TEST-ONLY: Do not set SEI nalu as a keyframe.

webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc

 /*
  *  Copyright (c) 2014 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 507 matching lines @@
           LOG(LS_WARNING) << "Failed to parse PPS id from slice of type: "
                           << static_cast<int>(nalu.type);
         }
         break;
       }
       // Slices below don't contain SPS or PPS ids.
       case H264::NaluType::kAud:
       case H264::NaluType::kEndOfSequence:
       case H264::NaluType::kEndOfStream:
       case H264::NaluType::kFiller:
-        break;
       case H264::NaluType::kSei:
-        parsed_payload->frame_type = kVideoFrameKey;
         break;
       case H264::NaluType::kStapA:
       case H264::NaluType::kFuA:
         LOG(LS_WARNING) << "Unexpected STAP-A or FU-A received.";
         return false;
     }
     RTPVideoHeaderH264* h264 = &parsed_payload->type.Video.codecHeader.H264;
     if (h264->nalus_length == kMaxNalusPerPacket) {
       LOG(LS_WARNING)
           << "Received packet containing more than " << kMaxNalusPerPacket
@@ skipping 54 matching lines @@
   h264->packetization_type = kH264FuA;
   h264->nalu_type = original_nal_type;
   if (first_fragment) {
     h264->nalus[h264->nalus_length] = nalu;
     h264->nalus_length = 1;
   }
   return true;
 }
 
 }  // namespace webrtc