Deprecate the static RtpRtcp::Create() method.

The method is being used externally to create instances
of the deprecated internal implementation.

Instead, I'm moving how we instantiate the internal implementation into
the implementation itself and move towards keeping the interface
separate from a single implementation.

Change-Id: I743aa86dc4c812b545699c546c253c104719260e
Bug: webrtc:11581
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176404
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31420}

20 files changed, 71 insertions, 32 deletions

diff --git a/audio/channel_receive.cc b/audio/channel_receive.cc
index 66b4bb11f5..c4278444ab 100644
--- a/audio/channel_receive.cc
+++ b/audio/channel_receive.cc
@@ -33,11 +33,11 @@
 #include "modules/pacing/packet_router.h"
 #include "modules/rtp_rtcp/include/receive_statistics.h"
 #include "modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
-#include "modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "modules/rtp_rtcp/source/absolute_capture_time_receiver.h"
 #include "modules/rtp_rtcp/source/rtp_header_extensions.h"
 #include "modules/rtp_rtcp/source/rtp_packet_received.h"
 #include "modules/rtp_rtcp/source/rtp_rtcp_config.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
 #include "modules/utility/include/process_thread.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/critical_section.h"
@@ -507,7 +507,7 @@ ChannelReceive::ChannelReceive(
   if (frame_transformer)
     InitFrameTransformerDelegate(std::move(frame_transformer));
 
-  _rtpRtcpModule = RtpRtcp::Create(configuration);
+  _rtpRtcpModule = ModuleRtpRtcpImpl2::Create(configuration);
   _rtpRtcpModule->SetSendingMediaStatus(false);
   _rtpRtcpModule->SetRemoteSSRC(remote_ssrc_);
 
diff --git a/audio/channel_send.cc b/audio/channel_send.cc
index 3387f271ba..1c18a8b9b7 100644
--- a/audio/channel_send.cc
+++ b/audio/channel_send.cc
@@ -29,6 +29,7 @@
 #include "modules/audio_coding/include/audio_coding_module.h"
 #include "modules/audio_processing/rms_level.h"
 #include "modules/pacing/packet_router.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
 #include "modules/utility/include/process_thread.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/event.h"
@@ -530,7 +531,7 @@ ChannelSend::ChannelSend(
 
   configuration.local_media_ssrc = ssrc;
 
-  _rtpRtcpModule = RtpRtcp::Create(configuration);
+  _rtpRtcpModule = ModuleRtpRtcpImpl2::Create(configuration);
   _rtpRtcpModule->SetSendingMediaStatus(false);
 
   rtp_sender_audio_ = std::make_unique<RTPSenderAudio>(
diff --git a/audio/voip/audio_channel.cc b/audio/voip/audio_channel.cc
index b9ce7accd1..455c43c48b 100644
--- a/audio/voip/audio_channel.cc
+++ b/audio/voip/audio_channel.cc
@@ -16,6 +16,7 @@
 #include "api/audio_codecs/audio_format.h"
 #include "api/task_queue/task_queue_factory.h"
 #include "modules/rtp_rtcp/include/receive_statistics.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
 #include "rtc_base/critical_section.h"
 #include "rtc_base/location.h"
 #include "rtc_base/logging.h"
@@ -51,7 +52,7 @@ AudioChannel::AudioChannel(
   rtp_config.outgoing_transport = transport;
   rtp_config.local_media_ssrc = local_ssrc;
 
-  rtp_rtcp_ = RtpRtcp::Create(rtp_config);
+  rtp_rtcp_ = ModuleRtpRtcpImpl2::Create(rtp_config);
 
   rtp_rtcp_->SetSendingMediaStatus(false);
   rtp_rtcp_->SetRTCPStatus(RtcpMode::kCompound);
diff --git a/audio/voip/test/BUILD.gn b/audio/voip/test/BUILD.gn
index 39f100a3aa..d698b3321d 100644
--- a/audio/voip/test/BUILD.gn
+++ b/audio/voip/test/BUILD.gn
@@ -36,6 +36,7 @@ if (rtc_include_tests) {
       "../../../api/task_queue:default_task_queue_factory",
       "../../../modules/audio_mixer:audio_mixer_impl",
       "../../../modules/audio_mixer:audio_mixer_test_utils",
+      "../../../modules/rtp_rtcp:rtp_rtcp",
       "../../../modules/rtp_rtcp:rtp_rtcp_format",
       "../../../modules/utility",
       "../../../rtc_base:logging",
@@ -56,6 +57,7 @@ if (rtc_include_tests) {
       "../../../api/audio_codecs:builtin_audio_encoder_factory",
       "../../../api/task_queue:default_task_queue_factory",
       "../../../modules/audio_mixer:audio_mixer_test_utils",
+      "../../../modules/rtp_rtcp:rtp_rtcp",
       "../../../rtc_base:logging",
       "../../../rtc_base:rtc_event",
       "../../../test:mock_transport",
@@ -72,6 +74,7 @@ if (rtc_include_tests) {
       "../../../api/audio_codecs:builtin_audio_encoder_factory",
       "../../../api/task_queue:default_task_queue_factory",
       "../../../modules/audio_mixer:audio_mixer_test_utils",
+      "../../../modules/rtp_rtcp:rtp_rtcp",
       "../../../modules/rtp_rtcp:rtp_rtcp_format",
       "../../../rtc_base:logging",
       "../../../rtc_base:rtc_event",
diff --git a/audio/voip/test/audio_egress_unittest.cc b/audio/voip/test/audio_egress_unittest.cc
index 3391265880..ebb1772b30 100644
--- a/audio/voip/test/audio_egress_unittest.cc
+++ b/audio/voip/test/audio_egress_unittest.cc
@@ -14,6 +14,7 @@
 #include "api/task_queue/default_task_queue_factory.h"
 #include "modules/audio_mixer/sine_wave_generator.h"
 #include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
 #include "rtc_base/event.h"
 #include "rtc_base/logging.h"
 #include "test/gmock.h"
@@ -36,7 +37,7 @@ std::unique_ptr<RtpRtcp> CreateRtpStack(Clock* clock,
   rtp_config.rtcp_report_interval_ms = 5000;
   rtp_config.outgoing_transport = transport;
   rtp_config.local_media_ssrc = remote_ssrc;
-  auto rtp_rtcp = RtpRtcp::Create(rtp_config);
+  auto rtp_rtcp = ModuleRtpRtcpImpl2::Create(rtp_config);
   rtp_rtcp->SetSendingMediaStatus(false);
   rtp_rtcp->SetRTCPStatus(RtcpMode::kCompound);
   return rtp_rtcp;
diff --git a/audio/voip/test/audio_ingress_unittest.cc b/audio/voip/test/audio_ingress_unittest.cc
index bedb82e211..91d114c52d 100644
--- a/audio/voip/test/audio_ingress_unittest.cc
+++ b/audio/voip/test/audio_ingress_unittest.cc
@@ -15,6 +15,7 @@
 #include "api/task_queue/default_task_queue_factory.h"
 #include "audio/voip/audio_egress.h"
 #include "modules/audio_mixer/sine_wave_generator.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
 #include "rtc_base/event.h"
 #include "rtc_base/logging.h"
 #include "test/gmock.h"
@@ -45,7 +46,7 @@ class AudioIngressTest : public ::testing::Test {
     rtp_config.rtcp_report_interval_ms = 5000;
     rtp_config.outgoing_transport = &transport_;
     rtp_config.local_media_ssrc = 0xdeadc0de;
-    rtp_rtcp_ = RtpRtcp::Create(rtp_config);
+    rtp_rtcp_ = ModuleRtpRtcpImpl2::Create(rtp_config);
 
     rtp_rtcp_->SetSendingMediaStatus(false);
     rtp_rtcp_->SetRTCPStatus(RtcpMode::kCompound);
diff --git a/call/flexfec_receive_stream_impl.cc b/call/flexfec_receive_stream_impl.cc
index 40005efe83..2689195fbd 100644
--- a/call/flexfec_receive_stream_impl.cc
+++ b/call/flexfec_receive_stream_impl.cc
@@ -22,8 +22,8 @@
 #include "call/rtp_stream_receiver_controller_interface.h"
 #include "modules/rtp_rtcp/include/flexfec_receiver.h"
 #include "modules/rtp_rtcp/include/receive_statistics.h"
-#include "modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
 #include "modules/utility/include/process_thread.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/location.h"
@@ -132,7 +132,7 @@ std::unique_ptr<RtpRtcp> CreateRtpRtcpModule(
   configuration.outgoing_transport = config.rtcp_send_transport;
   configuration.rtt_stats = rtt_stats;
   configuration.local_media_ssrc = config.local_ssrc;
-  return RtpRtcp::Create(configuration);
+  return ModuleRtpRtcpImpl2::Create(configuration);
 }
 
 }  // namespace
diff --git a/call/rtp_video_sender.cc b/call/rtp_video_sender.cc
index fd35876daa..e7dbb20888 100644
--- a/call/rtp_video_sender.cc
+++ b/call/rtp_video_sender.cc
@@ -22,8 +22,8 @@
 #include "api/video_codecs/video_codec.h"
 #include "call/rtp_transport_controller_send_interface.h"
 #include "modules/pacing/packet_router.h"
-#include "modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
 #include "modules/rtp_rtcp/source/rtp_sender.h"
 #include "modules/utility/include/process_thread.h"
 #include "modules/video_coding/include/video_codec_interface.h"
@@ -253,7 +253,7 @@ std::vector<RtpStreamSender> CreateRtpStreamSenders(
 
     configuration.need_rtp_packet_infos = rtp_config.lntf.enabled;
 
-    auto rtp_rtcp = RtpRtcp::Create(configuration);
+    auto rtp_rtcp = ModuleRtpRtcpImpl2::Create(configuration);
     rtp_rtcp->SetSendingStatus(false);
     rtp_rtcp->SetSendingMediaStatus(false);
     rtp_rtcp->SetRTCPStatus(RtcpMode::kCompound);
diff --git a/modules/rtp_rtcp/include/rtp_rtcp.h b/modules/rtp_rtcp/include/rtp_rtcp.h
index 2db523caaf..be36ec8a3d 100644
--- a/modules/rtp_rtcp/include/rtp_rtcp.h
+++ b/modules/rtp_rtcp/include/rtp_rtcp.h
@@ -50,6 +50,7 @@ namespace rtcp {
 class TransportFeedback;
 }
 
+// TODO(tommi): See if we can remove Module.
 class RtpRtcp : public Module, public RtcpFeedbackSenderInterface {
  public:
   struct Configuration {
@@ -158,8 +159,15 @@ class RtpRtcp : public Module, public RtcpFeedbackSenderInterface {
     RTC_DISALLOW_COPY_AND_ASSIGN(Configuration);
   };
 
-  // Creates an RTP/RTCP module object using provided |configuration|.
-  static std::unique_ptr<RtpRtcp> Create(const Configuration& configuration);
+  // DEPRECATED. Do not use. Currently instantiates a deprecated version of the
+  // RtpRtcp module.
+  static std::unique_ptr<RtpRtcp> RTC_DEPRECATED
+  Create(const Configuration& configuration) {
+    return DEPRECATED_Create(configuration);
+  }
+
+  static std::unique_ptr<RtpRtcp> DEPRECATED_Create(
+      const Configuration& configuration);
 
   // **************************************************************************
   // Receiver functions
diff --git a/modules/rtp_rtcp/source/nack_rtx_unittest.cc b/modules/rtp_rtcp/source/nack_rtx_unittest.cc
index 55e1e44ebe..0bfd18eaf6 100644
--- a/modules/rtp_rtcp/source/nack_rtx_unittest.cc
+++ b/modules/rtp_rtcp/source/nack_rtx_unittest.cc
@@ -19,9 +19,9 @@
 #include "call/rtp_stream_receiver_controller.h"
 #include "call/rtx_receive_stream.h"
 #include "modules/rtp_rtcp/include/receive_statistics.h"
-#include "modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
 #include "modules/rtp_rtcp/source/rtp_sender_video.h"
 #include "rtc_base/rate_limiter.h"
 #include "test/gtest.h"
@@ -134,7 +134,7 @@ class RtpRtcpRtxNackTest : public ::testing::Test {
     configuration.retransmission_rate_limiter = &retransmission_rate_limiter_;
     configuration.local_media_ssrc = kTestSsrc;
     configuration.rtx_send_ssrc = kTestRtxSsrc;
-    rtp_rtcp_module_ = RtpRtcp::Create(configuration);
+    rtp_rtcp_module_ = ModuleRtpRtcpImpl2::Create(configuration);
     FieldTrialBasedConfig field_trials;
     RTPSenderVideo::Config video_config;
     video_config.clock = &fake_clock;
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index 0bd37ebdd7..795ac29cc2 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -48,6 +48,14 @@ ModuleRtpRtcpImpl::RtpSenderContext::RtpSenderContext(
           &packet_history,
           config.paced_sender ? config.paced_sender : &non_paced_sender) {}
 
+std::unique_ptr<RtpRtcp> RtpRtcp::DEPRECATED_Create(
+    const Configuration& configuration) {
+  RTC_DCHECK(configuration.clock);
+  RTC_LOG(LS_ERROR)
+      << "*********** USING WebRTC INTERNAL IMPLEMENTATION DETAILS ***********";
+  return std::make_unique<ModuleRtpRtcpImpl>(configuration);
+}
+
 ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration)
     : rtcp_sender_(configuration),
       rtcp_receiver_(configuration, this),
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc b/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc
index c8f10ac481..76335f7430 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc
@@ -48,11 +48,6 @@ ModuleRtpRtcpImpl2::RtpSenderContext::RtpSenderContext(
           &packet_history,
           config.paced_sender ? config.paced_sender : &non_paced_sender) {}
 
-std::unique_ptr<RtpRtcp> RtpRtcp::Create(const Configuration& configuration) {
-  RTC_DCHECK(configuration.clock);
-  return std::make_unique<ModuleRtpRtcpImpl2>(configuration);
-}
-
 ModuleRtpRtcpImpl2::ModuleRtpRtcpImpl2(const Configuration& configuration)
     : rtcp_sender_(configuration),
       rtcp_receiver_(configuration, this),
@@ -86,6 +81,14 @@ ModuleRtpRtcpImpl2::~ModuleRtpRtcpImpl2() {
   RTC_DCHECK_RUN_ON(&construction_thread_checker_);
 }
 
+// static
+std::unique_ptr<RtpRtcp> ModuleRtpRtcpImpl2::Create(
+    const Configuration& configuration) {
+  RTC_DCHECK(configuration.clock);
+  RTC_DCHECK(TaskQueueBase::Current());
+  return std::make_unique<ModuleRtpRtcpImpl2>(configuration);
+}
+
 // Returns the number of milliseconds until the module want a worker thread
 // to call Process.
 int64_t ModuleRtpRtcpImpl2::TimeUntilNextProcess() {
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl2.h b/modules/rtp_rtcp/source/rtp_rtcp_impl2.h
index 67409c059f..87a8107156 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl2.h
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl2.h
@@ -24,13 +24,13 @@
 #include "api/video/video_bitrate_allocation.h"
 #include "modules/include/module_fec_types.h"
 #include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
-#include "modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"  // RTCPPacketType
 #include "modules/rtp_rtcp/source/rtcp_packet/tmmb_item.h"
 #include "modules/rtp_rtcp/source/rtcp_receiver.h"
 #include "modules/rtp_rtcp/source/rtcp_sender.h"
 #include "modules/rtp_rtcp/source/rtp_packet_history.h"
 #include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
 #include "modules/rtp_rtcp/source/rtp_sender.h"
 #include "modules/rtp_rtcp/source/rtp_sender_egress.h"
 #include "rtc_base/critical_section.h"
@@ -49,6 +49,14 @@ class ModuleRtpRtcpImpl2 final : public RtpRtcp,
   explicit ModuleRtpRtcpImpl2(const RtpRtcp::Configuration& configuration);
   ~ModuleRtpRtcpImpl2() override;
 
+  // This method is provided to easy with migrating away from the
+  // RtpRtcp::Create factory method. Since this is an internal implementation
+  // detail though, creating an instance of ModuleRtpRtcpImpl2 directly should
+  // be fine.
+  static std::unique_ptr<RtpRtcp> Create(const Configuration& configuration);
+
+  // TODO(tommi): Make implementation private?
+
   // Returns the number of milliseconds until the module want a worker thread to
   // call Process.
   int64_t TimeUntilNextProcess() override;
diff --git a/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc b/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc
index 3e35f42bff..e406b53c0c 100644
--- a/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc
@@ -18,6 +18,7 @@
 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "modules/rtp_rtcp/source/rtp_header_extensions.h"
 #include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
 #include "modules/rtp_rtcp/source/time_util.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
@@ -67,7 +68,7 @@ class RtpSenderAudioTest : public ::testing::Test {
  public:
   RtpSenderAudioTest()
       : fake_clock_(kStartTime),
-        rtp_module_(RtpRtcp::Create([&] {
+        rtp_module_(ModuleRtpRtcpImpl2::Create([&] {
           RtpRtcp::Configuration config;
           config.audio = true;
           config.clock = &fake_clock_;
diff --git a/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc b/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc
index 80481dc2e5..2dbb2e723e 100644
--- a/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc
@@ -24,7 +24,6 @@
 #include "common_video/generic_frame_descriptor/generic_frame_info.h"
 #include "modules/rtp_rtcp/include/rtp_cvo.h"
 #include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
-#include "modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "modules/rtp_rtcp/source/rtp_dependency_descriptor_extension.h"
 #include "modules/rtp_rtcp/source/rtp_descriptor_authentication.h"
@@ -33,6 +32,7 @@
 #include "modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension.h"
 #include "modules/rtp_rtcp/source/rtp_header_extensions.h"
 #include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
 #include "modules/rtp_rtcp/source/time_util.h"
 #include "rtc_base/arraysize.h"
 #include "rtc_base/rate_limiter.h"
@@ -169,7 +169,7 @@ class RtpSenderVideoTest : public ::testing::TestWithParam<bool> {
       : field_trials_(GetParam()),
         fake_clock_(kStartTime),
         retransmission_rate_limiter_(&fake_clock_, 1000),
-        rtp_module_(RtpRtcp::Create([&] {
+        rtp_module_(ModuleRtpRtcpImpl2::Create([&] {
           RtpRtcp::Configuration config;
           config.clock = &fake_clock_;
           config.outgoing_transport = &transport_;
@@ -920,7 +920,7 @@ class RtpSenderVideoWithFrameTransformerTest : public ::testing::Test {
   RtpSenderVideoWithFrameTransformerTest()
       : fake_clock_(kStartTime),
         retransmission_rate_limiter_(&fake_clock_, 1000),
-        rtp_module_(RtpRtcp::Create([&] {
+        rtp_module_(ModuleRtpRtcpImpl2::Create([&] {
           RtpRtcp::Configuration config;
           config.clock = &fake_clock_;
           config.outgoing_transport = &transport_;
diff --git a/video/end_to_end_tests/bandwidth_tests.cc b/video/end_to_end_tests/bandwidth_tests.cc
index 6e8e11d76f..d0c2c27f48 100644
--- a/video/end_to_end_tests/bandwidth_tests.cc
+++ b/video/end_to_end_tests/bandwidth_tests.cc
@@ -16,7 +16,7 @@
 #include "api/video/video_bitrate_allocation.h"
 #include "call/fake_network_pipe.h"
 #include "call/simulated_network.h"
-#include "modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
 #include "rtc_base/rate_limiter.h"
 #include "rtc_base/task_queue_for_test.h"
 #include "rtc_base/task_utils/to_queued_task.h"
@@ -244,7 +244,7 @@ TEST_F(BandwidthEndToEndTest, RembWithSendSideBwe) {
       config.outgoing_transport = receive_transport_;
       config.retransmission_rate_limiter = &retransmission_rate_limiter_;
       config.local_media_ssrc = (*receive_configs)[0].rtp.local_ssrc;
-      rtp_rtcp_ = RtpRtcp::Create(config);
+      rtp_rtcp_ = ModuleRtpRtcpImpl2::Create(config);
       rtp_rtcp_->SetRemoteSSRC((*receive_configs)[0].rtp.remote_ssrc);
       rtp_rtcp_->SetRTCPStatus(RtcpMode::kReducedSize);
     }
diff --git a/video/rtp_video_stream_receiver.cc b/video/rtp_video_stream_receiver.cc
index ad8b0383c8..a4c102e21f 100644
--- a/video/rtp_video_stream_receiver.cc
+++ b/video/rtp_video_stream_receiver.cc
@@ -25,7 +25,6 @@
 #include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
 #include "modules/rtp_rtcp/include/receive_statistics.h"
 #include "modules/rtp_rtcp/include/rtp_cvo.h"
-#include "modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "modules/rtp_rtcp/include/ulpfec_receiver.h"
 #include "modules/rtp_rtcp/source/create_video_rtp_depacketizer.h"
 #include "modules/rtp_rtcp/source/rtp_dependency_descriptor_extension.h"
@@ -35,6 +34,7 @@
 #include "modules/rtp_rtcp/source/rtp_header_extensions.h"
 #include "modules/rtp_rtcp/source/rtp_packet_received.h"
 #include "modules/rtp_rtcp/source/rtp_rtcp_config.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
 #include "modules/rtp_rtcp/source/video_rtp_depacketizer.h"
 #include "modules/rtp_rtcp/source/video_rtp_depacketizer_raw.h"
 #include "modules/utility/include/process_thread.h"
@@ -97,7 +97,7 @@ std::unique_ptr<RtpRtcp> CreateRtpRtcpModule(
   configuration.rtcp_cname_callback = rtcp_cname_callback;
   configuration.local_media_ssrc = local_ssrc;
 
-  std::unique_ptr<RtpRtcp> rtp_rtcp = RtpRtcp::Create(configuration);
+  std::unique_ptr<RtpRtcp> rtp_rtcp = RtpRtcp::DEPRECATED_Create(configuration);
   rtp_rtcp->SetRTCPStatus(RtcpMode::kCompound);
 
   return rtp_rtcp;
diff --git a/video/rtp_video_stream_receiver2.cc b/video/rtp_video_stream_receiver2.cc
index 2c7bd4bb4e..54a25d22fe 100644
--- a/video/rtp_video_stream_receiver2.cc
+++ b/video/rtp_video_stream_receiver2.cc
@@ -25,7 +25,6 @@
 #include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
 #include "modules/rtp_rtcp/include/receive_statistics.h"
 #include "modules/rtp_rtcp/include/rtp_cvo.h"
-#include "modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "modules/rtp_rtcp/include/ulpfec_receiver.h"
 #include "modules/rtp_rtcp/source/create_video_rtp_depacketizer.h"
 #include "modules/rtp_rtcp/source/rtp_dependency_descriptor_extension.h"
@@ -35,6 +34,7 @@
 #include "modules/rtp_rtcp/source/rtp_header_extensions.h"
 #include "modules/rtp_rtcp/source/rtp_packet_received.h"
 #include "modules/rtp_rtcp/source/rtp_rtcp_config.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
 #include "modules/rtp_rtcp/source/video_rtp_depacketizer.h"
 #include "modules/rtp_rtcp/source/video_rtp_depacketizer_raw.h"
 #include "modules/utility/include/process_thread.h"
@@ -97,7 +97,7 @@ std::unique_ptr<RtpRtcp> CreateRtpRtcpModule(
   configuration.rtcp_cname_callback = rtcp_cname_callback;
   configuration.local_media_ssrc = local_ssrc;
 
-  std::unique_ptr<RtpRtcp> rtp_rtcp = RtpRtcp::Create(configuration);
+  std::unique_ptr<RtpRtcp> rtp_rtcp = ModuleRtpRtcpImpl2::Create(configuration);
   rtp_rtcp->SetRTCPStatus(RtcpMode::kCompound);
 
   return rtp_rtcp;
diff --git a/video/rtp_video_stream_receiver2_unittest.cc b/video/rtp_video_stream_receiver2_unittest.cc
index 57fba8f9cf..22ca595605 100644
--- a/video/rtp_video_stream_receiver2_unittest.cc
+++ b/video/rtp_video_stream_receiver2_unittest.cc
@@ -37,6 +37,7 @@
 #include "test/gmock.h"
 #include "test/gtest.h"
 #include "test/mock_frame_transformer.h"
+#include "test/time_controller/simulated_task_queue.h"
 
 using ::testing::_;
 using ::testing::ElementsAre;
@@ -237,6 +238,9 @@ class RtpVideoStreamReceiver2Test : public ::testing::Test {
     return config;
   }
 
+  TokenTaskQueue task_queue_;
+  TokenTaskQueue::CurrentTaskQueueSetter task_queue_setter_{&task_queue_};
+
   const webrtc::test::ScopedFieldTrials override_field_trials_;
   VideoReceiveStream::Config config_;
   MockNackSender mock_nack_sender_;
diff --git a/video/video_send_stream_tests.cc b/video/video_send_stream_tests.cc
index 449e194ed2..cb77f13884 100644
--- a/video/video_send_stream_tests.cc
+++ b/video/video_send_stream_tests.cc
@@ -25,10 +25,10 @@
 #include "call/simulated_network.h"
 #include "call/video_send_stream.h"
 #include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
-#include "modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "modules/rtp_rtcp/source/rtcp_sender.h"
 #include "modules/rtp_rtcp/source/rtp_header_extensions.h"
 #include "modules/rtp_rtcp/source/rtp_packet.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
 #include "modules/rtp_rtcp/source/video_rtp_depacketizer_vp9.h"
 #include "modules/video_coding/codecs/vp8/include/vp8.h"
 #include "modules/video_coding/codecs/vp9/include/vp9.h"
@@ -1677,7 +1677,7 @@ TEST_F(VideoSendStreamTest, MinTransmitBitrateRespectsRemb) {
       config.clock = Clock::GetRealTimeClock();
       config.outgoing_transport = feedback_transport_.get();
       config.retransmission_rate_limiter = &retranmission_rate_limiter_;
-      rtp_rtcp_ = RtpRtcp::Create(config);
+      rtp_rtcp_ = ModuleRtpRtcpImpl2::Create(config);
       rtp_rtcp_->SetRTCPStatus(RtcpMode::kReducedSize);
     }