diff --git a/webrtc/modules/rtp_rtcp/source/rtp_packet.cc b/webrtc/modules/rtp_rtcp/source/rtp_packet.cc
index e720eebc4a..ec240a8635 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_packet.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_packet.cc
@@ -274,26 +274,23 @@ void Packet::SetCsrcs(const std::vector<uint32_t>& csrcs) {
 }
 
 uint8_t* Packet::AllocatePayload(size_t size_bytes) {
+  // Reset payload size to 0. If CopyOnWrite buffer_ was shared, this will cause
+  // reallocation and memcpy. Keeping just header reduces memcpy size.
+  SetPayloadSize(0);
+  return SetPayloadSize(size_bytes);
+}
+
+uint8_t* Packet::SetPayloadSize(size_t size_bytes) {
   RTC_DCHECK_EQ(padding_size_, 0);
   if (payload_offset_ + size_bytes > capacity()) {
     LOG(LS_WARNING) << "Cannot set payload, not enough space in buffer.";
     return nullptr;
   }
-  // Reset payload size to 0. If CopyOnWrite buffer_ was shared, this will cause
-  // reallocation and memcpy. Setting size to just headers reduces memcpy size.
-  buffer_.SetSize(payload_offset_);
   payload_size_ = size_bytes;
   buffer_.SetSize(payload_offset_ + payload_size_);
   return WriteAt(payload_offset_);
 }
 
-void Packet::SetPayloadSize(size_t size_bytes) {
-  RTC_DCHECK_EQ(padding_size_, 0);
-  RTC_DCHECK_LE(size_bytes, payload_size_);
-  payload_size_ = size_bytes;
-  buffer_.SetSize(payload_offset_ + payload_size_);
-}
-
 bool Packet::SetPadding(uint8_t size_bytes, Random* random) {
   RTC_DCHECK(random);
   if (payload_offset_ + payload_size_ + size_bytes > capacity()) {
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_packet.h b/webrtc/modules/rtp_rtcp/source/rtp_packet.h
index 3c02125e08..75c73cfbcc 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_packet.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_packet.h
@@ -96,8 +96,9 @@ class Packet {
   bool ReserveExtension();
 
   // Reserve size_bytes for payload. Returns nullptr on failure.
+  uint8_t* SetPayloadSize(size_t size_bytes);
+  // Same as SetPayloadSize but doesn't guarantee to keep current payload.
   uint8_t* AllocatePayload(size_t size_bytes);
-  void SetPayloadSize(size_t size_bytes);
   bool SetPadding(uint8_t size_bytes, Random* random);
 
  protected:
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_packet_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_packet_unittest.cc
index fc88497b19..2f1259e1bc 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_packet_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_packet_unittest.cc
@@ -125,7 +125,7 @@ TEST(RtpPacketTest, SetReservedExtensionsAfterPayload) {
   RtpPacketToSend packet(&extensions);
 
   EXPECT_TRUE(packet.ReserveExtension<TransmissionOffset>());
-  packet.AllocatePayload(kPayloadSize);
+  packet.SetPayloadSize(kPayloadSize);
   // Can't set extension after payload.
   EXPECT_FALSE(packet.SetExtension<AudioLevel>(kVoiceActive, kAudioLevel));
   // Unless reserved.
@@ -154,7 +154,7 @@ TEST(RtpPacketTest, CreateUnalignedPadding) {
   packet.SetSequenceNumber(kSeqNum);
   packet.SetTimestamp(kTimestamp);
   packet.SetSsrc(kSsrc);
-  packet.AllocatePayload(kPayloadSize);
+  packet.SetPayloadSize(kPayloadSize);
   Random r(0x123456789);
 
   EXPECT_LT(packet.size(), packet.capacity());