diff --git a/modules/rtp_rtcp/BUILD.gn b/modules/rtp_rtcp/BUILD.gn
index 14cd3fcc97..b9e92cdf94 100644
--- a/modules/rtp_rtcp/BUILD.gn
+++ b/modules/rtp_rtcp/BUILD.gn
@@ -270,12 +270,6 @@ rtc_library("rtp_rtcp") {
     ]
   }
 
-  if (rtc_enable_bwe_test_logging) {
-    defines = [ "BWE_TEST_LOGGING_COMPILE_TIME_ENABLE=1" ]
-  } else {
-    defines = [ "BWE_TEST_LOGGING_COMPILE_TIME_ENABLE=0" ]
-  }
-
   deps = [
     ":leb128",
     ":rtp_rtcp_format",
@@ -349,7 +343,6 @@ rtc_library("rtp_rtcp") {
     "../../rtc_base/task_utils:repeating_task",
     "../../system_wrappers",
     "../../system_wrappers:metrics",
-    "../remote_bitrate_estimator",
     "../video_coding:codec_globals_headers",
     "//third_party/abseil-cpp/absl/algorithm:container",
     "//third_party/abseil-cpp/absl/base:core_headers",
@@ -388,7 +381,6 @@ rtc_source_set("rtp_rtcp_legacy") {
     "../../rtc_base:macromagic",
     "../../rtc_base/synchronization:mutex",
     "../../system_wrappers",
-    "../remote_bitrate_estimator",
     "//third_party/abseil-cpp/absl/base:core_headers",
     "//third_party/abseil-cpp/absl/strings",
     "//third_party/abseil-cpp/absl/strings:string_view",
diff --git a/modules/rtp_rtcp/source/deprecated/deprecated_rtp_sender_egress.cc b/modules/rtp_rtcp/source/deprecated/deprecated_rtp_sender_egress.cc
index 6b259440d0..445b141c81 100644
--- a/modules/rtp_rtcp/source/deprecated/deprecated_rtp_sender_egress.cc
+++ b/modules/rtp_rtcp/source/deprecated/deprecated_rtp_sender_egress.cc
@@ -17,7 +17,6 @@
 #include "absl/strings/match.h"
 #include "api/units/timestamp.h"
 #include "logging/rtc_event_log/events/rtc_event_rtp_packet_outgoing.h"
-#include "modules/remote_bitrate_estimator/test/bwe_test_logging.h"
 #include "rtc_base/logging.h"
 
 namespace webrtc {
@@ -69,7 +68,6 @@ DEPRECATED_RtpSenderEgress::DEPRECATED_RtpSenderEgress(
       packet_history_(packet_history),
       transport_(config.outgoing_transport),
       event_log_(config.event_log),
-      is_audio_(config.audio),
       need_rtp_packet_infos_(config.need_rtp_packet_infos),
       transport_feedback_observer_(config.transport_feedback_callback),
       send_packet_observer_(config.send_packet_observer),
@@ -94,27 +92,6 @@ void DEPRECATED_RtpSenderEgress::SendPacket(
   RTC_DCHECK(HasCorrectSsrc(*packet));
   Timestamp now = clock_->CurrentTime();
   int64_t now_ms = now.ms();
-
-  if (is_audio_) {
-#if BWE_TEST_LOGGING_COMPILE_TIME_ENABLE
-    BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "AudioTotBitrate_kbps", now_ms,
-                                    GetSendRates().Sum().kbps(), packet_ssrc);
-    BWE_TEST_LOGGING_PLOT_WITH_SSRC(
-        1, "AudioNackBitrate_kbps", now_ms,
-        GetSendRates()[RtpPacketMediaType::kRetransmission].kbps(),
-        packet_ssrc);
-#endif
-  } else {
-#if BWE_TEST_LOGGING_COMPILE_TIME_ENABLE
-    BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "VideoTotBitrate_kbps", now_ms,
-                                    GetSendRates().Sum().kbps(), packet_ssrc);
-    BWE_TEST_LOGGING_PLOT_WITH_SSRC(
-        1, "VideoNackBitrate_kbps", now_ms,
-        GetSendRates()[RtpPacketMediaType::kRetransmission].kbps(),
-        packet_ssrc);
-#endif
-  }
-
   PacketOptions options;
   {
     MutexLock lock(&lock_);
diff --git a/modules/rtp_rtcp/source/deprecated/deprecated_rtp_sender_egress.h b/modules/rtp_rtcp/source/deprecated/deprecated_rtp_sender_egress.h
index 9d343c2d08..d29a434bbb 100644
--- a/modules/rtp_rtcp/source/deprecated/deprecated_rtp_sender_egress.h
+++ b/modules/rtp_rtcp/source/deprecated/deprecated_rtp_sender_egress.h
@@ -106,7 +106,6 @@ class DEPRECATED_RtpSenderEgress {
   RtpPacketHistory* const packet_history_;
   Transport* const transport_;
   RtcEventLog* const event_log_;
-  const bool is_audio_;
   const bool need_rtp_packet_infos_;
 
   TransportFeedbackObserver* const transport_feedback_observer_;
diff --git a/modules/rtp_rtcp/source/receive_statistics_impl.cc b/modules/rtp_rtcp/source/receive_statistics_impl.cc
index 1dc56bb96f..1ca7795dbf 100644
--- a/modules/rtp_rtcp/source/receive_statistics_impl.cc
+++ b/modules/rtp_rtcp/source/receive_statistics_impl.cc
@@ -17,7 +17,6 @@
 #include <vector>
 
 #include "api/units/time_delta.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/rtcp_packet/report_block.h"
 #include "modules/rtp_rtcp/source/rtp_packet_received.h"
@@ -282,11 +281,6 @@ void StreamStatisticianImpl::MaybeAppendReportBlockAndReset(
   // Only for report blocks in RTCP SR and RR.
   last_report_cumulative_loss_ = cumulative_loss_;
   last_report_seq_max_ = received_seq_max_;
-  BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "cumulative_loss_pkts", now.ms(),
-                                  cumulative_loss_, ssrc_);
-  BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "received_seq_max_pkts", now.ms(),
-                                  (received_seq_max_ - received_seq_first_),
-                                  ssrc_);
 }
 
 absl::optional<int> StreamStatisticianImpl::GetFractionLostInPercent() const {
diff --git a/modules/rtp_rtcp/source/rtcp_sender.h b/modules/rtp_rtcp/source/rtcp_sender.h
index 0ceec9a64a..fe2ec9b9f9 100644
--- a/modules/rtp_rtcp/source/rtcp_sender.h
+++ b/modules/rtp_rtcp/source/rtcp_sender.h
@@ -24,7 +24,6 @@
 #include "api/units/time_delta.h"
 #include "api/units/timestamp.h"
 #include "api/video/video_bitrate_allocation.h"
-#include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
 #include "modules/rtp_rtcp/include/receive_statistics.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "modules/rtp_rtcp/source/rtcp_nack_stats.h"
diff --git a/modules/rtp_rtcp/source/rtp_sender_egress.cc b/modules/rtp_rtcp/source/rtp_sender_egress.cc
index 8292489896..5c97ef8304 100644
--- a/modules/rtp_rtcp/source/rtp_sender_egress.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_egress.cc
@@ -161,9 +161,6 @@ void RtpSenderEgress::SendPacket(std::unique_ptr<RtpPacketToSend> packet,
   }
 
   const Timestamp now = clock_->CurrentTime();
-#if BWE_TEST_LOGGING_COMPILE_TIME_ENABLE
-  BweTestLoggingPlot(now, packet->Ssrc());
-#endif
   if (need_rtp_packet_infos_ &&
       packet->packet_type() == RtpPacketToSend::Type::kVideo) {
     // Last packet of a frame, add it to sequence number info map.
@@ -501,25 +498,4 @@ void RtpSenderEgress::PeriodicUpdate() {
       send_rates.Sum().bps(),
       send_rates[RtpPacketMediaType::kRetransmission].bps(), ssrc_);
 }
-
-#if BWE_TEST_LOGGING_COMPILE_TIME_ENABLE
-void RtpSenderEgress::BweTestLoggingPlot(Timestamp now, uint32_t packet_ssrc) {
-  RTC_DCHECK_RUN_ON(worker_queue_);
-
-  const auto rates = GetSendRates(now);
-  if (is_audio_) {
-    BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "AudioTotBitrate_kbps", now.ms(),
-                                    rates.Sum().kbps(), packet_ssrc);
-    BWE_TEST_LOGGING_PLOT_WITH_SSRC(
-        1, "AudioNackBitrate_kbps", now.ms(),
-        rates[RtpPacketMediaType::kRetransmission].kbps(), packet_ssrc);
-  } else {
-    BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "VideoTotBitrate_kbps", now.ms(),
-                                    rates.Sum().kbps(), packet_ssrc);
-    BWE_TEST_LOGGING_PLOT_WITH_SSRC(
-        1, "VideoNackBitrate_kbps", now.ms(),
-        rates[RtpPacketMediaType::kRetransmission].kbps(), packet_ssrc);
-  }
-}
-#endif  // BWE_TEST_LOGGING_COMPILE_TIME_ENABLE
 }  // namespace webrtc
diff --git a/modules/rtp_rtcp/source/rtp_sender_egress.h b/modules/rtp_rtcp/source/rtp_sender_egress.h
index 692757cb9d..49ee8c1278 100644
--- a/modules/rtp_rtcp/source/rtp_sender_egress.h
+++ b/modules/rtp_rtcp/source/rtp_sender_egress.h
@@ -25,7 +25,6 @@
 #include "api/units/data_rate.h"
 #include "api/units/time_delta.h"
 #include "api/units/timestamp.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/packet_sequencer.h"
 #include "modules/rtp_rtcp/source/rtp_packet_history.h"
@@ -120,9 +119,6 @@ class RtpSenderEgress {
                       RtpPacketMediaType packet_type,
                       RtpPacketCounter counter,
                       size_t packet_size);
-#if BWE_TEST_LOGGING_COMPILE_TIME_ENABLE
-  void BweTestLoggingPlot(Timestamp now, uint32_t packet_ssrc);
-#endif
 
   // Called on a timer, once a second, on the worker_queue_.
   void PeriodicUpdate();
diff --git a/modules/rtp_rtcp/source/rtp_sender_video.cc b/modules/rtp_rtcp/source/rtp_sender_video.cc
index c60305888b..6a53245c4e 100644
--- a/modules/rtp_rtcp/source/rtp_sender_video.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_video.cc
@@ -27,7 +27,6 @@
 #include "api/units/frequency.h"
 #include "api/units/time_delta.h"
 #include "api/units/timestamp.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/absolute_capture_time_sender.h"
 #include "modules/rtp_rtcp/source/byte_io.h"