Use RtpPacketToSend in RtpSenderVideo

webrtc/modules/rtp_rtcp/source/rtp_packet.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.
  */
 
@@ skipping 256 matching lines @@
   }
   buffer_.SetSize(payload_offset_);
 }
 
 uint8_t* Packet::AllocatePayload(size_t size_bytes) {
   RTC_DCHECK_EQ(padding_size_, 0u);
   if (payload_offset_ + size_bytes > capacity()) {
     LOG(LS_WARNING) << "Cannot set payload, not enough space in buffer.";
     return nullptr;
   }
+  buffer_.SetSize(payload_offset_);  // Reset payload size to avoid copying it.

[sprang_webrtc, 2016/09/27 10:45:00]

I'm not sure I follow this. How does doing this SetSize before the next two
lines below help avoid a copy?

[danilchap, 2016/09/27 11:47:05]

comment expanded.
this optimization was added to RtpPacket (though used only when creating red
packet from media_packet) because it use internal knowledge how
CopyOnWriteBuffer buffer_ works.

   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_, 0u);
   RTC_DCHECK_LE(size_bytes, payload_size_);
   payload_size_ = size_bytes;
   buffer_.SetSize(payload_offset_ + payload_size_);
@@ skipping 235 matching lines @@
 uint8_t* Packet::WriteAt(size_t offset) {
   return buffer_.data() + offset;
 }
 
 void Packet::WriteAt(size_t offset, uint8_t byte) {
   buffer_.data()[offset] = byte;
 }
 
 }  // namespace rtp
 }  // namespace webrtc