Allow RtpPacket::SetPayloadSize to increase payload size

webrtc/modules/rtp_rtcp/source/rtp_packet_unittest.cc

Index: webrtc/modules/rtp_rtcp/source/rtp_packet_unittest.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_packet_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_packet_unittest.cc
index fc88497b192138647579d36d4862263049e59153..2f1259e1bc4ed615ece0efe7ac542b4f6cf03d95 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());