Moving src/webrtc into src/.

In order to eliminate the WebRTC Subtree mirror in Chromium, WebRTC is moving the content of the src/webrtc directory up to the src/ directory. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38 Reviewed-on: https://webrtc-review.googlesource.com/1560 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19845}
2017-09-15 06:15:48 +02:00
parent 6674846b4a
commit bb547203bf
4576 changed files with 1092 additions and 1196 deletions
--- a/modules/rtp_rtcp/source/rtp_packet_history.h
+++ b/modules/rtp_rtcp/source/rtp_packet_history.h
@ -0,0 +1,85 @@
+/*
+ *  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.
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_PACKET_HISTORY_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_PACKET_HISTORY_H_
+
+#include <memory>
+#include <vector>
+
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/rtc_base/constructormagic.h"
+#include "webrtc/rtc_base/criticalsection.h"
+#include "webrtc/rtc_base/thread_annotations.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+class Clock;
+class RtpPacketToSend;
+
+class RtpPacketHistory {
+ public:
+  static constexpr size_t kMaxCapacity = 9600;
+  explicit RtpPacketHistory(Clock* clock);
+  ~RtpPacketHistory();
+
+  void SetStorePacketsStatus(bool enable, uint16_t number_to_store);
+  bool StorePackets() const;
+
+  void PutRtpPacket(std::unique_ptr<RtpPacketToSend> packet,
+                    StorageType type,
+                    bool sent);
+
+  // Gets stored RTP packet corresponding to the input |sequence number|.
+  // Returns nullptr if packet is not found.
+  // |min_elapsed_time_ms| is 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 elapsed, returns
+  // nullptr.
+  std::unique_ptr<RtpPacketToSend> GetPacketAndSetSendTime(
+      uint16_t sequence_number,
+      int64_t min_elapsed_time_ms,
+      bool retransmit);
+
+  std::unique_ptr<RtpPacketToSend> GetBestFittingPacket(
+      size_t packet_size) const;
+
+  bool HasRtpPacket(uint16_t sequence_number) const;
+
+ private:
+  struct StoredPacket {
+    uint16_t sequence_number = 0;
+    int64_t send_time = 0;
+    StorageType storage_type = kDontRetransmit;
+    bool has_been_retransmitted = false;
+
+    std::unique_ptr<RtpPacketToSend> packet;
+  };
+
+  std::unique_ptr<RtpPacketToSend> GetPacket(int index) const
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(critsect_);
+  void Allocate(size_t number_to_store) RTC_EXCLUSIVE_LOCKS_REQUIRED(critsect_);
+  void Free() RTC_EXCLUSIVE_LOCKS_REQUIRED(critsect_);
+  bool FindSeqNum(uint16_t sequence_number, int* index) const
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(critsect_);
+  int FindBestFittingPacket(size_t size) const
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(critsect_);
+
+  Clock* clock_;
+  rtc::CriticalSection critsect_;
+  bool store_ RTC_GUARDED_BY(critsect_);
+  uint32_t prev_index_ RTC_GUARDED_BY(critsect_);
+  std::vector<StoredPacket> stored_packets_ RTC_GUARDED_BY(critsect_);
+
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RtpPacketHistory);
+};
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_PACKET_HISTORY_H_