Update layer indices for non-flexible mode according to updates in the RTP payload profile.

webrtc/modules/video_coding/main/source/jitter_buffer.h

 /*
  *  Copyright (c) 2012 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 320 matching lines @@
   CriticalSectionWrapper* crit_sect_;
   // Event to signal when we have a frame ready for decoder.
   rtc::scoped_ptr<EventWrapper> frame_event_;
   // Number of allocated frames.
   int max_number_of_frames_;
   UnorderedFrameList free_frames_ GUARDED_BY(crit_sect_);
   FrameList decodable_frames_ GUARDED_BY(crit_sect_);
   FrameList incomplete_frames_ GUARDED_BY(crit_sect_);
   VCMDecodingState last_decoded_state_ GUARDED_BY(crit_sect_);
   bool first_packet_since_reset_;
-  // Contains scalability structure data for VP9.
-  Vp9SsMap vp9_ss_map_ GUARDED_BY(crit_sect_);
 
   // Statistics.
   VCMReceiveStatisticsCallback* stats_callback_ GUARDED_BY(crit_sect_);
   // Frame counts for each type (key, delta, ...)
   FrameCounts receive_statistics_;
   // Latest calculated frame rates of incoming stream.
   unsigned int incoming_frame_rate_;
   unsigned int incoming_frame_count_;
   int64_t time_last_incoming_frame_count_;
   unsigned int incoming_bit_count_;
@@ skipping 34 matching lines @@
   // Estimated rolling average of packets per frame
   float average_packets_per_frame_;
   // average_packets_per_frame converges fast if we have fewer than this many
   // frames.
   int frame_counter_;
   RTC_DISALLOW_COPY_AND_ASSIGN(VCMJitterBuffer);
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_JITTER_BUFFER_H_