diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index 4207f7bb98..3589d6a890 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -17,6 +17,7 @@
 #include <string>
 #include <utility>
 
+#include "absl/memory/memory.h"
 #include "modules/rtp_rtcp/source/rtcp_packet/dlrr.h"
 #include "modules/rtp_rtcp/source/rtp_rtcp_config.h"
 #include "rtc_base/checks.h"
@@ -83,7 +84,6 @@ ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration)
                                                 : kDefaultVideoReportInterval),
                      this),
       clock_(configuration.clock),
-      audio_(configuration.audio),
       keepalive_config_(configuration.keepalive_config),
       last_bitrate_process_time_(clock_->TimeInMilliseconds()),
       last_rtt_process_time_(clock_->TimeInMilliseconds()),
@@ -101,7 +101,9 @@ ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration)
     rtp_sender_.reset(new RTPSender(
         configuration.audio, configuration.clock,
         configuration.outgoing_transport, configuration.paced_sender,
-        configuration.flexfec_sender,
+        configuration.flexfec_sender
+            ? absl::make_optional(configuration.flexfec_sender->ssrc())
+            : absl::nullopt,
         configuration.transport_sequence_number_allocator,
         configuration.transport_feedback_callback,
         configuration.send_bitrate_observer,
@@ -112,6 +114,14 @@ ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration)
         configuration.populate_network2_timestamp,
         configuration.frame_encryptor, configuration.require_frame_encryption,
         configuration.extmap_allow_mixed));
+    if (configuration.audio) {
+      audio_ = absl::make_unique<RTPSenderAudio>(clock_, rtp_sender_.get());
+    } else {
+      video_ = absl::make_unique<RTPSenderVideo>(
+          clock_, rtp_sender_.get(), configuration.flexfec_sender,
+          configuration.frame_encryptor,
+          configuration.require_frame_encryption);
+    }
     // Make sure rtcp sender use same timestamp offset as rtp sender.
     rtcp_sender_.SetTimestampOffset(rtp_sender_->TimestampOffset());
 
@@ -268,22 +278,21 @@ void ModuleRtpRtcpImpl::RegisterAudioSendPayload(int payload_type,
                                                  int frequency,
                                                  int channels,
                                                  int rate) {
+  RTC_DCHECK(audio_);
   rtcp_sender_.SetRtpClockRate(payload_type, frequency);
-  RTC_CHECK_EQ(0,
-               rtp_sender_->RegisterPayload(payload_name, payload_type,
-                                            frequency, channels, rate));
+  RTC_CHECK_EQ(0, audio_->RegisterAudioPayload(payload_name, payload_type,
+                                               frequency, channels, rate));
 }
 
 void ModuleRtpRtcpImpl::RegisterVideoSendPayload(int payload_type,
                                                  const char* payload_name) {
+  RTC_DCHECK(video_);
   rtcp_sender_.SetRtpClockRate(payload_type, kVideoPayloadTypeFrequency);
-  RTC_CHECK_EQ(0,
-               rtp_sender_->RegisterPayload(payload_name, payload_type,
-                                            kVideoPayloadTypeFrequency, 0, 0));
+  video_->RegisterPayloadType(payload_type, payload_name);
 }
 
 int32_t ModuleRtpRtcpImpl::DeRegisterSendPayload(const int8_t payload_type) {
-  return rtp_sender_->DeRegisterSendPayload(payload_type);
+  return 0;
 }
 
 uint32_t ModuleRtpRtcpImpl::StartTimestamp() const {
@@ -446,10 +455,22 @@ bool ModuleRtpRtcpImpl::SendOutgoingData(
       expected_retransmission_time_ms = kDefaultExpectedRetransmissionTimeMs;
     }
   }
-  return rtp_sender_->SendOutgoingData(
-      frame_type, payload_type, time_stamp, capture_time_ms, payload_data,
-      payload_size, fragmentation, rtp_video_header, transport_frame_id_out,
-      expected_retransmission_time_ms);
+
+  const uint32_t rtp_timestamp = time_stamp + rtp_sender_->TimestampOffset();
+  if (transport_frame_id_out)
+    *transport_frame_id_out = rtp_timestamp;
+
+  if (audio_) {
+    RTC_DCHECK(fragmentation == nullptr);
+
+    return audio_->SendAudio(frame_type, payload_type, rtp_timestamp,
+                             payload_data, payload_size);
+  } else {
+    return video_->SendVideo(frame_type, payload_type, rtp_timestamp,
+                             capture_time_ms, payload_data, payload_size,
+                             fragmentation, rtp_video_header,
+                             expected_retransmission_time_ms);
+  }
 }
 
 bool ModuleRtpRtcpImpl::TimeToSendPacket(uint32_t ssrc,
@@ -764,11 +785,11 @@ bool ModuleRtpRtcpImpl::SendFeedbackPacket(
 int32_t ModuleRtpRtcpImpl::SendTelephoneEventOutband(const uint8_t key,
                                                      const uint16_t time_ms,
                                                      const uint8_t level) {
-  return rtp_sender_->SendTelephoneEvent(key, time_ms, level);
+  return audio_ ? audio_->SendTelephoneEvent(key, time_ms, level) : -1;
 }
 
 int32_t ModuleRtpRtcpImpl::SetAudioLevel(const uint8_t level_d_bov) {
-  return rtp_sender_->SetAudioLevel(level_d_bov);
+  return audio_ ? audio_->SetAudioLevel(level_d_bov) : -1;
 }
 
 int32_t ModuleRtpRtcpImpl::SetKeyFrameRequestMethod(
@@ -789,13 +810,18 @@ int32_t ModuleRtpRtcpImpl::RequestKeyFrame() {
 
 void ModuleRtpRtcpImpl::SetUlpfecConfig(int red_payload_type,
                                         int ulpfec_payload_type) {
-  rtp_sender_->SetUlpfecConfig(red_payload_type, ulpfec_payload_type);
+  RTC_DCHECK(video_);
+  video_->SetUlpfecConfig(red_payload_type, ulpfec_payload_type);
 }
 
 bool ModuleRtpRtcpImpl::SetFecParameters(
     const FecProtectionParams& delta_params,
     const FecProtectionParams& key_params) {
-  return rtp_sender_->SetFecParameters(delta_params, key_params);
+  if (!video_) {
+    return false;
+  }
+  video_->SetFecParameters(delta_params, key_params);
+  return true;
 }
 
 void ModuleRtpRtcpImpl::SetRemoteSSRC(const uint32_t ssrc) {
@@ -809,13 +835,13 @@ void ModuleRtpRtcpImpl::BitrateSent(uint32_t* total_rate,
                                     uint32_t* fec_rate,
                                     uint32_t* nack_rate) const {
   *total_rate = rtp_sender_->BitrateSent();
-  *video_rate = rtp_sender_->VideoBitrateSent();
-  *fec_rate = rtp_sender_->FecOverheadRate();
+  *video_rate = video_ ? video_->VideoBitrateSent() : 0;
+  *fec_rate = video_ ? video_->FecOverheadRate() : 0;
   *nack_rate = rtp_sender_->NackOverheadRate();
 }
 
 uint32_t ModuleRtpRtcpImpl::PacketizationOverheadBps() const {
-  return rtp_sender_->PacketizationOverheadBps();
+  return video_ ? video_->PacketizationOverheadBps() : 0;
 }
 
 void ModuleRtpRtcpImpl::OnRequestSendReport() {
@@ -843,8 +869,15 @@ void ModuleRtpRtcpImpl::OnReceivedNack(
 
 void ModuleRtpRtcpImpl::OnReceivedRtcpReportBlocks(
     const ReportBlockList& report_blocks) {
-  if (rtp_sender_)
-    rtp_sender_->OnReceivedRtcpReportBlocks(report_blocks);
+  if (video_) {
+    uint32_t ssrc = SSRC();
+
+    for (const RTCPReportBlock& report_block : report_blocks) {
+      if (ssrc == report_block.source_ssrc) {
+        video_->OnReceivedAck(report_block.extended_highest_sequence_number);
+      }
+    }
+  }
 }
 
 bool ModuleRtpRtcpImpl::LastReceivedNTP(
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl.h b/modules/rtp_rtcp/source/rtp_rtcp_impl.h
index 61adb314b5..114e897c88 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl.h
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl.h
@@ -32,6 +32,8 @@
 #include "modules/rtp_rtcp/source/rtcp_receiver.h"
 #include "modules/rtp_rtcp/source/rtcp_sender.h"
 #include "modules/rtp_rtcp/source/rtp_sender.h"
+#include "modules/rtp_rtcp/source/rtp_sender_audio.h"
+#include "modules/rtp_rtcp/source/rtp_sender_video.h"
 #include "rtc_base/critical_section.h"
 #include "rtc_base/gtest_prod_util.h"
 
@@ -336,13 +338,13 @@ class ModuleRtpRtcpImpl : public RtpRtcp, public RTCPReceiver::ModuleRtpRtcp {
   bool TimeToSendFullNackList(int64_t now) const;
 
   std::unique_ptr<RTPSender> rtp_sender_;
+  std::unique_ptr<RTPSenderAudio> audio_;
+  std::unique_ptr<RTPSenderVideo> video_;
   RTCPSender rtcp_sender_;
   RTCPReceiver rtcp_receiver_;
 
   Clock* const clock_;
 
-  const bool audio_;
-
   const RtpKeepAliveConfig keepalive_config_;
   int64_t last_bitrate_process_time_;
   int64_t last_rtt_process_time_;
diff --git a/modules/rtp_rtcp/source/rtp_sender.cc b/modules/rtp_rtcp/source/rtp_sender.cc
index 7f42a62e96..ad7e63157b 100644
--- a/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/modules/rtp_rtcp/source/rtp_sender.cc
@@ -20,14 +20,11 @@
 #include "api/array_view.h"
 #include "logging/rtc_event_log/events/rtc_event_rtp_packet_outgoing.h"
 #include "logging/rtc_event_log/rtc_event_log.h"
-#include "modules/remote_bitrate_estimator/test/bwe_test_logging.h"
 #include "modules/rtp_rtcp/include/rtp_cvo.h"
 #include "modules/rtp_rtcp/source/byte_io.h"
 #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_to_send.h"
-#include "modules/rtp_rtcp/source/rtp_sender_audio.h"
-#include "modules/rtp_rtcp/source/rtp_sender_video.h"
 #include "modules/rtp_rtcp/source/time_util.h"
 #include "rtc_base/arraysize.h"
 #include "rtc_base/checks.h"
@@ -35,7 +32,6 @@
 #include "rtc_base/numerics/safe_minmax.h"
 #include "rtc_base/rate_limiter.h"
 #include "rtc_base/time_utils.h"
-#include "rtc_base/trace_event.h"
 #include "system_wrappers/include/field_trial.h"
 
 namespace webrtc {
@@ -87,21 +83,6 @@ constexpr RtpExtensionSize kVideoExtensionSizes[] = {
      RtpGenericFrameDescriptorExtension::kMaxSizeBytes},
 };
 
-const char* FrameTypeToString(FrameType frame_type) {
-  switch (frame_type) {
-    case kEmptyFrame:
-      return "empty";
-    case kAudioFrameSpeech:
-      return "audio_speech";
-    case kAudioFrameCN:
-      return "audio_cn";
-    case kVideoFrameKey:
-      return "video_key";
-    case kVideoFrameDelta:
-      return "video_delta";
-  }
-  return "";
-}
 }  // namespace
 
 RTPSender::RTPSender(
@@ -109,7 +90,7 @@ RTPSender::RTPSender(
     Clock* clock,
     Transport* transport,
     RtpPacketSender* paced_sender,
-    FlexfecSender* flexfec_sender,
+    absl::optional<uint32_t> flexfec_ssrc,
     TransportSequenceNumberAllocator* sequence_number_allocator,
     TransportFeedbackObserver* transport_feedback_observer,
     BitrateStatisticsObserver* bitrate_callback,
@@ -127,13 +108,7 @@ RTPSender::RTPSender(
       clock_delta_ms_(clock_->TimeInMilliseconds() - rtc::TimeMillis()),
       random_(clock_->TimeInMicroseconds()),
       audio_configured_(audio),
-      audio_(audio ? new RTPSenderAudio(clock, this) : nullptr),
-      video_(audio ? nullptr
-                   : new RTPSenderVideo(clock,
-                                        this,
-                                        flexfec_sender,
-                                        frame_encryptor,
-                                        require_frame_encryption)),
+      flexfec_ssrc_(flexfec_ssrc),
       paced_sender_(paced_sender),
       transport_sequence_number_allocator_(sequence_number_allocator),
       transport_feedback_observer_(transport_feedback_observer),
@@ -180,7 +155,7 @@ RTPSender::RTPSender(
 
   // Store FlexFEC packets in the packet history data structure, so they can
   // be found when paced.
-  if (flexfec_sender) {
+  if (flexfec_ssrc_) {
     flexfec_packet_history_.SetStorePacketsStatus(
         RtpPacketHistory::StorageMode::kStore,
         kMinFlexfecPacketsToStoreForPacing);
@@ -216,29 +191,11 @@ uint16_t RTPSender::ActualSendBitrateKbit() const {
       1000);
 }
 
-uint32_t RTPSender::VideoBitrateSent() const {
-  if (video_) {
-    return video_->VideoBitrateSent();
-  }
-  return 0;
-}
-
-uint32_t RTPSender::FecOverheadRate() const {
-  if (video_) {
-    return video_->FecOverheadRate();
-  }
-  return 0;
-}
-
 uint32_t RTPSender::NackOverheadRate() const {
   rtc::CritScope cs(&statistics_crit_);
   return nack_bitrate_sent_.Rate(clock_->TimeInMilliseconds()).value_or(0);
 }
 
-uint32_t RTPSender::PacketizationOverheadBps() const {
-  return video_ ? video_->PacketizationOverheadBps() : 0;
-}
-
 void RTPSender::SetExtmapAllowMixed(bool extmap_allow_mixed) {
   rtc::CritScope lock(&send_critsect_);
   rtp_header_extension_map_.SetExtmapAllowMixed(extmap_allow_mixed);
@@ -265,29 +222,6 @@ int32_t RTPSender::DeregisterRtpHeaderExtension(RTPExtensionType type) {
   return rtp_header_extension_map_.Deregister(type);
 }
 
-int32_t RTPSender::RegisterPayload(absl::string_view payload_name,
-                                   int8_t payload_number,
-                                   uint32_t frequency,
-                                   size_t channels,
-                                   uint32_t rate) {
-  rtc::CritScope lock(&send_critsect_);
-
-  int32_t ret_val = 0;
-  if (audio_configured_) {
-    // TODO(mflodman): Change to CreateAudioPayload and make static.
-    ret_val = audio_->RegisterAudioPayload(payload_name, payload_number,
-                                           frequency, channels, rate);
-  } else {
-    video_->RegisterPayloadType(payload_number, payload_name);
-  }
-
-  return ret_val;
-}
-
-int32_t RTPSender::DeRegisterSendPayload(int8_t /* payload_type */) {
-  return 0;
-}
-
 void RTPSender::SetMaxRtpPacketSize(size_t max_packet_size) {
   RTC_DCHECK_GE(max_packet_size, 100);
   RTC_DCHECK_LE(max_packet_size, IP_PACKET_SIZE);
@@ -333,67 +267,6 @@ void RTPSender::SetRtxPayloadType(int payload_type,
   rtx_payload_type_map_[associated_payload_type] = payload_type;
 }
 
-bool RTPSender::SendOutgoingData(FrameType frame_type,
-                                 int8_t payload_type,
-                                 uint32_t capture_timestamp,
-                                 int64_t capture_time_ms,
-                                 const uint8_t* payload_data,
-                                 size_t payload_size,
-                                 const RTPFragmentationHeader* fragmentation,
-                                 const RTPVideoHeader* rtp_header,
-                                 uint32_t* transport_frame_id_out,
-                                 int64_t expected_retransmission_time_ms) {
-  uint16_t sequence_number;
-  uint32_t rtp_timestamp;
-  {
-    // Drop this packet if we're not sending media packets.
-    rtc::CritScope lock(&send_critsect_);
-    RTC_DCHECK(ssrc_);
-
-    sequence_number = sequence_number_;
-    rtp_timestamp = timestamp_offset_ + capture_timestamp;
-    if (transport_frame_id_out)
-      *transport_frame_id_out = rtp_timestamp;
-    if (!sending_media_)
-      return true;
-  }
-  switch (frame_type) {
-    case kAudioFrameSpeech:
-    case kAudioFrameCN:
-      RTC_CHECK(audio_configured_);
-      break;
-    case kVideoFrameKey:
-    case kVideoFrameDelta:
-      RTC_CHECK(!audio_configured_);
-      break;
-    case kEmptyFrame:
-      break;
-  }
-
-  bool result;
-  if (audio_configured_) {
-    TRACE_EVENT_ASYNC_STEP1("webrtc", "Audio", rtp_timestamp, "Send", "type",
-                            FrameTypeToString(frame_type));
-    // The only known way to produce of RTPFragmentationHeader for audio is
-    // to use the AudioCodingModule directly.
-    RTC_DCHECK(fragmentation == nullptr);
-    result = audio_->SendAudio(frame_type, payload_type, rtp_timestamp,
-                               payload_data, payload_size);
-  } else {
-    TRACE_EVENT_ASYNC_STEP1("webrtc", "Video", capture_time_ms, "Send", "type",
-                            FrameTypeToString(frame_type));
-    if (frame_type == kEmptyFrame)
-      return true;
-
-    result = video_->SendVideo(frame_type, payload_type, rtp_timestamp,
-                               capture_time_ms, payload_data, payload_size,
-                               fragmentation, rtp_header,
-                               expected_retransmission_time_ms);
-  }
-
-  return result;
-}
-
 size_t RTPSender::TrySendRedundantPayloads(size_t bytes_to_send,
                                            const PacedPacketInfo& pacing_info) {
   {
@@ -643,26 +516,6 @@ void RTPSender::OnReceivedNack(
   }
 }
 
-void RTPSender::OnReceivedRtcpReportBlocks(
-    const ReportBlockList& report_blocks) {
-  if (!video_) {
-    return;
-  }
-  uint32_t ssrc;
-  {
-    rtc::CritScope lock(&send_critsect_);
-    if (!ssrc_)
-      return;
-    ssrc = *ssrc_;
-  }
-
-  for (const RTCPReportBlock& report_block : report_blocks) {
-    if (ssrc == report_block.source_ssrc) {
-      video_->OnReceivedAck(report_block.extended_highest_sequence_number);
-    }
-  }
-}
-
 // Called from pacer when we can send the packet.
 bool RTPSender::TimeToSendPacket(uint32_t ssrc,
                                  uint16_t sequence_number,
@@ -809,29 +662,14 @@ bool RTPSender::SendToNetwork(std::unique_ptr<RtpPacketToSend> packet,
   RTC_DCHECK(packet);
   int64_t now_ms = clock_->TimeInMilliseconds();
 
-  if (video_) {
-    BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "VideoTotBitrate_kbps", now_ms,
-                                    ActualSendBitrateKbit(), packet->Ssrc());
-    BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "VideoFecBitrate_kbps", now_ms,
-                                    FecOverheadRate() / 1000, packet->Ssrc());
-    BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "VideoNackBitrate_kbps", now_ms,
-                                    NackOverheadRate() / 1000, packet->Ssrc());
-  } else {
-    BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "AudioTotBitrate_kbps", now_ms,
-                                    ActualSendBitrateKbit(), packet->Ssrc());
-    BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "AudioNackBitrate_kbps", now_ms,
-                                    NackOverheadRate() / 1000, packet->Ssrc());
-  }
-
   uint32_t ssrc = packet->Ssrc();
-  absl::optional<uint32_t> flexfec_ssrc = FlexfecSsrc();
   if (paced_sender_) {
     uint16_t seq_no = packet->SequenceNumber();
     // Correct offset between implementations of millisecond time stamps in
     // TickTime and Clock.
     int64_t corrected_time_ms = packet->capture_time_ms() + clock_delta_ms_;
     size_t payload_length = packet->payload_size();
-    if (ssrc == flexfec_ssrc) {
+    if (ssrc == FlexfecSsrc()) {
       // Store FlexFEC packets in the history here, so they can be found
       // when the pacer calls TimeToSendPacket.
       flexfec_packet_history_.PutRtpPacket(std::move(packet), storage,
@@ -1164,10 +1002,7 @@ void RTPSender::SetMid(const std::string& mid) {
 }
 
 absl::optional<uint32_t> RTPSender::FlexfecSsrc() const {
-  if (video_) {
-    return video_->FlexfecSsrc();
-  }
-  return absl::nullopt;
+  return flexfec_ssrc_;
 }
 
 void RTPSender::SetCsrcs(const std::vector<uint32_t>& csrcs) {
@@ -1187,34 +1022,6 @@ uint16_t RTPSender::SequenceNumber() const {
   return sequence_number_;
 }
 
-// Audio.
-int32_t RTPSender::SendTelephoneEvent(uint8_t key,
-                                      uint16_t time_ms,
-                                      uint8_t level) {
-  if (!audio_configured_) {
-    return -1;
-  }
-  return audio_->SendTelephoneEvent(key, time_ms, level);
-}
-
-int32_t RTPSender::SetAudioLevel(uint8_t level_d_bov) {
-  return audio_->SetAudioLevel(level_d_bov);
-}
-
-void RTPSender::SetUlpfecConfig(int red_payload_type, int ulpfec_payload_type) {
-  RTC_DCHECK(!audio_configured_);
-  video_->SetUlpfecConfig(red_payload_type, ulpfec_payload_type);
-}
-
-bool RTPSender::SetFecParameters(const FecProtectionParams& delta_params,
-                                 const FecProtectionParams& key_params) {
-  if (audio_configured_) {
-    return false;
-  }
-  video_->SetFecParameters(delta_params, key_params);
-  return true;
-}
-
 static std::unique_ptr<RtpPacketToSend> CreateRtxPacket(
     const RtpPacketToSend& packet,
     RtpHeaderExtensionMap* extension_map) {
diff --git a/modules/rtp_rtcp/source/rtp_sender.h b/modules/rtp_rtcp/source/rtp_sender.h
index 401a8ff067..622f41a08d 100644
--- a/modules/rtp_rtcp/source/rtp_sender.h
+++ b/modules/rtp_rtcp/source/rtp_sender.h
@@ -41,8 +41,6 @@ class OverheadObserver;
 class RateLimiter;
 class RtcEventLog;
 class RtpPacketToSend;
-class RTPSenderAudio;
-class RTPSenderVideo;
 
 class RTPSender {
  public:
@@ -50,9 +48,7 @@ class RTPSender {
             Clock* clock,
             Transport* transport,
             RtpPacketSender* paced_sender,
-            // TODO(brandtr): Remove |flexfec_sender| when that is hooked up
-            // to PacedSender instead.
-            FlexfecSender* flexfec_sender,
+            absl::optional<uint32_t> flexfec_ssrc,
             TransportSequenceNumberAllocator* sequence_number_allocator,
             TransportFeedbackObserver* transport_feedback_callback,
             BitrateStatisticsObserver* bitrate_callback,
@@ -72,18 +68,7 @@ class RTPSender {
 
   uint16_t ActualSendBitrateKbit() const;
 
-  uint32_t VideoBitrateSent() const;
-  uint32_t FecOverheadRate() const;
   uint32_t NackOverheadRate() const;
-  uint32_t PacketizationOverheadBps() const;
-
-  int32_t RegisterPayload(absl::string_view payload_name,
-                          const int8_t payload_type,
-                          const uint32_t frequency,
-                          const size_t channels,
-                          const uint32_t rate);
-
-  int32_t DeRegisterSendPayload(const int8_t payload_type);
 
   void SetSendingMediaStatus(bool enabled);
   bool SendingMedia() const;
@@ -109,17 +94,6 @@ class RTPSender {
 
   void SetMaxRtpPacketSize(size_t max_packet_size);
 
-  bool SendOutgoingData(FrameType frame_type,
-                        int8_t payload_type,
-                        uint32_t timestamp,
-                        int64_t capture_time_ms,
-                        const uint8_t* payload_data,
-                        size_t payload_size,
-                        const RTPFragmentationHeader* fragmentation,
-                        const RTPVideoHeader* rtp_header,
-                        uint32_t* transport_frame_id_out,
-                        int64_t expected_retransmission_time_ms);
-
   void SetExtmapAllowMixed(bool extmap_allow_mixed);
 
   // RTP header extension
@@ -145,10 +119,6 @@ class RTPSender {
 
   int32_t ReSendPacket(uint16_t packet_id);
 
-  // Feedback to decide when to stop sending the playout delay and MID header
-  // extensions.
-  void OnReceivedRtcpReportBlocks(const ReportBlockList& report_blocks);
-
   // RTX.
   void SetRtxStatus(int mode);
   int RtxStatus() const;
@@ -187,21 +157,6 @@ class RTPSender {
                      StorageType storage,
                      RtpPacketSender::Priority priority);
 
-  // Audio.
-
-  // Send a DTMF tone using RFC 2833 (4733).
-  int32_t SendTelephoneEvent(uint8_t key, uint16_t time_ms, uint8_t level);
-
-  // Store the audio level in d_bov for
-  // header-extension-for-audio-level-indication.
-  int32_t SetAudioLevel(uint8_t level_d_bov);
-
-  // ULPFEC.
-  void SetUlpfecConfig(int red_payload_type, int ulpfec_payload_type);
-
-  bool SetFecParameters(const FecProtectionParams& delta_params,
-                        const FecProtectionParams& key_params);
-
   // Called on update of RTP statistics.
   void RegisterRtpStatisticsCallback(StreamDataCountersCallback* callback);
   StreamDataCountersCallback* GetRtpStatisticsCallback() const;
@@ -269,8 +224,8 @@ class RTPSender {
   Random random_ RTC_GUARDED_BY(send_critsect_);
 
   const bool audio_configured_;
-  const std::unique_ptr<RTPSenderAudio> audio_;
-  const std::unique_ptr<RTPSenderVideo> video_;
+
+  const absl::optional<uint32_t> flexfec_ssrc_;
 
   RtpPacketSender* const paced_sender_;
   TransportSequenceNumberAllocator* const transport_sequence_number_allocator_;
diff --git a/modules/rtp_rtcp/source/rtp_sender_audio.cc b/modules/rtp_rtcp/source/rtp_sender_audio.cc
index 9acc098555..56d0884e75 100644
--- a/modules/rtp_rtcp/source/rtp_sender_audio.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_audio.cc
@@ -16,6 +16,7 @@
 
 #include "absl/strings/match.h"
 #include "api/audio_codecs/audio_format.h"
+#include "modules/remote_bitrate_estimator/test/bwe_test_logging.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "modules/rtp_rtcp/source/byte_io.h"
 #include "modules/rtp_rtcp/source/rtp_header_extensions.h"
@@ -27,6 +28,24 @@
 
 namespace webrtc {
 
+namespace {
+
+const char* FrameTypeToString(FrameType frame_type) {
+  switch (frame_type) {
+    case kEmptyFrame:
+      return "empty";
+    case kAudioFrameSpeech:
+      return "audio_speech";
+    case kAudioFrameCN:
+      return "audio_cn";
+    default:
+      RTC_NOTREACHED();
+      return "";
+  }
+}
+
+}  // namespace
+
 RTPSenderAudio::RTPSenderAudio(Clock* clock, RTPSender* rtp_sender)
     : clock_(clock), rtp_sender_(rtp_sender) {}
 
@@ -115,6 +134,12 @@ bool RTPSenderAudio::SendAudio(FrameType frame_type,
                                uint32_t rtp_timestamp,
                                const uint8_t* payload_data,
                                size_t payload_size) {
+  RTC_DCHECK(frame_type == kAudioFrameSpeech || frame_type == kAudioFrameCN ||
+             frame_type == kEmptyFrame);
+
+  TRACE_EVENT_ASYNC_STEP1("webrtc", "Audio", rtp_timestamp, "Send", "type",
+                          FrameTypeToString(frame_type));
+
   // From RFC 4733:
   // A source has wide latitude as to how often it sends event updates. A
   // natural interval is the spacing between non-event audio packets. [...]
@@ -233,7 +258,7 @@ bool RTPSenderAudio::SendAudio(FrameType frame_type,
   TRACE_EVENT_ASYNC_END2("webrtc", "Audio", rtp_timestamp, "timestamp",
                          packet->Timestamp(), "seqnum",
                          packet->SequenceNumber());
-  bool send_result = rtp_sender_->SendToNetwork(
+  bool send_result = LogAndSendToNetwork(
       std::move(packet), kAllowRetransmission, RtpPacketSender::kHighPriority);
   if (first_packet_sent_()) {
     RTC_LOG(LS_INFO) << "First audio RTP packet sent to pacer";
@@ -317,11 +342,29 @@ bool RTPSenderAudio::SendTelephoneEventPacket(bool ended,
     dtmfbuffer[1] = E | R | volume;
     ByteWriter<uint16_t>::WriteBigEndian(dtmfbuffer + 2, duration);
 
-    result = rtp_sender_->SendToNetwork(std::move(packet), kAllowRetransmission,
-                                        RtpPacketSender::kHighPriority);
+    result = LogAndSendToNetwork(std::move(packet), kAllowRetransmission,
+                                 RtpPacketSender::kHighPriority);
     send_count--;
   } while (send_count > 0 && result);
 
   return result;
 }
+
+bool RTPSenderAudio::LogAndSendToNetwork(
+    std::unique_ptr<RtpPacketToSend> packet,
+    StorageType storage,
+    RtpPacketSender::Priority priority) {
+#if BWE_TEST_LOGGING_COMPILE_TIME_ENABLE
+  int64_t now_ms = clock_->TimeInMilliseconds();
+  BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "AudioTotBitrate_kbps", now_ms,
+                                  rtp_sender_->ActualSendBitrateKbit(),
+                                  packet->Ssrc());
+  BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "AudioNackBitrate_kbps", now_ms,
+                                  rtp_sender_->NackOverheadRate() / 1000,
+                                  packet->Ssrc());
+#endif
+
+  return rtp_sender_->SendToNetwork(std::move(packet), storage, priority);
+}
+
 }  // namespace webrtc
diff --git a/modules/rtp_rtcp/source/rtp_sender_audio.h b/modules/rtp_rtcp/source/rtp_sender_audio.h
index f002023e9a..fa5894367d 100644
--- a/modules/rtp_rtcp/source/rtp_sender_audio.h
+++ b/modules/rtp_rtcp/source/rtp_sender_audio.h
@@ -14,6 +14,8 @@
 #include <stddef.h>
 #include <stdint.h>
 
+#include <memory>
+
 #include "absl/strings/string_view.h"
 #include "common_types.h"  // NOLINT(build/include)
 #include "modules/rtp_rtcp/source/dtmf_queue.h"
@@ -61,6 +63,10 @@ class RTPSenderAudio {
   bool MarkerBit(FrameType frame_type, int8_t payload_type);
 
  private:
+  bool LogAndSendToNetwork(std::unique_ptr<RtpPacketToSend> packet,
+                           StorageType storage,
+                           RtpPacketSender::Priority priority);
+
   Clock* const clock_ = nullptr;
   RTPSender* const rtp_sender_ = nullptr;
 
diff --git a/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc b/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc
index b1c46c1cea..7f9d72f40a 100644
--- a/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_audio_unittest.cc
@@ -64,7 +64,7 @@ class RtpSenderAudioTest : public ::testing::Test {
                     &fake_clock_,
                     &transport_,
                     nullptr,
-                    nullptr,
+                    absl::nullopt,
                     nullptr,
                     nullptr,
                     nullptr,
diff --git a/modules/rtp_rtcp/source/rtp_sender_unittest.cc b/modules/rtp_rtcp/source/rtp_sender_unittest.cc
index b13875acc0..714fac708a 100644
--- a/modules/rtp_rtcp/source/rtp_sender_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_unittest.cc
@@ -184,7 +184,7 @@ class RtpSenderTest : public ::testing::TestWithParam<bool> {
   void SetUpRtpSender(bool pacer, bool populate_network2) {
     rtp_sender_.reset(new RTPSender(
         false, &fake_clock_, &transport_, pacer ? &mock_paced_sender_ : nullptr,
-        nullptr, &seq_num_allocator_, nullptr, nullptr, nullptr,
+        absl::nullopt, &seq_num_allocator_, nullptr, nullptr, nullptr,
         &mock_rtc_event_log_, &send_packet_observer_,
         &retransmission_rate_limiter_, nullptr, populate_network2, nullptr,
         false, false));
@@ -324,9 +324,10 @@ TEST_P(RtpSenderTest, AssignSequenceNumberAllowsPaddingOnAudio) {
   MockTransport transport;
   const bool kEnableAudio = true;
   rtp_sender_.reset(new RTPSender(
-      kEnableAudio, &fake_clock_, &transport, &mock_paced_sender_, nullptr,
-      nullptr, nullptr, nullptr, nullptr, &mock_rtc_event_log_, nullptr,
-      &retransmission_rate_limiter_, nullptr, false, nullptr, false, false));
+      kEnableAudio, &fake_clock_, &transport, &mock_paced_sender_,
+      absl::nullopt, nullptr, nullptr, nullptr, nullptr, &mock_rtc_event_log_,
+      nullptr, &retransmission_rate_limiter_, nullptr, false, nullptr, false,
+      false));
   rtp_sender_->SetTimestampOffset(0);
   rtp_sender_->SetSSRC(kSsrc);
 
@@ -370,10 +371,10 @@ TEST_P(RtpSenderTestWithoutPacer,
   constexpr int kRtpOverheadBytesPerPacket = 12 + 8;
   testing::NiceMock<MockOverheadObserver> mock_overhead_observer;
   rtp_sender_.reset(new RTPSender(
-      false, &fake_clock_, &transport_, nullptr, nullptr, &seq_num_allocator_,
-      &feedback_observer_, nullptr, nullptr, &mock_rtc_event_log_, nullptr,
-      &retransmission_rate_limiter_, &mock_overhead_observer, false, nullptr,
-      false, false));
+      false, &fake_clock_, &transport_, nullptr, absl::nullopt,
+      &seq_num_allocator_, &feedback_observer_, nullptr, nullptr,
+      &mock_rtc_event_log_, nullptr, &retransmission_rate_limiter_,
+      &mock_overhead_observer, false, nullptr, false, false));
   rtp_sender_->SetSSRC(kSsrc);
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
                    kRtpExtensionTransportSequenceNumber,
@@ -397,10 +398,10 @@ TEST_P(RtpSenderTestWithoutPacer,
 
 TEST_P(RtpSenderTestWithoutPacer, SendsPacketsWithTransportSequenceNumber) {
   rtp_sender_.reset(new RTPSender(
-      false, &fake_clock_, &transport_, nullptr, nullptr, &seq_num_allocator_,
-      &feedback_observer_, nullptr, nullptr, &mock_rtc_event_log_,
-      &send_packet_observer_, &retransmission_rate_limiter_, nullptr, false,
-      nullptr, false, false));
+      false, &fake_clock_, &transport_, nullptr, absl::nullopt,
+      &seq_num_allocator_, &feedback_observer_, nullptr, nullptr,
+      &mock_rtc_event_log_, &send_packet_observer_,
+      &retransmission_rate_limiter_, nullptr, false, nullptr, false, false));
   rtp_sender_->SetSSRC(kSsrc);
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
                    kRtpExtensionTransportSequenceNumber,
@@ -429,10 +430,10 @@ TEST_P(RtpSenderTestWithoutPacer, SendsPacketsWithTransportSequenceNumber) {
 
 TEST_P(RtpSenderTestWithoutPacer, PacketOptionsNoRetransmission) {
   rtp_sender_.reset(new RTPSender(
-      false, &fake_clock_, &transport_, nullptr, nullptr, &seq_num_allocator_,
-      &feedback_observer_, nullptr, nullptr, &mock_rtc_event_log_,
-      &send_packet_observer_, &retransmission_rate_limiter_, nullptr, false,
-      nullptr, false, false));
+      false, &fake_clock_, &transport_, nullptr, absl::nullopt,
+      &seq_num_allocator_, &feedback_observer_, nullptr, nullptr,
+      &mock_rtc_event_log_, &send_packet_observer_,
+      &retransmission_rate_limiter_, nullptr, false, nullptr, false, false));
   rtp_sender_->SetSSRC(kSsrc);
 
   SendGenericPacket();
@@ -484,18 +485,20 @@ TEST_P(RtpSenderTestWithoutPacer, DoesnSetIncludedInAllocationByDefault) {
 TEST_P(RtpSenderTestWithoutPacer, OnSendSideDelayUpdated) {
   testing::StrictMock<MockSendSideDelayObserver> send_side_delay_observer_;
   rtp_sender_.reset(new RTPSender(
-      false, &fake_clock_, &transport_, nullptr, nullptr, nullptr, nullptr,
-      nullptr, &send_side_delay_observer_, &mock_rtc_event_log_, nullptr,
-      nullptr, nullptr, false, nullptr, false, false));
+      false, &fake_clock_, &transport_, nullptr, absl::nullopt, nullptr,
+      nullptr, nullptr, &send_side_delay_observer_, &mock_rtc_event_log_,
+      nullptr, nullptr, nullptr, false, nullptr, false, false));
   rtp_sender_->SetSSRC(kSsrc);
+  RTPSenderVideo rtp_sender_video(&fake_clock_, rtp_sender_.get(), nullptr,
+                                  nullptr, false);
 
   const uint8_t kPayloadType = 127;
-  const uint32_t kCaptureTimeMsToRtpTimestamp = 90;  // 90 kHz clock
   const char payload_name[] = "GENERIC";
+
+  rtp_sender_video.RegisterPayloadType(kPayloadType, payload_name);
+
+  const uint32_t kCaptureTimeMsToRtpTimestamp = 90;  // 90 kHz clock
   RTPVideoHeader video_header;
-  EXPECT_EQ(0, rtp_sender_->RegisterPayload(payload_name, kPayloadType,
-                                            1000 * kCaptureTimeMsToRtpTimestamp,
-                                            0, 1500));
 
   // Send packet with 10 ms send-side delay. The average and max should be 10
   // ms.
@@ -503,10 +506,10 @@ TEST_P(RtpSenderTestWithoutPacer, OnSendSideDelayUpdated) {
       .Times(1);
   int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
   fake_clock_.AdvanceTimeMilliseconds(10);
-  EXPECT_TRUE(rtp_sender_->SendOutgoingData(
+  EXPECT_TRUE(rtp_sender_video.SendVideo(
       kVideoFrameKey, kPayloadType,
       capture_time_ms * kCaptureTimeMsToRtpTimestamp, capture_time_ms,
-      kPayloadData, sizeof(kPayloadData), nullptr, &video_header, nullptr,
+      kPayloadData, sizeof(kPayloadData), nullptr, &video_header,
       kDefaultExpectedRetransmissionTimeMs));
 
   // Send another packet with 20 ms delay. The average
@@ -514,10 +517,10 @@ TEST_P(RtpSenderTestWithoutPacer, OnSendSideDelayUpdated) {
   EXPECT_CALL(send_side_delay_observer_, SendSideDelayUpdated(15, 20, kSsrc))
       .Times(1);
   fake_clock_.AdvanceTimeMilliseconds(10);
-  EXPECT_TRUE(rtp_sender_->SendOutgoingData(
+  EXPECT_TRUE(rtp_sender_video.SendVideo(
       kVideoFrameKey, kPayloadType,
       capture_time_ms * kCaptureTimeMsToRtpTimestamp, capture_time_ms,
-      kPayloadData, sizeof(kPayloadData), nullptr, &video_header, nullptr,
+      kPayloadData, sizeof(kPayloadData), nullptr, &video_header,
       kDefaultExpectedRetransmissionTimeMs));
 
   // Send another packet at the same time, which replaces the last packet.
@@ -526,10 +529,10 @@ TEST_P(RtpSenderTestWithoutPacer, OnSendSideDelayUpdated) {
   EXPECT_CALL(send_side_delay_observer_, SendSideDelayUpdated(5, 10, kSsrc))
       .Times(1);
   capture_time_ms = fake_clock_.TimeInMilliseconds();
-  EXPECT_TRUE(rtp_sender_->SendOutgoingData(
+  EXPECT_TRUE(rtp_sender_video.SendVideo(
       kVideoFrameKey, kPayloadType,
       capture_time_ms * kCaptureTimeMsToRtpTimestamp, capture_time_ms,
-      kPayloadData, sizeof(kPayloadData), nullptr, &video_header, nullptr,
+      kPayloadData, sizeof(kPayloadData), nullptr, &video_header,
       kDefaultExpectedRetransmissionTimeMs));
 
   // Send a packet 1 second later. The earlier packets should have timed
@@ -539,10 +542,10 @@ TEST_P(RtpSenderTestWithoutPacer, OnSendSideDelayUpdated) {
   fake_clock_.AdvanceTimeMilliseconds(1);
   EXPECT_CALL(send_side_delay_observer_, SendSideDelayUpdated(1, 1, kSsrc))
       .Times(1);
-  EXPECT_TRUE(rtp_sender_->SendOutgoingData(
+  EXPECT_TRUE(rtp_sender_video.SendVideo(
       kVideoFrameKey, kPayloadType,
       capture_time_ms * kCaptureTimeMsToRtpTimestamp, capture_time_ms,
-      kPayloadData, sizeof(kPayloadData), nullptr, &video_header, nullptr,
+      kPayloadData, sizeof(kPayloadData), nullptr, &video_header,
       kDefaultExpectedRetransmissionTimeMs));
 }
 
@@ -561,7 +564,7 @@ TEST_P(RtpSenderTestWithoutPacer, OnSendPacketUpdated) {
 
 TEST_P(RtpSenderTest, SendsPacketsWithTransportSequenceNumber) {
   rtp_sender_.reset(new RTPSender(
-      false, &fake_clock_, &transport_, &mock_paced_sender_, nullptr,
+      false, &fake_clock_, &transport_, &mock_paced_sender_, absl::nullopt,
       &seq_num_allocator_, &feedback_observer_, nullptr, nullptr,
       &mock_rtc_event_log_, &send_packet_observer_,
       &retransmission_rate_limiter_, nullptr, false, nullptr, false, false));
@@ -946,7 +949,7 @@ TEST_P(RtpSenderTest, OnSendPacketNotUpdatedForRetransmits) {
 
 TEST_P(RtpSenderTest, OnSendPacketNotUpdatedWithoutSeqNumAllocator) {
   rtp_sender_.reset(new RTPSender(
-      false, &fake_clock_, &transport_, &mock_paced_sender_, nullptr,
+      false, &fake_clock_, &transport_, &mock_paced_sender_, absl::nullopt,
       nullptr /* TransportSequenceNumberAllocator */, nullptr, nullptr, nullptr,
       nullptr, &send_packet_observer_, &retransmission_rate_limiter_, nullptr,
       false, nullptr, false, false));
@@ -973,8 +976,8 @@ TEST_P(RtpSenderTest, OnSendPacketNotUpdatedWithoutSeqNumAllocator) {
 TEST_P(RtpSenderTest, SendRedundantPayloads) {
   MockTransport transport;
   rtp_sender_.reset(new RTPSender(
-      false, &fake_clock_, &transport, &mock_paced_sender_, nullptr, nullptr,
-      nullptr, nullptr, nullptr, &mock_rtc_event_log_, nullptr,
+      false, &fake_clock_, &transport, &mock_paced_sender_, absl::nullopt,
+      nullptr, nullptr, nullptr, nullptr, &mock_rtc_event_log_, nullptr,
       &retransmission_rate_limiter_, nullptr, false, nullptr, false, false));
   rtp_sender_->SetSequenceNumber(kSeqNum);
   rtp_sender_->SetSSRC(kSsrc);
@@ -1050,15 +1053,16 @@ TEST_P(RtpSenderTest, SendRedundantPayloads) {
 TEST_P(RtpSenderTestWithoutPacer, SendGenericVideo) {
   const char payload_name[] = "GENERIC";
   const uint8_t payload_type = 127;
-  ASSERT_EQ(0, rtp_sender_->RegisterPayload(payload_name, payload_type, 90000,
-                                            0, 1500));
+  RTPSenderVideo rtp_sender_video(&fake_clock_, rtp_sender_.get(), nullptr,
+                                  nullptr, false);
+  rtp_sender_video.RegisterPayloadType(payload_type, payload_name);
   uint8_t payload[] = {47, 11, 32, 93, 89};
 
   // Send keyframe
   RTPVideoHeader video_header;
-  ASSERT_TRUE(rtp_sender_->SendOutgoingData(
+  ASSERT_TRUE(rtp_sender_video.SendVideo(
       kVideoFrameKey, payload_type, 1234, 4321, payload, sizeof(payload),
-      nullptr, &video_header, nullptr, kDefaultExpectedRetransmissionTimeMs));
+      nullptr, &video_header, kDefaultExpectedRetransmissionTimeMs));
 
   auto sent_payload = transport_.last_sent_packet().payload();
   uint8_t generic_header = sent_payload[0];
@@ -1071,9 +1075,9 @@ TEST_P(RtpSenderTestWithoutPacer, SendGenericVideo) {
   payload[1] = 42;
   payload[4] = 13;
 
-  ASSERT_TRUE(rtp_sender_->SendOutgoingData(
+  ASSERT_TRUE(rtp_sender_video.SendVideo(
       kVideoFrameDelta, payload_type, 1234, 4321, payload, sizeof(payload),
-      nullptr, &video_header, nullptr, kDefaultExpectedRetransmissionTimeMs));
+      nullptr, &video_header, kDefaultExpectedRetransmissionTimeMs));
 
   sent_payload = transport_.last_sent_packet().payload();
   generic_header = sent_payload[0];
@@ -1096,7 +1100,7 @@ TEST_P(RtpSenderTest, SendFlexfecPackets) {
 
   // Reset |rtp_sender_| to use FlexFEC.
   rtp_sender_.reset(new RTPSender(
-      false, &fake_clock_, &transport_, &mock_paced_sender_, &flexfec_sender,
+      false, &fake_clock_, &transport_, &mock_paced_sender_, kFlexfecSsrc,
       &seq_num_allocator_, nullptr, nullptr, nullptr, &mock_rtc_event_log_,
       &send_packet_observer_, &retransmission_rate_limiter_, nullptr, false,
       nullptr, false, false));
@@ -1167,10 +1171,10 @@ TEST_P(RtpSenderTest, NoFlexfecForTimingFrames) {
 
   // Reset |rtp_sender_| to use FlexFEC.
   rtp_sender_.reset(new RTPSender(
-      false, &fake_clock_, &transport_, &mock_paced_sender_, &flexfec_sender,
-      &seq_num_allocator_, nullptr, nullptr, nullptr, &mock_rtc_event_log_,
-      &send_packet_observer_, &retransmission_rate_limiter_, nullptr, false,
-      nullptr, false, false));
+      false, &fake_clock_, &transport_, &mock_paced_sender_,
+      flexfec_sender.ssrc(), &seq_num_allocator_, nullptr, nullptr, nullptr,
+      &mock_rtc_event_log_, &send_packet_observer_,
+      &retransmission_rate_limiter_, nullptr, false, nullptr, false, false));
   rtp_sender_->SetSSRC(kMediaSsrc);
   rtp_sender_->SetSequenceNumber(kSeqNum);
   rtp_sender_->SetStorePacketsStatus(true, 10);
@@ -1264,7 +1268,7 @@ TEST_P(RtpSenderTestWithoutPacer, SendFlexfecPackets) {
 
   // Reset |rtp_sender_| to use FlexFEC.
   rtp_sender_.reset(new RTPSender(
-      false, &fake_clock_, &transport_, nullptr, &flexfec_sender,
+      false, &fake_clock_, &transport_, nullptr, flexfec_sender.ssrc(),
       &seq_num_allocator_, nullptr, nullptr, nullptr, &mock_rtc_event_log_,
       &send_packet_observer_, &retransmission_rate_limiter_, nullptr, false,
       nullptr, false, false));
@@ -1391,10 +1395,10 @@ TEST_P(RtpSenderTest, FecOverheadRate) {
 
   // Reset |rtp_sender_| to use FlexFEC.
   rtp_sender_.reset(new RTPSender(
-      false, &fake_clock_, &transport_, &mock_paced_sender_, &flexfec_sender,
-      &seq_num_allocator_, nullptr, nullptr, nullptr, &mock_rtc_event_log_,
-      &send_packet_observer_, &retransmission_rate_limiter_, nullptr, false,
-      nullptr, false, false));
+      false, &fake_clock_, &transport_, &mock_paced_sender_,
+      flexfec_sender.ssrc(), &seq_num_allocator_, nullptr, nullptr, nullptr,
+      &mock_rtc_event_log_, &send_packet_observer_,
+      &retransmission_rate_limiter_, nullptr, false, nullptr, false, false));
   rtp_sender_->SetSSRC(kMediaSsrc);
   rtp_sender_->SetSequenceNumber(kSeqNum);
 
@@ -1460,11 +1464,17 @@ TEST_P(RtpSenderTest, BitrateCallbacks) {
     uint32_t retransmit_bitrate_;
   } callback;
   rtp_sender_.reset(new RTPSender(
-      false, &fake_clock_, &transport_, nullptr, nullptr, nullptr, nullptr,
-      &callback, nullptr, nullptr, nullptr, &retransmission_rate_limiter_,
-      nullptr, false, nullptr, false, false));
+      false, &fake_clock_, &transport_, nullptr, absl::nullopt, nullptr,
+      nullptr, &callback, nullptr, nullptr, nullptr,
+      &retransmission_rate_limiter_, nullptr, false, nullptr, false, false));
   rtp_sender_->SetSSRC(kSsrc);
 
+  RTPSenderVideo rtp_sender_video(&fake_clock_, rtp_sender_.get(), nullptr,
+                                  nullptr, false);
+  const char payload_name[] = "GENERIC";
+  const uint8_t payload_type = 127;
+  rtp_sender_video.RegisterPayloadType(payload_type, payload_name);
+
   // Simulate kNumPackets sent with kPacketInterval ms intervals, with the
   // number of packets selected so that we fill (but don't overflow) the one
   // second averaging window.
@@ -1475,10 +1485,6 @@ TEST_P(RtpSenderTest, BitrateCallbacks) {
   // Overhead = 12 bytes RTP header + 1 byte generic header.
   const uint32_t kPacketOverhead = 13;
 
-  const char payload_name[] = "GENERIC";
-  const uint8_t payload_type = 127;
-  ASSERT_EQ(0, rtp_sender_->RegisterPayload(payload_name, payload_type, 90000,
-                                            0, 1500));
   uint8_t payload[] = {47, 11, 32, 93, 89};
   rtp_sender_->SetStorePacketsStatus(true, 1);
   uint32_t ssrc = rtp_sender_->SSRC();
@@ -1489,9 +1495,9 @@ TEST_P(RtpSenderTest, BitrateCallbacks) {
   // Send a few frames.
   RTPVideoHeader video_header;
   for (uint32_t i = 0; i < kNumPackets; ++i) {
-    ASSERT_TRUE(rtp_sender_->SendOutgoingData(
+    ASSERT_TRUE(rtp_sender_video.SendVideo(
         kVideoFrameKey, payload_type, 1234, 4321, payload, sizeof(payload),
-        nullptr, &video_header, nullptr, kDefaultExpectedRetransmissionTimeMs));
+        nullptr, &video_header, kDefaultExpectedRetransmissionTimeMs));
     fake_clock_.AdvanceTimeMilliseconds(kPacketInterval);
   }
 
@@ -1548,8 +1554,9 @@ TEST_P(RtpSenderTestWithoutPacer, StreamDataCountersCallbacks) {
   const uint8_t kUlpfecPayloadType = 97;
   const char payload_name[] = "GENERIC";
   const uint8_t payload_type = 127;
-  ASSERT_EQ(0, rtp_sender_->RegisterPayload(payload_name, payload_type, 90000,
-                                            0, 1500));
+  RTPSenderVideo rtp_sender_video(&fake_clock_, rtp_sender_.get(), nullptr,
+                                  nullptr, false);
+  rtp_sender_video.RegisterPayloadType(payload_type, payload_name);
   uint8_t payload[] = {47, 11, 32, 93, 89};
   rtp_sender_->SetStorePacketsStatus(true, 1);
   uint32_t ssrc = rtp_sender_->SSRC();
@@ -1558,9 +1565,9 @@ TEST_P(RtpSenderTestWithoutPacer, StreamDataCountersCallbacks) {
 
   // Send a frame.
   RTPVideoHeader video_header;
-  ASSERT_TRUE(rtp_sender_->SendOutgoingData(
+  ASSERT_TRUE(rtp_sender_video.SendVideo(
       kVideoFrameKey, payload_type, 1234, 4321, payload, sizeof(payload),
-      nullptr, &video_header, nullptr, kDefaultExpectedRetransmissionTimeMs));
+      nullptr, &video_header, kDefaultExpectedRetransmissionTimeMs));
   StreamDataCounters expected;
   expected.transmitted.payload_bytes = 6;
   expected.transmitted.header_bytes = 12;
@@ -1594,15 +1601,15 @@ TEST_P(RtpSenderTestWithoutPacer, StreamDataCountersCallbacks) {
   callback.Matches(ssrc, expected);
 
   // Send ULPFEC.
-  rtp_sender_->SetUlpfecConfig(kRedPayloadType, kUlpfecPayloadType);
+  rtp_sender_video.SetUlpfecConfig(kRedPayloadType, kUlpfecPayloadType);
   FecProtectionParams fec_params;
   fec_params.fec_mask_type = kFecMaskRandom;
   fec_params.fec_rate = 1;
   fec_params.max_fec_frames = 1;
-  rtp_sender_->SetFecParameters(fec_params, fec_params);
-  ASSERT_TRUE(rtp_sender_->SendOutgoingData(
+  rtp_sender_video.SetFecParameters(fec_params, fec_params);
+  ASSERT_TRUE(rtp_sender_video.SendVideo(
       kVideoFrameDelta, payload_type, 1234, 4321, payload, sizeof(payload),
-      nullptr, &video_header, nullptr, kDefaultExpectedRetransmissionTimeMs));
+      nullptr, &video_header, kDefaultExpectedRetransmissionTimeMs));
   expected.transmitted.payload_bytes = 40;
   expected.transmitted.header_bytes = 60;
   expected.transmitted.packets = 5;
@@ -1613,22 +1620,14 @@ TEST_P(RtpSenderTestWithoutPacer, StreamDataCountersCallbacks) {
 }
 
 TEST_P(RtpSenderTestWithoutPacer, BytesReportedCorrectly) {
-  const char* kPayloadName = "GENERIC";
+  // XXX const char* kPayloadName = "GENERIC";
   const uint8_t kPayloadType = 127;
   rtp_sender_->SetSSRC(1234);
   rtp_sender_->SetRtxSsrc(4321);
   rtp_sender_->SetRtxPayloadType(kPayloadType - 1, kPayloadType);
   rtp_sender_->SetRtxStatus(kRtxRetransmitted | kRtxRedundantPayloads);
 
-  ASSERT_EQ(0, rtp_sender_->RegisterPayload(kPayloadName, kPayloadType, 90000,
-                                            0, 1500));
-  uint8_t payload[] = {47, 11, 32, 93, 89};
-
-  RTPVideoHeader video_header;
-  ASSERT_TRUE(rtp_sender_->SendOutgoingData(
-      kVideoFrameKey, kPayloadType, 1234, 4321, payload, sizeof(payload),
-      nullptr, &video_header, nullptr, kDefaultExpectedRetransmissionTimeMs));
-
+  SendGenericPacket();
   // Will send 2 full-size padding packets.
   rtp_sender_->TimeToSendPadding(1, PacedPacketInfo());
   rtp_sender_->TimeToSendPadding(1, PacedPacketInfo());
@@ -1637,9 +1636,9 @@ TEST_P(RtpSenderTestWithoutPacer, BytesReportedCorrectly) {
   StreamDataCounters rtx_stats;
   rtp_sender_->GetDataCounters(&rtp_stats, &rtx_stats);
 
-  // Payload + 1-byte generic header.
+  // Payload
   EXPECT_GT(rtp_stats.first_packet_time_ms, -1);
-  EXPECT_EQ(rtp_stats.transmitted.payload_bytes, sizeof(payload) + 1);
+  EXPECT_EQ(rtp_stats.transmitted.payload_bytes, sizeof(kPayloadData));
   EXPECT_EQ(rtp_stats.transmitted.header_bytes, 12u);
   EXPECT_EQ(rtp_stats.transmitted.padding_bytes, 0u);
   EXPECT_EQ(rtx_stats.transmitted.payload_bytes, 0u);
@@ -1694,10 +1693,11 @@ TEST_P(RtpSenderTestWithoutPacer, RespectsNackBitrateLimit) {
 
 TEST_P(RtpSenderTest, OnOverheadChanged) {
   MockOverheadObserver mock_overhead_observer;
-  rtp_sender_.reset(new RTPSender(
-      false, &fake_clock_, &transport_, nullptr, nullptr, nullptr, nullptr,
-      nullptr, nullptr, nullptr, nullptr, &retransmission_rate_limiter_,
-      &mock_overhead_observer, false, nullptr, false, false));
+  rtp_sender_.reset(
+      new RTPSender(false, &fake_clock_, &transport_, nullptr, absl::nullopt,
+                    nullptr, nullptr, nullptr, nullptr, nullptr, nullptr,
+                    &retransmission_rate_limiter_, &mock_overhead_observer,
+                    false, nullptr, false, false));
   rtp_sender_->SetSSRC(kSsrc);
 
   // RTP overhead is 12B.
@@ -1715,10 +1715,11 @@ TEST_P(RtpSenderTest, OnOverheadChanged) {
 
 TEST_P(RtpSenderTest, DoesNotUpdateOverheadOnEqualSize) {
   MockOverheadObserver mock_overhead_observer;
-  rtp_sender_.reset(new RTPSender(
-      false, &fake_clock_, &transport_, nullptr, nullptr, nullptr, nullptr,
-      nullptr, nullptr, nullptr, nullptr, &retransmission_rate_limiter_,
-      &mock_overhead_observer, false, nullptr, false, false));
+  rtp_sender_.reset(
+      new RTPSender(false, &fake_clock_, &transport_, nullptr, absl::nullopt,
+                    nullptr, nullptr, nullptr, nullptr, nullptr, nullptr,
+                    &retransmission_rate_limiter_, &mock_overhead_observer,
+                    false, nullptr, false, false));
   rtp_sender_->SetSSRC(kSsrc);
 
   EXPECT_CALL(mock_overhead_observer, OnOverheadChanged(_)).Times(1);
@@ -1729,7 +1730,7 @@ TEST_P(RtpSenderTest, DoesNotUpdateOverheadOnEqualSize) {
 TEST_P(RtpSenderTest, SendsKeepAlive) {
   MockTransport transport;
   rtp_sender_.reset(new RTPSender(
-      false, &fake_clock_, &transport, nullptr, nullptr, nullptr, nullptr,
+      false, &fake_clock_, &transport, nullptr, absl::nullopt, nullptr, nullptr,
       nullptr, nullptr, &mock_rtc_event_log_, nullptr,
       &retransmission_rate_limiter_, nullptr, false, nullptr, false, false));
   rtp_sender_->SetSequenceNumber(kSeqNum);
diff --git a/modules/rtp_rtcp/source/rtp_sender_video.cc b/modules/rtp_rtcp/source/rtp_sender_video.cc
index 949af14b7d..c63f0d7ea9 100644
--- a/modules/rtp_rtcp/source/rtp_sender_video.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_video.cc
@@ -21,6 +21,7 @@
 #include "absl/memory/memory.h"
 #include "absl/strings/match.h"
 #include "api/crypto/frame_encryptor_interface.h"
+#include "modules/remote_bitrate_estimator/test/bwe_test_logging.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "modules/rtp_rtcp/source/byte_io.h"
 #include "modules/rtp_rtcp/source/rtp_format_video_generic.h"
@@ -152,6 +153,20 @@ bool IsBaseLayer(const RTPVideoHeader& video_header) {
   return true;
 }
 
+const char* FrameTypeToString(FrameType frame_type) {
+  switch (frame_type) {
+    case kEmptyFrame:
+      return "empty";
+    case kVideoFrameKey:
+      return "video_key";
+    case kVideoFrameDelta:
+      return "video_delta";
+    default:
+      RTC_NOTREACHED();
+      return "";
+  }
+}
+
 }  // namespace
 
 RTPSenderVideo::RTPSenderVideo(Clock* clock,
@@ -207,8 +222,8 @@ void RTPSenderVideo::SendVideoPacket(std::unique_ptr<RtpPacketToSend> packet,
   // Remember some values about the packet before sending it away.
   size_t packet_size = packet->size();
   uint16_t seq_num = packet->SequenceNumber();
-  if (!rtp_sender_->SendToNetwork(std::move(packet), storage,
-                                  RtpPacketSender::kLowPriority)) {
+  if (!LogAndSendToNetwork(std::move(packet), storage,
+                           RtpPacketSender::kLowPriority)) {
     RTC_LOG(LS_WARNING) << "Failed to send video packet " << seq_num;
     return;
   }
@@ -249,8 +264,8 @@ void RTPSenderVideo::SendVideoPacketAsRedMaybeWithUlpfec(
   }
   // Send |red_packet| instead of |packet| for allocated sequence number.
   size_t red_packet_size = red_packet->size();
-  if (rtp_sender_->SendToNetwork(std::move(red_packet), media_packet_storage,
-                                 RtpPacketSender::kLowPriority)) {
+  if (LogAndSendToNetwork(std::move(red_packet), media_packet_storage,
+                          RtpPacketSender::kLowPriority)) {
     rtc::CritScope cs(&stats_crit_);
     video_bitrate_.Update(red_packet_size, clock_->TimeInMilliseconds());
   } else {
@@ -265,8 +280,8 @@ void RTPSenderVideo::SendVideoPacketAsRedMaybeWithUlpfec(
     rtp_packet->set_capture_time_ms(media_packet->capture_time_ms());
     rtp_packet->set_is_fec(true);
     uint16_t fec_sequence_number = rtp_packet->SequenceNumber();
-    if (rtp_sender_->SendToNetwork(std::move(rtp_packet), kDontRetransmit,
-                                   RtpPacketSender::kLowPriority)) {
+    if (LogAndSendToNetwork(std::move(rtp_packet), kDontRetransmit,
+                            RtpPacketSender::kLowPriority)) {
       rtc::CritScope cs(&stats_crit_);
       fec_bitrate_.Update(fec_packet->length(), clock_->TimeInMilliseconds());
     } else {
@@ -293,8 +308,8 @@ void RTPSenderVideo::SendVideoPacketWithFlexfec(
     for (auto& fec_packet : fec_packets) {
       size_t packet_length = fec_packet->size();
       uint16_t seq_num = fec_packet->SequenceNumber();
-      if (rtp_sender_->SendToNetwork(std::move(fec_packet), kDontRetransmit,
-                                     RtpPacketSender::kLowPriority)) {
+      if (LogAndSendToNetwork(std::move(fec_packet), kDontRetransmit,
+                              RtpPacketSender::kLowPriority)) {
         rtc::CritScope cs(&stats_crit_);
         fec_bitrate_.Update(packet_length, clock_->TimeInMilliseconds());
       } else {
@@ -304,6 +319,24 @@ void RTPSenderVideo::SendVideoPacketWithFlexfec(
   }
 }
 
+bool RTPSenderVideo::LogAndSendToNetwork(
+    std::unique_ptr<RtpPacketToSend> packet,
+    StorageType storage,
+    RtpPacketSender::Priority priority) {
+#if BWE_TEST_LOGGING_COMPILE_TIME_ENABLE
+  int64_t now_ms = clock_->TimeInMilliseconds();
+  BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "VideoTotBitrate_kbps", now_ms,
+                                  rtp_sender_->ActualSendBitrateKbit(),
+                                  packet->Ssrc());
+  BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "VideoFecBitrate_kbps", now_ms,
+                                  FecOverheadRate() / 1000, packet->Ssrc());
+  BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "VideoNackBitrate_kbps", now_ms,
+                                  rtp_sender_->NackOverheadRate() / 1000,
+                                  packet->Ssrc());
+#endif
+  return rtp_sender_->SendToNetwork(std::move(packet), storage, priority);
+}
+
 void RTPSenderVideo::SetUlpfecConfig(int red_payload_type,
                                      int ulpfec_payload_type) {
   // Sanity check. Per the definition of UlpfecConfig (see config.h),
@@ -371,6 +404,15 @@ bool RTPSenderVideo::SendVideo(FrameType frame_type,
                                const RTPFragmentationHeader* fragmentation,
                                const RTPVideoHeader* video_header,
                                int64_t expected_retransmission_time_ms) {
+  RTC_DCHECK(frame_type == kVideoFrameKey || frame_type == kVideoFrameDelta ||
+             frame_type == kEmptyFrame);
+
+  TRACE_EVENT_ASYNC_STEP1("webrtc", "Video", capture_time_ms, "Send", "type",
+                          FrameTypeToString(frame_type));
+
+  if (frame_type == kEmptyFrame)
+    return true;
+
   if (payload_size == 0)
     return false;
   RTC_CHECK(video_header);
diff --git a/modules/rtp_rtcp/source/rtp_sender_video.h b/modules/rtp_rtcp/source/rtp_sender_video.h
index 8e26206699..d29934f92d 100644
--- a/modules/rtp_rtcp/source/rtp_sender_video.h
+++ b/modules/rtp_rtcp/source/rtp_sender_video.h
@@ -121,6 +121,10 @@ class RTPSenderVideo {
                                   StorageType media_packet_storage,
                                   bool protect_media_packet);
 
+  bool LogAndSendToNetwork(std::unique_ptr<RtpPacketToSend> packet,
+                           StorageType storage,
+                           RtpPacketSender::Priority priority);
+
   bool red_enabled() const RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_) {
     return red_payload_type_ >= 0;
   }
diff --git a/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc b/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc
index 9b155e786c..85bbc80ac9 100644
--- a/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_video_unittest.cc
@@ -112,7 +112,7 @@ class RtpSenderVideoTest : public ::testing::TestWithParam<bool> {
                     &fake_clock_,
                     &transport_,
                     nullptr,
-                    nullptr,
+                    absl::nullopt,
                     nullptr,
                     nullptr,
                     nullptr,