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

webrtc/modules/rtp_rtcp/source/rtp_format.cc

Index: webrtc/modules/rtp_rtcp/source/rtp_format.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_format.cc b/webrtc/modules/rtp_rtcp/source/rtp_format.cc
index 7f799bb863846122a968365e70e04f26d7099433..325f18efe7f6df2362a2dcbf8ad92e7d6bff8850 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_format.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_format.cc
@@ -12,8 +12,10 @@
 
 #include <utility>
 
+#include "webrtc/base/logging.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_format_h264.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_format_video_stereo.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_format_vp9.h"
 
@@ -22,6 +24,7 @@ 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:
@@ -39,6 +42,9 @@ RtpPacketizer* RtpPacketizer::Create(RtpVideoCodecTypes type,
     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();
   }
@@ -55,6 +61,8 @@ RtpDepacketizer* RtpDepacketizer::Create(RtpVideoCodecTypes type) {
       return new RtpDepacketizerVp9();
     case kRtpVideoGeneric:
       return new RtpDepacketizerGeneric();
+    case kRtpVideoStereo:
+      return new RtpDepacketizerStereo();
     case kRtpVideoNone:
       assert(false);
   }