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

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 299 matching lines @@
   num_discarded_packets_ = 0;
   time_first_packet_ms_ = 0;
 
   // Start in a non-signaled state.
   waiting_for_completion_.frame_size = 0;
   waiting_for_completion_.timestamp = 0;
   waiting_for_completion_.latest_packet_time = -1;
   first_packet_since_reset_ = true;
   rtt_ms_ = kDefaultRtt;
   last_decoded_state_.Reset();
-  vp9_ss_map_.Reset();
 }
 
 void VCMJitterBuffer::Stop() {
   crit_sect_->Enter();
   UpdateHistograms();
   running_ = false;
   last_decoded_state_.Reset();
-  vp9_ss_map_.Reset();
 
   // Make sure all frames are free and reset.
   for (FrameList::iterator it = decodable_frames_.begin();
        it != decodable_frames_.end(); ++it) {
     free_frames_.push_back(it->second);
   }
   for (FrameList::iterator it = incomplete_frames_.begin();
        it != incomplete_frames_.end(); ++it) {
     free_frames_.push_back(it->second);
   }
@@ skipping 11 matching lines @@
 bool VCMJitterBuffer::Running() const {
   CriticalSectionScoped cs(crit_sect_);
   return running_;
 }
 
 void VCMJitterBuffer::Flush() {
   CriticalSectionScoped cs(crit_sect_);
   decodable_frames_.Reset(&free_frames_);
   incomplete_frames_.Reset(&free_frames_);
   last_decoded_state_.Reset();  // TODO(mikhal): sync reset.
-  vp9_ss_map_.Reset();
   num_consecutive_old_packets_ = 0;
   // Also reset the jitter and delay estimates
   jitter_estimate_.Reset();
   inter_frame_delay_.Reset(clock_->TimeInMilliseconds());
   waiting_for_completion_.frame_size = 0;
   waiting_for_completion_.timestamp = 0;
   waiting_for_completion_.latest_packet_time = -1;
   first_packet_since_reset_ = true;
   missing_sequence_numbers_.clear();
 }
@@ skipping 311 matching lines @@
           << num_consecutive_old_packets_
           << " consecutive old packets received. Flushing the jitter buffer.";
       Flush();
       return kFlushIndicator;
     }
     return kOldPacket;
   }
 
   num_consecutive_old_packets_ = 0;
 
-  if (packet.codec == kVideoCodecVP9) {
-    if (packet.codecSpecificHeader.codecHeader.VP9.flexible_mode) {
-      // TODO(asapersson): Add support for flexible mode.
-      return kGeneralError;
-    }
-    if (!packet.codecSpecificHeader.codecHeader.VP9.flexible_mode) {
-      if (vp9_ss_map_.Insert(packet))
-        vp9_ss_map_.UpdateFrames(&incomplete_frames_);
-
-      vp9_ss_map_.UpdatePacket(const_cast<VCMPacket*>(&packet));
-    }
-    if (!last_decoded_state_.in_initial_state())
-      vp9_ss_map_.RemoveOld(last_decoded_state_.time_stamp());
+  if (packet.codec == kVideoCodecVP9 &&
+      packet.codecSpecificHeader.codecHeader.VP9.flexible_mode) {
+    // TODO(asapersson): Add support for flexible mode.
+    return kGeneralError;
   }
 
   VCMFrameBuffer* frame;
   FrameList* frame_list;
   const VCMFrameBufferEnum error = GetFrame(packet, &frame, &frame_list);
   if (error != kNoError)
     return error;
 
   int64_t now_ms = clock_->TimeInMilliseconds();
   // We are keeping track of the first and latest seq numbers, and
@@ skipping 616 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