diff --git a/modules/rtp_rtcp/BUILD.gn b/modules/rtp_rtcp/BUILD.gn
index b8dd23ed86..90055480b6 100644
--- a/modules/rtp_rtcp/BUILD.gn
+++ b/modules/rtp_rtcp/BUILD.gn
@@ -156,7 +156,6 @@ rtc_library("rtp_rtcp") {
     "source/forward_error_correction_internal.h",
     "source/packet_loss_stats.cc",
     "source/packet_loss_stats.h",
-    "source/playout_delay_oracle.h",
     "source/receive_statistics_impl.cc",
     "source/receive_statistics_impl.h",
     "source/remote_ntp_time_estimator.cc",
diff --git a/modules/rtp_rtcp/source/playout_delay_oracle.h b/modules/rtp_rtcp/source/playout_delay_oracle.h
deleted file mode 100644
index 04465e3cfc..0000000000
--- a/modules/rtp_rtcp/source/playout_delay_oracle.h
+++ /dev/null
@@ -1,24 +0,0 @@
-/*
- *  Copyright (c) 2016 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 MODULES_RTP_RTCP_SOURCE_PLAYOUT_DELAY_ORACLE_H_
-#define MODULES_RTP_RTCP_SOURCE_PLAYOUT_DELAY_ORACLE_H_
-
-namespace webrtc {
-
-// TODO(sprang): Remove once downstream usage is gone.
-class PlayoutDelayOracle {
- public:
-  PlayoutDelayOracle() = default;
-};
-
-}  // namespace webrtc
-
-#endif  // MODULES_RTP_RTCP_SOURCE_PLAYOUT_DELAY_ORACLE_H_
diff --git a/modules/rtp_rtcp/source/rtp_sender_video.cc b/modules/rtp_rtcp/source/rtp_sender_video.cc
index 532e06950c..2696514ffb 100644
--- a/modules/rtp_rtcp/source/rtp_sender_video.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_video.cc
@@ -246,26 +246,6 @@ bool IsNoopDelay(const PlayoutDelay& delay) {
 
 }  // namespace
 
-RTPSenderVideo::RTPSenderVideo(Clock* clock,
-                               RTPSender* rtp_sender,
-                               FlexfecSender* flexfec_sender,
-                               PlayoutDelayOracle* playout_delay_oracle,
-                               FrameEncryptorInterface* frame_encryptor,
-                               bool require_frame_encryption,
-                               bool enable_retransmit_all_layers,
-                               const WebRtcKeyValueConfig& field_trials)
-    : RTPSenderVideo([&] {
-        Config config;
-        config.clock = clock;
-        config.rtp_sender = rtp_sender;
-        config.flexfec_sender = flexfec_sender;
-        config.frame_encryptor = frame_encryptor;
-        config.require_frame_encryption = require_frame_encryption;
-        config.enable_retransmit_all_layers = enable_retransmit_all_layers;
-        config.field_trials = &field_trials;
-        return config;
-      }()) {}
-
 RTPSenderVideo::RTPSenderVideo(const Config& config)
     : rtp_sender_(config.rtp_sender),
       clock_(config.clock),
diff --git a/modules/rtp_rtcp/source/rtp_sender_video.h b/modules/rtp_rtcp/source/rtp_sender_video.h
index 2df62adf21..5c9657e56f 100644
--- a/modules/rtp_rtcp/source/rtp_sender_video.h
+++ b/modules/rtp_rtcp/source/rtp_sender_video.h
@@ -25,7 +25,6 @@
 #include "modules/rtp_rtcp/include/flexfec_sender.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/playout_delay_oracle.h"
 #include "modules/rtp_rtcp/source/rtp_rtcp_config.h"
 #include "modules/rtp_rtcp/source/rtp_sender.h"
 #include "modules/rtp_rtcp/source/rtp_video_header.h"
@@ -69,8 +68,6 @@ class RTPSenderVideo {
     Clock* clock = nullptr;
     RTPSender* rtp_sender = nullptr;
     FlexfecSender* flexfec_sender = nullptr;
-    // TODO(sprang): Remove when downstream usage is gone.
-    PlayoutDelayOracle* playout_delay_oracle = nullptr;
     FrameEncryptorInterface* frame_encryptor = nullptr;
     bool require_frame_encryption = false;
     bool enable_retransmit_all_layers = false;
@@ -81,15 +78,6 @@ class RTPSenderVideo {
 
   explicit RTPSenderVideo(const Config& config);
 
-  // TODO(bugs.webrtc.org/10809): Remove when downstream usage is gone.
-  RTPSenderVideo(Clock* clock,
-                 RTPSender* rtpSender,
-                 FlexfecSender* flexfec_sender,
-                 PlayoutDelayOracle* playout_delay_oracle,
-                 FrameEncryptorInterface* frame_encryptor,
-                 bool require_frame_encryption,
-                 bool enable_retransmit_all_layers,
-                 const WebRtcKeyValueConfig& field_trials);
   virtual ~RTPSenderVideo();
 
   // expected_retransmission_time_ms.has_value() -> retransmission allowed.