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/*
* 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 MODULES_RTP_RTCP_SOURCE_RTP_PACKET_HISTORY_H_
#define MODULES_RTP_RTCP_SOURCE_RTP_PACKET_HISTORY_H_
#include <memory>
#include <vector>
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "rtc_base/constructormagic.h"
#include "rtc_base/criticalsection.h"
#include "rtc_base/thread_annotations.h"
#include "typedefs.h" // NOLINT(build/include)
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);
// Set RTT, used to avoid premature retransmission and to prevent over-writing
// a packet in the history before we are reasonably sure it has been received.
void SetRtt(int64_t rtt_ms);
// 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 {
StoredPacket();
StoredPacket(StoredPacket&&);
StoredPacket& operator=(StoredPacket&&);
~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_);
size_t prev_index_ RTC_GUARDED_BY(critsect_);
std::vector<StoredPacket> stored_packets_ RTC_GUARDED_BY(critsect_);
int64_t rtt_ms_ RTC_GUARDED_BY(critsect_);
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RtpPacketHistory);
};
} // namespace webrtc
#endif // MODULES_RTP_RTCP_SOURCE_RTP_PACKET_HISTORY_H_