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

modules/rtp_rtcp/source/rtp_format.h

 /*
  *  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.
  */
 
 #ifndef MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H_
 #define MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H_
 
 #include <string>
 
 #include "modules/include/module_common_types.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "rtc_base/constructormagic.h"
 
 namespace webrtc {
 class RtpPacketToSend;
 
 class RtpPacketizer {
  public:
   static 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);
 
   virtual ~RtpPacketizer() {}
 
   // Returns total number of packets which would be produced by the packetizer.
   virtual size_t SetPayloadData(
       const uint8_t* payload_data,
       size_t payload_size,
       const RTPFragmentationHeader* fragmentation) = 0;
 
@@ skipping 22 matching lines @@
 
   virtual ~RtpDepacketizer() {}
 
   // Parses the RTP payload, parsed result will be saved in |parsed_payload|.
   virtual bool Parse(ParsedPayload* parsed_payload,
                      const uint8_t* payload_data,
                      size_t payload_data_length) = 0;
 };
 }  // namespace webrtc
 #endif  // MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_H_