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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.
*
* Class for storing RTP packets.
*/
#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_PACKET_HISTORY_H_
#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_PACKET_HISTORY_H_
#include <vector>
#include "webrtc/base/thread_annotations.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/typedefs.h"
namespace webrtc {
class Clock;
class CriticalSectionWrapper;
static const size_t kMaxHistoryCapacity = 9600;
class RTPPacketHistory {
public:
explicit RTPPacketHistory(Clock* clock);
~RTPPacketHistory();
void SetStorePacketsStatus(bool enable, uint16_t number_to_store);
bool StorePackets() const;
// 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.
// 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 elapsed, no bytes are
// copied.
// stored_time_ms: returns the time when the packet was stored.
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);
bool HasRTPPacket(uint16_t sequence_number) const;
bool SetSent(uint16_t sequence_number);
private:
void GetPacket(int index,
uint8_t* packet,
size_t* packet_length,
int64_t* stored_time_ms) const
EXCLUSIVE_LOCKS_REQUIRED(*critsect_);
void Allocate(size_t number_to_store) EXCLUSIVE_LOCKS_REQUIRED(*critsect_);
void Free() EXCLUSIVE_LOCKS_REQUIRED(*critsect_);
void VerifyAndAllocatePacketLength(size_t packet_length, uint32_t start_index)
EXCLUSIVE_LOCKS_REQUIRED(*critsect_);
bool FindSeqNum(uint16_t sequence_number, int32_t* index) const
EXCLUSIVE_LOCKS_REQUIRED(*critsect_);
int FindBestFittingPacket(size_t size) const
EXCLUSIVE_LOCKS_REQUIRED(*critsect_);
private:
Clock* clock_;
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_SOURCE_RTP_PACKET_HISTORY_H_
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