundle_transport,
-            answer_audio_codecs, header_extensions, &current_streams,
-            answer.get(), &ice_credentials);
-        break;
       case MEDIA_TYPE_VIDEO:
-        error = AddVideoContentForAnswer(
+        error = AddRtpContentForAnswer(
             media_description_options, session_options, offer_content, offer,
             current_content, current_description, bundle_transport,
-            answer_video_codecs, header_extensions, &current_streams,
-            answer.get(), &ice_credentials);
+            media_description_options.type == MEDIA_TYPE_AUDIO
+                ? answer_audio_codecs
+                : answer_video_codecs,
+            header_extensions, &current_streams, answer.get(),
+            &ice_credentials);
         break;
       case MEDIA_TYPE_DATA:
         error = AddDataContentForAnswer(
@@ -2035,15 +2146,14 @@ void MergeCodecsFromDescription(
     const std::vector<const ContentInfo*>& current_active_contents,
     AudioCodecs* audio_codecs,
     VideoCodecs* video_codecs,
-    UsedPayloadTypes* used_pltypes,
-    const webrtc::FieldTrialsView* field_trials) {
+    UsedPayloadTypes* used_pltypes) {
   for (const ContentInfo* content : current_active_contents) {
     if (IsMediaContentOfType(content, MEDIA_TYPE_AUDIO)) {
       MergeCodecs(content->media_description()->codecs(), audio_codecs,
-                  used_pltypes, field_trials);
+                  used_pltypes);
     } else if (IsMediaContentOfType(content, MEDIA_TYPE_VIDEO)) {
       MergeCodecs(content->media_description()->codecs(), video_codecs,
-                  used_pltypes, field_trials);
+                  used_pltypes);
     }
   }
 }
@@ -2058,18 +2168,16 @@ void MediaSessionDescriptionFactory::GetCodecsForOffer(
     const std::vector<const ContentInfo*>& current_active_contents,
     AudioCodecs* audio_codecs,
     VideoCodecs* video_codecs) const {
-  const webrtc::FieldTrialsView* field_trials =
-      &transport_desc_factory_->trials();
   // First - get all codecs from the current description if the media type
   // is used. Add them to `used_pltypes` so the payload type is not reused if a
   // new media type is added.
   UsedPayloadTypes used_pltypes;
   MergeCodecsFromDescription(current_active_contents, audio_codecs,
-                             video_codecs, &used_pltypes, field_trials);
+                             video_codecs, &used_pltypes);
 
   // Add our codecs that are not in the current description.
-  MergeCodecs(all_audio_codecs_, audio_codecs, &used_pltypes, field_trials);
-  MergeCodecs(all_video_codecs_, video_codecs, &used_pltypes, field_trials);
+  MergeCodecs(all_audio_codecs_, audio_codecs, &used_pltypes);
+  MergeCodecs(all_video_codecs_, video_codecs, &used_pltypes);
 }
 
 // Getting codecs for an answer involves these steps:
@@ -2084,14 +2192,12 @@ void MediaSessionDescriptionFactory::GetCodecsForAnswer(
     const SessionDescription& remote_offer,
     AudioCodecs* audio_codecs,
     VideoCodecs* video_codecs) const {
-  const webrtc::FieldTrialsView* field_trials =
-      &transport_desc_factory_->trials();
   // First - get all codecs from the current description if the media type
   // is used. Add them to `used_pltypes` so the payload type is not reused if a
   // new media type is added.
   UsedPayloadTypes used_pltypes;
   MergeCodecsFromDescription(current_active_contents, audio_codecs,
-                             video_codecs, &used_pltypes, field_trials);
+                             video_codecs, &used_pltypes);
 
   // Second - filter out codecs that we don't support at all and should ignore.
   AudioCodecs filtered_offered_audio_codecs;
@@ -2101,9 +2207,9 @@ void MediaSessionDescriptionFactory::GetCodecsForAnswer(
       std::vector<Codec> offered_codecs = content.media_description()->codecs();
       for (const Codec& offered_audio_codec : offered_codecs) {
         if (!FindMatchingCodec(offered_codecs, filtered_offered_audio_codecs,
-                               offered_audio_codec, field_trials) &&
+                               offered_audio_codec) &&
             FindMatchingCodec(offered_codecs, all_audio_codecs_,
-                              offered_audio_codec, field_trials)) {
+                              offered_audio_codec)) {
           filtered_offered_audio_codecs.push_back(offered_audio_codec);
         }
       }
@@ -2111,9 +2217,9 @@ void MediaSessionDescriptionFactory::GetCodecsForAnswer(
       std::vector<Codec> offered_codecs = content.media_description()->codecs();
       for (const Codec& offered_video_codec : offered_codecs) {
         if (!FindMatchingCodec(offered_codecs, filtered_offered_video_codecs,
-                               offered_video_codec, field_trials) &&
+                               offered_video_codec) &&
             FindMatchingCodec(offered_codecs, all_video_codecs_,
-                              offered_video_codec, field_trials)) {
+                              offered_video_codec)) {
           filtered_offered_video_codecs.push_back(offered_video_codec);
         }
       }
@@ -2122,10 +2228,8 @@ void MediaSessionDescriptionFactory::GetCodecsForAnswer(
 
   // Add codecs that are not in the current description but were in
   // `remote_offer`.
-  MergeCodecs(filtered_offered_audio_codecs, audio_codecs, &used_pltypes,
-              field_trials);
-  MergeCodecs(filtered_offered_video_codecs, video_codecs, &used_pltypes,
-              field_trials);
+  MergeCodecs(filtered_offered_audio_codecs, audio_codecs, &used_pltypes);
+  MergeCodecs(filtered_offered_video_codecs, video_codecs, &used_pltypes);
 }
 
 MediaSessionDescriptionFactory::AudioVideoRtpHeaderExtensions
@@ -2237,228 +2341,85 @@ RTCError MediaSessionDescriptionFactory::AddTransportAnswer(
   return RTCError::OK();
 }
 
-// `audio_codecs` = set of all possible codecs that can be used, with correct
+// `codecs` = set of all possible codecs that can be used, with correct
 // payload type mappings
 //
-// `supported_audio_codecs` = set of codecs that are supported for the direction
+// `supported_codecs` = set of codecs that are supported for the direction
 // of this m= section
+// `current_content` = current description, may be null.
+// current_content->codecs() = set of previously negotiated codecs for this m=
+// section
 //
-// mcd->codecs() = set of previously negotiated codecs for this m= section
-//
-// The payload types should come from audio_codecs, but the order should come
-// from mcd->codecs() and then supported_codecs, to ensure that re-offers don't
-// change existing codec priority, and that new codecs are added with the right
-// priority.
-RTCError MediaSessionDescriptionFactory::AddAudioContentForOffer(
+// The payload types should come from codecs, but the order should come
+// from current_content->codecs() and then supported_codecs, to ensure that
+// re-offers don't change existing codec priority, and that new codecs are added
+// with the right priority.
+RTCError MediaSessionDescriptionFactory::AddRtpContentForOffer(
     const MediaDescriptionOptions& media_description_options,
     const MediaSessionOptions& session_options,
     const ContentInfo* current_content,
     const SessionDescription* current_description,
-    const RtpHeaderExtensions& audio_rtp_extensions,
-    const AudioCodecs& audio_codecs,
+    const RtpHeaderExtensions& header_extensions,
+    const std::vector<Codec>& codecs,
     StreamParamsVec* current_streams,
-    SessionDescription* desc,
+    SessionDescription* session_description,
     IceCredentialsIterator* ice_credentials) const {
-  const webrtc::FieldTrialsView* field_trials =
-      &transport_desc_factory_->trials();
-  // Filter audio_codecs (which includes all codecs, with correctly remapped
-  // payload types) based on transceiver direction.
-  const AudioCodecs& supported_audio_codecs =
-      GetAudioCodecsForOffer(media_description_options.direction);
-
-  AudioCodecs filtered_codecs;
+  RTC_DCHECK(media_description_options.type == MEDIA_TYPE_AUDIO ||
+             media_description_options.type == MEDIA_TYPE_VIDEO);
 
-  if (!media_description_options.codec_preferences.empty()) {
-    // Add the codecs from the current transceiver's codec preferences.
-    // They override any existing codecs from previous negotiations.
-    filtered_codecs = MatchCodecPreference(
-        media_description_options.codec_preferences, audio_codecs,
-        supported_audio_codecs, field_trials);
-  } else {
-    // Add the codecs from current content if it exists and is not rejected nor
-    // recycled.
-    if (current_content && !current_content->rejected &&
-        current_content->name == media_description_options.mid) {
-      if (!IsMediaContentOfType(current_content, MEDIA_TYPE_AUDIO)) {
-        // Can happen if the remote side re-uses a MID while recycling.
-        LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
-                             "Media type for content with mid='" +
-                                 current_content->name +
-                                 "' does not match previous type.");
-      }
-      const MediaContentDescription* mcd = current_content->media_description();
-      for (const Codec& codec : mcd->codecs()) {
-        if (FindMatchingCodec(mcd->codecs(), audio_codecs, codec,
-                              field_trials)) {
-          filtered_codecs.push_back(codec);
-        }
-      }
-    }
-    // Add other supported audio codecs.
-    for (const Codec& codec : supported_audio_codecs) {
-      absl::optional<Codec> found_codec = FindMatchingCodec(
-          supported_audio_codecs, audio_codecs, codec, field_trials);
-      if (found_codec &&
-          !FindMatchingCodec(supported_audio_codecs, filtered_codecs, codec,
-                             field_trials)) {
-        // Use the `found_codec` from `audio_codecs` because it has the
-        // correctly mapped payload type.
-        filtered_codecs.push_back(*found_codec);
-      }
-    }
-  }
-  if (!session_options.vad_enabled) {
-    // If application doesn't want CN codecs in offer.
-    StripCNCodecs(&filtered_codecs);
+  const std::vector<Codec>& supported_codecs =
+      media_description_options.type == MEDIA_TYPE_AUDIO
+          ? GetAudioCodecsForOffer(media_description_options.direction)
+          : GetVideoCodecsForOffer(media_description_options.direction);
+  webrtc::RTCErrorOr<std::vector<Codec>> error_or_filtered_codecs =
+      GetNegotiatedCodecsForOffer(media_description_options, session_options,
+                                  current_content, codecs, supported_codecs);
+  if (!error_or_filtered_codecs.ok()) {
+    return error_or_filtered_codecs.MoveError();
   }
 
   cricket::SecurePolicy sdes_policy =
       IsDtlsActive(current_content, current_description) ? cricket::SEC_DISABLED
                                                          : secure();
 
-  auto audio = std::make_unique<AudioContentDescription>();
   std::vector<std::string> crypto_suites;
-  GetSupportedAudioSdesCryptoSuiteNames(session_options.crypto_options,
-                                        &crypto_suites);
-  auto error = CreateMediaContentOffer(
-      media_description_options, session_options, filtered_codecs, sdes_policy,
-      GetCryptos(current_content), crypto_suites, audio_rtp_extensions,
-      ssrc_generator(), current_streams, audio.get(),
-      transport_desc_factory_->trials());
-  if (!error.ok()) {
-    return error;
-  }
-
-  bool secure_transport = (transport_desc_factory_->secure() != SEC_DISABLED);
-  SetMediaProtocol(secure_transport, audio.get());
-
-  audio->set_direction(media_description_options.direction);
-
-  desc->AddContent(media_description_options.mid, MediaProtocolType::kRtp,
-                   media_description_options.stopped, std::move(audio));
-  error = AddTransportOffer(media_description_options.mid,
-                            media_description_options.transport_options,
-                            current_description, desc, ice_credentials);
-  if (!error.ok()) {
-    return error;
-  }
-
-  return RTCError::OK();
-}
-
-// TODO(kron): This function is very similar to AddAudioContentForOffer.
-// Refactor to reuse shared code.
-RTCError MediaSessionDescriptionFactory::AddVideoContentForOffer(
-    const MediaDescriptionOptions& media_description_options,
-    const MediaSessionOptions& session_options,
-    const ContentInfo* current_content,
-    const SessionDescription* current_description,
-    const RtpHeaderExtensions& video_rtp_extensions,
-    const VideoCodecs& video_codecs,
-    StreamParamsVec* current_streams,
-    SessionDescription* desc,
-    IceCredentialsIterator* ice_credentials) const {
-  const webrtc::FieldTrialsView* field_trials =
-      &transport_desc_factory_->trials();
-  // Filter video_codecs (which includes all codecs, with correctly remapped
-  // payload types) based on transceiver direction.
-  const VideoCodecs& supported_video_codecs =
-      GetVideoCodecsForOffer(media_description_options.direction);
-
-  VideoCodecs filtered_codecs;
-
-  if (!media_description_options.codec_preferences.empty()) {
-    // Add the codecs from the current transceiver's codec preferences.
-    // They override any existing codecs from previous negotiations.
-    filtered_codecs = MatchCodecPreference(
-        media_description_options.codec_preferences, video_codecs,
-        supported_video_codecs, field_trials);
+  if (media_description_options.type == MEDIA_TYPE_AUDIO) {
+    GetSupportedAudioSdesCryptoSuiteNames(session_options.crypto_options,
+                                          &crypto_suites);
   } else {
-    // Add the codecs from current content if it exists and is not rejected nor
-    // recycled.
-    if (current_content && !current_content->rejected &&
-        current_content->name == media_description_options.mid) {
-      if (!IsMediaContentOfType(current_content, MEDIA_TYPE_VIDEO)) {
-        // Can happen if the remote side re-uses a MID while recycling.
-        LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
-                             "Media type for content with mid='" +
-                                 current_content->name +
-                                 "' does not match previous type.");
-      }
-      const MediaContentDescription* mcd = current_content->media_description();
-      for (const Codec& codec : mcd->codecs()) {
-        if (FindMatchingCodec(mcd->codecs(), video_codecs, codec,
-                              field_trials)) {
-          filtered_codecs.push_back(codec);
-        }
-      }
-    }
-    // Add other supported video codecs.
-    for (const Codec& codec : supported_video_codecs) {
-      absl::optional<Codec> found_codec = FindMatchingCodec(
-          supported_video_codecs, video_codecs, codec, field_trials);
-      if (found_codec &&
-          !FindMatchingCodec(supported_video_codecs, filtered_codecs, codec,
-                             field_trials)) {
-        // Use the `found_codec` from `video_codecs` because it has the
-        // correctly mapped payload type.
-        if (IsRtxCodec(codec)) {
-          // For RTX we might need to adjust the apt parameter if we got a
-          // remote offer without RTX for a codec for which we support RTX.
-          auto referenced_codec =
-              GetAssociatedCodecForRtx(supported_video_codecs, codec);
-          RTC_DCHECK(referenced_codec);
-
-          // Find the codec we should be referencing and point to it.
-          absl::optional<Codec> changed_referenced_codec =
-              FindMatchingCodec(supported_video_codecs, filtered_codecs,
-                                *referenced_codec, field_trials);
-          if (changed_referenced_codec) {
-            found_codec->SetParam(kCodecParamAssociatedPayloadType,
-                                  changed_referenced_codec->id);
-          }
-        }
-        filtered_codecs.push_back(*found_codec);
-      }
-    }
+    GetSupportedVideoSdesCryptoSuiteNames(session_options.crypto_options,
+                                          &crypto_suites);
   }
 
-  if (session_options.raw_packetization_for_video) {
-    for (Codec& codec : filtered_codecs) {
-      if (codec.IsMediaCodec()) {
-        codec.packetization = kPacketizationParamRaw;
-      }
-    }
+  std::unique_ptr<MediaContentDescription> content_description;
+  if (media_description_options.type == MEDIA_TYPE_AUDIO) {
+    content_description = std::make_unique<AudioContentDescription>();
+  } else {
+    content_description = std::make_unique<VideoContentDescription>();
   }
 
-  cricket::SecurePolicy sdes_policy =
-      IsDtlsActive(current_content, current_description) ? cricket::SEC_DISABLED
-                                                         : secure();
-  auto video = std::make_unique<VideoContentDescription>();
-  std::vector<std::string> crypto_suites;
-  GetSupportedVideoSdesCryptoSuiteNames(session_options.crypto_options,
-                                        &crypto_suites);
   auto error = CreateMediaContentOffer(
-      media_description_options, session_options, filtered_codecs, sdes_policy,
-      GetCryptos(current_content), crypto_suites, video_rtp_extensions,
-      ssrc_generator(), current_streams, video.get(),
+      media_description_options, session_options,
+      error_or_filtered_codecs.MoveValue(), sdes_policy,
+      GetCryptos(current_content), crypto_suites, header_extensions,
+      ssrc_generator(), current_streams, content_description.get(),
       transport_desc_factory_->trials());
   if (!error.ok()) {
     return error;
   }
 
-  video->set_bandwidth(kAutoBandwidth);
-
-  bool secure_transport = (transport_desc_factory_->secure() != SEC_DISABLED);
-  SetMediaProtocol(secure_transport, video.get());
+  bool secure_transport = transport_desc_factory_->secure() != SEC_DISABLED;
+  SetMediaProtocol(secure_transport, content_description.get());
 
-  video->set_direction(media_description_options.direction);
+  content_description->set_direction(media_description_options.direction);
 
-  desc->AddContent(media_description_options.mid, MediaProtocolType::kRtp,
-                   media_description_options.stopped, std::move(video));
+  session_description->AddContent(
+      media_description_options.mid, MediaProtocolType::kRtp,
+      media_description_options.stopped, std::move(content_description));
   return AddTransportOffer(media_description_options.mid,
                            media_description_options.transport_options,
-                           current_description, desc, ice_credentials);
+                           current_description, session_description,
+                           ice_credentials);
 }
 
 RTCError MediaSessionDescriptionFactory::AddDataContentForOffer(
@@ -2526,19 +2487,19 @@ RTCError MediaSessionDescriptionFactory::AddUnsupportedContentForOffer(
                            current_description, desc, ice_credentials);
 }
 
-// `audio_codecs` = set of all possible codecs that can be used, with correct
+// `codecs` = set of all possible codecs that can be used, with correct
 // payload type mappings
 //
-// `supported_audio_codecs` = set of codecs that are supported for the direction
+// `supported_codecs` = set of codecs that are supported for the direction
 // of this m= section
 //
 // mcd->codecs() = set of previously negotiated codecs for this m= section
 //
-// The payload types should come from audio_codecs, but the order should come
+// The payload types should come from codecs, but the order should come
 // from mcd->codecs() and then supported_codecs, to ensure that re-offers don't
 // change existing codec priority, and that new codecs are added with the right
 // priority.
-RTCError MediaSessionDescriptionFactory::AddAudioContentForAnswer(
+RTCError MediaSessionDescriptionFactory::AddRtpContentForAnswer(
     const MediaDescriptionOptions& media_description_options,
     const MediaSessionOptions& session_options,
     const ContentInfo* offer_content,
@@ -2546,272 +2507,104 @@ RTCError MediaSessionDescriptionFactory::AddAudioContentForAnswer(
     const ContentInfo* current_content,
     const SessionDescription* current_description,
     const TransportInfo* bundle_transport,
-    const AudioCodecs& audio_codecs,
-    const RtpHeaderExtensions& rtp_header_extensions,
+    const std::vector<Codec>& codecs,
+    const RtpHeaderExtensions& header_extensions,
     StreamParamsVec* current_streams,
     SessionDescription* answer,
     IceCredentialsIterator* ice_credentials) const {
-  const webrtc::FieldTrialsView* field_trials =
-      &transport_desc_factory_->trials();
-  RTC_CHECK(IsMediaContentOfType(offer_content, MEDIA_TYPE_AUDIO));
-  const AudioContentDescription* offer_audio_description =
-      offer_content->media_description()->as_audio();
-
-  std::unique_ptr<TransportDescription> audio_transport = CreateTransportAnswer(
+  RTC_DCHECK(media_description_options.type == MEDIA_TYPE_AUDIO ||
+             media_description_options.type == MEDIA_TYPE_VIDEO);
+  RTC_CHECK(
+      IsMediaContentOfType(offer_content, media_description_options.type));
+  const RtpMediaContentDescription* offer_content_description;
+  if (media_description_options.type == MEDIA_TYPE_AUDIO) {
+    offer_content_description = offer_content->media_description()->as_audio();
+  } else {
+    offer_content_description = offer_content->media_description()->as_video();
+  }
+  std::unique_ptr<TransportDescription> transport = CreateTransportAnswer(
       media_description_options.mid, offer_description,
       media_description_options.transport_options, current_description,
       bundle_transport != nullptr, ice_credentials);
-  if (!audio_transport) {
+  if (!transport) {
     LOG_AND_RETURN_ERROR(
         RTCErrorType::INTERNAL_ERROR,
-        "Failed to create transport answer, audio transport is missing");
+        "Failed to create transport answer, transport is missing");
   }
 
   // Pick codecs based on the requested communications direction in the offer
   // and the selected direction in the answer.
   // Note these will be filtered one final time in CreateMediaContentAnswer.
   auto wants_rtd = media_description_options.direction;
-  auto offer_rtd = offer_audio_description->direction();
+  auto offer_rtd = offer_content_description->direction();
   auto answer_rtd = NegotiateRtpTransceiverDirection(offer_rtd, wants_rtd);
-  AudioCodecs supported_audio_codecs =
-      GetAudioCodecsForAnswer(offer_rtd, answer_rtd);
 
-  AudioCodecs filtered_codecs;
-
-  if (!media_description_options.codec_preferences.empty()) {
-    filtered_codecs = MatchCodecPreference(
-        media_description_options.codec_preferences, audio_codecs,
-        supported_audio_codecs, field_trials);
-  } else {
-    // Add the codecs from current content if it exists and is not rejected nor
-    // recycled.
-    if (current_content && !current_content->rejected &&
-        current_content->name == media_description_options.mid) {
-      if (!IsMediaContentOfType(current_content, MEDIA_TYPE_AUDIO)) {
-        // Can happen if the remote side re-uses a MID while recycling.
-        LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
-                             "Media type for content with mid='" +
-                                 current_content->name +
-                                 "' does not match previous type.");
-      }
-      const MediaContentDescription* mcd = current_content->media_description();
-      for (const Codec& codec : mcd->codecs()) {
-        if (FindMatchingCodec(mcd->codecs(), audio_codecs, codec,
-                              field_trials)) {
-          filtered_codecs.push_back(codec);
-        }
-      }
-    }
-    // Add other supported audio codecs.
-    for (const Codec& codec : supported_audio_codecs) {
-      if (FindMatchingCodec(supported_audio_codecs, audio_codecs, codec,
-                            field_trials) &&
-          !FindMatchingCodec(supported_audio_codecs, filtered_codecs, codec,
-                             field_trials)) {
-        // We should use the local codec with local parameters and the codec id
-        // would be correctly mapped in `NegotiateCodecs`.
-        filtered_codecs.push_back(codec);
-      }
-    }
-  }
-  if (!session_options.vad_enabled) {
-    // If application doesn't want CN codecs in answer.
-    StripCNCodecs(&filtered_codecs);
+  const std::vector<Codec>& supported_codecs =
+      media_description_options.type == MEDIA_TYPE_AUDIO
+          ? GetAudioCodecsForAnswer(offer_rtd, answer_rtd)
+          : GetVideoCodecsForAnswer(offer_rtd, answer_rtd);
+  webrtc::RTCErrorOr<std::vector<Codec>> error_or_filtered_codecs =
+      GetNegotiatedCodecsForAnswer(media_description_options, session_options,
+                                   current_content, codecs, supported_codecs);
+  if (!error_or_filtered_codecs.ok()) {
+    return error_or_filtered_codecs.MoveError();
   }
+  auto filtered_codecs = error_or_filtered_codecs.MoveValue();
 
   // Determine if we have media codecs in common.
   bool has_common_media_codecs =
       std::find_if(filtered_codecs.begin(), filtered_codecs.end(),
-                   [](const AudioCodec& c) {
-                     return !(IsRedCodec(c) || IsComfortNoiseCodec(c));
+                   [](const Codec& c) {
+                     return c.IsMediaCodec() && !IsComfortNoiseCodec(c);
                    }) != filtered_codecs.end();
 
   bool bundle_enabled = offer_description->HasGroup(GROUP_TYPE_BUNDLE) &&
                         session_options.bundle_enabled;
-  auto audio_answer = std::make_unique<AudioContentDescription>();
+  std::unique_ptr<MediaContentDescription> answer_content;
+  if (media_description_options.type == MEDIA_TYPE_AUDIO) {
+    answer_content = std::make_unique<AudioContentDescription>();
+  } else {
+    answer_content = std::make_unique<VideoContentDescription>();
+  }
   // Do not require or create SDES cryptos if DTLS is used.
   cricket::SecurePolicy sdes_policy =
-      audio_transport->secure() ? cricket::SEC_DISABLED : secure();
-  if (!SetCodecsInAnswer(offer_audio_description, filtered_codecs,
-                         media_description_options, session_options,
-                         ssrc_generator(), current_streams, audio_answer.get(),
-                         transport_desc_factory_->trials())) {
+      transport->secure() ? cricket::SEC_DISABLED : secure();
+  if (!SetCodecsInAnswer(
+          offer_content_description, filtered_codecs, media_description_options,
+          session_options, ssrc_generator(), current_streams,
+          answer_content.get(), transport_desc_factory_->trials())) {
     LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
                          "Failed to set codecs in answer");
   }
   if (!CreateMediaContentAnswer(
-          offer_audio_description, media_description_options, session_options,
+          offer_content_description, media_description_options, session_options,
           sdes_policy, GetCryptos(current_content),
-          filtered_rtp_header_extensions(rtp_header_extensions),
-          ssrc_generator(), enable_encrypted_rtp_header_extensions_,
-          current_streams, bundle_enabled, audio_answer.get())) {
+          filtered_rtp_header_extensions(header_extensions), ssrc_generator(),
+          enable_encrypted_rtp_header_extensions_, current_streams,
+          bundle_enabled, answer_content.get())) {
     LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
                          "Failed to create answer");
   }
 
   bool secure = bundle_transport ? bundle_transport->description.secure()
-                                 : audio_transport->secure();
+                                 : transport->secure();
   bool rejected = media_description_options.stopped ||
                   offer_content->rejected || !has_common_media_codecs ||
                   !IsMediaProtocolSupported(MEDIA_TYPE_AUDIO,
-                                            audio_answer->protocol(), secure);
-  auto error = AddTransportAnswer(media_description_options.mid,
-                                  *(audio_transport.get()), answer);
-  if (!error.ok()) {
-    return error;
-  }
-
+                                            answer_content->protocol(), secure);
   if (rejected) {
-    RTC_LOG(LS_INFO) << "Audio m= section '" << media_description_options.mid
+    RTC_LOG(LS_INFO) << "m= section '" << media_description_options.mid
                      << "' being rejected in answer.";
   }
 
-  answer->AddContent(media_description_options.mid, offer_content->type,
-                     rejected, std::move(audio_answer));
-  return RTCError::OK();
-}
-
-// TODO(kron): This function is very similar to AddAudioContentForAnswer.
-// Refactor to reuse shared code.
-RTCError MediaSessionDescriptionFactory::AddVideoContentForAnswer(
-    const MediaDescriptionOptions& media_description_options,
-    const MediaSessionOptions& session_options,
-    const ContentInfo* offer_content,
-    const SessionDescription* offer_description,
-    const ContentInfo* current_content,
-    const SessionDescription* current_description,
-    const TransportInfo* bundle_transport,
-    const VideoCodecs& video_codecs,
-    const RtpHeaderExtensions& default_video_rtp_header_extensions,
-    StreamParamsVec* current_streams,
-    SessionDescription* answer,
-    IceCredentialsIterator* ice_credentials) const {
-  const webrtc::FieldTrialsView* field_trials =
-      &transport_desc_factory_->trials();
-  RTC_CHECK(IsMediaContentOfType(offer_content, MEDIA_TYPE_VIDEO));
-  const VideoContentDescription* offer_video_description =
-      offer_content->media_description()->as_video();
-
-  std::unique_ptr<TransportDescription> video_transport = CreateTransportAnswer(
-      media_description_options.mid, offer_description,
-      media_description_options.transport_options, current_description,
-      bundle_transport != nullptr, ice_credentials);
-  if (!video_transport) {
-    LOG_AND_RETURN_ERROR(
-        RTCErrorType::INTERNAL_ERROR,
-        "Failed to create transport answer, video transport is missing");
-  }
-
-  // Pick codecs based on the requested communications direction in the offer
-  // and the selected direction in the answer.
-  // Note these will be filtered one final time in CreateMediaContentAnswer.
-  auto wants_rtd = media_description_options.direction;
-  auto offer_rtd = offer_video_description->direction();
-  auto answer_rtd = NegotiateRtpTransceiverDirection(offer_rtd, wants_rtd);
-  VideoCodecs supported_video_codecs =
-      GetVideoCodecsForAnswer(offer_rtd, answer_rtd);
-
-  VideoCodecs filtered_codecs;
-
-  if (!media_description_options.codec_preferences.empty()) {
-    filtered_codecs = MatchCodecPreference(
-        media_description_options.codec_preferences, video_codecs,
-        supported_video_codecs, field_trials);
-  } else {
-    // Add the codecs from current content if it exists and is not rejected nor
-    // recycled.
-    if (current_content && !current_content->rejected &&
-        current_content->name == media_description_options.mid) {
-      if (!IsMediaContentOfType(current_content, MEDIA_TYPE_VIDEO)) {
-        // Can happen if the remote side re-uses a MID while recycling.
-        LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
-                             "Media type for content with mid='" +
-                                 current_content->name +
-                                 "' does not match previous type.");
-      }
-      const MediaContentDescription* mcd = current_content->media_description();
-      for (const Codec& codec : mcd->codecs()) {
-        if (FindMatchingCodec(mcd->codecs(), video_codecs, codec,
-                              field_trials)) {
-          filtered_codecs.push_back(codec);
-        }
-      }
-    }
-
-    // Add other supported video codecs.
-    VideoCodecs other_video_codecs;
-    for (const Codec& codec : supported_video_codecs) {
-      if (FindMatchingCodec(supported_video_codecs, video_codecs, codec,
-                            field_trials) &&
-          !FindMatchingCodec(supported_video_codecs, filtered_codecs, codec,
-                             field_trials)) {
-        // We should use the local codec with local parameters and the codec id
-        // would be correctly mapped in `NegotiateCodecs`.
-        other_video_codecs.push_back(codec);
-      }
-    }
-
-    // Use ComputeCodecsUnion to avoid having duplicate payload IDs
-    filtered_codecs =
-        ComputeCodecsUnion(filtered_codecs, other_video_codecs, field_trials);
-  }
-  // Determine if we have media codecs in common.
-  bool has_common_media_codecs =
-      std::find_if(
-          filtered_codecs.begin(), filtered_codecs.end(), [](const Codec& c) {
-            return !(IsRedCodec(c) || IsUlpfecCodec(c) || IsFlexfecCodec(c));
-          }) != filtered_codecs.end();
-
-  if (session_options.raw_packetization_for_video) {
-    for (Codec& codec : filtered_codecs) {
-      if (codec.IsMediaCodec()) {
-        codec.packetization = kPacketizationParamRaw;
-      }
-    }
-  }
-
-  bool bundle_enabled = offer_description->HasGroup(GROUP_TYPE_BUNDLE) &&
-                        session_options.bundle_enabled;
-  auto video_answer = std::make_unique<VideoContentDescription>();
-  // Do not require or create SDES cryptos if DTLS is used.
-  cricket::SecurePolicy sdes_policy =
-      video_transport->secure() ? cricket::SEC_DISABLED : secure();
-  if (!SetCodecsInAnswer(offer_video_description, filtered_codecs,
-                         media_description_options, session_options,
-                         ssrc_generator(), current_streams, video_answer.get(),
-                         transport_desc_factory_->trials())) {
-    LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
-                         "Failed to set codecs in answer");
-  }
-  if (!CreateMediaContentAnswer(
-          offer_video_description, media_description_options, session_options,
-          sdes_policy, GetCryptos(current_content),
-          filtered_rtp_header_extensions(default_video_rtp_header_extensions),
-          ssrc_generator(), enable_encrypted_rtp_header_extensions_,
-          current_streams, bundle_enabled, video_answer.get())) {
-    LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
-                         "Failed to create answer");
-  }
-  bool secure = bundle_transport ? bundle_transport->description.secure()
-                                 : video_transport->secure();
-  bool rejected = media_description_options.stopped ||
-                  offer_content->rejected || !has_common_media_codecs ||
-                  !IsMediaProtocolSupported(MEDIA_TYPE_VIDEO,
-                                            video_answer->protocol(), secure);
   auto error = AddTransportAnswer(media_description_options.mid,
-                                  *(video_transport.get()), answer);
+                                  *(transport.get()), answer);
   if (!error.ok()) {
     return error;
   }
 
-  if (!rejected) {
-    video_answer->set_bandwidth(kAutoBandwidth);
-  } else {
-    RTC_LOG(LS_INFO) << "Video m= section '" << media_description_options.mid
-                     << "' being rejected in answer.";
-  }
   answer->AddContent(media_description_options.mid, offer_content->type,
-                     rejected, std::move(video_answer));
+                     rejected, std::move(answer_content));
   return RTCError::OK();
 }
 
@@ -2934,23 +2727,19 @@ RTCError MediaSessionDescriptionFactory::AddUnsupportedContentForAnswer(
 }
 
 void MediaSessionDescriptionFactory::ComputeAudioCodecsIntersectionAndUnion() {
-  const webrtc::FieldTrialsView* field_trials =
-      &transport_desc_factory_->trials();
   audio_sendrecv_codecs_.clear();
   all_audio_codecs_.clear();
   // Compute the audio codecs union.
   for (const Codec& send : audio_send_codecs_) {
     all_audio_codecs_.push_back(send);
-    if (!FindMatchingCodec(audio_send_codecs_, audio_recv_codecs_, send,
-                           field_trials)) {
+    if (!FindMatchingCodec(audio_send_codecs_, audio_recv_codecs_, send)) {
       // It doesn't make sense to have an RTX codec we support sending but not
       // receiving.
-      RTC_DCHECK(!IsRtxCodec(send));
+      RTC_DCHECK(send.GetResiliencyType() != Codec::ResiliencyType::kRtx);
     }
   }
   for (const Codec& recv : audio_recv_codecs_) {
-    if (!FindMatchingCodec(audio_recv_codecs_, audio_send_codecs_, recv,
-                           field_trials)) {
+    if (!FindMatchingCodec(audio_recv_codecs_, audio_send_codecs_, recv)) {
       all_audio_codecs_.push_back(recv);
     }
   }
@@ -2960,17 +2749,15 @@ void MediaSessionDescriptionFactory::ComputeAudioCodecsIntersectionAndUnion() {
   // expensive than decoding, and prioritizing a codec in the send list probably
   // means it's a codec we can handle efficiently.
   NegotiateCodecs(audio_recv_codecs_, audio_send_codecs_,
-                  &audio_sendrecv_codecs_, true, field_trials);
+                  &audio_sendrecv_codecs_, true);
 }
 
 void MediaSessionDescriptionFactory::ComputeVideoCodecsIntersectionAndUnion() {
-  const webrtc::FieldTrialsView* field_trials =
-      &transport_desc_factory_->trials();
   video_sendrecv_codecs_.clear();
 
   // Use ComputeCodecsUnion to avoid having duplicate payload IDs
   all_video_codecs_ =
-      ComputeCodecsUnion(video_recv_codecs_, video_send_codecs_, field_trials);
+      ComputeCodecsUnion(video_recv_codecs_, video_send_codecs_);
 
   // Use NegotiateCodecs to merge our codec lists, since the operation is
   // essentially the same. Put send_codecs as the offered_codecs, which is the
@@ -2978,7 +2765,7 @@ void MediaSessionDescriptionFactory::ComputeVideoCodecsIntersectionAndUnion() {
   // expensive than decoding, and prioritizing a codec in the send list probably
   // means it's a codec we can handle efficiently.
   NegotiateCodecs(video_recv_codecs_, video_send_codecs_,
-                  &video_sendrecv_codecs_, true, field_trials);
+                  &video_sendrecv_codecs_, true);
 }
 
 bool IsMediaContent(const ContentInfo* content) {
diff --git a/third_party/libwebrtc/pc/media_session.h b/third_party/libwebrtc/pc/media_session.h
index 3100fb6fdb..0b3cb67c35 100644
--- a/third_party/libwebrtc/pc/media_session.h
+++ b/third_party/libwebrtc/pc/media_session.h
@@ -19,7 +19,6 @@
 #include <vector>
 
 #include "api/crypto/crypto_options.h"
-#include "api/field_trials_view.h"
 #include "api/media_types.h"
 #include "api/rtp_parameters.h"
 #include "api/rtp_transceiver_direction.h"
@@ -230,28 +229,14 @@ class MediaSessionDescriptionFactory {
       const TransportDescription& transport_desc,
       SessionDescription* answer_desc) const;
 
-  // Helpers for adding media contents to the SessionDescription. Returns true
-  // it succeeds or the media content is not needed, or false if there is any
-  // error.
-
-  webrtc::RTCError AddAudioContentForOffer(
-      const MediaDescriptionOptions& media_description_options,
-      const MediaSessionOptions& session_options,
-      const ContentInfo* current_content,
-      const SessionDescription* current_description,
-      const RtpHeaderExtensions& audio_rtp_extensions,
-      const AudioCodecs& audio_codecs,
-      StreamParamsVec* current_streams,
-      SessionDescription* desc,
-      IceCredentialsIterator* ice_credentials) const;
-
-  webrtc::RTCError AddVideoContentForOffer(
+  // Helpers for adding media contents to the SessionDescription.
+  webrtc::RTCError AddRtpContentForOffer(
       const MediaDescriptionOptions& media_description_options,
       const MediaSessionOptions& session_options,
       const ContentInfo* current_content,
       const SessionDescription* current_description,
-      const RtpHeaderExtensions& video_rtp_extensions,
-      const VideoCodecs& video_codecs,
+      const RtpHeaderExtensions& header_extensions,
+      const std::vector<Codec>& codecs,
       StreamParamsVec* current_streams,
       SessionDescription* desc,
       IceCredentialsIterator* ice_credentials) const;
@@ -273,21 +258,7 @@ class MediaSessionDescriptionFactory {
       SessionDescription* desc,
       IceCredentialsIterator* ice_credentials) const;
 
-  webrtc::RTCError AddAudioContentForAnswer(
-      const MediaDescriptionOptions& media_description_options,
-      const MediaSessionOptions& session_options,
-      const ContentInfo* offer_content,
-      const SessionDescription* offer_description,
-      const ContentInfo* current_content,
-      const SessionDescription* current_description,
-      const TransportInfo* bundle_transport,
-      const AudioCodecs& audio_codecs,
-      const RtpHeaderExtensions& rtp_header_extensions,
-      StreamParamsVec* current_streams,
-      SessionDescription* answer,
-      IceCredentialsIterator* ice_credentials) const;
-
-  webrtc::RTCError AddVideoContentForAnswer(
+  webrtc::RTCError AddRtpContentForAnswer(
       const MediaDescriptionOptions& media_description_options,
       const MediaSessionOptions& session_options,
       const ContentInfo* offer_content,
@@ -295,8 +266,8 @@ class MediaSessionDescriptionFactory {
       const ContentInfo* current_content,
       const SessionDescription* current_description,
       const TransportInfo* bundle_transport,
-      const VideoCodecs& video_codecs,
-      const RtpHeaderExtensions& rtp_header_extensions,
+      const std::vector<Codec>& codecs,
+      const RtpHeaderExtensions& header_extensions,
       StreamParamsVec* current_streams,
       SessionDescription* answer,
       IceCredentialsIterator* ice_credentials) const;
diff --git a/third_party/libwebrtc/pc/media_session_unittest.cc b/third_party/libwebrtc/pc/media_session_unittest.cc
index a1770c18c5..641f638e72 100644
--- a/third_party/libwebrtc/pc/media_session_unittest.cc
+++ b/third_party/libwebrtc/pc/media_session_unittest.cc
@@ -54,50 +54,16 @@
   ASSERT_EQ(s, cd->cryptos().size()); \
   ASSERT_EQ(cs, cd->cryptos()[0].crypto_suite)
 
-typedef std::vector<cricket::Candidate> Candidates;
-
-using cricket::AudioCodec;
-using cricket::AudioContentDescription;
-using cricket::ContentInfo;
-using cricket::CryptoParamsVec;
-using cricket::GetFirstAudioContent;
-using cricket::GetFirstAudioContentDescription;
-using cricket::GetFirstDataContent;
-using cricket::GetFirstVideoContent;
-using cricket::GetFirstVideoContentDescription;
-using cricket::kAutoBandwidth;
-using cricket::MEDIA_TYPE_AUDIO;
-using cricket::MEDIA_TYPE_DATA;
-using cricket::MEDIA_TYPE_VIDEO;
-using cricket::MediaContentDescription;
-using cricket::MediaDescriptionOptions;
-using cricket::MediaProtocolType;
-using cricket::MediaSessionDescriptionFactory;
-using cricket::MediaSessionOptions;
-using cricket::MediaType;
-using cricket::RidDescription;
-using cricket::RidDirection;
-using cricket::SctpDataContentDescription;
-using cricket::SEC_DISABLED;
-using cricket::SEC_ENABLED;
-using cricket::SEC_REQUIRED;
-using cricket::SessionDescription;
-using cricket::SimulcastDescription;
-using cricket::SimulcastLayer;
-using cricket::SimulcastLayerList;
-using cricket::SsrcGroup;
-using cricket::StreamParams;
-using cricket::StreamParamsVec;
-using cricket::TransportDescription;
-using cricket::TransportDescriptionFactory;
-using cricket::TransportInfo;
-using cricket::VideoCodec;
-using cricket::VideoContentDescription;
-using rtc::kCsAeadAes128Gcm;
-using rtc::kCsAeadAes256Gcm;
-using rtc::kCsAesCm128HmacSha1_32;
-using rtc::kCsAesCm128HmacSha1_80;
-using rtc::UniqueRandomIdGenerator;
+namespace cricket {
+namespace {
+
+using ::rtc::kCsAeadAes128Gcm;
+using ::rtc::kCsAeadAes256Gcm;
+using ::rtc::kCsAesCm128HmacSha1_32;
+using ::rtc::kCsAesCm128HmacSha1_80;
+using ::rtc::UniqueRandomIdGenerator;
+using ::testing::Bool;
+using ::testing::Combine;
 using ::testing::Contains;
 using ::testing::Each;
 using ::testing::ElementsAre;
@@ -110,146 +76,145 @@ using ::testing::Ne;
 using ::testing::Not;
 using ::testing::Pointwise;
 using ::testing::SizeIs;
-using webrtc::RtpExtension;
-using webrtc::RtpTransceiverDirection;
+using ::testing::Values;
+using ::testing::ValuesIn;
+using ::webrtc::RtpExtension;
+using ::webrtc::RtpTransceiverDirection;
+
+using Candidates = std::vector<Candidate>;
 
-static AudioCodec createRedAudioCodec(absl::string_view encoding_id) {
-  AudioCodec red = cricket::CreateAudioCodec(63, "red", 48000, 2);
-  red.SetParam(cricket::kCodecParamNotInNameValueFormat,
+AudioCodec CreateRedAudioCodec(absl::string_view encoding_id) {
+  AudioCodec red = CreateAudioCodec(63, "red", 48000, 2);
+  red.SetParam(kCodecParamNotInNameValueFormat,
                std::string(encoding_id) + '/' + std::string(encoding_id));
   return red;
 }
 
-static const AudioCodec kAudioCodecs1[] = {
-    cricket::CreateAudioCodec(111, "opus", 48000, 2),
-    createRedAudioCodec("111"),
-    cricket::CreateAudioCodec(102, "iLBC", 8000, 1),
-    cricket::CreateAudioCodec(0, "PCMU", 8000, 1),
-    cricket::CreateAudioCodec(8, "PCMA", 8000, 1),
-    cricket::CreateAudioCodec(117, "red", 8000, 1),
-    cricket::CreateAudioCodec(107, "CN", 48000, 1)};
+const AudioCodec kAudioCodecs1[] = {CreateAudioCodec(111, "opus", 48000, 2),
+                                    CreateRedAudioCodec("111"),
+                                    CreateAudioCodec(102, "iLBC", 8000, 1),
+                                    CreateAudioCodec(0, "PCMU", 8000, 1),
+                                    CreateAudioCodec(8, "PCMA", 8000, 1),
+                                    CreateAudioCodec(117, "red", 8000, 1),
+                                    CreateAudioCodec(107, "CN", 48000, 1)};
 
-static const AudioCodec kAudioCodecs2[] = {
-    cricket::CreateAudioCodec(126, "foo", 16000, 1),
-    cricket::CreateAudioCodec(0, "PCMU", 8000, 1),
-    cricket::CreateAudioCodec(127, "iLBC", 8000, 1),
+const AudioCodec kAudioCodecs2[] = {
+    CreateAudioCodec(126, "foo", 16000, 1),
+    CreateAudioCodec(0, "PCMU", 8000, 1),
+    CreateAudioCodec(127, "iLBC", 8000, 1),
 };
 
-static const AudioCodec kAudioCodecsAnswer[] = {
-    cricket::CreateAudioCodec(102, "iLBC", 8000, 1),
-    cricket::CreateAudioCodec(0, "PCMU", 8000, 1),
+const AudioCodec kAudioCodecsAnswer[] = {
+    CreateAudioCodec(102, "iLBC", 8000, 1),
+    CreateAudioCodec(0, "PCMU", 8000, 1),
 };
 
-static const VideoCodec kVideoCodecs1[] = {
-    cricket::CreateVideoCodec(96, "H264-SVC"),
-    cricket::CreateVideoCodec(97, "H264")};
+const VideoCodec kVideoCodecs1[] = {CreateVideoCodec(96, "H264-SVC"),
+                                    CreateVideoCodec(97, "H264")};
 
-static const VideoCodec kVideoCodecs1Reverse[] = {
-    cricket::CreateVideoCodec(97, "H264"),
-    cricket::CreateVideoCodec(96, "H264-SVC")};
+const VideoCodec kVideoCodecs1Reverse[] = {CreateVideoCodec(97, "H264"),
+                                           CreateVideoCodec(96, "H264-SVC")};
 
-static const VideoCodec kVideoCodecs2[] = {
-    cricket::CreateVideoCodec(126, "H264"),
-    cricket::CreateVideoCodec(127, "H263")};
+const VideoCodec kVideoCodecs2[] = {CreateVideoCodec(126, "H264"),
+                                    CreateVideoCodec(127, "H263")};
 
-static const VideoCodec kVideoCodecsAnswer[] = {
-    cricket::CreateVideoCodec(97, "H264")};
+const VideoCodec kVideoCodecsAnswer[] = {CreateVideoCodec(97, "H264")};
 
-static const RtpExtension kAudioRtpExtension1[] = {
+const RtpExtension kAudioRtpExtension1[] = {
     RtpExtension("urn:ietf:params:rtp-hdrext:ssrc-audio-level", 8),
     RtpExtension("http://google.com/testing/audio_something", 10),
 };
 
-static const RtpExtension kAudioRtpExtensionEncrypted1[] = {
+const RtpExtension kAudioRtpExtensionEncrypted1[] = {
     RtpExtension("urn:ietf:params:rtp-hdrext:ssrc-audio-level", 8),
     RtpExtension("http://google.com/testing/audio_something", 10),
     RtpExtension("urn:ietf:params:rtp-hdrext:ssrc-audio-level", 12, true),
     RtpExtension("http://google.com/testing/audio_something", 11, true),
 };
 
-static const RtpExtension kAudioRtpExtension2[] = {
+const RtpExtension kAudioRtpExtension2[] = {
     RtpExtension("urn:ietf:params:rtp-hdrext:ssrc-audio-level", 2),
     RtpExtension("http://google.com/testing/audio_something_else", 8),
     RtpExtension("http://google.com/testing/both_audio_and_video", 7),
 };
 
-static const RtpExtension kAudioRtpExtension3[] = {
+const RtpExtension kAudioRtpExtension3[] = {
     RtpExtension("http://google.com/testing/audio_something", 2),
     RtpExtension("http://google.com/testing/both_audio_and_video", 3),
 };
 
-static const RtpExtension kAudioRtpExtension3ForEncryption[] = {
+const RtpExtension kAudioRtpExtension3ForEncryption[] = {
     RtpExtension("http://google.com/testing/audio_something", 2),
     // Use RTP extension that supports encryption.
     RtpExtension("urn:ietf:params:rtp-hdrext:toffset", 3),
 };
 
-static const RtpExtension kAudioRtpExtension3ForEncryptionOffer[] = {
+const RtpExtension kAudioRtpExtension3ForEncryptionOffer[] = {
     RtpExtension("http://google.com/testing/audio_something", 2),
     RtpExtension("urn:ietf:params:rtp-hdrext:toffset", 3),
     RtpExtension("http://google.com/testing/audio_something", 14, true),
     RtpExtension("urn:ietf:params:rtp-hdrext:toffset", 13, true),
 };
 
-static const RtpExtension kVideoRtpExtension3ForEncryptionOffer[] = {
+const RtpExtension kVideoRtpExtension3ForEncryptionOffer[] = {
     RtpExtension("http://google.com/testing/video_something", 4),
     RtpExtension("urn:ietf:params:rtp-hdrext:toffset", 3),
     RtpExtension("http://google.com/testing/video_something", 12, true),
     RtpExtension("urn:ietf:params:rtp-hdrext:toffset", 13, true),
 };
 
-static const RtpExtension kAudioRtpExtensionAnswer[] = {
+const RtpExtension kAudioRtpExtensionAnswer[] = {
     RtpExtension("urn:ietf:params:rtp-hdrext:ssrc-audio-level", 8),
 };
 
-static const RtpExtension kAudioRtpExtensionEncryptedAnswer[] = {
+const RtpExtension kAudioRtpExtensionEncryptedAnswer[] = {
     RtpExtension("urn:ietf:params:rtp-hdrext:ssrc-audio-level", 12, true),
 };
 
-static const RtpExtension kVideoRtpExtension1[] = {
+const RtpExtension kVideoRtpExtension1[] = {
     RtpExtension("urn:ietf:params:rtp-hdrext:toffset", 14),
     RtpExtension("http://google.com/testing/video_something", 13),
 };
 
-static const RtpExtension kVideoRtpExtensionEncrypted1[] = {
+const RtpExtension kVideoRtpExtensionEncrypted1[] = {
     RtpExtension("urn:ietf:params:rtp-hdrext:toffset", 14),
     RtpExtension("http://google.com/testing/video_something", 13),
     RtpExtension("urn:ietf:params:rtp-hdrext:toffset", 9, true),
     RtpExtension("http://google.com/testing/video_something", 7, true),
 };
 
-static const RtpExtension kVideoRtpExtension2[] = {
+const RtpExtension kVideoRtpExtension2[] = {
     RtpExtension("urn:ietf:params:rtp-hdrext:toffset", 2),
     RtpExtension("http://google.com/testing/video_something_else", 14),
     RtpExtension("http://google.com/testing/both_audio_and_video", 7),
 };
 
-static const RtpExtension kVideoRtpExtension3[] = {
+const RtpExtension kVideoRtpExtension3[] = {
     RtpExtension("http://google.com/testing/video_something", 4),
     RtpExtension("http://google.com/testing/both_audio_and_video", 5),
 };
 
-static const RtpExtension kVideoRtpExtension3ForEncryption[] = {
+const RtpExtension kVideoRtpExtension3ForEncryption[] = {
     RtpExtension("http://google.com/testing/video_something", 4),
     // Use RTP extension that supports encryption.
     RtpExtension("urn:ietf:params:rtp-hdrext:toffset", 5),
 };
 
-static const RtpExtension kVideoRtpExtensionAnswer[] = {
+const RtpExtension kVideoRtpExtensionAnswer[] = {
     RtpExtension("urn:ietf:params:rtp-hdrext:toffset", 14),
 };
 
-static const RtpExtension kVideoRtpExtensionEncryptedAnswer[] = {
+const RtpExtension kVideoRtpExtensionEncryptedAnswer[] = {
     RtpExtension("urn:ietf:params:rtp-hdrext:toffset", 9, true),
 };
 
-static const RtpExtension kRtpExtensionTransportSequenceNumber01[] = {
+const RtpExtension kRtpExtensionTransportSequenceNumber01[] = {
     RtpExtension("http://www.ietf.org/id/"
                  "draft-holmer-rmcat-transport-wide-cc-extensions-01",
                  1),
 };
 
-static const RtpExtension kRtpExtensionTransportSequenceNumber01And02[] = {
+const RtpExtension kRtpExtensionTransportSequenceNumber01And02[] = {
     RtpExtension("http://www.ietf.org/id/"
                  "draft-holmer-rmcat-transport-wide-cc-extensions-01",
                  1),
@@ -258,68 +223,64 @@ static const RtpExtension kRtpExtensionTransportSequenceNumber01And02[] = {
         2),
 };
 
-static const RtpExtension kRtpExtensionTransportSequenceNumber02[] = {
+const RtpExtension kRtpExtensionTransportSequenceNumber02[] = {
     RtpExtension(
         "http://www.webrtc.org/experiments/rtp-hdrext/transport-wide-cc-02",
         2),
 };
 
-static const RtpExtension kRtpExtensionGenericFrameDescriptorUri00[] = {
+const RtpExtension kRtpExtensionGenericFrameDescriptorUri00[] = {
     RtpExtension("http://www.webrtc.org/experiments/rtp-hdrext/"
                  "generic-frame-descriptor-00",
                  3),
 };
 
-static const uint32_t kSimulcastParamsSsrc[] = {10, 11, 20, 21, 30, 31};
-static const uint32_t kSimSsrc[] = {10, 20, 30};
-static const uint32_t kFec1Ssrc[] = {10, 11};
-static const uint32_t kFec2Ssrc[] = {20, 21};
-static const uint32_t kFec3Ssrc[] = {30, 31};
-
-static const char kMediaStream1[] = "stream_1";
-static const char kMediaStream2[] = "stream_2";
-static const char kVideoTrack1[] = "video_1";
-static const char kVideoTrack2[] = "video_2";
-static const char kAudioTrack1[] = "audio_1";
-static const char kAudioTrack2[] = "audio_2";
-static const char kAudioTrack3[] = "audio_3";
-
-static const char* kMediaProtocols[] = {"RTP/AVP", "RTP/SAVP", "RTP/AVPF",
-                                        "RTP/SAVPF"};
-static const char* kMediaProtocolsDtls[] = {
-    "TCP/TLS/RTP/SAVPF", "TCP/TLS/RTP/SAVP", "UDP/TLS/RTP/SAVPF",
-    "UDP/TLS/RTP/SAVP"};
+const uint32_t kSimulcastParamsSsrc[] = {10, 11, 20, 21, 30, 31};
+const uint32_t kSimSsrc[] = {10, 20, 30};
+const uint32_t kFec1Ssrc[] = {10, 11};
+const uint32_t kFec2Ssrc[] = {20, 21};
+const uint32_t kFec3Ssrc[] = {30, 31};
+
+const char kMediaStream1[] = "stream_1";
+const char kMediaStream2[] = "stream_2";
+const char kVideoTrack1[] = "video_1";
+const char kVideoTrack2[] = "video_2";
+const char kAudioTrack1[] = "audio_1";
+const char kAudioTrack2[] = "audio_2";
+const char kAudioTrack3[] = "audio_3";
+
+const char* kMediaProtocols[] = {"RTP/AVP", "RTP/SAVP", "RTP/AVPF",
+                                 "RTP/SAVPF"};
+const char* kMediaProtocolsDtls[] = {"TCP/TLS/RTP/SAVPF", "TCP/TLS/RTP/SAVP",
+                                     "UDP/TLS/RTP/SAVPF", "UDP/TLS/RTP/SAVP"};
 
 // SRTP cipher name negotiated by the tests. This must be updated if the
 // default changes.
-static const char* kDefaultSrtpCryptoSuite = kCsAesCm128HmacSha1_80;
-static const char* kDefaultSrtpCryptoSuiteGcm = kCsAeadAes256Gcm;
-static const uint8_t kDefaultCryptoSuiteSize = 3U;
+const char* kDefaultSrtpCryptoSuite = kCsAesCm128HmacSha1_80;
+const char* kDefaultSrtpCryptoSuiteGcm = kCsAeadAes256Gcm;
+const uint8_t kDefaultCryptoSuiteSize = 3U;
 
 // These constants are used to make the code using "AddMediaDescriptionOptions"
 // more readable.
-static constexpr bool kStopped = true;
-static constexpr bool kActive = false;
+constexpr bool kStopped = true;
+constexpr bool kActive = false;
 
-static bool IsMediaContentOfType(const ContentInfo* content,
-                                 MediaType media_type) {
+bool IsMediaContentOfType(const ContentInfo* content, MediaType media_type) {
   RTC_DCHECK(content);
   return content->media_description()->type() == media_type;
 }
 
-static RtpTransceiverDirection GetMediaDirection(const ContentInfo* content) {
+RtpTransceiverDirection GetMediaDirection(const ContentInfo* content) {
   RTC_DCHECK(content);
   return content->media_description()->direction();
 }
 
-static void AddRtxCodec(const VideoCodec& rtx_codec,
-                        std::vector<VideoCodec>* codecs) {
-  ASSERT_FALSE(cricket::FindCodecById(*codecs, rtx_codec.id));
+void AddRtxCodec(const VideoCodec& rtx_codec, std::vector<VideoCodec>* codecs) {
+  ASSERT_FALSE(FindCodecById(*codecs, rtx_codec.id));
   codecs->push_back(rtx_codec);
 }
 
-static std::vector<std::string> GetCodecNames(
-    const std::vector<cricket::Codec>& codecs) {
+std::vector<std::string> GetCodecNames(const std::vector<Codec>& codecs) {
   std::vector<std::string> codec_names;
   codec_names.reserve(codecs.size());
   for (const auto& codec : codecs) {
@@ -348,29 +309,29 @@ FindFirstMediaDescriptionByMid(const std::string& mid,
 }
 
 // Add a media section to the `session_options`.
-static void AddMediaDescriptionOptions(MediaType type,
-                                       const std::string& mid,
-                                       RtpTransceiverDirection direction,
-                                       bool stopped,
-                                       MediaSessionOptions* opts) {
+void AddMediaDescriptionOptions(MediaType type,
+                                const std::string& mid,
+                                RtpTransceiverDirection direction,
+                                bool stopped,
+                                MediaSessionOptions* opts) {
   opts->media_description_options.push_back(
       MediaDescriptionOptions(type, mid, direction, stopped));
 }
 
-static void AddAudioVideoSections(RtpTransceiverDirection direction,
-                                  MediaSessionOptions* opts) {
+void AddAudioVideoSections(RtpTransceiverDirection direction,
+                           MediaSessionOptions* opts) {
   AddMediaDescriptionOptions(MEDIA_TYPE_AUDIO, "audio", direction, kActive,
                              opts);
   AddMediaDescriptionOptions(MEDIA_TYPE_VIDEO, "video", direction, kActive,
                              opts);
 }
 
-static void AddDataSection(RtpTransceiverDirection direction,
-                           MediaSessionOptions* opts) {
+void AddDataSection(RtpTransceiverDirection direction,
+                    MediaSessionOptions* opts) {
   AddMediaDescriptionOptions(MEDIA_TYPE_DATA, "data", direction, kActive, opts);
 }
 
-static void AttachSenderToMediaDescriptionOptions(
+void AttachSenderToMediaDescriptionOptions(
     const std::string& mid,
     MediaType type,
     const std::string& track_id,
@@ -393,7 +354,7 @@ static void AttachSenderToMediaDescriptionOptions(
   }
 }
 
-static void AttachSenderToMediaDescriptionOptions(
+void AttachSenderToMediaDescriptionOptions(
     const std::string& mid,
     MediaType type,
     const std::string& track_id,
@@ -405,23 +366,21 @@ static void AttachSenderToMediaDescriptionOptions(
                                         session_options);
 }
 
-static void DetachSenderFromMediaSection(const std::string& mid,
-                                         const std::string& track_id,
-                                         MediaSessionOptions* session_options) {
-  std::vector<cricket::SenderOptions>& sender_options_list =
+void DetachSenderFromMediaSection(const std::string& mid,
+                                  const std::string& track_id,
+                                  MediaSessionOptions* session_options) {
+  std::vector<SenderOptions>& sender_options_list =
       FindFirstMediaDescriptionByMid(mid, session_options)->sender_options;
-  auto sender_it =
-      absl::c_find_if(sender_options_list,
-                      [track_id](const cricket::SenderOptions& sender_options) {
-                        return sender_options.track_id == track_id;
-                      });
+  auto sender_it = absl::c_find_if(
+      sender_options_list, [track_id](const SenderOptions& sender_options) {
+        return sender_options.track_id == track_id;
+      });
   RTC_DCHECK(sender_it != sender_options_list.end());
   sender_options_list.erase(sender_it);
 }
 
-// Helper function used to create a default MediaSessionOptions for Plan B SDP.
-// (https://tools.ietf.org/html/draft-uberti-rtcweb-plan-00).
-static MediaSessionOptions CreatePlanBMediaSessionOptions() {
+// Helper function used to create recv-only audio MediaSessionOptions.
+MediaSessionOptions CreateAudioMediaSession() {
   MediaSessionOptions session_options;
   AddMediaDescriptionOptions(MEDIA_TYPE_AUDIO, "audio",
                              RtpTransceiverDirection::kRecvOnly, kActive,
@@ -433,7 +392,7 @@ static MediaSessionOptions CreatePlanBMediaSessionOptions() {
 void PreferGcmCryptoParameters(CryptoParamsVec* cryptos) {
   cryptos->erase(
       std::remove_if(cryptos->begin(), cryptos->end(),
-                     [](const cricket::CryptoParams& crypto) {
+                     [](const CryptoParams& crypto) {
                        return crypto.crypto_suite != kCsAeadAes256Gcm &&
                               crypto.crypto_suite != kCsAeadAes128Gcm;
                      }),
@@ -444,7 +403,7 @@ void PreferGcmCryptoParameters(CryptoParamsVec* cryptos) {
 // was designed for Plan B SDP, where only one audio "m=" section and one video
 // "m=" section could be generated, and ordering couldn't be controlled. Many of
 // these tests may be obsolete as a result, and should be refactored or removed.
-class MediaSessionDescriptionFactoryTest : public ::testing::Test {
+class MediaSessionDescriptionFactoryTest : public testing::Test {
  public:
   MediaSessionDescriptionFactoryTest()
       : tdf1_(field_trials),
@@ -548,9 +507,9 @@ class MediaSessionDescriptionFactoryTest : public ::testing::Test {
         EXPECT_EQ(current_audio_ufrag, ti_audio->description.ice_ufrag);
         EXPECT_EQ(current_audio_pwd, ti_audio->description.ice_pwd);
       } else {
-        EXPECT_EQ(static_cast<size_t>(cricket::ICE_UFRAG_LENGTH),
+        EXPECT_EQ(static_cast<size_t>(ICE_UFRAG_LENGTH),
                   ti_audio->description.ice_ufrag.size());
-        EXPECT_EQ(static_cast<size_t>(cricket::ICE_PWD_LENGTH),
+        EXPECT_EQ(static_cast<size_t>(ICE_PWD_LENGTH),
                   ti_audio->description.ice_pwd.size());
       }
       auto media_desc_options_it =
@@ -572,9 +531,9 @@ class MediaSessionDescriptionFactoryTest : public ::testing::Test {
           EXPECT_EQ(current_video_ufrag, ti_video->description.ice_ufrag);
           EXPECT_EQ(current_video_pwd, ti_video->description.ice_pwd);
         } else {
-          EXPECT_EQ(static_cast<size_t>(cricket::ICE_UFRAG_LENGTH),
+          EXPECT_EQ(static_cast<size_t>(ICE_UFRAG_LENGTH),
                     ti_video->description.ice_ufrag.size());
-          EXPECT_EQ(static_cast<size_t>(cricket::ICE_PWD_LENGTH),
+          EXPECT_EQ(static_cast<size_t>(ICE_PWD_LENGTH),
                     ti_video->description.ice_pwd.size());
         }
       }
@@ -593,9 +552,9 @@ class MediaSessionDescriptionFactoryTest : public ::testing::Test {
           EXPECT_EQ(current_data_ufrag, ti_data->description.ice_ufrag);
           EXPECT_EQ(current_data_pwd, ti_data->description.ice_pwd);
         } else {
-          EXPECT_EQ(static_cast<size_t>(cricket::ICE_UFRAG_LENGTH),
+          EXPECT_EQ(static_cast<size_t>(ICE_UFRAG_LENGTH),
                     ti_data->description.ice_ufrag.size());
-          EXPECT_EQ(static_cast<size_t>(cricket::ICE_PWD_LENGTH),
+          EXPECT_EQ(static_cast<size_t>(ICE_PWD_LENGTH),
                     ti_data->description.ice_pwd.size());
         }
       }
@@ -625,10 +584,10 @@ class MediaSessionDescriptionFactoryTest : public ::testing::Test {
           f1_.CreateAnswerOrError(ref_desc.get(), options, nullptr).MoveValue();
     }
     ASSERT_TRUE(desc);
-    const cricket::MediaContentDescription* audio_media_desc =
+    const MediaContentDescription* audio_media_desc =
         desc->GetContentDescriptionByName("audio");
     ASSERT_TRUE(audio_media_desc);
-    const cricket::MediaContentDescription* video_media_desc =
+    const MediaContentDescription* video_media_desc =
         desc->GetContentDescriptionByName("video");
     ASSERT_TRUE(video_media_desc);
     EXPECT_TRUE(CompareCryptoParams(audio_media_desc->cryptos(),
@@ -638,7 +597,7 @@ class MediaSessionDescriptionFactoryTest : public ::testing::Test {
 
     // Verify the selected crypto is one from the reference audio
     // media content.
-    const cricket::MediaContentDescription* ref_audio_media_desc =
+    const MediaContentDescription* ref_audio_media_desc =
         ref_desc->GetContentDescriptionByName("audio");
     bool found = false;
     for (size_t i = 0; i < ref_audio_media_desc->cryptos().size(); ++i) {
@@ -680,10 +639,10 @@ class MediaSessionDescriptionFactoryTest : public ::testing::Test {
     EXPECT_EQ(expected_direction_in_answer, vcd_answer->direction());
   }
 
-  bool VerifyNoCNCodecs(const cricket::ContentInfo* content) {
+  bool VerifyNoCNCodecs(const ContentInfo* content) {
     RTC_DCHECK(content);
     RTC_CHECK(content->media_description());
-    for (const cricket::Codec& codec : content->media_description()->codecs()) {
+    for (const Codec& codec : content->media_description()->codecs()) {
       if (codec.name == "CN") {
         return false;
       }
@@ -706,7 +665,7 @@ class MediaSessionDescriptionFactoryTest : public ::testing::Test {
         f1_.CreateOfferOrError(offer_opts, nullptr).MoveValue();
     ASSERT_TRUE(offer.get());
     if (gcm_offer && gcm_answer) {
-      for (cricket::ContentInfo& content : offer->contents()) {
+      for (ContentInfo& content : offer->contents()) {
         auto cryptos = content.media_description()->cryptos();
         PreferGcmCryptoParameters(&cryptos);
         content.media_description()->set_cryptos(cryptos);
@@ -720,8 +679,8 @@ class MediaSessionDescriptionFactoryTest : public ::testing::Test {
     ASSERT_TRUE(vc);
     EXPECT_EQ(MediaProtocolType::kRtp, ac->type);
     EXPECT_EQ(MediaProtocolType::kRtp, vc->type);
-    const AudioContentDescription* acd = ac->media_description()->as_audio();
-    const VideoContentDescription* vcd = vc->media_description()->as_video();
+    const MediaContentDescription* acd = ac->media_description();
+    const MediaContentDescription* vcd = vc->media_description();
     EXPECT_EQ(MEDIA_TYPE_AUDIO, acd->type());
     EXPECT_THAT(acd->codecs(), ElementsAreArray(kAudioCodecsAnswer));
     EXPECT_EQ(kAutoBandwidth, acd->bandwidth());  // negotiated auto bw
@@ -741,13 +700,13 @@ class MediaSessionDescriptionFactoryTest : public ::testing::Test {
     } else {
       ASSERT_CRYPTO(vcd, 1U, kDefaultSrtpCryptoSuite);
     }
-    EXPECT_EQ(cricket::kMediaProtocolSavpf, vcd->protocol());
+    EXPECT_EQ(kMediaProtocolSavpf, vcd->protocol());
   }
 
   void TestTransportSequenceNumberNegotiation(
-      const cricket::RtpHeaderExtensions& local,
-      const cricket::RtpHeaderExtensions& offered,
-      const cricket::RtpHeaderExtensions& expectedAnswer) {
+      const RtpHeaderExtensions& local,
+      const RtpHeaderExtensions& offered,
+      const RtpHeaderExtensions& expectedAnswer) {
     MediaSessionOptions opts;
     AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &opts);
     SetAudioVideoRtpHeaderExtensions(offered, offered, &opts);
@@ -767,8 +726,7 @@ class MediaSessionDescriptionFactoryTest : public ::testing::Test {
   }
 
   std::vector<webrtc::RtpHeaderExtensionCapability>
-  HeaderExtensionCapabilitiesFromRtpExtensions(
-      cricket::RtpHeaderExtensions extensions) {
+  HeaderExtensionCapabilitiesFromRtpExtensions(RtpHeaderExtensions extensions) {
     std::vector<webrtc::RtpHeaderExtensionCapability> capabilities;
     for (const auto& extension : extensions) {
       webrtc::RtpHeaderExtensionCapability capability(
@@ -779,8 +737,8 @@ class MediaSessionDescriptionFactoryTest : public ::testing::Test {
     return capabilities;
   }
 
-  void SetAudioVideoRtpHeaderExtensions(cricket::RtpHeaderExtensions audio_exts,
-                                        cricket::RtpHeaderExtensions video_exts,
+  void SetAudioVideoRtpHeaderExtensions(RtpHeaderExtensions audio_exts,
+                                        RtpHeaderExtensions video_exts,
                                         MediaSessionOptions* opts) {
     auto audio_caps = HeaderExtensionCapabilitiesFromRtpExtensions(audio_exts);
     auto video_caps = HeaderExtensionCapabilitiesFromRtpExtensions(video_exts);
@@ -812,22 +770,21 @@ class MediaSessionDescriptionFactoryTest : public ::testing::Test {
 TEST_F(MediaSessionDescriptionFactoryTest, TestCreateAudioOffer) {
   f1_.set_secure(SEC_ENABLED);
   std::unique_ptr<SessionDescription> offer =
-      f1_.CreateOfferOrError(CreatePlanBMediaSessionOptions(), nullptr)
-          .MoveValue();
+      f1_.CreateOfferOrError(CreateAudioMediaSession(), nullptr).MoveValue();
   ASSERT_TRUE(offer.get());
   const ContentInfo* ac = offer->GetContentByName("audio");
   const ContentInfo* vc = offer->GetContentByName("video");
   ASSERT_TRUE(ac);
   EXPECT_FALSE(vc);
   EXPECT_EQ(MediaProtocolType::kRtp, ac->type);
-  const AudioContentDescription* acd = ac->media_description()->as_audio();
+  const MediaContentDescription* acd = ac->media_description();
   EXPECT_EQ(MEDIA_TYPE_AUDIO, acd->type());
   EXPECT_EQ(f1_.audio_sendrecv_codecs(), acd->codecs());
   EXPECT_EQ(0U, acd->first_ssrc());             // no sender is attached.
   EXPECT_EQ(kAutoBandwidth, acd->bandwidth());  // default bandwidth (auto)
   EXPECT_TRUE(acd->rtcp_mux());                 // rtcp-mux defaults on
   ASSERT_CRYPTO(acd, kDefaultCryptoSuiteSize, kDefaultSrtpCryptoSuite);
-  EXPECT_EQ(cricket::kMediaProtocolSavpf, acd->protocol());
+  EXPECT_EQ(kMediaProtocolSavpf, acd->protocol());
 }
 
 // Create an offer with just Opus and RED.
@@ -843,7 +800,7 @@ TEST_F(MediaSessionDescriptionFactoryTest,
   EXPECT_EQ("opus", preferences[0].name);
   EXPECT_EQ("red", preferences[1].name);
 
-  auto opts = CreatePlanBMediaSessionOptions();
+  auto opts = CreateAudioMediaSession();
   opts.media_description_options.at(0).codec_preferences = preferences;
   std::unique_ptr<SessionDescription> offer =
       f1_.CreateOfferOrError(opts, nullptr).MoveValue();
@@ -853,7 +810,7 @@ TEST_F(MediaSessionDescriptionFactoryTest,
   ASSERT_TRUE(ac != NULL);
   ASSERT_TRUE(vc == NULL);
   EXPECT_EQ(MediaProtocolType::kRtp, ac->type);
-  const AudioContentDescription* acd = ac->media_description()->as_audio();
+  const MediaContentDescription* acd = ac->media_description();
   EXPECT_EQ(MEDIA_TYPE_AUDIO, acd->type());
   EXPECT_EQ(2U, acd->codecs().size());
   EXPECT_EQ("opus", acd->codecs()[0].name);
@@ -872,7 +829,7 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCreateAudioOfferWithRedForOpus) {
   EXPECT_EQ("red", preferences[0].name);
   EXPECT_EQ("opus", preferences[1].name);
 
-  auto opts = CreatePlanBMediaSessionOptions();
+  auto opts = CreateAudioMediaSession();
   opts.media_description_options.at(0).codec_preferences = preferences;
   std::unique_ptr<SessionDescription> offer =
       f1_.CreateOfferOrError(opts, nullptr).MoveValue();
@@ -882,7 +839,7 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCreateAudioOfferWithRedForOpus) {
   ASSERT_TRUE(ac != NULL);
   ASSERT_TRUE(vc == NULL);
   EXPECT_EQ(MediaProtocolType::kRtp, ac->type);
-  const AudioContentDescription* acd = ac->media_description()->as_audio();
+  const MediaContentDescription* acd = ac->media_description();
   EXPECT_EQ(MEDIA_TYPE_AUDIO, acd->type());
   EXPECT_EQ(2U, acd->codecs().size());
   EXPECT_EQ("red", acd->codecs()[0].name);
@@ -903,22 +860,22 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCreateVideoOffer) {
   ASSERT_TRUE(vc);
   EXPECT_EQ(MediaProtocolType::kRtp, ac->type);
   EXPECT_EQ(MediaProtocolType::kRtp, vc->type);
-  const AudioContentDescription* acd = ac->media_description()->as_audio();
-  const VideoContentDescription* vcd = vc->media_description()->as_video();
+  const MediaContentDescription* acd = ac->media_description();
+  const MediaContentDescription* vcd = vc->media_description();
   EXPECT_EQ(MEDIA_TYPE_AUDIO, acd->type());
   EXPECT_EQ(f1_.audio_sendrecv_codecs(), acd->codecs());
   EXPECT_EQ(0U, acd->first_ssrc());             // no sender is attached
   EXPECT_EQ(kAutoBandwidth, acd->bandwidth());  // default bandwidth (auto)
   EXPECT_TRUE(acd->rtcp_mux());                 // rtcp-mux defaults on
   ASSERT_CRYPTO(acd, kDefaultCryptoSuiteSize, kDefaultSrtpCryptoSuite);
-  EXPECT_EQ(cricket::kMediaProtocolSavpf, acd->protocol());
+  EXPECT_EQ(kMediaProtocolSavpf, acd->protocol());
   EXPECT_EQ(MEDIA_TYPE_VIDEO, vcd->type());
   EXPECT_EQ(f1_.video_sendrecv_codecs(), vcd->codecs());
   EXPECT_EQ(0U, vcd->first_ssrc());             // no sender is attached
   EXPECT_EQ(kAutoBandwidth, vcd->bandwidth());  // default bandwidth (auto)
   EXPECT_TRUE(vcd->rtcp_mux());                 // rtcp-mux defaults on
   ASSERT_CRYPTO(vcd, kDefaultCryptoSuiteSize, kDefaultSrtpCryptoSuite);
-  EXPECT_EQ(cricket::kMediaProtocolSavpf, vcd->protocol());
+  EXPECT_EQ(kMediaProtocolSavpf, vcd->protocol());
 }
 
 // Test creating an offer with bundle where the Codecs have the same dynamic
@@ -978,9 +935,9 @@ TEST_F(MediaSessionDescriptionFactoryTest,
   EXPECT_TRUE(acd);
 
   ASSERT_CRYPTO(acd, 1U, kDefaultSrtpCryptoSuite);
-  EXPECT_EQ(cricket::kMediaProtocolSavpf, acd->protocol());
+  EXPECT_EQ(kMediaProtocolSavpf, acd->protocol());
   ASSERT_CRYPTO(vcd, 1U, kDefaultSrtpCryptoSuite);
-  EXPECT_EQ(cricket::kMediaProtocolSavpf, vcd->protocol());
+  EXPECT_EQ(kMediaProtocolSavpf, vcd->protocol());
 }
 
 // Create an SCTP data offer with bundle without error.
@@ -996,7 +953,7 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCreateSctpDataOffer) {
   auto dcd = GetFirstSctpDataContentDescription(offer.get());
   ASSERT_TRUE(dcd);
   // Since this transport is insecure, the protocol should be "SCTP".
-  EXPECT_EQ(cricket::kMediaProtocolSctp, dcd->protocol());
+  EXPECT_EQ(kMediaProtocolSctp, dcd->protocol());
 }
 
 // Create an SCTP data offer with bundle without error.
@@ -1013,7 +970,7 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCreateSecureSctpDataOffer) {
   auto dcd = GetFirstSctpDataContentDescription(offer.get());
   ASSERT_TRUE(dcd);
   // The protocol should now be "UDP/DTLS/SCTP"
-  EXPECT_EQ(cricket::kMediaProtocolUdpDtlsSctp, dcd->protocol());
+  EXPECT_EQ(kMediaProtocolUdpDtlsSctp, dcd->protocol());
 }
 
 // Test creating an sctp data channel from an already generated offer.
@@ -1027,13 +984,13 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCreateImplicitSctpDataOffer) {
   ASSERT_TRUE(offer1.get());
   const ContentInfo* data = offer1->GetContentByName("data");
   ASSERT_TRUE(data);
-  ASSERT_EQ(cricket::kMediaProtocolSctp, data->media_description()->protocol());
+  ASSERT_EQ(kMediaProtocolSctp, data->media_description()->protocol());
 
   std::unique_ptr<SessionDescription> offer2(
       f1_.CreateOfferOrError(opts, offer1.get()).MoveValue());
   data = offer2->GetContentByName("data");
   ASSERT_TRUE(data);
-  EXPECT_EQ(cricket::kMediaProtocolSctp, data->media_description()->protocol());
+  EXPECT_EQ(kMediaProtocolSctp, data->media_description()->protocol());
 }
 
 // Test that if BUNDLE is enabled and all media sections are rejected then the
@@ -1051,7 +1008,7 @@ TEST_F(MediaSessionDescriptionFactoryTest, ReOfferNoBundleGroupIfAllRejected) {
   std::unique_ptr<SessionDescription> reoffer =
       f1_.CreateOfferOrError(opts, offer.get()).MoveValue();
 
-  EXPECT_FALSE(reoffer->GetGroupByName(cricket::GROUP_TYPE_BUNDLE));
+  EXPECT_FALSE(reoffer->GetGroupByName(GROUP_TYPE_BUNDLE));
 }
 
 // Test that if BUNDLE is enabled and the remote re-offer does not include a
@@ -1074,7 +1031,7 @@ TEST_F(MediaSessionDescriptionFactoryTest, ReAnswerNoBundleGroupIfAllRejected) {
   std::unique_ptr<SessionDescription> reanswer =
       f2_.CreateAnswerOrError(reoffer.get(), opts, answer.get()).MoveValue();
 
-  EXPECT_FALSE(reanswer->GetGroupByName(cricket::GROUP_TYPE_BUNDLE));
+  EXPECT_FALSE(reanswer->GetGroupByName(GROUP_TYPE_BUNDLE));
 }
 
 // Test that if BUNDLE is enabled and the previous offerer-tagged media section
@@ -1097,8 +1054,7 @@ TEST_F(MediaSessionDescriptionFactoryTest, ReOfferChangeBundleOffererTagged) {
   std::unique_ptr<SessionDescription> reoffer =
       f1_.CreateOfferOrError(opts, offer.get()).MoveValue();
 
-  const cricket::ContentGroup* bundle_group =
-      reoffer->GetGroupByName(cricket::GROUP_TYPE_BUNDLE);
+  const ContentGroup* bundle_group = reoffer->GetGroupByName(GROUP_TYPE_BUNDLE);
   ASSERT_TRUE(bundle_group);
   EXPECT_FALSE(bundle_group->HasContentName("audio"));
   EXPECT_TRUE(bundle_group->HasContentName("video"));
@@ -1128,8 +1084,8 @@ TEST_F(MediaSessionDescriptionFactoryTest, ReAnswerChangedBundleOffererTagged) {
   std::unique_ptr<SessionDescription> reanswer =
       f2_.CreateAnswerOrError(reoffer.get(), opts, answer.get()).MoveValue();
 
-  const cricket::ContentGroup* bundle_group =
-      reanswer->GetGroupByName(cricket::GROUP_TYPE_BUNDLE);
+  const ContentGroup* bundle_group =
+      reanswer->GetGroupByName(GROUP_TYPE_BUNDLE);
   ASSERT_TRUE(bundle_group);
   EXPECT_FALSE(bundle_group->HasContentName("audio"));
   EXPECT_TRUE(bundle_group->HasContentName("video"));
@@ -1159,10 +1115,10 @@ TEST_F(MediaSessionDescriptionFactoryTest,
   // Munge the offer to have two groups. Offers like these cannot be generated
   // without munging, but it is valid to receive such offers from remote
   // endpoints.
-  cricket::ContentGroup bundle_group1(cricket::GROUP_TYPE_BUNDLE);
+  ContentGroup bundle_group1(GROUP_TYPE_BUNDLE);
   bundle_group1.AddContentName("1");
   bundle_group1.AddContentName("2");
-  cricket::ContentGroup bundle_group2(cricket::GROUP_TYPE_BUNDLE);
+  ContentGroup bundle_group2(GROUP_TYPE_BUNDLE);
   bundle_group2.AddContentName("3");
   bundle_group2.AddContentName("4");
   offer->AddGroup(bundle_group1);
@@ -1174,8 +1130,8 @@ TEST_F(MediaSessionDescriptionFactoryTest,
   std::unique_ptr<SessionDescription> answer =
       f2_.CreateAnswerOrError(offer.get(), opts, nullptr).MoveValue();
 
-  std::vector<const cricket::ContentGroup*> answer_groups =
-      answer->GetGroupsByName(cricket::GROUP_TYPE_BUNDLE);
+  std::vector<const ContentGroup*> answer_groups =
+      answer->GetGroupsByName(GROUP_TYPE_BUNDLE);
   ASSERT_EQ(answer_groups.size(), 2u);
   EXPECT_EQ(answer_groups[0]->content_names().size(), 2u);
   EXPECT_TRUE(answer_groups[0]->HasContentName("1"));
@@ -1189,7 +1145,7 @@ TEST_F(MediaSessionDescriptionFactoryTest,
   opts.bundle_enabled = false;
   answer = f2_.CreateAnswerOrError(offer.get(), opts, nullptr).MoveValue();
 
-  answer_groups = answer->GetGroupsByName(cricket::GROUP_TYPE_BUNDLE);
+  answer_groups = answer->GetGroupsByName(GROUP_TYPE_BUNDLE);
   // Rejected groups are still listed, but they are empty.
   ASSERT_EQ(answer_groups.size(), 2u);
   EXPECT_TRUE(answer_groups[0]->content_names().empty());
@@ -1268,8 +1224,8 @@ TEST_F(MediaSessionDescriptionFactoryTest,
   const ContentInfo* vc = offer->GetContentByName("video");
   ASSERT_TRUE(ac);
   ASSERT_TRUE(vc);
-  const AudioContentDescription* acd = ac->media_description()->as_audio();
-  const VideoContentDescription* vcd = vc->media_description()->as_video();
+  const MediaContentDescription* acd = ac->media_description();
+  const MediaContentDescription* vcd = vc->media_description();
 
   EXPECT_FALSE(vcd->has_ssrcs());  // No StreamParams.
   EXPECT_FALSE(acd->has_ssrcs());  // No StreamParams.
@@ -1336,26 +1292,24 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCreateAudioAnswer) {
   f1_.set_secure(SEC_ENABLED);
   f2_.set_secure(SEC_ENABLED);
   std::unique_ptr<SessionDescription> offer =
-      f1_.CreateOfferOrError(CreatePlanBMediaSessionOptions(), nullptr)
-          .MoveValue();
+      f1_.CreateOfferOrError(CreateAudioMediaSession(), nullptr).MoveValue();
   ASSERT_TRUE(offer.get());
   std::unique_ptr<SessionDescription> answer =
-      f2_.CreateAnswerOrError(offer.get(), CreatePlanBMediaSessionOptions(),
-                              nullptr)
+      f2_.CreateAnswerOrError(offer.get(), CreateAudioMediaSession(), nullptr)
           .MoveValue();
   const ContentInfo* ac = answer->GetContentByName("audio");
   const ContentInfo* vc = answer->GetContentByName("video");
   ASSERT_TRUE(ac);
   EXPECT_FALSE(vc);
   EXPECT_EQ(MediaProtocolType::kRtp, ac->type);
-  const AudioContentDescription* acd = ac->media_description()->as_audio();
+  const MediaContentDescription* acd = ac->media_description();
   EXPECT_EQ(MEDIA_TYPE_AUDIO, acd->type());
   EXPECT_THAT(acd->codecs(), ElementsAreArray(kAudioCodecsAnswer));
   EXPECT_EQ(0U, acd->first_ssrc());             // no sender is attached
   EXPECT_EQ(kAutoBandwidth, acd->bandwidth());  // negotiated auto bw
   EXPECT_TRUE(acd->rtcp_mux());                 // negotiated rtcp-mux
   ASSERT_CRYPTO(acd, 1U, kDefaultSrtpCryptoSuite);
-  EXPECT_EQ(cricket::kMediaProtocolSavpf, acd->protocol());
+  EXPECT_EQ(kMediaProtocolSavpf, acd->protocol());
 }
 
 // Create a typical audio answer with GCM ciphers enabled, and ensure it
@@ -1363,11 +1317,11 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCreateAudioAnswer) {
 TEST_F(MediaSessionDescriptionFactoryTest, TestCreateAudioAnswerGcm) {
   f1_.set_secure(SEC_ENABLED);
   f2_.set_secure(SEC_ENABLED);
-  MediaSessionOptions opts = CreatePlanBMediaSessionOptions();
+  MediaSessionOptions opts = CreateAudioMediaSession();
   std::unique_ptr<SessionDescription> offer =
       f1_.CreateOfferOrError(opts, nullptr).MoveValue();
   ASSERT_TRUE(offer.get());
-  for (cricket::ContentInfo& content : offer->contents()) {
+  for (ContentInfo& content : offer->contents()) {
     auto cryptos = content.media_description()->cryptos();
     PreferGcmCryptoParameters(&cryptos);
     content.media_description()->set_cryptos(cryptos);
@@ -1379,14 +1333,14 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCreateAudioAnswerGcm) {
   ASSERT_TRUE(ac);
   EXPECT_FALSE(vc);
   EXPECT_EQ(MediaProtocolType::kRtp, ac->type);
-  const AudioContentDescription* acd = ac->media_description()->as_audio();
+  const MediaContentDescription* acd = ac->media_description();
   EXPECT_EQ(MEDIA_TYPE_AUDIO, acd->type());
   EXPECT_THAT(acd->codecs(), ElementsAreArray(kAudioCodecsAnswer));
   EXPECT_EQ(0U, acd->first_ssrc());             // no sender is attached
   EXPECT_EQ(kAutoBandwidth, acd->bandwidth());  // negotiated auto bw
   EXPECT_TRUE(acd->rtcp_mux());                 // negotiated rtcp-mux
   ASSERT_CRYPTO(acd, 1U, kDefaultSrtpCryptoSuiteGcm);
-  EXPECT_EQ(cricket::kMediaProtocolSavpf, acd->protocol());
+  EXPECT_EQ(kMediaProtocolSavpf, acd->protocol());
 }
 
 // Create an audio answer with no common codecs, and ensure it is rejected.
@@ -1396,10 +1350,10 @@ TEST_F(MediaSessionDescriptionFactoryTest,
   AddMediaDescriptionOptions(MEDIA_TYPE_AUDIO, "audio",
                              RtpTransceiverDirection::kSendRecv, kActive,
                              &opts);
-  std::vector f1_codecs = {cricket::CreateAudioCodec(96, "opus", 48000, 1)};
+  std::vector f1_codecs = {CreateAudioCodec(96, "opus", 48000, 1)};
   f1_.set_audio_codecs(f1_codecs, f1_codecs);
 
-  std::vector f2_codecs = {cricket::CreateAudioCodec(0, "PCMU", 8000, 1)};
+  std::vector f2_codecs = {CreateAudioCodec(0, "PCMU", 8000, 1)};
   f2_.set_audio_codecs(f2_codecs, f2_codecs);
 
   std::unique_ptr<SessionDescription> offer =
@@ -1428,8 +1382,8 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCreateVideoAnswer) {
   ASSERT_TRUE(vc);
   EXPECT_EQ(MediaProtocolType::kRtp, ac->type);
   EXPECT_EQ(MediaProtocolType::kRtp, vc->type);
-  const AudioContentDescription* acd = ac->media_description()->as_audio();
-  const VideoContentDescription* vcd = vc->media_description()->as_video();
+  const MediaContentDescription* acd = ac->media_description();
+  const MediaContentDescription* vcd = vc->media_description();
   EXPECT_EQ(MEDIA_TYPE_AUDIO, acd->type());
   EXPECT_THAT(acd->codecs(), ElementsAreArray(kAudioCodecsAnswer));
   EXPECT_EQ(kAutoBandwidth, acd->bandwidth());  // negotiated auto bw
@@ -1441,7 +1395,7 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCreateVideoAnswer) {
   EXPECT_EQ(0U, vcd->first_ssrc());  // no sender is attached
   EXPECT_TRUE(vcd->rtcp_mux());      // negotiated rtcp-mux
   ASSERT_CRYPTO(vcd, 1U, kDefaultSrtpCryptoSuite);
-  EXPECT_EQ(cricket::kMediaProtocolSavpf, vcd->protocol());
+  EXPECT_EQ(kMediaProtocolSavpf, vcd->protocol());
 }
 
 // Create a typical video answer with GCM ciphers enabled, and ensure it
@@ -1469,10 +1423,10 @@ TEST_F(MediaSessionDescriptionFactoryTest,
   AddMediaDescriptionOptions(MEDIA_TYPE_VIDEO, "video",
                              RtpTransceiverDirection::kSendRecv, kActive,
                              &opts);
-  std::vector f1_codecs = {cricket::CreateVideoCodec(96, "H264")};
+  std::vector f1_codecs = {CreateVideoCodec(96, "H264")};
   f1_.set_video_codecs(f1_codecs, f1_codecs);
 
-  std::vector f2_codecs = {cricket::CreateVideoCodec(97, "VP8")};
+  std::vector f2_codecs = {CreateVideoCodec(97, "VP8")};
   f2_.set_video_codecs(f2_codecs, f2_codecs);
 
   std::unique_ptr<SessionDescription> offer =
@@ -1492,12 +1446,12 @@ TEST_F(MediaSessionDescriptionFactoryTest,
   AddMediaDescriptionOptions(MEDIA_TYPE_VIDEO, "video",
                              RtpTransceiverDirection::kSendRecv, kActive,
                              &opts);
-  std::vector f1_codecs = {cricket::CreateVideoCodec(96, "H264"),
-                           cricket::CreateVideoCodec(118, "flexfec-03")};
+  std::vector f1_codecs = {CreateVideoCodec(96, "H264"),
+                           CreateVideoCodec(118, "flexfec-03")};
   f1_.set_video_codecs(f1_codecs, f1_codecs);
 
-  std::vector f2_codecs = {cricket::CreateVideoCodec(97, "VP8"),
-                           cricket::CreateVideoCodec(118, "flexfec-03")};
+  std::vector f2_codecs = {CreateVideoCodec(97, "VP8"),
+                           CreateVideoCodec(118, "flexfec-03")};
   f2_.set_video_codecs(f2_codecs, f2_codecs);
 
   std::unique_ptr<SessionDescription> offer =
@@ -1648,7 +1602,7 @@ TEST_F(MediaSessionDescriptionFactoryTest,
   const SctpDataContentDescription* dcd_answer =
       dc_answer->media_description()->as_sctp();
   EXPECT_FALSE(dc_answer->rejected);
-  EXPECT_EQ(cricket::kSctpSendBufferSize, dcd_answer->max_message_size());
+  EXPECT_EQ(kSctpSendBufferSize, dcd_answer->max_message_size());
 }
 
 // Verifies that the order of the media contents in the offer is preserved in
@@ -1720,7 +1674,7 @@ TEST_F(MediaSessionDescriptionFactoryTest, CreateAnswerToInactiveOffer) {
 
 // Test that the media protocol is RTP/AVPF if DTLS and SDES are disabled.
 TEST_F(MediaSessionDescriptionFactoryTest, AudioOfferAnswerWithCryptoDisabled) {
-  MediaSessionOptions opts = CreatePlanBMediaSessionOptions();
+  MediaSessionOptions opts = CreateAudioMediaSession();
   f1_.set_secure(SEC_DISABLED);
   f2_.set_secure(SEC_DISABLED);
   tdf1_.set_secure(SEC_DISABLED);
@@ -1731,7 +1685,7 @@ TEST_F(MediaSessionDescriptionFactoryTest, AudioOfferAnswerWithCryptoDisabled) {
   const AudioContentDescription* offer_acd =
       GetFirstAudioContentDescription(offer.get());
   ASSERT_TRUE(offer_acd);
-  EXPECT_EQ(cricket::kMediaProtocolAvpf, offer_acd->protocol());
+  EXPECT_EQ(kMediaProtocolAvpf, offer_acd->protocol());
 
   std::unique_ptr<SessionDescription> answer =
       f2_.CreateAnswerOrError(offer.get(), opts, nullptr).MoveValue();
@@ -1743,7 +1697,7 @@ TEST_F(MediaSessionDescriptionFactoryTest, AudioOfferAnswerWithCryptoDisabled) {
   const AudioContentDescription* answer_acd =
       GetFirstAudioContentDescription(answer.get());
   ASSERT_TRUE(answer_acd);
-  EXPECT_EQ(cricket::kMediaProtocolAvpf, answer_acd->protocol());
+  EXPECT_EQ(kMediaProtocolAvpf, answer_acd->protocol());
 }
 
 // Create a video offer and answer and ensure the RTP header extensions
@@ -1886,9 +1840,9 @@ TEST_F(MediaSessionDescriptionFactoryTest,
   MediaSessionOptions opts;
   AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &opts);
 
-  const cricket::RtpHeaderExtensions offered_extensions = {
+  const RtpHeaderExtensions offered_extensions = {
       RtpExtension(RtpExtension::kAbsoluteCaptureTimeUri, 7)};
-  const cricket::RtpHeaderExtensions local_extensions = {
+  const RtpHeaderExtensions local_extensions = {
       RtpExtension(RtpExtension::kTransportSequenceNumberUri, 5)};
   SetAudioVideoRtpHeaderExtensions(offered_extensions, offered_extensions,
                                    &opts);
@@ -1910,9 +1864,9 @@ TEST_F(MediaSessionDescriptionFactoryTest,
   MediaSessionOptions opts;
   AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &opts);
 
-  const cricket::RtpHeaderExtensions offered_extensions = {
+  const RtpHeaderExtensions offered_extensions = {
       RtpExtension(RtpExtension::kAbsoluteCaptureTimeUri, 7)};
-  const cricket::RtpHeaderExtensions local_extensions = {
+  const RtpHeaderExtensions local_extensions = {
       RtpExtension(RtpExtension::kAbsoluteCaptureTimeUri, 5)};
   SetAudioVideoRtpHeaderExtensions(offered_extensions, offered_extensions,
                                    &opts);
@@ -1934,9 +1888,9 @@ TEST_F(MediaSessionDescriptionFactoryTest,
   MediaSessionOptions opts;
   AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &opts);
 
-  const cricket::RtpHeaderExtensions offered_extensions = {
+  const RtpHeaderExtensions offered_extensions = {
       RtpExtension(RtpExtension::kTransportSequenceNumberUri, 7)};
-  const cricket::RtpHeaderExtensions local_extensions = {
+  const RtpHeaderExtensions local_extensions = {
       RtpExtension(RtpExtension::kAbsoluteCaptureTimeUri, 5)};
   SetAudioVideoRtpHeaderExtensions(offered_extensions, offered_extensions,
                                    &opts);
@@ -2257,8 +2211,8 @@ TEST_F(MediaSessionDescriptionFactoryTest,
   const ContentInfo* vc = answer->GetContentByName("video");
   ASSERT_TRUE(ac);
   ASSERT_TRUE(vc);
-  const AudioContentDescription* acd = ac->media_description()->as_audio();
-  const VideoContentDescription* vcd = vc->media_description()->as_video();
+  const MediaContentDescription* acd = ac->media_description();
+  const MediaContentDescription* vcd = vc->media_description();
 
   EXPECT_FALSE(acd->has_ssrcs());  // No StreamParams.
   EXPECT_FALSE(vcd->has_ssrcs());  // No StreamParams.
@@ -2523,8 +2477,8 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCreateMultiStreamVideoOffer) {
   const ContentInfo* vc = offer->GetContentByName("video");
   ASSERT_TRUE(ac);
   ASSERT_TRUE(vc);
-  const AudioContentDescription* acd = ac->media_description()->as_audio();
-  const VideoContentDescription* vcd = vc->media_description()->as_video();
+  const MediaContentDescription* acd = ac->media_description();
+  const MediaContentDescription* vcd = vc->media_description();
   EXPECT_EQ(MEDIA_TYPE_AUDIO, acd->type());
   EXPECT_EQ(f1_.audio_sendrecv_codecs(), acd->codecs());
 
@@ -2568,10 +2522,8 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCreateMultiStreamVideoOffer) {
   vc = updated_offer->GetContentByName("video");
   ASSERT_TRUE(ac);
   ASSERT_TRUE(vc);
-  const AudioContentDescription* updated_acd =
-      ac->media_description()->as_audio();
-  const VideoContentDescription* updated_vcd =
-      vc->media_description()->as_video();
+  const MediaContentDescription* updated_acd = ac->media_description();
+  const MediaContentDescription* updated_vcd = vc->media_description();
 
   EXPECT_EQ(acd->type(), updated_acd->type());
   EXPECT_EQ(acd->codecs(), updated_acd->codecs());
@@ -2616,24 +2568,24 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCreateSimulcastVideoOffer) {
   ASSERT_TRUE(offer.get());
   const ContentInfo* vc = offer->GetContentByName("video");
   ASSERT_TRUE(vc);
-  const VideoContentDescription* vcd = vc->media_description()->as_video();
+  const MediaContentDescription* vcd = vc->media_description();
 
   const StreamParamsVec& video_streams = vcd->streams();
   ASSERT_EQ(1U, video_streams.size());
   EXPECT_EQ(kVideoTrack1, video_streams[0].id);
   const SsrcGroup* sim_ssrc_group =
-      video_streams[0].get_ssrc_group(cricket::kSimSsrcGroupSemantics);
+      video_streams[0].get_ssrc_group(kSimSsrcGroupSemantics);
   ASSERT_TRUE(sim_ssrc_group);
   EXPECT_EQ(static_cast<size_t>(num_sim_layers), sim_ssrc_group->ssrcs.size());
 }
 
 MATCHER(RidDescriptionEquals, "Verifies that two RidDescriptions are equal.") {
-  const RidDescription& rid1 = ::testing::get<0>(arg);
-  const RidDescription& rid2 = ::testing::get<1>(arg);
+  const RidDescription& rid1 = std::get<0>(arg);
+  const RidDescription& rid2 = std::get<1>(arg);
   return rid1.rid == rid2.rid && rid1.direction == rid2.direction;
 }
 
-static void CheckSimulcastInSessionDescription(
+void CheckSimulcastInSessionDescription(
     const SessionDescription* description,
     const std::string& content_name,
     const std::vector<RidDescription>& send_rids,
@@ -2828,8 +2780,8 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCreateMultiStreamVideoAnswer) {
   const ContentInfo* vc = answer->GetContentByName("video");
   ASSERT_TRUE(ac);
   ASSERT_TRUE(vc);
-  const AudioContentDescription* acd = ac->media_description()->as_audio();
-  const VideoContentDescription* vcd = vc->media_description()->as_video();
+  const MediaContentDescription* acd = ac->media_description();
+  const MediaContentDescription* vcd = vc->media_description();
   ASSERT_CRYPTO(acd, 1U, kDefaultSrtpCryptoSuite);
   ASSERT_CRYPTO(vcd, 1U, kDefaultSrtpCryptoSuite);
 
@@ -2873,10 +2825,8 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCreateMultiStreamVideoAnswer) {
   vc = updated_answer->GetContentByName("video");
   ASSERT_TRUE(ac);
   ASSERT_TRUE(vc);
-  const AudioContentDescription* updated_acd =
-      ac->media_description()->as_audio();
-  const VideoContentDescription* updated_vcd =
-      vc->media_description()->as_video();
+  const MediaContentDescription* updated_acd = ac->media_description();
+  const MediaContentDescription* updated_vcd = vc->media_description();
 
   ASSERT_CRYPTO(updated_acd, 1U, kDefaultSrtpCryptoSuite);
   EXPECT_TRUE(CompareCryptoParams(acd->cryptos(), updated_acd->cryptos()));
@@ -3082,14 +3032,12 @@ TEST_F(MediaSessionDescriptionFactoryTest,
                              &opts);
   std::vector<VideoCodec> f1_codecs = MAKE_VECTOR(kVideoCodecs1);
   // This creates rtx for H264 with the payload type `f1_` uses.
-  AddRtxCodec(cricket::CreateVideoRtxCodec(126, kVideoCodecs1[1].id),
-              &f1_codecs);
+  AddRtxCodec(CreateVideoRtxCodec(126, kVideoCodecs1[1].id), &f1_codecs);
   f1_.set_video_codecs(f1_codecs, f1_codecs);
 
   std::vector<VideoCodec> f2_codecs = MAKE_VECTOR(kVideoCodecs2);
   // This creates rtx for H264 with the payload type `f2_` uses.
-  AddRtxCodec(cricket::CreateVideoRtxCodec(125, kVideoCodecs2[0].id),
-              &f2_codecs);
+  AddRtxCodec(CreateVideoRtxCodec(125, kVideoCodecs2[0].id), &f2_codecs);
   f2_.set_video_codecs(f2_codecs, f2_codecs);
 
   std::unique_ptr<SessionDescription> offer =
@@ -3102,8 +3050,7 @@ TEST_F(MediaSessionDescriptionFactoryTest,
       GetFirstVideoContentDescription(answer.get());
 
   std::vector<VideoCodec> expected_codecs = MAKE_VECTOR(kVideoCodecsAnswer);
-  AddRtxCodec(cricket::CreateVideoRtxCodec(126, kVideoCodecs1[1].id),
-              &expected_codecs);
+  AddRtxCodec(CreateVideoRtxCodec(126, kVideoCodecs1[1].id), &expected_codecs);
 
   EXPECT_EQ(expected_codecs, vcd->codecs());
 
@@ -3136,14 +3083,12 @@ TEST_F(MediaSessionDescriptionFactoryTest,
                              &opts);
   // We specifically choose different preferred payload types for VP8 to
   // trigger the issue.
-  cricket::VideoCodec vp8_offerer = cricket::CreateVideoCodec(100, "VP8");
-  cricket::VideoCodec vp8_offerer_rtx =
-      cricket::CreateVideoRtxCodec(101, vp8_offerer.id);
-  cricket::VideoCodec vp8_answerer = cricket::CreateVideoCodec(110, "VP8");
-  cricket::VideoCodec vp8_answerer_rtx =
-      cricket::CreateVideoRtxCodec(111, vp8_answerer.id);
-  cricket::VideoCodec vp9 = cricket::CreateVideoCodec(120, "VP9");
-  cricket::VideoCodec vp9_rtx = cricket::CreateVideoRtxCodec(121, vp9.id);
+  VideoCodec vp8_offerer = CreateVideoCodec(100, "VP8");
+  VideoCodec vp8_offerer_rtx = CreateVideoRtxCodec(101, vp8_offerer.id);
+  VideoCodec vp8_answerer = CreateVideoCodec(110, "VP8");
+  VideoCodec vp8_answerer_rtx = CreateVideoRtxCodec(111, vp8_answerer.id);
+  VideoCodec vp9 = CreateVideoCodec(120, "VP9");
+  VideoCodec vp9_rtx = CreateVideoRtxCodec(121, vp9.id);
 
   std::vector<VideoCodec> f1_codecs = {vp8_offerer, vp8_offerer_rtx};
   // We also specifically cause the answerer to prefer VP9, such that if it
@@ -3171,7 +3116,7 @@ TEST_F(MediaSessionDescriptionFactoryTest,
 
   const VideoContentDescription* vcd =
       GetFirstVideoContentDescription(updated_offer.get());
-  std::vector<cricket::VideoCodec> codecs = vcd->codecs();
+  std::vector<VideoCodec> codecs = vcd->codecs();
   ASSERT_EQ(4u, codecs.size());
   EXPECT_EQ(vp8_offerer, codecs[0]);
   EXPECT_EQ(vp8_offerer_rtx, codecs[1]);
@@ -3187,8 +3132,7 @@ TEST_F(MediaSessionDescriptionFactoryTest,
        RespondentCreatesOfferWithVideoAndRtxAfterCreatingAudioAnswer) {
   std::vector<VideoCodec> f1_codecs = MAKE_VECTOR(kVideoCodecs1);
   // This creates rtx for H264 with the payload type `f1_` uses.
-  AddRtxCodec(cricket::CreateVideoRtxCodec(126, kVideoCodecs1[1].id),
-              &f1_codecs);
+  AddRtxCodec(CreateVideoRtxCodec(126, kVideoCodecs1[1].id), &f1_codecs);
   f1_.set_video_codecs(f1_codecs, f1_codecs);
 
   MediaSessionOptions opts;
@@ -3214,7 +3158,7 @@ TEST_F(MediaSessionDescriptionFactoryTest,
   std::vector<VideoCodec> f2_codecs = MAKE_VECTOR(kVideoCodecs2);
   int used_pl_type = acd->codecs()[0].id;
   f2_codecs[0].id = used_pl_type;  // Set the payload type for H264.
-  AddRtxCodec(cricket::CreateVideoRtxCodec(125, used_pl_type), &f2_codecs);
+  AddRtxCodec(CreateVideoRtxCodec(125, used_pl_type), &f2_codecs);
   f2_.set_video_codecs(f2_codecs, f2_codecs);
 
   std::unique_ptr<SessionDescription> updated_offer(
@@ -3232,12 +3176,12 @@ TEST_F(MediaSessionDescriptionFactoryTest,
       GetFirstVideoContentDescription(updated_answer.get());
 
   ASSERT_EQ("H264", updated_vcd->codecs()[0].name);
-  ASSERT_EQ(cricket::kRtxCodecName, updated_vcd->codecs()[1].name);
+  ASSERT_EQ(kRtxCodecName, updated_vcd->codecs()[1].name);
   int new_h264_pl_type = updated_vcd->codecs()[0].id;
   EXPECT_NE(used_pl_type, new_h264_pl_type);
   VideoCodec rtx = updated_vcd->codecs()[1];
-  int pt_referenced_by_rtx = rtc::FromString<int>(
-      rtx.params[cricket::kCodecParamAssociatedPayloadType]);
+  int pt_referenced_by_rtx =
+      rtc::FromString<int>(rtx.params[kCodecParamAssociatedPayloadType]);
   EXPECT_EQ(new_h264_pl_type, pt_referenced_by_rtx);
 }
 
@@ -3251,8 +3195,7 @@ TEST_F(MediaSessionDescriptionFactoryTest,
 
   std::vector<VideoCodec> f2_codecs = MAKE_VECTOR(kVideoCodecs2);
   // This creates rtx for H264 with the payload type `f2_` uses.
-  AddRtxCodec(cricket::CreateVideoRtxCodec(125, kVideoCodecs2[0].id),
-              &f2_codecs);
+  AddRtxCodec(CreateVideoRtxCodec(125, kVideoCodecs2[0].id), &f2_codecs);
   f2_.set_video_codecs(f2_codecs, f2_codecs);
 
   std::unique_ptr<SessionDescription> offer =
@@ -3279,8 +3222,7 @@ TEST_F(MediaSessionDescriptionFactoryTest,
 
   // New offer should attempt to add H263, and RTX for H264.
   expected_codecs.push_back(kVideoCodecs2[1]);
-  AddRtxCodec(cricket::CreateVideoRtxCodec(125, kVideoCodecs1[1].id),
-              &expected_codecs);
+  AddRtxCodec(CreateVideoRtxCodec(125, kVideoCodecs1[1].id), &expected_codecs);
   EXPECT_EQ(expected_codecs, updated_vcd->codecs());
 }
 
@@ -3292,14 +3234,12 @@ TEST_F(MediaSessionDescriptionFactoryTest, RtxWithoutApt) {
                              &opts);
   std::vector<VideoCodec> f1_codecs = MAKE_VECTOR(kVideoCodecs1);
   // This creates RTX without associated payload type parameter.
-  AddRtxCodec(cricket::CreateVideoCodec(126, cricket::kRtxCodecName),
-              &f1_codecs);
+  AddRtxCodec(CreateVideoCodec(126, kRtxCodecName), &f1_codecs);
   f1_.set_video_codecs(f1_codecs, f1_codecs);
 
   std::vector<VideoCodec> f2_codecs = MAKE_VECTOR(kVideoCodecs2);
   // This creates RTX for H264 with the payload type `f2_` uses.
-  AddRtxCodec(cricket::CreateVideoRtxCodec(125, kVideoCodecs2[0].id),
-              &f2_codecs);
+  AddRtxCodec(CreateVideoRtxCodec(125, kVideoCodecs2[0].id), &f2_codecs);
   f2_.set_video_codecs(f2_codecs, f2_codecs);
 
   std::unique_ptr<SessionDescription> offer =
@@ -3310,23 +3250,22 @@ TEST_F(MediaSessionDescriptionFactoryTest, RtxWithoutApt) {
   // is possible to test that that RTX is dropped when
   // kCodecParamAssociatedPayloadType is missing in the offer.
   MediaContentDescription* media_desc =
-      offer->GetContentDescriptionByName(cricket::CN_VIDEO);
+      offer->GetContentDescriptionByName(CN_VIDEO);
   ASSERT_TRUE(media_desc);
-  VideoContentDescription* desc = media_desc->as_video();
-  std::vector<VideoCodec> codecs = desc->codecs();
-  for (VideoCodec& codec : codecs) {
-    if (absl::StartsWith(codec.name, cricket::kRtxCodecName)) {
+  std::vector<Codec> codecs = media_desc->codecs();
+  for (Codec& codec : codecs) {
+    if (absl::StartsWith(codec.name, kRtxCodecName)) {
       codec.params.clear();
     }
   }
-  desc->set_codecs(codecs);
+  media_desc->set_codecs(codecs);
 
   std::unique_ptr<SessionDescription> answer =
       f2_.CreateAnswerOrError(offer.get(), opts, nullptr).MoveValue();
 
   EXPECT_THAT(
       GetCodecNames(GetFirstVideoContentDescription(answer.get())->codecs()),
-      Not(Contains(cricket::kRtxCodecName)));
+      Not(Contains(kRtxCodecName)));
 }
 
 // Test that RTX will be filtered out in the answer if its associated payload
@@ -3338,14 +3277,12 @@ TEST_F(MediaSessionDescriptionFactoryTest, FilterOutRtxIfAptDoesntMatch) {
                              &opts);
   std::vector<VideoCodec> f1_codecs = MAKE_VECTOR(kVideoCodecs1);
   // This creates RTX for H264 in sender.
-  AddRtxCodec(cricket::CreateVideoRtxCodec(126, kVideoCodecs1[1].id),
-              &f1_codecs);
+  AddRtxCodec(CreateVideoRtxCodec(126, kVideoCodecs1[1].id), &f1_codecs);
   f1_.set_video_codecs(f1_codecs, f1_codecs);
 
   std::vector<VideoCodec> f2_codecs = MAKE_VECTOR(kVideoCodecs2);
   // This creates RTX for H263 in receiver.
-  AddRtxCodec(cricket::CreateVideoRtxCodec(125, kVideoCodecs2[1].id),
-              &f2_codecs);
+  AddRtxCodec(CreateVideoRtxCodec(125, kVideoCodecs2[1].id), &f2_codecs);
   f2_.set_video_codecs(f2_codecs, f2_codecs);
 
   std::unique_ptr<SessionDescription> offer =
@@ -3358,7 +3295,7 @@ TEST_F(MediaSessionDescriptionFactoryTest, FilterOutRtxIfAptDoesntMatch) {
 
   EXPECT_THAT(
       GetCodecNames(GetFirstVideoContentDescription(answer.get())->codecs()),
-      Not(Contains(cricket::kRtxCodecName)));
+      Not(Contains(kRtxCodecName)));
 }
 
 // Test that when multiple RTX codecs are offered, only the matched RTX codec
@@ -3371,19 +3308,16 @@ TEST_F(MediaSessionDescriptionFactoryTest,
                              &opts);
   std::vector<VideoCodec> f1_codecs = MAKE_VECTOR(kVideoCodecs1);
   // This creates RTX for H264-SVC in sender.
-  AddRtxCodec(cricket::CreateVideoRtxCodec(125, kVideoCodecs1[0].id),
-              &f1_codecs);
+  AddRtxCodec(CreateVideoRtxCodec(125, kVideoCodecs1[0].id), &f1_codecs);
   f1_.set_video_codecs(f1_codecs, f1_codecs);
 
   // This creates RTX for H264 in sender.
-  AddRtxCodec(cricket::CreateVideoRtxCodec(126, kVideoCodecs1[1].id),
-              &f1_codecs);
+  AddRtxCodec(CreateVideoRtxCodec(126, kVideoCodecs1[1].id), &f1_codecs);
   f1_.set_video_codecs(f1_codecs, f1_codecs);
 
   std::vector<VideoCodec> f2_codecs = MAKE_VECTOR(kVideoCodecs2);
   // This creates RTX for H264 in receiver.
-  AddRtxCodec(cricket::CreateVideoRtxCodec(124, kVideoCodecs2[0].id),
-              &f2_codecs);
+  AddRtxCodec(CreateVideoRtxCodec(124, kVideoCodecs2[0].id), &f2_codecs);
   f2_.set_video_codecs(f2_codecs, f1_codecs);
 
   // H264-SVC codec is removed in the answer, therefore, associated RTX codec
@@ -3396,8 +3330,7 @@ TEST_F(MediaSessionDescriptionFactoryTest,
   const VideoContentDescription* vcd =
       GetFirstVideoContentDescription(answer.get());
   std::vector<VideoCodec> expected_codecs = MAKE_VECTOR(kVideoCodecsAnswer);
-  AddRtxCodec(cricket::CreateVideoRtxCodec(126, kVideoCodecs1[1].id),
-              &expected_codecs);
+  AddRtxCodec(CreateVideoRtxCodec(126, kVideoCodecs1[1].id), &expected_codecs);
 
   EXPECT_EQ(expected_codecs, vcd->codecs());
 }
@@ -3411,8 +3344,7 @@ TEST_F(MediaSessionDescriptionFactoryTest, AddSecondRtxInNewOffer) {
                              &opts);
   std::vector<VideoCodec> f1_codecs = MAKE_VECTOR(kVideoCodecs1);
   // This creates RTX for H264 for the offerer.
-  AddRtxCodec(cricket::CreateVideoRtxCodec(126, kVideoCodecs1[1].id),
-              &f1_codecs);
+  AddRtxCodec(CreateVideoRtxCodec(126, kVideoCodecs1[1].id), &f1_codecs);
   f1_.set_video_codecs(f1_codecs, f1_codecs);
 
   std::unique_ptr<SessionDescription> offer =
@@ -3422,13 +3354,11 @@ TEST_F(MediaSessionDescriptionFactoryTest, AddSecondRtxInNewOffer) {
       GetFirstVideoContentDescription(offer.get());
 
   std::vector<VideoCodec> expected_codecs = MAKE_VECTOR(kVideoCodecs1);
-  AddRtxCodec(cricket::CreateVideoRtxCodec(126, kVideoCodecs1[1].id),
-              &expected_codecs);
+  AddRtxCodec(CreateVideoRtxCodec(126, kVideoCodecs1[1].id), &expected_codecs);
   EXPECT_EQ(expected_codecs, vcd->codecs());
 
   // Now, attempt to add RTX for H264-SVC.
-  AddRtxCodec(cricket::CreateVideoRtxCodec(125, kVideoCodecs1[0].id),
-              &f1_codecs);
+  AddRtxCodec(CreateVideoRtxCodec(125, kVideoCodecs1[0].id), &f1_codecs);
   f1_.set_video_codecs(f1_codecs, f1_codecs);
 
   std::unique_ptr<SessionDescription> updated_offer(
@@ -3436,8 +3366,7 @@ TEST_F(MediaSessionDescriptionFactoryTest, AddSecondRtxInNewOffer) {
   ASSERT_TRUE(updated_offer);
   vcd = GetFirstVideoContentDescription(updated_offer.get());
 
-  AddRtxCodec(cricket::CreateVideoRtxCodec(125, kVideoCodecs1[0].id),
-              &expected_codecs);
+  AddRtxCodec(CreateVideoRtxCodec(125, kVideoCodecs1[0].id), &expected_codecs);
   EXPECT_EQ(expected_codecs, vcd->codecs());
 }
 
@@ -3454,8 +3383,8 @@ TEST_F(MediaSessionDescriptionFactoryTest, SimSsrcsGenerateMultipleRtxSsrcs) {
 
   // Use a single real codec, and then add RTX for it.
   std::vector<VideoCodec> f1_codecs;
-  f1_codecs.push_back(cricket::CreateVideoCodec(97, "H264"));
-  AddRtxCodec(cricket::CreateVideoRtxCodec(125, 97), &f1_codecs);
+  f1_codecs.push_back(CreateVideoCodec(97, "H264"));
+  AddRtxCodec(CreateVideoRtxCodec(125, 97), &f1_codecs);
   f1_.set_video_codecs(f1_codecs, f1_codecs);
 
   // Ensure that the offer has an RTX ssrc for each regular ssrc, and that there
@@ -3464,10 +3393,9 @@ TEST_F(MediaSessionDescriptionFactoryTest, SimSsrcsGenerateMultipleRtxSsrcs) {
       f1_.CreateOfferOrError(opts, nullptr).MoveValue();
   ASSERT_TRUE(offer.get());
   MediaContentDescription* media_desc =
-      offer->GetContentDescriptionByName(cricket::CN_VIDEO);
+      offer->GetContentDescriptionByName(CN_VIDEO);
   ASSERT_TRUE(media_desc);
-  VideoContentDescription* desc = media_desc->as_video();
-  const StreamParamsVec& streams = desc->streams();
+  const StreamParamsVec& streams = media_desc->streams();
   // Single stream.
   ASSERT_EQ(1u, streams.size());
   // Stream should have 6 ssrcs: 3 for video, 3 for RTX.
@@ -3499,8 +3427,8 @@ TEST_F(MediaSessionDescriptionFactoryTest, GenerateFlexfecSsrc) {
 
   // Use a single real codec, and then add FlexFEC for it.
   std::vector<VideoCodec> f1_codecs;
-  f1_codecs.push_back(cricket::CreateVideoCodec(97, "H264"));
-  f1_codecs.push_back(cricket::CreateVideoCodec(118, "flexfec-03"));
+  f1_codecs.push_back(CreateVideoCodec(97, "H264"));
+  f1_codecs.push_back(CreateVideoCodec(118, "flexfec-03"));
   f1_.set_video_codecs(f1_codecs, f1_codecs);
 
   // Ensure that the offer has a single FlexFEC ssrc and that
@@ -3509,10 +3437,9 @@ TEST_F(MediaSessionDescriptionFactoryTest, GenerateFlexfecSsrc) {
       f1_.CreateOfferOrError(opts, nullptr).MoveValue();
   ASSERT_TRUE(offer.get());
   MediaContentDescription* media_desc =
-      offer->GetContentDescriptionByName(cricket::CN_VIDEO);
+      offer->GetContentDescriptionByName(CN_VIDEO);
   ASSERT_TRUE(media_desc);
-  VideoContentDescription* desc = media_desc->as_video();
-  const StreamParamsVec& streams = desc->streams();
+  const StreamParamsVec& streams = media_desc->streams();
   // Single stream.
   ASSERT_EQ(1u, streams.size());
   // Stream should have 2 ssrcs: 1 for video, 1 for FlexFEC.
@@ -3543,8 +3470,8 @@ TEST_F(MediaSessionDescriptionFactoryTest, SimSsrcsGenerateNoFlexfecSsrcs) {
 
   // Use a single real codec, and then add FlexFEC for it.
   std::vector<VideoCodec> f1_codecs;
-  f1_codecs.push_back(cricket::CreateVideoCodec(97, "H264"));
-  f1_codecs.push_back(cricket::CreateVideoCodec(118, "flexfec-03"));
+  f1_codecs.push_back(CreateVideoCodec(97, "H264"));
+  f1_codecs.push_back(CreateVideoCodec(118, "flexfec-03"));
   f1_.set_video_codecs(f1_codecs, f1_codecs);
 
   // Ensure that the offer has no FlexFEC ssrcs for each regular ssrc, and that
@@ -3553,10 +3480,9 @@ TEST_F(MediaSessionDescriptionFactoryTest, SimSsrcsGenerateNoFlexfecSsrcs) {
       f1_.CreateOfferOrError(opts, nullptr).MoveValue();
   ASSERT_TRUE(offer.get());
   MediaContentDescription* media_desc =
-      offer->GetContentDescriptionByName(cricket::CN_VIDEO);
+      offer->GetContentDescriptionByName(CN_VIDEO);
   ASSERT_TRUE(media_desc);
-  VideoContentDescription* desc = media_desc->as_video();
-  const StreamParamsVec& streams = desc->streams();
+  const StreamParamsVec& streams = media_desc->streams();
   // Single stream.
   ASSERT_EQ(1u, streams.size());
   // Stream should have 3 ssrcs: 3 for video, 0 for FlexFEC.
@@ -3704,33 +3630,33 @@ TEST_F(MediaSessionDescriptionFactoryTest, RtpExtensionIdReusedEncrypted) {
 
 TEST(MediaSessionDescription, CopySessionDescription) {
   SessionDescription source;
-  cricket::ContentGroup group(cricket::CN_AUDIO);
+  ContentGroup group(CN_AUDIO);
   source.AddGroup(group);
   std::unique_ptr<AudioContentDescription> acd =
       std::make_unique<AudioContentDescription>();
   acd->set_codecs(MAKE_VECTOR(kAudioCodecs1));
   acd->AddLegacyStream(1);
-  source.AddContent(cricket::CN_AUDIO, MediaProtocolType::kRtp, acd->Clone());
+  source.AddContent(CN_AUDIO, MediaProtocolType::kRtp, acd->Clone());
   std::unique_ptr<VideoContentDescription> vcd =
       std::make_unique<VideoContentDescription>();
   vcd->set_codecs(MAKE_VECTOR(kVideoCodecs1));
   vcd->AddLegacyStream(2);
-  source.AddContent(cricket::CN_VIDEO, MediaProtocolType::kRtp, vcd->Clone());
+  source.AddContent(CN_VIDEO, MediaProtocolType::kRtp, vcd->Clone());
 
   std::unique_ptr<SessionDescription> copy = source.Clone();
   ASSERT_TRUE(copy.get());
-  EXPECT_TRUE(copy->HasGroup(cricket::CN_AUDIO));
+  EXPECT_TRUE(copy->HasGroup(CN_AUDIO));
   const ContentInfo* ac = copy->GetContentByName("audio");
   const ContentInfo* vc = copy->GetContentByName("video");
   ASSERT_TRUE(ac);
   ASSERT_TRUE(vc);
   EXPECT_EQ(MediaProtocolType::kRtp, ac->type);
-  const AudioContentDescription* acd_copy = ac->media_description()->as_audio();
+  const MediaContentDescription* acd_copy = ac->media_description();
   EXPECT_EQ(acd->codecs(), acd_copy->codecs());
   EXPECT_EQ(1u, acd->first_ssrc());
 
   EXPECT_EQ(MediaProtocolType::kRtp, vc->type);
-  const VideoContentDescription* vcd_copy = vc->media_description()->as_video();
+  const MediaContentDescription* vcd_copy = vc->media_description();
   EXPECT_EQ(vcd->codecs(), vcd_copy->codecs());
   EXPECT_EQ(2u, vcd->first_ssrc());
 }
@@ -3870,18 +3796,16 @@ TEST_F(MediaSessionDescriptionFactoryTest,
   tdf2_.set_secure(SEC_DISABLED);
 
   std::unique_ptr<SessionDescription> offer =
-      f1_.CreateOfferOrError(CreatePlanBMediaSessionOptions(), nullptr)
-          .MoveValue();
+      f1_.CreateOfferOrError(CreateAudioMediaSession(), nullptr).MoveValue();
   ASSERT_TRUE(offer.get());
   ContentInfo* offer_content = offer->GetContentByName("audio");
   ASSERT_TRUE(offer_content);
-  AudioContentDescription* offer_audio_desc =
-      offer_content->media_description()->as_audio();
-  offer_audio_desc->set_protocol(cricket::kMediaProtocolDtlsSavpf);
+  MediaContentDescription* offer_audio_desc =
+      offer_content->media_description();
+  offer_audio_desc->set_protocol(kMediaProtocolDtlsSavpf);
 
   std::unique_ptr<SessionDescription> answer =
-      f2_.CreateAnswerOrError(offer.get(), CreatePlanBMediaSessionOptions(),
-                              nullptr)
+      f2_.CreateAnswerOrError(offer.get(), CreateAudioMediaSession(), nullptr)
           .MoveValue();
   ASSERT_TRUE(answer);
   ContentInfo* answer_content = answer->GetContentByName("audio");
@@ -3899,18 +3823,16 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestOfferDtlsSavpfCreateAnswer) {
   tdf2_.set_secure(SEC_ENABLED);
 
   std::unique_ptr<SessionDescription> offer =
-      f1_.CreateOfferOrError(CreatePlanBMediaSessionOptions(), nullptr)
-          .MoveValue();
+      f1_.CreateOfferOrError(CreateAudioMediaSession(), nullptr).MoveValue();
   ASSERT_TRUE(offer.get());
   ContentInfo* offer_content = offer->GetContentByName("audio");
   ASSERT_TRUE(offer_content);
-  AudioContentDescription* offer_audio_desc =
-      offer_content->media_description()->as_audio();
-  offer_audio_desc->set_protocol(cricket::kMediaProtocolDtlsSavpf);
+  MediaContentDescription* offer_audio_desc =
+      offer_content->media_description();
+  offer_audio_desc->set_protocol(kMediaProtocolDtlsSavpf);
 
   std::unique_ptr<SessionDescription> answer =
-      f2_.CreateAnswerOrError(offer.get(), CreatePlanBMediaSessionOptions(),
-                              nullptr)
+      f2_.CreateAnswerOrError(offer.get(), CreateAudioMediaSession(), nullptr)
           .MoveValue();
   ASSERT_TRUE(answer);
 
@@ -3918,9 +3840,9 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestOfferDtlsSavpfCreateAnswer) {
   ASSERT_TRUE(answer_content);
   ASSERT_FALSE(answer_content->rejected);
 
-  const AudioContentDescription* answer_audio_desc =
-      answer_content->media_description()->as_audio();
-  EXPECT_EQ(cricket::kMediaProtocolDtlsSavpf, answer_audio_desc->protocol());
+  const MediaContentDescription* answer_audio_desc =
+      answer_content->media_description();
+  EXPECT_EQ(kMediaProtocolDtlsSavpf, answer_audio_desc->protocol());
 }
 
 // Test that we include both SDES and DTLS in the offer, but only include SDES
@@ -3933,10 +3855,10 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCryptoDtls) {
   MediaSessionOptions options;
   AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &options);
   std::unique_ptr<SessionDescription> offer, answer;
-  const cricket::MediaContentDescription* audio_media_desc;
-  const cricket::MediaContentDescription* video_media_desc;
-  const cricket::TransportDescription* audio_trans_desc;
-  const cricket::TransportDescription* video_trans_desc;
+  const MediaContentDescription* audio_media_desc;
+  const MediaContentDescription* video_media_desc;
+  const TransportDescription* audio_trans_desc;
+  const TransportDescription* video_trans_desc;
 
   // Generate an offer with SDES and DTLS support.
   offer = f1_.CreateOfferOrError(options, nullptr).MoveValue();
@@ -3985,8 +3907,8 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCryptoDtls) {
   ASSERT_TRUE(video_media_desc);
   EXPECT_TRUE(audio_media_desc->cryptos().empty());
   EXPECT_TRUE(video_media_desc->cryptos().empty());
-  EXPECT_EQ(cricket::kMediaProtocolSavpf, audio_media_desc->protocol());
-  EXPECT_EQ(cricket::kMediaProtocolSavpf, video_media_desc->protocol());
+  EXPECT_EQ(kMediaProtocolSavpf, audio_media_desc->protocol());
+  EXPECT_EQ(kMediaProtocolSavpf, video_media_desc->protocol());
 
   audio_trans_desc = answer->GetTransportDescriptionByName("audio");
   ASSERT_TRUE(audio_trans_desc);
@@ -4017,7 +3939,7 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCryptoDtls) {
 // Test that an answer can't be created if cryptos are required but the offer is
 // unsecure.
 TEST_F(MediaSessionDescriptionFactoryTest, TestSecureAnswerToUnsecureOffer) {
-  MediaSessionOptions options = CreatePlanBMediaSessionOptions();
+  MediaSessionOptions options = CreateAudioMediaSession();
   f1_.set_secure(SEC_DISABLED);
   tdf1_.set_secure(SEC_DISABLED);
   f2_.set_secure(SEC_REQUIRED);
@@ -4053,10 +3975,10 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCryptoOfferDtlsButNotSdes) {
       GetFirstVideoContentDescription(offer.get());
   ASSERT_TRUE(video_offer->cryptos().empty());
 
-  const cricket::TransportDescription* audio_offer_trans_desc =
+  const TransportDescription* audio_offer_trans_desc =
       offer->GetTransportDescriptionByName("audio");
   ASSERT_TRUE(audio_offer_trans_desc->identity_fingerprint.get());
-  const cricket::TransportDescription* video_offer_trans_desc =
+  const TransportDescription* video_offer_trans_desc =
       offer->GetTransportDescriptionByName("video");
   ASSERT_TRUE(video_offer_trans_desc->identity_fingerprint.get());
 
@@ -4065,10 +3987,10 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCryptoOfferDtlsButNotSdes) {
       f2_.CreateAnswerOrError(offer.get(), options, nullptr).MoveValue();
   ASSERT_TRUE(answer.get());
 
-  const cricket::TransportDescription* audio_answer_trans_desc =
+  const TransportDescription* audio_answer_trans_desc =
       answer->GetTransportDescriptionByName("audio");
   EXPECT_TRUE(audio_answer_trans_desc->identity_fingerprint.get());
-  const cricket::TransportDescription* video_answer_trans_desc =
+  const TransportDescription* video_answer_trans_desc =
       answer->GetTransportDescriptionByName("video");
   EXPECT_TRUE(video_answer_trans_desc->identity_fingerprint.get());
 }
@@ -4160,27 +4082,25 @@ TEST_F(MediaSessionDescriptionFactoryTest,
 
   ASSERT_EQ(4u, offer->contents().size());
   EXPECT_FALSE(offer->contents()[0].rejected);
-  const AudioContentDescription* acd =
-      offer->contents()[0].media_description()->as_audio();
+  const MediaContentDescription* acd = offer->contents()[0].media_description();
   ASSERT_EQ(1u, acd->streams().size());
   EXPECT_EQ(kAudioTrack1, acd->streams()[0].id);
   EXPECT_EQ(RtpTransceiverDirection::kSendRecv, acd->direction());
 
   EXPECT_FALSE(offer->contents()[1].rejected);
-  const VideoContentDescription* vcd =
-      offer->contents()[1].media_description()->as_video();
+  const MediaContentDescription* vcd = offer->contents()[1].media_description();
   ASSERT_EQ(1u, vcd->streams().size());
   EXPECT_EQ(kVideoTrack1, vcd->streams()[0].id);
   EXPECT_EQ(RtpTransceiverDirection::kSendRecv, vcd->direction());
 
   EXPECT_FALSE(offer->contents()[2].rejected);
-  acd = offer->contents()[2].media_description()->as_audio();
+  acd = offer->contents()[2].media_description();
   ASSERT_EQ(1u, acd->streams().size());
   EXPECT_EQ(kAudioTrack2, acd->streams()[0].id);
   EXPECT_EQ(RtpTransceiverDirection::kSendRecv, acd->direction());
 
   EXPECT_FALSE(offer->contents()[3].rejected);
-  vcd = offer->contents()[3].media_description()->as_video();
+  vcd = offer->contents()[3].media_description();
   ASSERT_EQ(1u, vcd->streams().size());
   EXPECT_EQ(kVideoTrack2, vcd->streams()[0].id);
   EXPECT_EQ(RtpTransceiverDirection::kSendRecv, vcd->direction());
@@ -4223,27 +4143,27 @@ TEST_F(MediaSessionDescriptionFactoryTest,
 
   ASSERT_EQ(4u, answer->contents().size());
   EXPECT_FALSE(answer->contents()[0].rejected);
-  const AudioContentDescription* acd =
-      answer->contents()[0].media_description()->as_audio();
+  const MediaContentDescription* acd =
+      answer->contents()[0].media_description();
   ASSERT_EQ(1u, acd->streams().size());
   EXPECT_EQ(kAudioTrack1, acd->streams()[0].id);
   EXPECT_EQ(RtpTransceiverDirection::kSendRecv, acd->direction());
 
   EXPECT_FALSE(answer->contents()[1].rejected);
-  const VideoContentDescription* vcd =
-      answer->contents()[1].media_description()->as_video();
+  const MediaContentDescription* vcd =
+      answer->contents()[1].media_description();
   ASSERT_EQ(1u, vcd->streams().size());
   EXPECT_EQ(kVideoTrack1, vcd->streams()[0].id);
   EXPECT_EQ(RtpTransceiverDirection::kSendRecv, vcd->direction());
 
   EXPECT_FALSE(answer->contents()[2].rejected);
-  acd = answer->contents()[2].media_description()->as_audio();
+  acd = answer->contents()[2].media_description();
   ASSERT_EQ(1u, acd->streams().size());
   EXPECT_EQ(kAudioTrack2, acd->streams()[0].id);
   EXPECT_EQ(RtpTransceiverDirection::kSendRecv, acd->direction());
 
   EXPECT_FALSE(answer->contents()[3].rejected);
-  vcd = answer->contents()[3].media_description()->as_video();
+  vcd = answer->contents()[3].media_description();
   ASSERT_EQ(1u, vcd->streams().size());
   EXPECT_EQ(kVideoTrack2, vcd->streams()[0].id);
   EXPECT_EQ(RtpTransceiverDirection::kSendRecv, vcd->direction());
@@ -4379,10 +4299,10 @@ TEST_F(MediaSessionDescriptionFactoryTest,
       f1_.CreateOfferOrError(opts, nullptr).MoveValue();
   ASSERT_TRUE(offer);
   ASSERT_EQ(2u, offer->contents().size());
-  const VideoContentDescription* vcd1 =
-      offer->contents()[0].media_description()->as_video();
-  const VideoContentDescription* vcd2 =
-      offer->contents()[1].media_description()->as_video();
+  const MediaContentDescription* vcd1 =
+      offer->contents()[0].media_description();
+  const MediaContentDescription* vcd2 =
+      offer->contents()[1].media_description();
   EXPECT_EQ(vcd1->codecs().size(), vcd2->codecs().size());
   ASSERT_EQ(2u, vcd1->codecs().size());
   EXPECT_EQ(vcd1->codecs()[0].name, vcd2->codecs()[0].name);
@@ -4395,8 +4315,8 @@ TEST_F(MediaSessionDescriptionFactoryTest,
       f2_.CreateAnswerOrError(offer.get(), opts, nullptr).MoveValue();
   ASSERT_TRUE(answer);
   ASSERT_EQ(2u, answer->contents().size());
-  vcd1 = answer->contents()[0].media_description()->as_video();
-  vcd2 = answer->contents()[1].media_description()->as_video();
+  vcd1 = answer->contents()[0].media_description();
+  vcd2 = answer->contents()[1].media_description();
   EXPECT_EQ(vcd1->codecs().size(), vcd2->codecs().size());
   ASSERT_EQ(1u, vcd1->codecs().size());
   EXPECT_EQ(vcd1->codecs()[0].name, vcd2->codecs()[0].name);
@@ -4406,11 +4326,11 @@ TEST_F(MediaSessionDescriptionFactoryTest,
 // Test verifying that negotiating codecs with the same packetization retains
 // the packetization value.
 TEST_F(MediaSessionDescriptionFactoryTest, PacketizationIsEqual) {
-  std::vector f1_codecs = {cricket::CreateVideoCodec(96, "H264")};
+  std::vector f1_codecs = {CreateVideoCodec(96, "H264")};
   f1_codecs.back().packetization = "raw";
   f1_.set_video_codecs(f1_codecs, f1_codecs);
 
-  std::vector f2_codecs = {cricket::CreateVideoCodec(96, "H264")};
+  std::vector f2_codecs = {CreateVideoCodec(96, "H264")};
   f2_codecs.back().packetization = "raw";
   f2_.set_video_codecs(f2_codecs, f2_codecs);
 
@@ -4424,8 +4344,8 @@ TEST_F(MediaSessionDescriptionFactoryTest, PacketizationIsEqual) {
       f1_.CreateOfferOrError(opts, nullptr).MoveValue();
   ASSERT_TRUE(offer);
   ASSERT_EQ(1u, offer->contents().size());
-  const VideoContentDescription* vcd1 =
-      offer->contents()[0].media_description()->as_video();
+  const MediaContentDescription* vcd1 =
+      offer->contents()[0].media_description();
   ASSERT_EQ(1u, vcd1->codecs().size());
   EXPECT_EQ(vcd1->codecs()[0].packetization, "raw");
 
@@ -4434,7 +4354,7 @@ TEST_F(MediaSessionDescriptionFactoryTest, PacketizationIsEqual) {
       f2_.CreateAnswerOrError(offer.get(), opts, nullptr).MoveValue();
   ASSERT_TRUE(answer);
   ASSERT_EQ(1u, answer->contents().size());
-  vcd1 = answer->contents()[0].media_description()->as_video();
+  vcd1 = answer->contents()[0].media_description();
   ASSERT_EQ(1u, vcd1->codecs().size());
   EXPECT_EQ(vcd1->codecs()[0].packetization, "raw");
 }
@@ -4442,11 +4362,11 @@ TEST_F(MediaSessionDescriptionFactoryTest, PacketizationIsEqual) {
 // Test verifying that negotiating codecs with different packetization removes
 // the packetization value.
 TEST_F(MediaSessionDescriptionFactoryTest, PacketizationIsDifferent) {
-  std::vector f1_codecs = {cricket::CreateVideoCodec(96, "H264")};
+  std::vector f1_codecs = {CreateVideoCodec(96, "H264")};
   f1_codecs.back().packetization = "raw";
   f1_.set_video_codecs(f1_codecs, f1_codecs);
 
-  std::vector f2_codecs = {cricket::CreateVideoCodec(96, "H264")};
+  std::vector f2_codecs = {CreateVideoCodec(96, "H264")};
   f2_codecs.back().packetization = "notraw";
   f2_.set_video_codecs(f2_codecs, f2_codecs);
 
@@ -4491,10 +4411,9 @@ TEST_F(MediaSessionDescriptionFactoryTest,
       f1_.CreateOfferOrError(opts, nullptr).MoveValue();
   ASSERT_TRUE(offer);
   ASSERT_EQ(2u, offer->contents().size());
-  VideoContentDescription* vcd1 =
-      offer->contents()[0].media_description()->as_video();
-  const VideoContentDescription* vcd2 =
-      offer->contents()[1].media_description()->as_video();
+  MediaContentDescription* vcd1 = offer->contents()[0].media_description();
+  const MediaContentDescription* vcd2 =
+      offer->contents()[1].media_description();
   auto video_codecs = MAKE_VECTOR(kVideoCodecs1);
   EXPECT_EQ(video_codecs, vcd1->codecs());
   EXPECT_EQ(video_codecs, vcd2->codecs());
@@ -4505,8 +4424,8 @@ TEST_F(MediaSessionDescriptionFactoryTest,
   vcd1->set_codecs(video_codecs_reverse);
   std::unique_ptr<SessionDescription> updated_offer(
       f1_.CreateOfferOrError(opts, offer.get()).MoveValue());
-  vcd1 = updated_offer->contents()[0].media_description()->as_video();
-  vcd2 = updated_offer->contents()[1].media_description()->as_video();
+  vcd1 = updated_offer->contents()[0].media_description();
+  vcd2 = updated_offer->contents()[1].media_description();
   // The video codec preference order should be respected.
   EXPECT_EQ(video_codecs_reverse, vcd1->codecs());
   EXPECT_EQ(video_codecs, vcd2->codecs());
@@ -4528,10 +4447,9 @@ TEST_F(MediaSessionDescriptionFactoryTest,
       f1_.CreateOfferOrError(opts, nullptr).MoveValue();
   ASSERT_TRUE(offer);
   ASSERT_EQ(2u, offer->contents().size());
-  VideoContentDescription* vcd1 =
-      offer->contents()[0].media_description()->as_video();
-  const VideoContentDescription* vcd2 =
-      offer->contents()[1].media_description()->as_video();
+  MediaContentDescription* vcd1 = offer->contents()[0].media_description();
+  const MediaContentDescription* vcd2 =
+      offer->contents()[1].media_description();
   auto video_codecs = MAKE_VECTOR(kVideoCodecs1);
   EXPECT_EQ(video_codecs, vcd1->codecs());
   EXPECT_EQ(video_codecs, vcd2->codecs());
@@ -4542,8 +4460,8 @@ TEST_F(MediaSessionDescriptionFactoryTest,
   vcd1->set_codecs(video_codecs_reverse);
   std::unique_ptr<SessionDescription> answer =
       f1_.CreateAnswerOrError(offer.get(), opts, nullptr).MoveValue();
-  vcd1 = answer->contents()[0].media_description()->as_video();
-  vcd2 = answer->contents()[1].media_description()->as_video();
+  vcd1 = answer->contents()[0].media_description();
+  vcd2 = answer->contents()[1].media_description();
   // The video codec preference order should be respected.
   EXPECT_EQ(video_codecs_reverse, vcd1->codecs());
   EXPECT_EQ(video_codecs, vcd2->codecs());
@@ -4586,8 +4504,8 @@ TEST_F(MediaSessionDescriptionFactoryTest, CreateAnswerWithLocalCodecParams) {
   std::unique_ptr<SessionDescription> offer =
       f1_.CreateOfferOrError(opts, nullptr).MoveValue();
   ASSERT_TRUE(offer);
-  auto offer_acd = offer->contents()[0].media_description()->as_audio();
-  auto offer_vcd = offer->contents()[1].media_description()->as_video();
+  auto offer_acd = offer->contents()[0].media_description();
+  auto offer_vcd = offer->contents()[1].media_description();
   std::string value;
   EXPECT_TRUE(offer_acd->codecs()[0].GetParam(audio_param_name, &value));
   EXPECT_EQ(audio_value1, value);
@@ -4597,8 +4515,8 @@ TEST_F(MediaSessionDescriptionFactoryTest, CreateAnswerWithLocalCodecParams) {
   std::unique_ptr<SessionDescription> answer =
       f2_.CreateAnswerOrError(offer.get(), opts, nullptr).MoveValue();
   ASSERT_TRUE(answer);
-  auto answer_acd = answer->contents()[0].media_description()->as_audio();
-  auto answer_vcd = answer->contents()[1].media_description()->as_video();
+  auto answer_acd = answer->contents()[0].media_description();
+  auto answer_vcd = answer->contents()[1].media_description();
   // Use the parameters from the local codecs.
   EXPECT_TRUE(answer_acd->codecs()[0].GetParam(audio_param_name, &value));
   EXPECT_EQ(audio_value2, value);
@@ -4615,12 +4533,12 @@ TEST_F(MediaSessionDescriptionFactoryTest,
        H264MatchCriteriaIncludesPacketizationMode) {
   // Create two H264 codecs with the same profile level ID and different
   // packetization modes.
-  VideoCodec h264_pm0 = cricket::CreateVideoCodec(96, "H264");
-  h264_pm0.params[cricket::kH264FmtpProfileLevelId] = "42c01f";
-  h264_pm0.params[cricket::kH264FmtpPacketizationMode] = "0";
-  VideoCodec h264_pm1 = cricket::CreateVideoCodec(97, "H264");
-  h264_pm1.params[cricket::kH264FmtpProfileLevelId] = "42c01f";
-  h264_pm1.params[cricket::kH264FmtpPacketizationMode] = "1";
+  VideoCodec h264_pm0 = CreateVideoCodec(96, "H264");
+  h264_pm0.params[kH264FmtpProfileLevelId] = "42c01f";
+  h264_pm0.params[kH264FmtpPacketizationMode] = "0";
+  VideoCodec h264_pm1 = CreateVideoCodec(97, "H264");
+  h264_pm1.params[kH264FmtpProfileLevelId] = "42c01f";
+  h264_pm1.params[kH264FmtpPacketizationMode] = "1";
 
   // Offerer will send both codecs, answerer should choose the one with matching
   // packetization mode (and not the first one it sees).
@@ -4643,13 +4561,13 @@ TEST_F(MediaSessionDescriptionFactoryTest,
   // Answer should have one negotiated codec with packetization-mode=1 using the
   // offered payload type.
   ASSERT_EQ(1u, answer->contents().size());
-  auto answer_vcd = answer->contents()[0].media_description()->as_video();
+  auto answer_vcd = answer->contents()[0].media_description();
   ASSERT_EQ(1u, answer_vcd->codecs().size());
   auto answer_codec = answer_vcd->codecs()[0];
   EXPECT_EQ(h264_pm1.id, answer_codec.id);
 }
 
-class MediaProtocolTest : public ::testing::TestWithParam<const char*> {
+class MediaProtocolTest : public testing::TestWithParam<const char*> {
  public:
   MediaProtocolTest()
       : tdf1_(field_trials_),
@@ -4702,18 +4620,18 @@ TEST_P(MediaProtocolTest, TestAudioVideoAcceptance) {
   ASSERT_TRUE(vc);
   EXPECT_FALSE(ac->rejected);  // the offer is accepted
   EXPECT_FALSE(vc->rejected);
-  const AudioContentDescription* acd = ac->media_description()->as_audio();
-  const VideoContentDescription* vcd = vc->media_description()->as_video();
+  const MediaContentDescription* acd = ac->media_description();
+  const MediaContentDescription* vcd = vc->media_description();
   EXPECT_EQ(GetParam(), acd->protocol());
   EXPECT_EQ(GetParam(), vcd->protocol());
 }
 
 INSTANTIATE_TEST_SUITE_P(MediaProtocolPatternTest,
                          MediaProtocolTest,
-                         ::testing::ValuesIn(kMediaProtocols));
+                         ValuesIn(kMediaProtocols));
 INSTANTIATE_TEST_SUITE_P(MediaProtocolDtlsPatternTest,
                          MediaProtocolTest,
-                         ::testing::ValuesIn(kMediaProtocolsDtls));
+                         ValuesIn(kMediaProtocolsDtls));
 
 TEST_F(MediaSessionDescriptionFactoryTest, TestSetAudioCodecs) {
   webrtc::test::ScopedKeyValueConfig field_trials;
@@ -4769,17 +4687,15 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestSetAudioCodecs) {
   EXPECT_EQ(no_codecs, sf.audio_sendrecv_codecs());
 }
 
-namespace {
 // Compare the two vectors of codecs ignoring the payload type.
-bool CodecsMatch(const std::vector<cricket::Codec>& codecs1,
-                 const std::vector<cricket::Codec>& codecs2,
-                 const webrtc::FieldTrialsView* field_trials) {
+bool CodecsMatch(const std::vector<Codec>& codecs1,
+                 const std::vector<Codec>& codecs2) {
   if (codecs1.size() != codecs2.size()) {
     return false;
   }
 
   for (size_t i = 0; i < codecs1.size(); ++i) {
-    if (!codecs1[i].Matches(codecs2[i], field_trials)) {
+    if (!codecs1[i].Matches(codecs2[i])) {
       return false;
     }
   }
@@ -4817,29 +4733,29 @@ void TestAudioCodecsOffer(RtpTransceiverDirection direction) {
   // send nor receive audio. The checks are still in place if at some point
   // we'd instead create an inactive stream.
   if (ac) {
-    AudioContentDescription* acd = ac->media_description()->as_audio();
+    MediaContentDescription* acd = ac->media_description();
     // sendrecv and inactive should both present lists as if the channel was
     // to be used for sending and receiving. Inactive essentially means it
     // might eventually be used anything, but we don't know more at this
     // moment.
     if (acd->direction() == RtpTransceiverDirection::kSendOnly) {
-      EXPECT_TRUE(CodecsMatch(send_codecs, acd->codecs(), &field_trials));
+      EXPECT_TRUE(CodecsMatch(send_codecs, acd->codecs()));
     } else if (acd->direction() == RtpTransceiverDirection::kRecvOnly) {
-      EXPECT_TRUE(CodecsMatch(recv_codecs, acd->codecs(), &field_trials));
+      EXPECT_TRUE(CodecsMatch(recv_codecs, acd->codecs()));
     } else {
-      EXPECT_TRUE(CodecsMatch(sendrecv_codecs, acd->codecs(), &field_trials));
+      EXPECT_TRUE(CodecsMatch(sendrecv_codecs, acd->codecs()));
     }
   }
 }
 
-static const AudioCodec kOfferAnswerCodecs[] = {
-    cricket::CreateAudioCodec(0, "codec0", 16000, 1),
-    cricket::CreateAudioCodec(1, "codec1", 8000, 1),
-    cricket::CreateAudioCodec(2, "codec2", 8000, 1),
-    cricket::CreateAudioCodec(3, "codec3", 8000, 1),
-    cricket::CreateAudioCodec(4, "codec4", 8000, 2),
-    cricket::CreateAudioCodec(5, "codec5", 32000, 1),
-    cricket::CreateAudioCodec(6, "codec6", 48000, 1)};
+const AudioCodec kOfferAnswerCodecs[] = {
+    CreateAudioCodec(0, "codec0", 16000, 1),
+    CreateAudioCodec(1, "codec1", 8000, 1),
+    CreateAudioCodec(2, "codec2", 8000, 1),
+    CreateAudioCodec(3, "codec3", 8000, 1),
+    CreateAudioCodec(4, "codec4", 8000, 2),
+    CreateAudioCodec(5, "codec5", 32000, 1),
+    CreateAudioCodec(6, "codec6", 48000, 1)};
 
 /* The codecs groups below are chosen as per the matrix below. The objective
  * is to have different sets of codecs in the inputs, to get unique sets of
@@ -4860,18 +4776,18 @@ static const AudioCodec kOfferAnswerCodecs[] = {
  *     6   | x    x    x |  x    x    x |  x    x    x    x    x
  */
 // Codecs used by offerer in the AudioCodecsAnswerTest
-static const int kOfferSendCodecs[] = {0, 1, 3, 5, 6};
-static const int kOfferRecvCodecs[] = {1, 2, 3, 4, 6};
+const int kOfferSendCodecs[] = {0, 1, 3, 5, 6};
+const int kOfferRecvCodecs[] = {1, 2, 3, 4, 6};
 // Codecs used in the answerer in the AudioCodecsAnswerTest.  The order is
 // jumbled to catch the answer not following the order in the offer.
-static const int kAnswerSendCodecs[] = {6, 5, 2, 3, 4};
-static const int kAnswerRecvCodecs[] = {6, 5, 4, 1, 0};
+const int kAnswerSendCodecs[] = {6, 5, 2, 3, 4};
+const int kAnswerRecvCodecs[] = {6, 5, 4, 1, 0};
 // The resulting sets of codecs in the answer in the AudioCodecsAnswerTest
-static const int kResultSend_RecvCodecs[] = {0, 1, 5, 6};
-static const int kResultRecv_SendCodecs[] = {2, 3, 4, 6};
-static const int kResultSendrecv_SendCodecs[] = {3, 6};
-static const int kResultSendrecv_RecvCodecs[] = {1, 6};
-static const int kResultSendrecv_SendrecvCodecs[] = {6};
+const int kResultSend_RecvCodecs[] = {0, 1, 5, 6};
+const int kResultRecv_SendCodecs[] = {2, 3, 4, 6};
+const int kResultSendrecv_SendCodecs[] = {3, 6};
+const int kResultSendrecv_RecvCodecs[] = {1, 6};
+const int kResultSendrecv_SendrecvCodecs[] = {6};
 
 template <typename T, int IDXS>
 std::vector<T> VectorFromIndices(const T* array, const int (&indices)[IDXS]) {
@@ -4936,7 +4852,7 @@ void TestAudioCodecsAnswer(RtpTransceiverDirection offer_direction,
   // we'd instead create an inactive stream.
   if (ac) {
     ASSERT_EQ(MEDIA_TYPE_AUDIO, ac->media_description()->type());
-    const AudioContentDescription* acd = ac->media_description()->as_audio();
+    const MediaContentDescription* acd = ac->media_description();
 
     std::vector<AudioCodec> target_codecs;
     // For offers with sendrecv or inactive, we should never reply with more
@@ -4998,10 +4914,7 @@ void TestAudioCodecsAnswer(RtpTransceiverDirection offer_direction,
   }
 }
 
-}  // namespace
-
-class AudioCodecsOfferTest
-    : public ::testing::TestWithParam<RtpTransceiverDirection> {};
+using AudioCodecsOfferTest = testing::TestWithParam<RtpTransceiverDirection>;
 
 TEST_P(AudioCodecsOfferTest, TestCodecsInOffer) {
   TestAudioCodecsOffer(GetParam());
@@ -5009,31 +4922,30 @@ TEST_P(AudioCodecsOfferTest, TestCodecsInOffer) {
 
 INSTANTIATE_TEST_SUITE_P(MediaSessionDescriptionFactoryTest,
                          AudioCodecsOfferTest,
-                         ::testing::Values(RtpTransceiverDirection::kSendOnly,
-                                           RtpTransceiverDirection::kRecvOnly,
-                                           RtpTransceiverDirection::kSendRecv,
-                                           RtpTransceiverDirection::kInactive));
+                         Values(RtpTransceiverDirection::kSendOnly,
+                                RtpTransceiverDirection::kRecvOnly,
+                                RtpTransceiverDirection::kSendRecv,
+                                RtpTransceiverDirection::kInactive));
 
-class AudioCodecsAnswerTest
-    : public ::testing::TestWithParam<::testing::tuple<RtpTransceiverDirection,
-                                                       RtpTransceiverDirection,
-                                                       bool>> {};
+using AudioCodecsAnswerTest = testing::TestWithParam<
+    std::tuple<RtpTransceiverDirection, RtpTransceiverDirection, bool>>;
 
 TEST_P(AudioCodecsAnswerTest, TestCodecsInAnswer) {
-  TestAudioCodecsAnswer(::testing::get<0>(GetParam()),
-                        ::testing::get<1>(GetParam()),
-                        ::testing::get<2>(GetParam()));
-}
-
-INSTANTIATE_TEST_SUITE_P(
-    MediaSessionDescriptionFactoryTest,
-    AudioCodecsAnswerTest,
-    ::testing::Combine(::testing::Values(RtpTransceiverDirection::kSendOnly,
-                                         RtpTransceiverDirection::kRecvOnly,
-                                         RtpTransceiverDirection::kSendRecv,
-                                         RtpTransceiverDirection::kInactive),
-                       ::testing::Values(RtpTransceiverDirection::kSendOnly,
-                                         RtpTransceiverDirection::kRecvOnly,
-                                         RtpTransceiverDirection::kSendRecv,
-                                         RtpTransceiverDirection::kInactive),
-                       ::testing::Bool()));
+  TestAudioCodecsAnswer(std::get<0>(GetParam()), std::get<1>(GetParam()),
+                        std::get<2>(GetParam()));
+}
+
+INSTANTIATE_TEST_SUITE_P(MediaSessionDescriptionFactoryTest,
+                         AudioCodecsAnswerTest,
+                         Combine(Values(RtpTransceiverDirection::kSendOnly,
+                                        RtpTransceiverDirection::kRecvOnly,
+                                        RtpTransceiverDirection::kSendRecv,
+                                        RtpTransceiverDirection::kInactive),
+                                 Values(RtpTransceiverDirection::kSendOnly,
+                                        RtpTransceiverDirection::kRecvOnly,
+                                        RtpTransceiverDirection::kSendRecv,
+                                        RtpTransceiverDirection::kInactive),
+                                 Bool()));
+
+}  // namespace
+}  // namespace cricket
diff --git a/third_party/libwebrtc/pc/media_stream_unittest.cc b/third_party/libwebrtc/pc/media_stream_unittest.cc
index f55ea203fb..d6c79efae9 100644
--- a/third_party/libwebrtc/pc/media_stream_unittest.cc
+++ b/third_party/libwebrtc/pc/media_stream_unittest.cc
@@ -91,7 +91,7 @@ TEST_F(MediaStreamTest, GetTrackInfo) {
   ASSERT_EQ(1u, stream_->GetAudioTracks().size());
 
   // Verify the video track.
-  scoped_refptr<webrtc::MediaStreamTrackInterface> video_track(
+  scoped_refptr<MediaStreamTrackInterface> video_track(
       stream_->GetVideoTracks()[0]);
   EXPECT_EQ(0, video_track->id().compare(kVideoTrackId));
   EXPECT_TRUE(video_track->enabled());
@@ -105,7 +105,7 @@ TEST_F(MediaStreamTest, GetTrackInfo) {
   EXPECT_TRUE(video_track->enabled());
 
   // Verify the audio track.
-  scoped_refptr<webrtc::MediaStreamTrackInterface> audio_track(
+  scoped_refptr<MediaStreamTrackInterface> audio_track(
       stream_->GetAudioTracks()[0]);
   EXPECT_EQ(0, audio_track->id().compare(kAudioTrackId));
   EXPECT_TRUE(audio_track->enabled());
@@ -139,14 +139,12 @@ TEST_F(MediaStreamTest, RemoveTrack) {
 }
 
 TEST_F(MediaStreamTest, ChangeVideoTrack) {
-  scoped_refptr<webrtc::VideoTrackInterface> video_track(
-      stream_->GetVideoTracks()[0]);
+  scoped_refptr<VideoTrackInterface> video_track(stream_->GetVideoTracks()[0]);
   ChangeTrack(video_track.get());
 }
 
 TEST_F(MediaStreamTest, ChangeAudioTrack) {
-  scoped_refptr<webrtc::AudioTrackInterface> audio_track(
-      stream_->GetAudioTracks()[0]);
+  scoped_refptr<AudioTrackInterface> audio_track(stream_->GetAudioTracks()[0]);
   ChangeTrack(audio_track.get());
 }
 
diff --git a/third_party/libwebrtc/pc/peer_connection.cc b/third_party/libwebrtc/pc/peer_connection.cc
index 183cbeb7cd..26b70c63db 100644
--- a/third_party/libwebrtc/pc/peer_connection.cc
+++ b/third_party/libwebrtc/pc/peer_connection.cc
@@ -23,6 +23,7 @@
 #include "absl/strings/match.h"
 #include "absl/strings/string_view.h"
 #include "absl/types/optional.h"
+#include "api/environment/environment.h"
 #include "api/jsep_ice_candidate.h"
 #include "api/media_types.h"
 #include "api/rtp_parameters.h"
@@ -185,7 +186,7 @@ IceCandidatePairType GetIceCandidatePairCounter(
 absl::optional<int> RTCConfigurationToIceConfigOptionalInt(
     int rtc_configuration_parameter) {
   if (rtc_configuration_parameter ==
-      webrtc::PeerConnectionInterface::RTCConfiguration::kUndefined) {
+      PeerConnectionInterface::RTCConfiguration::kUndefined) {
     return absl::nullopt;
   }
   return rtc_configuration_parameter;
@@ -449,7 +450,7 @@ bool PeerConnectionInterface::RTCConfiguration::operator==(
     absl::optional<int> ice_unwritable_min_checks;
     absl::optional<int> ice_inactive_timeout;
     absl::optional<int> stun_candidate_keepalive_interval;
-    webrtc::TurnCustomizer* turn_customizer;
+    TurnCustomizer* turn_customizer;
     SdpSemantics sdp_semantics;
     absl::optional<rtc::AdapterType> network_preference;
     bool active_reset_srtp_params;
@@ -459,7 +460,7 @@ bool PeerConnectionInterface::RTCConfiguration::operator==(
     bool enable_implicit_rollback;
     absl::optional<int> report_usage_pattern_delay_ms;
     absl::optional<int> stable_writable_connection_ping_interval_ms;
-    webrtc::VpnPreference vpn_preference;
+    VpnPreference vpn_preference;
     std::vector<rtc::NetworkMask> vpn_list;
     PortAllocatorConfig port_allocator_config;
     absl::optional<TimeDelta> pacer_burst_interval;
@@ -536,9 +537,9 @@ bool PeerConnectionInterface::RTCConfiguration::operator!=(
 }
 
 RTCErrorOr<rtc::scoped_refptr<PeerConnection>> PeerConnection::Create(
+    const Environment& env,
     rtc::scoped_refptr<ConnectionContext> context,
     const PeerConnectionFactoryInterface::Options& options,
-    std::unique_ptr<RtcEventLog> event_log,
     std::unique_ptr<Call> call,
     const PeerConnectionInterface::RTCConfiguration& configuration,
     PeerConnectionDependencies dependencies) {
@@ -577,36 +578,15 @@ RTCErrorOr<rtc::scoped_refptr<PeerConnection>> PeerConnection::Create(
       configuration.sdp_semantics == SdpSemantics::kUnifiedPlan;
   bool dtls_enabled = DtlsEnabled(configuration, options, dependencies);
 
-  // Interim code: If an AsyncResolverFactory is given, but not an
-  // AsyncDnsResolverFactory, wrap it in a WrappingAsyncDnsResolverFactory
-  // If neither is given, create a BasicAsyncDnsResolverFactory.
-  // TODO(bugs.webrtc.org/12598): Remove code once all callers pass a
-  // AsyncDnsResolverFactory.
-#pragma clang diagnostic push
-#pragma clang diagnostic ignored "-Wdeprecated-declarations"
-  if (dependencies.async_dns_resolver_factory &&
-      dependencies.async_resolver_factory) {
-    RTC_LOG(LS_ERROR)
-        << "Attempt to set both old and new type of DNS resolver factory";
-    return RTCError(RTCErrorType::INVALID_PARAMETER,
-                    "Both old and new type of DNS resolver given");
-  }
   if (!dependencies.async_dns_resolver_factory) {
-    if (dependencies.async_resolver_factory) {
-      dependencies.async_dns_resolver_factory =
-          std::make_unique<WrappingAsyncDnsResolverFactory>(
-              std::move(dependencies.async_resolver_factory));
-    } else {
       dependencies.async_dns_resolver_factory =
           std::make_unique<BasicAsyncDnsResolverFactory>();
-    }
   }
-#pragma clang diagnostic pop
 
   // The PeerConnection constructor consumes some, but not all, dependencies.
   auto pc = rtc::make_ref_counted<PeerConnection>(
-      context, options, is_unified_plan, std::move(event_log), std::move(call),
-      dependencies, dtls_enabled);
+      env, context, options, is_unified_plan, std::move(call), dependencies,
+      dtls_enabled);
   RTCError init_error = pc->Initialize(configuration, std::move(dependencies));
   if (!init_error.ok()) {
     RTC_LOG(LS_ERROR) << "PeerConnection initialization failed";
@@ -616,20 +596,18 @@ RTCErrorOr<rtc::scoped_refptr<PeerConnection>> PeerConnection::Create(
 }
 
 PeerConnection::PeerConnection(
+    const Environment& env,
     rtc::scoped_refptr<ConnectionContext> context,
     const PeerConnectionFactoryInterface::Options& options,
     bool is_unified_plan,
-    std::unique_ptr<RtcEventLog> event_log,
     std::unique_ptr<Call> call,
     PeerConnectionDependencies& dependencies,
     bool dtls_enabled)
-    : context_(context),
-      trials_(std::move(dependencies.trials), &context->field_trials()),
+    : env_(env),
+      context_(context),
       options_(options),
       observer_(dependencies.observer),
       is_unified_plan_(is_unified_plan),
-      event_log_(std::move(event_log)),
-      event_log_ptr_(event_log_.get()),
       async_dns_resolver_factory_(
           std::move(dependencies.async_dns_resolver_factory)),
       port_allocator_(std::move(dependencies.allocator)),
@@ -648,7 +626,10 @@ PeerConnection::PeerConnection(
       dtls_enabled_(dtls_enabled),
       data_channel_controller_(this),
       message_handler_(signaling_thread()),
-      weak_factory_(this) {}
+      weak_factory_(this) {
+  // Field trials specific to the peerconnection should be owned by the `env`,
+  RTC_DCHECK(dependencies.trials == nullptr);
+}
 
 PeerConnection::~PeerConnection() {
   TRACE_EVENT0("webrtc", "PeerConnection::~PeerConnection");
@@ -699,13 +680,11 @@ PeerConnection::~PeerConnection() {
   sctp_mid_s_.reset();
   SetSctpTransportName("");
 
-  // call_ and event_log_ must be destroyed on the worker thread.
+  // call_ must be destroyed on the worker thread.
   worker_thread()->BlockingCall([this] {
     RTC_DCHECK_RUN_ON(worker_thread());
     worker_thread_safety_->SetNotAlive();
     call_.reset();
-    // The event log must outlive call (and any other object that uses it).
-    event_log_.reset();
   });
 
   data_channel_controller_.PrepareForShutdown();
@@ -797,7 +776,7 @@ JsepTransportController* PeerConnection::InitializeTransportController_n(
   config.transport_observer = this;
   config.rtcp_handler = InitializeRtcpCallback();
   config.un_demuxable_packet_handler = InitializeUnDemuxablePacketHandler();
-  config.event_log = event_log_ptr_;
+  config.event_log = &env_.event_log();
 #if defined(ENABLE_EXTERNAL_AUTH)
   config.enable_external_auth = true;
 #endif
@@ -816,7 +795,7 @@ JsepTransportController* PeerConnection::InitializeTransportController_n(
         }
       };
 
-  config.field_trials = trials_.get();
+  config.field_trials = &env_.field_trials();
 
   transport_controller_.reset(new JsepTransportController(
       network_thread(), port_allocator_.get(),
@@ -1685,7 +1664,7 @@ void PeerConnection::AddIceCandidate(
     std::function<void(RTCError)> callback) {
   RTC_DCHECK_RUN_ON(signaling_thread());
   sdp_handler_->AddIceCandidate(std::move(candidate),
-                                [this, callback](webrtc::RTCError result) {
+                                [this, callback](RTCError result) {
                                   ClearStatsCache();
                                   callback(result);
                                 });
@@ -1789,7 +1768,7 @@ bool PeerConnection::StartRtcEventLog(
     std::unique_ptr<RtcEventLogOutput> output) {
   int64_t output_period_ms = 5000;
   if (trials().IsDisabled("WebRTC-RtcEventLogNewFormat")) {
-    output_period_ms = webrtc::RtcEventLog::kImmediateOutput;
+    output_period_ms = RtcEventLog::kImmediateOutput;
   }
   return StartRtcEventLog(std::move(output), output_period_ms);
 }
@@ -1931,8 +1910,7 @@ void PeerConnection::Close() {
     RTC_DCHECK_RUN_ON(worker_thread());
     worker_thread_safety_->SetNotAlive();
     call_.reset();
-    // The event log must outlive call (and any other object that uses it).
-    event_log_.reset();
+    StopRtcEventLog_w();
   });
   ReportUsagePattern();
 
@@ -2029,13 +2007,13 @@ void PeerConnection::ReportFirstConnectUsageMetrics() {
   RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.ProvisionalAnswer", pranswer,
                             kProvisionalAnswerMax);
 
-  // Record the number of valid / invalid ice-ufrag. We do allow certain
-  // non-spec ice-char for backward-compat reasons. At this point we know
-  // that the ufrag/pwd consists of a valid ice-char or one of the four
-  // not allowed characters since we have passed the IsIceChar check done
-  // by the p2p transport description on setRemoteDescription calls.
   auto transport_infos = remote_description()->description()->transport_infos();
-  if (transport_infos.size() > 0) {
+  if (!transport_infos.empty()) {
+    // Record the number of valid / invalid ice-ufrag. We do allow certain
+    // non-spec ice-char for backward-compat reasons. At this point we know
+    // that the ufrag/pwd consists of a valid ice-char or one of the four
+    // not allowed characters since we have passed the IsIceChar check done
+    // by the p2p transport description on setRemoteDescription calls.
     auto ice_parameters = transport_infos[0].description.GetIceParameters();
     auto is_invalid_char = [](char c) {
       return c == '-' || c == '=' || c == '#' || c == '_';
@@ -2047,6 +2025,16 @@ void PeerConnection::ReportFirstConnectUsageMetrics() {
     RTC_HISTOGRAM_BOOLEAN(
         "WebRTC.PeerConnection.ValidIceChars",
         !(isUsingInvalidIceCharInUfrag || isUsingInvalidIceCharInPwd));
+
+    // Record whether the hash algorithm of the first transport's
+    // DTLS fingerprint is still using SHA-1.
+    if (transport_infos[0].description.identity_fingerprint) {
+      RTC_HISTOGRAM_BOOLEAN(
+          "WebRTC.PeerConnection.DtlsFingerprintLegacySha1",
+          absl::EqualsIgnoreCase(
+              transport_infos[0].description.identity_fingerprint->algorithm,
+              "sha-1"));
+    }
   }
 
   // Record RtcpMuxPolicy setting.
@@ -2222,7 +2210,7 @@ bool PeerConnection::ReconfigurePortAllocator_n(
     IceTransportsType type,
     int candidate_pool_size,
     PortPrunePolicy turn_port_prune_policy,
-    webrtc::TurnCustomizer* turn_customizer,
+    TurnCustomizer* turn_customizer,
     absl::optional<int> stun_candidate_keepalive_interval,
     bool have_local_description) {
   RTC_DCHECK_RUN_ON(network_thread());
@@ -2245,17 +2233,15 @@ bool PeerConnection::StartRtcEventLog_w(
     std::unique_ptr<RtcEventLogOutput> output,
     int64_t output_period_ms) {
   RTC_DCHECK_RUN_ON(worker_thread());
-  if (!event_log_) {
+  if (!worker_thread_safety_->alive()) {
     return false;
   }
-  return event_log_->StartLogging(std::move(output), output_period_ms);
+  return env_.event_log().StartLogging(std::move(output), output_period_ms);
 }
 
 void PeerConnection::StopRtcEventLog_w() {
   RTC_DCHECK_RUN_ON(worker_thread());
-  if (event_log_) {
-    event_log_->StopLogging();
-  }
+  env_.event_log().StopLogging();
 }
 
 absl::optional<rtc::SSLRole> PeerConnection::GetSctpSslRole_n() {
diff --git a/third_party/libwebrtc/pc/peer_connection.h b/third_party/libwebrtc/pc/peer_connection.h
index ea1a9d9d90..e6037a2698 100644
--- a/third_party/libwebrtc/pc/peer_connection.h
+++ b/third_party/libwebrtc/pc/peer_connection.h
@@ -27,6 +27,7 @@
 #include "api/crypto/crypto_options.h"
 #include "api/data_channel_interface.h"
 #include "api/dtls_transport_interface.h"
+#include "api/environment/environment.h"
 #include "api/field_trials_view.h"
 #include "api/ice_transport_interface.h"
 #include "api/jsep.h"
@@ -110,9 +111,9 @@ class PeerConnection : public PeerConnectionInternal,
   // Note that the function takes ownership of dependencies, and will
   // either use them or release them, whether it succeeds or fails.
   static RTCErrorOr<rtc::scoped_refptr<PeerConnection>> Create(
+      const Environment& env,
       rtc::scoped_refptr<ConnectionContext> context,
       const PeerConnectionFactoryInterface::Options& options,
-      std::unique_ptr<RtcEventLog> event_log,
       std::unique_ptr<Call> call,
       const PeerConnectionInterface::RTCConfiguration& configuration,
       PeerConnectionDependencies dependencies);
@@ -163,7 +164,7 @@ class PeerConnection : public PeerConnectionInternal,
       const DataChannelInit* config) override;
   // WARNING: LEGACY. See peerconnectioninterface.h
   bool GetStats(StatsObserver* observer,
-                webrtc::MediaStreamTrackInterface* track,
+                MediaStreamTrackInterface* track,
                 StatsOutputLevel level) override;
   // Spec-complaint GetStats(). See peerconnectioninterface.h
   void GetStats(RTCStatsCollectorCallback* callback) override;
@@ -428,7 +429,7 @@ class PeerConnection : public PeerConnectionInternal,
   void TeardownDataChannelTransport_n(RTCError error)
       RTC_RUN_ON(network_thread());
 
-  const FieldTrialsView& trials() const override { return *trials_; }
+  const FieldTrialsView& trials() const override { return env_.field_trials(); }
 
   bool ConfiguredForMedia() const;
 
@@ -441,10 +442,10 @@ class PeerConnection : public PeerConnectionInternal,
 
  protected:
   // Available for rtc::scoped_refptr creation
-  PeerConnection(rtc::scoped_refptr<ConnectionContext> context,
+  PeerConnection(const Environment& env,
+                 rtc::scoped_refptr<ConnectionContext> context,
                  const PeerConnectionFactoryInterface::Options& options,
                  bool is_unified_plan,
-                 std::unique_ptr<RtcEventLog> event_log,
                  std::unique_ptr<Call> call,
                  PeerConnectionDependencies& dependencies,
                  bool dtls_enabled);
@@ -510,7 +511,7 @@ class PeerConnection : public PeerConnectionInternal,
       IceTransportsType type,
       int candidate_pool_size,
       PortPrunePolicy turn_port_prune_policy,
-      webrtc::TurnCustomizer* turn_customizer,
+      TurnCustomizer* turn_customizer,
       absl::optional<int> stun_candidate_keepalive_interval,
       bool have_local_description);
 
@@ -597,26 +598,14 @@ class PeerConnection : public PeerConnectionInternal,
   std::function<void(const RtpPacketReceived& parsed_packet)>
   InitializeUnDemuxablePacketHandler();
 
+  const Environment env_;
   const rtc::scoped_refptr<ConnectionContext> context_;
-  // Field trials active for this PeerConnection is the first of:
-  // a) Specified in PeerConnectionDependencies (owned).
-  // b) Accessed via ConnectionContext (e.g PeerConnectionFactoryDependencies>
-  // c) Created as Default (FieldTrialBasedConfig).
-  const webrtc::AlwaysValidPointer<const FieldTrialsView, FieldTrialBasedConfig>
-      trials_;
   const PeerConnectionFactoryInterface::Options options_;
   PeerConnectionObserver* observer_ RTC_GUARDED_BY(signaling_thread()) =
       nullptr;
 
   const bool is_unified_plan_;
 
-  // The EventLog needs to outlive `call_` (and any other object that uses it).
-  std::unique_ptr<RtcEventLog> event_log_ RTC_GUARDED_BY(worker_thread());
-
-  // Points to the same thing as `event_log_`. Since it's const, we may read the
-  // pointer (but not touch the object) from any thread.
-  RtcEventLog* const event_log_ptr_ RTC_PT_GUARDED_BY(worker_thread());
-
   IceConnectionState ice_connection_state_ RTC_GUARDED_BY(signaling_thread()) =
       kIceConnectionNew;
   PeerConnectionInterface::IceConnectionState standardized_ice_connection_state_
@@ -634,7 +623,7 @@ class PeerConnection : public PeerConnectionInternal,
   std::unique_ptr<cricket::PortAllocator>
       port_allocator_;  // TODO(bugs.webrtc.org/9987): Accessed on both
                         // signaling and network thread.
-  const std::unique_ptr<webrtc::IceTransportFactory>
+  const std::unique_ptr<IceTransportFactory>
       ice_transport_factory_;  // TODO(bugs.webrtc.org/9987): Accessed on the
                                // signaling thread but the underlying raw
                                // pointer is given to
diff --git a/third_party/libwebrtc/pc/peer_connection_crypto_unittest.cc b/third_party/libwebrtc/pc/peer_connection_crypto_unittest.cc
index dc350b2be0..a65988ab05 100644
--- a/third_party/libwebrtc/pc/peer_connection_crypto_unittest.cc
+++ b/third_party/libwebrtc/pc/peer_connection_crypto_unittest.cc
@@ -162,7 +162,7 @@ class PeerConnectionCryptoBaseTest : public ::testing::Test {
     return transport_info->description.connection_role;
   }
 
-  webrtc::test::ScopedKeyValueConfig field_trials_;
+  test::ScopedKeyValueConfig field_trials_;
   std::unique_ptr<rtc::VirtualSocketServer> vss_;
   rtc::AutoSocketServerThread main_;
   rtc::scoped_refptr<PeerConnectionFactoryInterface> pc_factory_;
diff --git a/third_party/libwebrtc/pc/peer_connection_data_channel_unittest.cc b/third_party/libwebrtc/pc/peer_connection_data_channel_unittest.cc
index 3bb2088866..8682cfa9fb 100644
--- a/third_party/libwebrtc/pc/peer_connection_data_channel_unittest.cc
+++ b/third_party/libwebrtc/pc/peer_connection_data_channel_unittest.cc
@@ -15,7 +15,6 @@
 #include <vector>
 
 #include "absl/types/optional.h"
-#include "api/call/call_factory_interface.h"
 #include "api/jsep.h"
 #include "api/media_types.h"
 #include "api/peer_connection_interface.h"
@@ -24,8 +23,6 @@
 #include "api/task_queue/default_task_queue_factory.h"
 #include "api/task_queue/task_queue_factory.h"
 #include "api/transport/sctp_transport_factory_interface.h"
-#include "media/base/fake_media_engine.h"
-#include "media/base/media_engine.h"
 #include "p2p/base/p2p_constants.h"
 #include "p2p/base/port_allocator.h"
 #include "pc/media_session.h"
@@ -35,6 +32,7 @@
 #include "pc/sctp_transport.h"
 #include "pc/sdp_utils.h"
 #include "pc/session_description.h"
+#include "pc/test/enable_fake_media.h"
 #include "pc/test/mock_peer_connection_observers.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/logging.h"
@@ -64,8 +62,7 @@ PeerConnectionFactoryDependencies CreatePeerConnectionFactoryDependencies() {
   deps.worker_thread = rtc::Thread::Current();
   deps.signaling_thread = rtc::Thread::Current();
   deps.task_queue_factory = CreateDefaultTaskQueueFactory();
-  deps.media_engine = std::make_unique<cricket::FakeMediaEngine>();
-  deps.call_factory = CreateCallFactory();
+  EnableFakeMedia(deps);
   deps.sctp_factory = std::make_unique<FakeSctpTransportFactory>();
   return deps;
 }
diff --git a/third_party/libwebrtc/pc/peer_connection_encodings_integrationtest.cc b/third_party/libwebrtc/pc/peer_connection_encodings_integrationtest.cc
index c7181c53ae..ae238671c2 100644
--- a/third_party/libwebrtc/pc/peer_connection_encodings_integrationtest.cc
+++ b/third_party/libwebrtc/pc/peer_connection_encodings_integrationtest.cc
@@ -77,18 +77,17 @@ struct StringParamToString {
 // RTX, RED and FEC are reliability mechanisms used in combinations with other
 // codecs, but are not themselves a specific codec. Typically you don't want to
 // filter these out of the list of codec preferences.
-bool IsReliabilityMechanism(const webrtc::RtpCodecCapability& codec) {
+bool IsReliabilityMechanism(const RtpCodecCapability& codec) {
   return absl::EqualsIgnoreCase(codec.name, cricket::kRtxCodecName) ||
          absl::EqualsIgnoreCase(codec.name, cricket::kRedCodecName) ||
          absl::EqualsIgnoreCase(codec.name, cricket::kUlpfecCodecName);
 }
 
 std::string GetCurrentCodecMimeType(
-    rtc::scoped_refptr<const webrtc::RTCStatsReport> report,
-    const webrtc::RTCOutboundRtpStreamStats& outbound_rtp) {
+    rtc::scoped_refptr<const RTCStatsReport> report,
+    const RTCOutboundRtpStreamStats& outbound_rtp) {
   return outbound_rtp.codec_id.is_defined()
-             ? *report->GetAs<webrtc::RTCCodecStats>(*outbound_rtp.codec_id)
-                    ->mime_type
+             ? *report->GetAs<RTCCodecStats>(*outbound_rtp.codec_id)->mime_type
              : "";
 }
 
@@ -98,8 +97,8 @@ struct RidAndResolution {
   uint32_t height;
 };
 
-const webrtc::RTCOutboundRtpStreamStats* FindOutboundRtpByRid(
-    const std::vector<const webrtc::RTCOutboundRtpStreamStats*>& outbound_rtps,
+const RTCOutboundRtpStreamStats* FindOutboundRtpByRid(
+    const std::vector<const RTCOutboundRtpStreamStats*>& outbound_rtps,
     const absl::string_view& rid) {
   for (const auto* outbound_rtp : outbound_rtps) {
     if (outbound_rtp->rid.is_defined() && *outbound_rtp->rid == rid) {
@@ -121,8 +120,8 @@ class PeerConnectionEncodingsIntegrationTest : public ::testing::Test {
   rtc::scoped_refptr<PeerConnectionTestWrapper> CreatePc() {
     auto pc_wrapper = rtc::make_ref_counted<PeerConnectionTestWrapper>(
         "pc", &pss_, background_thread_.get(), background_thread_.get());
-    pc_wrapper->CreatePc({}, webrtc::CreateBuiltinAudioEncoderFactory(),
-                         webrtc::CreateBuiltinAudioDecoderFactory());
+    pc_wrapper->CreatePc({}, CreateBuiltinAudioEncoderFactory(),
+                         CreateBuiltinAudioDecoderFactory());
     return pc_wrapper;
   }
 
@@ -130,10 +129,9 @@ class PeerConnectionEncodingsIntegrationTest : public ::testing::Test {
       rtc::scoped_refptr<PeerConnectionTestWrapper> local,
       rtc::scoped_refptr<PeerConnectionTestWrapper> remote,
       std::vector<cricket::SimulcastLayer> init_layers) {
-    rtc::scoped_refptr<webrtc::MediaStreamInterface> stream =
-        local->GetUserMedia(
-            /*audio=*/false, cricket::AudioOptions(), /*video=*/true,
-            {.width = 1280, .height = 720});
+    rtc::scoped_refptr<MediaStreamInterface> stream = local->GetUserMedia(
+        /*audio=*/false, cricket::AudioOptions(), /*video=*/true,
+        {.width = 1280, .height = 720});
     rtc::scoped_refptr<VideoTrackInterface> track = stream->GetVideoTracks()[0];
 
     RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
@@ -973,8 +971,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest,
       local_pc_wrapper->pc()->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
   rtc::scoped_refptr<RtpTransceiverInterface> audio_transceiver =
       transceiver_or_error.MoveValue();
-  webrtc::RtpParameters parameters =
-      audio_transceiver->sender()->GetParameters();
+  RtpParameters parameters = audio_transceiver->sender()->GetParameters();
   EXPECT_FALSE(parameters.encodings[0].codec.has_value());
 }
 
@@ -986,8 +983,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest,
       local_pc_wrapper->pc()->AddTransceiver(cricket::MEDIA_TYPE_VIDEO);
   rtc::scoped_refptr<RtpTransceiverInterface> video_transceiver =
       transceiver_or_error.MoveValue();
-  webrtc::RtpParameters parameters =
-      video_transceiver->sender()->GetParameters();
+  RtpParameters parameters = video_transceiver->sender()->GetParameters();
   EXPECT_FALSE(parameters.encodings[0].codec.has_value());
 }
 
@@ -997,19 +993,19 @@ TEST_F(PeerConnectionEncodingsIntegrationTest,
   rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc();
   ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper);
 
-  rtc::scoped_refptr<webrtc::MediaStreamInterface> stream =
+  rtc::scoped_refptr<MediaStreamInterface> stream =
       local_pc_wrapper->GetUserMedia(
           /*audio=*/true, {}, /*video=*/false, {});
   rtc::scoped_refptr<AudioTrackInterface> track = stream->GetAudioTracks()[0];
 
-  absl::optional<webrtc::RtpCodecCapability> pcmu =
+  absl::optional<RtpCodecCapability> pcmu =
       local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_AUDIO,
                                                    "pcmu");
   ASSERT_TRUE(pcmu);
 
-  webrtc::RtpTransceiverInit init;
-  init.direction = webrtc::RtpTransceiverDirection::kSendOnly;
-  webrtc::RtpEncodingParameters encoding_parameters;
+  RtpTransceiverInit init;
+  init.direction = RtpTransceiverDirection::kSendOnly;
+  RtpEncodingParameters encoding_parameters;
   encoding_parameters.codec = pcmu;
   init.send_encodings.push_back(encoding_parameters);
 
@@ -1017,8 +1013,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest,
       local_pc_wrapper->pc()->AddTransceiver(track, init);
   rtc::scoped_refptr<RtpTransceiverInterface> audio_transceiver =
       transceiver_or_error.MoveValue();
-  webrtc::RtpParameters parameters =
-      audio_transceiver->sender()->GetParameters();
+  RtpParameters parameters = audio_transceiver->sender()->GetParameters();
   EXPECT_EQ(*parameters.encodings[0].codec, *pcmu);
 
   NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper);
@@ -1039,19 +1034,19 @@ TEST_F(PeerConnectionEncodingsIntegrationTest,
   rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc();
   ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper);
 
-  rtc::scoped_refptr<webrtc::MediaStreamInterface> stream =
+  rtc::scoped_refptr<MediaStreamInterface> stream =
       local_pc_wrapper->GetUserMedia(
           /*audio=*/false, {}, /*video=*/true, {.width = 1280, .height = 720});
   rtc::scoped_refptr<VideoTrackInterface> track = stream->GetVideoTracks()[0];
 
-  absl::optional<webrtc::RtpCodecCapability> vp9 =
+  absl::optional<RtpCodecCapability> vp9 =
       local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO,
                                                    "vp9");
   ASSERT_TRUE(vp9);
 
-  webrtc::RtpTransceiverInit init;
-  init.direction = webrtc::RtpTransceiverDirection::kSendOnly;
-  webrtc::RtpEncodingParameters encoding_parameters;
+  RtpTransceiverInit init;
+  init.direction = RtpTransceiverDirection::kSendOnly;
+  RtpEncodingParameters encoding_parameters;
   encoding_parameters.codec = vp9;
   encoding_parameters.scalability_mode = "L3T3";
   init.send_encodings.push_back(encoding_parameters);
@@ -1060,8 +1055,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest,
       local_pc_wrapper->pc()->AddTransceiver(track, init);
   rtc::scoped_refptr<RtpTransceiverInterface> audio_transceiver =
       transceiver_or_error.MoveValue();
-  webrtc::RtpParameters parameters =
-      audio_transceiver->sender()->GetParameters();
+  RtpParameters parameters = audio_transceiver->sender()->GetParameters();
   EXPECT_EQ(*parameters.encodings[0].codec, *vp9);
 
   NegotiateWithSimulcastTweaks(local_pc_wrapper, remote_pc_wrapper);
@@ -1087,20 +1081,19 @@ TEST_F(PeerConnectionEncodingsIntegrationTest,
   rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc();
   ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper);
 
-  rtc::scoped_refptr<webrtc::MediaStreamInterface> stream =
+  rtc::scoped_refptr<MediaStreamInterface> stream =
       local_pc_wrapper->GetUserMedia(
           /*audio=*/true, {}, /*video=*/false, {});
   rtc::scoped_refptr<AudioTrackInterface> track = stream->GetAudioTracks()[0];
 
-  absl::optional<webrtc::RtpCodecCapability> pcmu =
+  absl::optional<RtpCodecCapability> pcmu =
       local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_AUDIO,
                                                    "pcmu");
 
   auto transceiver_or_error = local_pc_wrapper->pc()->AddTransceiver(track);
   rtc::scoped_refptr<RtpTransceiverInterface> audio_transceiver =
       transceiver_or_error.MoveValue();
-  webrtc::RtpParameters parameters =
-      audio_transceiver->sender()->GetParameters();
+  RtpParameters parameters = audio_transceiver->sender()->GetParameters();
   parameters.encodings[0].codec = pcmu;
   EXPECT_TRUE(audio_transceiver->sender()->SetParameters(parameters).ok());
 
@@ -1125,12 +1118,12 @@ TEST_F(PeerConnectionEncodingsIntegrationTest,
   rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc();
   ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper);
 
-  rtc::scoped_refptr<webrtc::MediaStreamInterface> stream =
+  rtc::scoped_refptr<MediaStreamInterface> stream =
       local_pc_wrapper->GetUserMedia(
           /*audio=*/true, {}, /*video=*/false, {});
   rtc::scoped_refptr<AudioTrackInterface> track = stream->GetAudioTracks()[0];
 
-  absl::optional<webrtc::RtpCodecCapability> pcmu =
+  absl::optional<RtpCodecCapability> pcmu =
       local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_AUDIO,
                                                    "pcmu");
 
@@ -1150,8 +1143,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest,
   EXPECT_STRCASENE(("audio/" + pcmu->name).c_str(), codec_name.c_str());
   std::string last_codec_id = outbound_rtps[0]->codec_id.value();
 
-  webrtc::RtpParameters parameters =
-      audio_transceiver->sender()->GetParameters();
+  RtpParameters parameters = audio_transceiver->sender()->GetParameters();
   parameters.encodings[0].codec = pcmu;
   EXPECT_TRUE(audio_transceiver->sender()->SetParameters(parameters).ok());
 
@@ -1174,20 +1166,19 @@ TEST_F(PeerConnectionEncodingsIntegrationTest,
   rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc();
   ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper);
 
-  rtc::scoped_refptr<webrtc::MediaStreamInterface> stream =
+  rtc::scoped_refptr<MediaStreamInterface> stream =
       local_pc_wrapper->GetUserMedia(
           /*audio=*/false, {}, /*video=*/true, {.width = 1280, .height = 720});
   rtc::scoped_refptr<VideoTrackInterface> track = stream->GetVideoTracks()[0];
 
-  absl::optional<webrtc::RtpCodecCapability> vp9 =
+  absl::optional<RtpCodecCapability> vp9 =
       local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO,
                                                    "vp9");
 
   auto transceiver_or_error = local_pc_wrapper->pc()->AddTransceiver(track);
   rtc::scoped_refptr<RtpTransceiverInterface> video_transceiver =
       transceiver_or_error.MoveValue();
-  webrtc::RtpParameters parameters =
-      video_transceiver->sender()->GetParameters();
+  RtpParameters parameters = video_transceiver->sender()->GetParameters();
   parameters.encodings[0].codec = vp9;
   parameters.encodings[0].scalability_mode = "L3T3";
   EXPECT_TRUE(video_transceiver->sender()->SetParameters(parameters).ok());
@@ -1218,12 +1209,12 @@ TEST_F(PeerConnectionEncodingsIntegrationTest,
   rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc();
   ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper);
 
-  rtc::scoped_refptr<webrtc::MediaStreamInterface> stream =
+  rtc::scoped_refptr<MediaStreamInterface> stream =
       local_pc_wrapper->GetUserMedia(
           /*audio=*/false, {}, /*video=*/true, {.width = 1280, .height = 720});
   rtc::scoped_refptr<VideoTrackInterface> track = stream->GetVideoTracks()[0];
 
-  absl::optional<webrtc::RtpCodecCapability> vp9 =
+  absl::optional<RtpCodecCapability> vp9 =
       local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO,
                                                    "vp9");
 
@@ -1243,8 +1234,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest,
   EXPECT_STRCASENE(("audio/" + vp9->name).c_str(), codec_name.c_str());
   std::string last_codec_id = outbound_rtps[0]->codec_id.value();
 
-  webrtc::RtpParameters parameters =
-      video_transceiver->sender()->GetParameters();
+  RtpParameters parameters = video_transceiver->sender()->GetParameters();
   parameters.encodings[0].codec = vp9;
   parameters.encodings[0].scalability_mode = "L3T3";
   EXPECT_TRUE(video_transceiver->sender()->SetParameters(parameters).ok());
@@ -1269,15 +1259,15 @@ TEST_F(PeerConnectionEncodingsIntegrationTest,
        AddTransceiverRejectsUnknownCodecParameterAudio) {
   rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc();
 
-  webrtc::RtpCodec dummy_codec;
+  RtpCodec dummy_codec;
   dummy_codec.kind = cricket::MEDIA_TYPE_AUDIO;
   dummy_codec.name = "FOOBAR";
   dummy_codec.clock_rate = 90000;
   dummy_codec.num_channels = 2;
 
-  webrtc::RtpTransceiverInit init;
-  init.direction = webrtc::RtpTransceiverDirection::kSendOnly;
-  webrtc::RtpEncodingParameters encoding_parameters;
+  RtpTransceiverInit init;
+  init.direction = RtpTransceiverDirection::kSendOnly;
+  RtpEncodingParameters encoding_parameters;
   encoding_parameters.codec = dummy_codec;
   init.send_encodings.push_back(encoding_parameters);
 
@@ -1292,14 +1282,14 @@ TEST_F(PeerConnectionEncodingsIntegrationTest,
        AddTransceiverRejectsUnknownCodecParameterVideo) {
   rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc();
 
-  webrtc::RtpCodec dummy_codec;
+  RtpCodec dummy_codec;
   dummy_codec.kind = cricket::MEDIA_TYPE_VIDEO;
   dummy_codec.name = "FOOBAR";
   dummy_codec.clock_rate = 90000;
 
-  webrtc::RtpTransceiverInit init;
-  init.direction = webrtc::RtpTransceiverDirection::kSendOnly;
-  webrtc::RtpEncodingParameters encoding_parameters;
+  RtpTransceiverInit init;
+  init.direction = RtpTransceiverDirection::kSendOnly;
+  RtpEncodingParameters encoding_parameters;
   encoding_parameters.codec = dummy_codec;
   init.send_encodings.push_back(encoding_parameters);
 
@@ -1314,7 +1304,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest,
        SetParametersRejectsUnknownCodecParameterAudio) {
   rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc();
 
-  webrtc::RtpCodec dummy_codec;
+  RtpCodec dummy_codec;
   dummy_codec.kind = cricket::MEDIA_TYPE_AUDIO;
   dummy_codec.name = "FOOBAR";
   dummy_codec.clock_rate = 90000;
@@ -1326,8 +1316,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest,
   rtc::scoped_refptr<RtpTransceiverInterface> audio_transceiver =
       transceiver_or_error.MoveValue();
 
-  webrtc::RtpParameters parameters =
-      audio_transceiver->sender()->GetParameters();
+  RtpParameters parameters = audio_transceiver->sender()->GetParameters();
   parameters.encodings[0].codec = dummy_codec;
   RTCError error = audio_transceiver->sender()->SetParameters(parameters);
   EXPECT_EQ(error.type(), RTCErrorType::INVALID_MODIFICATION);
@@ -1337,7 +1326,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest,
        SetParametersRejectsUnknownCodecParameterVideo) {
   rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc();
 
-  webrtc::RtpCodec dummy_codec;
+  RtpCodec dummy_codec;
   dummy_codec.kind = cricket::MEDIA_TYPE_VIDEO;
   dummy_codec.name = "FOOBAR";
   dummy_codec.clock_rate = 90000;
@@ -1348,8 +1337,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest,
   rtc::scoped_refptr<RtpTransceiverInterface> video_transceiver =
       transceiver_or_error.MoveValue();
 
-  webrtc::RtpParameters parameters =
-      video_transceiver->sender()->GetParameters();
+  RtpParameters parameters = video_transceiver->sender()->GetParameters();
   parameters.encodings[0].codec = dummy_codec;
   RTCError error = video_transceiver->sender()->SetParameters(parameters);
   EXPECT_EQ(error.type(), RTCErrorType::INVALID_MODIFICATION);
@@ -1359,12 +1347,12 @@ TEST_F(PeerConnectionEncodingsIntegrationTest,
        SetParametersRejectsNonPreferredCodecParameterAudio) {
   rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc();
 
-  absl::optional<webrtc::RtpCodecCapability> opus =
+  absl::optional<RtpCodecCapability> opus =
       local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_AUDIO,
                                                    "opus");
   ASSERT_TRUE(opus);
 
-  std::vector<webrtc::RtpCodecCapability> not_opus_codecs =
+  std::vector<RtpCodecCapability> not_opus_codecs =
       local_pc_wrapper->pc_factory()
           ->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_AUDIO)
           .codecs;
@@ -1382,8 +1370,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest,
       transceiver_or_error.MoveValue();
   ASSERT_TRUE(audio_transceiver->SetCodecPreferences(not_opus_codecs).ok());
 
-  webrtc::RtpParameters parameters =
-      audio_transceiver->sender()->GetParameters();
+  RtpParameters parameters = audio_transceiver->sender()->GetParameters();
   parameters.encodings[0].codec = opus;
   RTCError error = audio_transceiver->sender()->SetParameters(parameters);
   EXPECT_EQ(error.type(), RTCErrorType::INVALID_MODIFICATION);
@@ -1393,12 +1380,12 @@ TEST_F(PeerConnectionEncodingsIntegrationTest,
        SetParametersRejectsNonPreferredCodecParameterVideo) {
   rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc();
 
-  absl::optional<webrtc::RtpCodecCapability> vp8 =
+  absl::optional<RtpCodecCapability> vp8 =
       local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO,
                                                    "vp8");
   ASSERT_TRUE(vp8);
 
-  std::vector<webrtc::RtpCodecCapability> not_vp8_codecs =
+  std::vector<RtpCodecCapability> not_vp8_codecs =
       local_pc_wrapper->pc_factory()
           ->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_VIDEO)
           .codecs;
@@ -1416,8 +1403,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest,
       transceiver_or_error.MoveValue();
   ASSERT_TRUE(video_transceiver->SetCodecPreferences(not_vp8_codecs).ok());
 
-  webrtc::RtpParameters parameters =
-      video_transceiver->sender()->GetParameters();
+  RtpParameters parameters = video_transceiver->sender()->GetParameters();
   parameters.encodings[0].codec = vp8;
   RTCError error = video_transceiver->sender()->SetParameters(parameters);
   EXPECT_EQ(error.type(), RTCErrorType::INVALID_MODIFICATION);
@@ -1429,12 +1415,12 @@ TEST_F(PeerConnectionEncodingsIntegrationTest,
   rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc();
   ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper);
 
-  absl::optional<webrtc::RtpCodecCapability> opus =
+  absl::optional<RtpCodecCapability> opus =
       local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_AUDIO,
                                                    "opus");
   ASSERT_TRUE(opus);
 
-  std::vector<webrtc::RtpCodecCapability> not_opus_codecs =
+  std::vector<RtpCodecCapability> not_opus_codecs =
       local_pc_wrapper->pc_factory()
           ->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_AUDIO)
           .codecs;
@@ -1456,8 +1442,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest,
   local_pc_wrapper->WaitForConnection();
   remote_pc_wrapper->WaitForConnection();
 
-  webrtc::RtpParameters parameters =
-      audio_transceiver->sender()->GetParameters();
+  RtpParameters parameters = audio_transceiver->sender()->GetParameters();
   parameters.encodings[0].codec = opus;
   RTCError error = audio_transceiver->sender()->SetParameters(parameters);
   EXPECT_EQ(error.type(), RTCErrorType::INVALID_MODIFICATION);
@@ -1469,12 +1454,12 @@ TEST_F(PeerConnectionEncodingsIntegrationTest,
   rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc();
   ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper);
 
-  absl::optional<webrtc::RtpCodecCapability> opus =
+  absl::optional<RtpCodecCapability> opus =
       local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_AUDIO,
                                                    "opus");
   ASSERT_TRUE(opus);
 
-  std::vector<webrtc::RtpCodecCapability> not_opus_codecs =
+  std::vector<RtpCodecCapability> not_opus_codecs =
       local_pc_wrapper->pc_factory()
           ->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_AUDIO)
           .codecs;
@@ -1519,8 +1504,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest,
   local_pc_wrapper->WaitForConnection();
   remote_pc_wrapper->WaitForConnection();
 
-  webrtc::RtpParameters parameters =
-      audio_transceiver->sender()->GetParameters();
+  RtpParameters parameters = audio_transceiver->sender()->GetParameters();
   parameters.encodings[0].codec = opus;
   RTCError error = audio_transceiver->sender()->SetParameters(parameters);
   EXPECT_EQ(error.type(), RTCErrorType::INVALID_MODIFICATION);
@@ -1532,12 +1516,12 @@ TEST_F(PeerConnectionEncodingsIntegrationTest,
   rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc();
   ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper);
 
-  absl::optional<webrtc::RtpCodecCapability> vp8 =
+  absl::optional<RtpCodecCapability> vp8 =
       local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO,
                                                    "vp8");
   ASSERT_TRUE(vp8);
 
-  std::vector<webrtc::RtpCodecCapability> not_vp8_codecs =
+  std::vector<RtpCodecCapability> not_vp8_codecs =
       local_pc_wrapper->pc_factory()
           ->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_VIDEO)
           .codecs;
@@ -1559,8 +1543,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest,
   local_pc_wrapper->WaitForConnection();
   remote_pc_wrapper->WaitForConnection();
 
-  webrtc::RtpParameters parameters =
-      video_transceiver->sender()->GetParameters();
+  RtpParameters parameters = video_transceiver->sender()->GetParameters();
   parameters.encodings[0].codec = vp8;
   RTCError error = video_transceiver->sender()->SetParameters(parameters);
   EXPECT_EQ(error.type(), RTCErrorType::INVALID_MODIFICATION);
@@ -1572,12 +1555,12 @@ TEST_F(PeerConnectionEncodingsIntegrationTest,
   rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc();
   ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper);
 
-  absl::optional<webrtc::RtpCodecCapability> vp8 =
+  absl::optional<RtpCodecCapability> vp8 =
       local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO,
                                                    "vp8");
   ASSERT_TRUE(vp8);
 
-  std::vector<webrtc::RtpCodecCapability> not_vp8_codecs =
+  std::vector<RtpCodecCapability> not_vp8_codecs =
       local_pc_wrapper->pc_factory()
           ->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_VIDEO)
           .codecs;
@@ -1622,8 +1605,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest,
   local_pc_wrapper->WaitForConnection();
   remote_pc_wrapper->WaitForConnection();
 
-  webrtc::RtpParameters parameters =
-      video_transceiver->sender()->GetParameters();
+  RtpParameters parameters = video_transceiver->sender()->GetParameters();
   parameters.encodings[0].codec = vp8;
   RTCError error = video_transceiver->sender()->SetParameters(parameters);
   EXPECT_EQ(error.type(), RTCErrorType::INVALID_MODIFICATION);
@@ -1635,12 +1617,12 @@ TEST_F(PeerConnectionEncodingsIntegrationTest,
   rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc();
   ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper);
 
-  absl::optional<webrtc::RtpCodecCapability> opus =
+  absl::optional<RtpCodecCapability> opus =
       local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_AUDIO,
                                                    "opus");
   ASSERT_TRUE(opus);
 
-  std::vector<webrtc::RtpCodecCapability> not_opus_codecs =
+  std::vector<RtpCodecCapability> not_opus_codecs =
       local_pc_wrapper->pc_factory()
           ->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_AUDIO)
           .codecs;
@@ -1651,9 +1633,9 @@ TEST_F(PeerConnectionEncodingsIntegrationTest,
                      }),
       not_opus_codecs.end());
 
-  webrtc::RtpTransceiverInit init;
-  init.direction = webrtc::RtpTransceiverDirection::kSendOnly;
-  webrtc::RtpEncodingParameters encoding_parameters;
+  RtpTransceiverInit init;
+  init.direction = RtpTransceiverDirection::kSendOnly;
+  RtpEncodingParameters encoding_parameters;
   encoding_parameters.codec = opus;
   init.send_encodings.push_back(encoding_parameters);
 
@@ -1667,8 +1649,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest,
   local_pc_wrapper->WaitForConnection();
   remote_pc_wrapper->WaitForConnection();
 
-  webrtc::RtpParameters parameters =
-      audio_transceiver->sender()->GetParameters();
+  RtpParameters parameters = audio_transceiver->sender()->GetParameters();
   EXPECT_EQ(parameters.encodings[0].codec, opus);
 
   ASSERT_TRUE(audio_transceiver->SetCodecPreferences(not_opus_codecs).ok());
@@ -1684,24 +1665,24 @@ TEST_F(PeerConnectionEncodingsIntegrationTest,
   rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc();
   ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper);
 
-  std::vector<webrtc::RtpCodecCapability> send_codecs =
+  std::vector<RtpCodecCapability> send_codecs =
       local_pc_wrapper->pc_factory()
           ->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_AUDIO)
           .codecs;
 
-  absl::optional<webrtc::RtpCodecCapability> opus =
+  absl::optional<RtpCodecCapability> opus =
       local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_AUDIO,
                                                    "opus");
   ASSERT_TRUE(opus);
 
-  absl::optional<webrtc::RtpCodecCapability> red =
+  absl::optional<RtpCodecCapability> red =
       local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_AUDIO,
                                                    "red");
   ASSERT_TRUE(red);
 
-  webrtc::RtpTransceiverInit init;
-  init.direction = webrtc::RtpTransceiverDirection::kSendOnly;
-  webrtc::RtpEncodingParameters encoding_parameters;
+  RtpTransceiverInit init;
+  init.direction = RtpTransceiverDirection::kSendOnly;
+  RtpEncodingParameters encoding_parameters;
   encoding_parameters.codec = opus;
   init.send_encodings.push_back(encoding_parameters);
 
@@ -1720,8 +1701,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest,
   local_pc_wrapper->WaitForConnection();
   remote_pc_wrapper->WaitForConnection();
 
-  webrtc::RtpParameters parameters =
-      audio_transceiver->sender()->GetParameters();
+  RtpParameters parameters = audio_transceiver->sender()->GetParameters();
   EXPECT_EQ(parameters.encodings[0].codec, opus);
   EXPECT_EQ(parameters.codecs[0].payload_type, red->preferred_payload_type);
   EXPECT_EQ(parameters.codecs[0].name, red->name);
@@ -1743,14 +1723,14 @@ TEST_F(PeerConnectionEncodingsIntegrationTest,
        SetParametersRejectsScalabilityModeForSelectedCodec) {
   rtc::scoped_refptr<PeerConnectionTestWrapper> local_pc_wrapper = CreatePc();
 
-  absl::optional<webrtc::RtpCodecCapability> vp8 =
+  absl::optional<RtpCodecCapability> vp8 =
       local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO,
                                                    "vp8");
   ASSERT_TRUE(vp8);
 
-  webrtc::RtpTransceiverInit init;
-  init.direction = webrtc::RtpTransceiverDirection::kSendOnly;
-  webrtc::RtpEncodingParameters encoding_parameters;
+  RtpTransceiverInit init;
+  init.direction = RtpTransceiverDirection::kSendOnly;
+  RtpEncodingParameters encoding_parameters;
   encoding_parameters.codec = vp8;
   encoding_parameters.scalability_mode = "L1T3";
   init.send_encodings.push_back(encoding_parameters);
@@ -1761,8 +1741,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest,
   rtc::scoped_refptr<RtpTransceiverInterface> video_transceiver =
       transceiver_or_error.MoveValue();
 
-  webrtc::RtpParameters parameters =
-      video_transceiver->sender()->GetParameters();
+  RtpParameters parameters = video_transceiver->sender()->GetParameters();
   parameters.encodings[0].scalability_mode = "L3T3";
   RTCError error = video_transceiver->sender()->SetParameters(parameters);
   EXPECT_EQ(error.type(), RTCErrorType::INVALID_MODIFICATION);
@@ -1774,12 +1753,12 @@ TEST_F(PeerConnectionEncodingsIntegrationTest,
   rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc();
   ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper);
 
-  absl::optional<webrtc::RtpCodecCapability> vp8 =
+  absl::optional<RtpCodecCapability> vp8 =
       local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO,
                                                    "vp8");
   ASSERT_TRUE(vp8);
 
-  std::vector<webrtc::RtpCodecCapability> not_vp8_codecs =
+  std::vector<RtpCodecCapability> not_vp8_codecs =
       local_pc_wrapper->pc_factory()
           ->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_VIDEO)
           .codecs;
@@ -1790,9 +1769,9 @@ TEST_F(PeerConnectionEncodingsIntegrationTest,
                      }),
       not_vp8_codecs.end());
 
-  webrtc::RtpTransceiverInit init;
-  init.direction = webrtc::RtpTransceiverDirection::kSendOnly;
-  webrtc::RtpEncodingParameters encoding_parameters;
+  RtpTransceiverInit init;
+  init.direction = RtpTransceiverDirection::kSendOnly;
+  RtpEncodingParameters encoding_parameters;
   encoding_parameters.rid = "h";
   encoding_parameters.codec = vp8;
   encoding_parameters.scale_resolution_down_by = 2;
@@ -1811,8 +1790,7 @@ TEST_F(PeerConnectionEncodingsIntegrationTest,
   local_pc_wrapper->WaitForConnection();
   remote_pc_wrapper->WaitForConnection();
 
-  webrtc::RtpParameters parameters =
-      video_transceiver->sender()->GetParameters();
+  RtpParameters parameters = video_transceiver->sender()->GetParameters();
   ASSERT_EQ(parameters.encodings.size(), 2u);
   EXPECT_EQ(parameters.encodings[0].codec, vp8);
   EXPECT_EQ(parameters.encodings[1].codec, vp8);
@@ -1833,17 +1811,17 @@ TEST_F(PeerConnectionEncodingsIntegrationTest,
   rtc::scoped_refptr<PeerConnectionTestWrapper> remote_pc_wrapper = CreatePc();
   ExchangeIceCandidates(local_pc_wrapper, remote_pc_wrapper);
 
-  absl::optional<webrtc::RtpCodecCapability> vp8 =
+  absl::optional<RtpCodecCapability> vp8 =
       local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO,
                                                    "vp8");
   ASSERT_TRUE(vp8);
-  absl::optional<webrtc::RtpCodecCapability> vp9 =
+  absl::optional<RtpCodecCapability> vp9 =
       local_pc_wrapper->FindFirstSendCodecWithName(cricket::MEDIA_TYPE_VIDEO,
                                                    "vp9");
 
-  webrtc::RtpTransceiverInit init;
-  init.direction = webrtc::RtpTransceiverDirection::kSendOnly;
-  webrtc::RtpEncodingParameters encoding_parameters;
+  RtpTransceiverInit init;
+  init.direction = RtpTransceiverDirection::kSendOnly;
+  RtpEncodingParameters encoding_parameters;
   encoding_parameters.rid = "h";
   encoding_parameters.codec = vp8;
   encoding_parameters.scale_resolution_down_by = 2;
diff --git a/third_party/libwebrtc/pc/peer_connection_factory.cc b/third_party/libwebrtc/pc/peer_connection_factory.cc
index 81780cf51e..8ce44d374f 100644
--- a/third_party/libwebrtc/pc/peer_connection_factory.cc
+++ b/third_party/libwebrtc/pc/peer_connection_factory.cc
@@ -14,8 +14,8 @@
 #include <utility>
 
 #include "absl/strings/match.h"
-#include "api/async_resolver_factory.h"
-#include "api/call/call_factory_interface.h"
+#include "api/environment/environment.h"
+#include "api/environment/environment_factory.h"
 #include "api/fec_controller.h"
 #include "api/ice_transport_interface.h"
 #include "api/network_state_predictor.h"
@@ -27,13 +27,13 @@
 #include "call/audio_state.h"
 #include "call/rtp_transport_controller_send_factory.h"
 #include "media/base/media_engine.h"
-#include "p2p/base/basic_async_resolver_factory.h"
 #include "p2p/base/basic_packet_socket_factory.h"
 #include "p2p/base/default_ice_transport_factory.h"
 #include "p2p/base/port_allocator.h"
 #include "p2p/client/basic_port_allocator.h"
 #include "pc/audio_track.h"
 #include "pc/local_audio_source.h"
+#include "pc/media_factory.h"
 #include "pc/media_stream.h"
 #include "pc/media_stream_proxy.h"
 #include "pc/media_stream_track_proxy.h"
@@ -78,7 +78,10 @@ CreateModularPeerConnectionFactory(
 // Static
 rtc::scoped_refptr<PeerConnectionFactory> PeerConnectionFactory::Create(
     PeerConnectionFactoryDependencies dependencies) {
-  auto context = ConnectionContext::Create(&dependencies);
+  auto context = ConnectionContext::Create(
+      CreateEnvironment(std::move(dependencies.trials),
+                        std::move(dependencies.task_queue_factory)),
+      &dependencies);
   if (!context) {
     return nullptr;
   }
@@ -89,7 +92,6 @@ PeerConnectionFactory::PeerConnectionFactory(
     rtc::scoped_refptr<ConnectionContext> context,
     PeerConnectionFactoryDependencies* dependencies)
     : context_(context),
-      task_queue_factory_(std::move(dependencies->task_queue_factory)),
       event_log_factory_(std::move(dependencies->event_log_factory)),
       fec_controller_factory_(std::move(dependencies->fec_controller_factory)),
       network_state_predictor_factory_(
@@ -105,8 +107,12 @@ PeerConnectionFactory::PeerConnectionFactory(
 
 PeerConnectionFactory::PeerConnectionFactory(
     PeerConnectionFactoryDependencies dependencies)
-    : PeerConnectionFactory(ConnectionContext::Create(&dependencies),
-                            &dependencies) {}
+    : PeerConnectionFactory(
+          ConnectionContext::Create(
+              CreateEnvironment(std::move(dependencies.trials),
+                                std::move(dependencies.task_queue_factory)),
+              &dependencies),
+          &dependencies) {}
 
 PeerConnectionFactory::~PeerConnectionFactory() {
   RTC_DCHECK_RUN_ON(signaling_thread());
@@ -205,6 +211,23 @@ PeerConnectionFactory::CreatePeerConnectionOrError(
     PeerConnectionDependencies dependencies) {
   RTC_DCHECK_RUN_ON(signaling_thread());
 
+  EnvironmentFactory env_factory(context_->env());
+
+  // Field trials active for this PeerConnection is the first of:
+  // a) Specified in the PeerConnectionDependencies
+  // b) Specified in the PeerConnectionFactoryDependencies
+  // c) Created as default by the EnvironmentFactory.
+  env_factory.Set(std::move(dependencies.trials));
+
+  if (event_log_factory_ != nullptr) {
+    worker_thread()->BlockingCall([&] {
+      Environment env_for_rtc_event_log = env_factory.Create();
+      env_factory.Set(event_log_factory_->Create(env_for_rtc_event_log));
+    });
+  }
+
+  const Environment env = env_factory.Create();
+
   // Set internal defaults if optional dependencies are not set.
   if (!dependencies.cert_generator) {
     dependencies.cert_generator =
@@ -212,11 +235,10 @@ PeerConnectionFactory::CreatePeerConnectionOrError(
                                                        network_thread());
   }
   if (!dependencies.allocator) {
-    const FieldTrialsView* trials =
-        dependencies.trials ? dependencies.trials.get() : &field_trials();
     dependencies.allocator = std::make_unique<cricket::BasicPortAllocator>(
         context_->default_network_manager(), context_->default_socket_factory(),
-        configuration.turn_customizer, /*relay_port_factory=*/nullptr, trials);
+        configuration.turn_customizer, /*relay_port_factory=*/nullptr,
+        &env.field_trials());
     dependencies.allocator->SetPortRange(
         configuration.port_allocator_config.min_port,
         configuration.port_allocator_config.max_port);
@@ -232,19 +254,13 @@ PeerConnectionFactory::CreatePeerConnectionOrError(
   dependencies.allocator->SetNetworkIgnoreMask(options().network_ignore_mask);
   dependencies.allocator->SetVpnList(configuration.vpn_list);
 
-  std::unique_ptr<RtcEventLog> event_log =
-      worker_thread()->BlockingCall([this] { return CreateRtcEventLog_w(); });
-
-  const FieldTrialsView* trials =
-      dependencies.trials ? dependencies.trials.get() : &field_trials();
   std::unique_ptr<Call> call =
-      worker_thread()->BlockingCall([this, &event_log, trials, &configuration] {
-        return CreateCall_w(event_log.get(), *trials, configuration);
+      worker_thread()->BlockingCall([this, &env, &configuration] {
+        return CreateCall_w(env, configuration);
       });
 
-  auto result = PeerConnection::Create(context_, options_, std::move(event_log),
-                                       std::move(call), configuration,
-                                       std::move(dependencies));
+  auto result = PeerConnection::Create(env, context_, options_, std::move(call),
+                                       configuration, std::move(dependencies));
   if (!result.ok()) {
     return result.MoveError();
   }
@@ -285,23 +301,12 @@ rtc::scoped_refptr<AudioTrackInterface> PeerConnectionFactory::CreateAudioTrack(
   return AudioTrackProxy::Create(signaling_thread(), track);
 }
 
-std::unique_ptr<RtcEventLog> PeerConnectionFactory::CreateRtcEventLog_w() {
-  RTC_DCHECK_RUN_ON(worker_thread());
-
-  auto encoding_type = RtcEventLog::EncodingType::NewFormat;
-  if (field_trials().IsDisabled("WebRTC-RtcEventLogNewFormat"))
-    encoding_type = RtcEventLog::EncodingType::Legacy;
-  return event_log_factory_ ? event_log_factory_->Create(encoding_type)
-                            : std::make_unique<RtcEventLogNull>();
-}
-
 std::unique_ptr<Call> PeerConnectionFactory::CreateCall_w(
-    RtcEventLog* event_log,
-    const FieldTrialsView& field_trials,
+    const Environment& env,
     const PeerConnectionInterface::RTCConfiguration& configuration) {
   RTC_DCHECK_RUN_ON(worker_thread());
 
-  CallConfig call_config(event_log, network_thread());
+  CallConfig call_config(env, network_thread());
   if (!media_engine() || !context_->call_factory()) {
     return nullptr;
   }
@@ -314,7 +319,7 @@ std::unique_ptr<Call> PeerConnectionFactory::CreateCall_w(
   FieldTrialParameter<DataRate> max_bandwidth("max",
                                               DataRate::KilobitsPerSec(2000));
   ParseFieldTrial({&min_bandwidth, &start_bandwidth, &max_bandwidth},
-                  field_trials.Lookup("WebRTC-PcFactoryDefaultBitrates"));
+                  env.field_trials().Lookup("WebRTC-PcFactoryDefaultBitrates"));
 
   call_config.bitrate_config.min_bitrate_bps =
       rtc::saturated_cast<int>(min_bandwidth->bps());
@@ -324,7 +329,6 @@ std::unique_ptr<Call> PeerConnectionFactory::CreateCall_w(
       rtc::saturated_cast<int>(max_bandwidth->bps());
 
   call_config.fec_controller_factory = fec_controller_factory_.get();
-  call_config.task_queue_factory = task_queue_factory_.get();
   call_config.network_state_predictor_factory =
       network_state_predictor_factory_.get();
   call_config.neteq_factory = neteq_factory_.get();
@@ -337,11 +341,9 @@ std::unique_ptr<Call> PeerConnectionFactory::CreateCall_w(
     RTC_LOG(LS_INFO) << "Using default network controller factory";
   }
 
-  call_config.trials = &field_trials;
   call_config.rtp_transport_controller_send_factory =
       transport_controller_send_factory_.get();
   call_config.metronome = metronome_.get();
-  call_config.pacer_burst_interval = configuration.pacer_burst_interval;
   return context_->call_factory()->CreateCall(call_config);
 }
 
diff --git a/third_party/libwebrtc/pc/peer_connection_factory.h b/third_party/libwebrtc/pc/peer_connection_factory.h
index f55d09f6d8..c3760c02c9 100644
--- a/third_party/libwebrtc/pc/peer_connection_factory.h
+++ b/third_party/libwebrtc/pc/peer_connection_factory.h
@@ -29,12 +29,10 @@
 #include "api/network_state_predictor.h"
 #include "api/peer_connection_interface.h"
 #include "api/rtc_error.h"
-#include "api/rtc_event_log/rtc_event_log.h"
 #include "api/rtc_event_log/rtc_event_log_factory_interface.h"
 #include "api/rtp_parameters.h"
 #include "api/scoped_refptr.h"
 #include "api/sequence_checker.h"
-#include "api/task_queue/task_queue_factory.h"
 #include "api/transport/network_control.h"
 #include "api/transport/sctp_transport_factory_interface.h"
 #include "call/call.h"
@@ -53,8 +51,6 @@ class BasicPacketSocketFactory;
 
 namespace webrtc {
 
-class RtcEventLog;
-
 class PeerConnectionFactory : public PeerConnectionFactoryInterface {
  public:
   // Creates a PeerConnectionFactory. It returns nullptr on initialization
@@ -135,16 +131,13 @@ class PeerConnectionFactory : public PeerConnectionFactoryInterface {
 
   bool IsTrialEnabled(absl::string_view key) const;
 
-  std::unique_ptr<RtcEventLog> CreateRtcEventLog_w();
   std::unique_ptr<Call> CreateCall_w(
-      RtcEventLog* event_log,
-      const FieldTrialsView& field_trials,
+      const Environment& env,
       const PeerConnectionInterface::RTCConfiguration& configuration);
 
   rtc::scoped_refptr<ConnectionContext> context_;
   PeerConnectionFactoryInterface::Options options_
       RTC_GUARDED_BY(signaling_thread());
-  std::unique_ptr<TaskQueueFactory> task_queue_factory_;
   std::unique_ptr<RtcEventLogFactoryInterface> event_log_factory_;
   std::unique_ptr<FecControllerFactoryInterface> fec_controller_factory_;
   std::unique_ptr<NetworkStatePredictorFactoryInterface>
diff --git a/third_party/libwebrtc/pc/peer_connection_factory_proxy.h b/third_party/libwebrtc/pc/peer_connection_factory_proxy.h
index 4781497642..b9bd1cbf0e 100644
--- a/third_party/libwebrtc/pc/peer_connection_factory_proxy.h
+++ b/third_party/libwebrtc/pc/peer_connection_factory_proxy.h
@@ -29,10 +29,10 @@ PROXY_METHOD2(RTCErrorOr<rtc::scoped_refptr<PeerConnectionInterface>>,
               CreatePeerConnectionOrError,
               const PeerConnectionInterface::RTCConfiguration&,
               PeerConnectionDependencies)
-PROXY_CONSTMETHOD1(webrtc::RtpCapabilities,
+PROXY_CONSTMETHOD1(RtpCapabilities,
                    GetRtpSenderCapabilities,
                    cricket::MediaType)
-PROXY_CONSTMETHOD1(webrtc::RtpCapabilities,
+PROXY_CONSTMETHOD1(RtpCapabilities,
                    GetRtpReceiverCapabilities,
                    cricket::MediaType)
 PROXY_METHOD1(rtc::scoped_refptr<MediaStreamInterface>,
diff --git a/third_party/libwebrtc/pc/peer_connection_factory_unittest.cc b/third_party/libwebrtc/pc/peer_connection_factory_unittest.cc
index 11e232c01f..618ac68388 100644
--- a/third_party/libwebrtc/pc/peer_connection_factory_unittest.cc
+++ b/third_party/libwebrtc/pc/peer_connection_factory_unittest.cc
@@ -20,6 +20,8 @@
 #include "api/audio_codecs/builtin_audio_encoder_factory.h"
 #include "api/create_peerconnection_factory.h"
 #include "api/data_channel_interface.h"
+#include "api/enable_media.h"
+#include "api/environment/environment_factory.h"
 #include "api/jsep.h"
 #include "api/media_stream_interface.h"
 #include "api/task_queue/default_task_queue_factory.h"
@@ -35,7 +37,6 @@
 #include "api/video_codecs/video_encoder_factory_template_libvpx_vp9_adapter.h"
 #include "api/video_codecs/video_encoder_factory_template_open_h264_adapter.h"
 #include "media/base/fake_frame_source.h"
-#include "media/engine/webrtc_media_engine.h"
 #include "modules/audio_device/include/audio_device.h"
 #include "modules/audio_processing/include/audio_processing.h"
 #include "p2p/base/fake_port_allocator.h"
@@ -106,8 +107,7 @@ class NullPeerConnectionObserver : public PeerConnectionObserver {
       PeerConnectionInterface::IceConnectionState new_state) override {}
   void OnIceGatheringChange(
       PeerConnectionInterface::IceGatheringState new_state) override {}
-  void OnIceCandidate(const webrtc::IceCandidateInterface* candidate) override {
-  }
+  void OnIceCandidate(const IceCandidateInterface* candidate) override {}
 };
 
 class MockNetworkManager : public rtc::NetworkManager {
@@ -133,17 +133,15 @@ class PeerConnectionFactoryTest : public ::testing::Test {
  private:
   void SetUp() {
 #ifdef WEBRTC_ANDROID
-    webrtc::InitializeAndroidObjects();
+    InitializeAndroidObjects();
 #endif
     // Use fake audio device module since we're only testing the interface
     // level, and using a real one could make tests flaky e.g. when run in
     // parallel.
-    factory_ = webrtc::CreatePeerConnectionFactory(
+    factory_ = CreatePeerConnectionFactory(
         rtc::Thread::Current(), rtc::Thread::Current(), rtc::Thread::Current(),
-        rtc::scoped_refptr<webrtc::AudioDeviceModule>(
-            FakeAudioCaptureModule::Create()),
-        webrtc::CreateBuiltinAudioEncoderFactory(),
-        webrtc::CreateBuiltinAudioDecoderFactory(),
+        rtc::scoped_refptr<AudioDeviceModule>(FakeAudioCaptureModule::Create()),
+        CreateBuiltinAudioEncoderFactory(), CreateBuiltinAudioDecoderFactory(),
         std::make_unique<VideoEncoderFactoryTemplate<
             LibvpxVp8EncoderTemplateAdapter, LibvpxVp9EncoderTemplateAdapter,
             OpenH264EncoderTemplateAdapter, LibaomAv1EncoderTemplateAdapter>>(),
@@ -182,64 +180,64 @@ class PeerConnectionFactoryTest : public ::testing::Test {
     }
   }
 
-  void VerifyAudioCodecCapability(const webrtc::RtpCodecCapability& codec) {
+  void VerifyAudioCodecCapability(const RtpCodecCapability& codec) {
     EXPECT_EQ(codec.kind, cricket::MEDIA_TYPE_AUDIO);
     EXPECT_FALSE(codec.name.empty());
     EXPECT_GT(codec.clock_rate, 0);
     EXPECT_GT(codec.num_channels, 0);
   }
 
-  void VerifyVideoCodecCapability(const webrtc::RtpCodecCapability& codec,
+  void VerifyVideoCodecCapability(const RtpCodecCapability& codec,
                                   bool sender) {
     EXPECT_EQ(codec.kind, cricket::MEDIA_TYPE_VIDEO);
     EXPECT_FALSE(codec.name.empty());
     EXPECT_GT(codec.clock_rate, 0);
     if (sender) {
       if (codec.name == "VP8" || codec.name == "H264") {
-        EXPECT_THAT(codec.scalability_modes,
-                    UnorderedElementsAre(webrtc::ScalabilityMode::kL1T1,
-                                         webrtc::ScalabilityMode::kL1T2,
-                                         webrtc::ScalabilityMode::kL1T3))
+        EXPECT_THAT(
+            codec.scalability_modes,
+            UnorderedElementsAre(ScalabilityMode::kL1T1, ScalabilityMode::kL1T2,
+                                 ScalabilityMode::kL1T3))
             << "Codec: " << codec.name;
       } else if (codec.name == "VP9" || codec.name == "AV1") {
         EXPECT_THAT(
             codec.scalability_modes,
             UnorderedElementsAre(
                 // clang-format off
-                webrtc::ScalabilityMode::kL1T1,
-                webrtc::ScalabilityMode::kL1T2,
-                webrtc::ScalabilityMode::kL1T3,
-                webrtc::ScalabilityMode::kL2T1,
-                webrtc::ScalabilityMode::kL2T1h,
-                webrtc::ScalabilityMode::kL2T1_KEY,
-                webrtc::ScalabilityMode::kL2T2,
-                webrtc::ScalabilityMode::kL2T2h,
-                webrtc::ScalabilityMode::kL2T2_KEY,
-                webrtc::ScalabilityMode::kL2T2_KEY_SHIFT,
-                webrtc::ScalabilityMode::kL2T3,
-                webrtc::ScalabilityMode::kL2T3h,
-                webrtc::ScalabilityMode::kL2T3_KEY,
-                webrtc::ScalabilityMode::kL3T1,
-                webrtc::ScalabilityMode::kL3T1h,
-                webrtc::ScalabilityMode::kL3T1_KEY,
-                webrtc::ScalabilityMode::kL3T2,
-                webrtc::ScalabilityMode::kL3T2h,
-                webrtc::ScalabilityMode::kL3T2_KEY,
-                webrtc::ScalabilityMode::kL3T3,
-                webrtc::ScalabilityMode::kL3T3h,
-                webrtc::ScalabilityMode::kL3T3_KEY,
-                webrtc::ScalabilityMode::kS2T1,
-                webrtc::ScalabilityMode::kS2T1h,
-                webrtc::ScalabilityMode::kS2T2,
-                webrtc::ScalabilityMode::kS2T2h,
-                webrtc::ScalabilityMode::kS2T3,
-                webrtc::ScalabilityMode::kS2T3h,
-                webrtc::ScalabilityMode::kS3T1,
-                webrtc::ScalabilityMode::kS3T1h,
-                webrtc::ScalabilityMode::kS3T2,
-                webrtc::ScalabilityMode::kS3T2h,
-                webrtc::ScalabilityMode::kS3T3,
-                webrtc::ScalabilityMode::kS3T3h)
+                ScalabilityMode::kL1T1,
+                ScalabilityMode::kL1T2,
+                ScalabilityMode::kL1T3,
+                ScalabilityMode::kL2T1,
+                ScalabilityMode::kL2T1h,
+                ScalabilityMode::kL2T1_KEY,
+                ScalabilityMode::kL2T2,
+                ScalabilityMode::kL2T2h,
+                ScalabilityMode::kL2T2_KEY,
+                ScalabilityMode::kL2T2_KEY_SHIFT,
+                ScalabilityMode::kL2T3,
+                ScalabilityMode::kL2T3h,
+                ScalabilityMode::kL2T3_KEY,
+                ScalabilityMode::kL3T1,
+                ScalabilityMode::kL3T1h,
+                ScalabilityMode::kL3T1_KEY,
+                ScalabilityMode::kL3T2,
+                ScalabilityMode::kL3T2h,
+                ScalabilityMode::kL3T2_KEY,
+                ScalabilityMode::kL3T3,
+                ScalabilityMode::kL3T3h,
+                ScalabilityMode::kL3T3_KEY,
+                ScalabilityMode::kS2T1,
+                ScalabilityMode::kS2T1h,
+                ScalabilityMode::kS2T2,
+                ScalabilityMode::kS2T2h,
+                ScalabilityMode::kS2T3,
+                ScalabilityMode::kS2T3h,
+                ScalabilityMode::kS3T1,
+                ScalabilityMode::kS3T1h,
+                ScalabilityMode::kS3T2,
+                ScalabilityMode::kS3T2h,
+                ScalabilityMode::kS3T3,
+                ScalabilityMode::kS3T3h)
             // clang-format on
             )
             << "Codec: " << codec.name;
@@ -251,7 +249,7 @@ class PeerConnectionFactoryTest : public ::testing::Test {
     }
   }
 
-  webrtc::test::ScopedKeyValueConfig field_trials_;
+  test::ScopedKeyValueConfig field_trials_;
   std::unique_ptr<rtc::SocketServer> socket_server_;
   rtc::AutoSocketServerThread main_thread_;
   rtc::scoped_refptr<PeerConnectionFactoryInterface> factory_;
@@ -267,37 +265,27 @@ class PeerConnectionFactoryTest : public ::testing::Test {
 // to reconstruct factory with our own ConnectionContext.
 rtc::scoped_refptr<PeerConnectionFactoryInterface>
 CreatePeerConnectionFactoryWithRtxDisabled() {
-  webrtc::PeerConnectionFactoryDependencies pcf_dependencies;
+  PeerConnectionFactoryDependencies pcf_dependencies;
   pcf_dependencies.signaling_thread = rtc::Thread::Current();
   pcf_dependencies.worker_thread = rtc::Thread::Current();
   pcf_dependencies.network_thread = rtc::Thread::Current();
   pcf_dependencies.task_queue_factory = CreateDefaultTaskQueueFactory();
-  pcf_dependencies.call_factory = CreateCallFactory();
-  pcf_dependencies.trials = std::make_unique<webrtc::FieldTrialBasedConfig>();
-
-  cricket::MediaEngineDependencies media_dependencies;
-  media_dependencies.task_queue_factory =
-      pcf_dependencies.task_queue_factory.get();
-  media_dependencies.adm = rtc::scoped_refptr<webrtc::AudioDeviceModule>(
-      FakeAudioCaptureModule::Create());
-  media_dependencies.audio_encoder_factory =
-      webrtc::CreateBuiltinAudioEncoderFactory();
-  media_dependencies.audio_decoder_factory =
-      webrtc::CreateBuiltinAudioDecoderFactory();
-  media_dependencies.video_encoder_factory =
+
+  pcf_dependencies.adm = FakeAudioCaptureModule::Create();
+  pcf_dependencies.audio_encoder_factory = CreateBuiltinAudioEncoderFactory();
+  pcf_dependencies.audio_decoder_factory = CreateBuiltinAudioDecoderFactory();
+  pcf_dependencies.video_encoder_factory =
       std::make_unique<VideoEncoderFactoryTemplate<
           LibvpxVp8EncoderTemplateAdapter, LibvpxVp9EncoderTemplateAdapter,
           OpenH264EncoderTemplateAdapter, LibaomAv1EncoderTemplateAdapter>>();
-  media_dependencies.video_decoder_factory =
+  pcf_dependencies.video_decoder_factory =
       std::make_unique<VideoDecoderFactoryTemplate<
           LibvpxVp8DecoderTemplateAdapter, LibvpxVp9DecoderTemplateAdapter,
           OpenH264DecoderTemplateAdapter, Dav1dDecoderTemplateAdapter>>(),
-  media_dependencies.trials = pcf_dependencies.trials.get();
-  pcf_dependencies.media_engine =
-      cricket::CreateMediaEngine(std::move(media_dependencies));
+  EnableMedia(pcf_dependencies);
 
-  rtc::scoped_refptr<webrtc::ConnectionContext> context =
-      ConnectionContext::Create(&pcf_dependencies);
+  rtc::scoped_refptr<ConnectionContext> context =
+      ConnectionContext::Create(CreateEnvironment(), &pcf_dependencies);
   context->set_use_rtx(false);
   return rtc::make_ref_counted<PeerConnectionFactory>(context,
                                                       &pcf_dependencies);
@@ -311,26 +299,26 @@ CreatePeerConnectionFactoryWithRtxDisabled() {
 // See https://bugs.chromium.org/p/webrtc/issues/detail?id=7806 for details.
 TEST(PeerConnectionFactoryTestInternal, DISABLED_CreatePCUsingInternalModules) {
 #ifdef WEBRTC_ANDROID
-  webrtc::InitializeAndroidObjects();
+  InitializeAndroidObjects();
 #endif
 
   rtc::scoped_refptr<PeerConnectionFactoryInterface> factory(
-      webrtc::CreatePeerConnectionFactory(
+      CreatePeerConnectionFactory(
           nullptr /* network_thread */, nullptr /* worker_thread */,
           nullptr /* signaling_thread */, nullptr /* default_adm */,
-          webrtc::CreateBuiltinAudioEncoderFactory(),
-          webrtc::CreateBuiltinAudioDecoderFactory(),
+          CreateBuiltinAudioEncoderFactory(),
+          CreateBuiltinAudioDecoderFactory(),
           nullptr /* video_encoder_factory */,
           nullptr /* video_decoder_factory */, nullptr /* audio_mixer */,
           nullptr /* audio_processing */));
 
   NullPeerConnectionObserver observer;
-  webrtc::PeerConnectionInterface::RTCConfiguration config;
-  config.sdp_semantics = webrtc::SdpSemantics::kUnifiedPlan;
+  PeerConnectionInterface::RTCConfiguration config;
+  config.sdp_semantics = SdpSemantics::kUnifiedPlan;
 
   std::unique_ptr<FakeRTCCertificateGenerator> cert_generator(
       new FakeRTCCertificateGenerator());
-  webrtc::PeerConnectionDependencies pc_dependencies(&observer);
+  PeerConnectionDependencies pc_dependencies(&observer);
   pc_dependencies.cert_generator = std::move(cert_generator);
   auto result =
       factory->CreatePeerConnectionOrError(config, std::move(pc_dependencies));
@@ -339,7 +327,7 @@ TEST(PeerConnectionFactoryTestInternal, DISABLED_CreatePCUsingInternalModules) {
 }
 
 TEST_F(PeerConnectionFactoryTest, CheckRtpSenderAudioCapabilities) {
-  webrtc::RtpCapabilities audio_capabilities =
+  RtpCapabilities audio_capabilities =
       factory_->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_AUDIO);
   EXPECT_FALSE(audio_capabilities.codecs.empty());
   for (const auto& codec : audio_capabilities.codecs) {
@@ -352,7 +340,7 @@ TEST_F(PeerConnectionFactoryTest, CheckRtpSenderAudioCapabilities) {
 }
 
 TEST_F(PeerConnectionFactoryTest, CheckRtpSenderVideoCapabilities) {
-  webrtc::RtpCapabilities video_capabilities =
+  RtpCapabilities video_capabilities =
       factory_->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_VIDEO);
   EXPECT_FALSE(video_capabilities.codecs.empty());
   for (const auto& codec : video_capabilities.codecs) {
@@ -365,7 +353,7 @@ TEST_F(PeerConnectionFactoryTest, CheckRtpSenderVideoCapabilities) {
 }
 
 TEST_F(PeerConnectionFactoryTest, CheckRtpSenderRtxEnabledCapabilities) {
-  webrtc::RtpCapabilities video_capabilities =
+  RtpCapabilities video_capabilities =
       factory_->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_VIDEO);
   const auto it = std::find_if(
       video_capabilities.codecs.begin(), video_capabilities.codecs.end(),
@@ -375,7 +363,7 @@ TEST_F(PeerConnectionFactoryTest, CheckRtpSenderRtxEnabledCapabilities) {
 
 TEST(PeerConnectionFactoryTestInternal, CheckRtpSenderRtxDisabledCapabilities) {
   auto factory = CreatePeerConnectionFactoryWithRtxDisabled();
-  webrtc::RtpCapabilities video_capabilities =
+  RtpCapabilities video_capabilities =
       factory->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_VIDEO);
   const auto it = std::find_if(
       video_capabilities.codecs.begin(), video_capabilities.codecs.end(),
@@ -384,14 +372,14 @@ TEST(PeerConnectionFactoryTestInternal, CheckRtpSenderRtxDisabledCapabilities) {
 }
 
 TEST_F(PeerConnectionFactoryTest, CheckRtpSenderDataCapabilities) {
-  webrtc::RtpCapabilities data_capabilities =
+  RtpCapabilities data_capabilities =
       factory_->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_DATA);
   EXPECT_TRUE(data_capabilities.codecs.empty());
   EXPECT_TRUE(data_capabilities.header_extensions.empty());
 }
 
 TEST_F(PeerConnectionFactoryTest, CheckRtpReceiverAudioCapabilities) {
-  webrtc::RtpCapabilities audio_capabilities =
+  RtpCapabilities audio_capabilities =
       factory_->GetRtpReceiverCapabilities(cricket::MEDIA_TYPE_AUDIO);
   EXPECT_FALSE(audio_capabilities.codecs.empty());
   for (const auto& codec : audio_capabilities.codecs) {
@@ -404,7 +392,7 @@ TEST_F(PeerConnectionFactoryTest, CheckRtpReceiverAudioCapabilities) {
 }
 
 TEST_F(PeerConnectionFactoryTest, CheckRtpReceiverVideoCapabilities) {
-  webrtc::RtpCapabilities video_capabilities =
+  RtpCapabilities video_capabilities =
       factory_->GetRtpReceiverCapabilities(cricket::MEDIA_TYPE_VIDEO);
   EXPECT_FALSE(video_capabilities.codecs.empty());
   for (const auto& codec : video_capabilities.codecs) {
@@ -417,7 +405,7 @@ TEST_F(PeerConnectionFactoryTest, CheckRtpReceiverVideoCapabilities) {
 }
 
 TEST_F(PeerConnectionFactoryTest, CheckRtpReceiverRtxEnabledCapabilities) {
-  webrtc::RtpCapabilities video_capabilities =
+  RtpCapabilities video_capabilities =
       factory_->GetRtpReceiverCapabilities(cricket::MEDIA_TYPE_VIDEO);
   const auto it = std::find_if(
       video_capabilities.codecs.begin(), video_capabilities.codecs.end(),
@@ -428,7 +416,7 @@ TEST_F(PeerConnectionFactoryTest, CheckRtpReceiverRtxEnabledCapabilities) {
 TEST(PeerConnectionFactoryTestInternal,
      CheckRtpReceiverRtxDisabledCapabilities) {
   auto factory = CreatePeerConnectionFactoryWithRtxDisabled();
-  webrtc::RtpCapabilities video_capabilities =
+  RtpCapabilities video_capabilities =
       factory->GetRtpReceiverCapabilities(cricket::MEDIA_TYPE_VIDEO);
   const auto it = std::find_if(
       video_capabilities.codecs.begin(), video_capabilities.codecs.end(),
@@ -437,7 +425,7 @@ TEST(PeerConnectionFactoryTestInternal,
 }
 
 TEST_F(PeerConnectionFactoryTest, CheckRtpReceiverDataCapabilities) {
-  webrtc::RtpCapabilities data_capabilities =
+  RtpCapabilities data_capabilities =
       factory_->GetRtpReceiverCapabilities(cricket::MEDIA_TYPE_DATA);
   EXPECT_TRUE(data_capabilities.codecs.empty());
   EXPECT_TRUE(data_capabilities.header_extensions.empty());
@@ -447,8 +435,8 @@ TEST_F(PeerConnectionFactoryTest, CheckRtpReceiverDataCapabilities) {
 // configuration. Also verifies the URL's parsed correctly as expected.
 TEST_F(PeerConnectionFactoryTest, CreatePCUsingIceServers) {
   PeerConnectionInterface::RTCConfiguration config;
-  config.sdp_semantics = webrtc::SdpSemantics::kUnifiedPlan;
-  webrtc::PeerConnectionInterface::IceServer ice_server;
+  config.sdp_semantics = SdpSemantics::kUnifiedPlan;
+  PeerConnectionInterface::IceServer ice_server;
   ice_server.uri = kStunIceServer;
   config.servers.push_back(ice_server);
   ice_server.uri = kTurnIceServer;
@@ -459,7 +447,7 @@ TEST_F(PeerConnectionFactoryTest, CreatePCUsingIceServers) {
   ice_server.username = kTurnUsername;
   ice_server.password = kTurnPassword;
   config.servers.push_back(ice_server);
-  webrtc::PeerConnectionDependencies pc_dependencies(&observer_);
+  PeerConnectionDependencies pc_dependencies(&observer_);
   pc_dependencies.cert_generator =
       std::make_unique<FakeRTCCertificateGenerator>();
   pc_dependencies.allocator = std::move(port_allocator_);
@@ -484,15 +472,15 @@ TEST_F(PeerConnectionFactoryTest, CreatePCUsingIceServers) {
 // configuration. Also verifies the list of URL's parsed correctly as expected.
 TEST_F(PeerConnectionFactoryTest, CreatePCUsingIceServersUrls) {
   PeerConnectionInterface::RTCConfiguration config;
-  config.sdp_semantics = webrtc::SdpSemantics::kUnifiedPlan;
-  webrtc::PeerConnectionInterface::IceServer ice_server;
+  config.sdp_semantics = SdpSemantics::kUnifiedPlan;
+  PeerConnectionInterface::IceServer ice_server;
   ice_server.urls.push_back(kStunIceServer);
   ice_server.urls.push_back(kTurnIceServer);
   ice_server.urls.push_back(kTurnIceServerWithTransport);
   ice_server.username = kTurnUsername;
   ice_server.password = kTurnPassword;
   config.servers.push_back(ice_server);
-  webrtc::PeerConnectionDependencies pc_dependencies(&observer_);
+  PeerConnectionDependencies pc_dependencies(&observer_);
   pc_dependencies.cert_generator =
       std::make_unique<FakeRTCCertificateGenerator>();
   pc_dependencies.allocator = std::move(port_allocator_);
@@ -515,15 +503,15 @@ TEST_F(PeerConnectionFactoryTest, CreatePCUsingIceServersUrls) {
 
 TEST_F(PeerConnectionFactoryTest, CreatePCUsingNoUsernameInUri) {
   PeerConnectionInterface::RTCConfiguration config;
-  config.sdp_semantics = webrtc::SdpSemantics::kUnifiedPlan;
-  webrtc::PeerConnectionInterface::IceServer ice_server;
+  config.sdp_semantics = SdpSemantics::kUnifiedPlan;
+  PeerConnectionInterface::IceServer ice_server;
   ice_server.uri = kStunIceServer;
   config.servers.push_back(ice_server);
   ice_server.uri = kTurnIceServerWithNoUsernameInUri;
   ice_server.username = kTurnUsername;
   ice_server.password = kTurnPassword;
   config.servers.push_back(ice_server);
-  webrtc::PeerConnectionDependencies pc_dependencies(&observer_);
+  PeerConnectionDependencies pc_dependencies(&observer_);
   pc_dependencies.cert_generator =
       std::make_unique<FakeRTCCertificateGenerator>();
   pc_dependencies.allocator = std::move(port_allocator_);
@@ -541,13 +529,13 @@ TEST_F(PeerConnectionFactoryTest, CreatePCUsingNoUsernameInUri) {
 // has transport parameter in it.
 TEST_F(PeerConnectionFactoryTest, CreatePCUsingTurnUrlWithTransportParam) {
   PeerConnectionInterface::RTCConfiguration config;
-  config.sdp_semantics = webrtc::SdpSemantics::kUnifiedPlan;
-  webrtc::PeerConnectionInterface::IceServer ice_server;
+  config.sdp_semantics = SdpSemantics::kUnifiedPlan;
+  PeerConnectionInterface::IceServer ice_server;
   ice_server.uri = kTurnIceServerWithTransport;
   ice_server.username = kTurnUsername;
   ice_server.password = kTurnPassword;
   config.servers.push_back(ice_server);
-  webrtc::PeerConnectionDependencies pc_dependencies(&observer_);
+  PeerConnectionDependencies pc_dependencies(&observer_);
   pc_dependencies.cert_generator =
       std::make_unique<FakeRTCCertificateGenerator>();
   pc_dependencies.allocator = std::move(port_allocator_);
@@ -563,8 +551,8 @@ TEST_F(PeerConnectionFactoryTest, CreatePCUsingTurnUrlWithTransportParam) {
 
 TEST_F(PeerConnectionFactoryTest, CreatePCUsingSecureTurnUrl) {
   PeerConnectionInterface::RTCConfiguration config;
-  config.sdp_semantics = webrtc::SdpSemantics::kUnifiedPlan;
-  webrtc::PeerConnectionInterface::IceServer ice_server;
+  config.sdp_semantics = SdpSemantics::kUnifiedPlan;
+  PeerConnectionInterface::IceServer ice_server;
   ice_server.uri = kSecureTurnIceServer;
   ice_server.username = kTurnUsername;
   ice_server.password = kTurnPassword;
@@ -577,7 +565,7 @@ TEST_F(PeerConnectionFactoryTest, CreatePCUsingSecureTurnUrl) {
   ice_server.username = kTurnUsername;
   ice_server.password = kTurnPassword;
   config.servers.push_back(ice_server);
-  webrtc::PeerConnectionDependencies pc_dependencies(&observer_);
+  PeerConnectionDependencies pc_dependencies(&observer_);
   pc_dependencies.cert_generator =
       std::make_unique<FakeRTCCertificateGenerator>();
   pc_dependencies.allocator = std::move(port_allocator_);
@@ -602,8 +590,8 @@ TEST_F(PeerConnectionFactoryTest, CreatePCUsingSecureTurnUrl) {
 
 TEST_F(PeerConnectionFactoryTest, CreatePCUsingIPLiteralAddress) {
   PeerConnectionInterface::RTCConfiguration config;
-  config.sdp_semantics = webrtc::SdpSemantics::kUnifiedPlan;
-  webrtc::PeerConnectionInterface::IceServer ice_server;
+  config.sdp_semantics = SdpSemantics::kUnifiedPlan;
+  PeerConnectionInterface::IceServer ice_server;
   ice_server.uri = kStunIceServerWithIPv4Address;
   config.servers.push_back(ice_server);
   ice_server.uri = kStunIceServerWithIPv4AddressWithoutPort;
@@ -616,7 +604,7 @@ TEST_F(PeerConnectionFactoryTest, CreatePCUsingIPLiteralAddress) {
   ice_server.username = kTurnUsername;
   ice_server.password = kTurnPassword;
   config.servers.push_back(ice_server);
-  webrtc::PeerConnectionDependencies pc_dependencies(&observer_);
+  PeerConnectionDependencies pc_dependencies(&observer_);
   pc_dependencies.cert_generator =
       std::make_unique<FakeRTCCertificateGenerator>();
   pc_dependencies.allocator = std::move(port_allocator_);
@@ -644,8 +632,8 @@ TEST_F(PeerConnectionFactoryTest, CreatePCUsingIPLiteralAddress) {
 // This test verifies the captured stream is rendered locally using a
 // local video track.
 TEST_F(PeerConnectionFactoryTest, LocalRendering) {
-  rtc::scoped_refptr<webrtc::FakeVideoTrackSource> source =
-      webrtc::FakeVideoTrackSource::Create(/*is_screencast=*/false);
+  rtc::scoped_refptr<FakeVideoTrackSource> source =
+      FakeVideoTrackSource::Create(/*is_screencast=*/false);
 
   cricket::FakeFrameSource frame_source(1280, 720,
                                         rtc::kNumMicrosecsPerSec / 30);
@@ -673,7 +661,7 @@ TEST_F(PeerConnectionFactoryTest, LocalRendering) {
 }
 
 TEST(PeerConnectionFactoryDependenciesTest, UsesNetworkManager) {
-  constexpr webrtc::TimeDelta kWaitTimeout = webrtc::TimeDelta::Seconds(10);
+  constexpr TimeDelta kWaitTimeout = TimeDelta::Seconds(10);
   auto mock_network_manager = std::make_unique<NiceMock<MockNetworkManager>>();
 
   rtc::Event called;
@@ -681,24 +669,24 @@ TEST(PeerConnectionFactoryDependenciesTest, UsesNetworkManager) {
       .Times(AtLeast(1))
       .WillRepeatedly(InvokeWithoutArgs([&] { called.Set(); }));
 
-  webrtc::PeerConnectionFactoryDependencies pcf_dependencies;
+  PeerConnectionFactoryDependencies pcf_dependencies;
   pcf_dependencies.network_manager = std::move(mock_network_manager);
 
-  rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> pcf =
+  rtc::scoped_refptr<PeerConnectionFactoryInterface> pcf =
       CreateModularPeerConnectionFactory(std::move(pcf_dependencies));
 
   PeerConnectionInterface::RTCConfiguration config;
   config.ice_candidate_pool_size = 2;
   NullPeerConnectionObserver observer;
   auto pc = pcf->CreatePeerConnectionOrError(
-      config, webrtc::PeerConnectionDependencies(&observer));
+      config, PeerConnectionDependencies(&observer));
   ASSERT_TRUE(pc.ok());
 
   called.Wait(kWaitTimeout);
 }
 
 TEST(PeerConnectionFactoryDependenciesTest, UsesPacketSocketFactory) {
-  constexpr webrtc::TimeDelta kWaitTimeout = webrtc::TimeDelta::Seconds(10);
+  constexpr TimeDelta kWaitTimeout = TimeDelta::Seconds(10);
   auto mock_socket_factory =
       std::make_unique<NiceMock<rtc::MockPacketSocketFactory>>();
 
@@ -710,10 +698,10 @@ TEST(PeerConnectionFactoryDependenciesTest, UsesPacketSocketFactory) {
       }))
       .WillRepeatedly(Return(nullptr));
 
-  webrtc::PeerConnectionFactoryDependencies pcf_dependencies;
+  PeerConnectionFactoryDependencies pcf_dependencies;
   pcf_dependencies.packet_socket_factory = std::move(mock_socket_factory);
 
-  rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> pcf =
+  rtc::scoped_refptr<PeerConnectionFactoryInterface> pcf =
       CreateModularPeerConnectionFactory(std::move(pcf_dependencies));
 
   // By default, localhost addresses are ignored, which makes tests fail if test
@@ -726,7 +714,7 @@ TEST(PeerConnectionFactoryDependenciesTest, UsesPacketSocketFactory) {
   config.ice_candidate_pool_size = 2;
   NullPeerConnectionObserver observer;
   auto pc = pcf->CreatePeerConnectionOrError(
-      config, webrtc::PeerConnectionDependencies(&observer));
+      config, PeerConnectionDependencies(&observer));
   ASSERT_TRUE(pc.ok());
 
   called.Wait(kWaitTimeout);
diff --git a/third_party/libwebrtc/pc/peer_connection_field_trial_tests.cc b/third_party/libwebrtc/pc/peer_connection_field_trial_tests.cc
index 7799c9d6e3..4cbe24986c 100644
--- a/third_party/libwebrtc/pc/peer_connection_field_trial_tests.cc
+++ b/third_party/libwebrtc/pc/peer_connection_field_trial_tests.cc
@@ -16,13 +16,13 @@
 #include "api/audio_codecs/builtin_audio_decoder_factory.h"
 #include "api/audio_codecs/builtin_audio_encoder_factory.h"
 #include "api/create_peerconnection_factory.h"
+#include "api/enable_media_with_defaults.h"
 #include "api/peer_connection_interface.h"
 #include "api/stats/rtcstats_objects.h"
 #include "api/task_queue/default_task_queue_factory.h"
 #include "api/video_codecs/builtin_video_decoder_factory.h"
 #include "api/video_codecs/builtin_video_encoder_factory.h"
 #include "media/engine/webrtc_media_engine.h"
-#include "media/engine/webrtc_media_engine_defaults.h"
 #include "pc/peer_connection_wrapper.h"
 #include "pc/session_description.h"
 #include "pc/test/fake_audio_capture_module.h"
@@ -68,7 +68,7 @@ class PeerConnectionFieldTrialTest : public ::testing::Test {
 #ifdef WEBRTC_ANDROID
     InitializeAndroidObjects();
 #endif
-    webrtc::PeerConnectionInterface::IceServer ice_server;
+    PeerConnectionInterface::IceServer ice_server;
     ice_server.uri = "stun:stun.l.google.com:19302";
     config_.servers.push_back(ice_server);
     config_.sdp_semantics = SdpSemantics::kUnifiedPlan;
@@ -81,13 +81,8 @@ class PeerConnectionFieldTrialTest : public ::testing::Test {
     pcf_deps.signaling_thread = rtc::Thread::Current();
     pcf_deps.trials = std::move(field_trials);
     pcf_deps.task_queue_factory = CreateDefaultTaskQueueFactory();
-    pcf_deps.call_factory = webrtc::CreateCallFactory();
-    cricket::MediaEngineDependencies media_deps;
-    media_deps.task_queue_factory = pcf_deps.task_queue_factory.get();
-    media_deps.adm = FakeAudioCaptureModule::Create();
-    media_deps.trials = pcf_deps.trials.get();
-    webrtc::SetMediaEngineDefaults(&media_deps);
-    pcf_deps.media_engine = cricket::CreateMediaEngine(std::move(media_deps));
+    pcf_deps.adm = FakeAudioCaptureModule::Create();
+    EnableMediaWithDefaults(pcf_deps);
     pc_factory_ = CreateModularPeerConnectionFactory(std::move(pcf_deps));
 
     // Allow ADAPTER_TYPE_LOOPBACK to create PeerConnections with loopback in
@@ -113,7 +108,7 @@ class PeerConnectionFieldTrialTest : public ::testing::Test {
   std::unique_ptr<rtc::SocketServer> socket_server_;
   rtc::AutoSocketServerThread main_thread_;
   rtc::scoped_refptr<PeerConnectionFactoryInterface> pc_factory_ = nullptr;
-  webrtc::PeerConnectionInterface::RTCConfiguration config_;
+  PeerConnectionInterface::RTCConfiguration config_;
 };
 
 // Tests for the dependency descriptor field trial. The dependency descriptor
@@ -138,7 +133,7 @@ TEST_F(PeerConnectionFieldTrialTest, EnableDependencyDescriptorAdvertised) {
       media_description1->rtp_header_extensions();
 
   bool found = absl::c_find_if(rtp_header_extensions1,
-                               [](const webrtc::RtpExtension& rtp_extension) {
+                               [](const RtpExtension& rtp_extension) {
                                  return rtp_extension.uri ==
                                         RtpExtension::kDependencyDescriptorUri;
                                }) != rtp_header_extensions1.end();
@@ -168,14 +163,14 @@ TEST_F(PeerConnectionFieldTrialTest, InjectDependencyDescriptor) {
       media_description1->rtp_header_extensions();
 
   bool found1 = absl::c_find_if(rtp_header_extensions1,
-                                [](const webrtc::RtpExtension& rtp_extension) {
+                                [](const RtpExtension& rtp_extension) {
                                   return rtp_extension.uri ==
                                          RtpExtension::kDependencyDescriptorUri;
                                 }) != rtp_header_extensions1.end();
   EXPECT_FALSE(found1);
 
   std::set<int> existing_ids;
-  for (const webrtc::RtpExtension& rtp_extension : rtp_header_extensions1) {
+  for (const RtpExtension& rtp_extension : rtp_header_extensions1) {
     existing_ids.insert(rtp_extension.id);
   }
 
@@ -212,7 +207,7 @@ TEST_F(PeerConnectionFieldTrialTest, InjectDependencyDescriptor) {
       media_description2->rtp_header_extensions();
 
   bool found2 = absl::c_find_if(rtp_header_extensions2,
-                                [](const webrtc::RtpExtension& rtp_extension) {
+                                [](const RtpExtension& rtp_extension) {
                                   return rtp_extension.uri ==
                                          RtpExtension::kDependencyDescriptorUri;
                                 }) != rtp_header_extensions2.end();
diff --git a/third_party/libwebrtc/pc/peer_connection_header_extension_unittest.cc b/third_party/libwebrtc/pc/peer_connection_header_extension_unittest.cc
index b1c6c3cfb5..277979b330 100644
--- a/third_party/libwebrtc/pc/peer_connection_header_extension_unittest.cc
+++ b/third_party/libwebrtc/pc/peer_connection_header_extension_unittest.cc
@@ -16,7 +16,6 @@
 
 #include "absl/strings/string_view.h"
 #include "absl/types/optional.h"
-#include "api/call/call_factory_interface.h"
 #include "api/jsep.h"
 #include "api/media_types.h"
 #include "api/peer_connection_interface.h"
@@ -35,6 +34,7 @@
 #include "p2p/base/port_allocator.h"
 #include "pc/peer_connection_wrapper.h"
 #include "pc/session_description.h"
+#include "pc/test/enable_fake_media.h"
 #include "pc/test/mock_peer_connection_observers.h"
 #include "rtc_base/internal/default_socket_server.h"
 #include "rtc_base/rtc_certificate_generator.h"
@@ -77,21 +77,18 @@ class PeerConnectionHeaderExtensionTest
   std::unique_ptr<PeerConnectionWrapper> CreatePeerConnection(
       cricket::MediaType media_type,
       absl::optional<SdpSemantics> semantics) {
-    auto voice = std::make_unique<cricket::FakeVoiceEngine>();
-    auto video = std::make_unique<cricket::FakeVideoEngine>();
+    auto media_engine = std::make_unique<cricket::FakeMediaEngine>();
     if (media_type == cricket::MediaType::MEDIA_TYPE_AUDIO)
-      voice->SetRtpHeaderExtensions(extensions_);
+      media_engine->fake_voice_engine()->SetRtpHeaderExtensions(extensions_);
     else
-      video->SetRtpHeaderExtensions(extensions_);
-    auto media_engine = std::make_unique<cricket::CompositeMediaEngine>(
-        std::move(voice), std::move(video));
+      media_engine->fake_video_engine()->SetRtpHeaderExtensions(extensions_);
     PeerConnectionFactoryDependencies factory_dependencies;
     factory_dependencies.network_thread = rtc::Thread::Current();
     factory_dependencies.worker_thread = rtc::Thread::Current();
     factory_dependencies.signaling_thread = rtc::Thread::Current();
     factory_dependencies.task_queue_factory = CreateDefaultTaskQueueFactory();
-    factory_dependencies.media_engine = std::move(media_engine);
-    factory_dependencies.call_factory = CreateCallFactory();
+    EnableFakeMedia(factory_dependencies, std::move(media_engine));
+
     factory_dependencies.event_log_factory =
         std::make_unique<RtcEventLogFactory>(
             factory_dependencies.task_queue_factory.get());
@@ -117,7 +114,7 @@ class PeerConnectionHeaderExtensionTest
         pc_factory, result.MoveValue(), std::move(observer));
   }
 
-  webrtc::test::ScopedKeyValueConfig field_trials_;
+  test::ScopedKeyValueConfig field_trials_;
   std::unique_ptr<rtc::SocketServer> socket_server_;
   rtc::AutoSocketServerThread main_thread_;
   std::vector<RtpHeaderExtensionCapability> extensions_;
diff --git a/third_party/libwebrtc/pc/peer_connection_histogram_unittest.cc b/third_party/libwebrtc/pc/peer_connection_histogram_unittest.cc
index 68a4dbc361..58bd6ebb48 100644
--- a/third_party/libwebrtc/pc/peer_connection_histogram_unittest.cc
+++ b/third_party/libwebrtc/pc/peer_connection_histogram_unittest.cc
@@ -15,7 +15,6 @@
 #include <vector>
 
 #include "absl/types/optional.h"
-#include "api/async_resolver_factory.h"
 #include "api/call/call_factory_interface.h"
 #include "api/jsep.h"
 #include "api/jsep_session_description.h"
@@ -25,9 +24,7 @@
 #include "api/task_queue/default_task_queue_factory.h"
 #include "api/task_queue/task_queue_factory.h"
 #include "api/test/mock_async_dns_resolver.h"
-#include "media/base/fake_media_engine.h"
 #include "media/base/media_engine.h"
-#include "p2p/base/mock_async_resolver.h"
 #include "p2p/base/port_allocator.h"
 #include "p2p/client/basic_port_allocator.h"
 #include "pc/peer_connection.h"
@@ -35,6 +32,7 @@
 #include "pc/peer_connection_proxy.h"
 #include "pc/peer_connection_wrapper.h"
 #include "pc/sdp_utils.h"
+#include "pc/test/enable_fake_media.h"
 #include "pc/test/mock_peer_connection_observers.h"
 #include "pc/usage_pattern.h"
 #include "pc/webrtc_sdp.h"
@@ -84,9 +82,7 @@ class PeerConnectionFactoryForUsageHistogramTest
           dependencies.worker_thread = rtc::Thread::Current();
           dependencies.signaling_thread = rtc::Thread::Current();
           dependencies.task_queue_factory = CreateDefaultTaskQueueFactory();
-          dependencies.media_engine =
-              std::make_unique<cricket::FakeMediaEngine>();
-          dependencies.call_factory = CreateCallFactory();
+          EnableFakeMedia(dependencies);
           return dependencies;
         }()) {}
 };
@@ -97,7 +93,7 @@ typedef PeerConnectionWrapperForUsageHistogramTest* RawWrapperPtr;
 
 class ObserverForUsageHistogramTest : public MockPeerConnectionObserver {
  public:
-  void OnIceCandidate(const webrtc::IceCandidateInterface* candidate) override;
+  void OnIceCandidate(const IceCandidateInterface* candidate) override;
 
   void OnInterestingUsage(int usage_pattern) override {
     interesting_usage_detected_ = usage_pattern;
@@ -160,12 +156,11 @@ class PeerConnectionWrapperForUsageHistogramTest
     return static_cast<ObserverForUsageHistogramTest*>(observer())
         ->HaveDataChannel();
   }
-  void BufferIceCandidate(const webrtc::IceCandidateInterface* candidate) {
+  void BufferIceCandidate(const IceCandidateInterface* candidate) {
     std::string sdp;
     EXPECT_TRUE(candidate->ToString(&sdp));
-    std::unique_ptr<webrtc::IceCandidateInterface> candidate_copy(
-        CreateIceCandidate(candidate->sdp_mid(), candidate->sdp_mline_index(),
-                           sdp, nullptr));
+    std::unique_ptr<IceCandidateInterface> candidate_copy(CreateIceCandidate(
+        candidate->sdp_mid(), candidate->sdp_mline_index(), sdp, nullptr));
     buffered_candidates_.push_back(std::move(candidate_copy));
   }
 
@@ -216,19 +211,18 @@ class PeerConnectionWrapperForUsageHistogramTest
     return true;
   }
 
-  webrtc::PeerConnectionInterface::IceGatheringState ice_gathering_state() {
+  PeerConnectionInterface::IceGatheringState ice_gathering_state() {
     return pc()->ice_gathering_state();
   }
 
  private:
   // Candidates that have been sent but not yet configured
-  std::vector<std::unique_ptr<webrtc::IceCandidateInterface>>
-      buffered_candidates_;
+  std::vector<std::unique_ptr<IceCandidateInterface>> buffered_candidates_;
 };
 
 // Buffers candidates until we add them via AddBufferedIceCandidates.
 void ObserverForUsageHistogramTest::OnIceCandidate(
-    const webrtc::IceCandidateInterface* candidate) {
+    const IceCandidateInterface* candidate) {
   // If target is not set, ignore. This happens in one-ended unit tests.
   if (candidate_target_) {
     this->candidate_target_->BufferIceCandidate(candidate);
@@ -245,12 +239,12 @@ class PeerConnectionUsageHistogramTest : public ::testing::Test {
       : vss_(new rtc::VirtualSocketServer()),
         socket_factory_(new rtc::BasicPacketSocketFactory(vss_.get())),
         main_(vss_.get()) {
-    webrtc::metrics::Reset();
+    metrics::Reset();
   }
 
   WrapperPtr CreatePeerConnection() {
     RTCConfiguration config;
-    config.sdp_semantics = webrtc::SdpSemantics::kUnifiedPlan;
+    config.sdp_semantics = SdpSemantics::kUnifiedPlan;
     return CreatePeerConnection(
         config, PeerConnectionFactoryInterface::Options(), nullptr);
   }
@@ -262,13 +256,13 @@ class PeerConnectionUsageHistogramTest : public ::testing::Test {
 
   WrapperPtr CreatePeerConnectionWithMdns(const RTCConfiguration& config) {
     auto resolver_factory =
-        std::make_unique<NiceMock<webrtc::MockAsyncDnsResolverFactory>>();
+        std::make_unique<NiceMock<MockAsyncDnsResolverFactory>>();
 
-    webrtc::PeerConnectionDependencies deps(nullptr /* observer_in */);
+    PeerConnectionDependencies deps(nullptr /* observer_in */);
 
     auto fake_network = NewFakeNetwork();
     fake_network->set_mdns_responder(
-        std::make_unique<webrtc::FakeMdnsResponder>(rtc::Thread::Current()));
+        std::make_unique<FakeMdnsResponder>(rtc::Thread::Current()));
     fake_network->AddInterface(NextLocalAddress());
 
     std::unique_ptr<cricket::BasicPortAllocator> port_allocator(
@@ -283,7 +277,7 @@ class PeerConnectionUsageHistogramTest : public ::testing::Test {
 
   WrapperPtr CreatePeerConnectionWithImmediateReport() {
     RTCConfiguration configuration;
-    configuration.sdp_semantics = webrtc::SdpSemantics::kUnifiedPlan;
+    configuration.sdp_semantics = SdpSemantics::kUnifiedPlan;
     configuration.report_usage_pattern_delay_ms = 0;
     return CreatePeerConnection(
         configuration, PeerConnectionFactoryInterface::Options(), nullptr);
@@ -364,7 +358,7 @@ class PeerConnectionUsageHistogramTest : public ::testing::Test {
     // This works correctly only if there is only one sample value
     // that has been counted.
     // Returns -1 for "not found".
-    return webrtc::metrics::MinSample(kUsagePatternMetric);
+    return metrics::MinSample(kUsagePatternMetric);
   }
 
   // The PeerConnection's port allocator is tied to the PeerConnection's
@@ -393,10 +387,10 @@ TEST_F(PeerConnectionUsageHistogramTest, UsageFingerprintHistogramFromTimeout) {
   auto pc = CreatePeerConnectionWithImmediateReport();
 
   int expected_fingerprint = MakeUsageFingerprint({});
-  EXPECT_METRIC_EQ_WAIT(1, webrtc::metrics::NumSamples(kUsagePatternMetric),
+  EXPECT_METRIC_EQ_WAIT(1, metrics::NumSamples(kUsagePatternMetric),
                         kDefaultTimeout);
   EXPECT_METRIC_EQ(
-      1, webrtc::metrics::NumEvents(kUsagePatternMetric, expected_fingerprint));
+      1, metrics::NumEvents(kUsagePatternMetric, expected_fingerprint));
 }
 
 #ifndef WEBRTC_ANDROID
@@ -421,11 +415,10 @@ TEST_F(PeerConnectionUsageHistogramTest, FingerprintAudioVideo) {
        UsageEvent::DIRECT_CONNECTION_SELECTED, UsageEvent::CLOSE_CALLED});
   // In this case, we may or may not have PRIVATE_CANDIDATE_COLLECTED,
   // depending on the machine configuration.
-  EXPECT_METRIC_EQ(2, webrtc::metrics::NumSamples(kUsagePatternMetric));
+  EXPECT_METRIC_EQ(2, metrics::NumSamples(kUsagePatternMetric));
   EXPECT_METRIC_TRUE(
-      webrtc::metrics::NumEvents(kUsagePatternMetric, expected_fingerprint) ==
-          2 ||
-      webrtc::metrics::NumEvents(
+      metrics::NumEvents(kUsagePatternMetric, expected_fingerprint) == 2 ||
+      metrics::NumEvents(
           kUsagePatternMetric,
           expected_fingerprint |
               static_cast<int>(UsageEvent::PRIVATE_CANDIDATE_COLLECTED)) == 2);
@@ -466,11 +459,11 @@ TEST_F(PeerConnectionUsageHistogramTest, FingerprintWithMdnsCaller) {
        UsageEvent::CANDIDATE_COLLECTED, UsageEvent::ADD_ICE_CANDIDATE_SUCCEEDED,
        UsageEvent::REMOTE_MDNS_CANDIDATE_ADDED, UsageEvent::ICE_STATE_CONNECTED,
        UsageEvent::REMOTE_CANDIDATE_ADDED, UsageEvent::CLOSE_CALLED});
-  EXPECT_METRIC_EQ(2, webrtc::metrics::NumSamples(kUsagePatternMetric));
-  EXPECT_METRIC_EQ(1, webrtc::metrics::NumEvents(kUsagePatternMetric,
-                                                 expected_fingerprint_caller));
-  EXPECT_METRIC_EQ(1, webrtc::metrics::NumEvents(kUsagePatternMetric,
-                                                 expected_fingerprint_callee));
+  EXPECT_METRIC_EQ(2, metrics::NumSamples(kUsagePatternMetric));
+  EXPECT_METRIC_EQ(
+      1, metrics::NumEvents(kUsagePatternMetric, expected_fingerprint_caller));
+  EXPECT_METRIC_EQ(
+      1, metrics::NumEvents(kUsagePatternMetric, expected_fingerprint_callee));
 }
 
 // Test getting the usage fingerprint when the callee collects an mDNS
@@ -507,11 +500,11 @@ TEST_F(PeerConnectionUsageHistogramTest, FingerprintWithMdnsCallee) {
        UsageEvent::ADD_ICE_CANDIDATE_SUCCEEDED, UsageEvent::ICE_STATE_CONNECTED,
        UsageEvent::REMOTE_CANDIDATE_ADDED,
        UsageEvent::DIRECT_CONNECTION_SELECTED, UsageEvent::CLOSE_CALLED});
-  EXPECT_METRIC_EQ(2, webrtc::metrics::NumSamples(kUsagePatternMetric));
-  EXPECT_METRIC_EQ(1, webrtc::metrics::NumEvents(kUsagePatternMetric,
-                                                 expected_fingerprint_caller));
-  EXPECT_METRIC_EQ(1, webrtc::metrics::NumEvents(kUsagePatternMetric,
-                                                 expected_fingerprint_callee));
+  EXPECT_METRIC_EQ(2, metrics::NumSamples(kUsagePatternMetric));
+  EXPECT_METRIC_EQ(
+      1, metrics::NumEvents(kUsagePatternMetric, expected_fingerprint_caller));
+  EXPECT_METRIC_EQ(
+      1, metrics::NumEvents(kUsagePatternMetric, expected_fingerprint_callee));
 }
 
 #ifdef WEBRTC_HAVE_SCTP
@@ -529,11 +522,10 @@ TEST_F(PeerConnectionUsageHistogramTest, FingerprintDataOnly) {
        UsageEvent::CANDIDATE_COLLECTED, UsageEvent::ADD_ICE_CANDIDATE_SUCCEEDED,
        UsageEvent::ICE_STATE_CONNECTED, UsageEvent::REMOTE_CANDIDATE_ADDED,
        UsageEvent::DIRECT_CONNECTION_SELECTED, UsageEvent::CLOSE_CALLED});
-  EXPECT_METRIC_EQ(2, webrtc::metrics::NumSamples(kUsagePatternMetric));
+  EXPECT_METRIC_EQ(2, metrics::NumSamples(kUsagePatternMetric));
   EXPECT_METRIC_TRUE(
-      webrtc::metrics::NumEvents(kUsagePatternMetric, expected_fingerprint) ==
-          2 ||
-      webrtc::metrics::NumEvents(
+      metrics::NumEvents(kUsagePatternMetric, expected_fingerprint) == 2 ||
+      metrics::NumEvents(
           kUsagePatternMetric,
           expected_fingerprint |
               static_cast<int>(UsageEvent::PRIVATE_CANDIDATE_COLLECTED)) == 2);
@@ -557,9 +549,9 @@ TEST_F(PeerConnectionUsageHistogramTest, FingerprintStunTurn) {
   int expected_fingerprint = MakeUsageFingerprint(
       {UsageEvent::STUN_SERVER_ADDED, UsageEvent::TURN_SERVER_ADDED,
        UsageEvent::CLOSE_CALLED});
-  EXPECT_METRIC_EQ(1, webrtc::metrics::NumSamples(kUsagePatternMetric));
+  EXPECT_METRIC_EQ(1, metrics::NumSamples(kUsagePatternMetric));
   EXPECT_METRIC_EQ(
-      1, webrtc::metrics::NumEvents(kUsagePatternMetric, expected_fingerprint));
+      1, metrics::NumEvents(kUsagePatternMetric, expected_fingerprint));
 }
 
 TEST_F(PeerConnectionUsageHistogramTest, FingerprintStunTurnInReconfiguration) {
@@ -579,9 +571,9 @@ TEST_F(PeerConnectionUsageHistogramTest, FingerprintStunTurnInReconfiguration) {
   int expected_fingerprint = MakeUsageFingerprint(
       {UsageEvent::STUN_SERVER_ADDED, UsageEvent::TURN_SERVER_ADDED,
        UsageEvent::CLOSE_CALLED});
-  EXPECT_METRIC_EQ(1, webrtc::metrics::NumSamples(kUsagePatternMetric));
+  EXPECT_METRIC_EQ(1, metrics::NumSamples(kUsagePatternMetric));
   EXPECT_METRIC_EQ(
-      1, webrtc::metrics::NumEvents(kUsagePatternMetric, expected_fingerprint));
+      1, metrics::NumEvents(kUsagePatternMetric, expected_fingerprint));
 }
 
 TEST_F(PeerConnectionUsageHistogramTest, FingerprintWithPrivateIPCaller) {
@@ -607,11 +599,11 @@ TEST_F(PeerConnectionUsageHistogramTest, FingerprintWithPrivateIPCaller) {
        UsageEvent::REMOTE_PRIVATE_CANDIDATE_ADDED,
        UsageEvent::ICE_STATE_CONNECTED, UsageEvent::REMOTE_CANDIDATE_ADDED,
        UsageEvent::DIRECT_CONNECTION_SELECTED, UsageEvent::CLOSE_CALLED});
-  EXPECT_METRIC_EQ(2, webrtc::metrics::NumSamples(kUsagePatternMetric));
-  EXPECT_METRIC_EQ(1, webrtc::metrics::NumEvents(kUsagePatternMetric,
-                                                 expected_fingerprint_caller));
-  EXPECT_METRIC_EQ(1, webrtc::metrics::NumEvents(kUsagePatternMetric,
-                                                 expected_fingerprint_callee));
+  EXPECT_METRIC_EQ(2, metrics::NumSamples(kUsagePatternMetric));
+  EXPECT_METRIC_EQ(
+      1, metrics::NumEvents(kUsagePatternMetric, expected_fingerprint_caller));
+  EXPECT_METRIC_EQ(
+      1, metrics::NumEvents(kUsagePatternMetric, expected_fingerprint_callee));
 }
 
 TEST_F(PeerConnectionUsageHistogramTest, FingerprintWithPrivateIpv6Callee) {
@@ -639,11 +631,11 @@ TEST_F(PeerConnectionUsageHistogramTest, FingerprintWithPrivateIpv6Callee) {
        UsageEvent::ADD_ICE_CANDIDATE_SUCCEEDED,
        UsageEvent::REMOTE_CANDIDATE_ADDED, UsageEvent::ICE_STATE_CONNECTED,
        UsageEvent::DIRECT_CONNECTION_SELECTED, UsageEvent::CLOSE_CALLED});
-  EXPECT_METRIC_EQ(2, webrtc::metrics::NumSamples(kUsagePatternMetric));
-  EXPECT_METRIC_EQ(1, webrtc::metrics::NumEvents(kUsagePatternMetric,
-                                                 expected_fingerprint_caller));
-  EXPECT_METRIC_EQ(1, webrtc::metrics::NumEvents(kUsagePatternMetric,
-                                                 expected_fingerprint_callee));
+  EXPECT_METRIC_EQ(2, metrics::NumSamples(kUsagePatternMetric));
+  EXPECT_METRIC_EQ(
+      1, metrics::NumEvents(kUsagePatternMetric, expected_fingerprint_caller));
+  EXPECT_METRIC_EQ(
+      1, metrics::NumEvents(kUsagePatternMetric, expected_fingerprint_callee));
 }
 
 #ifndef WEBRTC_ANDROID
@@ -667,7 +659,7 @@ TEST_F(PeerConnectionUsageHistogramTest,
   ASSERT_TRUE(caller->SetLocalDescription(caller->CreateOffer()));
   // Wait until the gathering completes so that the session description would
   // have contained ICE candidates.
-  EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceGatheringComplete,
+  EXPECT_EQ_WAIT(PeerConnectionInterface::kIceGatheringComplete,
                  caller->ice_gathering_state(), kDefaultTimeout);
   EXPECT_TRUE(caller->observer()->candidate_gathered());
   // Get the current offer that contains candidates and pass it to the callee.
@@ -716,11 +708,11 @@ TEST_F(PeerConnectionUsageHistogramTest,
        UsageEvent::REMOTE_PRIVATE_CANDIDATE_ADDED,
        UsageEvent::REMOTE_IPV6_CANDIDATE_ADDED, UsageEvent::ICE_STATE_CONNECTED,
        UsageEvent::DIRECT_CONNECTION_SELECTED, UsageEvent::CLOSE_CALLED});
-  EXPECT_METRIC_EQ(2, webrtc::metrics::NumSamples(kUsagePatternMetric));
-  EXPECT_METRIC_EQ(1, webrtc::metrics::NumEvents(kUsagePatternMetric,
-                                                 expected_fingerprint_caller));
-  EXPECT_METRIC_EQ(1, webrtc::metrics::NumEvents(kUsagePatternMetric,
-                                                 expected_fingerprint_callee));
+  EXPECT_METRIC_EQ(2, metrics::NumSamples(kUsagePatternMetric));
+  EXPECT_METRIC_EQ(
+      1, metrics::NumEvents(kUsagePatternMetric, expected_fingerprint_caller));
+  EXPECT_METRIC_EQ(
+      1, metrics::NumEvents(kUsagePatternMetric, expected_fingerprint_callee));
 }
 
 TEST_F(PeerConnectionUsageHistogramTest, NotableUsageNoted) {
@@ -731,7 +723,7 @@ TEST_F(PeerConnectionUsageHistogramTest, NotableUsageNoted) {
   int expected_fingerprint = MakeUsageFingerprint(
       {UsageEvent::DATA_ADDED, UsageEvent::SET_LOCAL_DESCRIPTION_SUCCEEDED,
        UsageEvent::CANDIDATE_COLLECTED, UsageEvent::CLOSE_CALLED});
-  EXPECT_METRIC_EQ(1, webrtc::metrics::NumSamples(kUsagePatternMetric));
+  EXPECT_METRIC_EQ(1, metrics::NumSamples(kUsagePatternMetric));
   EXPECT_METRIC_TRUE(
       expected_fingerprint == ObservedFingerprint() ||
       (expected_fingerprint |
@@ -748,9 +740,9 @@ TEST_F(PeerConnectionUsageHistogramTest, NotableUsageOnEventFiring) {
   int expected_fingerprint = MakeUsageFingerprint(
       {UsageEvent::DATA_ADDED, UsageEvent::SET_LOCAL_DESCRIPTION_SUCCEEDED,
        UsageEvent::CANDIDATE_COLLECTED});
-  EXPECT_METRIC_EQ(0, webrtc::metrics::NumSamples(kUsagePatternMetric));
+  EXPECT_METRIC_EQ(0, metrics::NumSamples(kUsagePatternMetric));
   caller->GetInternalPeerConnection()->RequestUsagePatternReportForTesting();
-  EXPECT_METRIC_EQ_WAIT(1, webrtc::metrics::NumSamples(kUsagePatternMetric),
+  EXPECT_METRIC_EQ_WAIT(1, metrics::NumSamples(kUsagePatternMetric),
                         kDefaultTimeout);
   EXPECT_METRIC_TRUE(
       expected_fingerprint == ObservedFingerprint() ||
@@ -769,12 +761,12 @@ TEST_F(PeerConnectionUsageHistogramTest,
   int expected_fingerprint = MakeUsageFingerprint(
       {UsageEvent::DATA_ADDED, UsageEvent::SET_LOCAL_DESCRIPTION_SUCCEEDED,
        UsageEvent::CANDIDATE_COLLECTED, UsageEvent::CLOSE_CALLED});
-  EXPECT_METRIC_EQ(0, webrtc::metrics::NumSamples(kUsagePatternMetric));
+  EXPECT_METRIC_EQ(0, metrics::NumSamples(kUsagePatternMetric));
   caller->pc()->Close();
-  EXPECT_METRIC_EQ(1, webrtc::metrics::NumSamples(kUsagePatternMetric));
+  EXPECT_METRIC_EQ(1, metrics::NumSamples(kUsagePatternMetric));
   caller->GetInternalPeerConnection()->RequestUsagePatternReportForTesting();
   caller->observer()->ClearInterestingUsageDetector();
-  EXPECT_METRIC_EQ_WAIT(2, webrtc::metrics::NumSamples(kUsagePatternMetric),
+  EXPECT_METRIC_EQ_WAIT(2, metrics::NumSamples(kUsagePatternMetric),
                         kDefaultTimeout);
   EXPECT_METRIC_TRUE(
       expected_fingerprint == ObservedFingerprint() ||
diff --git a/third_party/libwebrtc/pc/peer_connection_ice_unittest.cc b/third_party/libwebrtc/pc/peer_connection_ice_unittest.cc
index 532583f307..492e108cbc 100644
--- a/third_party/libwebrtc/pc/peer_connection_ice_unittest.cc
+++ b/third_party/libwebrtc/pc/peer_connection_ice_unittest.cc
@@ -342,7 +342,7 @@ class PeerConnectionIceTest
       public ::testing::WithParamInterface<SdpSemantics> {
  protected:
   PeerConnectionIceTest() : PeerConnectionIceBaseTest(GetParam()) {
-    webrtc::metrics::Reset();
+    metrics::Reset();
   }
 };
 
@@ -514,7 +514,7 @@ TEST_P(PeerConnectionIceTest, CannotAddCandidateWhenRemoteDescriptionNotSet) {
 
   EXPECT_FALSE(caller->pc()->AddIceCandidate(jsep_candidate.get()));
   EXPECT_METRIC_THAT(
-      webrtc::metrics::Samples("WebRTC.PeerConnection.AddIceCandidate"),
+      metrics::Samples("WebRTC.PeerConnection.AddIceCandidate"),
       ElementsAre(Pair(kAddIceCandidateFailNoRemoteDescription, 2)));
 }
 
@@ -1457,7 +1457,7 @@ class PeerConnectionIceConfigTest : public ::testing::Test {
     pc_ = result.MoveValue();
   }
 
-  webrtc::test::ScopedKeyValueConfig field_trials_;
+  test::ScopedKeyValueConfig field_trials_;
   std::unique_ptr<rtc::SocketServer> socket_server_;
   rtc::AutoSocketServerThread main_thread_;
   rtc::scoped_refptr<PeerConnectionFactoryInterface> pc_factory_ = nullptr;
diff --git a/third_party/libwebrtc/pc/peer_connection_integrationtest.cc b/third_party/libwebrtc/pc/peer_connection_integrationtest.cc
index d76e5e27d5..bfff86ee93 100644
--- a/third_party/libwebrtc/pc/peer_connection_integrationtest.cc
+++ b/third_party/libwebrtc/pc/peer_connection_integrationtest.cc
@@ -28,7 +28,6 @@
 #include "absl/memory/memory.h"
 #include "absl/strings/string_view.h"
 #include "absl/types/optional.h"
-#include "api/async_resolver_factory.h"
 #include "api/candidate.h"
 #include "api/crypto/crypto_options.h"
 #include "api/dtmf_sender_interface.h"
@@ -61,7 +60,6 @@
 #include "media/base/codec.h"
 #include "media/base/media_constants.h"
 #include "media/base/stream_params.h"
-#include "p2p/base/mock_async_resolver.h"
 #include "p2p/base/port.h"
 #include "p2p/base/port_allocator.h"
 #include "p2p/base/port_interface.h"
@@ -104,6 +102,12 @@ namespace webrtc {
 
 namespace {
 
+using ::testing::AtLeast;
+using ::testing::InSequence;
+using ::testing::MockFunction;
+using ::testing::NiceMock;
+using ::testing::Return;
+
 class PeerConnectionIntegrationTest
     : public PeerConnectionIntegrationBaseTest,
       public ::testing::WithParamInterface<SdpSemantics> {
@@ -124,7 +128,7 @@ class FakeClockForTest : public rtc::ScopedFakeClock {
     // Some things use a time of "0" as a special value, so we need to start out
     // the fake clock at a nonzero time.
     // TODO(deadbeef): Fix this.
-    AdvanceTime(webrtc::TimeDelta::Seconds(1));
+    AdvanceTime(TimeDelta::Seconds(1));
   }
 
   // Explicit handle.
@@ -324,7 +328,7 @@ TEST_P(PeerConnectionIntegrationTest,
   ConnectFakeSignaling();
 
   // Add video tracks with 16:9 aspect ratio, size 1280 x 720.
-  webrtc::FakePeriodicVideoSource::Config config;
+  FakePeriodicVideoSource::Config config;
   config.width = 1280;
   config.height = 720;
   config.timestamp_offset_ms = rtc::TimeMillis();
@@ -366,7 +370,7 @@ TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithSendOnlyVideo) {
       CreateOneDirectionalPeerConnectionWrappers(/*caller_to_callee=*/true));
   ConnectFakeSignaling();
   // Add one-directional video, from caller to callee.
-  rtc::scoped_refptr<webrtc::VideoTrackInterface> caller_track =
+  rtc::scoped_refptr<VideoTrackInterface> caller_track =
       caller()->CreateLocalVideoTrack();
   caller()->AddTrack(caller_track);
   PeerConnectionInterface::RTCOfferAnswerOptions options;
@@ -391,7 +395,7 @@ TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithReceiveOnlyVideo) {
       CreateOneDirectionalPeerConnectionWrappers(/*caller_to_callee=*/false));
   ConnectFakeSignaling();
   // Add one-directional video, from callee to caller.
-  rtc::scoped_refptr<webrtc::VideoTrackInterface> callee_track =
+  rtc::scoped_refptr<VideoTrackInterface> callee_track =
       callee()->CreateLocalVideoTrack();
   callee()->AddTrack(callee_track);
   PeerConnectionInterface::RTCOfferAnswerOptions options;
@@ -414,14 +418,14 @@ TEST_P(PeerConnectionIntegrationTest,
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();
   // Add one-directional video, from caller to callee.
-  rtc::scoped_refptr<webrtc::VideoTrackInterface> caller_track =
+  rtc::scoped_refptr<VideoTrackInterface> caller_track =
       caller()->CreateLocalVideoTrack();
   caller()->AddTrack(caller_track);
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
 
   // Add receive video.
-  rtc::scoped_refptr<webrtc::VideoTrackInterface> callee_track =
+  rtc::scoped_refptr<VideoTrackInterface> callee_track =
       callee()->CreateLocalVideoTrack();
   callee()->AddTrack(callee_track);
   caller()->CreateAndSetAndSignalOffer();
@@ -438,14 +442,14 @@ TEST_P(PeerConnectionIntegrationTest,
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();
   // Add one-directional video, from callee to caller.
-  rtc::scoped_refptr<webrtc::VideoTrackInterface> callee_track =
+  rtc::scoped_refptr<VideoTrackInterface> callee_track =
       callee()->CreateLocalVideoTrack();
   callee()->AddTrack(callee_track);
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
 
   // Add send video.
-  rtc::scoped_refptr<webrtc::VideoTrackInterface> caller_track =
+  rtc::scoped_refptr<VideoTrackInterface> caller_track =
       caller()->CreateLocalVideoTrack();
   caller()->AddTrack(caller_track);
   caller()->CreateAndSetAndSignalOffer();
@@ -462,15 +466,15 @@ TEST_P(PeerConnectionIntegrationTest,
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();
   // Add send video, from caller to callee.
-  rtc::scoped_refptr<webrtc::VideoTrackInterface> caller_track =
+  rtc::scoped_refptr<VideoTrackInterface> caller_track =
       caller()->CreateLocalVideoTrack();
-  rtc::scoped_refptr<webrtc::RtpSenderInterface> caller_sender =
+  rtc::scoped_refptr<RtpSenderInterface> caller_sender =
       caller()->AddTrack(caller_track);
   // Add receive video, from callee to caller.
-  rtc::scoped_refptr<webrtc::VideoTrackInterface> callee_track =
+  rtc::scoped_refptr<VideoTrackInterface> callee_track =
       callee()->CreateLocalVideoTrack();
 
-  rtc::scoped_refptr<webrtc::RtpSenderInterface> callee_sender =
+  rtc::scoped_refptr<RtpSenderInterface> callee_sender =
       callee()->AddTrack(callee_track);
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
@@ -494,15 +498,15 @@ TEST_P(PeerConnectionIntegrationTest,
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();
   // Add send video, from caller to callee.
-  rtc::scoped_refptr<webrtc::VideoTrackInterface> caller_track =
+  rtc::scoped_refptr<VideoTrackInterface> caller_track =
       caller()->CreateLocalVideoTrack();
-  rtc::scoped_refptr<webrtc::RtpSenderInterface> caller_sender =
+  rtc::scoped_refptr<RtpSenderInterface> caller_sender =
       caller()->AddTrack(caller_track);
   // Add receive video, from callee to caller.
-  rtc::scoped_refptr<webrtc::VideoTrackInterface> callee_track =
+  rtc::scoped_refptr<VideoTrackInterface> callee_track =
       callee()->CreateLocalVideoTrack();
 
-  rtc::scoped_refptr<webrtc::RtpSenderInterface> callee_sender =
+  rtc::scoped_refptr<RtpSenderInterface> callee_sender =
       callee()->AddTrack(callee_track);
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
@@ -654,9 +658,9 @@ TEST_P(PeerConnectionIntegrationTest, RotatedVideoWithCVOExtension) {
   ConnectFakeSignaling();
   // Add rotated video tracks.
   caller()->AddTrack(
-      caller()->CreateLocalVideoTrackWithRotation(webrtc::kVideoRotation_90));
+      caller()->CreateLocalVideoTrackWithRotation(kVideoRotation_90));
   callee()->AddTrack(
-      callee()->CreateLocalVideoTrackWithRotation(webrtc::kVideoRotation_270));
+      callee()->CreateLocalVideoTrackWithRotation(kVideoRotation_270));
 
   // Wait for video frames to be received by both sides.
   caller()->CreateAndSetAndSignalOffer();
@@ -673,8 +677,8 @@ TEST_P(PeerConnectionIntegrationTest, RotatedVideoWithCVOExtension) {
   EXPECT_EQ(4.0 / 3, callee()->local_rendered_aspect_ratio());
   EXPECT_EQ(4.0 / 3, callee()->rendered_aspect_ratio());
   // Ensure that the CVO bits were surfaced to the renderer.
-  EXPECT_EQ(webrtc::kVideoRotation_270, caller()->rendered_rotation());
-  EXPECT_EQ(webrtc::kVideoRotation_90, callee()->rendered_rotation());
+  EXPECT_EQ(kVideoRotation_270, caller()->rendered_rotation());
+  EXPECT_EQ(kVideoRotation_90, callee()->rendered_rotation());
 }
 
 // Test that when the CVO extension isn't supported, video is rotated the
@@ -684,9 +688,9 @@ TEST_P(PeerConnectionIntegrationTest, RotatedVideoWithoutCVOExtension) {
   ConnectFakeSignaling();
   // Add rotated video tracks.
   caller()->AddTrack(
-      caller()->CreateLocalVideoTrackWithRotation(webrtc::kVideoRotation_90));
+      caller()->CreateLocalVideoTrackWithRotation(kVideoRotation_90));
   callee()->AddTrack(
-      callee()->CreateLocalVideoTrackWithRotation(webrtc::kVideoRotation_270));
+      callee()->CreateLocalVideoTrackWithRotation(kVideoRotation_270));
 
   // Remove the CVO extension from the offered SDP.
   callee()->SetReceivedSdpMunger([](cricket::SessionDescription* desc) {
@@ -710,8 +714,8 @@ TEST_P(PeerConnectionIntegrationTest, RotatedVideoWithoutCVOExtension) {
   EXPECT_EQ(3.0 / 4, callee()->local_rendered_aspect_ratio());
   EXPECT_EQ(3.0 / 4, callee()->rendered_aspect_ratio());
   // Expect that each endpoint is unaware of the rotation of the other endpoint.
-  EXPECT_EQ(webrtc::kVideoRotation_0, caller()->rendered_rotation());
-  EXPECT_EQ(webrtc::kVideoRotation_0, callee()->rendered_rotation());
+  EXPECT_EQ(kVideoRotation_0, caller()->rendered_rotation());
+  EXPECT_EQ(kVideoRotation_0, callee()->rendered_rotation());
 }
 
 // Test that if the answerer rejects the audio m= section, no audio is sent or
@@ -899,9 +903,9 @@ TEST_F(PeerConnectionIntegrationTestPlanB, EnableAudioAfterRejecting) {
   ConnectFakeSignaling();
 
   // Add audio track, do normal offer/answer.
-  rtc::scoped_refptr<webrtc::AudioTrackInterface> track =
+  rtc::scoped_refptr<AudioTrackInterface> track =
       caller()->CreateLocalAudioTrack();
-  rtc::scoped_refptr<webrtc::RtpSenderInterface> sender =
+  rtc::scoped_refptr<RtpSenderInterface> sender =
       caller()->pc()->AddTrack(track, {"stream"}).MoveValue();
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
@@ -974,7 +978,7 @@ TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();
   // Add one-directional video, from caller to callee.
-  rtc::scoped_refptr<webrtc::VideoTrackInterface> track =
+  rtc::scoped_refptr<VideoTrackInterface> track =
       caller()->CreateLocalVideoTrack();
 
   RtpTransceiverInit video_transceiver_init;
@@ -988,7 +992,7 @@ TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
   // Add receive direction.
   video_sender->SetDirectionWithError(RtpTransceiverDirection::kSendRecv);
 
-  rtc::scoped_refptr<webrtc::VideoTrackInterface> callee_track =
+  rtc::scoped_refptr<VideoTrackInterface> callee_track =
       callee()->CreateLocalVideoTrack();
 
   callee()->AddTrack(callee_track);
@@ -1086,11 +1090,7 @@ void ModifyPayloadTypesAndRemoveMidExtension(
                                     }),
                      extensions.end());
     media->set_rtp_header_extensions(extensions);
-    cricket::VideoContentDescription* video = media->as_video();
-    ASSERT_TRUE(video != nullptr);
-    std::vector<cricket::VideoCodec> codecs = {
-        cricket::CreateVideoCodec(pt++, "VP8")};
-    video->set_codecs(codecs);
+    media->set_codecs({cricket::CreateVideoCodec(pt++, "VP8")});
   }
 }
 
@@ -1348,11 +1348,11 @@ TEST_P(PeerConnectionIntegrationTest, NewGetStatsManyAudioAndManyVideoStreams) {
       audio_sender_1->track()->id(), video_sender_1->track()->id(),
       audio_sender_2->track()->id(), video_sender_2->track()->id()};
 
-  rtc::scoped_refptr<const webrtc::RTCStatsReport> caller_report =
+  rtc::scoped_refptr<const RTCStatsReport> caller_report =
       caller()->NewGetStats();
   ASSERT_TRUE(caller_report);
   auto outbound_stream_stats =
-      caller_report->GetStatsOfType<webrtc::RTCOutboundRtpStreamStats>();
+      caller_report->GetStatsOfType<RTCOutboundRtpStreamStats>();
   ASSERT_EQ(outbound_stream_stats.size(), 4u);
   std::vector<std::string> outbound_track_ids;
   for (const auto& stat : outbound_stream_stats) {
@@ -1373,11 +1373,11 @@ TEST_P(PeerConnectionIntegrationTest, NewGetStatsManyAudioAndManyVideoStreams) {
   }
   EXPECT_THAT(outbound_track_ids, UnorderedElementsAreArray(track_ids));
 
-  rtc::scoped_refptr<const webrtc::RTCStatsReport> callee_report =
+  rtc::scoped_refptr<const RTCStatsReport> callee_report =
       callee()->NewGetStats();
   ASSERT_TRUE(callee_report);
   auto inbound_stream_stats =
-      callee_report->GetStatsOfType<webrtc::RTCInboundRtpStreamStats>();
+      callee_report->GetStatsOfType<RTCInboundRtpStreamStats>();
   ASSERT_EQ(4u, inbound_stream_stats.size());
   std::vector<std::string> inbound_track_ids;
   for (const auto& stat : inbound_stream_stats) {
@@ -1412,11 +1412,10 @@ TEST_P(PeerConnectionIntegrationTest,
 
   // We received a frame, so we should have nonzero "bytes received" stats for
   // the unsignaled stream, if stats are working for it.
-  rtc::scoped_refptr<const webrtc::RTCStatsReport> report =
-      callee()->NewGetStats();
+  rtc::scoped_refptr<const RTCStatsReport> report = callee()->NewGetStats();
   ASSERT_NE(nullptr, report);
   auto inbound_stream_stats =
-      report->GetStatsOfType<webrtc::RTCInboundRtpStreamStats>();
+      report->GetStatsOfType<RTCInboundRtpStreamStats>();
   ASSERT_EQ(1U, inbound_stream_stats.size());
   ASSERT_TRUE(inbound_stream_stats[0]->bytes_received.is_defined());
   ASSERT_GT(*inbound_stream_stats[0]->bytes_received, 0U);
@@ -1459,12 +1458,10 @@ TEST_P(PeerConnectionIntegrationTest,
   media_expectations.CalleeExpectsSomeVideo(1);
   ASSERT_TRUE(ExpectNewFrames(media_expectations));
 
-  rtc::scoped_refptr<const webrtc::RTCStatsReport> report =
-      callee()->NewGetStats();
+  rtc::scoped_refptr<const RTCStatsReport> report = callee()->NewGetStats();
   ASSERT_NE(nullptr, report);
 
-  auto inbound_rtps =
-      report->GetStatsOfType<webrtc::RTCInboundRtpStreamStats>();
+  auto inbound_rtps = report->GetStatsOfType<RTCInboundRtpStreamStats>();
   auto index = FindFirstMediaStatsIndexByKind("audio", inbound_rtps);
   ASSERT_GE(index, 0);
   EXPECT_TRUE(inbound_rtps[index]->audio_level.is_defined());
@@ -1655,18 +1652,18 @@ TEST_P(PeerConnectionIntegrationTest, IceStatesReachCompletion) {
   callee()->AddAudioVideoTracks();
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
-  EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceGatheringComplete,
+  EXPECT_EQ_WAIT(PeerConnectionInterface::kIceGatheringComplete,
                  caller()->ice_gathering_state(), kMaxWaitForFramesMs);
-  EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceGatheringComplete,
+  EXPECT_EQ_WAIT(PeerConnectionInterface::kIceGatheringComplete,
                  callee()->ice_gathering_state(), kMaxWaitForFramesMs);
   // After the best candidate pair is selected and all candidates are signaled,
   // the ICE connection state should reach "complete".
   // TODO(deadbeef): Currently, the ICE "controlled" agent (the
   // answerer/"callee" by default) only reaches "connected". When this is
   // fixed, this test should be updated.
-  EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted,
+  EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionCompleted,
                  caller()->ice_connection_state(), kDefaultTimeout);
-  EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
+  EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionConnected,
                  callee()->ice_connection_state(), kDefaultTimeout);
 }
 
@@ -1679,9 +1676,9 @@ constexpr int kOnlyLocalPorts = cricket::PORTALLOCATOR_DISABLE_STUN |
 TEST_P(PeerConnectionIntegrationTest,
        IceStatesReachCompletionWithRemoteHostname) {
   auto caller_resolver_factory =
-      std::make_unique<NiceMock<webrtc::MockAsyncDnsResolverFactory>>();
+      std::make_unique<NiceMock<MockAsyncDnsResolverFactory>>();
   auto callee_resolver_factory =
-      std::make_unique<NiceMock<webrtc::MockAsyncDnsResolverFactory>>();
+      std::make_unique<NiceMock<MockAsyncDnsResolverFactory>>();
   auto callee_async_resolver =
       std::make_unique<NiceMock<MockAsyncDnsResolver>>();
   auto caller_async_resolver =
@@ -1695,12 +1692,12 @@ TEST_P(PeerConnectionIntegrationTest,
   // P2PTransportChannel.
   EXPECT_CALL(*caller_resolver_factory, Create())
       .WillOnce(Return(ByMove(std::move(caller_async_resolver))));
-  webrtc::PeerConnectionDependencies caller_deps(nullptr);
+  PeerConnectionDependencies caller_deps(nullptr);
   caller_deps.async_dns_resolver_factory = std::move(caller_resolver_factory);
 
   EXPECT_CALL(*callee_resolver_factory, Create())
       .WillOnce(Return(ByMove(std::move(callee_async_resolver))));
-  webrtc::PeerConnectionDependencies callee_deps(nullptr);
+  PeerConnectionDependencies callee_deps(nullptr);
   callee_deps.async_dns_resolver_factory = std::move(callee_resolver_factory);
 
   PeerConnectionInterface::RTCConfiguration config;
@@ -1719,9 +1716,9 @@ TEST_P(PeerConnectionIntegrationTest,
 
   // Enable hostname candidates with mDNS names.
   caller()->SetMdnsResponder(
-      std::make_unique<webrtc::FakeMdnsResponder>(network_thread()));
+      std::make_unique<FakeMdnsResponder>(network_thread()));
   callee()->SetMdnsResponder(
-      std::make_unique<webrtc::FakeMdnsResponder>(network_thread()));
+      std::make_unique<FakeMdnsResponder>(network_thread()));
 
   SetPortAllocatorFlags(kOnlyLocalPorts, kOnlyLocalPorts);
 
@@ -1730,18 +1727,18 @@ TEST_P(PeerConnectionIntegrationTest,
   callee()->AddAudioVideoTracks();
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
-  EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted,
+  EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionCompleted,
                  caller()->ice_connection_state(), kDefaultTimeout);
-  EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
+  EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionConnected,
                  callee()->ice_connection_state(), kDefaultTimeout);
 
   // Part of reporting the stats will occur on the network thread, so flush it
   // before checking NumEvents.
   SendTask(network_thread(), [] {});
 
-  EXPECT_METRIC_EQ(1, webrtc::metrics::NumEvents(
-                          "WebRTC.PeerConnection.CandidatePairType_UDP",
-                          webrtc::kIceCandidatePairHostNameHostName));
+  EXPECT_METRIC_EQ(
+      1, metrics::NumEvents("WebRTC.PeerConnection.CandidatePairType_UDP",
+                            kIceCandidatePairHostNameHostName));
   DestroyPeerConnections();
 }
 
@@ -1758,8 +1755,8 @@ class PeerConnectionIntegrationIceStatesTest
   }
 
   void StartStunServer(const SocketAddress& server_address) {
-    stun_server_.reset(
-        cricket::TestStunServer::Create(firewall(), server_address));
+    stun_server_ = cricket::TestStunServer::Create(firewall(), server_address,
+                                                   *network_thread());
   }
 
   bool TestIPv6() {
@@ -1805,7 +1802,7 @@ class PeerConnectionIntegrationIceStatesTest
 
  private:
   uint32_t port_allocator_flags_;
-  std::unique_ptr<cricket::TestStunServer> stun_server_;
+  cricket::TestStunServer::StunServerPtr stun_server_;
 };
 
 // Ensure FakeClockForTest is constructed first (see class for rationale).
@@ -1862,9 +1859,9 @@ TEST_P(PeerConnectionIntegrationIceStatesTest, MAYBE_VerifyBestConnection) {
   caller()->CreateAndSetAndSignalOffer();
 
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
-  EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted,
+  EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionCompleted,
                  caller()->ice_connection_state(), kDefaultTimeout);
-  EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
+  EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionConnected,
                  callee()->ice_connection_state(), kDefaultTimeout);
 
   // Part of reporting the stats will occur on the network thread, so flush it
@@ -1872,10 +1869,10 @@ TEST_P(PeerConnectionIntegrationIceStatesTest, MAYBE_VerifyBestConnection) {
   SendTask(network_thread(), [] {});
 
   // TODO(bugs.webrtc.org/9456): Fix it.
-  const int num_best_ipv4 = webrtc::metrics::NumEvents(
-      "WebRTC.PeerConnection.IPMetrics", webrtc::kBestConnections_IPv4);
-  const int num_best_ipv6 = webrtc::metrics::NumEvents(
-      "WebRTC.PeerConnection.IPMetrics", webrtc::kBestConnections_IPv6);
+  const int num_best_ipv4 = metrics::NumEvents(
+      "WebRTC.PeerConnection.IPMetrics", kBestConnections_IPv4);
+  const int num_best_ipv6 = metrics::NumEvents(
+      "WebRTC.PeerConnection.IPMetrics", kBestConnections_IPv6);
   if (TestIPv6()) {
     // When IPv6 is enabled, we should prefer an IPv6 connection over an IPv4
     // connection.
@@ -1886,12 +1883,12 @@ TEST_P(PeerConnectionIntegrationIceStatesTest, MAYBE_VerifyBestConnection) {
     EXPECT_METRIC_EQ(0, num_best_ipv6);
   }
 
-  EXPECT_METRIC_EQ(0, webrtc::metrics::NumEvents(
-                          "WebRTC.PeerConnection.CandidatePairType_UDP",
-                          webrtc::kIceCandidatePairHostHost));
-  EXPECT_METRIC_EQ(1, webrtc::metrics::NumEvents(
-                          "WebRTC.PeerConnection.CandidatePairType_UDP",
-                          webrtc::kIceCandidatePairHostPublicHostPublic));
+  EXPECT_METRIC_EQ(
+      0, metrics::NumEvents("WebRTC.PeerConnection.CandidatePairType_UDP",
+                            kIceCandidatePairHostHost));
+  EXPECT_METRIC_EQ(
+      1, metrics::NumEvents("WebRTC.PeerConnection.CandidatePairType_UDP",
+                            kIceCandidatePairHostPublicHostPublic));
 }
 
 constexpr uint32_t kFlagsIPv4NoStun = cricket::PORTALLOCATOR_DISABLE_TCP |
@@ -1931,17 +1928,17 @@ TEST_P(PeerConnectionIntegrationTest, MediaContinuesFlowingAfterIceRestart) {
   callee()->AddAudioVideoTracks();
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
-  EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted,
+  EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionCompleted,
                  caller()->ice_connection_state(), kMaxWaitForFramesMs);
-  EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
+  EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionConnected,
                  callee()->ice_connection_state(), kMaxWaitForFramesMs);
 
   // To verify that the ICE restart actually occurs, get
   // ufrag/password/candidates before and after restart.
   // Create an SDP string of the first audio candidate for both clients.
-  const webrtc::IceCandidateCollection* audio_candidates_caller =
+  const IceCandidateCollection* audio_candidates_caller =
       caller()->pc()->local_description()->candidates(0);
-  const webrtc::IceCandidateCollection* audio_candidates_callee =
+  const IceCandidateCollection* audio_candidates_callee =
       callee()->pc()->local_description()->candidates(0);
   ASSERT_GT(audio_candidates_caller->count(), 0u);
   ASSERT_GT(audio_candidates_callee->count(), 0u);
@@ -1964,9 +1961,9 @@ TEST_P(PeerConnectionIntegrationTest, MediaContinuesFlowingAfterIceRestart) {
   caller()->SetOfferAnswerOptions(IceRestartOfferAnswerOptions());
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
-  EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted,
+  EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionCompleted,
                  caller()->ice_connection_state(), kMaxWaitForFramesMs);
-  EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
+  EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionConnected,
                  callee()->ice_connection_state(), kMaxWaitForFramesMs);
 
   // Grab the ufrags/candidates again.
@@ -2141,9 +2138,9 @@ TEST_F(PeerConnectionIntegrationTestPlanB,
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kMaxWaitForActivationMs);
   // Wait for ICE to complete, without any tracks being set.
-  EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted,
+  EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionCompleted,
                  caller()->ice_connection_state(), kMaxWaitForFramesMs);
-  EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
+  EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionConnected,
                  callee()->ice_connection_state(), kMaxWaitForFramesMs);
   // Now set the tracks, and expect frames to immediately start flowing.
   EXPECT_TRUE(
@@ -2182,9 +2179,9 @@ TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kMaxWaitForActivationMs);
   // Wait for ICE to complete, without any tracks being set.
-  EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted,
+  EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionCompleted,
                  caller()->ice_connection_state(), kMaxWaitForFramesMs);
-  EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
+  EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionConnected,
                  callee()->ice_connection_state(), kMaxWaitForFramesMs);
   // Now set the tracks, and expect frames to immediately start flowing.
   auto callee_audio_sender = callee()->pc()->GetSenders()[0];
@@ -2279,21 +2276,21 @@ TEST_P(PeerConnectionIntegrationTestWithFakeClock,
   });
 
   PeerConnectionInterface::RTCConfiguration client_1_config;
-  webrtc::PeerConnectionInterface::IceServer ice_server_1;
+  PeerConnectionInterface::IceServer ice_server_1;
   ice_server_1.urls.push_back("turn:88.88.88.0:3478");
   ice_server_1.username = "test";
   ice_server_1.password = "test";
   client_1_config.servers.push_back(ice_server_1);
-  client_1_config.type = webrtc::PeerConnectionInterface::kRelay;
+  client_1_config.type = PeerConnectionInterface::kRelay;
   client_1_config.presume_writable_when_fully_relayed = true;
 
   PeerConnectionInterface::RTCConfiguration client_2_config;
-  webrtc::PeerConnectionInterface::IceServer ice_server_2;
+  PeerConnectionInterface::IceServer ice_server_2;
   ice_server_2.urls.push_back("turn:99.99.99.0:3478");
   ice_server_2.username = "test";
   ice_server_2.password = "test";
   client_2_config.servers.push_back(ice_server_2);
-  client_2_config.type = webrtc::PeerConnectionInterface::kRelay;
+  client_2_config.type = PeerConnectionInterface::kRelay;
   client_2_config.presume_writable_when_fully_relayed = true;
 
   ASSERT_TRUE(
@@ -2326,22 +2323,22 @@ TEST_P(PeerConnectionIntegrationTestWithFakeClock,
   caller()->AddAudioTrack();
 
   // Call getStats, assert there are no candidates.
-  rtc::scoped_refptr<const webrtc::RTCStatsReport> first_report =
+  rtc::scoped_refptr<const RTCStatsReport> first_report =
       caller()->NewGetStats();
   ASSERT_TRUE(first_report);
   auto first_candidate_stats =
-      first_report->GetStatsOfType<webrtc::RTCLocalIceCandidateStats>();
+      first_report->GetStatsOfType<RTCLocalIceCandidateStats>();
   ASSERT_EQ(first_candidate_stats.size(), 0u);
 
   // Create an offer at the caller and set it as remote description on the
   // callee.
   caller()->CreateAndSetAndSignalOffer();
   // Call getStats again, assert there are candidates now.
-  rtc::scoped_refptr<const webrtc::RTCStatsReport> second_report =
+  rtc::scoped_refptr<const RTCStatsReport> second_report =
       caller()->NewGetStats();
   ASSERT_TRUE(second_report);
   auto second_candidate_stats =
-      second_report->GetStatsOfType<webrtc::RTCLocalIceCandidateStats>();
+      second_report->GetStatsOfType<RTCLocalIceCandidateStats>();
   ASSERT_NE(second_candidate_stats.size(), 0u);
 
   // The fake clock ensures that no time has passed so the cache must have been
@@ -2362,17 +2359,17 @@ TEST_P(PeerConnectionIntegrationTestWithFakeClock,
                              kDefaultTimeout, FakeClock());
 
   // Call getStats, assert there are no candidates.
-  rtc::scoped_refptr<const webrtc::RTCStatsReport> first_report =
+  rtc::scoped_refptr<const RTCStatsReport> first_report =
       caller()->NewGetStats();
   ASSERT_TRUE(first_report);
   auto first_candidate_stats =
-      first_report->GetStatsOfType<webrtc::RTCRemoteIceCandidateStats>();
+      first_report->GetStatsOfType<RTCRemoteIceCandidateStats>();
   ASSERT_EQ(first_candidate_stats.size(), 0u);
 
   // Add a "fake" candidate.
   absl::optional<RTCError> result;
   caller()->pc()->AddIceCandidate(
-      absl::WrapUnique(webrtc::CreateIceCandidate(
+      absl::WrapUnique(CreateIceCandidate(
           "", 0,
           "candidate:2214029314 1 udp 2122260223 127.0.0.1 49152 typ host",
           nullptr)),
@@ -2381,11 +2378,11 @@ TEST_P(PeerConnectionIntegrationTestWithFakeClock,
   ASSERT_TRUE(result.value().ok());
 
   // Call getStats again, assert there is a remote candidate now.
-  rtc::scoped_refptr<const webrtc::RTCStatsReport> second_report =
+  rtc::scoped_refptr<const RTCStatsReport> second_report =
       caller()->NewGetStats();
   ASSERT_TRUE(second_report);
   auto second_candidate_stats =
-      second_report->GetStatsOfType<webrtc::RTCRemoteIceCandidateStats>();
+      second_report->GetStatsOfType<RTCRemoteIceCandidateStats>();
   ASSERT_EQ(second_candidate_stats.size(), 1u);
 
   // The fake clock ensures that no time has passed so the cache must have been
@@ -2413,22 +2410,22 @@ TEST_P(PeerConnectionIntegrationTest, TurnCustomizerUsedForTurnConnections) {
                    turn_server_2_external_address);
 
   PeerConnectionInterface::RTCConfiguration client_1_config;
-  webrtc::PeerConnectionInterface::IceServer ice_server_1;
+  PeerConnectionInterface::IceServer ice_server_1;
   ice_server_1.urls.push_back("turn:88.88.88.0:3478");
   ice_server_1.username = "test";
   ice_server_1.password = "test";
   client_1_config.servers.push_back(ice_server_1);
-  client_1_config.type = webrtc::PeerConnectionInterface::kRelay;
+  client_1_config.type = PeerConnectionInterface::kRelay;
   auto* customizer1 = CreateTurnCustomizer();
   client_1_config.turn_customizer = customizer1;
 
   PeerConnectionInterface::RTCConfiguration client_2_config;
-  webrtc::PeerConnectionInterface::IceServer ice_server_2;
+  PeerConnectionInterface::IceServer ice_server_2;
   ice_server_2.urls.push_back("turn:99.99.99.0:3478");
   ice_server_2.username = "test";
   ice_server_2.password = "test";
   client_2_config.servers.push_back(ice_server_2);
-  client_2_config.type = webrtc::PeerConnectionInterface::kRelay;
+  client_2_config.type = PeerConnectionInterface::kRelay;
   auto* customizer2 = CreateTurnCustomizer();
   client_2_config.turn_customizer = customizer2;
 
@@ -2460,18 +2457,18 @@ TEST_P(PeerConnectionIntegrationTest, TCPUsedForTurnConnections) {
   CreateTurnServer(turn_server_internal_address, turn_server_external_address,
                    cricket::PROTO_TCP);
 
-  webrtc::PeerConnectionInterface::IceServer ice_server;
+  PeerConnectionInterface::IceServer ice_server;
   ice_server.urls.push_back("turn:88.88.88.0:3478?transport=tcp");
   ice_server.username = "test";
   ice_server.password = "test";
 
   PeerConnectionInterface::RTCConfiguration client_1_config;
   client_1_config.servers.push_back(ice_server);
-  client_1_config.type = webrtc::PeerConnectionInterface::kRelay;
+  client_1_config.type = PeerConnectionInterface::kRelay;
 
   PeerConnectionInterface::RTCConfiguration client_2_config;
   client_2_config.servers.push_back(ice_server);
-  client_2_config.type = webrtc::PeerConnectionInterface::kRelay;
+  client_2_config.type = PeerConnectionInterface::kRelay;
 
   ASSERT_TRUE(
       CreatePeerConnectionWrappersWithConfig(client_1_config, client_2_config));
@@ -2482,7 +2479,7 @@ TEST_P(PeerConnectionIntegrationTest, TCPUsedForTurnConnections) {
   callee()->AddAudioVideoTracks();
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
-  EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
+  EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionConnected,
                  callee()->ice_connection_state(), kMaxWaitForFramesMs);
 
   MediaExpectations media_expectations;
@@ -2506,20 +2503,20 @@ TEST_P(PeerConnectionIntegrationTest,
   CreateTurnServer(turn_server_internal_address, turn_server_external_address,
                    cricket::PROTO_TLS, "88.88.88.0");
 
-  webrtc::PeerConnectionInterface::IceServer ice_server;
+  PeerConnectionInterface::IceServer ice_server;
   ice_server.urls.push_back("turns:88.88.88.0:3478?transport=tcp");
   ice_server.username = "test";
   ice_server.password = "test";
 
   PeerConnectionInterface::RTCConfiguration client_1_config;
   client_1_config.servers.push_back(ice_server);
-  client_1_config.type = webrtc::PeerConnectionInterface::kRelay;
+  client_1_config.type = PeerConnectionInterface::kRelay;
 
   PeerConnectionInterface::RTCConfiguration client_2_config;
   client_2_config.servers.push_back(ice_server);
   // Setting the type to kRelay forces the connection to go through a TURN
   // server.
-  client_2_config.type = webrtc::PeerConnectionInterface::kRelay;
+  client_2_config.type = PeerConnectionInterface::kRelay;
 
   // Get a copy to the pointer so we can verify calls later.
   rtc::TestCertificateVerifier* client_1_cert_verifier =
@@ -2530,10 +2527,10 @@ TEST_P(PeerConnectionIntegrationTest,
   client_2_cert_verifier->verify_certificate_ = true;
 
   // Create the dependencies with the test certificate verifier.
-  webrtc::PeerConnectionDependencies client_1_deps(nullptr);
+  PeerConnectionDependencies client_1_deps(nullptr);
   client_1_deps.tls_cert_verifier =
       std::unique_ptr<rtc::TestCertificateVerifier>(client_1_cert_verifier);
-  webrtc::PeerConnectionDependencies client_2_deps(nullptr);
+  PeerConnectionDependencies client_2_deps(nullptr);
   client_2_deps.tls_cert_verifier =
       std::unique_ptr<rtc::TestCertificateVerifier>(client_2_cert_verifier);
 
@@ -2644,7 +2641,7 @@ TEST_P(PeerConnectionIntegrationTest, GetSourcesAudio) {
   ASSERT_GT(receiver->GetParameters().encodings.size(), 0u);
   EXPECT_EQ(receiver->GetParameters().encodings[0].ssrc,
             sources[0].source_id());
-  EXPECT_EQ(webrtc::RtpSourceType::SSRC, sources[0].source_type());
+  EXPECT_EQ(RtpSourceType::SSRC, sources[0].source_type());
 }
 
 TEST_P(PeerConnectionIntegrationTest, GetSourcesVideo) {
@@ -2665,7 +2662,7 @@ TEST_P(PeerConnectionIntegrationTest, GetSourcesVideo) {
   ASSERT_GT(sources.size(), 0u);
   EXPECT_EQ(receiver->GetParameters().encodings[0].ssrc,
             sources[0].source_id());
-  EXPECT_EQ(webrtc::RtpSourceType::SSRC, sources[0].source_type());
+  EXPECT_EQ(RtpSourceType::SSRC, sources[0].source_type());
 }
 
 TEST_P(PeerConnectionIntegrationTest, UnsignaledSsrcGetSourcesAudio) {
@@ -2684,7 +2681,7 @@ TEST_P(PeerConnectionIntegrationTest, UnsignaledSsrcGetSourcesAudio) {
                    })(),
                    kDefaultTimeout);
   ASSERT_GT(sources.size(), 0u);
-  EXPECT_EQ(webrtc::RtpSourceType::SSRC, sources[0].source_type());
+  EXPECT_EQ(RtpSourceType::SSRC, sources[0].source_type());
 }
 
 TEST_P(PeerConnectionIntegrationTest, UnsignaledSsrcGetSourcesVideo) {
@@ -2703,7 +2700,7 @@ TEST_P(PeerConnectionIntegrationTest, UnsignaledSsrcGetSourcesVideo) {
                    })(),
                    kDefaultTimeout);
   ASSERT_GT(sources.size(), 0u);
-  EXPECT_EQ(webrtc::RtpSourceType::SSRC, sources[0].source_type());
+  EXPECT_EQ(RtpSourceType::SSRC, sources[0].source_type());
 }
 
 // Similar to the above test, except instead of waiting until GetSources() is
@@ -2728,7 +2725,7 @@ TEST_P(PeerConnectionIntegrationTest,
   std::vector<RtpSource> sources = receiver->GetSources();
   // SSRC history must not be cleared since the reception of the first frame.
   ASSERT_GT(sources.size(), 0u);
-  EXPECT_EQ(webrtc::RtpSourceType::SSRC, sources[0].source_type());
+  EXPECT_EQ(RtpSourceType::SSRC, sources[0].source_type());
 }
 
 TEST_P(PeerConnectionIntegrationTest, UnsignaledSsrcGetParametersAudio) {
@@ -2791,9 +2788,9 @@ TEST_F(PeerConnectionIntegrationTestPlanB, RemoveAndAddTrackWithNewStreamId) {
   ConnectFakeSignaling();
 
   // Add track using stream 1, do offer/answer.
-  rtc::scoped_refptr<webrtc::AudioTrackInterface> track =
+  rtc::scoped_refptr<AudioTrackInterface> track =
       caller()->CreateLocalAudioTrack();
-  rtc::scoped_refptr<webrtc::RtpSenderInterface> sender =
+  rtc::scoped_refptr<RtpSenderInterface> sender =
       caller()->AddTrack(track, {"stream_1"});
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
@@ -2819,18 +2816,70 @@ TEST_P(PeerConnectionIntegrationTest, RtcEventLogOutputWriteCalled) {
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();
 
-  auto output = std::make_unique<testing::NiceMock<MockRtcEventLogOutput>>();
-  ON_CALL(*output, IsActive()).WillByDefault(::testing::Return(true));
-  ON_CALL(*output, Write(::testing::A<absl::string_view>()))
-      .WillByDefault(::testing::Return(true));
-  EXPECT_CALL(*output, Write(::testing::A<absl::string_view>()))
-      .Times(::testing::AtLeast(1));
-  EXPECT_TRUE(caller()->pc()->StartRtcEventLog(
-      std::move(output), webrtc::RtcEventLog::kImmediateOutput));
+  auto output = std::make_unique<NiceMock<MockRtcEventLogOutput>>();
+  ON_CALL(*output, IsActive).WillByDefault(Return(true));
+  ON_CALL(*output, Write).WillByDefault(Return(true));
+  EXPECT_CALL(*output, Write).Times(AtLeast(1));
+  EXPECT_TRUE(caller()->pc()->StartRtcEventLog(std::move(output),
+                                               RtcEventLog::kImmediateOutput));
+
+  caller()->AddAudioVideoTracks();
+  caller()->CreateAndSetAndSignalOffer();
+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+}
+
+TEST_P(PeerConnectionIntegrationTest, RtcEventLogOutputWriteCalledOnStop) {
+  // This test uses check point to ensure log is written before peer connection
+  // is destroyed.
+  // https://google.github.io/googletest/gmock_cook_book.html#UsingCheckPoints
+  MockFunction<void()> test_is_complete;
+  ASSERT_TRUE(CreatePeerConnectionWrappers());
+  ConnectFakeSignaling();
+
+  auto output = std::make_unique<NiceMock<MockRtcEventLogOutput>>();
+  ON_CALL(*output, IsActive).WillByDefault(Return(true));
+  ON_CALL(*output, Write).WillByDefault(Return(true));
+  InSequence s;
+  EXPECT_CALL(*output, Write).Times(AtLeast(1));
+  EXPECT_CALL(test_is_complete, Call);
+
+  // Use large output period to prevent this test pass for the wrong reason.
+  EXPECT_TRUE(caller()->pc()->StartRtcEventLog(std::move(output),
+                                               /*output_period_ms=*/100'000));
 
   caller()->AddAudioVideoTracks();
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+
+  caller()->pc()->StopRtcEventLog();
+  test_is_complete.Call();
+}
+
+TEST_P(PeerConnectionIntegrationTest, RtcEventLogOutputWriteCalledOnClose) {
+  // This test uses check point to ensure log is written before peer connection
+  // is destroyed.
+  // https://google.github.io/googletest/gmock_cook_book.html#UsingCheckPoints
+  MockFunction<void()> test_is_complete;
+  ASSERT_TRUE(CreatePeerConnectionWrappers());
+  ConnectFakeSignaling();
+
+  auto output = std::make_unique<NiceMock<MockRtcEventLogOutput>>();
+  ON_CALL(*output, IsActive).WillByDefault(Return(true));
+  ON_CALL(*output, Write).WillByDefault(Return(true));
+  InSequence s;
+  EXPECT_CALL(*output, Write).Times(AtLeast(1));
+  EXPECT_CALL(test_is_complete, Call);
+
+  // Use large output period to prevent this test pass for the wrong reason.
+  EXPECT_TRUE(caller()->pc()->StartRtcEventLog(std::move(output),
+                                               /*output_period_ms=*/100'000));
+
+  caller()->AddAudioVideoTracks();
+  caller()->CreateAndSetAndSignalOffer();
+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+
+  caller()->pc()->Close();
+  test_is_complete.Call();
 }
 
 // Test that if candidates are only signaled by applying full session
@@ -2900,8 +2949,7 @@ TEST_P(PeerConnectionIntegrationTest, DisableAndEnableAudioPlayout) {
 
 double GetAudioEnergyStat(PeerConnectionIntegrationWrapper* pc) {
   auto report = pc->NewGetStats();
-  auto inbound_rtps =
-      report->GetStatsOfType<webrtc::RTCInboundRtpStreamStats>();
+  auto inbound_rtps = report->GetStatsOfType<RTCInboundRtpStreamStats>();
   RTC_CHECK(!inbound_rtps.empty());
   auto* inbound_rtp = inbound_rtps[0];
   if (!inbound_rtp->total_audio_energy.is_defined()) {
@@ -2974,20 +3022,20 @@ TEST_P(PeerConnectionIntegrationTest,
   ASSERT_TRUE_WAIT(DtlsConnected(), kDefaultTimeout);
   ASSERT_NE(nullptr, caller()->event_log_factory());
   ASSERT_NE(nullptr, callee()->event_log_factory());
-  webrtc::FakeRtcEventLog* caller_event_log =
+  FakeRtcEventLog* caller_event_log =
       caller()->event_log_factory()->last_log_created();
-  webrtc::FakeRtcEventLog* callee_event_log =
+  FakeRtcEventLog* callee_event_log =
       callee()->event_log_factory()->last_log_created();
   ASSERT_NE(nullptr, caller_event_log);
   ASSERT_NE(nullptr, callee_event_log);
-  int caller_ice_config_count = caller_event_log->GetEventCount(
-      webrtc::RtcEvent::Type::IceCandidatePairConfig);
-  int caller_ice_event_count = caller_event_log->GetEventCount(
-      webrtc::RtcEvent::Type::IceCandidatePairEvent);
-  int callee_ice_config_count = callee_event_log->GetEventCount(
-      webrtc::RtcEvent::Type::IceCandidatePairConfig);
-  int callee_ice_event_count = callee_event_log->GetEventCount(
-      webrtc::RtcEvent::Type::IceCandidatePairEvent);
+  int caller_ice_config_count =
+      caller_event_log->GetEventCount(RtcEvent::Type::IceCandidatePairConfig);
+  int caller_ice_event_count =
+      caller_event_log->GetEventCount(RtcEvent::Type::IceCandidatePairEvent);
+  int callee_ice_config_count =
+      callee_event_log->GetEventCount(RtcEvent::Type::IceCandidatePairConfig);
+  int callee_ice_event_count =
+      callee_event_log->GetEventCount(RtcEvent::Type::IceCandidatePairEvent);
   EXPECT_LT(0, caller_ice_config_count);
   EXPECT_LT(0, caller_ice_event_count);
   EXPECT_LT(0, callee_ice_config_count);
@@ -3001,20 +3049,20 @@ TEST_P(PeerConnectionIntegrationTest, RegatherAfterChangingIceTransportType) {
 
   CreateTurnServer(turn_server_internal_address, turn_server_external_address);
 
-  webrtc::PeerConnectionInterface::IceServer ice_server;
+  PeerConnectionInterface::IceServer ice_server;
   ice_server.urls.push_back("turn:88.88.88.0:3478");
   ice_server.username = "test";
   ice_server.password = "test";
 
   PeerConnectionInterface::RTCConfiguration caller_config;
   caller_config.servers.push_back(ice_server);
-  caller_config.type = webrtc::PeerConnectionInterface::kRelay;
+  caller_config.type = PeerConnectionInterface::kRelay;
   caller_config.continual_gathering_policy = PeerConnection::GATHER_CONTINUALLY;
   caller_config.surface_ice_candidates_on_ice_transport_type_changed = true;
 
   PeerConnectionInterface::RTCConfiguration callee_config;
   callee_config.servers.push_back(ice_server);
-  callee_config.type = webrtc::PeerConnectionInterface::kRelay;
+  callee_config.type = PeerConnectionInterface::kRelay;
   callee_config.continual_gathering_policy = PeerConnection::GATHER_CONTINUALLY;
   callee_config.surface_ice_candidates_on_ice_transport_type_changed = true;
 
@@ -3031,9 +3079,9 @@ TEST_P(PeerConnectionIntegrationTest, RegatherAfterChangingIceTransportType) {
   // kIceGatheringComplete (see
   // P2PTransportChannel::OnCandidatesAllocationDone), and consequently not
   // kIceConnectionComplete.
-  EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
+  EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionConnected,
                  caller()->ice_connection_state(), kDefaultTimeout);
-  EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
+  EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionConnected,
                  callee()->ice_connection_state(), kDefaultTimeout);
   // Note that we cannot use the metric
   // `WebRTC.PeerConnection.CandidatePairType_UDP` in this test since this
@@ -3046,7 +3094,7 @@ TEST_P(PeerConnectionIntegrationTest, RegatherAfterChangingIceTransportType) {
 
   // Loosen the caller's candidate filter.
   caller_config = caller()->pc()->GetConfiguration();
-  caller_config.type = webrtc::PeerConnectionInterface::kAll;
+  caller_config.type = PeerConnectionInterface::kAll;
   caller()->pc()->SetConfiguration(caller_config);
   // We should have gathered a new host candidate.
   EXPECT_EQ_WAIT(cricket::LOCAL_PORT_TYPE,
@@ -3054,7 +3102,7 @@ TEST_P(PeerConnectionIntegrationTest, RegatherAfterChangingIceTransportType) {
 
   // Loosen the callee's candidate filter.
   callee_config = callee()->pc()->GetConfiguration();
-  callee_config.type = webrtc::PeerConnectionInterface::kAll;
+  callee_config.type = PeerConnectionInterface::kAll;
   callee()->pc()->SetConfiguration(callee_config);
   EXPECT_EQ_WAIT(cricket::LOCAL_PORT_TYPE,
                  callee()->last_candidate_gathered().type(), kDefaultTimeout);
@@ -3084,19 +3132,19 @@ TEST_P(PeerConnectionIntegrationTest, OnIceCandidateError) {
 
   CreateTurnServer(turn_server_internal_address, turn_server_external_address);
 
-  webrtc::PeerConnectionInterface::IceServer ice_server;
+  PeerConnectionInterface::IceServer ice_server;
   ice_server.urls.push_back("turn:88.88.88.0:3478");
   ice_server.username = "test";
   ice_server.password = "123";
 
   PeerConnectionInterface::RTCConfiguration caller_config;
   caller_config.servers.push_back(ice_server);
-  caller_config.type = webrtc::PeerConnectionInterface::kRelay;
+  caller_config.type = PeerConnectionInterface::kRelay;
   caller_config.continual_gathering_policy = PeerConnection::GATHER_CONTINUALLY;
 
   PeerConnectionInterface::RTCConfiguration callee_config;
   callee_config.servers.push_back(ice_server);
-  callee_config.type = webrtc::PeerConnectionInterface::kRelay;
+  callee_config.type = PeerConnectionInterface::kRelay;
   callee_config.continual_gathering_policy = PeerConnection::GATHER_CONTINUALLY;
 
   ASSERT_TRUE(
@@ -3115,19 +3163,19 @@ TEST_P(PeerConnectionIntegrationTest, OnIceCandidateError) {
 }
 
 TEST_P(PeerConnectionIntegrationTest, OnIceCandidateErrorWithEmptyAddress) {
-  webrtc::PeerConnectionInterface::IceServer ice_server;
+  PeerConnectionInterface::IceServer ice_server;
   ice_server.urls.push_back("turn:127.0.0.1:3478?transport=tcp");
   ice_server.username = "test";
   ice_server.password = "test";
 
   PeerConnectionInterface::RTCConfiguration caller_config;
   caller_config.servers.push_back(ice_server);
-  caller_config.type = webrtc::PeerConnectionInterface::kRelay;
+  caller_config.type = PeerConnectionInterface::kRelay;
   caller_config.continual_gathering_policy = PeerConnection::GATHER_CONTINUALLY;
 
   PeerConnectionInterface::RTCConfiguration callee_config;
   callee_config.servers.push_back(ice_server);
-  callee_config.type = webrtc::PeerConnectionInterface::kRelay;
+  callee_config.type = PeerConnectionInterface::kRelay;
   callee_config.continual_gathering_policy = PeerConnection::GATHER_CONTINUALLY;
 
   ASSERT_TRUE(
@@ -3697,7 +3745,7 @@ TEST_P(PeerConnectionIntegrationTest, EndToEndRtpSenderVideoEncoderSelector) {
       CreateOneDirectionalPeerConnectionWrappers(/*caller_to_callee=*/true));
   ConnectFakeSignaling();
   // Add one-directional video, from caller to callee.
-  rtc::scoped_refptr<webrtc::VideoTrackInterface> caller_track =
+  rtc::scoped_refptr<VideoTrackInterface> caller_track =
       caller()->CreateLocalVideoTrack();
   auto sender = caller()->AddTrack(caller_track);
   PeerConnectionInterface::RTCOfferAnswerOptions options;
@@ -3722,7 +3770,7 @@ TEST_P(PeerConnectionIntegrationTest, EndToEndRtpSenderVideoEncoderSelector) {
 }
 
 int NacksReceivedCount(PeerConnectionIntegrationWrapper& pc) {
-  rtc::scoped_refptr<const webrtc::RTCStatsReport> report = pc.NewGetStats();
+  rtc::scoped_refptr<const RTCStatsReport> report = pc.NewGetStats();
   auto sender_stats = report->GetStatsOfType<RTCOutboundRtpStreamStats>();
   if (sender_stats.size() != 1) {
     ADD_FAILURE();
@@ -3735,7 +3783,7 @@ int NacksReceivedCount(PeerConnectionIntegrationWrapper& pc) {
 }
 
 int NacksSentCount(PeerConnectionIntegrationWrapper& pc) {
-  rtc::scoped_refptr<const webrtc::RTCStatsReport> report = pc.NewGetStats();
+  rtc::scoped_refptr<const RTCStatsReport> report = pc.NewGetStats();
   auto receiver_stats = report->GetStatsOfType<RTCInboundRtpStreamStats>();
   if (receiver_stats.size() != 1) {
     ADD_FAILURE();
@@ -3761,11 +3809,10 @@ TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
   // codecs.
   caller()->SetGeneratedSdpMunger([](cricket::SessionDescription* desc) {
     for (ContentInfo& content : desc->contents()) {
-      cricket::AudioContentDescription* media =
-          content.media_description()->as_audio();
-      std::vector<cricket::AudioCodec> codecs = media->codecs();
-      std::vector<cricket::AudioCodec> codecs_out;
-      for (cricket::AudioCodec codec : codecs) {
+      cricket::MediaContentDescription* media = content.media_description();
+      std::vector<cricket::Codec> codecs = media->codecs();
+      std::vector<cricket::Codec> codecs_out;
+      for (cricket::Codec codec : codecs) {
         if (codec.name == "opus") {
           codec.AddFeedbackParam(cricket::FeedbackParam(
               cricket::kRtcpFbParamNack, cricket::kParamValueEmpty));
@@ -3812,11 +3859,10 @@ TEST_F(PeerConnectionIntegrationTestUnifiedPlan, VideoPacketLossCausesNack) {
   // codecs.
   caller()->SetGeneratedSdpMunger([](cricket::SessionDescription* desc) {
     for (ContentInfo& content : desc->contents()) {
-      cricket::VideoContentDescription* media =
-          content.media_description()->as_video();
-      std::vector<cricket::VideoCodec> codecs = media->codecs();
-      std::vector<cricket::VideoCodec> codecs_out;
-      for (cricket::VideoCodec codec : codecs) {
+      cricket::MediaContentDescription* media = content.media_description();
+      std::vector<cricket::Codec> codecs = media->codecs();
+      std::vector<cricket::Codec> codecs_out;
+      for (cricket::Codec codec : codecs) {
         if (codec.name == "VP8") {
           ASSERT_TRUE(codec.HasFeedbackParam(cricket::FeedbackParam(
               cricket::kRtcpFbParamNack, cricket::kParamValueEmpty)));
diff --git a/third_party/libwebrtc/pc/peer_connection_interface_unittest.cc b/third_party/libwebrtc/pc/peer_connection_interface_unittest.cc
index 1f5ab2f449..5ee9881b84 100644
--- a/third_party/libwebrtc/pc/peer_connection_interface_unittest.cc
+++ b/third_party/libwebrtc/pc/peer_connection_interface_unittest.cc
@@ -22,9 +22,9 @@
 #include "api/audio/audio_mixer.h"
 #include "api/audio_codecs/builtin_audio_decoder_factory.h"
 #include "api/audio_codecs/builtin_audio_encoder_factory.h"
-#include "api/call/call_factory_interface.h"
 #include "api/create_peerconnection_factory.h"
 #include "api/data_channel_interface.h"
+#include "api/enable_media_with_defaults.h"
 #include "api/jsep.h"
 #include "api/media_stream_interface.h"
 #include "api/media_types.h"
@@ -53,7 +53,6 @@
 #include "media/base/media_engine.h"
 #include "media/base/stream_params.h"
 #include "media/engine/webrtc_media_engine.h"
-#include "media/engine/webrtc_media_engine_defaults.h"
 #include "media/sctp/sctp_transport_internal.h"
 #include "modules/audio_device/include/audio_device.h"
 #include "modules/audio_processing/include/audio_processing.h"
@@ -475,8 +474,7 @@ bool GetFirstSsrc(const cricket::ContentInfo* content_info, int* ssrc) {
 
 // Get the ufrags out of an SDP blob. Useful for testing ICE restart
 // behavior.
-std::vector<std::string> GetUfrags(
-    const webrtc::SessionDescriptionInterface* desc) {
+std::vector<std::string> GetUfrags(const SessionDescriptionInterface* desc) {
   std::vector<std::string> ufrags;
   for (const cricket::TransportInfo& info :
        desc->description()->transport_infos()) {
@@ -545,21 +543,19 @@ rtc::scoped_refptr<StreamCollection> CreateStreamCollection(
       StreamCollection::Create());
 
   for (int i = 0; i < number_of_streams; ++i) {
-    rtc::scoped_refptr<webrtc::MediaStreamInterface> stream(
-        webrtc::MediaStream::Create(kStreams[i]));
+    rtc::scoped_refptr<MediaStreamInterface> stream(
+        MediaStream::Create(kStreams[i]));
 
     for (int j = 0; j < tracks_per_stream; ++j) {
       // Add a local audio track.
-      rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track(
-          webrtc::AudioTrack::Create(kAudioTracks[i * tracks_per_stream + j],
-                                     nullptr));
+      rtc::scoped_refptr<AudioTrackInterface> audio_track(
+          AudioTrack::Create(kAudioTracks[i * tracks_per_stream + j], nullptr));
       stream->AddTrack(audio_track);
 
       // Add a local video track.
-      rtc::scoped_refptr<webrtc::VideoTrackInterface> video_track(
-          webrtc::VideoTrack::Create(kVideoTracks[i * tracks_per_stream + j],
-                                     webrtc::FakeVideoTrackSource::Create(),
-                                     rtc::Thread::Current()));
+      rtc::scoped_refptr<VideoTrackInterface> video_track(VideoTrack::Create(
+          kVideoTracks[i * tracks_per_stream + j],
+          FakeVideoTrackSource::Create(), rtc::Thread::Current()));
       stream->AddTrack(video_track);
     }
 
@@ -579,10 +575,10 @@ bool CompareStreamCollections(StreamCollectionInterface* s1,
     if (s1->at(i)->id() != s2->at(i)->id()) {
       return false;
     }
-    webrtc::AudioTrackVector audio_tracks1 = s1->at(i)->GetAudioTracks();
-    webrtc::AudioTrackVector audio_tracks2 = s2->at(i)->GetAudioTracks();
-    webrtc::VideoTrackVector video_tracks1 = s1->at(i)->GetVideoTracks();
-    webrtc::VideoTrackVector video_tracks2 = s2->at(i)->GetVideoTracks();
+    AudioTrackVector audio_tracks1 = s1->at(i)->GetAudioTracks();
+    AudioTrackVector audio_tracks2 = s2->at(i)->GetAudioTracks();
+    VideoTrackVector video_tracks1 = s1->at(i)->GetVideoTracks();
+    VideoTrackVector video_tracks2 = s2->at(i)->GetVideoTracks();
 
     if (audio_tracks1.size() != audio_tracks2.size()) {
       return false;
@@ -631,7 +627,7 @@ class MockTrackObserver : public ObserverInterface {
 // constraints are propagated into the PeerConnection's MediaConfig. These
 // settings are intended for MediaChannel constructors, but that is not
 // exercised by these unittest.
-class PeerConnectionFactoryForTest : public webrtc::PeerConnectionFactory {
+class PeerConnectionFactoryForTest : public PeerConnectionFactory {
  public:
   static rtc::scoped_refptr<PeerConnectionFactoryForTest>
   CreatePeerConnectionFactoryForTest() {
@@ -641,16 +637,10 @@ class PeerConnectionFactoryForTest : public webrtc::PeerConnectionFactory {
     dependencies.signaling_thread = rtc::Thread::Current();
     dependencies.task_queue_factory = CreateDefaultTaskQueueFactory();
     dependencies.trials = std::make_unique<FieldTrialBasedConfig>();
-    cricket::MediaEngineDependencies media_deps;
-    media_deps.task_queue_factory = dependencies.task_queue_factory.get();
     // Use fake audio device module since we're only testing the interface
     // level, and using a real one could make tests flaky when run in parallel.
-    media_deps.adm = FakeAudioCaptureModule::Create();
-    SetMediaEngineDefaults(&media_deps);
-    media_deps.trials = dependencies.trials.get();
-    dependencies.media_engine =
-        cricket::CreateMediaEngine(std::move(media_deps));
-    dependencies.call_factory = webrtc::CreateCallFactory();
+    dependencies.adm = FakeAudioCaptureModule::Create();
+    EnableMediaWithDefaults(dependencies);
     dependencies.event_log_factory = std::make_unique<RtcEventLogFactory>(
         dependencies.task_queue_factory.get());
 
@@ -672,7 +662,7 @@ class PeerConnectionInterfaceBaseTest : public ::testing::Test {
         main_(vss_.get()),
         sdp_semantics_(sdp_semantics) {
 #ifdef WEBRTC_ANDROID
-    webrtc::InitializeAndroidObjects();
+    InitializeAndroidObjects();
 #endif
   }
 
@@ -680,22 +670,16 @@ class PeerConnectionInterfaceBaseTest : public ::testing::Test {
     // Use fake audio capture module since we're only testing the interface
     // level, and using a real one could make tests flaky when run in parallel.
     fake_audio_capture_module_ = FakeAudioCaptureModule::Create();
-    pc_factory_ = webrtc::CreatePeerConnectionFactory(
+    pc_factory_ = CreatePeerConnectionFactory(
         rtc::Thread::Current(), rtc::Thread::Current(), rtc::Thread::Current(),
-        rtc::scoped_refptr<webrtc::AudioDeviceModule>(
-            fake_audio_capture_module_),
-        webrtc::CreateBuiltinAudioEncoderFactory(),
-        webrtc::CreateBuiltinAudioDecoderFactory(),
-        std::make_unique<webrtc::VideoEncoderFactoryTemplate<
-            webrtc::LibvpxVp8EncoderTemplateAdapter,
-            webrtc::LibvpxVp9EncoderTemplateAdapter,
-            webrtc::OpenH264EncoderTemplateAdapter,
-            webrtc::LibaomAv1EncoderTemplateAdapter>>(),
-        std::make_unique<webrtc::VideoDecoderFactoryTemplate<
-            webrtc::LibvpxVp8DecoderTemplateAdapter,
-            webrtc::LibvpxVp9DecoderTemplateAdapter,
-            webrtc::OpenH264DecoderTemplateAdapter,
-            webrtc::Dav1dDecoderTemplateAdapter>>(),
+        rtc::scoped_refptr<AudioDeviceModule>(fake_audio_capture_module_),
+        CreateBuiltinAudioEncoderFactory(), CreateBuiltinAudioDecoderFactory(),
+        std::make_unique<VideoEncoderFactoryTemplate<
+            LibvpxVp8EncoderTemplateAdapter, LibvpxVp9EncoderTemplateAdapter,
+            OpenH264EncoderTemplateAdapter, LibaomAv1EncoderTemplateAdapter>>(),
+        std::make_unique<VideoDecoderFactoryTemplate<
+            LibvpxVp8DecoderTemplateAdapter, LibvpxVp9DecoderTemplateAdapter,
+            OpenH264DecoderTemplateAdapter, Dav1dDecoderTemplateAdapter>>(),
         nullptr /* audio_mixer */, nullptr /* audio_processing */);
     ASSERT_TRUE(pc_factory_);
   }
@@ -953,8 +937,7 @@ class PeerConnectionInterfaceBaseTest : public ::testing::Test {
 
   // Call the standards-compliant GetStats function.
   bool DoGetRTCStats() {
-    auto callback =
-        rtc::make_ref_counted<webrtc::MockRTCStatsCollectorCallback>();
+    auto callback = rtc::make_ref_counted<MockRTCStatsCollectorCallback>();
     pc_->GetStats(callback.get());
     EXPECT_TRUE_WAIT(callback->called(), kTimeout);
     return callback->called();
@@ -994,14 +977,14 @@ class PeerConnectionInterfaceBaseTest : public ::testing::Test {
     std::string sdp;
     EXPECT_TRUE(offer->ToString(&sdp));
     std::unique_ptr<SessionDescriptionInterface> remote_offer(
-        webrtc::CreateSessionDescription(SdpType::kOffer, sdp));
+        CreateSessionDescription(SdpType::kOffer, sdp));
     EXPECT_TRUE(DoSetRemoteDescription(std::move(remote_offer)));
     EXPECT_EQ(PeerConnectionInterface::kHaveRemoteOffer, observer_.state_);
   }
 
   void CreateAndSetRemoteOffer(const std::string& sdp) {
     std::unique_ptr<SessionDescriptionInterface> remote_offer(
-        webrtc::CreateSessionDescription(SdpType::kOffer, sdp));
+        CreateSessionDescription(SdpType::kOffer, sdp));
     EXPECT_TRUE(DoSetRemoteDescription(std::move(remote_offer)));
     EXPECT_EQ(PeerConnectionInterface::kHaveRemoteOffer, observer_.state_);
   }
@@ -1020,7 +1003,7 @@ class PeerConnectionInterfaceBaseTest : public ::testing::Test {
     std::string sdp;
     EXPECT_TRUE(answer->ToString(&sdp));
     std::unique_ptr<SessionDescriptionInterface> new_answer(
-        webrtc::CreateSessionDescription(SdpType::kAnswer, sdp));
+        CreateSessionDescription(SdpType::kAnswer, sdp));
     EXPECT_TRUE(DoSetLocalDescription(std::move(new_answer)));
     EXPECT_EQ(PeerConnectionInterface::kStable, observer_.state_);
   }
@@ -1032,7 +1015,7 @@ class PeerConnectionInterfaceBaseTest : public ::testing::Test {
     std::string sdp;
     EXPECT_TRUE(answer->ToString(&sdp));
     std::unique_ptr<SessionDescriptionInterface> pr_answer(
-        webrtc::CreateSessionDescription(SdpType::kPrAnswer, sdp));
+        CreateSessionDescription(SdpType::kPrAnswer, sdp));
     EXPECT_TRUE(DoSetLocalDescription(std::move(pr_answer)));
     EXPECT_EQ(PeerConnectionInterface::kHaveLocalPrAnswer, observer_.state_);
   }
@@ -1057,7 +1040,7 @@ class PeerConnectionInterfaceBaseTest : public ::testing::Test {
     std::string sdp;
     EXPECT_TRUE(offer->ToString(&sdp));
     std::unique_ptr<SessionDescriptionInterface> new_offer(
-        webrtc::CreateSessionDescription(SdpType::kOffer, sdp));
+        CreateSessionDescription(SdpType::kOffer, sdp));
 
     EXPECT_TRUE(DoSetLocalDescription(std::move(new_offer)));
     EXPECT_EQ(PeerConnectionInterface::kHaveLocalOffer, observer_.state_);
@@ -1067,7 +1050,7 @@ class PeerConnectionInterfaceBaseTest : public ::testing::Test {
 
   void CreateAnswerAsRemoteDescription(const std::string& sdp) {
     std::unique_ptr<SessionDescriptionInterface> answer(
-        webrtc::CreateSessionDescription(SdpType::kAnswer, sdp));
+        CreateSessionDescription(SdpType::kAnswer, sdp));
     ASSERT_TRUE(answer);
     EXPECT_TRUE(DoSetRemoteDescription(std::move(answer)));
     EXPECT_EQ(PeerConnectionInterface::kStable, observer_.state_);
@@ -1075,12 +1058,12 @@ class PeerConnectionInterfaceBaseTest : public ::testing::Test {
 
   void CreatePrAnswerAndAnswerAsRemoteDescription(const std::string& sdp) {
     std::unique_ptr<SessionDescriptionInterface> pr_answer(
-        webrtc::CreateSessionDescription(SdpType::kPrAnswer, sdp));
+        CreateSessionDescription(SdpType::kPrAnswer, sdp));
     ASSERT_TRUE(pr_answer);
     EXPECT_TRUE(DoSetRemoteDescription(std::move(pr_answer)));
     EXPECT_EQ(PeerConnectionInterface::kHaveRemotePrAnswer, observer_.state_);
     std::unique_ptr<SessionDescriptionInterface> answer(
-        webrtc::CreateSessionDescription(SdpType::kAnswer, sdp));
+        CreateSessionDescription(SdpType::kAnswer, sdp));
     ASSERT_TRUE(answer);
     EXPECT_TRUE(DoSetRemoteDescription(std::move(answer)));
     EXPECT_EQ(PeerConnectionInterface::kStable, observer_.state_);
@@ -1124,8 +1107,8 @@ class PeerConnectionInterfaceBaseTest : public ::testing::Test {
 
     std::string mediastream_id = kStreams[0];
 
-    rtc::scoped_refptr<webrtc::MediaStreamInterface> stream(
-        webrtc::MediaStream::Create(mediastream_id));
+    rtc::scoped_refptr<MediaStreamInterface> stream(
+        MediaStream::Create(mediastream_id));
     reference_collection_->AddStream(stream);
 
     if (number_of_audio_tracks > 0) {
@@ -1149,22 +1132,20 @@ class PeerConnectionInterfaceBaseTest : public ::testing::Test {
     }
 
     return std::unique_ptr<SessionDescriptionInterface>(
-        webrtc::CreateSessionDescription(SdpType::kOffer, sdp_ms1));
+        CreateSessionDescription(SdpType::kOffer, sdp_ms1));
   }
 
   void AddAudioTrack(const std::string& track_id,
                      MediaStreamInterface* stream) {
-    rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track(
-        webrtc::AudioTrack::Create(track_id, nullptr));
+    rtc::scoped_refptr<AudioTrackInterface> audio_track(
+        AudioTrack::Create(track_id, nullptr));
     ASSERT_TRUE(stream->AddTrack(audio_track));
   }
 
   void AddVideoTrack(const std::string& track_id,
                      MediaStreamInterface* stream) {
-    rtc::scoped_refptr<webrtc::VideoTrackInterface> video_track(
-        webrtc::VideoTrack::Create(track_id,
-                                   webrtc::FakeVideoTrackSource::Create(),
-                                   rtc::Thread::Current()));
+    rtc::scoped_refptr<VideoTrackInterface> video_track(VideoTrack::Create(
+        track_id, FakeVideoTrackSource::Create(), rtc::Thread::Current()));
     ASSERT_TRUE(stream->AddTrack(video_track));
   }
 
@@ -1224,7 +1205,7 @@ class PeerConnectionInterfaceBaseTest : public ::testing::Test {
     std::string sdp;
     EXPECT_TRUE((*desc)->ToString(&sdp));
     std::unique_ptr<SessionDescriptionInterface> remote_offer(
-        webrtc::CreateSessionDescription(SdpType::kOffer, sdp));
+        CreateSessionDescription(SdpType::kOffer, sdp));
     EXPECT_TRUE(DoSetRemoteDescription(std::move(remote_offer)));
     EXPECT_EQ(PeerConnectionInterface::kHaveRemoteOffer, observer_.state_);
   }
@@ -1237,7 +1218,7 @@ class PeerConnectionInterfaceBaseTest : public ::testing::Test {
     std::string sdp;
     EXPECT_TRUE((*desc)->ToString(&sdp));
     std::unique_ptr<SessionDescriptionInterface> new_offer(
-        webrtc::CreateSessionDescription(SdpType::kOffer, sdp));
+        CreateSessionDescription(SdpType::kOffer, sdp));
 
     EXPECT_TRUE(DoSetLocalDescription(std::move(new_offer)));
     EXPECT_EQ(PeerConnectionInterface::kHaveLocalOffer, observer_.state_);
@@ -1246,8 +1227,7 @@ class PeerConnectionInterfaceBaseTest : public ::testing::Test {
   bool HasCNCodecs(const cricket::ContentInfo* content) {
     RTC_DCHECK(content);
     RTC_DCHECK(content->media_description());
-    for (const cricket::AudioCodec& codec :
-         content->media_description()->as_audio()->codecs()) {
+    for (const cricket::Codec& codec : content->media_description()->codecs()) {
       if (codec.name == "CN") {
         return true;
       }
@@ -1273,13 +1253,13 @@ class PeerConnectionInterfaceBaseTest : public ::testing::Test {
 
   rtc::SocketServer* socket_server() const { return vss_.get(); }
 
-  webrtc::test::ScopedKeyValueConfig field_trials_;
+  test::ScopedKeyValueConfig field_trials_;
   std::unique_ptr<rtc::VirtualSocketServer> vss_;
   rtc::AutoSocketServerThread main_;
   rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_;
   cricket::FakePortAllocator* port_allocator_ = nullptr;
   FakeRTCCertificateGenerator* fake_certificate_generator_ = nullptr;
-  rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> pc_factory_;
+  rtc::scoped_refptr<PeerConnectionFactoryInterface> pc_factory_;
   rtc::scoped_refptr<PeerConnectionInterface> pc_;
   MockPeerConnectionObserver observer_;
   rtc::scoped_refptr<StreamCollection> reference_collection_;
@@ -1399,22 +1379,19 @@ TEST_P(PeerConnectionInterfaceTest,
   config.prune_turn_ports = true;
 
   // Create the PC factory and PC with the above config.
-  rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> pc_factory(
-      webrtc::CreatePeerConnectionFactory(
+  rtc::scoped_refptr<PeerConnectionFactoryInterface> pc_factory(
+      CreatePeerConnectionFactory(
           rtc::Thread::Current(), rtc::Thread::Current(),
           rtc::Thread::Current(), fake_audio_capture_module_,
-          webrtc::CreateBuiltinAudioEncoderFactory(),
-          webrtc::CreateBuiltinAudioDecoderFactory(),
-          std::make_unique<webrtc::VideoEncoderFactoryTemplate<
-              webrtc::LibvpxVp8EncoderTemplateAdapter,
-              webrtc::LibvpxVp9EncoderTemplateAdapter,
-              webrtc::OpenH264EncoderTemplateAdapter,
-              webrtc::LibaomAv1EncoderTemplateAdapter>>(),
-          std::make_unique<webrtc::VideoDecoderFactoryTemplate<
-              webrtc::LibvpxVp8DecoderTemplateAdapter,
-              webrtc::LibvpxVp9DecoderTemplateAdapter,
-              webrtc::OpenH264DecoderTemplateAdapter,
-              webrtc::Dav1dDecoderTemplateAdapter>>(),
+          CreateBuiltinAudioEncoderFactory(),
+          CreateBuiltinAudioDecoderFactory(),
+          std::make_unique<VideoEncoderFactoryTemplate<
+              LibvpxVp8EncoderTemplateAdapter, LibvpxVp9EncoderTemplateAdapter,
+              OpenH264EncoderTemplateAdapter,
+              LibaomAv1EncoderTemplateAdapter>>(),
+          std::make_unique<VideoDecoderFactoryTemplate<
+              LibvpxVp8DecoderTemplateAdapter, LibvpxVp9DecoderTemplateAdapter,
+              OpenH264DecoderTemplateAdapter, Dav1dDecoderTemplateAdapter>>(),
           nullptr /* audio_mixer */, nullptr /* audio_processing */));
   PeerConnectionDependencies pc_dependencies(&observer_);
   pc_dependencies.allocator = std::move(port_allocator);
@@ -1431,7 +1408,7 @@ TEST_P(PeerConnectionInterfaceTest,
   EXPECT_TRUE(raw_port_allocator->flags() & cricket::PORTALLOCATOR_DISABLE_TCP);
   EXPECT_TRUE(raw_port_allocator->flags() &
               cricket::PORTALLOCATOR_DISABLE_COSTLY_NETWORKS);
-  EXPECT_EQ(webrtc::PRUNE_BASED_ON_PRIORITY,
+  EXPECT_EQ(PRUNE_BASED_ON_PRIORITY,
             raw_port_allocator->turn_port_prune_policy());
 }
 
@@ -1453,8 +1430,7 @@ TEST_P(PeerConnectionInterfaceTest, GetConfigurationAfterCreatePeerConnection) {
 TEST_P(PeerConnectionInterfaceTest, GetConfigurationAfterSetConfiguration) {
   PeerConnectionInterface::RTCConfiguration starting_config;
   starting_config.sdp_semantics = sdp_semantics_;
-  starting_config.bundle_policy =
-      webrtc::PeerConnection::kBundlePolicyMaxBundle;
+  starting_config.bundle_policy = PeerConnection::kBundlePolicyMaxBundle;
   CreatePeerConnection(starting_config);
 
   PeerConnectionInterface::RTCConfiguration config = pc_->GetConfiguration();
@@ -1985,7 +1961,7 @@ TEST_P(PeerConnectionInterfaceTest, CreateSctpDataChannel) {
   RTCConfiguration rtc_config;
   CreatePeerConnection(rtc_config);
 
-  webrtc::DataChannelInit config;
+  DataChannelInit config;
   auto channel = pc_->CreateDataChannelOrError("1", &config);
   EXPECT_TRUE(channel.ok());
   EXPECT_TRUE(channel.value()->reliable());
@@ -2017,7 +1993,7 @@ TEST_P(PeerConnectionInterfaceTest, CreateSctpDataChannelWhenClosed) {
   RTCConfiguration rtc_config;
   CreatePeerConnection(rtc_config);
   pc_->Close();
-  webrtc::DataChannelInit config;
+  DataChannelInit config;
   auto ret = pc_->CreateDataChannelOrError("1", &config);
   ASSERT_FALSE(ret.ok());
   EXPECT_EQ(ret.error().type(), RTCErrorType::INVALID_STATE);
@@ -2029,7 +2005,7 @@ TEST_P(PeerConnectionInterfaceTest, CreateSctpDataChannelWithMinusOne) {
   RTCConfiguration rtc_config;
   CreatePeerConnection(rtc_config);
 
-  webrtc::DataChannelInit config;
+  DataChannelInit config;
   config.maxRetransmitTime = -1;
   config.maxRetransmits = -1;
   auto channel = pc_->CreateDataChannelOrError("1", &config);
@@ -2044,7 +2020,7 @@ TEST_P(PeerConnectionInterfaceTest,
   CreatePeerConnection(rtc_config);
 
   std::string label = "test";
-  webrtc::DataChannelInit config;
+  DataChannelInit config;
   config.maxRetransmits = 0;
   config.maxRetransmitTime = 0;
 
@@ -2059,7 +2035,7 @@ TEST_P(PeerConnectionInterfaceTest,
   RTCConfiguration rtc_config;
   CreatePeerConnection(rtc_config);
 
-  webrtc::DataChannelInit config;
+  DataChannelInit config;
 
   config.id = 1;
   config.negotiated = true;
@@ -2113,7 +2089,7 @@ TEST_P(PeerConnectionInterfaceTest, DISABLED_TestRejectSctpDataChannelInAnswer)
   std::string sdp;
   EXPECT_TRUE(pc_->local_description()->ToString(&sdp));
   std::unique_ptr<SessionDescriptionInterface> answer(
-      webrtc::CreateSessionDescription(SdpType::kAnswer, sdp));
+      CreateSessionDescription(SdpType::kAnswer, sdp));
   ASSERT_TRUE(answer);
   cricket::ContentInfo* data_info =
       cricket::GetFirstDataContent(answer->description());
@@ -2132,8 +2108,7 @@ TEST_P(PeerConnectionInterfaceTest, ReceiveFireFoxOffer) {
   AddAudioTrack("audio_label");
   AddVideoTrack("video_label");
   std::unique_ptr<SessionDescriptionInterface> desc(
-      webrtc::CreateSessionDescription(SdpType::kOffer,
-                                       webrtc::kFireFoxSdpOffer, nullptr));
+      CreateSessionDescription(SdpType::kOffer, kFireFoxSdpOffer, nullptr));
   EXPECT_TRUE(DoSetSessionDescription(std::move(desc), false));
   CreateAnswerAsLocalDescription();
   ASSERT_TRUE(pc_->local_description() != nullptr);
@@ -2170,8 +2145,7 @@ TEST_P(PeerConnectionInterfaceTest, DtlsSdesFallbackNotSupported) {
   EXPECT_EQ_WAIT(1, fake_certificate_generator_->generated_certificates(),
                  kTimeout);
   std::unique_ptr<SessionDescriptionInterface> desc(
-      webrtc::CreateSessionDescription(SdpType::kOffer, kDtlsSdesFallbackSdp,
-                                       nullptr));
+      CreateSessionDescription(SdpType::kOffer, kDtlsSdesFallbackSdp, nullptr));
   EXPECT_FALSE(DoSetSessionDescription(std::move(desc), /*local=*/false));
 }
 
@@ -2184,18 +2158,17 @@ TEST_P(PeerConnectionInterfaceTest, ReceiveUpdatedAudioOfferWithBadCodecs) {
   CreateOfferAsLocalDescription();
 
   const char* answer_sdp = (sdp_semantics_ == SdpSemantics::kPlanB_DEPRECATED
-                                ? webrtc::kAudioSdpPlanB
-                                : webrtc::kAudioSdpUnifiedPlan);
+                                ? kAudioSdpPlanB
+                                : kAudioSdpUnifiedPlan);
   std::unique_ptr<SessionDescriptionInterface> answer(
-      webrtc::CreateSessionDescription(SdpType::kAnswer, answer_sdp, nullptr));
+      CreateSessionDescription(SdpType::kAnswer, answer_sdp, nullptr));
   EXPECT_TRUE(DoSetSessionDescription(std::move(answer), false));
 
-  const char* reoffer_sdp =
-      (sdp_semantics_ == SdpSemantics::kPlanB_DEPRECATED
-           ? webrtc::kAudioSdpWithUnsupportedCodecsPlanB
-           : webrtc::kAudioSdpWithUnsupportedCodecsUnifiedPlan);
+  const char* reoffer_sdp = (sdp_semantics_ == SdpSemantics::kPlanB_DEPRECATED
+                                 ? kAudioSdpWithUnsupportedCodecsPlanB
+                                 : kAudioSdpWithUnsupportedCodecsUnifiedPlan);
   std::unique_ptr<SessionDescriptionInterface> updated_offer(
-      webrtc::CreateSessionDescription(SdpType::kOffer, reoffer_sdp, nullptr));
+      CreateSessionDescription(SdpType::kOffer, reoffer_sdp, nullptr));
   EXPECT_TRUE(DoSetSessionDescription(std::move(updated_offer), false));
   CreateAnswerAsLocalDescription();
 }
@@ -2282,12 +2255,11 @@ TEST_P(PeerConnectionInterfaceTest, SetConfigurationChangesPruneTurnPortsFlag) {
   config.prune_turn_ports = false;
   CreatePeerConnection(config);
   config = pc_->GetConfiguration();
-  EXPECT_EQ(webrtc::NO_PRUNE, port_allocator_->turn_port_prune_policy());
+  EXPECT_EQ(NO_PRUNE, port_allocator_->turn_port_prune_policy());
 
   config.prune_turn_ports = true;
   EXPECT_TRUE(pc_->SetConfiguration(config).ok());
-  EXPECT_EQ(webrtc::PRUNE_BASED_ON_PRIORITY,
-            port_allocator_->turn_port_prune_policy());
+  EXPECT_EQ(PRUNE_BASED_ON_PRIORITY, port_allocator_->turn_port_prune_policy());
 }
 
 // Test that the ice check interval can be changed. This does not verify that
@@ -2556,12 +2528,12 @@ TEST_F(PeerConnectionInterfaceTestPlanB, CloseAndTestMethods) {
   std::string sdp;
   ASSERT_TRUE(pc_->remote_description()->ToString(&sdp));
   std::unique_ptr<SessionDescriptionInterface> remote_offer(
-      webrtc::CreateSessionDescription(SdpType::kOffer, sdp));
+      CreateSessionDescription(SdpType::kOffer, sdp));
   EXPECT_FALSE(DoSetRemoteDescription(std::move(remote_offer)));
 
   ASSERT_TRUE(pc_->local_description()->ToString(&sdp));
   std::unique_ptr<SessionDescriptionInterface> local_offer(
-      webrtc::CreateSessionDescription(SdpType::kOffer, sdp));
+      CreateSessionDescription(SdpType::kOffer, sdp));
   EXPECT_FALSE(DoSetLocalDescription(std::move(local_offer)));
 }
 
@@ -2621,10 +2593,10 @@ TEST_F(PeerConnectionInterfaceTestPlanB,
                                        reference_collection_.get()));
   rtc::scoped_refptr<AudioTrackInterface> audio_track2 =
       observer_.remote_streams()->at(0)->GetAudioTracks()[1];
-  EXPECT_EQ(webrtc::MediaStreamTrackInterface::kLive, audio_track2->state());
+  EXPECT_EQ(MediaStreamTrackInterface::kLive, audio_track2->state());
   rtc::scoped_refptr<VideoTrackInterface> video_track2 =
       observer_.remote_streams()->at(0)->GetVideoTracks()[1];
-  EXPECT_EQ(webrtc::MediaStreamTrackInterface::kLive, video_track2->state());
+  EXPECT_EQ(MediaStreamTrackInterface::kLive, video_track2->state());
 
   // Remove the extra audio and video tracks.
   std::unique_ptr<SessionDescriptionInterface> desc_ms2 =
@@ -2638,10 +2610,10 @@ TEST_F(PeerConnectionInterfaceTestPlanB,
   EXPECT_TRUE(CompareStreamCollections(observer_.remote_streams(),
                                        reference_collection_.get()));
   // Track state may be updated asynchronously.
-  EXPECT_EQ_WAIT(webrtc::MediaStreamTrackInterface::kEnded,
-                 audio_track2->state(), kTimeout);
-  EXPECT_EQ_WAIT(webrtc::MediaStreamTrackInterface::kEnded,
-                 video_track2->state(), kTimeout);
+  EXPECT_EQ_WAIT(MediaStreamTrackInterface::kEnded, audio_track2->state(),
+                 kTimeout);
+  EXPECT_EQ_WAIT(MediaStreamTrackInterface::kEnded, video_track2->state(),
+                 kTimeout);
 }
 
 // This tests that remote tracks are ended if a local session description is set
@@ -2659,7 +2631,7 @@ TEST_P(PeerConnectionInterfaceTest, RejectMediaContent) {
 
   rtc::scoped_refptr<MediaStreamTrackInterface> remote_audio =
       audio_receiver->track();
-  EXPECT_EQ(webrtc::MediaStreamTrackInterface::kLive, remote_audio->state());
+  EXPECT_EQ(MediaStreamTrackInterface::kLive, remote_audio->state());
   rtc::scoped_refptr<MediaStreamTrackInterface> remote_video =
       video_receiver->track();
   EXPECT_EQ(MediaStreamTrackInterface::kLive, remote_video->state());
@@ -2703,8 +2675,8 @@ TEST_F(PeerConnectionInterfaceTestPlanB, RemoveTrackThenRejectMediaContent) {
   remote_stream->RemoveTrack(remote_stream->GetAudioTracks()[0]);
 
   std::unique_ptr<SessionDescriptionInterface> local_answer(
-      webrtc::CreateSessionDescription(SdpType::kAnswer,
-                                       GetSdpStringWithStream1(), nullptr));
+      CreateSessionDescription(SdpType::kAnswer, GetSdpStringWithStream1(),
+                               nullptr));
   cricket::ContentInfo* video_info =
       local_answer->description()->GetContentByName("video");
   video_info->rejected = true;
@@ -2993,9 +2965,9 @@ TEST_P(PeerConnectionInterfaceTest,
   ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
   // Grab a copy of the offer before it gets passed into the PC.
   std::unique_ptr<SessionDescriptionInterface> modified_offer =
-      webrtc::CreateSessionDescription(
-          webrtc::SdpType::kOffer, offer->session_id(),
-          offer->session_version(), offer->description()->Clone());
+      CreateSessionDescription(SdpType::kOffer, offer->session_id(),
+                               offer->session_version(),
+                               offer->description()->Clone());
   EXPECT_TRUE(DoSetLocalDescription(std::move(offer)));
 
   auto senders = pc_->GetSenders();
@@ -3051,8 +3023,8 @@ TEST_F(PeerConnectionInterfaceTestPlanB,
   EXPECT_TRUE(ContainsSender(senders, kVideoTracks[0], kStreams[0]));
 
   // Add a new MediaStream but with the same tracks as in the first stream.
-  rtc::scoped_refptr<webrtc::MediaStreamInterface> stream_1(
-      webrtc::MediaStream::Create(kStreams[1]));
+  rtc::scoped_refptr<MediaStreamInterface> stream_1(
+      MediaStream::Create(kStreams[1]));
   stream_1->AddTrack(stream_collection->at(0)->GetVideoTracks()[0]);
   stream_1->AddTrack(stream_collection->at(0)->GetAudioTracks()[0]);
   pc_->AddStream(stream_1.get());
@@ -3173,9 +3145,9 @@ TEST_P(PeerConnectionInterfaceTest, SetConfigurationCausingPartialIceRestart) {
   EXPECT_TRUE(pc_->SetConfiguration(config).ok());
 
   // Do ICE restart for the first m= section, initiated by remote peer.
-  std::unique_ptr<webrtc::SessionDescriptionInterface> remote_offer(
-      webrtc::CreateSessionDescription(SdpType::kOffer,
-                                       GetSdpStringWithStream1(), nullptr));
+  std::unique_ptr<SessionDescriptionInterface> remote_offer(
+      CreateSessionDescription(SdpType::kOffer, GetSdpStringWithStream1(),
+                               nullptr));
   ASSERT_TRUE(remote_offer);
   remote_offer->description()->transport_infos()[0].description.ice_ufrag =
       "modified";
@@ -3221,7 +3193,7 @@ TEST_P(PeerConnectionInterfaceTest, CurrentAndPendingDescriptions) {
 
   // Set remote pranswer.
   std::unique_ptr<SessionDescriptionInterface> remote_pranswer(
-      webrtc::CreateSessionDescription(SdpType::kPrAnswer, sdp));
+      CreateSessionDescription(SdpType::kPrAnswer, sdp));
   SessionDescriptionInterface* remote_pranswer_ptr = remote_pranswer.get();
   EXPECT_TRUE(DoSetRemoteDescription(std::move(remote_pranswer)));
   EXPECT_EQ(local_offer_ptr, pc_->pending_local_description());
@@ -3231,7 +3203,7 @@ TEST_P(PeerConnectionInterfaceTest, CurrentAndPendingDescriptions) {
 
   // Set remote answer.
   std::unique_ptr<SessionDescriptionInterface> remote_answer(
-      webrtc::CreateSessionDescription(SdpType::kAnswer, sdp));
+      CreateSessionDescription(SdpType::kAnswer, sdp));
   SessionDescriptionInterface* remote_answer_ptr = remote_answer.get();
   EXPECT_TRUE(DoSetRemoteDescription(std::move(remote_answer)));
   EXPECT_EQ(nullptr, pc_->pending_local_description());
@@ -3241,7 +3213,7 @@ TEST_P(PeerConnectionInterfaceTest, CurrentAndPendingDescriptions) {
 
   // Set remote offer.
   std::unique_ptr<SessionDescriptionInterface> remote_offer(
-      webrtc::CreateSessionDescription(SdpType::kOffer, sdp));
+      CreateSessionDescription(SdpType::kOffer, sdp));
   SessionDescriptionInterface* remote_offer_ptr = remote_offer.get();
   EXPECT_TRUE(DoSetRemoteDescription(std::move(remote_offer)));
   EXPECT_EQ(remote_offer_ptr, pc_->pending_remote_description());
@@ -3251,7 +3223,7 @@ TEST_P(PeerConnectionInterfaceTest, CurrentAndPendingDescriptions) {
 
   // Set local pranswer.
   std::unique_ptr<SessionDescriptionInterface> local_pranswer(
-      webrtc::CreateSessionDescription(SdpType::kPrAnswer, sdp));
+      CreateSessionDescription(SdpType::kPrAnswer, sdp));
   SessionDescriptionInterface* local_pranswer_ptr = local_pranswer.get();
   EXPECT_TRUE(DoSetLocalDescription(std::move(local_pranswer)));
   EXPECT_EQ(remote_offer_ptr, pc_->pending_remote_description());
@@ -3261,7 +3233,7 @@ TEST_P(PeerConnectionInterfaceTest, CurrentAndPendingDescriptions) {
 
   // Set local answer.
   std::unique_ptr<SessionDescriptionInterface> local_answer(
-      webrtc::CreateSessionDescription(SdpType::kAnswer, sdp));
+      CreateSessionDescription(SdpType::kAnswer, sdp));
   SessionDescriptionInterface* local_answer_ptr = local_answer.get();
   EXPECT_TRUE(DoSetLocalDescription(std::move(local_answer)));
   EXPECT_EQ(nullptr, pc_->pending_remote_description());
@@ -3280,9 +3252,8 @@ TEST_P(PeerConnectionInterfaceTest,
   // The RtcEventLog will be reset when the PeerConnection is closed.
   pc_->Close();
 
-  EXPECT_FALSE(
-      pc_->StartRtcEventLog(std::make_unique<webrtc::RtcEventLogOutputNull>(),
-                            webrtc::RtcEventLog::kImmediateOutput));
+  EXPECT_FALSE(pc_->StartRtcEventLog(std::make_unique<RtcEventLogOutputNull>(),
+                                     RtcEventLog::kImmediateOutput));
   pc_->StopRtcEventLog();
 }
 
diff --git a/third_party/libwebrtc/pc/peer_connection_jsep_unittest.cc b/third_party/libwebrtc/pc/peer_connection_jsep_unittest.cc
index 1369253ad6..786b519a93 100644
--- a/third_party/libwebrtc/pc/peer_connection_jsep_unittest.cc
+++ b/third_party/libwebrtc/pc/peer_connection_jsep_unittest.cc
@@ -20,7 +20,7 @@
 
 #include "absl/strings/string_view.h"
 #include "absl/types/optional.h"
-#include "api/call/call_factory_interface.h"
+#include "api/enable_media_with_defaults.h"
 #include "api/field_trials_view.h"
 #include "api/jsep.h"
 #include "api/media_stream_interface.h"
@@ -40,7 +40,6 @@
 #include "media/base/media_engine.h"
 #include "media/base/stream_params.h"
 #include "media/engine/webrtc_media_engine.h"
-#include "media/engine/webrtc_media_engine_defaults.h"
 #include "modules/audio_device/include/audio_device.h"
 #include "p2p/base/p2p_constants.h"
 #include "p2p/base/port_allocator.h"
@@ -84,13 +83,8 @@ PeerConnectionFactoryDependencies CreatePeerConnectionFactoryDependencies() {
   dependencies.signaling_thread = rtc::Thread::Current();
   dependencies.task_queue_factory = CreateDefaultTaskQueueFactory();
   dependencies.trials = std::make_unique<FieldTrialBasedConfig>();
-  cricket::MediaEngineDependencies media_deps;
-  media_deps.task_queue_factory = dependencies.task_queue_factory.get();
-  media_deps.adm = FakeAudioCaptureModule::Create();
-  media_deps.trials = dependencies.trials.get();
-  SetMediaEngineDefaults(&media_deps);
-  dependencies.media_engine = cricket::CreateMediaEngine(std::move(media_deps));
-  dependencies.call_factory = CreateCallFactory();
+  dependencies.adm = FakeAudioCaptureModule::Create();
+  EnableMediaWithDefaults(dependencies);
   dependencies.sctp_factory = std::make_unique<FakeSctpTransportFactory>();
   return dependencies;
 }
diff --git a/third_party/libwebrtc/pc/peer_connection_media_unittest.cc b/third_party/libwebrtc/pc/peer_connection_media_unittest.cc
index 485541981e..387094cc4f 100644
--- a/third_party/libwebrtc/pc/peer_connection_media_unittest.cc
+++ b/third_party/libwebrtc/pc/peer_connection_media_unittest.cc
@@ -27,7 +27,6 @@
 #include "absl/algorithm/container.h"
 #include "absl/types/optional.h"
 #include "api/audio_options.h"
-#include "api/call/call_factory_interface.h"
 #include "api/jsep.h"
 #include "api/media_types.h"
 #include "api/peer_connection_interface.h"
@@ -57,6 +56,7 @@
 #include "pc/rtp_media_utils.h"
 #include "pc/rtp_transceiver.h"
 #include "pc/session_description.h"
+#include "pc/test/enable_fake_media.h"
 #include "pc/test/mock_peer_connection_observers.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/rtc_certificate_generator.h"
@@ -82,9 +82,9 @@ using ::testing::NotNull;
 using ::testing::Values;
 
 cricket::MediaSendChannelInterface* SendChannelInternal(
-    rtc::scoped_refptr<webrtc::RtpTransceiverInterface> transceiver) {
-  auto transceiver_with_internal = static_cast<rtc::RefCountedObject<
-      webrtc::RtpTransceiverProxyWithInternal<webrtc::RtpTransceiver>>*>(
+    rtc::scoped_refptr<RtpTransceiverInterface> transceiver) {
+  auto transceiver_with_internal = static_cast<
+      rtc::RefCountedObject<RtpTransceiverProxyWithInternal<RtpTransceiver>>*>(
       transceiver.get());
   auto transceiver_internal =
       static_cast<RtpTransceiver*>(transceiver_with_internal->internal());
@@ -92,9 +92,9 @@ cricket::MediaSendChannelInterface* SendChannelInternal(
 }
 
 cricket::MediaReceiveChannelInterface* ReceiveChannelInternal(
-    rtc::scoped_refptr<webrtc::RtpTransceiverInterface> transceiver) {
-  auto transceiver_with_internal = static_cast<rtc::RefCountedObject<
-      webrtc::RtpTransceiverProxyWithInternal<webrtc::RtpTransceiver>>*>(
+    rtc::scoped_refptr<RtpTransceiverInterface> transceiver) {
+  auto transceiver_with_internal = static_cast<
+      rtc::RefCountedObject<RtpTransceiverProxyWithInternal<RtpTransceiver>>*>(
       transceiver.get());
   auto transceiver_internal =
       static_cast<RtpTransceiver*>(transceiver_with_internal->internal());
@@ -102,22 +102,22 @@ cricket::MediaReceiveChannelInterface* ReceiveChannelInternal(
 }
 
 cricket::FakeVideoMediaSendChannel* VideoMediaSendChannel(
-    rtc::scoped_refptr<webrtc::RtpTransceiverInterface> transceiver) {
+    rtc::scoped_refptr<RtpTransceiverInterface> transceiver) {
   return static_cast<cricket::FakeVideoMediaSendChannel*>(
       SendChannelInternal(transceiver));
 }
 cricket::FakeVideoMediaReceiveChannel* VideoMediaReceiveChannel(
-    rtc::scoped_refptr<webrtc::RtpTransceiverInterface> transceiver) {
+    rtc::scoped_refptr<RtpTransceiverInterface> transceiver) {
   return static_cast<cricket::FakeVideoMediaReceiveChannel*>(
       ReceiveChannelInternal(transceiver));
 }
 cricket::FakeVoiceMediaSendChannel* VoiceMediaSendChannel(
-    rtc::scoped_refptr<webrtc::RtpTransceiverInterface> transceiver) {
+    rtc::scoped_refptr<RtpTransceiverInterface> transceiver) {
   return static_cast<cricket::FakeVoiceMediaSendChannel*>(
       SendChannelInternal(transceiver));
 }
 cricket::FakeVoiceMediaReceiveChannel* VoiceMediaReceiveChannel(
-    rtc::scoped_refptr<webrtc::RtpTransceiverInterface> transceiver) {
+    rtc::scoped_refptr<RtpTransceiverInterface> transceiver) {
   return static_cast<cricket::FakeVoiceMediaReceiveChannel*>(
       ReceiveChannelInternal(transceiver));
 }
@@ -173,8 +173,7 @@ class PeerConnectionMediaBaseTest : public ::testing::Test {
     factory_dependencies.worker_thread = rtc::Thread::Current();
     factory_dependencies.signaling_thread = rtc::Thread::Current();
     factory_dependencies.task_queue_factory = CreateDefaultTaskQueueFactory();
-    factory_dependencies.media_engine = std::move(media_engine);
-    factory_dependencies.call_factory = CreateCallFactory();
+    EnableFakeMedia(factory_dependencies, std::move(media_engine));
     factory_dependencies.event_log_factory =
         std::make_unique<RtcEventLogFactory>(
             factory_dependencies.task_queue_factory.get());
@@ -255,7 +254,7 @@ class PeerConnectionMediaBaseTest : public ::testing::Test {
     return sdp_semantics_ == SdpSemantics::kUnifiedPlan;
   }
 
-  webrtc::test::ScopedKeyValueConfig field_trials_;
+  test::ScopedKeyValueConfig field_trials_;
   std::unique_ptr<rtc::VirtualSocketServer> vss_;
   rtc::AutoSocketServerThread main_;
   const SdpSemantics sdp_semantics_;
@@ -1496,10 +1495,10 @@ TEST_P(PeerConnectionMediaTest, RedFmtpPayloadDifferentRedundancy) {
 }
 
 template <typename C>
-bool CompareCodecs(const std::vector<webrtc::RtpCodecCapability>& capabilities,
+bool CompareCodecs(const std::vector<RtpCodecCapability>& capabilities,
                    const std::vector<C>& codecs) {
   bool capability_has_rtx =
-      absl::c_any_of(capabilities, [](const webrtc::RtpCodecCapability& codec) {
+      absl::c_any_of(capabilities, [](const RtpCodecCapability& codec) {
         return codec.name == cricket::kRtxCodecName;
       });
   bool codecs_has_rtx = absl::c_any_of(codecs, [](const C& codec) {
@@ -1511,16 +1510,16 @@ bool CompareCodecs(const std::vector<webrtc::RtpCodecCapability>& capabilities,
       codecs, std::back_inserter(codecs_no_rtx),
       [](const C& codec) { return codec.name != cricket::kRtxCodecName; });
 
-  std::vector<webrtc::RtpCodecCapability> capabilities_no_rtx;
+  std::vector<RtpCodecCapability> capabilities_no_rtx;
   absl::c_copy_if(capabilities, std::back_inserter(capabilities_no_rtx),
-                  [](const webrtc::RtpCodecCapability& codec) {
+                  [](const RtpCodecCapability& codec) {
                     return codec.name != cricket::kRtxCodecName;
                   });
 
   return capability_has_rtx == codecs_has_rtx &&
          absl::c_equal(
              capabilities_no_rtx, codecs_no_rtx,
-             [](const webrtc::RtpCodecCapability& capability, const C& codec) {
+             [](const RtpCodecCapability& capability, const C& codec) {
                return codec.MatchesRtpCodec(capability);
              });
 }
@@ -1539,9 +1538,9 @@ TEST_F(PeerConnectionMediaTestUnifiedPlan,
   auto capabilities = caller->pc_factory()->GetRtpSenderCapabilities(
       cricket::MediaType::MEDIA_TYPE_AUDIO);
 
-  std::vector<webrtc::RtpCodecCapability> codecs;
+  std::vector<RtpCodecCapability> codecs;
   absl::c_copy_if(capabilities.codecs, std::back_inserter(codecs),
-                  [](const webrtc::RtpCodecCapability& codec) {
+                  [](const RtpCodecCapability& codec) {
                     return codec.name.find("_only_") != std::string::npos;
                   });
 
@@ -1562,9 +1561,9 @@ TEST_F(PeerConnectionMediaTestUnifiedPlan,
   auto capabilities = caller->pc_factory()->GetRtpReceiverCapabilities(
       cricket::MediaType::MEDIA_TYPE_AUDIO);
 
-  std::vector<webrtc::RtpCodecCapability> codecs;
+  std::vector<RtpCodecCapability> codecs;
   absl::c_copy_if(capabilities.codecs, std::back_inserter(codecs),
-                  [](const webrtc::RtpCodecCapability& codec) {
+                  [](const RtpCodecCapability& codec) {
                     return codec.name.find("_only_") != std::string::npos;
                   });
 
@@ -1612,7 +1611,7 @@ TEST_F(PeerConnectionMediaTestUnifiedPlan,
   auto codecs_only_rtx_red_fec = codecs;
   auto it = std::remove_if(codecs_only_rtx_red_fec.begin(),
                            codecs_only_rtx_red_fec.end(),
-                           [](const webrtc::RtpCodecCapability& codec) {
+                           [](const RtpCodecCapability& codec) {
                              return !(codec.name == cricket::kRtxCodecName ||
                                       codec.name == cricket::kRedCodecName ||
                                       codec.name == cricket::kUlpfecCodecName);
@@ -1639,7 +1638,6 @@ TEST_F(PeerConnectionMediaTestUnifiedPlan, SetCodecPreferencesAllAudioCodecs) {
   auto codecs = offer->description()
                     ->contents()[0]
                     .media_description()
-                    ->as_audio()
                     ->codecs();
   EXPECT_TRUE(CompareCodecs(sender_audio_codecs, codecs));
 }
@@ -1652,7 +1650,7 @@ TEST_F(PeerConnectionMediaTestUnifiedPlan,
       caller->pc_factory()
           ->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_AUDIO)
           .codecs;
-  std::vector<webrtc::RtpCodecCapability> empty_codecs = {};
+  std::vector<RtpCodecCapability> empty_codecs = {};
 
   auto audio_transceiver = caller->pc()->GetTransceivers().front();
 
@@ -1662,7 +1660,6 @@ TEST_F(PeerConnectionMediaTestUnifiedPlan,
   auto codecs = offer->description()
                     ->contents()[0]
                     .media_description()
-                    ->as_audio()
                     ->codecs();
   EXPECT_TRUE(CompareCodecs(sender_audio_codecs, codecs));
 }
@@ -1707,7 +1704,7 @@ TEST_F(PeerConnectionMediaTestUnifiedPlan,
   auto codecs_only_rtx_red_fec = codecs;
   auto it = std::remove_if(codecs_only_rtx_red_fec.begin(),
                            codecs_only_rtx_red_fec.end(),
-                           [](const webrtc::RtpCodecCapability& codec) {
+                           [](const RtpCodecCapability& codec) {
                              return !(codec.name == cricket::kRtxCodecName ||
                                       codec.name == cricket::kRedCodecName ||
                                       codec.name == cricket::kUlpfecCodecName);
@@ -1734,7 +1731,6 @@ TEST_F(PeerConnectionMediaTestUnifiedPlan, SetCodecPreferencesAllVideoCodecs) {
   auto codecs = offer->description()
                     ->contents()[0]
                     .media_description()
-                    ->as_video()
                     ->codecs();
   EXPECT_TRUE(CompareCodecs(sender_video_codecs, codecs));
 }
@@ -1748,7 +1744,7 @@ TEST_F(PeerConnectionMediaTestUnifiedPlan,
           ->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_VIDEO)
           .codecs;
 
-  std::vector<webrtc::RtpCodecCapability> empty_codecs = {};
+  std::vector<RtpCodecCapability> empty_codecs = {};
 
   auto video_transceiver = caller->pc()->GetTransceivers().front();
 
@@ -1758,7 +1754,6 @@ TEST_F(PeerConnectionMediaTestUnifiedPlan,
   auto codecs = offer->description()
                     ->contents()[0]
                     .media_description()
-                    ->as_video()
                     ->codecs();
   EXPECT_TRUE(CompareCodecs(sender_video_codecs, codecs));
 }
@@ -1787,7 +1782,6 @@ TEST_F(PeerConnectionMediaTestUnifiedPlan,
   auto codecs = offer->description()
                     ->contents()[0]
                     .media_description()
-                    ->as_video()
                     ->codecs();
   EXPECT_TRUE(CompareCodecs(single_codec, codecs));
 }
@@ -1818,7 +1812,7 @@ TEST_F(PeerConnectionMediaTestUnifiedPlan, SetCodecPreferencesVideoWithRtx) {
   auto video_codecs_vpx_rtx = sender_video_codecs;
   auto it =
       std::remove_if(video_codecs_vpx_rtx.begin(), video_codecs_vpx_rtx.end(),
-                     [](const webrtc::RtpCodecCapability& codec) {
+                     [](const RtpCodecCapability& codec) {
                        return codec.name != cricket::kRtxCodecName &&
                               codec.name != cricket::kVp8CodecName &&
                               codec.name != cricket::kVp9CodecName;
@@ -1832,7 +1826,6 @@ TEST_F(PeerConnectionMediaTestUnifiedPlan, SetCodecPreferencesVideoWithRtx) {
   auto codecs = offer->description()
                     ->contents()[0]
                     .media_description()
-                    ->as_video()
                     ->codecs();
 
   EXPECT_TRUE(CompareCodecs(video_codecs_vpx_rtx, codecs));
@@ -1867,7 +1860,7 @@ TEST_F(PeerConnectionMediaTestUnifiedPlan,
 
   auto video_codecs_vpx = video_codecs;
   auto it = std::remove_if(video_codecs_vpx.begin(), video_codecs_vpx.end(),
-                           [](const webrtc::RtpCodecCapability& codec) {
+                           [](const RtpCodecCapability& codec) {
                              return codec.name != cricket::kVp8CodecName &&
                                     codec.name != cricket::kVp9CodecName;
                            });
@@ -1879,7 +1872,6 @@ TEST_F(PeerConnectionMediaTestUnifiedPlan,
   auto codecs = offer->description()
                     ->contents()[0]
                     .media_description()
-                    ->as_video()
                     ->codecs();
 
   EXPECT_EQ(codecs.size(), 2u);  // VP8, VP9
@@ -1890,7 +1882,7 @@ TEST_F(PeerConnectionMediaTestUnifiedPlan,
   auto recv_transceiver = callee->pc()->GetTransceivers().front();
   auto video_codecs_vp8_rtx = video_codecs;
   it = std::remove_if(video_codecs_vp8_rtx.begin(), video_codecs_vp8_rtx.end(),
-                      [](const webrtc::RtpCodecCapability& codec) {
+                      [](const RtpCodecCapability& codec) {
                         bool r = codec.name != cricket::kVp8CodecName &&
                                  codec.name != cricket::kRtxCodecName;
                         return r;
@@ -1904,7 +1896,6 @@ TEST_F(PeerConnectionMediaTestUnifiedPlan,
   auto recv_codecs = answer->description()
                          ->contents()[0]
                          .media_description()
-                         ->as_video()
                          ->codecs();
   EXPECT_EQ(recv_codecs.size(), 1u);  // VP8
 }
@@ -1937,7 +1928,7 @@ TEST_F(PeerConnectionMediaTestUnifiedPlan,
 
   auto video_codecs_vpx = video_codecs;
   auto it = std::remove_if(video_codecs_vpx.begin(), video_codecs_vpx.end(),
-                           [](const webrtc::RtpCodecCapability& codec) {
+                           [](const RtpCodecCapability& codec) {
                              return codec.name != cricket::kVp8CodecName &&
                                     codec.name != cricket::kVp9CodecName;
                            });
@@ -1952,7 +1943,6 @@ TEST_F(PeerConnectionMediaTestUnifiedPlan,
   auto codecs = offer->description()
                     ->contents()[0]
                     .media_description()
-                    ->as_video()
                     ->codecs();
   EXPECT_EQ(codecs.size(), 2u);  // VP9, VP8
   EXPECT_TRUE(CompareCodecs(video_codecs_vpx, codecs));
@@ -1967,7 +1957,6 @@ TEST_F(PeerConnectionMediaTestUnifiedPlan,
   auto recv_codecs = answer->description()
                          ->contents()[0]
                          .media_description()
-                         ->as_video()
                          ->codecs();
 
   EXPECT_TRUE(CompareCodecs(video_codecs_vpx_reverse, recv_codecs));
diff --git a/third_party/libwebrtc/pc/peer_connection_rampup_tests.cc b/third_party/libwebrtc/pc/peer_connection_rampup_tests.cc
index 545a1d53d0..0fd3c27f7d 100644
--- a/third_party/libwebrtc/pc/peer_connection_rampup_tests.cc
+++ b/third_party/libwebrtc/pc/peer_connection_rampup_tests.cc
@@ -201,7 +201,7 @@ class PeerConnectionRampUpTest : public ::testing::Test {
     fake_network_managers_.emplace_back(fake_network_manager);
 
     auto observer = std::make_unique<MockPeerConnectionObserver>();
-    webrtc::PeerConnectionDependencies dependencies(observer.get());
+    PeerConnectionDependencies dependencies(observer.get());
     cricket::BasicPortAllocator* port_allocator =
         new cricket::BasicPortAllocator(fake_network_manager,
                                         firewall_socket_factory_.get());
diff --git a/third_party/libwebrtc/pc/peer_connection_rtp_unittest.cc b/third_party/libwebrtc/pc/peer_connection_rtp_unittest.cc
index b93e5923bb..1a97a4bb44 100644
--- a/third_party/libwebrtc/pc/peer_connection_rtp_unittest.cc
+++ b/third_party/libwebrtc/pc/peer_connection_rtp_unittest.cc
@@ -75,13 +75,13 @@ using ::testing::UnorderedElementsAre;
 using ::testing::Values;
 
 template <typename MethodFunctor>
-class OnSuccessObserver : public webrtc::SetRemoteDescriptionObserverInterface {
+class OnSuccessObserver : public SetRemoteDescriptionObserverInterface {
  public:
   explicit OnSuccessObserver(MethodFunctor on_success)
       : on_success_(std::move(on_success)) {}
 
-  // webrtc::SetRemoteDescriptionObserverInterface implementation.
-  void OnSetRemoteDescriptionComplete(webrtc::RTCError error) override {
+  // SetRemoteDescriptionObserverInterface implementation.
+  void OnSetRemoteDescriptionComplete(RTCError error) override {
     RTC_CHECK(error.ok());
     on_success_();
   }
@@ -113,7 +113,7 @@ class PeerConnectionRtpBaseTest : public ::testing::Test {
                                             Dav1dDecoderTemplateAdapter>>(),
             nullptr /* audio_mixer */,
             nullptr /* audio_processing */)) {
-    webrtc::metrics::Reset();
+    metrics::Reset();
   }
 
   std::unique_ptr<PeerConnectionWrapper> CreatePeerConnection() {
@@ -201,7 +201,7 @@ class PeerConnectionRtpTestUnifiedPlan : public PeerConnectionRtpBaseTest {
   }
 };
 
-// These tests cover `webrtc::PeerConnectionObserver` callbacks firing upon
+// These tests cover `PeerConnectionObserver` callbacks firing upon
 // setting the remote description.
 
 TEST_P(PeerConnectionRtpTest, AddTrackWithoutStreamFiresOnAddTrack) {
@@ -934,8 +934,8 @@ TEST_P(PeerConnectionRtpTest,
   auto caller = CreatePeerConnection();
   auto callee = CreatePeerConnection();
 
-  rtc::scoped_refptr<webrtc::MockSetSessionDescriptionObserver> observer =
-      rtc::make_ref_counted<webrtc::MockSetSessionDescriptionObserver>();
+  rtc::scoped_refptr<MockSetSessionDescriptionObserver> observer =
+      rtc::make_ref_counted<MockSetSessionDescriptionObserver>();
 
   auto offer = caller->CreateOfferAndSetAsLocal();
   callee->pc()->SetRemoteDescription(observer.get(), offer.release());
diff --git a/third_party/libwebrtc/pc/peer_connection_signaling_unittest.cc b/third_party/libwebrtc/pc/peer_connection_signaling_unittest.cc
index 8ca59fc20c..190fb38b43 100644
--- a/third_party/libwebrtc/pc/peer_connection_signaling_unittest.cc
+++ b/third_party/libwebrtc/pc/peer_connection_signaling_unittest.cc
@@ -896,8 +896,8 @@ TEST_P(PeerConnectionSignalingTest, UnsupportedContentType) {
       "m=bogus 9 FOO 0 8\r\n"
       "c=IN IP4 0.0.0.0\r\n"
       "a=mid:bogusmid\r\n";
-  std::unique_ptr<webrtc::SessionDescriptionInterface> remote_description =
-      webrtc::CreateSessionDescription(SdpType::kOffer, sdp, nullptr);
+  std::unique_ptr<SessionDescriptionInterface> remote_description =
+      CreateSessionDescription(SdpType::kOffer, sdp, nullptr);
 
   EXPECT_TRUE(caller->SetRemoteDescription(std::move(remote_description)));
 
@@ -977,20 +977,17 @@ TEST_P(PeerConnectionSignalingTest, ReceiveFlexFec) {
       "a=ssrc-group:FEC-FR 1224551896 1953032773\r\n"
       "a=ssrc:1224551896 cname:/exJcmhSLpyu9FgV\r\n"
       "a=ssrc:1953032773 cname:/exJcmhSLpyu9FgV\r\n";
-  std::unique_ptr<webrtc::SessionDescriptionInterface> remote_description =
-      webrtc::CreateSessionDescription(SdpType::kOffer, sdp, nullptr);
+  std::unique_ptr<SessionDescriptionInterface> remote_description =
+      CreateSessionDescription(SdpType::kOffer, sdp, nullptr);
 
   EXPECT_TRUE(caller->SetRemoteDescription(std::move(remote_description)));
 
   auto answer = caller->CreateAnswer();
   ASSERT_EQ(answer->description()->contents().size(), 1u);
-  ASSERT_NE(
-      answer->description()->contents()[0].media_description()->as_video(),
-      nullptr);
+  ASSERT_NE(answer->description()->contents()[0].media_description(), nullptr);
   auto codecs = answer->description()
                     ->contents()[0]
                     .media_description()
-                    ->as_video()
                     ->codecs();
   ASSERT_EQ(codecs.size(), 2u);
   EXPECT_EQ(codecs[1].name, "flexfec-03");
@@ -1033,20 +1030,17 @@ TEST_P(PeerConnectionSignalingTest, ReceiveFlexFecReoffer) {
       "a=ssrc-group:FEC-FR 1224551896 1953032773\r\n"
       "a=ssrc:1224551896 cname:/exJcmhSLpyu9FgV\r\n"
       "a=ssrc:1953032773 cname:/exJcmhSLpyu9FgV\r\n";
-  std::unique_ptr<webrtc::SessionDescriptionInterface> remote_description =
-      webrtc::CreateSessionDescription(SdpType::kOffer, sdp, nullptr);
+  std::unique_ptr<SessionDescriptionInterface> remote_description =
+      CreateSessionDescription(SdpType::kOffer, sdp, nullptr);
 
   EXPECT_TRUE(caller->SetRemoteDescription(std::move(remote_description)));
 
   auto answer = caller->CreateAnswer();
   ASSERT_EQ(answer->description()->contents().size(), 1u);
-  ASSERT_NE(
-      answer->description()->contents()[0].media_description()->as_video(),
-      nullptr);
+  ASSERT_NE(answer->description()->contents()[0].media_description(), nullptr);
   auto codecs = answer->description()
                     ->contents()[0]
                     .media_description()
-                    ->as_video()
                     ->codecs();
   ASSERT_EQ(codecs.size(), 2u);
   EXPECT_EQ(codecs[1].name, "flexfec-03");
@@ -1059,7 +1053,6 @@ TEST_P(PeerConnectionSignalingTest, ReceiveFlexFecReoffer) {
   auto offer_codecs = offer->description()
                           ->contents()[0]
                           .media_description()
-                          ->as_video()
                           ->codecs();
   auto flexfec_it = std::find_if(
       offer_codecs.begin(), offer_codecs.end(),
@@ -1104,8 +1097,8 @@ TEST_P(PeerConnectionSignalingTest, MidAttributeMaxLength) {
       "a=rtcp-fb:102 nack\r\n"
       "a=rtcp-fb:102 nack pli\r\n"
       "a=ssrc:1224551896 cname:/exJcmhSLpyu9FgV\r\n";
-  std::unique_ptr<webrtc::SessionDescriptionInterface> remote_description =
-      webrtc::CreateSessionDescription(SdpType::kOffer, sdp, nullptr);
+  std::unique_ptr<SessionDescriptionInterface> remote_description =
+      CreateSessionDescription(SdpType::kOffer, sdp, nullptr);
 
   EXPECT_FALSE(caller->SetRemoteDescription(std::move(remote_description)));
 }
@@ -1339,8 +1332,8 @@ TEST_F(PeerConnectionSignalingUnifiedPlanTest, RtxReofferApt) {
       "a=rtcp-fb:102 nack\r\n"
       "a=rtcp-fb:102 nack pli\r\n"
       "a=ssrc:1224551896 cname:/exJcmhSLpyu9FgV\r\n";
-  std::unique_ptr<webrtc::SessionDescriptionInterface> remote_description =
-      webrtc::CreateSessionDescription(SdpType::kOffer, sdp, nullptr);
+  std::unique_ptr<SessionDescriptionInterface> remote_description =
+      CreateSessionDescription(SdpType::kOffer, sdp, nullptr);
 
   EXPECT_TRUE(callee->SetRemoteDescription(std::move(remote_description)));
 
@@ -1353,7 +1346,6 @@ TEST_F(PeerConnectionSignalingUnifiedPlanTest, RtxReofferApt) {
   auto codecs = reoffer->description()
                     ->contents()[0]
                     .media_description()
-                    ->as_video()
                     ->codecs();
   ASSERT_GT(codecs.size(), 2u);
   EXPECT_EQ(codecs[0].name, "VP8");
diff --git a/third_party/libwebrtc/pc/peer_connection_simulcast_unittest.cc b/third_party/libwebrtc/pc/peer_connection_simulcast_unittest.cc
index 6b6a96c473..bffb5d9e9f 100644
--- a/third_party/libwebrtc/pc/peer_connection_simulcast_unittest.cc
+++ b/third_party/libwebrtc/pc/peer_connection_simulcast_unittest.cc
@@ -220,7 +220,7 @@ class PeerConnectionSimulcastMetricsTests
     : public PeerConnectionSimulcastTests,
       public ::testing::WithParamInterface<int> {
  protected:
-  PeerConnectionSimulcastMetricsTests() { webrtc::metrics::Reset(); }
+  PeerConnectionSimulcastMetricsTests() { metrics::Reset(); }
 };
 #endif
 
diff --git a/third_party/libwebrtc/pc/peer_connection_svc_integrationtest.cc b/third_party/libwebrtc/pc/peer_connection_svc_integrationtest.cc
index 672f3eef99..32ca451866 100644
--- a/third_party/libwebrtc/pc/peer_connection_svc_integrationtest.cc
+++ b/third_party/libwebrtc/pc/peer_connection_svc_integrationtest.cc
@@ -37,14 +37,13 @@ class PeerConnectionSVCIntegrationTest
       : PeerConnectionIntegrationBaseTest(SdpSemantics::kUnifiedPlan) {}
 
   RTCError SetCodecPreferences(
-      rtc::scoped_refptr<webrtc::RtpTransceiverInterface> transceiver,
+      rtc::scoped_refptr<RtpTransceiverInterface> transceiver,
       absl::string_view codec_name) {
-    webrtc::RtpCapabilities capabilities =
+    RtpCapabilities capabilities =
         caller()->pc_factory()->GetRtpSenderCapabilities(
             cricket::MEDIA_TYPE_VIDEO);
     std::vector<RtpCodecCapability> codecs;
-    for (const webrtc::RtpCodecCapability& codec_capability :
-         capabilities.codecs) {
+    for (const RtpCodecCapability& codec_capability : capabilities.codecs) {
       if (codec_capability.name == codec_name)
         codecs.push_back(codec_capability);
     }
@@ -55,8 +54,8 @@ class PeerConnectionSVCIntegrationTest
 TEST_F(PeerConnectionSVCIntegrationTest, AddTransceiverAcceptsL1T1) {
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();
-  webrtc::RtpTransceiverInit init;
-  webrtc::RtpEncodingParameters encoding_parameters;
+  RtpTransceiverInit init;
+  RtpEncodingParameters encoding_parameters;
   encoding_parameters.scalability_mode = "L1T1";
   init.send_encodings.push_back(encoding_parameters);
   auto transceiver_or_error =
@@ -67,8 +66,8 @@ TEST_F(PeerConnectionSVCIntegrationTest, AddTransceiverAcceptsL1T1) {
 TEST_F(PeerConnectionSVCIntegrationTest, AddTransceiverAcceptsL3T3) {
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();
-  webrtc::RtpTransceiverInit init;
-  webrtc::RtpEncodingParameters encoding_parameters;
+  RtpTransceiverInit init;
+  RtpEncodingParameters encoding_parameters;
   encoding_parameters.scalability_mode = "L3T3";
   init.send_encodings.push_back(encoding_parameters);
   auto transceiver_or_error =
@@ -80,33 +79,32 @@ TEST_F(PeerConnectionSVCIntegrationTest,
        AddTransceiverRejectsUnknownScalabilityMode) {
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();
-  webrtc::RtpTransceiverInit init;
-  webrtc::RtpEncodingParameters encoding_parameters;
+  RtpTransceiverInit init;
+  RtpEncodingParameters encoding_parameters;
   encoding_parameters.scalability_mode = "FOOBAR";
   init.send_encodings.push_back(encoding_parameters);
   auto transceiver_or_error =
       caller()->pc()->AddTransceiver(caller()->CreateLocalVideoTrack(), init);
   EXPECT_FALSE(transceiver_or_error.ok());
   EXPECT_EQ(transceiver_or_error.error().type(),
-            webrtc::RTCErrorType::UNSUPPORTED_OPERATION);
+            RTCErrorType::UNSUPPORTED_OPERATION);
 }
 
 TEST_F(PeerConnectionSVCIntegrationTest, SetParametersAcceptsL1T3WithVP8) {
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();
 
-  webrtc::RtpCapabilities capabilities =
+  RtpCapabilities capabilities =
       caller()->pc_factory()->GetRtpSenderCapabilities(
           cricket::MEDIA_TYPE_VIDEO);
   std::vector<RtpCodecCapability> vp8_codec;
-  for (const webrtc::RtpCodecCapability& codec_capability :
-       capabilities.codecs) {
+  for (const RtpCodecCapability& codec_capability : capabilities.codecs) {
     if (codec_capability.name == cricket::kVp8CodecName)
       vp8_codec.push_back(codec_capability);
   }
 
-  webrtc::RtpTransceiverInit init;
-  webrtc::RtpEncodingParameters encoding_parameters;
+  RtpTransceiverInit init;
+  RtpEncodingParameters encoding_parameters;
   init.send_encodings.push_back(encoding_parameters);
   auto transceiver_or_error =
       caller()->pc()->AddTransceiver(caller()->CreateLocalVideoTrack(), init);
@@ -114,7 +112,7 @@ TEST_F(PeerConnectionSVCIntegrationTest, SetParametersAcceptsL1T3WithVP8) {
   auto transceiver = transceiver_or_error.MoveValue();
   EXPECT_TRUE(transceiver->SetCodecPreferences(vp8_codec).ok());
 
-  webrtc::RtpParameters parameters = transceiver->sender()->GetParameters();
+  RtpParameters parameters = transceiver->sender()->GetParameters();
   ASSERT_EQ(parameters.encodings.size(), 1u);
   parameters.encodings[0].scalability_mode = "L1T3";
   auto result = transceiver->sender()->SetParameters(parameters);
@@ -125,8 +123,8 @@ TEST_F(PeerConnectionSVCIntegrationTest, SetParametersRejectsL3T3WithVP8) {
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();
 
-  webrtc::RtpTransceiverInit init;
-  webrtc::RtpEncodingParameters encoding_parameters;
+  RtpTransceiverInit init;
+  RtpEncodingParameters encoding_parameters;
   init.send_encodings.push_back(encoding_parameters);
   auto transceiver_or_error =
       caller()->pc()->AddTransceiver(caller()->CreateLocalVideoTrack(), init);
@@ -134,12 +132,12 @@ TEST_F(PeerConnectionSVCIntegrationTest, SetParametersRejectsL3T3WithVP8) {
   auto transceiver = transceiver_or_error.MoveValue();
   EXPECT_TRUE(SetCodecPreferences(transceiver, cricket::kVp8CodecName).ok());
 
-  webrtc::RtpParameters parameters = transceiver->sender()->GetParameters();
+  RtpParameters parameters = transceiver->sender()->GetParameters();
   ASSERT_EQ(parameters.encodings.size(), 1u);
   parameters.encodings[0].scalability_mode = "L3T3";
   auto result = transceiver->sender()->SetParameters(parameters);
   EXPECT_FALSE(result.ok());
-  EXPECT_EQ(result.type(), webrtc::RTCErrorType::INVALID_MODIFICATION);
+  EXPECT_EQ(result.type(), RTCErrorType::INVALID_MODIFICATION);
 }
 
 TEST_F(PeerConnectionSVCIntegrationTest,
@@ -147,8 +145,8 @@ TEST_F(PeerConnectionSVCIntegrationTest,
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();
 
-  webrtc::RtpTransceiverInit init;
-  webrtc::RtpEncodingParameters encoding_parameters;
+  RtpTransceiverInit init;
+  RtpEncodingParameters encoding_parameters;
   init.send_encodings.push_back(encoding_parameters);
   auto transceiver_or_error =
       caller()->pc()->AddTransceiver(caller()->CreateLocalVideoTrack(), init);
@@ -159,7 +157,7 @@ TEST_F(PeerConnectionSVCIntegrationTest,
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
 
-  webrtc::RtpParameters parameters = transceiver->sender()->GetParameters();
+  RtpParameters parameters = transceiver->sender()->GetParameters();
   ASSERT_EQ(parameters.encodings.size(), 1u);
   parameters.encodings[0].scalability_mode = "L1T3";
   auto result = transceiver->sender()->SetParameters(parameters);
@@ -171,8 +169,8 @@ TEST_F(PeerConnectionSVCIntegrationTest,
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();
 
-  webrtc::RtpTransceiverInit init;
-  webrtc::RtpEncodingParameters encoding_parameters;
+  RtpTransceiverInit init;
+  RtpEncodingParameters encoding_parameters;
   init.send_encodings.push_back(encoding_parameters);
   auto transceiver_or_error =
       caller()->pc()->AddTransceiver(caller()->CreateLocalVideoTrack(), init);
@@ -183,7 +181,7 @@ TEST_F(PeerConnectionSVCIntegrationTest,
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
 
-  webrtc::RtpParameters parameters = transceiver->sender()->GetParameters();
+  RtpParameters parameters = transceiver->sender()->GetParameters();
   ASSERT_EQ(parameters.encodings.size(), 1u);
   parameters.encodings[0].scalability_mode = "L3T3";
   auto result = transceiver->sender()->SetParameters(parameters);
@@ -195,8 +193,8 @@ TEST_F(PeerConnectionSVCIntegrationTest,
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();
 
-  webrtc::RtpTransceiverInit init;
-  webrtc::RtpEncodingParameters encoding_parameters;
+  RtpTransceiverInit init;
+  RtpEncodingParameters encoding_parameters;
   init.send_encodings.push_back(encoding_parameters);
   auto transceiver_or_error =
       caller()->pc()->AddTransceiver(caller()->CreateLocalVideoTrack(), init);
@@ -207,12 +205,12 @@ TEST_F(PeerConnectionSVCIntegrationTest,
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
 
-  webrtc::RtpParameters parameters = transceiver->sender()->GetParameters();
+  RtpParameters parameters = transceiver->sender()->GetParameters();
   ASSERT_EQ(parameters.encodings.size(), 1u);
   parameters.encodings[0].scalability_mode = "L3T3";
   auto result = transceiver->sender()->SetParameters(parameters);
   EXPECT_FALSE(result.ok());
-  EXPECT_EQ(result.type(), webrtc::RTCErrorType::INVALID_MODIFICATION);
+  EXPECT_EQ(result.type(), RTCErrorType::INVALID_MODIFICATION);
 }
 
 TEST_F(PeerConnectionSVCIntegrationTest,
@@ -220,8 +218,8 @@ TEST_F(PeerConnectionSVCIntegrationTest,
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();
 
-  webrtc::RtpTransceiverInit init;
-  webrtc::RtpEncodingParameters encoding_parameters;
+  RtpTransceiverInit init;
+  RtpEncodingParameters encoding_parameters;
   init.send_encodings.push_back(encoding_parameters);
   auto transceiver_or_error =
       caller()->pc()->AddTransceiver(caller()->CreateLocalVideoTrack(), init);
@@ -232,27 +230,27 @@ TEST_F(PeerConnectionSVCIntegrationTest,
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
 
-  webrtc::RtpParameters parameters = transceiver->sender()->GetParameters();
+  RtpParameters parameters = transceiver->sender()->GetParameters();
   ASSERT_EQ(parameters.encodings.size(), 1u);
   parameters.encodings[0].scalability_mode = "FOOBAR";
   auto result = transceiver->sender()->SetParameters(parameters);
   EXPECT_FALSE(result.ok());
-  EXPECT_EQ(result.type(), webrtc::RTCErrorType::INVALID_MODIFICATION);
+  EXPECT_EQ(result.type(), RTCErrorType::INVALID_MODIFICATION);
 }
 
 TEST_F(PeerConnectionSVCIntegrationTest, FallbackToL1Tx) {
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();
 
-  webrtc::RtpTransceiverInit init;
-  webrtc::RtpEncodingParameters encoding_parameters;
+  RtpTransceiverInit init;
+  RtpEncodingParameters encoding_parameters;
   init.send_encodings.push_back(encoding_parameters);
   auto transceiver_or_error =
       caller()->pc()->AddTransceiver(caller()->CreateLocalVideoTrack(), init);
   ASSERT_TRUE(transceiver_or_error.ok());
   auto caller_transceiver = transceiver_or_error.MoveValue();
 
-  webrtc::RtpCapabilities capabilities =
+  RtpCapabilities capabilities =
       caller()->pc_factory()->GetRtpSenderCapabilities(
           cricket::MEDIA_TYPE_VIDEO);
   std::vector<RtpCodecCapability> send_codecs = capabilities.codecs;
@@ -267,8 +265,7 @@ TEST_F(PeerConnectionSVCIntegrationTest, FallbackToL1Tx) {
   caller_transceiver->SetCodecPreferences(send_codecs);
 
   // L3T3 should be supported by VP9
-  webrtc::RtpParameters parameters =
-      caller_transceiver->sender()->GetParameters();
+  RtpParameters parameters = caller_transceiver->sender()->GetParameters();
   ASSERT_EQ(parameters.encodings.size(), 1u);
   parameters.encodings[0].scalability_mode = "L3T3";
   auto result = caller_transceiver->sender()->SetParameters(parameters);
diff --git a/third_party/libwebrtc/pc/peer_connection_wrapper.cc b/third_party/libwebrtc/pc/peer_connection_wrapper.cc
index 44f4256b10..557d0c8422 100644
--- a/third_party/libwebrtc/pc/peer_connection_wrapper.cc
+++ b/third_party/libwebrtc/pc/peer_connection_wrapper.cc
@@ -339,8 +339,7 @@ bool PeerConnectionWrapper::IsIceConnected() {
   return observer()->ice_connected_;
 }
 
-rtc::scoped_refptr<const webrtc::RTCStatsReport>
-PeerConnectionWrapper::GetStats() {
+rtc::scoped_refptr<const RTCStatsReport> PeerConnectionWrapper::GetStats() {
   auto callback = rtc::make_ref_counted<MockRTCStatsCollectorCallback>();
   pc()->GetStats(callback.get());
   EXPECT_TRUE_WAIT(callback->called(), kDefaultTimeout);
diff --git a/third_party/libwebrtc/pc/rtc_stats_collector.cc b/third_party/libwebrtc/pc/rtc_stats_collector.cc
index 0797ba2a76..2bac176aac 100644
--- a/third_party/libwebrtc/pc/rtc_stats_collector.cc
+++ b/third_party/libwebrtc/pc/rtc_stats_collector.cc
@@ -336,7 +336,7 @@ const char* QualityLimitationReasonToRTCQualityLimitationReason(
 
 std::map<std::string, double>
 QualityLimitationDurationToRTCQualityLimitationDuration(
-    std::map<webrtc::QualityLimitationReason, int64_t> durations_ms) {
+    std::map<QualityLimitationReason, int64_t> durations_ms) {
   std::map<std::string, double> result;
   // The internal duration is defined in milliseconds while the spec defines
   // the value in seconds:
@@ -513,7 +513,7 @@ std::unique_ptr<RTCInboundRtpStreamStats> CreateInboundAudioStreamStats(
 
 std::unique_ptr<RTCAudioPlayoutStats> CreateAudioPlayoutStats(
     const AudioDeviceModule::Stats& audio_device_stats,
-    webrtc::Timestamp timestamp) {
+    Timestamp timestamp) {
   auto stats = std::make_unique<RTCAudioPlayoutStats>(
       /*id=*/kAudioPlayoutSingletonId, timestamp);
   stats->synthesized_samples_duration =
diff --git a/third_party/libwebrtc/pc/rtc_stats_collector.h b/third_party/libwebrtc/pc/rtc_stats_collector.h
index e94d23944c..4c68e77086 100644
--- a/third_party/libwebrtc/pc/rtc_stats_collector.h
+++ b/third_party/libwebrtc/pc/rtc_stats_collector.h
@@ -317,7 +317,7 @@ class RTCStatsCollector : public rtc::RefCountInterface {
     uint32_t data_channels_closed;
     // Identifies channels that have been opened, whose internal id is stored in
     // the set until they have been fully closed.
-    webrtc::flat_set<int> opened_data_channels;
+    flat_set<int> opened_data_channels;
   };
   InternalRecord internal_record_;
 };
diff --git a/third_party/libwebrtc/pc/rtc_stats_collector_unittest.cc b/third_party/libwebrtc/pc/rtc_stats_collector_unittest.cc
index 37821ac829..055be6fe99 100644
--- a/third_party/libwebrtc/pc/rtc_stats_collector_unittest.cc
+++ b/third_party/libwebrtc/pc/rtc_stats_collector_unittest.cc
@@ -263,9 +263,9 @@ class FakeAudioTrackForStats : public MediaStreamTrack<AudioTrackInterface> {
   std::string kind() const override {
     return MediaStreamTrackInterface::kAudioKind;
   }
-  webrtc::AudioSourceInterface* GetSource() const override { return nullptr; }
-  void AddSink(webrtc::AudioTrackSinkInterface* sink) override {}
-  void RemoveSink(webrtc::AudioTrackSinkInterface* sink) override {}
+  AudioSourceInterface* GetSource() const override { return nullptr; }
+  void AddSink(AudioTrackSinkInterface* sink) override {}
+  void RemoveSink(AudioTrackSinkInterface* sink) override {}
   bool GetSignalLevel(int* level) override { return false; }
   rtc::scoped_refptr<AudioProcessorInterface> GetAudioProcessor() override {
     return processor_;
@@ -2030,7 +2030,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCIceCandidatePairStats) {
   EXPECT_TRUE(report->Get(*expected_pair.transport_id));
 
   // Set bandwidth and "GetStats" again.
-  webrtc::Call::Stats call_stats;
+  Call::Stats call_stats;
   const int kSendBandwidth = 888;
   call_stats.send_bandwidth_bps = kSendBandwidth;
   const int kRecvBandwidth = 999;
@@ -2339,12 +2339,9 @@ TEST_F(RTCStatsCollectorTest, CollectRTCInboundRtpStreamStats_Video) {
   video_media_info.receivers[0].key_frames_decoded = 3;
   video_media_info.receivers[0].frames_dropped = 13;
   video_media_info.receivers[0].qp_sum = absl::nullopt;
-  video_media_info.receivers[0].total_decode_time =
-      webrtc::TimeDelta::Seconds(9);
-  video_media_info.receivers[0].total_processing_delay =
-      webrtc::TimeDelta::Millis(600);
-  video_media_info.receivers[0].total_assembly_time =
-      webrtc::TimeDelta::Millis(500);
+  video_media_info.receivers[0].total_decode_time = TimeDelta::Seconds(9);
+  video_media_info.receivers[0].total_processing_delay = TimeDelta::Millis(600);
+  video_media_info.receivers[0].total_assembly_time = TimeDelta::Millis(500);
   video_media_info.receivers[0].frames_assembled_from_multiple_packets = 23;
   video_media_info.receivers[0].total_inter_frame_delay = 0.123;
   video_media_info.receivers[0].total_squared_inter_frame_delay = 0.00456;
@@ -2617,12 +2614,12 @@ TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRtpStreamStats_Video) {
   video_media_info.senders[0].key_frames_encoded = 3;
   video_media_info.senders[0].total_encode_time_ms = 9000;
   video_media_info.senders[0].total_encoded_bytes_target = 1234;
-  video_media_info.senders[0].total_packet_send_delay =
-      webrtc::TimeDelta::Seconds(10);
+  video_media_info.senders[0].total_packet_send_delay = TimeDelta::Seconds(10);
   video_media_info.senders[0].quality_limitation_reason =
       QualityLimitationReason::kBandwidth;
-  video_media_info.senders[0].quality_limitation_durations_ms
-      [webrtc::QualityLimitationReason::kBandwidth] = 300;
+  video_media_info.senders[0]
+      .quality_limitation_durations_ms[QualityLimitationReason::kBandwidth] =
+      300;
   video_media_info.senders[0].quality_limitation_resolution_changes = 56u;
   video_media_info.senders[0].qp_sum = absl::nullopt;
   video_media_info.senders[0].content_type = VideoContentType::UNSPECIFIED;
diff --git a/third_party/libwebrtc/pc/rtc_stats_traversal_unittest.cc b/third_party/libwebrtc/pc/rtc_stats_traversal_unittest.cc
index 72ad255564..8205ebedc0 100644
--- a/third_party/libwebrtc/pc/rtc_stats_traversal_unittest.cc
+++ b/third_party/libwebrtc/pc/rtc_stats_traversal_unittest.cc
@@ -47,7 +47,7 @@ class RTCStatsTraversalTest : public ::testing::Test {
     for (const RTCStats* start_node : start_nodes) {
       start_ids.push_back(start_node->id());
     }
-    result_ = webrtc::TakeReferencedStats(initial_report_, start_ids);
+    result_ = ::webrtc::TakeReferencedStats(initial_report_, start_ids);
   }
 
   void EXPECT_VISITED(const RTCStats* stats) {
diff --git a/third_party/libwebrtc/pc/rtp_sender.cc b/third_party/libwebrtc/pc/rtp_sender.cc
index cdae1595b3..b0c32eff85 100644
--- a/third_party/libwebrtc/pc/rtp_sender.cc
+++ b/third_party/libwebrtc/pc/rtp_sender.cc
@@ -115,13 +115,13 @@ class SignalingThreadCallback {
     if (!signaling_thread_->IsCurrent()) {
       signaling_thread_->PostTask(
           [callback = std::move(callback_), error]() mutable {
-            webrtc::InvokeSetParametersCallback(callback, error);
+            InvokeSetParametersCallback(callback, error);
           });
       callback_ = nullptr;
       return;
     }
 
-    webrtc::InvokeSetParametersCallback(callback_, error);
+    InvokeSetParametersCallback(callback_, error);
     callback_ = nullptr;
   }
 
@@ -243,7 +243,7 @@ void RtpSenderBase::SetParametersInternal(const RtpParameters& parameters,
         "Attempted to set an unimplemented parameter of RtpParameters.");
     RTC_LOG(LS_ERROR) << error.message() << " ("
                       << ::webrtc::ToString(error.type()) << ")";
-    webrtc::InvokeSetParametersCallback(callback, error);
+    InvokeSetParametersCallback(callback, error);
     return;
   }
   if (!media_channel_ || !ssrc_) {
@@ -252,7 +252,7 @@ void RtpSenderBase::SetParametersInternal(const RtpParameters& parameters,
     if (result.ok()) {
       init_parameters_ = parameters;
     }
-    webrtc::InvokeSetParametersCallback(callback, result);
+    InvokeSetParametersCallback(callback, result);
     return;
   }
   auto task = [&, callback = std::move(callback),
@@ -268,13 +268,13 @@ void RtpSenderBase::SetParametersInternal(const RtpParameters& parameters,
     RTCError result = cricket::CheckRtpParametersInvalidModificationAndValues(
         old_parameters, rtp_parameters);
     if (!result.ok()) {
-      webrtc::InvokeSetParametersCallback(callback, result);
+      InvokeSetParametersCallback(callback, result);
       return;
     }
 
     result = CheckCodecParameters(rtp_parameters);
     if (!result.ok()) {
-      webrtc::InvokeSetParametersCallback(callback, result);
+      InvokeSetParametersCallback(callback, result);
       return;
     }
 
@@ -389,7 +389,7 @@ void RtpSenderBase::SetParametersAsync(const RtpParameters& parameters,
   TRACE_EVENT0("webrtc", "RtpSenderBase::SetParametersAsync");
   RTCError result = CheckSetParameters(parameters);
   if (!result.ok()) {
-    webrtc::InvokeSetParametersCallback(callback, result);
+    InvokeSetParametersCallback(callback, result);
     return;
   }
 
@@ -399,7 +399,7 @@ void RtpSenderBase::SetParametersAsync(const RtpParameters& parameters,
           signaling_thread_,
           [this, callback = std::move(callback)](RTCError error) mutable {
             last_transaction_id_.reset();
-            webrtc::InvokeSetParametersCallback(callback, error);
+            InvokeSetParametersCallback(callback, error);
           }),
       false);
 }
diff --git a/third_party/libwebrtc/pc/rtp_sender.h b/third_party/libwebrtc/pc/rtp_sender.h
index d29c3760e6..8925230636 100644
--- a/third_party/libwebrtc/pc/rtp_sender.h
+++ b/third_party/libwebrtc/pc/rtp_sender.h
@@ -86,9 +86,7 @@ class RtpSenderInternal : public RtpSenderInterface {
       const RtpParameters& parameters) = 0;
 
   // Additional checks that are specific to the current codec settings
-  virtual RTCError CheckCodecParameters(const RtpParameters& parameters) {
-    return webrtc::RTCError::OK();
-  }
+  virtual RTCError CheckCodecParameters(const RtpParameters& parameters) = 0;
 
   // Returns an ID that changes every time SetTrack() is called, but
   // otherwise remains constant. Used to generate IDs for stats.
diff --git a/third_party/libwebrtc/pc/rtp_sender_receiver_unittest.cc b/third_party/libwebrtc/pc/rtp_sender_receiver_unittest.cc
index 3092e53c2d..4387aedf53 100644
--- a/third_party/libwebrtc/pc/rtp_sender_receiver_unittest.cc
+++ b/third_party/libwebrtc/pc/rtp_sender_receiver_unittest.cc
@@ -105,7 +105,7 @@ class RtpSenderReceiverTest
       : network_thread_(rtc::Thread::Current()),
         worker_thread_(rtc::Thread::Current()),
         video_bitrate_allocator_factory_(
-            webrtc::CreateBuiltinVideoBitrateAllocatorFactory()),
+            CreateBuiltinVideoBitrateAllocatorFactory()),
         // Create fake media engine/etc. so we can create channels to use to
         // test RtpSenders/RtpReceivers.
         media_engine_(std::make_unique<cricket::FakeMediaEngine>()),
@@ -119,16 +119,16 @@ class RtpSenderReceiverTest
     // Fake media channels are owned by the media engine.
     voice_media_send_channel_ = media_engine_->voice().CreateSendChannel(
         &fake_call_, cricket::MediaConfig(), cricket::AudioOptions(),
-        webrtc::CryptoOptions(), webrtc::AudioCodecPairId::Create());
+        CryptoOptions(), AudioCodecPairId::Create());
     video_media_send_channel_ = media_engine_->video().CreateSendChannel(
         &fake_call_, cricket::MediaConfig(), cricket::VideoOptions(),
-        webrtc::CryptoOptions(), video_bitrate_allocator_factory_.get());
+        CryptoOptions(), video_bitrate_allocator_factory_.get());
     voice_media_receive_channel_ = media_engine_->voice().CreateReceiveChannel(
         &fake_call_, cricket::MediaConfig(), cricket::AudioOptions(),
-        webrtc::CryptoOptions(), webrtc::AudioCodecPairId::Create());
+        CryptoOptions(), AudioCodecPairId::Create());
     video_media_receive_channel_ = media_engine_->video().CreateReceiveChannel(
         &fake_call_, cricket::MediaConfig(), cricket::VideoOptions(),
-        webrtc::CryptoOptions());
+        CryptoOptions());
 
     // Create streams for predefined SSRCs. Streams need to exist in order
     // for the senders and receievers to apply parameters to them.
@@ -162,8 +162,8 @@ class RtpSenderReceiverTest
     audio_track_ = nullptr;
   }
 
-  std::unique_ptr<webrtc::RtpTransportInternal> CreateDtlsSrtpTransport() {
-    auto dtls_srtp_transport = std::make_unique<webrtc::DtlsSrtpTransport>(
+  std::unique_ptr<RtpTransportInternal> CreateDtlsSrtpTransport() {
+    auto dtls_srtp_transport = std::make_unique<DtlsSrtpTransport>(
         /*rtcp_mux_required=*/true, field_trials_);
     dtls_srtp_transport->SetDtlsTransports(rtp_dtls_transport_.get(),
                                            /*rtcp_dtls_transport=*/nullptr);
@@ -515,12 +515,12 @@ class RtpSenderReceiverTest
   test::RunLoop run_loop_;
   rtc::Thread* const network_thread_;
   rtc::Thread* const worker_thread_;
-  webrtc::RtcEventLogNull event_log_;
+  RtcEventLogNull event_log_;
   // The `rtp_dtls_transport_` and `rtp_transport_` should be destroyed after
   // the `channel_manager`.
   std::unique_ptr<cricket::DtlsTransportInternal> rtp_dtls_transport_;
-  std::unique_ptr<webrtc::RtpTransportInternal> rtp_transport_;
-  std::unique_ptr<webrtc::VideoBitrateAllocatorFactory>
+  std::unique_ptr<RtpTransportInternal> rtp_transport_;
+  std::unique_ptr<VideoBitrateAllocatorFactory>
       video_bitrate_allocator_factory_;
   std::unique_ptr<cricket::FakeMediaEngine> media_engine_;
   rtc::UniqueRandomIdGenerator ssrc_generator_;
@@ -540,7 +540,7 @@ class RtpSenderReceiverTest
   rtc::scoped_refptr<MediaStreamInterface> local_stream_;
   rtc::scoped_refptr<VideoTrackInterface> video_track_;
   rtc::scoped_refptr<AudioTrackInterface> audio_track_;
-  webrtc::test::ScopedKeyValueConfig field_trials_;
+  test::ScopedKeyValueConfig field_trials_;
 };
 
 // Test that `voice_channel_` is updated when an audio track is associated
@@ -651,15 +651,13 @@ TEST_F(RtpSenderReceiverTest, LocalVideoTrackDisable) {
 TEST_F(RtpSenderReceiverTest, RemoteVideoTrackState) {
   CreateVideoRtpReceiver();
 
-  EXPECT_EQ(webrtc::MediaStreamTrackInterface::kLive, video_track_->state());
-  EXPECT_EQ(webrtc::MediaSourceInterface::kLive,
-            video_track_->GetSource()->state());
+  EXPECT_EQ(MediaStreamTrackInterface::kLive, video_track_->state());
+  EXPECT_EQ(MediaSourceInterface::kLive, video_track_->GetSource()->state());
 
   DestroyVideoRtpReceiver();
 
-  EXPECT_EQ(webrtc::MediaStreamTrackInterface::kEnded, video_track_->state());
-  EXPECT_EQ(webrtc::MediaSourceInterface::kEnded,
-            video_track_->GetSource()->state());
+  EXPECT_EQ(MediaStreamTrackInterface::kEnded, video_track_->state());
+  EXPECT_EQ(MediaSourceInterface::kEnded, video_track_->GetSource()->state());
   DestroyVideoRtpReceiver();
 }
 
@@ -888,9 +886,9 @@ TEST_F(RtpSenderReceiverTest, AudioSenderCanSetParametersAsync) {
 
   RtpParameters params = audio_rtp_sender_->GetParameters();
   EXPECT_EQ(1u, params.encodings.size());
-  absl::optional<webrtc::RTCError> result;
+  absl::optional<RTCError> result;
   audio_rtp_sender_->SetParametersAsync(
-      params, [&result](webrtc::RTCError error) { result = error; });
+      params, [&result](RTCError error) { result = error; });
   run_loop_.Flush();
   EXPECT_TRUE(result->ok());
 
@@ -918,13 +916,13 @@ TEST_F(RtpSenderReceiverTest,
   audio_rtp_sender_ =
       AudioRtpSender::Create(worker_thread_, /*id=*/"", nullptr, nullptr);
 
-  absl::optional<webrtc::RTCError> result;
+  absl::optional<RTCError> result;
   RtpParameters params = audio_rtp_sender_->GetParameters();
   ASSERT_EQ(1u, params.encodings.size());
   params.encodings[0].max_bitrate_bps = 90000;
 
   audio_rtp_sender_->SetParametersAsync(
-      params, [&result](webrtc::RTCError error) { result = error; });
+      params, [&result](RTCError error) { result = error; });
   run_loop_.Flush();
   EXPECT_TRUE(result->ok());
 
@@ -932,7 +930,7 @@ TEST_F(RtpSenderReceiverTest,
   EXPECT_EQ(params.encodings[0].max_bitrate_bps, 90000);
 
   audio_rtp_sender_->SetParametersAsync(
-      params, [&result](webrtc::RTCError error) { result = error; });
+      params, [&result](RTCError error) { result = error; });
   run_loop_.Flush();
   EXPECT_TRUE(result->ok());
 
@@ -1016,13 +1014,13 @@ TEST_F(RtpSenderReceiverTest,
 
   RtpParameters params = audio_rtp_sender_->GetParameters();
   EXPECT_EQ(1u, params.encodings.size());
-  absl::optional<webrtc::RTCError> result;
+  absl::optional<RTCError> result;
   audio_rtp_sender_->SetParametersAsync(
-      params, [&result](webrtc::RTCError error) { result = error; });
+      params, [&result](RTCError error) { result = error; });
   run_loop_.Flush();
   EXPECT_TRUE(result->ok());
   audio_rtp_sender_->SetParametersAsync(
-      params, [&result](webrtc::RTCError error) { result = error; });
+      params, [&result](RTCError error) { result = error; });
   run_loop_.Flush();
   EXPECT_EQ(RTCErrorType::INVALID_STATE, result->type());
 
@@ -1081,7 +1079,7 @@ TEST_F(RtpSenderReceiverTest, SetAudioMaxSendBitrate) {
   CreateAudioRtpSender();
 
   EXPECT_EQ(-1, voice_media_send_channel()->max_bps());
-  webrtc::RtpParameters params = audio_rtp_sender_->GetParameters();
+  RtpParameters params = audio_rtp_sender_->GetParameters();
   EXPECT_EQ(1U, params.encodings.size());
   EXPECT_FALSE(params.encodings[0].max_bitrate_bps);
   params.encodings[0].max_bitrate_bps = 1000;
@@ -1106,10 +1104,9 @@ TEST_F(RtpSenderReceiverTest, SetAudioMaxSendBitrate) {
 TEST_F(RtpSenderReceiverTest, SetAudioBitratePriority) {
   CreateAudioRtpSender();
 
-  webrtc::RtpParameters params = audio_rtp_sender_->GetParameters();
+  RtpParameters params = audio_rtp_sender_->GetParameters();
   EXPECT_EQ(1U, params.encodings.size());
-  EXPECT_EQ(webrtc::kDefaultBitratePriority,
-            params.encodings[0].bitrate_priority);
+  EXPECT_EQ(kDefaultBitratePriority, params.encodings[0].bitrate_priority);
   double new_bitrate_priority = 2.0;
   params.encodings[0].bitrate_priority = new_bitrate_priority;
   EXPECT_TRUE(audio_rtp_sender_->SetParameters(params).ok());
@@ -1140,9 +1137,9 @@ TEST_F(RtpSenderReceiverTest, VideoSenderCanSetParametersAsync) {
 
   RtpParameters params = video_rtp_sender_->GetParameters();
   EXPECT_EQ(1u, params.encodings.size());
-  absl::optional<webrtc::RTCError> result;
+  absl::optional<RTCError> result;
   video_rtp_sender_->SetParametersAsync(
-      params, [&result](webrtc::RTCError error) { result = error; });
+      params, [&result](RTCError error) { result = error; });
   run_loop_.Flush();
   EXPECT_TRUE(result->ok());
 
@@ -1170,19 +1167,19 @@ TEST_F(RtpSenderReceiverTest,
   video_rtp_sender_ =
       VideoRtpSender::Create(worker_thread_, /*id=*/"", nullptr);
 
-  absl::optional<webrtc::RTCError> result;
+  absl::optional<RTCError> result;
   RtpParameters params = video_rtp_sender_->GetParameters();
   ASSERT_EQ(1u, params.encodings.size());
   params.encodings[0].max_bitrate_bps = 90000;
   video_rtp_sender_->SetParametersAsync(
-      params, [&result](webrtc::RTCError error) { result = error; });
+      params, [&result](RTCError error) { result = error; });
   run_loop_.Flush();
   EXPECT_TRUE(result->ok());
 
   params = video_rtp_sender_->GetParameters();
   EXPECT_EQ(params.encodings[0].max_bitrate_bps, 90000);
   video_rtp_sender_->SetParametersAsync(
-      params, [&result](webrtc::RTCError error) { result = error; });
+      params, [&result](RTCError error) { result = error; });
   run_loop_.Flush();
   EXPECT_TRUE(result->ok());
 
@@ -1350,13 +1347,13 @@ TEST_F(RtpSenderReceiverTest,
 
   RtpParameters params = video_rtp_sender_->GetParameters();
   EXPECT_EQ(1u, params.encodings.size());
-  absl::optional<webrtc::RTCError> result;
+  absl::optional<RTCError> result;
   video_rtp_sender_->SetParametersAsync(
-      params, [&result](webrtc::RTCError error) { result = error; });
+      params, [&result](RTCError error) { result = error; });
   run_loop_.Flush();
   EXPECT_TRUE(result->ok());
   video_rtp_sender_->SetParametersAsync(
-      params, [&result](webrtc::RTCError error) { result = error; });
+      params, [&result](RTCError error) { result = error; });
   run_loop_.Flush();
   EXPECT_EQ(RTCErrorType::INVALID_STATE, result->type());
 
@@ -1453,7 +1450,7 @@ TEST_F(RtpSenderReceiverTest, VideoSenderDetectInvalidNumTemporalLayers) {
   CreateVideoRtpSender();
 
   RtpParameters params = video_rtp_sender_->GetParameters();
-  params.encodings[0].num_temporal_layers = webrtc::kMaxTemporalStreams + 1;
+  params.encodings[0].num_temporal_layers = kMaxTemporalStreams + 1;
   RTCError result = video_rtp_sender_->SetParameters(params);
   EXPECT_EQ(RTCErrorType::INVALID_RANGE, result.type());
 
@@ -1536,7 +1533,7 @@ TEST_F(RtpSenderReceiverTest, SetVideoMinMaxSendBitrate) {
   CreateVideoRtpSender();
 
   EXPECT_EQ(-1, video_media_send_channel()->max_bps());
-  webrtc::RtpParameters params = video_rtp_sender_->GetParameters();
+  RtpParameters params = video_rtp_sender_->GetParameters();
   EXPECT_EQ(1U, params.encodings.size());
   EXPECT_FALSE(params.encodings[0].min_bitrate_bps);
   EXPECT_FALSE(params.encodings[0].max_bitrate_bps);
@@ -1589,10 +1586,9 @@ TEST_F(RtpSenderReceiverTest, SetVideoMinMaxSendBitrateSimulcast) {
 TEST_F(RtpSenderReceiverTest, SetVideoBitratePriority) {
   CreateVideoRtpSender();
 
-  webrtc::RtpParameters params = video_rtp_sender_->GetParameters();
+  RtpParameters params = video_rtp_sender_->GetParameters();
   EXPECT_EQ(1U, params.encodings.size());
-  EXPECT_EQ(webrtc::kDefaultBitratePriority,
-            params.encodings[0].bitrate_priority);
+  EXPECT_EQ(kDefaultBitratePriority, params.encodings[0].bitrate_priority);
   double new_bitrate_priority = 2.0;
   params.encodings[0].bitrate_priority = new_bitrate_priority;
   EXPECT_TRUE(video_rtp_sender_->SetParameters(params).ok());
diff --git a/third_party/libwebrtc/pc/rtp_transceiver.cc b/third_party/libwebrtc/pc/rtp_transceiver.cc
index 815ec9dece..ca626cc94b 100644
--- a/third_party/libwebrtc/pc/rtp_transceiver.cc
+++ b/third_party/libwebrtc/pc/rtp_transceiver.cc
@@ -542,7 +542,7 @@ bool RtpTransceiver::stopping() const {
 
 RtpTransceiverDirection RtpTransceiver::direction() const {
   if (unified_plan_ && stopping())
-    return webrtc::RtpTransceiverDirection::kStopped;
+    return RtpTransceiverDirection::kStopped;
 
   return direction_;
 }
@@ -570,7 +570,7 @@ RTCError RtpTransceiver::SetDirectionWithError(
 absl::optional<RtpTransceiverDirection> RtpTransceiver::current_direction()
     const {
   if (unified_plan_ && stopped())
-    return webrtc::RtpTransceiverDirection::kStopped;
+    return RtpTransceiverDirection::kStopped;
 
   return current_direction_;
 }
@@ -604,7 +604,7 @@ void RtpTransceiver::StopSendingAndReceiving() {
   });
 
   stopping_ = true;
-  direction_ = webrtc::RtpTransceiverDirection::kInactive;
+  direction_ = RtpTransceiverDirection::kInactive;
 }
 
 RTCError RtpTransceiver::StopStandard() {
diff --git a/third_party/libwebrtc/pc/rtp_transceiver.h b/third_party/libwebrtc/pc/rtp_transceiver.h
index deda5d7d61..88febb9429 100644
--- a/third_party/libwebrtc/pc/rtp_transceiver.h
+++ b/third_party/libwebrtc/pc/rtp_transceiver.h
@@ -358,20 +358,18 @@ PROXY_CONSTMETHOD0(rtc::scoped_refptr<RtpReceiverInterface>, receiver)
 PROXY_CONSTMETHOD0(bool, stopped)
 PROXY_CONSTMETHOD0(bool, stopping)
 PROXY_CONSTMETHOD0(RtpTransceiverDirection, direction)
-PROXY_METHOD1(webrtc::RTCError, SetDirectionWithError, RtpTransceiverDirection)
+PROXY_METHOD1(RTCError, SetDirectionWithError, RtpTransceiverDirection)
 PROXY_CONSTMETHOD0(absl::optional<RtpTransceiverDirection>, current_direction)
 PROXY_CONSTMETHOD0(absl::optional<RtpTransceiverDirection>, fired_direction)
-PROXY_METHOD0(webrtc::RTCError, StopStandard)
+PROXY_METHOD0(RTCError, StopStandard)
 PROXY_METHOD0(void, StopInternal)
-PROXY_METHOD1(webrtc::RTCError,
-              SetCodecPreferences,
-              rtc::ArrayView<RtpCodecCapability>)
+PROXY_METHOD1(RTCError, SetCodecPreferences, rtc::ArrayView<RtpCodecCapability>)
 PROXY_CONSTMETHOD0(std::vector<RtpCodecCapability>, codec_preferences)
 PROXY_CONSTMETHOD0(std::vector<RtpHeaderExtensionCapability>,
                    GetHeaderExtensionsToNegotiate)
 PROXY_CONSTMETHOD0(std::vector<RtpHeaderExtensionCapability>,
                    GetNegotiatedHeaderExtensions)
-PROXY_METHOD1(webrtc::RTCError,
+PROXY_METHOD1(RTCError,
               SetHeaderExtensionsToNegotiate,
               rtc::ArrayView<const RtpHeaderExtensionCapability>)
 END_PROXY_MAP(RtpTransceiver)
diff --git a/third_party/libwebrtc/pc/rtp_transceiver_unittest.cc b/third_party/libwebrtc/pc/rtp_transceiver_unittest.cc
index 8b4a2389ce..d75e964509 100644
--- a/third_party/libwebrtc/pc/rtp_transceiver_unittest.cc
+++ b/third_party/libwebrtc/pc/rtp_transceiver_unittest.cc
@@ -17,10 +17,11 @@
 
 #include "absl/strings/string_view.h"
 #include "absl/types/optional.h"
+#include "api/environment/environment_factory.h"
 #include "api/peer_connection_interface.h"
 #include "api/rtp_parameters.h"
-#include "media/base/fake_media_engine.h"
 #include "media/base/media_engine.h"
+#include "pc/test/enable_fake_media.h"
 #include "pc/test/mock_channel_interface.h"
 #include "pc/test/mock_rtp_receiver_internal.h"
 #include "pc/test/mock_rtp_sender_internal.h"
@@ -44,7 +45,8 @@ class RtpTransceiverTest : public testing::Test {
  public:
   RtpTransceiverTest()
       : dependencies_(MakeDependencies()),
-        context_(ConnectionContext::Create(&dependencies_)) {}
+        context_(
+            ConnectionContext::Create(CreateEnvironment(), &dependencies_)) {}
 
  protected:
   cricket::MediaEngineInterface* media_engine() {
@@ -60,7 +62,7 @@ class RtpTransceiverTest : public testing::Test {
     d.network_thread = rtc::Thread::Current();
     d.worker_thread = rtc::Thread::Current();
     d.signaling_thread = rtc::Thread::Current();
-    d.media_engine = std::make_unique<cricket::FakeMediaEngine>();
+    EnableFakeMedia(d);
     return d;
   }
 
@@ -420,8 +422,8 @@ TEST_F(RtpTransceiverTestForHeaderExtensions, ReturnsNegotiatedHdrExts) {
   EXPECT_CALL(*mock_channel, mid()).WillRepeatedly(ReturnRef(content_name));
   EXPECT_CALL(*mock_channel, SetRtpTransport(_)).WillRepeatedly(Return(true));
 
-  cricket::RtpHeaderExtensions extensions = {webrtc::RtpExtension("uri1", 1),
-                                             webrtc::RtpExtension("uri2", 2)};
+  cricket::RtpHeaderExtensions extensions = {RtpExtension("uri1", 1),
+                                             RtpExtension("uri2", 2)};
   cricket::AudioContentDescription description;
   description.set_rtp_header_extensions(extensions);
   transceiver_->OnNegotiationUpdate(SdpType::kAnswer, &description);
@@ -449,8 +451,8 @@ TEST_F(RtpTransceiverTestForHeaderExtensions,
   EXPECT_CALL(*sender_.get(), SetTransceiverAsStopped());
   EXPECT_CALL(*sender_.get(), Stop());
 
-  cricket::RtpHeaderExtensions extensions = {webrtc::RtpExtension("uri1", 1),
-                                             webrtc::RtpExtension("uri2", 2)};
+  cricket::RtpHeaderExtensions extensions = {RtpExtension("uri1", 1),
+                                             RtpExtension("uri2", 2)};
   cricket::AudioContentDescription description;
   description.set_rtp_header_extensions(extensions);
   transceiver_->OnNegotiationUpdate(SdpType::kAnswer, &description);
@@ -464,8 +466,7 @@ TEST_F(RtpTransceiverTestForHeaderExtensions,
                                 RtpTransceiverDirection::kStopped),
                           Field(&RtpHeaderExtensionCapability::direction,
                                 RtpTransceiverDirection::kStopped)));
-  extensions = {webrtc::RtpExtension("uri3", 4),
-                webrtc::RtpExtension("uri5", 6)};
+  extensions = {RtpExtension("uri3", 4), RtpExtension("uri5", 6)};
   description.set_rtp_header_extensions(extensions);
   transceiver_->OnNegotiationUpdate(SdpType::kAnswer, &description);
 
diff --git a/third_party/libwebrtc/pc/rtp_transport.cc b/third_party/libwebrtc/pc/rtp_transport.cc
index 653b51fd9e..7cf9fe0ace 100644
--- a/third_party/libwebrtc/pc/rtp_transport.cc
+++ b/third_party/libwebrtc/pc/rtp_transport.cc
@@ -180,12 +180,16 @@ bool RtpTransport::UnregisterRtpDemuxerSink(RtpPacketSinkInterface* sink) {
   return true;
 }
 
+flat_set<uint32_t> RtpTransport::GetSsrcsForSink(RtpPacketSinkInterface* sink) {
+  return rtp_demuxer_.GetSsrcsForSink(sink);
+}
+
 void RtpTransport::DemuxPacket(rtc::CopyOnWriteBuffer packet,
                                int64_t packet_time_us) {
-  webrtc::RtpPacketReceived parsed_packet(
-      &header_extension_map_, packet_time_us == -1
-                                  ? Timestamp::MinusInfinity()
-                                  : Timestamp::Micros(packet_time_us));
+  RtpPacketReceived parsed_packet(&header_extension_map_,
+                                  packet_time_us == -1
+                                      ? Timestamp::MinusInfinity()
+                                      : Timestamp::Micros(packet_time_us));
   if (!parsed_packet.Parse(std::move(packet))) {
     RTC_LOG(LS_ERROR)
         << "Failed to parse the incoming RTP packet before demuxing. Drop it.";
@@ -226,7 +230,14 @@ void RtpTransport::OnSentPacket(rtc::PacketTransportInternal* packet_transport,
                                 const rtc::SentPacket& sent_packet) {
   RTC_DCHECK(packet_transport == rtp_packet_transport_ ||
              packet_transport == rtcp_packet_transport_);
+  if (processing_sent_packet_) {
+    TaskQueueBase::Current()->PostTask(SafeTask(
+        safety_.flag(), [this, sent_packet] { SendSentPacket(sent_packet); }));
+    return;
+  }
+  processing_sent_packet_ = true;
   SendSentPacket(sent_packet);
+  processing_sent_packet_ = false;
 }
 
 void RtpTransport::OnRtpPacketReceived(rtc::CopyOnWriteBuffer packet,
diff --git a/third_party/libwebrtc/pc/rtp_transport.h b/third_party/libwebrtc/pc/rtp_transport.h
index 456c91c370..6d5d4bff57 100644
--- a/third_party/libwebrtc/pc/rtp_transport.h
+++ b/third_party/libwebrtc/pc/rtp_transport.h
@@ -96,6 +96,7 @@ class RtpTransport : public RtpTransportInternal {
                   rtc::CopyOnWriteBuffer* packet,
                   const rtc::PacketOptions& options,
                   int flags);
+  flat_set<uint32_t> GetSsrcsForSink(RtpPacketSinkInterface* sink);
 
   // Overridden by SrtpTransport.
   virtual void OnNetworkRouteChanged(
@@ -140,6 +141,7 @@ class RtpTransport : public RtpTransportInternal {
   RtpHeaderExtensionMap header_extension_map_;
   // Guard against recursive "ready to send" signals
   bool processing_ready_to_send_ = false;
+  bool processing_sent_packet_ = false;
   ScopedTaskSafety safety_;
 };
 
diff --git a/third_party/libwebrtc/pc/rtp_transport_internal.h b/third_party/libwebrtc/pc/rtp_transport_internal.h
index 4114fa9340..483a1cee38 100644
--- a/third_party/libwebrtc/pc/rtp_transport_internal.h
+++ b/third_party/libwebrtc/pc/rtp_transport_internal.h
@@ -72,7 +72,7 @@ class RtpTransportInternal : public sigslot::has_slots<> {
   // Called whenever a RTP packet that can not be demuxed by the transport is
   // received.
   void SetUnDemuxableRtpPacketReceivedHandler(
-      absl::AnyInvocable<void(webrtc::RtpPacketReceived&)> callback) {
+      absl::AnyInvocable<void(RtpPacketReceived&)> callback) {
     callback_undemuxable_rtp_packet_received_ = std::move(callback);
   }
 
@@ -160,7 +160,7 @@ class RtpTransportInternal : public sigslot::has_slots<> {
   CallbackList<bool> callback_list_ready_to_send_;
   CallbackList<rtc::CopyOnWriteBuffer*, int64_t>
       callback_list_rtcp_packet_received_;
-  absl::AnyInvocable<void(webrtc::RtpPacketReceived&)>
+  absl::AnyInvocable<void(RtpPacketReceived&)>
       callback_undemuxable_rtp_packet_received_ =
           [](RtpPacketReceived& packet) {};
   CallbackList<absl::optional<rtc::NetworkRoute>>
diff --git a/third_party/libwebrtc/pc/rtp_transport_unittest.cc b/third_party/libwebrtc/pc/rtp_transport_unittest.cc
index 5b6a8309e0..6b8e616799 100644
--- a/third_party/libwebrtc/pc/rtp_transport_unittest.cc
+++ b/third_party/libwebrtc/pc/rtp_transport_unittest.cc
@@ -349,4 +349,28 @@ TEST(RtpTransportTest, RecursiveSetSendDoesNotCrash) {
   EXPECT_FALSE(observer.ready_to_send());
 }
 
+TEST(RtpTransportTest, RecursiveOnSentPacketDoesNotCrash) {
+  const int kShortTimeout = 100;
+  test::RunLoop loop;
+  RtpTransport transport(kMuxEnabled);
+  rtc::FakePacketTransport fake_rtp("fake_rtp");
+  transport.SetRtpPacketTransport(&fake_rtp);
+  fake_rtp.SetDestination(&fake_rtp, true);
+  TransportObserver observer(&transport);
+  const rtc::PacketOptions options;
+  const int flags = 0;
+
+  fake_rtp.SetWritable(true);
+  observer.SetActionOnSentPacket([&]() {
+    rtc::CopyOnWriteBuffer rtp_data(kRtpData, kRtpLen);
+    if (observer.sent_packet_count() < 2) {
+      transport.SendRtpPacket(&rtp_data, options, flags);
+    }
+  });
+  rtc::CopyOnWriteBuffer rtp_data(kRtpData, kRtpLen);
+  transport.SendRtpPacket(&rtp_data, options, flags);
+  EXPECT_EQ(observer.sent_packet_count(), 1);
+  EXPECT_EQ_WAIT(observer.sent_packet_count(), 2, kShortTimeout);
+}
+
 }  // namespace webrtc
diff --git a/third_party/libwebrtc/pc/sctp_transport.h b/third_party/libwebrtc/pc/sctp_transport.h
index 35e7656100..076dee5318 100644
--- a/third_party/libwebrtc/pc/sctp_transport.h
+++ b/third_party/libwebrtc/pc/sctp_transport.h
@@ -61,7 +61,7 @@ class SctpTransport : public SctpTransportInterface,
   void Start(int local_port, int remote_port, int max_message_size);
 
   // TODO(https://bugs.webrtc.org/10629): Move functions that need
-  // internal() to be functions on the webrtc::SctpTransport interface,
+  // internal() to be functions on the SctpTransport interface,
   // and make the internal() function private.
   cricket::SctpTransportInternal* internal() {
     RTC_DCHECK_RUN_ON(owner_thread_);
diff --git a/third_party/libwebrtc/pc/sctp_utils.cc b/third_party/libwebrtc/pc/sctp_utils.cc
index 54742c27a7..60ffdc7919 100644
--- a/third_party/libwebrtc/pc/sctp_utils.cc
+++ b/third_party/libwebrtc/pc/sctp_utils.cc
@@ -64,7 +64,7 @@ bool ParseDataChannelOpenMessage(const rtc::CopyOnWriteBuffer& payload,
   // Format defined at
   // http://tools.ietf.org/html/draft-jesup-rtcweb-data-protocol-04
 
-  rtc::ByteBufferReader buffer(payload.data<char>(), payload.size());
+  rtc::ByteBufferReader buffer(payload);
   uint8_t message_type;
   if (!buffer.ReadUInt8(&message_type)) {
     RTC_LOG(LS_WARNING) << "Could not read OPEN message type.";
diff --git a/third_party/libwebrtc/pc/sctp_utils_unittest.cc b/third_party/libwebrtc/pc/sctp_utils_unittest.cc
index 3e49824b45..9ef2068a05 100644
--- a/third_party/libwebrtc/pc/sctp_utils_unittest.cc
+++ b/third_party/libwebrtc/pc/sctp_utils_unittest.cc
@@ -35,7 +35,7 @@ class SctpUtilsTest : public ::testing::Test {
     uint16_t label_length;
     uint16_t protocol_length;
 
-    rtc::ByteBufferReader buffer(packet.data<char>(), packet.size());
+    rtc::ByteBufferReader buffer(packet);
     ASSERT_TRUE(buffer.ReadUInt8(&message_type));
     EXPECT_EQ(0x03, message_type);
 
@@ -176,7 +176,7 @@ TEST_F(SctpUtilsTest, WriteParseAckMessage) {
   webrtc::WriteDataChannelOpenAckMessage(&packet);
 
   uint8_t message_type;
-  rtc::ByteBufferReader buffer(packet.data<char>(), packet.size());
+  rtc::ByteBufferReader buffer(packet);
   ASSERT_TRUE(buffer.ReadUInt8(&message_type));
   EXPECT_EQ(0x02, message_type);
 
diff --git a/third_party/libwebrtc/pc/sdp_offer_answer.cc b/third_party/libwebrtc/pc/sdp_offer_answer.cc
index 0261195bb0..1e43833a0b 100644
--- a/third_party/libwebrtc/pc/sdp_offer_answer.cc
+++ b/third_party/libwebrtc/pc/sdp_offer_answer.cc
@@ -86,8 +86,7 @@ namespace webrtc {
 
 namespace {
 
-typedef webrtc::PeerConnectionInterface::RTCOfferAnswerOptions
-    RTCOfferAnswerOptions;
+typedef PeerConnectionInterface::RTCOfferAnswerOptions RTCOfferAnswerOptions;
 
 // Error messages
 const char kInvalidSdp[] = "Invalid session description.";
@@ -566,22 +565,9 @@ RTCError ValidatePayloadTypes(const cricket::SessionDescription& description) {
       continue;
     }
     const auto type = media_description->type();
-    if (type == cricket::MEDIA_TYPE_AUDIO) {
-      RTC_DCHECK(media_description->as_audio());
-      for (const auto& codec : media_description->as_audio()->codecs()) {
-        if (!cricket::UsedPayloadTypes::IsIdValid(
-                codec, media_description->rtcp_mux())) {
-          LOG_AND_RETURN_ERROR(
-              RTCErrorType::INVALID_PARAMETER,
-              "The media section with MID='" + content.mid() +
-                  "' used an invalid payload type " + rtc::ToString(codec.id) +
-                  " for codec '" + codec.name + ", rtcp-mux:" +
-                  (media_description->rtcp_mux() ? "enabled" : "disabled"));
-        }
-      }
-    } else if (type == cricket::MEDIA_TYPE_VIDEO) {
-      RTC_DCHECK(media_description->as_video());
-      for (const auto& codec : media_description->as_video()->codecs()) {
+    if (type == cricket::MEDIA_TYPE_AUDIO ||
+        type == cricket::MEDIA_TYPE_VIDEO) {
+      for (const auto& codec : media_description->codecs()) {
         if (!cricket::UsedPayloadTypes::IsIdValid(
                 codec, media_description->rtcp_mux())) {
           LOG_AND_RETURN_ERROR(
@@ -834,8 +820,8 @@ std::string GenerateRtcpCname() {
 }
 
 // Check if we can send `new_stream` on a PeerConnection.
-bool CanAddLocalMediaStream(webrtc::StreamCollectionInterface* current_streams,
-                            webrtc::MediaStreamInterface* new_stream) {
+bool CanAddLocalMediaStream(StreamCollectionInterface* current_streams,
+                            MediaStreamInterface* new_stream) {
   if (!new_stream || !current_streams) {
     return false;
   }
@@ -847,7 +833,7 @@ bool CanAddLocalMediaStream(webrtc::StreamCollectionInterface* current_streams,
   return true;
 }
 
-rtc::scoped_refptr<webrtc::DtlsTransport> LookupDtlsTransportByMid(
+rtc::scoped_refptr<DtlsTransport> LookupDtlsTransportByMid(
     rtc::Thread* network_thread,
     JsepTransportController* controller,
     const std::string& mid) {
@@ -1888,8 +1874,8 @@ RTCError SdpOfferAnswerHandler::ApplyLocalDescription(
       if (audio_content->rejected) {
         RemoveSenders(cricket::MEDIA_TYPE_AUDIO);
       } else {
-        const cricket::AudioContentDescription* audio_desc =
-            audio_content->media_description()->as_audio();
+        const cricket::MediaContentDescription* audio_desc =
+            audio_content->media_description();
         UpdateLocalSenders(audio_desc->streams(), audio_desc->type());
       }
     }
@@ -1900,8 +1886,8 @@ RTCError SdpOfferAnswerHandler::ApplyLocalDescription(
       if (video_content->rejected) {
         RemoveSenders(cricket::MEDIA_TYPE_VIDEO);
       } else {
-        const cricket::VideoContentDescription* video_desc =
-            video_content->media_description()->as_video();
+        const cricket::MediaContentDescription* video_desc =
+            video_content->media_description();
         UpdateLocalSenders(video_desc->streams(), video_desc->type());
       }
     }
@@ -2010,8 +1996,11 @@ RTCError SdpOfferAnswerHandler::ReplaceRemoteDescription(
   const cricket::SessionDescription* session_desc =
       remote_description()->description();
 
+  const auto* local = local_description();
+
   // NOTE: This will perform a BlockingCall() to the network thread.
-  return transport_controller_s()->SetRemoteDescription(sdp_type, session_desc);
+  return transport_controller_s()->SetRemoteDescription(
+      sdp_type, local ? local->description() : nullptr, session_desc);
 }
 
 void SdpOfferAnswerHandler::ApplyRemoteDescription(
@@ -4932,13 +4921,15 @@ RTCError SdpOfferAnswerHandler::PushdownTransportDescription(
   if (source == cricket::CS_LOCAL) {
     const SessionDescriptionInterface* sdesc = local_description();
     RTC_DCHECK(sdesc);
-    return transport_controller_s()->SetLocalDescription(type,
-                                                         sdesc->description());
+    const auto* remote = remote_description();
+    return transport_controller_s()->SetLocalDescription(
+        type, sdesc->description(), remote ? remote->description() : nullptr);
   } else {
     const SessionDescriptionInterface* sdesc = remote_description();
     RTC_DCHECK(sdesc);
-    return transport_controller_s()->SetRemoteDescription(type,
-                                                          sdesc->description());
+    const auto* local = local_description();
+    return transport_controller_s()->SetRemoteDescription(
+        type, local ? local->description() : nullptr, sdesc->description());
   }
 }
 
@@ -5366,44 +5357,40 @@ bool SdpOfferAnswerHandler::UpdatePayloadTypeDemuxingState(
       // Ignore transceivers that are not receiving.
       continue;
     }
-    switch (content_info.media_description()->type()) {
-      case cricket::MediaType::MEDIA_TYPE_AUDIO: {
-        if (!mid_header_extension_missing_audio) {
-          mid_header_extension_missing_audio =
-              !ContentHasHeaderExtension(content_info, RtpExtension::kMidUri);
-        }
-        const cricket::AudioContentDescription* audio_desc =
-            content_info.media_description()->as_audio();
-        for (const cricket::AudioCodec& audio : audio_desc->codecs()) {
-          if (payload_types->audio_payload_types.count(audio.id)) {
+    const cricket::MediaType media_type =
+        content_info.media_description()->type();
+    if (media_type == cricket::MediaType::MEDIA_TYPE_AUDIO ||
+        media_type == cricket::MediaType::MEDIA_TYPE_VIDEO) {
+      if (media_type == cricket::MediaType::MEDIA_TYPE_AUDIO &&
+          !mid_header_extension_missing_audio) {
+        mid_header_extension_missing_audio =
+            !ContentHasHeaderExtension(content_info, RtpExtension::kMidUri);
+      } else if (media_type == cricket::MEDIA_TYPE_VIDEO &&
+                 !mid_header_extension_missing_video) {
+        mid_header_extension_missing_video =
+            !ContentHasHeaderExtension(content_info, RtpExtension::kMidUri);
+      }
+      const cricket::MediaContentDescription* media_desc =
+          content_info.media_description();
+      for (const cricket::Codec& codec : media_desc->codecs()) {
+        if (media_type == cricket::MediaType::MEDIA_TYPE_AUDIO) {
+          if (payload_types->audio_payload_types.count(codec.id)) {
             // Two m= sections are using the same payload type, thus demuxing
             // by payload type is not possible.
-            payload_types->pt_demuxing_possible_audio = false;
+            if (media_type == cricket::MediaType::MEDIA_TYPE_AUDIO) {
+              payload_types->pt_demuxing_possible_audio = false;
+            }
           }
-          payload_types->audio_payload_types.insert(audio.id);
-        }
-        break;
-      }
-      case cricket::MediaType::MEDIA_TYPE_VIDEO: {
-        if (!mid_header_extension_missing_video) {
-          mid_header_extension_missing_video =
-              !ContentHasHeaderExtension(content_info, RtpExtension::kMidUri);
-        }
-        const cricket::VideoContentDescription* video_desc =
-            content_info.media_description()->as_video();
-        for (const cricket::VideoCodec& video : video_desc->codecs()) {
-          if (payload_types->video_payload_types.count(video.id)) {
+          payload_types->audio_payload_types.insert(codec.id);
+        } else if (media_type == cricket::MEDIA_TYPE_VIDEO) {
+          if (payload_types->video_payload_types.count(codec.id)) {
             // Two m= sections are using the same payload type, thus demuxing
             // by payload type is not possible.
             payload_types->pt_demuxing_possible_video = false;
           }
-          payload_types->video_payload_types.insert(video.id);
+          payload_types->video_payload_types.insert(codec.id);
         }
-        break;
       }
-      default:
-        // Ignore data channels.
-        continue;
     }
   }
 
diff --git a/third_party/libwebrtc/pc/sdp_offer_answer.h b/third_party/libwebrtc/pc/sdp_offer_answer.h
index 8aa7040b16..88ddfe0332 100644
--- a/third_party/libwebrtc/pc/sdp_offer_answer.h
+++ b/third_party/libwebrtc/pc/sdp_offer_answer.h
@@ -674,8 +674,8 @@ class SdpOfferAnswerHandler : public SdpStateProvider {
   // or else the CreateBuiltinVideoBitrateAllocatorFactory() will be called.
   // Note that one can still choose to override this in a MediaEngine
   // if one wants too.
-  std::unique_ptr<webrtc::VideoBitrateAllocatorFactory>
-      video_bitrate_allocator_factory_ RTC_GUARDED_BY(signaling_thread());
+  std::unique_ptr<VideoBitrateAllocatorFactory> video_bitrate_allocator_factory_
+      RTC_GUARDED_BY(signaling_thread());
 
   // Whether we are the initial offerer on the association. This
   // determines the SSL role.
diff --git a/third_party/libwebrtc/pc/sdp_offer_answer_unittest.cc b/third_party/libwebrtc/pc/sdp_offer_answer_unittest.cc
index 94ceff10ac..f158febac7 100644
--- a/third_party/libwebrtc/pc/sdp_offer_answer_unittest.cc
+++ b/third_party/libwebrtc/pc/sdp_offer_answer_unittest.cc
@@ -88,7 +88,7 @@ class SdpOfferAnswerTest : public ::testing::Test {
                                             Dav1dDecoderTemplateAdapter>>(),
             nullptr /* audio_mixer */,
             nullptr /* audio_processing */)) {
-    webrtc::metrics::Reset();
+    metrics::Reset();
   }
 
   std::unique_ptr<PeerConnectionWrapper> CreatePeerConnection() {
@@ -168,8 +168,8 @@ TEST_F(SdpOfferAnswerTest, BundleRejectsCodecCollisionsAudioVideo) {
   // There is no error yet but the metrics counter will increase.
   EXPECT_TRUE(error.ok());
   EXPECT_METRIC_EQ(
-      1, webrtc::metrics::NumEvents(
-             "WebRTC.PeerConnection.ValidBundledPayloadTypes", false));
+      1, metrics::NumEvents("WebRTC.PeerConnection.ValidBundledPayloadTypes",
+                            false));
 
   // Tolerate codec collisions in rejected m-lines.
   pc = CreatePeerConnection();
@@ -178,9 +178,9 @@ TEST_F(SdpOfferAnswerTest, BundleRejectsCodecCollisionsAudioVideo) {
       absl::StrReplaceAll(sdp, {{"m=video 9 ", "m=video 0 "}}));
   pc->SetRemoteDescription(std::move(rejected_offer), &error);
   EXPECT_TRUE(error.ok());
-  EXPECT_METRIC_EQ(1,
-                   webrtc::metrics::NumEvents(
-                       "WebRTC.PeerConnection.ValidBundledPayloadTypes", true));
+  EXPECT_METRIC_EQ(
+      1, metrics::NumEvents("WebRTC.PeerConnection.ValidBundledPayloadTypes",
+                            true));
 }
 
 TEST_F(SdpOfferAnswerTest, BundleRejectsCodecCollisionsVideoFmtp) {
@@ -221,8 +221,8 @@ TEST_F(SdpOfferAnswerTest, BundleRejectsCodecCollisionsVideoFmtp) {
   pc->SetRemoteDescription(std::move(desc), &error);
   EXPECT_TRUE(error.ok());
   EXPECT_METRIC_EQ(
-      1, webrtc::metrics::NumEvents(
-             "WebRTC.PeerConnection.ValidBundledPayloadTypes", false));
+      1, metrics::NumEvents("WebRTC.PeerConnection.ValidBundledPayloadTypes",
+                            false));
 }
 
 TEST_F(SdpOfferAnswerTest, BundleCodecCollisionInDifferentBundlesAllowed) {
@@ -264,8 +264,8 @@ TEST_F(SdpOfferAnswerTest, BundleCodecCollisionInDifferentBundlesAllowed) {
   pc->SetRemoteDescription(std::move(desc), &error);
   EXPECT_TRUE(error.ok());
   EXPECT_METRIC_EQ(
-      0, webrtc::metrics::NumEvents(
-             "WebRTC.PeerConnection.ValidBundledPayloadTypes", false));
+      0, metrics::NumEvents("WebRTC.PeerConnection.ValidBundledPayloadTypes",
+                            false));
 }
 
 TEST_F(SdpOfferAnswerTest, BundleMeasuresHeaderExtensionIdCollision) {
@@ -990,8 +990,7 @@ TEST_F(SdpOfferAnswerTest, SdpMungingWithInvalidPayloadTypeIsRejected) {
 
   auto offer = pc->CreateOffer();
   ASSERT_EQ(offer->description()->contents().size(), 1u);
-  auto* audio =
-      offer->description()->contents()[0].media_description()->as_audio();
+  auto* audio = offer->description()->contents()[0].media_description();
   ASSERT_GT(audio->codecs().size(), 0u);
   EXPECT_TRUE(audio->rtcp_mux());
   auto codecs = audio->codecs();
diff --git a/third_party/libwebrtc/pc/slow_peer_connection_integration_test.cc b/third_party/libwebrtc/pc/slow_peer_connection_integration_test.cc
index fd9d3417df..9e49291d94 100644
--- a/third_party/libwebrtc/pc/slow_peer_connection_integration_test.cc
+++ b/third_party/libwebrtc/pc/slow_peer_connection_integration_test.cc
@@ -67,7 +67,7 @@ class FakeClockForTest : public rtc::ScopedFakeClock {
     // Some things use a time of "0" as a special value, so we need to start out
     // the fake clock at a nonzero time.
     // TODO(deadbeef): Fix this.
-    AdvanceTime(webrtc::TimeDelta::Seconds(1000));
+    AdvanceTime(TimeDelta::Seconds(1000));
   }
 
   // Explicit handle.
@@ -170,20 +170,20 @@ TEST_P(PeerConnectionIntegrationTest,
   CreateTurnServer(turn_server_internal_address, turn_server_external_address,
                    cricket::PROTO_TLS, "88.88.88.0");
 
-  webrtc::PeerConnectionInterface::IceServer ice_server;
+  PeerConnectionInterface::IceServer ice_server;
   ice_server.urls.push_back("turns:88.88.88.0:3478?transport=tcp");
   ice_server.username = "test";
   ice_server.password = "test";
 
   PeerConnectionInterface::RTCConfiguration client_1_config;
   client_1_config.servers.push_back(ice_server);
-  client_1_config.type = webrtc::PeerConnectionInterface::kRelay;
+  client_1_config.type = PeerConnectionInterface::kRelay;
 
   PeerConnectionInterface::RTCConfiguration client_2_config;
   client_2_config.servers.push_back(ice_server);
   // Setting the type to kRelay forces the connection to go through a TURN
   // server.
-  client_2_config.type = webrtc::PeerConnectionInterface::kRelay;
+  client_2_config.type = PeerConnectionInterface::kRelay;
 
   // Get a copy to the pointer so we can verify calls later.
   rtc::TestCertificateVerifier* client_1_cert_verifier =
@@ -194,10 +194,10 @@ TEST_P(PeerConnectionIntegrationTest,
   client_2_cert_verifier->verify_certificate_ = false;
 
   // Create the dependencies with the test certificate verifier.
-  webrtc::PeerConnectionDependencies client_1_deps(nullptr);
+  PeerConnectionDependencies client_1_deps(nullptr);
   client_1_deps.tls_cert_verifier =
       std::unique_ptr<rtc::TestCertificateVerifier>(client_1_cert_verifier);
-  webrtc::PeerConnectionDependencies client_2_deps(nullptr);
+  PeerConnectionDependencies client_2_deps(nullptr);
   client_2_deps.tls_cert_verifier =
       std::unique_ptr<rtc::TestCertificateVerifier>(client_2_cert_verifier);
 
@@ -262,8 +262,8 @@ class PeerConnectionIntegrationIceStatesTest
   }
 
   void StartStunServer(const SocketAddress& server_address) {
-    stun_server_.reset(
-        cricket::TestStunServer::Create(firewall(), server_address));
+    stun_server_ = cricket::TestStunServer::Create(firewall(), server_address,
+                                                   *network_thread());
   }
 
   bool TestIPv6() {
@@ -309,7 +309,7 @@ class PeerConnectionIntegrationIceStatesTest
 
  private:
   uint32_t port_allocator_flags_;
-  std::unique_ptr<cricket::TestStunServer> stun_server_;
+  cricket::TestStunServer::StunServerPtr stun_server_;
 };
 
 // Ensure FakeClockForTest is constructed first (see class for rationale).
diff --git a/third_party/libwebrtc/pc/srtp_session.cc b/third_party/libwebrtc/pc/srtp_session.cc
index 5408d3e0da..f16679cef1 100644
--- a/third_party/libwebrtc/pc/srtp_session.cc
+++ b/third_party/libwebrtc/pc/srtp_session.cc
@@ -332,6 +332,12 @@ bool SrtpSession::IsExternalAuthActive() const {
   return external_auth_active_;
 }
 
+bool SrtpSession::RemoveSsrcFromSession(uint32_t ssrc) {
+  RTC_DCHECK(session_);
+  // libSRTP expects the SSRC to be in network byte order.
+  return srtp_remove_stream(session_, htonl(ssrc)) == srtp_err_status_ok;
+}
+
 bool SrtpSession::GetSendStreamPacketIndex(void* p,
                                            int in_len,
                                            int64_t* index) {
diff --git a/third_party/libwebrtc/pc/srtp_session.h b/third_party/libwebrtc/pc/srtp_session.h
index 60f1860ada..f8fd3e3123 100644
--- a/third_party/libwebrtc/pc/srtp_session.h
+++ b/third_party/libwebrtc/pc/srtp_session.h
@@ -97,6 +97,14 @@ class SrtpSession {
   // been set.
   bool IsExternalAuthActive() const;
 
+  // Removes a SSRC from the underlying libSRTP session.
+  // Note: this should only be done for SSRCs that are received.
+  // Removing SSRCs that were sent and then reusing them leads to
+  // cryptographic weaknesses described in
+  // https://www.rfc-editor.org/rfc/rfc3711#section-8
+  // https://www.rfc-editor.org/rfc/rfc7714#section-8.4
+  bool RemoveSsrcFromSession(uint32_t ssrc);
+
  private:
   bool DoSetKey(int type,
                 int crypto_suite,
diff --git a/third_party/libwebrtc/pc/srtp_session_unittest.cc b/third_party/libwebrtc/pc/srtp_session_unittest.cc
index 16a840a307..7adfee86fd 100644
--- a/third_party/libwebrtc/pc/srtp_session_unittest.cc
+++ b/third_party/libwebrtc/pc/srtp_session_unittest.cc
@@ -251,4 +251,36 @@ TEST_F(SrtpSessionTest, TestReplay) {
       s1_.ProtectRtp(rtp_packet_, rtp_len_, sizeof(rtp_packet_), &out_len));
 }
 
+TEST_F(SrtpSessionTest, RemoveSsrc) {
+  EXPECT_TRUE(s1_.SetSend(kSrtpAes128CmSha1_80, kTestKey1, kTestKeyLen,
+                          kEncryptedHeaderExtensionIds));
+  EXPECT_TRUE(s2_.SetRecv(kSrtpAes128CmSha1_80, kTestKey1, kTestKeyLen,
+                          kEncryptedHeaderExtensionIds));
+  int out_len;
+  // Encrypt and decrypt the packet once.
+  EXPECT_TRUE(
+      s1_.ProtectRtp(rtp_packet_, rtp_len_, sizeof(rtp_packet_), &out_len));
+  EXPECT_TRUE(s2_.UnprotectRtp(rtp_packet_, out_len, &out_len));
+  EXPECT_EQ(rtp_len_, out_len);
+  EXPECT_EQ(0, memcmp(rtp_packet_, kPcmuFrame, out_len));
+
+  // Recreate the original packet and encrypt again.
+  memcpy(rtp_packet_, kPcmuFrame, rtp_len_);
+  EXPECT_TRUE(
+      s1_.ProtectRtp(rtp_packet_, rtp_len_, sizeof(rtp_packet_), &out_len));
+  // Attempting to decrypt will fail as a replay attack.
+  // (srtp_err_status_replay_fail) since the sequence number was already seen.
+  EXPECT_FALSE(s2_.UnprotectRtp(rtp_packet_, out_len, &out_len));
+
+  // Remove the fake packet SSRC 1 from the session.
+  EXPECT_TRUE(s2_.RemoveSsrcFromSession(1));
+  EXPECT_FALSE(s2_.RemoveSsrcFromSession(1));
+
+  // Since the SRTP state was discarded, this is no longer a replay attack.
+  EXPECT_TRUE(s2_.UnprotectRtp(rtp_packet_, out_len, &out_len));
+  EXPECT_EQ(rtp_len_, out_len);
+  EXPECT_EQ(0, memcmp(rtp_packet_, kPcmuFrame, out_len));
+  EXPECT_TRUE(s2_.RemoveSsrcFromSession(1));
+}
+
 }  // namespace rtc
diff --git a/third_party/libwebrtc/pc/srtp_transport.cc b/third_party/libwebrtc/pc/srtp_transport.cc
index cc20216672..c82839ca82 100644
--- a/third_party/libwebrtc/pc/srtp_transport.cc
+++ b/third_party/libwebrtc/pc/srtp_transport.cc
@@ -37,86 +37,6 @@ SrtpTransport::SrtpTransport(bool rtcp_mux_enabled,
                              const FieldTrialsView& field_trials)
     : RtpTransport(rtcp_mux_enabled), field_trials_(field_trials) {}
 
-RTCError SrtpTransport::SetSrtpSendKey(const cricket::CryptoParams& params) {
-  if (send_params_) {
-    LOG_AND_RETURN_ERROR(
-        webrtc::RTCErrorType::UNSUPPORTED_OPERATION,
-        "Setting the SRTP send key twice is currently unsupported.");
-  }
-  if (recv_params_ && recv_params_->crypto_suite != params.crypto_suite) {
-    LOG_AND_RETURN_ERROR(
-        webrtc::RTCErrorType::UNSUPPORTED_OPERATION,
-        "The send key and receive key must have the same cipher suite.");
-  }
-
-  send_crypto_suite_ = rtc::SrtpCryptoSuiteFromName(params.crypto_suite);
-  if (*send_crypto_suite_ == rtc::kSrtpInvalidCryptoSuite) {
-    return RTCError(RTCErrorType::INVALID_PARAMETER,
-                    "Invalid SRTP crypto suite");
-  }
-
-  int send_key_len, send_salt_len;
-  if (!rtc::GetSrtpKeyAndSaltLengths(*send_crypto_suite_, &send_key_len,
-                                     &send_salt_len)) {
-    return RTCError(RTCErrorType::INVALID_PARAMETER,
-                    "Could not get lengths for crypto suite(s):"
-                    " send crypto_suite ");
-  }
-
-  send_key_ = rtc::ZeroOnFreeBuffer<uint8_t>(send_key_len + send_salt_len);
-  if (!ParseKeyParams(params.key_params, send_key_.data(), send_key_.size())) {
-    return RTCError(RTCErrorType::INVALID_PARAMETER,
-                    "Failed to parse the crypto key params");
-  }
-
-  if (!MaybeSetKeyParams()) {
-    return RTCError(RTCErrorType::INVALID_PARAMETER,
-                    "Failed to set the crypto key params");
-  }
-  send_params_ = params;
-  return RTCError::OK();
-}
-
-RTCError SrtpTransport::SetSrtpReceiveKey(const cricket::CryptoParams& params) {
-  if (recv_params_) {
-    LOG_AND_RETURN_ERROR(
-        webrtc::RTCErrorType::UNSUPPORTED_OPERATION,
-        "Setting the SRTP send key twice is currently unsupported.");
-  }
-  if (send_params_ && send_params_->crypto_suite != params.crypto_suite) {
-    LOG_AND_RETURN_ERROR(
-        webrtc::RTCErrorType::UNSUPPORTED_OPERATION,
-        "The send key and receive key must have the same cipher suite.");
-  }
-
-  recv_crypto_suite_ = rtc::SrtpCryptoSuiteFromName(params.crypto_suite);
-  if (*recv_crypto_suite_ == rtc::kSrtpInvalidCryptoSuite) {
-    return RTCError(RTCErrorType::INVALID_PARAMETER,
-                    "Invalid SRTP crypto suite");
-  }
-
-  int recv_key_len, recv_salt_len;
-  if (!rtc::GetSrtpKeyAndSaltLengths(*recv_crypto_suite_, &recv_key_len,
-                                     &recv_salt_len)) {
-    return RTCError(RTCErrorType::INVALID_PARAMETER,
-                    "Could not get lengths for crypto suite(s):"
-                    " recv crypto_suite ");
-  }
-
-  recv_key_ = rtc::ZeroOnFreeBuffer<uint8_t>(recv_key_len + recv_salt_len);
-  if (!ParseKeyParams(params.key_params, recv_key_.data(), recv_key_.size())) {
-    return RTCError(RTCErrorType::INVALID_PARAMETER,
-                    "Failed to parse the crypto key params");
-  }
-
-  if (!MaybeSetKeyParams()) {
-    return RTCError(RTCErrorType::INVALID_PARAMETER,
-                    "Failed to set the crypto key params");
-  }
-  recv_params_ = params;
-  return RTCError::OK();
-}
-
 bool SrtpTransport::SendRtpPacket(rtc::CopyOnWriteBuffer* packet,
                                   const rtc::PacketOptions& options,
                                   int flags) {
@@ -519,4 +439,19 @@ void SrtpTransport::MaybeUpdateWritableState() {
   }
 }
 
+bool SrtpTransport::UnregisterRtpDemuxerSink(RtpPacketSinkInterface* sink) {
+  if (recv_session_ &&
+      field_trials_.IsEnabled("WebRTC-SrtpRemoveReceiveStream")) {
+    // Remove the SSRCs explicitly registered with the demuxer
+    // (via SDP negotiation) from the SRTP session.
+    for (const auto ssrc : GetSsrcsForSink(sink)) {
+      if (!recv_session_->RemoveSsrcFromSession(ssrc)) {
+        RTC_LOG(LS_WARNING)
+            << "Could not remove SSRC " << ssrc << " from SRTP session.";
+      }
+    }
+  }
+  return RtpTransport::UnregisterRtpDemuxerSink(sink);
+}
+
 }  // namespace webrtc
diff --git a/third_party/libwebrtc/pc/srtp_transport.h b/third_party/libwebrtc/pc/srtp_transport.h
index 46c11ed56d..29721f3b68 100644
--- a/third_party/libwebrtc/pc/srtp_transport.h
+++ b/third_party/libwebrtc/pc/srtp_transport.h
@@ -41,9 +41,6 @@ class SrtpTransport : public RtpTransport {
 
   virtual ~SrtpTransport() = default;
 
-  virtual RTCError SetSrtpSendKey(const cricket::CryptoParams& params);
-  virtual RTCError SetSrtpReceiveKey(const cricket::CryptoParams& params);
-
   bool SendRtpPacket(rtc::CopyOnWriteBuffer* packet,
                      const rtc::PacketOptions& options,
                      int flags) override;
@@ -109,6 +106,10 @@ class SrtpTransport : public RtpTransport {
     rtp_abs_sendtime_extn_id_ = rtp_abs_sendtime_extn_id;
   }
 
+  // In addition to unregistering the sink, the SRTP transport
+  // disassociates all SSRCs of the sink from libSRTP.
+  bool UnregisterRtpDemuxerSink(RtpPacketSinkInterface* sink) override;
+
  protected:
   // If the writable state changed, fire the SignalWritableState.
   void MaybeUpdateWritableState();
diff --git a/third_party/libwebrtc/pc/srtp_transport_unittest.cc b/third_party/libwebrtc/pc/srtp_transport_unittest.cc
index ac8be8762b..de4ff03179 100644
--- a/third_party/libwebrtc/pc/srtp_transport_unittest.cc
+++ b/third_party/libwebrtc/pc/srtp_transport_unittest.cc
@@ -342,7 +342,7 @@ class SrtpTransportTest : public ::testing::Test, public sigslot::has_slots<> {
   TransportObserver rtp_sink2_;
 
   int sequence_number_ = 0;
-  webrtc::test::ScopedKeyValueConfig field_trials_;
+  test::ScopedKeyValueConfig field_trials_;
 };
 
 class SrtpTransportTestWithExternalAuth
@@ -425,4 +425,66 @@ TEST_F(SrtpTransportTest, TestSetParamsKeyTooShort) {
       rtc::kSrtpAes128CmSha1_80, kTestKey1, kTestKeyLen - 1, extension_ids));
 }
 
+TEST_F(SrtpTransportTest, RemoveSrtpReceiveStream) {
+  test::ScopedKeyValueConfig field_trials(
+      "WebRTC-SrtpRemoveReceiveStream/Enabled/");
+  auto srtp_transport =
+      std::make_unique<SrtpTransport>(/*rtcp_mux_enabled=*/true, field_trials);
+  auto rtp_packet_transport = std::make_unique<rtc::FakePacketTransport>(
+      "fake_packet_transport_loopback");
+
+  bool asymmetric = false;
+  rtp_packet_transport->SetDestination(rtp_packet_transport.get(), asymmetric);
+  srtp_transport->SetRtpPacketTransport(rtp_packet_transport.get());
+
+  TransportObserver rtp_sink;
+
+  std::vector<int> extension_ids;
+  EXPECT_TRUE(srtp_transport->SetRtpParams(
+      rtc::kSrtpAeadAes128Gcm, kTestKeyGcm128_1, kTestKeyGcm128Len,
+      extension_ids, rtc::kSrtpAeadAes128Gcm, kTestKeyGcm128_1,
+      kTestKeyGcm128Len, extension_ids));
+
+  RtpDemuxerCriteria demuxer_criteria;
+  uint32_t ssrc = 0x1;  // SSRC of kPcmuFrame
+  demuxer_criteria.ssrcs().insert(ssrc);
+  EXPECT_TRUE(
+      srtp_transport->RegisterRtpDemuxerSink(demuxer_criteria, &rtp_sink));
+
+  // Create a packet and try to send it three times.
+  size_t rtp_len = sizeof(kPcmuFrame);
+  size_t packet_size = rtp_len + rtc::rtp_auth_tag_len(rtc::kCsAeadAes128Gcm);
+  rtc::Buffer rtp_packet_buffer(packet_size);
+  char* rtp_packet_data = rtp_packet_buffer.data<char>();
+  memcpy(rtp_packet_data, kPcmuFrame, rtp_len);
+
+  // First attempt will succeed.
+  rtc::CopyOnWriteBuffer first_try(rtp_packet_data, rtp_len, packet_size);
+  EXPECT_TRUE(srtp_transport->SendRtpPacket(&first_try, rtc::PacketOptions(),
+                                            cricket::PF_SRTP_BYPASS));
+  EXPECT_EQ(rtp_sink.rtp_count(), 1);
+
+  // Second attempt will be rejected by libSRTP as a replay attack
+  // (srtp_err_status_replay_fail) since the sequence number was already seen.
+  // Hence the packet never reaches the sink.
+  rtc::CopyOnWriteBuffer second_try(rtp_packet_data, rtp_len, packet_size);
+  EXPECT_TRUE(srtp_transport->SendRtpPacket(&second_try, rtc::PacketOptions(),
+                                            cricket::PF_SRTP_BYPASS));
+  EXPECT_EQ(rtp_sink.rtp_count(), 1);
+
+  // Reset the sink.
+  EXPECT_TRUE(srtp_transport->UnregisterRtpDemuxerSink(&rtp_sink));
+  EXPECT_TRUE(
+      srtp_transport->RegisterRtpDemuxerSink(demuxer_criteria, &rtp_sink));
+
+  // Third attempt will succeed again since libSRTP does not remember seeing
+  // the sequence number after the reset.
+  rtc::CopyOnWriteBuffer third_try(rtp_packet_data, rtp_len, packet_size);
+  EXPECT_TRUE(srtp_transport->SendRtpPacket(&third_try, rtc::PacketOptions(),
+                                            cricket::PF_SRTP_BYPASS));
+  EXPECT_EQ(rtp_sink.rtp_count(), 2);
+  // Clear the sink to clean up.
+  srtp_transport->UnregisterRtpDemuxerSink(&rtp_sink);
+}
+
 }  // namespace webrtc
diff --git a/third_party/libwebrtc/pc/test/android_test_initializer.cc b/third_party/libwebrtc/pc/test/android_test_initializer.cc
index 963544cb4b..88b4587789 100644
--- a/third_party/libwebrtc/pc/test/android_test_initializer.cc
+++ b/third_party/libwebrtc/pc/test/android_test_initializer.cc
@@ -39,7 +39,7 @@ void EnsureInitializedOnce() {
 
   RTC_CHECK(rtc::InitializeSSL()) << "Failed to InitializeSSL()";
 
-  webrtc::JVM::Initialize(jvm);
+  JVM::Initialize(jvm);
 }
 
 }  // anonymous namespace
diff --git a/third_party/libwebrtc/pc/test/enable_fake_media.cc b/third_party/libwebrtc/pc/test/enable_fake_media.cc
new file mode 100644
index 0000000000..5497c60724
--- /dev/null
+++ b/third_party/libwebrtc/pc/test/enable_fake_media.cc
@@ -0,0 +1,63 @@
+/*
+ *  Copyright 2023 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 "pc/test/enable_fake_media.h"
+
+#include <memory>
+#include <utility>
+
+#include "absl/base/nullability.h"
+#include "api/environment/environment.h"
+#include "api/peer_connection_interface.h"
+#include "call/call.h"
+#include "call/call_config.h"
+#include "media/base/fake_media_engine.h"
+#include "pc/media_factory.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+using ::cricket::FakeMediaEngine;
+using ::cricket::MediaEngineInterface;
+
+void EnableFakeMedia(
+    PeerConnectionFactoryDependencies& deps,
+    absl::Nonnull<std::unique_ptr<FakeMediaEngine>> fake_media_engine) {
+  class FakeMediaFactory : public MediaFactory {
+   public:
+    explicit FakeMediaFactory(
+        absl::Nonnull<std::unique_ptr<FakeMediaEngine>> fake)
+        : fake_(std::move(fake)) {}
+
+    std::unique_ptr<Call> CreateCall(const CallConfig& config) override {
+      return Call::Create(config);
+    }
+
+    std::unique_ptr<MediaEngineInterface> CreateMediaEngine(
+        const Environment& /*env*/,
+        PeerConnectionFactoryDependencies& /*dependencies*/) {
+      RTC_CHECK(fake_ != nullptr)
+          << "CreateMediaEngine can be called at most once.";
+      return std::move(fake_);
+    }
+
+   private:
+    absl::Nullable<std::unique_ptr<FakeMediaEngine>> fake_;
+  };
+
+  deps.media_factory =
+      std::make_unique<FakeMediaFactory>(std::move(fake_media_engine));
+}
+
+void EnableFakeMedia(PeerConnectionFactoryDependencies& deps) {
+  EnableFakeMedia(deps, std::make_unique<cricket::FakeMediaEngine>());
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/pc/test/enable_fake_media.h b/third_party/libwebrtc/pc/test/enable_fake_media.h
new file mode 100644
index 0000000000..82c55ad08b
--- /dev/null
+++ b/third_party/libwebrtc/pc/test/enable_fake_media.h
@@ -0,0 +1,38 @@
+/*
+ *  Copyright 2023 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.
+ */
+
+// Enables fake media support for PeerConnnectionFactory created from `deps` for
+// testing purposes. Such fake media support ignores media dependencies in the
+// `PeerConnectionFactoryDependencies`. Allows to test PeerConnection and
+// PeerConnectionFactory in the presence of the media, but doesn't test media
+// support itself.
+
+#ifndef PC_TEST_ENABLE_FAKE_MEDIA_H_
+#define PC_TEST_ENABLE_FAKE_MEDIA_H_
+
+#include <memory>
+
+#include "absl/base/nullability.h"
+#include "api/peer_connection_interface.h"
+#include "media/base/fake_media_engine.h"
+
+namespace webrtc {
+
+// Enables media support backed by the 'fake_media_engine'.
+void EnableFakeMedia(
+    PeerConnectionFactoryDependencies& deps,
+    absl::Nonnull<std::unique_ptr<cricket::FakeMediaEngine>> fake_media_engine);
+
+// Enables media support backed by unspecified lightweight fake implementation.
+void EnableFakeMedia(PeerConnectionFactoryDependencies& deps);
+
+}  // namespace webrtc
+
+#endif  //  PC_TEST_ENABLE_FAKE_MEDIA_H_
diff --git a/third_party/libwebrtc/pc/test/fake_peer_connection_base.h b/third_party/libwebrtc/pc/test/fake_peer_connection_base.h
index a1c8dca12e..1615088e99 100644
--- a/third_party/libwebrtc/pc/test/fake_peer_connection_base.h
+++ b/third_party/libwebrtc/pc/test/fake_peer_connection_base.h
@@ -363,7 +363,7 @@ class FakePeerConnectionBase : public PeerConnectionInternal {
   const FieldTrialsView& trials() const override { return field_trials_; }
 
  protected:
-  webrtc::test::ScopedKeyValueConfig field_trials_;
+  test::ScopedKeyValueConfig field_trials_;
 };
 
 }  // namespace webrtc
diff --git a/third_party/libwebrtc/pc/test/fake_peer_connection_for_stats.h b/third_party/libwebrtc/pc/test/fake_peer_connection_for_stats.h
index 7302182912..2883c86b58 100644
--- a/third_party/libwebrtc/pc/test/fake_peer_connection_for_stats.h
+++ b/third_party/libwebrtc/pc/test/fake_peer_connection_for_stats.h
@@ -18,10 +18,11 @@
 #include <utility>
 #include <vector>
 
-#include "media/base/fake_media_engine.h"
+#include "api/environment/environment_factory.h"
 #include "media/base/media_channel.h"
 #include "pc/channel.h"
 #include "pc/stream_collection.h"
+#include "pc/test/enable_fake_media.h"
 #include "pc/test/fake_data_channel_controller.h"
 #include "pc/test/fake_peer_connection_base.h"
 
@@ -150,7 +151,7 @@ class VoiceChannelForTesting : public cricket::VoiceChannel {
           receive_channel,
       const std::string& content_name,
       bool srtp_required,
-      webrtc::CryptoOptions crypto_options,
+      CryptoOptions crypto_options,
       rtc::UniqueRandomIdGenerator* ssrc_generator,
       std::string transport_name)
       : VoiceChannel(worker_thread,
@@ -183,7 +184,7 @@ class VideoChannelForTesting : public cricket::VideoChannel {
           receive_channel,
       const std::string& content_name,
       bool srtp_required,
-      webrtc::CryptoOptions crypto_options,
+      CryptoOptions crypto_options,
       rtc::UniqueRandomIdGenerator* ssrc_generator,
       std::string transport_name)
       : VideoChannel(worker_thread,
@@ -219,7 +220,8 @@ class FakePeerConnectionForStats : public FakePeerConnectionBase {
         signaling_thread_(rtc::Thread::Current()),
         // TODO(hta): remove separate thread variables and use context.
         dependencies_(MakeDependencies()),
-        context_(ConnectionContext::Create(&dependencies_)),
+        context_(
+            ConnectionContext::Create(CreateEnvironment(), &dependencies_)),
         local_streams_(StreamCollection::Create()),
         remote_streams_(StreamCollection::Create()),
         data_channel_controller_(network_thread_) {}
@@ -235,7 +237,7 @@ class FakePeerConnectionForStats : public FakePeerConnectionBase {
     dependencies.network_thread = rtc::Thread::Current();
     dependencies.worker_thread = rtc::Thread::Current();
     dependencies.signaling_thread = rtc::Thread::Current();
-    dependencies.media_engine = std::make_unique<cricket::FakeMediaEngine>();
+    EnableFakeMedia(dependencies);
     return dependencies;
   }
 
@@ -298,7 +300,7 @@ class FakePeerConnectionForStats : public FakePeerConnectionBase {
         worker_thread_, network_thread_, signaling_thread_,
         std::move(voice_media_send_channel),
         std::move(voice_media_receive_channel), mid, kDefaultSrtpRequired,
-        webrtc::CryptoOptions(), context_->ssrc_generator(), transport_name);
+        CryptoOptions(), context_->ssrc_generator(), transport_name);
     auto transceiver =
         GetOrCreateFirstTransceiverOfType(cricket::MEDIA_TYPE_AUDIO)
             ->internal();
@@ -332,7 +334,7 @@ class FakePeerConnectionForStats : public FakePeerConnectionBase {
         worker_thread_, network_thread_, signaling_thread_,
         std::move(video_media_send_channel),
         std::move(video_media_receive_channel), mid, kDefaultSrtpRequired,
-        webrtc::CryptoOptions(), context_->ssrc_generator(), transport_name);
+        CryptoOptions(), context_->ssrc_generator(), transport_name);
     auto transceiver =
         GetOrCreateFirstTransceiverOfType(cricket::MEDIA_TYPE_VIDEO)
             ->internal();
diff --git a/third_party/libwebrtc/pc/test/fake_periodic_video_source.h b/third_party/libwebrtc/pc/test/fake_periodic_video_source.h
index 452a8f6c30..65652bdf0d 100644
--- a/third_party/libwebrtc/pc/test/fake_periodic_video_source.h
+++ b/third_party/libwebrtc/pc/test/fake_periodic_video_source.h
@@ -65,12 +65,12 @@ class FakePeriodicVideoSource final
     return wants_;
   }
 
-  void RemoveSink(rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) override {
+  void RemoveSink(rtc::VideoSinkInterface<VideoFrame>* sink) override {
     RTC_DCHECK(thread_checker_.IsCurrent());
     broadcaster_.RemoveSink(sink);
   }
 
-  void AddOrUpdateSink(rtc::VideoSinkInterface<webrtc::VideoFrame>* sink,
+  void AddOrUpdateSink(rtc::VideoSinkInterface<VideoFrame>* sink,
                        const rtc::VideoSinkWants& wants) override {
     RTC_DCHECK(thread_checker_.IsCurrent());
     {
diff --git a/third_party/libwebrtc/pc/test/integration_test_helpers.cc b/third_party/libwebrtc/pc/test/integration_test_helpers.cc
index ede159d744..64d8debc09 100644
--- a/third_party/libwebrtc/pc/test/integration_test_helpers.cc
+++ b/third_party/libwebrtc/pc/test/integration_test_helpers.cc
@@ -46,7 +46,7 @@ void RemoveSsrcsAndKeepMsids(cricket::SessionDescription* desc) {
 
 int FindFirstMediaStatsIndexByKind(
     const std::string& kind,
-    const std::vector<const webrtc::RTCInboundRtpStreamStats*>& inbound_rtps) {
+    const std::vector<const RTCInboundRtpStreamStats*>& inbound_rtps) {
   for (size_t i = 0; i < inbound_rtps.size(); i++) {
     if (*inbound_rtps[i]->kind == kind) {
       return i;
diff --git a/third_party/libwebrtc/pc/test/integration_test_helpers.h b/third_party/libwebrtc/pc/test/integration_test_helpers.h
index 36b2111324..fb719e7ddd 100644
--- a/third_party/libwebrtc/pc/test/integration_test_helpers.h
+++ b/third_party/libwebrtc/pc/test/integration_test_helpers.h
@@ -31,10 +31,10 @@
 #include "absl/strings/string_view.h"
 #include "absl/types/optional.h"
 #include "api/audio_options.h"
-#include "api/call/call_factory_interface.h"
 #include "api/candidate.h"
 #include "api/crypto/crypto_options.h"
 #include "api/data_channel_interface.h"
+#include "api/enable_media_with_defaults.h"
 #include "api/field_trials_view.h"
 #include "api/ice_transport_interface.h"
 #include "api/jsep.h"
@@ -68,14 +68,11 @@
 #include "media/base/media_engine.h"
 #include "media/base/stream_params.h"
 #include "media/engine/fake_webrtc_video_engine.h"
-#include "media/engine/webrtc_media_engine.h"
-#include "media/engine/webrtc_media_engine_defaults.h"
 #include "modules/audio_device/include/audio_device.h"
 #include "modules/audio_processing/include/audio_processing.h"
 #include "modules/audio_processing/test/audio_processing_builder_for_testing.h"
 #include "p2p/base/fake_ice_transport.h"
 #include "p2p/base/ice_transport_internal.h"
-#include "p2p/base/mock_async_resolver.h"
 #include "p2p/base/p2p_constants.h"
 #include "p2p/base/port.h"
 #include "p2p/base/port_allocator.h"
@@ -179,14 +176,14 @@ void ReplaceFirstSsrc(StreamParams& stream, uint32_t ssrc);
 
 int FindFirstMediaStatsIndexByKind(
     const std::string& kind,
-    const std::vector<const webrtc::RTCInboundRtpStreamStats*>& inbound_rtps);
+    const std::vector<const RTCInboundRtpStreamStats*>& inbound_rtps);
 
-class TaskQueueMetronome : public webrtc::Metronome {
+class TaskQueueMetronome : public Metronome {
  public:
   explicit TaskQueueMetronome(TimeDelta tick_period);
   ~TaskQueueMetronome() override;
 
-  // webrtc::Metronome implementation.
+  // Metronome implementation.
   void RequestCallOnNextTick(absl::AnyInvocable<void() &&> callback) override;
   TimeDelta TickPeriod() const override;
 
@@ -209,7 +206,7 @@ class SignalingMessageReceiver {
   virtual ~SignalingMessageReceiver() {}
 };
 
-class MockRtpReceiverObserver : public webrtc::RtpReceiverObserverInterface {
+class MockRtpReceiverObserver : public RtpReceiverObserverInterface {
  public:
   explicit MockRtpReceiverObserver(cricket::MediaType media_type)
       : expected_media_type_(media_type) {}
@@ -236,14 +233,14 @@ class MockRtpReceiverObserver : public webrtc::RtpReceiverObserverInterface {
 // advertise support of any codecs.
 // TODO(steveanton): See how this could become a subclass of
 // PeerConnectionWrapper defined in peerconnectionwrapper.h.
-class PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver,
+class PeerConnectionIntegrationWrapper : public PeerConnectionObserver,
                                          public SignalingMessageReceiver {
  public:
-  webrtc::PeerConnectionFactoryInterface* pc_factory() const {
+  PeerConnectionFactoryInterface* pc_factory() const {
     return peer_connection_factory_.get();
   }
 
-  webrtc::PeerConnectionInterface* pc() const { return peer_connection_.get(); }
+  PeerConnectionInterface* pc() const { return peer_connection_.get(); }
 
   // If a signaling message receiver is set (via ConnectFakeSignaling), this
   // will set the whole offer/answer exchange in motion. Just need to wait for
@@ -341,11 +338,11 @@ class PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver,
     return AddTrack(CreateLocalVideoTrack());
   }
 
-  rtc::scoped_refptr<webrtc::AudioTrackInterface> CreateLocalAudioTrack() {
+  rtc::scoped_refptr<AudioTrackInterface> CreateLocalAudioTrack() {
     cricket::AudioOptions options;
     // Disable highpass filter so that we can get all the test audio frames.
     options.highpass_filter = false;
-    rtc::scoped_refptr<webrtc::AudioSourceInterface> source =
+    rtc::scoped_refptr<AudioSourceInterface> source =
         peer_connection_factory_->CreateAudioSource(options);
     // TODO(perkj): Test audio source when it is implemented. Currently audio
     // always use the default input.
@@ -353,21 +350,20 @@ class PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver,
                                                       source.get());
   }
 
-  rtc::scoped_refptr<webrtc::VideoTrackInterface> CreateLocalVideoTrack() {
-    webrtc::FakePeriodicVideoSource::Config config;
+  rtc::scoped_refptr<VideoTrackInterface> CreateLocalVideoTrack() {
+    FakePeriodicVideoSource::Config config;
     config.timestamp_offset_ms = rtc::TimeMillis();
     return CreateLocalVideoTrackInternal(config);
   }
 
-  rtc::scoped_refptr<webrtc::VideoTrackInterface>
-  CreateLocalVideoTrackWithConfig(
-      webrtc::FakePeriodicVideoSource::Config config) {
+  rtc::scoped_refptr<VideoTrackInterface> CreateLocalVideoTrackWithConfig(
+      FakePeriodicVideoSource::Config config) {
     return CreateLocalVideoTrackInternal(config);
   }
 
-  rtc::scoped_refptr<webrtc::VideoTrackInterface>
-  CreateLocalVideoTrackWithRotation(webrtc::VideoRotation rotation) {
-    webrtc::FakePeriodicVideoSource::Config config;
+  rtc::scoped_refptr<VideoTrackInterface> CreateLocalVideoTrackWithRotation(
+      VideoRotation rotation) {
+    FakePeriodicVideoSource::Config config;
     config.rotation = rotation;
     config.timestamp_offset_ms = rtc::TimeMillis();
     return CreateLocalVideoTrackInternal(config);
@@ -411,22 +407,22 @@ class PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver,
   }
 
   bool SignalingStateStable() {
-    return pc()->signaling_state() == webrtc::PeerConnectionInterface::kStable;
+    return pc()->signaling_state() == PeerConnectionInterface::kStable;
   }
 
   bool IceGatheringStateComplete() {
     return pc()->ice_gathering_state() ==
-           webrtc::PeerConnectionInterface::kIceGatheringComplete;
+           PeerConnectionInterface::kIceGatheringComplete;
   }
 
   void CreateDataChannel() { CreateDataChannel(nullptr); }
 
-  void CreateDataChannel(const webrtc::DataChannelInit* init) {
+  void CreateDataChannel(const DataChannelInit* init) {
     CreateDataChannel(kDataChannelLabel, init);
   }
 
   void CreateDataChannel(const std::string& label,
-                         const webrtc::DataChannelInit* init) {
+                         const DataChannelInit* init) {
     auto data_channel_or_error = pc()->CreateDataChannelOrError(label, init);
     ASSERT_TRUE(data_channel_or_error.ok());
     data_channels_.push_back(data_channel_or_error.MoveValue());
@@ -484,7 +480,7 @@ class PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver,
   // Returns a MockStatsObserver in a state after stats gathering finished,
   // which can be used to access the gathered stats.
   rtc::scoped_refptr<MockStatsObserver> OldGetStatsForTrack(
-      webrtc::MediaStreamTrackInterface* track) {
+      MediaStreamTrackInterface* track) {
     auto observer = rtc::make_ref_counted<MockStatsObserver>();
     EXPECT_TRUE(peer_connection_->GetStats(
         observer.get(), nullptr,
@@ -500,9 +496,8 @@ class PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver,
 
   // Synchronously gets stats and returns them. If it times out, fails the test
   // and returns null.
-  rtc::scoped_refptr<const webrtc::RTCStatsReport> NewGetStats() {
-    auto callback =
-        rtc::make_ref_counted<webrtc::MockRTCStatsCollectorCallback>();
+  rtc::scoped_refptr<const RTCStatsReport> NewGetStats() {
+    auto callback = rtc::make_ref_counted<MockRTCStatsCollectorCallback>();
     peer_connection_->GetStats(callback.get());
     EXPECT_TRUE_WAIT(callback->called(), kDefaultTimeout);
     return callback->report();
@@ -529,10 +524,10 @@ class PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver,
     return static_cast<double>(rendered_width()) / rendered_height();
   }
 
-  webrtc::VideoRotation rendered_rotation() {
+  VideoRotation rendered_rotation() {
     EXPECT_FALSE(fake_video_renderers_.empty());
     return fake_video_renderers_.empty()
-               ? webrtc::kVideoRotation_0
+               ? kVideoRotation_0
                : fake_video_renderers_.begin()->second->rotation();
   }
 
@@ -575,20 +570,20 @@ class PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver,
     return pc()->local_streams().get();
   }
 
-  webrtc::PeerConnectionInterface::SignalingState signaling_state() {
+  PeerConnectionInterface::SignalingState signaling_state() {
     return pc()->signaling_state();
   }
 
-  webrtc::PeerConnectionInterface::IceConnectionState ice_connection_state() {
+  PeerConnectionInterface::IceConnectionState ice_connection_state() {
     return pc()->ice_connection_state();
   }
 
-  webrtc::PeerConnectionInterface::IceConnectionState
+  PeerConnectionInterface::IceConnectionState
   standardized_ice_connection_state() {
     return pc()->standardized_ice_connection_state();
   }
 
-  webrtc::PeerConnectionInterface::IceGatheringState ice_gathering_state() {
+  PeerConnectionInterface::IceGatheringState ice_gathering_state() {
     return pc()->ice_gathering_state();
   }
 
@@ -617,7 +612,7 @@ class PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver,
   }
   cricket::PortAllocator* port_allocator() const { return port_allocator_; }
 
-  webrtc::FakeRtcEventLogFactory* event_log_factory() const {
+  FakeRtcEventLogFactory* event_log_factory() const {
     return event_log_factory_;
   }
 
@@ -630,8 +625,7 @@ class PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver,
 
   // Sets the mDNS responder for the owned fake network manager and keeps a
   // reference to the responder.
-  void SetMdnsResponder(
-      std::unique_ptr<webrtc::FakeMdnsResponder> mdns_responder) {
+  void SetMdnsResponder(std::unique_ptr<FakeMdnsResponder> mdns_responder) {
     RTC_DCHECK(mdns_responder != nullptr);
     mdns_responder_ = mdns_responder.get();
     network_manager()->set_mdns_responder(std::move(mdns_responder));
@@ -646,7 +640,7 @@ class PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver,
   }
   bool Rollback() {
     return SetRemoteDescription(
-        webrtc::CreateSessionDescription(SdpType::kRollback, ""));
+        CreateSessionDescription(SdpType::kRollback, ""));
   }
 
   // Functions for querying stats.
@@ -654,7 +648,7 @@ class PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver,
     // Get the baseline numbers for audio_packets and audio_delay.
     auto received_stats = NewGetStats();
     auto rtp_stats =
-        received_stats->GetStatsOfType<webrtc::RTCInboundRtpStreamStats>()[0];
+        received_stats->GetStatsOfType<RTCInboundRtpStreamStats>()[0];
     ASSERT_TRUE(rtp_stats->relative_packet_arrival_delay.is_defined());
     ASSERT_TRUE(rtp_stats->packets_received.is_defined());
     rtp_stats_id_ = rtp_stats->id();
@@ -666,8 +660,7 @@ class PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver,
 
   void UpdateDelayStats(std::string tag, int desc_size) {
     auto report = NewGetStats();
-    auto rtp_stats =
-        report->GetAs<webrtc::RTCInboundRtpStreamStats>(rtp_stats_id_);
+    auto rtp_stats = report->GetAs<RTCInboundRtpStreamStats>(rtp_stats_id_);
     ASSERT_TRUE(rtp_stats);
     auto delta_packets = *rtp_stats->packets_received - audio_packets_stat_;
     auto delta_rpad =
@@ -746,11 +739,11 @@ class PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver,
 
   bool Init(const PeerConnectionFactory::Options* options,
             const PeerConnectionInterface::RTCConfiguration* config,
-            webrtc::PeerConnectionDependencies dependencies,
+            PeerConnectionDependencies dependencies,
             rtc::SocketServer* socket_server,
             rtc::Thread* network_thread,
             rtc::Thread* worker_thread,
-            std::unique_ptr<webrtc::FakeRtcEventLogFactory> event_log_factory,
+            std::unique_ptr<FakeRtcEventLogFactory> event_log_factory,
             bool reset_encoder_factory,
             bool reset_decoder_factory,
             bool create_media_engine) {
@@ -773,51 +766,45 @@ class PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver,
     }
     rtc::Thread* const signaling_thread = rtc::Thread::Current();
 
-    webrtc::PeerConnectionFactoryDependencies pc_factory_dependencies;
+    PeerConnectionFactoryDependencies pc_factory_dependencies;
     pc_factory_dependencies.network_thread = network_thread;
     pc_factory_dependencies.worker_thread = worker_thread;
     pc_factory_dependencies.signaling_thread = signaling_thread;
     pc_factory_dependencies.task_queue_factory =
-        webrtc::CreateDefaultTaskQueueFactory();
+        CreateDefaultTaskQueueFactory();
     pc_factory_dependencies.trials = std::make_unique<FieldTrialBasedConfig>();
     pc_factory_dependencies.metronome =
         std::make_unique<TaskQueueMetronome>(TimeDelta::Millis(8));
-    cricket::MediaEngineDependencies media_deps;
-    media_deps.task_queue_factory =
-        pc_factory_dependencies.task_queue_factory.get();
-    media_deps.adm = fake_audio_capture_module_;
-    webrtc::SetMediaEngineDefaults(&media_deps);
+
+    pc_factory_dependencies.adm = fake_audio_capture_module_;
+    if (create_media_engine) {
+      EnableMediaWithDefaults(pc_factory_dependencies);
+    }
 
     if (reset_encoder_factory) {
-      media_deps.video_encoder_factory.reset();
+      pc_factory_dependencies.video_encoder_factory.reset();
     }
     if (reset_decoder_factory) {
-      media_deps.video_decoder_factory.reset();
+      pc_factory_dependencies.video_decoder_factory.reset();
     }
 
-    if (!media_deps.audio_processing) {
+    if (!pc_factory_dependencies.audio_processing) {
       // If the standard Creation method for APM returns a null pointer, instead
       // use the builder for testing to create an APM object.
-      media_deps.audio_processing = AudioProcessingBuilderForTesting().Create();
+      pc_factory_dependencies.audio_processing =
+          AudioProcessingBuilderForTesting().Create();
     }
 
-    media_deps.trials = pc_factory_dependencies.trials.get();
-
-    if (create_media_engine) {
-      pc_factory_dependencies.media_engine =
-          cricket::CreateMediaEngine(std::move(media_deps));
-    }
-    pc_factory_dependencies.call_factory = webrtc::CreateCallFactory();
     if (event_log_factory) {
       event_log_factory_ = event_log_factory.get();
       pc_factory_dependencies.event_log_factory = std::move(event_log_factory);
     } else {
       pc_factory_dependencies.event_log_factory =
-          std::make_unique<webrtc::RtcEventLogFactory>(
+          std::make_unique<RtcEventLogFactory>(
               pc_factory_dependencies.task_queue_factory.get());
     }
-    peer_connection_factory_ = webrtc::CreateModularPeerConnectionFactory(
-        std::move(pc_factory_dependencies));
+    peer_connection_factory_ =
+        CreateModularPeerConnectionFactory(std::move(pc_factory_dependencies));
 
     if (!peer_connection_factory_) {
       return false;
@@ -834,9 +821,9 @@ class PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver,
     return peer_connection_.get() != nullptr;
   }
 
-  rtc::scoped_refptr<webrtc::PeerConnectionInterface> CreatePeerConnection(
+  rtc::scoped_refptr<PeerConnectionInterface> CreatePeerConnection(
       const PeerConnectionInterface::RTCConfiguration* config,
-      webrtc::PeerConnectionDependencies dependencies) {
+      PeerConnectionDependencies dependencies) {
     PeerConnectionInterface::RTCConfiguration modified_config;
     modified_config.sdp_semantics = sdp_semantics_;
     // If `config` is null, this will result in a default configuration being
@@ -869,21 +856,20 @@ class PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver,
     signal_ice_candidates_ = signal;
   }
 
-  rtc::scoped_refptr<webrtc::VideoTrackInterface> CreateLocalVideoTrackInternal(
-      webrtc::FakePeriodicVideoSource::Config config) {
+  rtc::scoped_refptr<VideoTrackInterface> CreateLocalVideoTrackInternal(
+      FakePeriodicVideoSource::Config config) {
     // Set max frame rate to 10fps to reduce the risk of test flakiness.
     // TODO(deadbeef): Do something more robust.
     config.frame_interval_ms = 100;
 
     video_track_sources_.emplace_back(
-        rtc::make_ref_counted<webrtc::FakePeriodicVideoTrackSource>(
+        rtc::make_ref_counted<FakePeriodicVideoTrackSource>(
             config, false /* remote */));
-    rtc::scoped_refptr<webrtc::VideoTrackInterface> track =
+    rtc::scoped_refptr<VideoTrackInterface> track =
         peer_connection_factory_->CreateVideoTrack(video_track_sources_.back(),
                                                    rtc::CreateRandomUuid());
     if (!local_video_renderer_) {
-      local_video_renderer_.reset(
-          new webrtc::FakeVideoTrackRenderer(track.get()));
+      local_video_renderer_.reset(new FakeVideoTrackRenderer(track.get()));
     }
     return track;
   }
@@ -891,7 +877,7 @@ class PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver,
   void HandleIncomingOffer(const std::string& msg) {
     RTC_LOG(LS_INFO) << debug_name_ << ": HandleIncomingOffer";
     std::unique_ptr<SessionDescriptionInterface> desc =
-        webrtc::CreateSessionDescription(SdpType::kOffer, msg);
+        CreateSessionDescription(SdpType::kOffer, msg);
     if (received_sdp_munger_) {
       received_sdp_munger_(desc->description());
     }
@@ -911,7 +897,7 @@ class PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver,
   void HandleIncomingAnswer(const std::string& msg) {
     RTC_LOG(LS_INFO) << debug_name_ << ": HandleIncomingAnswer";
     std::unique_ptr<SessionDescriptionInterface> desc =
-        webrtc::CreateSessionDescription(SdpType::kAnswer, msg);
+        CreateSessionDescription(SdpType::kAnswer, msg);
     if (received_sdp_munger_) {
       received_sdp_munger_(desc->description());
     }
@@ -1062,7 +1048,7 @@ class PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver,
                          const std::string& msg) override {
     RTC_LOG(LS_INFO) << debug_name_ << ": ReceiveIceMessage";
     absl::optional<RTCError> result;
-    pc()->AddIceCandidate(absl::WrapUnique(webrtc::CreateIceCandidate(
+    pc()->AddIceCandidate(absl::WrapUnique(CreateIceCandidate(
                               sdp_mid, sdp_mline_index, msg, nullptr)),
                           [&result](RTCError r) { result = r; });
     EXPECT_TRUE_WAIT(result.has_value(), kDefaultTimeout);
@@ -1071,7 +1057,7 @@ class PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver,
 
   // PeerConnectionObserver callbacks.
   void OnSignalingChange(
-      webrtc::PeerConnectionInterface::SignalingState new_state) override {
+      PeerConnectionInterface::SignalingState new_state) override {
     EXPECT_EQ(pc()->signaling_state(), new_state);
     peer_connection_signaling_state_history_.push_back(new_state);
   }
@@ -1100,21 +1086,21 @@ class PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver,
   }
   void OnRenegotiationNeeded() override {}
   void OnIceConnectionChange(
-      webrtc::PeerConnectionInterface::IceConnectionState new_state) override {
+      PeerConnectionInterface::IceConnectionState new_state) override {
     EXPECT_EQ(pc()->ice_connection_state(), new_state);
     ice_connection_state_history_.push_back(new_state);
   }
   void OnStandardizedIceConnectionChange(
-      webrtc::PeerConnectionInterface::IceConnectionState new_state) override {
+      PeerConnectionInterface::IceConnectionState new_state) override {
     standardized_ice_connection_state_history_.push_back(new_state);
   }
   void OnConnectionChange(
-      webrtc::PeerConnectionInterface::PeerConnectionState new_state) override {
+      PeerConnectionInterface::PeerConnectionState new_state) override {
     peer_connection_state_history_.push_back(new_state);
   }
 
   void OnIceGatheringChange(
-      webrtc::PeerConnectionInterface::IceGatheringState new_state) override {
+      PeerConnectionInterface::IceGatheringState new_state) override {
     EXPECT_EQ(pc()->ice_gathering_state(), new_state);
     ice_gathering_state_history_.push_back(new_state);
   }
@@ -1124,7 +1110,7 @@ class PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver,
     ice_candidate_pair_change_history_.push_back(event);
   }
 
-  void OnIceCandidate(const webrtc::IceCandidateInterface* candidate) override {
+  void OnIceCandidate(const IceCandidateInterface* candidate) override {
     RTC_LOG(LS_INFO) << debug_name_ << ": OnIceCandidate";
 
     if (remote_async_dns_resolver_) {
@@ -1180,20 +1166,19 @@ class PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver,
   std::unique_ptr<rtc::FakeNetworkManager> fake_network_manager_;
   std::unique_ptr<rtc::BasicPacketSocketFactory> socket_factory_;
   // Reference to the mDNS responder owned by `fake_network_manager_` after set.
-  webrtc::FakeMdnsResponder* mdns_responder_ = nullptr;
+  FakeMdnsResponder* mdns_responder_ = nullptr;
 
-  rtc::scoped_refptr<webrtc::PeerConnectionInterface> peer_connection_;
-  rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface>
-      peer_connection_factory_;
+  rtc::scoped_refptr<PeerConnectionInterface> peer_connection_;
+  rtc::scoped_refptr<PeerConnectionFactoryInterface> peer_connection_factory_;
 
   cricket::PortAllocator* port_allocator_;
   // Needed to keep track of number of frames sent.
   rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_;
   // Needed to keep track of number of frames received.
-  std::map<std::string, std::unique_ptr<webrtc::FakeVideoTrackRenderer>>
+  std::map<std::string, std::unique_ptr<FakeVideoTrackRenderer>>
       fake_video_renderers_;
   // Needed to ensure frames aren't received for removed tracks.
-  std::vector<std::unique_ptr<webrtc::FakeVideoTrackRenderer>>
+  std::vector<std::unique_ptr<FakeVideoTrackRenderer>>
       removed_fake_video_renderers_;
 
   // For remote peer communication.
@@ -1205,10 +1190,9 @@ class PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver,
 
   // Store references to the video sources we've created, so that we can stop
   // them, if required.
-  std::vector<rtc::scoped_refptr<webrtc::VideoTrackSource>>
-      video_track_sources_;
+  std::vector<rtc::scoped_refptr<VideoTrackSource>> video_track_sources_;
   // `local_video_renderer_` attached to the first created local video track.
-  std::unique_ptr<webrtc::FakeVideoTrackRenderer> local_video_renderer_;
+  std::unique_ptr<FakeVideoTrackRenderer> local_video_renderer_;
 
   SdpSemantics sdp_semantics_;
   PeerConnectionInterface::RTCOfferAnswerOptions offer_answer_options_;
@@ -1238,7 +1222,7 @@ class PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver,
       ice_candidate_pair_change_history_;
   std::vector<PeerConnectionInterface::SignalingState>
       peer_connection_signaling_state_history_;
-  webrtc::FakeRtcEventLogFactory* event_log_factory_;
+  FakeRtcEventLogFactory* event_log_factory_;
 
   // Number of ICE candidates expected. The default is no limit.
   int candidates_expected_ = std::numeric_limits<int>::max();
@@ -1255,7 +1239,7 @@ class PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver,
   friend class PeerConnectionIntegrationBaseTest;
 };
 
-class MockRtcEventLogOutput : public webrtc::RtcEventLogOutput {
+class MockRtcEventLogOutput : public RtcEventLogOutput {
  public:
   virtual ~MockRtcEventLogOutput() = default;
   MOCK_METHOD(bool, IsActive, (), (const, override));
@@ -1367,7 +1351,7 @@ class MediaExpectations {
   int callee_video_frames_expected_ = 0;
 };
 
-class MockIceTransport : public webrtc::IceTransportInterface {
+class MockIceTransport : public IceTransportInterface {
  public:
   MockIceTransport(const std::string& name, int component)
       : internal_(std::make_unique<cricket::FakeIceTransport>(
@@ -1415,7 +1399,7 @@ class PeerConnectionIntegrationBaseTest : public ::testing::Test {
     worker_thread_->SetName("PCWorkerThread", this);
     RTC_CHECK(network_thread_->Start());
     RTC_CHECK(worker_thread_->Start());
-    webrtc::metrics::Reset();
+    metrics::Reset();
   }
 
   ~PeerConnectionIntegrationBaseTest() {
@@ -1452,13 +1436,13 @@ class PeerConnectionIntegrationBaseTest : public ::testing::Test {
     // are connected. This is an important distinction. Once we have separate
     // ICE and DTLS state, this check needs to use the DTLS state.
     return (callee()->ice_connection_state() ==
-                webrtc::PeerConnectionInterface::kIceConnectionConnected ||
+                PeerConnectionInterface::kIceConnectionConnected ||
             callee()->ice_connection_state() ==
-                webrtc::PeerConnectionInterface::kIceConnectionCompleted) &&
+                PeerConnectionInterface::kIceConnectionCompleted) &&
            (caller()->ice_connection_state() ==
-                webrtc::PeerConnectionInterface::kIceConnectionConnected ||
+                PeerConnectionInterface::kIceConnectionConnected ||
             caller()->ice_connection_state() ==
-                webrtc::PeerConnectionInterface::kIceConnectionCompleted);
+                PeerConnectionInterface::kIceConnectionCompleted);
   }
 
   // When `event_log_factory` is null, the default implementation of the event
@@ -1467,8 +1451,8 @@ class PeerConnectionIntegrationBaseTest : public ::testing::Test {
       const std::string& debug_name,
       const PeerConnectionFactory::Options* options,
       const RTCConfiguration* config,
-      webrtc::PeerConnectionDependencies dependencies,
-      std::unique_ptr<webrtc::FakeRtcEventLogFactory> event_log_factory,
+      PeerConnectionDependencies dependencies,
+      std::unique_ptr<FakeRtcEventLogFactory> event_log_factory,
       bool reset_encoder_factory,
       bool reset_decoder_factory,
       bool create_media_engine = true) {
@@ -1498,10 +1482,10 @@ class PeerConnectionIntegrationBaseTest : public ::testing::Test {
       const std::string& debug_name,
       const PeerConnectionFactory::Options* options,
       const RTCConfiguration* config,
-      webrtc::PeerConnectionDependencies dependencies) {
+      PeerConnectionDependencies dependencies) {
     return CreatePeerConnectionWrapper(
         debug_name, options, config, std::move(dependencies),
-        std::make_unique<webrtc::FakeRtcEventLogFactory>(),
+        std::make_unique<FakeRtcEventLogFactory>(),
         /*reset_encoder_factory=*/false,
         /*reset_decoder_factory=*/false);
   }
@@ -1522,17 +1506,17 @@ class PeerConnectionIntegrationBaseTest : public ::testing::Test {
     // callee PeerConnections.
     SdpSemantics original_semantics = sdp_semantics_;
     sdp_semantics_ = caller_semantics;
-    caller_ = CreatePeerConnectionWrapper(
-        "Caller", nullptr, nullptr, webrtc::PeerConnectionDependencies(nullptr),
-        nullptr,
-        /*reset_encoder_factory=*/false,
-        /*reset_decoder_factory=*/false);
+    caller_ = CreatePeerConnectionWrapper("Caller", nullptr, nullptr,
+                                          PeerConnectionDependencies(nullptr),
+                                          nullptr,
+                                          /*reset_encoder_factory=*/false,
+                                          /*reset_decoder_factory=*/false);
     sdp_semantics_ = callee_semantics;
-    callee_ = CreatePeerConnectionWrapper(
-        "Callee", nullptr, nullptr, webrtc::PeerConnectionDependencies(nullptr),
-        nullptr,
-        /*reset_encoder_factory=*/false,
-        /*reset_decoder_factory=*/false);
+    callee_ = CreatePeerConnectionWrapper("Callee", nullptr, nullptr,
+                                          PeerConnectionDependencies(nullptr),
+                                          nullptr,
+                                          /*reset_encoder_factory=*/false,
+                                          /*reset_decoder_factory=*/false);
     sdp_semantics_ = original_semantics;
     return caller_ && callee_;
   }
@@ -1540,24 +1524,24 @@ class PeerConnectionIntegrationBaseTest : public ::testing::Test {
   bool CreatePeerConnectionWrappersWithConfig(
       const PeerConnectionInterface::RTCConfiguration& caller_config,
       const PeerConnectionInterface::RTCConfiguration& callee_config) {
-    caller_ = CreatePeerConnectionWrapper(
-        "Caller", nullptr, &caller_config,
-        webrtc::PeerConnectionDependencies(nullptr), nullptr,
-        /*reset_encoder_factory=*/false,
-        /*reset_decoder_factory=*/false);
-    callee_ = CreatePeerConnectionWrapper(
-        "Callee", nullptr, &callee_config,
-        webrtc::PeerConnectionDependencies(nullptr), nullptr,
-        /*reset_encoder_factory=*/false,
-        /*reset_decoder_factory=*/false);
+    caller_ = CreatePeerConnectionWrapper("Caller", nullptr, &caller_config,
+                                          PeerConnectionDependencies(nullptr),
+                                          nullptr,
+                                          /*reset_encoder_factory=*/false,
+                                          /*reset_decoder_factory=*/false);
+    callee_ = CreatePeerConnectionWrapper("Callee", nullptr, &callee_config,
+                                          PeerConnectionDependencies(nullptr),
+                                          nullptr,
+                                          /*reset_encoder_factory=*/false,
+                                          /*reset_decoder_factory=*/false);
     return caller_ && callee_;
   }
 
   bool CreatePeerConnectionWrappersWithConfigAndDeps(
       const PeerConnectionInterface::RTCConfiguration& caller_config,
-      webrtc::PeerConnectionDependencies caller_dependencies,
+      PeerConnectionDependencies caller_dependencies,
       const PeerConnectionInterface::RTCConfiguration& callee_config,
-      webrtc::PeerConnectionDependencies callee_dependencies) {
+      PeerConnectionDependencies callee_dependencies) {
     caller_ =
         CreatePeerConnectionWrapper("Caller", nullptr, &caller_config,
                                     std::move(caller_dependencies), nullptr,
@@ -1574,16 +1558,16 @@ class PeerConnectionIntegrationBaseTest : public ::testing::Test {
   bool CreatePeerConnectionWrappersWithOptions(
       const PeerConnectionFactory::Options& caller_options,
       const PeerConnectionFactory::Options& callee_options) {
-    caller_ = CreatePeerConnectionWrapper(
-        "Caller", &caller_options, nullptr,
-        webrtc::PeerConnectionDependencies(nullptr), nullptr,
-        /*reset_encoder_factory=*/false,
-        /*reset_decoder_factory=*/false);
-    callee_ = CreatePeerConnectionWrapper(
-        "Callee", &callee_options, nullptr,
-        webrtc::PeerConnectionDependencies(nullptr), nullptr,
-        /*reset_encoder_factory=*/false,
-        /*reset_decoder_factory=*/false);
+    caller_ = CreatePeerConnectionWrapper("Caller", &caller_options, nullptr,
+                                          PeerConnectionDependencies(nullptr),
+                                          nullptr,
+                                          /*reset_encoder_factory=*/false,
+                                          /*reset_decoder_factory=*/false);
+    callee_ = CreatePeerConnectionWrapper("Callee", &callee_options, nullptr,
+                                          PeerConnectionDependencies(nullptr),
+                                          nullptr,
+                                          /*reset_encoder_factory=*/false,
+                                          /*reset_decoder_factory=*/false);
     return caller_ && callee_;
   }
 
@@ -1591,10 +1575,10 @@ class PeerConnectionIntegrationBaseTest : public ::testing::Test {
     PeerConnectionInterface::RTCConfiguration default_config;
     caller_ = CreatePeerConnectionWrapperWithFakeRtcEventLog(
         "Caller", nullptr, &default_config,
-        webrtc::PeerConnectionDependencies(nullptr));
+        PeerConnectionDependencies(nullptr));
     callee_ = CreatePeerConnectionWrapperWithFakeRtcEventLog(
         "Callee", nullptr, &default_config,
-        webrtc::PeerConnectionDependencies(nullptr));
+        PeerConnectionDependencies(nullptr));
     return caller_ && callee_;
   }
 
@@ -1604,7 +1588,7 @@ class PeerConnectionIntegrationBaseTest : public ::testing::Test {
         new FakeRTCCertificateGenerator());
     cert_generator->use_alternate_key();
 
-    webrtc::PeerConnectionDependencies dependencies(nullptr);
+    PeerConnectionDependencies dependencies(nullptr);
     dependencies.cert_generator = std::move(cert_generator);
     return CreatePeerConnectionWrapper("New Peer", nullptr, nullptr,
                                        std::move(dependencies), nullptr,
@@ -1614,12 +1598,12 @@ class PeerConnectionIntegrationBaseTest : public ::testing::Test {
 
   bool CreateOneDirectionalPeerConnectionWrappers(bool caller_to_callee) {
     caller_ = CreatePeerConnectionWrapper(
-        "Caller", nullptr, nullptr, webrtc::PeerConnectionDependencies(nullptr),
+        "Caller", nullptr, nullptr, PeerConnectionDependencies(nullptr),
         nullptr,
         /*reset_encoder_factory=*/!caller_to_callee,
         /*reset_decoder_factory=*/caller_to_callee);
     callee_ = CreatePeerConnectionWrapper(
-        "Callee", nullptr, nullptr, webrtc::PeerConnectionDependencies(nullptr),
+        "Callee", nullptr, nullptr, PeerConnectionDependencies(nullptr),
         nullptr,
         /*reset_encoder_factory=*/caller_to_callee,
         /*reset_decoder_factory=*/!caller_to_callee);
@@ -1627,18 +1611,18 @@ class PeerConnectionIntegrationBaseTest : public ::testing::Test {
   }
 
   bool CreatePeerConnectionWrappersWithoutMediaEngine() {
-    caller_ = CreatePeerConnectionWrapper(
-        "Caller", nullptr, nullptr, webrtc::PeerConnectionDependencies(nullptr),
-        nullptr,
-        /*reset_encoder_factory=*/false,
-        /*reset_decoder_factory=*/false,
-        /*create_media_engine=*/false);
-    callee_ = CreatePeerConnectionWrapper(
-        "Callee", nullptr, nullptr, webrtc::PeerConnectionDependencies(nullptr),
-        nullptr,
-        /*reset_encoder_factory=*/false,
-        /*reset_decoder_factory=*/false,
-        /*create_media_engine=*/false);
+    caller_ = CreatePeerConnectionWrapper("Caller", nullptr, nullptr,
+                                          PeerConnectionDependencies(nullptr),
+                                          nullptr,
+                                          /*reset_encoder_factory=*/false,
+                                          /*reset_decoder_factory=*/false,
+                                          /*create_media_engine=*/false);
+    callee_ = CreatePeerConnectionWrapper("Callee", nullptr, nullptr,
+                                          PeerConnectionDependencies(nullptr),
+                                          nullptr,
+                                          /*reset_encoder_factory=*/false,
+                                          /*reset_decoder_factory=*/false,
+                                          /*create_media_engine=*/false);
     return caller_ && callee_;
   }
 
@@ -1708,7 +1692,7 @@ class PeerConnectionIntegrationBaseTest : public ::testing::Test {
 
   // Messages may get lost on the unreliable DataChannel, so we send multiple
   // times to avoid test flakiness.
-  void SendRtpDataWithRetries(webrtc::DataChannelInterface* dc,
+  void SendRtpDataWithRetries(DataChannelInterface* dc,
                               const std::string& data,
                               int retries) {
     for (int i = 0; i < retries; ++i) {
diff --git a/third_party/libwebrtc/pc/test/mock_peer_connection_observers.h b/third_party/libwebrtc/pc/test/mock_peer_connection_observers.h
index e9d97a97f6..6222ef7719 100644
--- a/third_party/libwebrtc/pc/test/mock_peer_connection_observers.h
+++ b/third_party/libwebrtc/pc/test/mock_peer_connection_observers.h
@@ -254,7 +254,7 @@ class MockPeerConnectionObserver : public PeerConnectionObserver {
 };
 
 class MockCreateSessionDescriptionObserver
-    : public webrtc::CreateSessionDescriptionObserver {
+    : public CreateSessionDescriptionObserver {
  public:
   MockCreateSessionDescriptionObserver()
       : called_(false),
@@ -266,7 +266,7 @@ class MockCreateSessionDescriptionObserver
     error_ = "";
     desc_.reset(desc);
   }
-  void OnFailure(webrtc::RTCError error) override {
+  void OnFailure(RTCError error) override {
     MutexLock lock(&mutex_);
     called_ = true;
     error_ = error.message();
@@ -295,8 +295,7 @@ class MockCreateSessionDescriptionObserver
   std::unique_ptr<SessionDescriptionInterface> desc_ RTC_GUARDED_BY(mutex_);
 };
 
-class MockSetSessionDescriptionObserver
-    : public webrtc::SetSessionDescriptionObserver {
+class MockSetSessionDescriptionObserver : public SetSessionDescriptionObserver {
  public:
   static rtc::scoped_refptr<MockSetSessionDescriptionObserver> Create() {
     return rtc::make_ref_counted<MockSetSessionDescriptionObserver>();
@@ -312,7 +311,7 @@ class MockSetSessionDescriptionObserver
     called_ = true;
     error_ = "";
   }
-  void OnFailure(webrtc::RTCError error) override {
+  void OnFailure(RTCError error) override {
     MutexLock lock(&mutex_);
     called_ = true;
     error_ = error.message();
@@ -375,14 +374,14 @@ class FakeSetRemoteDescriptionObserver
   absl::optional<RTCError> error_;
 };
 
-class MockDataChannelObserver : public webrtc::DataChannelObserver {
+class MockDataChannelObserver : public DataChannelObserver {
  public:
   struct Message {
     std::string data;
     bool binary;
   };
 
-  explicit MockDataChannelObserver(webrtc::DataChannelInterface* channel)
+  explicit MockDataChannelObserver(DataChannelInterface* channel)
       : channel_(channel) {
     channel_->RegisterObserver(this);
     states_.push_back(channel_->state());
@@ -419,12 +418,12 @@ class MockDataChannelObserver : public webrtc::DataChannelObserver {
   }
 
  private:
-  rtc::scoped_refptr<webrtc::DataChannelInterface> channel_;
+  rtc::scoped_refptr<DataChannelInterface> channel_;
   std::vector<DataChannelInterface::DataState> states_;
   std::vector<Message> messages_;
 };
 
-class MockStatsObserver : public webrtc::StatsObserver {
+class MockStatsObserver : public StatsObserver {
  public:
   MockStatsObserver() : called_(false), stats_() {}
   virtual ~MockStatsObserver() {}
@@ -576,7 +575,7 @@ class MockStatsObserver : public webrtc::StatsObserver {
 };
 
 // Helper class that just stores the report from the callback.
-class MockRTCStatsCollectorCallback : public webrtc::RTCStatsCollectorCallback {
+class MockRTCStatsCollectorCallback : public RTCStatsCollectorCallback {
  public:
   rtc::scoped_refptr<const RTCStatsReport> report() { return report_; }
 
diff --git a/third_party/libwebrtc/pc/test/rtp_transport_test_util.h b/third_party/libwebrtc/pc/test/rtp_transport_test_util.h
index 593ee002c9..563014f94a 100644
--- a/third_party/libwebrtc/pc/test/rtp_transport_test_util.h
+++ b/third_party/libwebrtc/pc/test/rtp_transport_test_util.h
@@ -33,9 +33,14 @@ class TransportObserver : public RtpPacketSinkInterface {
     rtp_transport->SubscribeReadyToSend(
         this, [this](bool arg) { OnReadyToSend(arg); });
     rtp_transport->SetUnDemuxableRtpPacketReceivedHandler(
-        [this](webrtc::RtpPacketReceived& packet) {
-          OnUndemuxableRtpPacket(packet);
-        });
+        [this](RtpPacketReceived& packet) { OnUndemuxableRtpPacket(packet); });
+    rtp_transport->SubscribeSentPacket(this,
+                                       [this](const rtc::SentPacket& packet) {
+                                         sent_packet_count_++;
+                                         if (action_on_sent_packet_) {
+                                           action_on_sent_packet_();
+                                         }
+                                       });
   }
 
   // RtpPacketInterface override.
@@ -57,6 +62,7 @@ class TransportObserver : public RtpPacketSinkInterface {
   int rtp_count() const { return rtp_count_; }
   int un_demuxable_rtp_count() const { return un_demuxable_rtp_count_; }
   int rtcp_count() const { return rtcp_count_; }
+  int sent_packet_count() const { return sent_packet_count_; }
 
   rtc::CopyOnWriteBuffer last_recv_rtp_packet() {
     return last_recv_rtp_packet_;
@@ -81,16 +87,21 @@ class TransportObserver : public RtpPacketSinkInterface {
   void SetActionOnReadyToSend(absl::AnyInvocable<void(bool)> action) {
     action_on_ready_to_send_ = std::move(action);
   }
+  void SetActionOnSentPacket(absl::AnyInvocable<void()> action) {
+    action_on_sent_packet_ = std::move(action);
+  }
 
  private:
   bool ready_to_send_ = false;
   int rtp_count_ = 0;
   int un_demuxable_rtp_count_ = 0;
   int rtcp_count_ = 0;
+  int sent_packet_count_ = 0;
   int ready_to_send_signal_count_ = 0;
   rtc::CopyOnWriteBuffer last_recv_rtp_packet_;
   rtc::CopyOnWriteBuffer last_recv_rtcp_packet_;
   absl::AnyInvocable<void(bool)> action_on_ready_to_send_;
+  absl::AnyInvocable<void()> action_on_sent_packet_;
 };
 
 }  // namespace webrtc
diff --git a/third_party/libwebrtc/pc/test/svc_e2e_tests.cc b/third_party/libwebrtc/pc/test/svc_e2e_tests.cc
index ae35c7f676..3fde5a44e0 100644
--- a/third_party/libwebrtc/pc/test/svc_e2e_tests.cc
+++ b/third_party/libwebrtc/pc/test/svc_e2e_tests.cc
@@ -160,10 +160,9 @@ std::string SvcTestNameGenerator(
 // encoder and decoder level.
 class SvcVideoQualityAnalyzer : public DefaultVideoQualityAnalyzer {
  public:
-  using SpatialTemporalLayerCounts =
-      webrtc::flat_map<int, webrtc::flat_map<int, int>>;
+  using SpatialTemporalLayerCounts = flat_map<int, flat_map<int, int>>;
 
-  explicit SvcVideoQualityAnalyzer(webrtc::Clock* clock)
+  explicit SvcVideoQualityAnalyzer(Clock* clock)
       : DefaultVideoQualityAnalyzer(clock,
                                     test::GetGlobalMetricsLogger(),
                                     DefaultVideoQualityAnalyzerOptions{
@@ -315,9 +314,9 @@ TEST_P(SvcTest, ScalabilityModeSupported) {
   if (UseDependencyDescriptor()) {
     trials += "WebRTC-DependencyDescriptorAdvertised/Enabled/";
   }
-  webrtc::test::ScopedFieldTrials override_trials(AppendFieldTrials(trials));
+  test::ScopedFieldTrials override_trials(AppendFieldTrials(trials));
   std::unique_ptr<NetworkEmulationManager> network_emulation_manager =
-      CreateNetworkEmulationManager(webrtc::TimeMode::kSimulated);
+      CreateNetworkEmulationManager(TimeMode::kSimulated);
   auto analyzer = std::make_unique<SvcVideoQualityAnalyzer>(
       network_emulation_manager->time_controller()->GetClock());
   SvcVideoQualityAnalyzer* analyzer_ptr = analyzer.get();
@@ -475,16 +474,17 @@ INSTANTIATE_TEST_SUITE_P(
                 SvcTestParameters::Create(kAv1CodecName, "L2T3h"),
                 SvcTestParameters::Create(kAv1CodecName, "L2T3_KEY"),
                 // SvcTestParameters::Create(kAv1CodecName, "L2T3_KEY_SHIFT"),
-                SvcTestParameters::Create(kAv1CodecName, "L3T1"),
-                SvcTestParameters::Create(kAv1CodecName, "L3T1h"),
-                SvcTestParameters::Create(kAv1CodecName, "L3T1_KEY"),
-                SvcTestParameters::Create(kAv1CodecName, "L3T2"),
-                SvcTestParameters::Create(kAv1CodecName, "L3T2h"),
-                SvcTestParameters::Create(kAv1CodecName, "L3T2_KEY"),
+                // TODO(bugs.webrtc.org/15666): Investigate and reenable AV1
+                // L3 tests. SvcTestParameters::Create(kAv1CodecName, "L3T1"),
+                // SvcTestParameters::Create(kAv1CodecName, "L3T1h"),
+                // SvcTestParameters::Create(kAv1CodecName, "L3T1_KEY"),
+                // SvcTestParameters::Create(kAv1CodecName, "L3T2"),
+                // SvcTestParameters::Create(kAv1CodecName, "L3T2h"),
+                // SvcTestParameters::Create(kAv1CodecName, "L3T2_KEY"),
                 // SvcTestParameters::Create(kAv1CodecName, "L3T2_KEY_SHIFT"),
-                SvcTestParameters::Create(kAv1CodecName, "L3T3"),
-                SvcTestParameters::Create(kAv1CodecName, "L3T3h"),
-                SvcTestParameters::Create(kAv1CodecName, "L3T3_KEY"),
+                // SvcTestParameters::Create(kAv1CodecName, "L3T3"),
+                // SvcTestParameters::Create(kAv1CodecName, "L3T3h"),
+                // SvcTestParameters::Create(kAv1CodecName, "L3T3_KEY"),
                 // SvcTestParameters::Create(kAv1CodecName, "L3T3_KEY_SHIFT"),
                 SvcTestParameters::Create(kAv1CodecName, "S2T1"),
                 SvcTestParameters::Create(kAv1CodecName, "S2T1h"),
@@ -492,12 +492,14 @@ INSTANTIATE_TEST_SUITE_P(
                 SvcTestParameters::Create(kAv1CodecName, "S2T2h"),
                 SvcTestParameters::Create(kAv1CodecName, "S2T3"),
                 SvcTestParameters::Create(kAv1CodecName, "S2T3h"),
-                SvcTestParameters::Create(kAv1CodecName, "S3T1"),
-                SvcTestParameters::Create(kAv1CodecName, "S3T1h"),
-                SvcTestParameters::Create(kAv1CodecName, "S3T2"),
-                SvcTestParameters::Create(kAv1CodecName, "S3T2h"),
-                SvcTestParameters::Create(kAv1CodecName, "S3T3"),
-                SvcTestParameters::Create(kAv1CodecName, "S3T3h"),
+                // TODO(bugs.webrtc.org/15666): Investigate and reenable AV1
+                // S3 tests.
+                // SvcTestParameters::Create(kAv1CodecName, "S3T1"),
+                // SvcTestParameters::Create(kAv1CodecName, "S3T1h"),
+                // SvcTestParameters::Create(kAv1CodecName, "S3T2"),
+                // SvcTestParameters::Create(kAv1CodecName, "S3T2h"),
+                // SvcTestParameters::Create(kAv1CodecName, "S3T3"),
+                // SvcTestParameters::Create(kAv1CodecName, "S3T3h"),
             }),
             Values(UseDependencyDescriptor::Enabled)),
     SvcTestNameGenerator);
diff --git a/third_party/libwebrtc/pc/video_rtp_receiver_unittest.cc b/third_party/libwebrtc/pc/video_rtp_receiver_unittest.cc
index 5ff736084f..e9729170b7 100644
--- a/third_party/libwebrtc/pc/video_rtp_receiver_unittest.cc
+++ b/third_party/libwebrtc/pc/video_rtp_receiver_unittest.cc
@@ -94,7 +94,7 @@ class VideoRtpReceiverTest : public testing::Test {
              [&]() { receiver_->SetMediaChannel(media_channel); });
   }
 
-  webrtc::VideoTrackSourceInterface* Source() {
+  VideoTrackSourceInterface* Source() {
     return receiver_->streams()[0]->FindVideoTrack("receiver")->GetSource();
   }
 
diff --git a/third_party/libwebrtc/pc/video_rtp_track_source_unittest.cc b/third_party/libwebrtc/pc/video_rtp_track_source_unittest.cc
index 13728c7eff..55632cea42 100644
--- a/third_party/libwebrtc/pc/video_rtp_track_source_unittest.cc
+++ b/third_party/libwebrtc/pc/video_rtp_track_source_unittest.cc
@@ -109,11 +109,11 @@ TEST(VideoRtpTrackSourceTest, NoCallbacksAfterClearedCallback) {
 
 class TestFrame : public RecordableEncodedFrame {
  public:
-  rtc::scoped_refptr<const webrtc::EncodedImageBufferInterface> encoded_buffer()
+  rtc::scoped_refptr<const EncodedImageBufferInterface> encoded_buffer()
       const override {
     return nullptr;
   }
-  absl::optional<webrtc::ColorSpace> color_space() const override {
+  absl::optional<ColorSpace> color_space() const override {
     return absl::nullopt;
   }
   VideoCodecType codec() const override { return kVideoCodecGeneric; }
diff --git a/third_party/libwebrtc/pc/video_track.h b/third_party/libwebrtc/pc/video_track.h
index 13a51c454b..e504182c82 100644
--- a/third_party/libwebrtc/pc/video_track.h
+++ b/third_party/libwebrtc/pc/video_track.h
@@ -70,7 +70,7 @@ class VideoTrack : public MediaStreamTrack<VideoTrackInterface>,
   // Implements ObserverInterface. Observes `video_source_` state.
   void OnChanged() override;
 
-  RTC_NO_UNIQUE_ADDRESS webrtc::SequenceChecker signaling_thread_;
+  RTC_NO_UNIQUE_ADDRESS SequenceChecker signaling_thread_;
   rtc::Thread* const worker_thread_;
   const rtc::scoped_refptr<
       VideoTrackSourceProxyWithInternal<VideoTrackSourceInterface>>
diff --git a/third_party/libwebrtc/pc/video_track_source_proxy.h b/third_party/libwebrtc/pc/video_track_source_proxy.h
index 8500a98766..40d24234ad 100644
--- a/third_party/libwebrtc/pc/video_track_source_proxy.h
+++ b/third_party/libwebrtc/pc/video_track_source_proxy.h
@@ -52,7 +52,7 @@ PROXY_SECONDARY_METHOD1(void,
                         rtc::VideoSinkInterface<RecordableEncodedFrame>*)
 PROXY_SECONDARY_METHOD1(void,
                         ProcessConstraints,
-                        const webrtc::VideoTrackSourceConstraints&)
+                        const VideoTrackSourceConstraints&)
 END_PROXY_MAP(VideoTrackSource)
 
 }  // namespace webrtc
diff --git a/third_party/libwebrtc/pc/webrtc_sdp.cc b/third_party/libwebrtc/pc/webrtc_sdp.cc
index 2a3173cb02..da024eab81 100644
--- a/third_party/libwebrtc/pc/webrtc_sdp.cc
+++ b/third_party/libwebrtc/pc/webrtc_sdp.cc
@@ -267,11 +267,9 @@ struct SsrcInfo {
   std::string stream_id;
   std::string track_id;
 };
-typedef std::vector<SsrcInfo> SsrcInfoVec;
-typedef std::vector<SsrcGroup> SsrcGroupVec;
+using SsrcInfoVec = std::vector<SsrcInfo>;
+using SsrcGroupVec = std::vector<SsrcGroup>;
 
-template <class T>
-static void AddFmtpLine(const T& codec, std::string* message);
 static void BuildMediaDescription(const ContentInfo* content_info,
                                   const TransportInfo* transport_info,
                                   const cricket::MediaType media_type,
@@ -2628,7 +2626,7 @@ static std::unique_ptr<MediaContentDescription> ParseContentDescription(
     int* msid_signaling,
     TransportDescription* transport,
     std::vector<std::unique_ptr<JsepIceCandidate>>* candidates,
-    webrtc::SdpParseError* error) {
+    SdpParseError* error) {
   std::unique_ptr<MediaContentDescription> media_desc;
   if (media_type == cricket::MediaType::MEDIA_TYPE_AUDIO) {
     media_desc = std::make_unique<AudioContentDescription>();
diff --git a/third_party/libwebrtc/pc/webrtc_sdp_unittest.cc b/third_party/libwebrtc/pc/webrtc_sdp_unittest.cc
index 2c43c35d15..ae26ba0ce2 100644
--- a/third_party/libwebrtc/pc/webrtc_sdp_unittest.cc
+++ b/third_party/libwebrtc/pc/webrtc_sdp_unittest.cc
@@ -1316,8 +1316,9 @@ class WebRtcSdpTest : public ::testing::Test {
     return video;
   }
 
-  template <class MCD>
-  void CompareMediaContentDescription(const MCD* cd1, const MCD* cd2) {
+  void CompareMediaContentDescription(
+      const cricket::MediaContentDescription* cd1,
+      const cricket::MediaContentDescription* cd2) {
     // type
     EXPECT_EQ(cd1->type(), cd2->type());
 
@@ -1427,20 +1428,14 @@ class WebRtcSdpTest : public ::testing::Test {
 
       ASSERT_EQ(IsAudioContent(&c1), IsAudioContent(&c2));
       if (IsAudioContent(&c1)) {
-        const AudioContentDescription* acd1 =
-            c1.media_description()->as_audio();
-        const AudioContentDescription* acd2 =
-            c2.media_description()->as_audio();
-        CompareMediaContentDescription<AudioContentDescription>(acd1, acd2);
+        CompareMediaContentDescription(c1.media_description(),
+                                       c2.media_description());
       }
 
       ASSERT_EQ(IsVideoContent(&c1), IsVideoContent(&c2));
       if (IsVideoContent(&c1)) {
-        const VideoContentDescription* vcd1 =
-            c1.media_description()->as_video();
-        const VideoContentDescription* vcd2 =
-            c2.media_description()->as_video();
-        CompareMediaContentDescription<VideoContentDescription>(vcd1, vcd2);
+        CompareMediaContentDescription(c1.media_description(),
+                                       c2.media_description());
       }
 
       ASSERT_EQ(IsDataContent(&c1), IsDataContent(&c2));
@@ -5015,7 +5010,7 @@ TEST_F(WebRtcSdpTest, ParseSessionLevelExtmapAttributes) {
   EXPECT_TRUE(SdpDeserialize(sdp, &jdesc));
   ASSERT_EQ(1u, jdesc.description()->contents().size());
   const auto content = jdesc.description()->contents()[0];
-  const auto* audio_description = content.media_description()->as_audio();
+  const auto* audio_description = content.media_description();
   ASSERT_NE(audio_description, nullptr);
   const auto& extensions = audio_description->rtp_header_extensions();
   ASSERT_EQ(1u, extensions.size());