Allow RtpPacket::SetPayloadSize to increase payload size

webrtc/modules/rtp_rtcp/source/rtp_packet_unittest.cc

 /*
  *  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.
  */
 #include "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h"
@@ skipping 107 matching lines @@
 
 TEST(RtpPacketTest, SetReservedExtensionsAfterPayload) {
   const size_t kPayloadSize = 4;
   RtpPacketToSend::ExtensionManager extensions;
   extensions.Register(kRtpExtensionTransmissionTimeOffset,
                       kTransmissionOffsetExtensionId);
   extensions.Register(kRtpExtensionAudioLevel, kAudioLevelExtensionId);
   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.
   EXPECT_TRUE(packet.SetExtension<TransmissionOffset>(kTimeOffset));
 }
 
 TEST(RtpPacketTest, CreatePurePadding) {
   const size_t kPaddingSize = kMaxPaddingSize - 1;
   RtpPacketToSend packet(nullptr, 12 + kPaddingSize);
   packet.SetPayloadType(kPayloadType);
   packet.SetSequenceNumber(kSeqNum);
   packet.SetTimestamp(kTimestamp);
   packet.SetSsrc(kSsrc);
   Random random(0x123456789);
 
   EXPECT_LT(packet.size(), packet.capacity());
   EXPECT_FALSE(packet.SetPadding(kPaddingSize + 1, &random));
   EXPECT_TRUE(packet.SetPadding(kPaddingSize, &random));
   EXPECT_EQ(packet.size(), packet.capacity());
 }
 
 TEST(RtpPacketTest, CreateUnalignedPadding) {
   const size_t kPayloadSize = 3;  // Make padding start at unaligned address.
   RtpPacketToSend packet(nullptr, 12 + kPayloadSize + kMaxPaddingSize);
   packet.SetPayloadType(kPayloadType);
   packet.SetSequenceNumber(kSeqNum);
   packet.SetTimestamp(kTimestamp);
   packet.SetSsrc(kSsrc);
-  packet.AllocatePayload(kPayloadSize);
+  packet.SetPayloadSize(kPayloadSize);
   Random r(0x123456789);
 
   EXPECT_LT(packet.size(), packet.capacity());
   EXPECT_TRUE(packet.SetPadding(kMaxPaddingSize, &r));
   EXPECT_EQ(packet.size(), packet.capacity());
 }
 
 TEST(RtpPacketTest, ParseMinimum) {
   RtpPacketReceived packet;
   EXPECT_TRUE(packet.Parse(kMinimumPacket, sizeof(kMinimumPacket)));
@@ skipping 126 matching lines @@
   int32_t time_offset;
   EXPECT_FALSE(packet.GetExtension<TransmissionOffset>(&time_offset));
   packet.IdentifyExtensions(extensions);
   EXPECT_TRUE(packet.GetExtension<TransmissionOffset>(&time_offset));
   EXPECT_EQ(kTimeOffset, time_offset);
   EXPECT_EQ(0u, packet.payload_size());
   EXPECT_EQ(0u, packet.padding_size());
 }
 
 }  // namespace webrtc