diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
index 84f1fae216..b9d2bb9fff 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
@@ -533,19 +533,23 @@ size_t RTPSender::TrySendRedundantPayloads(size_t bytes_to_send,
   return bytes_to_send - bytes_left;
 }
 
-size_t RTPSender::SendPadData(size_t bytes,
-                              bool timestamp_provided,
-                              uint32_t timestamp,
-                              int64_t capture_time_ms) {
-  return SendPadData(bytes, timestamp_provided, timestamp, capture_time_ms,
-                     PacketInfo::kNotAProbe);
+size_t RTPSender::SendPadData(size_t bytes, int probe_cluster_id) {
+  return DeprecatedSendPadData(bytes, false, 0, 0, probe_cluster_id);
 }
 
 size_t RTPSender::SendPadData(size_t bytes,
                               bool timestamp_provided,
                               uint32_t timestamp,
-                              int64_t capture_time_ms,
-                              int probe_cluster_id) {
+                              int64_t capture_time_ms) {
+  return DeprecatedSendPadData(bytes, timestamp_provided, timestamp,
+                               capture_time_ms, PacketInfo::kNotAProbe);
+}
+
+size_t RTPSender::DeprecatedSendPadData(size_t bytes,
+                                        bool timestamp_provided,
+                                        uint32_t timestamp,
+                                        int64_t capture_time_ms,
+                                        int probe_cluster_id) {
   // Always send full padding packets. This is accounted for by the
   // RtpPacketSender,
   // which will make sure we don't send too much padding even if a single packet
@@ -872,8 +876,7 @@ size_t RTPSender::TimeToSendPadding(size_t bytes, int probe_cluster_id) {
     return 0;
   size_t bytes_sent = TrySendRedundantPayloads(bytes, probe_cluster_id);
   if (bytes_sent < bytes)
-    bytes_sent +=
-        SendPadData(bytes - bytes_sent, false, 0, 0, probe_cluster_id);
+    bytes_sent += SendPadData(bytes - bytes_sent, probe_cluster_id);
   return bytes_sent;
 }
 
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.h b/webrtc/modules/rtp_rtcp/source/rtp_sender.h
index 6c9928c793..c757eecf14 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.h
@@ -18,6 +18,7 @@
 
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/criticalsection.h"
+#include "webrtc/base/deprecation.h"
 #include "webrtc/base/random.h"
 #include "webrtc/base/rate_statistics.h"
 #include "webrtc/base/thread_annotations.h"
@@ -271,15 +272,11 @@ class RTPSender {
   int32_t SetFecParameters(const FecProtectionParams *delta_params,
                            const FecProtectionParams *key_params);
 
+  RTC_DEPRECATED
   size_t SendPadData(size_t bytes,
                      bool timestamp_provided,
                      uint32_t timestamp,
                      int64_t capture_time_ms);
-  size_t SendPadData(size_t bytes,
-                     bool timestamp_provided,
-                     uint32_t timestamp,
-                     int64_t capture_time_ms,
-                     int probe_cluster_id);
 
   // Called on update of RTP statistics.
   void RegisterRtpStatisticsCallback(StreamDataCountersCallback* callback);
@@ -302,6 +299,14 @@ class RTPSender {
   // time.
   typedef std::map<int64_t, int> SendDelayMap;
 
+  size_t SendPadData(size_t bytes, int probe_cluster_id);
+
+  size_t DeprecatedSendPadData(size_t bytes,
+                               bool timestamp_provided,
+                               uint32_t timestamp,
+                               int64_t capture_time_ms,
+                               int probe_cluster_id);
+
   size_t CreateRtpHeader(uint8_t* header,
                          int8_t payload_type,
                          uint32_t ssrc,
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
index 811ac77398..2cd15963b5 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
@@ -440,16 +440,16 @@ TEST_F(RtpSenderTestWithoutPacer, AssignSequenceNumberMayAllowPadding) {
   auto packet = rtp_sender_->AllocatePacket();
   ASSERT_TRUE(packet);
 
-  ASSERT_FALSE(rtp_sender_->SendPadData(kPaddingSize, false, 0, 0, -1));
+  ASSERT_FALSE(rtp_sender_->TimeToSendPadding(kPaddingSize, -1));
   packet->SetMarker(false);
   ASSERT_TRUE(rtp_sender_->AssignSequenceNumber(packet.get()));
   // Packet without marker bit doesn't allow padding.
-  EXPECT_FALSE(rtp_sender_->SendPadData(kPaddingSize, false, 0, 0, -1));
+  EXPECT_FALSE(rtp_sender_->TimeToSendPadding(kPaddingSize, -1));
 
   packet->SetMarker(true);
   ASSERT_TRUE(rtp_sender_->AssignSequenceNumber(packet.get()));
   // Packet with marker bit allows send padding.
-  EXPECT_TRUE(rtp_sender_->SendPadData(kPaddingSize, false, 0, 0, -1));
+  EXPECT_TRUE(rtp_sender_->TimeToSendPadding(kPaddingSize, -1));
 }
 
 TEST_F(RtpSenderTestWithoutPacer, AssignSequenceNumberSetPaddingTimestamps) {
@@ -460,7 +460,7 @@ TEST_F(RtpSenderTestWithoutPacer, AssignSequenceNumberSetPaddingTimestamps) {
   packet->SetTimestamp(kTimestamp);
 
   ASSERT_TRUE(rtp_sender_->AssignSequenceNumber(packet.get()));
-  ASSERT_TRUE(rtp_sender_->SendPadData(kPaddingSize, false, 0, 0, -1));
+  ASSERT_TRUE(rtp_sender_->TimeToSendPadding(kPaddingSize, -1));
 
   ASSERT_EQ(1u, transport_.sent_packets_.size());
   // Parse the padding packet and verify its timestamp.