Only create H264 frames if there are no gaps in the packet sequence number.

webrtc/modules/video_coding/packet_buffer.h

 /*
  *  Copyright (c) 2016 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 WEBRTC_MODULES_VIDEO_CODING_PACKET_BUFFER_H_
 #define WEBRTC_MODULES_VIDEO_CODING_PACKET_BUFFER_H_
 
+#include <memory>
+#include <set>
 #include <vector>
-#include <memory>
 
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/scoped_ref_ptr.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/video_coding/packet.h"
 #include "webrtc/modules/video_coding/rtp_frame_reference_finder.h"
 #include "webrtc/modules/video_coding/sequence_number_util.h"
 
 namespace webrtc {
@@ skipping 21 matching lines @@
       OnReceivedFrameCallback* frame_callback);
 
   virtual ~PacketBuffer();
 
   // Returns true if |packet| is inserted into the packet buffer, false
   // otherwise. The PacketBuffer will always take ownership of the
   // |packet.dataPtr| when this function is called. Made virtual for testing.
   virtual bool InsertPacket(VCMPacket* packet);
   void ClearTo(uint16_t seq_num);
   void Clear();
+  void PaddingReceived(uint16_t seq_num);
 
   // Timestamp (not RTP timestamp) of the last received packet/keyframe packet.
   rtc::Optional<int64_t> LastReceivedPacketMs() const;
   rtc::Optional<int64_t> LastReceivedKeyframePacketMs() const;
 
   int AddRef() const;
   int Release() const;
 
  protected:
   // Both |start_buffer_size| and |max_buffer_size| must be a power of 2.
@@ skipping 47 matching lines @@
 
   // Get the packet with sequence number |seq_num|.
   // Virtual for testing.
   virtual VCMPacket* GetPacket(uint16_t seq_num)
       EXCLUSIVE_LOCKS_REQUIRED(crit_);
 
   // Mark all slots used by |frame| as not used.
   // Virtual for testing.
   virtual void ReturnFrame(RtpFrameObject* frame);
 
+  void UpdateMissingPackets(uint16_t seq_num) EXCLUSIVE_LOCKS_REQUIRED(crit_);
+
   rtc::CriticalSection crit_;
 
   // Buffer size_ and max_size_ must always be a power of two.
   size_t size_ GUARDED_BY(crit_);
   const size_t max_size_;
 
   // The fist sequence number currently in the buffer.
   uint16_t first_seq_num_ GUARDED_BY(crit_);
 
   // If the packet buffer has received its first packet.
   bool first_packet_received_ GUARDED_BY(crit_);
 
   // If the buffer is cleared to |first_seq_num_|.
   bool is_cleared_to_first_seq_num_ GUARDED_BY(crit_);
 
   // Buffer that holds the inserted packets.
   std::vector<VCMPacket> data_buffer_ GUARDED_BY(crit_);
 
   // Buffer that holds the information about which slot that is currently in use
   // and information needed to determine the continuity between packets.
   std::vector<ContinuityInfo> sequence_buffer_ GUARDED_BY(crit_);
 
   // Called when a received frame is found.
   OnReceivedFrameCallback* const received_frame_callback_;
 
   // Timestamp (not RTP timestamp) of the last received packet/keyframe packet.
   rtc::Optional<int64_t> last_received_packet_ms_ GUARDED_BY(crit_);
   rtc::Optional<int64_t> last_received_keyframe_packet_ms_ GUARDED_BY(crit_);
 
+  rtc::Optional<uint16_t> newest_inserted_seq_num_ GUARDED_BY(crit_);
+  std::set<uint16_t, DescendingSeqNumComp<uint16_t>> missing_packets_
+      GUARDED_BY(crit_);
+
   mutable volatile int ref_count_ = 0;
 };
 
 }  // namespace video_coding
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_VIDEO_CODING_PACKET_BUFFER_H_