[EXPERIMENTAL] Generic stereo codec with index header sending single frames

modules/rtp_rtcp/source/rtp_format_h264_unittest.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 50 matching lines @@
   fragmentation->fragmentationLength[2] =
       kNalHeaderSize + frameSize - payloadOffset;
 }
 
 RtpPacketizer* CreateH264Packetizer(H264PacketizationMode mode,
                                     size_t max_payload_size,
                                     size_t last_packet_reduction) {
   RTPVideoTypeHeader type_header;
   type_header.H264.packetization_mode = mode;
   return RtpPacketizer::Create(kRtpVideoH264, max_payload_size,
-                               last_packet_reduction, &type_header,
+                               last_packet_reduction, &type_header, nullptr,
                                kEmptyFrame);
 }
 
 void VerifyFua(size_t fua_index,
                const uint8_t* expected_payload,
                int offset,
                rtc::ArrayView<const uint8_t> packet,
                const std::vector<size_t>& expected_sizes) {
   ASSERT_EQ(expected_sizes[fua_index] + kFuAHeaderSize, packet.size())
       << "FUA index: " << fua_index;
@@ skipping 855 matching lines @@
   EXPECT_EQ(kVideoFrameDelta, payload.frame_type);
   EXPECT_EQ(kH264SingleNalu, h264.packetization_type);
   EXPECT_EQ(kSei, h264.nalu_type);
   ASSERT_EQ(1u, h264.nalus_length);
   EXPECT_EQ(static_cast<H264::NaluType>(kSei), h264.nalus[0].type);
   EXPECT_EQ(-1, h264.nalus[0].sps_id);
   EXPECT_EQ(-1, h264.nalus[0].pps_id);
 }
 
 }  // namespace webrtc