Work on flexible mode and screen sharing.

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

 /*
  *  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.
  */
 #include "webrtc/modules/video_coding/main/source/jitter_buffer.h"
@@ skipping 577 matching lines @@
           << " consecutive old packets received. Flushing the jitter buffer.";
       Flush();
       return kFlushIndicator;
     }
     return kOldPacket;
   }
 
   if (packet.codec == kVideoCodecVP9) {
     // TODO(asapersson): Move this code to appropriate place.
     // TODO(asapersson): Handle out of order GOF.
-    if (packet.codecSpecificHeader.codecHeader.VP9.flexible_mode) {
-      // TODO(asapersson): Add support for flexible mode.
-      return kGeneralError;
-    }
     if (packet.codecSpecificHeader.codecHeader.VP9.ss_data_available) {
       if (!last_gof_valid_ ||
           IsNewerTimestamp(packet.timestamp, last_gof_timestamp_)) {
         last_gof_.CopyGofInfoVP9(
             packet.codecSpecificHeader.codecHeader.VP9.gof);
         last_gof_timestamp_ = packet.timestamp;
         last_gof_valid_ = true;
       }
     }
     if (last_gof_valid_ &&
@@ skipping 641 matching lines @@
   }
   // Evaluate if the RTT is higher than |high_rtt_nack_threshold_ms_|, and in
   // that case we don't wait for retransmissions.
   if (high_rtt_nack_threshold_ms_ >= 0 &&
       rtt_ms_ >= high_rtt_nack_threshold_ms_) {
     return false;
   }
   return true;
 }
 }  // namespace webrtc