Use RtpPacketToSend in RtpSenderVideo

webrtc/modules/rtp_rtcp/source/rtp_packet.cc

Index: webrtc/modules/rtp_rtcp/source/rtp_packet.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_packet.cc b/webrtc/modules/rtp_rtcp/source/rtp_packet.cc
index 283512cd8fa51e7d7dc74da62f5cc116f9051ad3..d089560e68cf25d59b6b14a6e22a0d6369588da0 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_packet.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_packet.cc
@@ -274,6 +274,7 @@ uint8_t* Packet::AllocatePayload(size_t size_bytes) {
     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_);