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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}
85 lines
2.9 KiB
C++
85 lines
2.9 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_PACKET_HISTORY_H_
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#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_PACKET_HISTORY_H_
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#include <memory>
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#include <vector>
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#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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#include "webrtc/rtc_base/constructormagic.h"
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#include "webrtc/rtc_base/criticalsection.h"
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#include "webrtc/rtc_base/thread_annotations.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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class Clock;
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class RtpPacketToSend;
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class RtpPacketHistory {
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public:
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static constexpr size_t kMaxCapacity = 9600;
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explicit RtpPacketHistory(Clock* clock);
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~RtpPacketHistory();
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void SetStorePacketsStatus(bool enable, uint16_t number_to_store);
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bool StorePackets() const;
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void PutRtpPacket(std::unique_ptr<RtpPacketToSend> packet,
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StorageType type,
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bool sent);
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// Gets stored RTP packet corresponding to the input |sequence number|.
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// Returns nullptr if packet is not found.
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// |min_elapsed_time_ms| is the minimum time that must have elapsed since
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// the last time the packet was resent (parameter is ignored if set to zero).
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// If the packet is found but the minimum time has not elapsed, returns
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// nullptr.
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std::unique_ptr<RtpPacketToSend> GetPacketAndSetSendTime(
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uint16_t sequence_number,
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int64_t min_elapsed_time_ms,
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bool retransmit);
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std::unique_ptr<RtpPacketToSend> GetBestFittingPacket(
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size_t packet_size) const;
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bool HasRtpPacket(uint16_t sequence_number) const;
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private:
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struct StoredPacket {
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uint16_t sequence_number = 0;
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int64_t send_time = 0;
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StorageType storage_type = kDontRetransmit;
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bool has_been_retransmitted = false;
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std::unique_ptr<RtpPacketToSend> packet;
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};
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std::unique_ptr<RtpPacketToSend> GetPacket(int index) const
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RTC_EXCLUSIVE_LOCKS_REQUIRED(critsect_);
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void Allocate(size_t number_to_store) RTC_EXCLUSIVE_LOCKS_REQUIRED(critsect_);
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void Free() RTC_EXCLUSIVE_LOCKS_REQUIRED(critsect_);
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bool FindSeqNum(uint16_t sequence_number, int* index) const
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RTC_EXCLUSIVE_LOCKS_REQUIRED(critsect_);
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int FindBestFittingPacket(size_t size) const
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RTC_EXCLUSIVE_LOCKS_REQUIRED(critsect_);
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Clock* clock_;
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rtc::CriticalSection critsect_;
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bool store_ RTC_GUARDED_BY(critsect_);
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uint32_t prev_index_ RTC_GUARDED_BY(critsect_);
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std::vector<StoredPacket> stored_packets_ RTC_GUARDED_BY(critsect_);
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RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RtpPacketHistory);
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_PACKET_HISTORY_H_
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