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

modules/rtp_rtcp/source/rtp_format.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.
  */
 
 #include "modules/rtp_rtcp/source/rtp_format.h"
 
 #include <utility>
 
 #include "modules/rtp_rtcp/source/rtp_format_h264.h"
 #include "modules/rtp_rtcp/source/rtp_format_video_generic.h"
+#include "modules/rtp_rtcp/source/rtp_format_video_stereo.h"
 #include "modules/rtp_rtcp/source/rtp_format_vp8.h"
 #include "modules/rtp_rtcp/source/rtp_format_vp9.h"
+#include "rtc_base/logging.h"
 
 namespace webrtc {
 RtpPacketizer* RtpPacketizer::Create(RtpVideoCodecTypes type,
                                      size_t max_payload_len,
                                      size_t last_packet_reduction_len,
                                      const RTPVideoTypeHeader* rtp_type_header,
+                                     const RTPVideoStereoInfo* stereoInfo,
                                      FrameType frame_type) {
   switch (type) {
     case kRtpVideoH264:
       RTC_CHECK(rtp_type_header);
       return new RtpPacketizerH264(max_payload_len, last_packet_reduction_len,
                                    rtp_type_header->H264.packetization_mode);
     case kRtpVideoVp8:
       RTC_CHECK(rtp_type_header);
       return new RtpPacketizerVp8(rtp_type_header->VP8, max_payload_len,
                                   last_packet_reduction_len);
     case kRtpVideoVp9:
       RTC_CHECK(rtp_type_header);
       return new RtpPacketizerVp9(rtp_type_header->VP9, max_payload_len,
                                   last_packet_reduction_len);
     case kRtpVideoGeneric:
       return new RtpPacketizerGeneric(frame_type, max_payload_len,
                                       last_packet_reduction_len);
+    case kRtpVideoStereo:
+      return new RtpPacketizerStereo(max_payload_len, last_packet_reduction_len,
+                                     rtp_type_header, stereoInfo);
     case kRtpVideoNone:
       RTC_NOTREACHED();
   }
   return nullptr;
 }
 
 RtpDepacketizer* RtpDepacketizer::Create(RtpVideoCodecTypes type) {
   switch (type) {
     case kRtpVideoH264:
       return new RtpDepacketizerH264();
     case kRtpVideoVp8:
       return new RtpDepacketizerVp8();
     case kRtpVideoVp9:
       return new RtpDepacketizerVp9();
     case kRtpVideoGeneric:
       return new RtpDepacketizerGeneric();
+    case kRtpVideoStereo:
+      return new RtpDepacketizerStereo();
     case kRtpVideoNone:
       assert(false);
   }
   return nullptr;
 }
 }  // namespace webrtc