Remove time constraint on first retransmit of a packet.

webrtc/modules/rtp_rtcp/source/rtp_packet_history.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.
  *
  *  Class for storing RTP packets.
@@ skipping 28 matching lines @@
   // Stores RTP packet.
   int32_t PutRTPPacket(const uint8_t* packet,
                        size_t packet_length,
                        int64_t capture_time_ms,
                        StorageType type);
 
   // Gets stored RTP packet corresponding to the input sequence number.
   // The packet is copied to the buffer pointed to by ptr_rtp_packet.
   // The rtp_packet_length should show the available buffer size.
   // Returns true if packet is found.
-  // rtp_packet_length: returns the copied packet length on success.
+  // packet_length: returns the copied packet length on success.
   // min_elapsed_time_ms: the minimum time that must have elapsed since the last
   // time the packet was resent (parameter is ignored if set to zero).
-  // If the packet is found but the minimum time has not elaped, no bytes are
+  // If the packet is found but the minimum time has not elapsed, no bytes are
   // copied.
   // stored_time_ms: returns the time when the packet was stored.
-  // type: returns the storage type set in PutRTPPacket.
   bool GetPacketAndSetSendTime(uint16_t sequence_number,
                                int64_t min_elapsed_time_ms,
                                bool retransmit,
                                uint8_t* packet,
                                size_t* packet_length,
                                int64_t* stored_time_ms);
 
   bool GetBestFittingPacket(uint8_t* packet, size_t* packet_length,
                             int64_t* stored_time_ms);
 
@@ skipping 21 matching lines @@
   rtc::scoped_ptr<CriticalSectionWrapper> critsect_;
   bool store_ GUARDED_BY(critsect_);
   uint32_t prev_index_ GUARDED_BY(critsect_);
 
   struct StoredPacket {
     StoredPacket();
     uint16_t sequence_number = 0;
     int64_t time_ms = 0;
     int64_t send_time = 0;
     StorageType storage_type = kDontRetransmit;
+    bool has_been_retransmitted = false;
 
     uint8_t data[IP_PACKET_SIZE];
     size_t length = 0;
   };
   std::vector<StoredPacket> stored_packets_ GUARDED_BY(critsect_);
 };
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_RTP_RTCP_RTP_PACKET_HISTORY_H_