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

webrtc/modules/rtp_rtcp/source/rtp_format.h

Index: webrtc/modules/rtp_rtcp/source/rtp_format.h
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_format.h b/webrtc/modules/rtp_rtcp/source/rtp_format.h
index 378f38a29bf2384b0d6505e890d56c65beb8b8bf..99a50ba8834b8006643daadb5ce4456b93a9ec61 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_format.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_format.h
@@ -26,6 +26,7 @@ class RtpPacketizer {
                                size_t max_payload_len,
                                size_t last_packet_reduction_len,
                                const RTPVideoTypeHeader* rtp_type_header,
+                               const RTPVideoStereoInfo* stereoInfo,
                                FrameType frame_type);
 
   virtual ~RtpPacketizer() {}