Cleanup RtpPacketizerGeneric

merge SetPayloadData into constructor.
Reuse payload size split function

Bug: webrtc:9680
Change-Id: If8de22b2fc39c0bdf6d60154ecaaf19e2dccdcc9
Reviewed-on: https://webrtc-review.googlesource.com/97640
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24598}

modules/rtp_rtcp/source/rtp_format.cc

@@ -53,11 +53,8 @@ std::unique_ptr<RtpPacketizer> RtpPacketizer::Create(
       return std::move(packetizer);
     }
     default: {
-      auto packetizer = absl::make_unique<RtpPacketizerGeneric>(
-          rtp_video_header, frame_type, limits.max_payload_len,
-          limits.last_packet_reduction_len);
-      packetizer->SetPayloadData(payload.data(), payload.size(), nullptr);
-      return std::move(packetizer);
+      return absl::make_unique<RtpPacketizerGeneric>(
+          payload, limits, rtp_video_header, frame_type);
     }
   }
 }