Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(341)

Side by Side Diff: webrtc/modules/rtp_rtcp/source/rtp_packet_history.h

Issue 1739273002: [Draft] RtpPacket sketched. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rebase to use landed version of rtp::Packet Created 4 years, 8 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
OLDNEW
1 /* 1 /*
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source 5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 * 9 *
10 * Class for storing RTP packets. 10 * Class for storing RTP packets.
11 */ 11 */
12 12
13 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_PACKET_HISTORY_H_ 13 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_PACKET_HISTORY_H_
14 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_PACKET_HISTORY_H_ 14 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_PACKET_HISTORY_H_
15 15
16 #include <memory>
16 #include <vector> 17 #include <vector>
17 18
18 #include "webrtc/base/criticalsection.h" 19 #include "webrtc/base/criticalsection.h"
19 #include "webrtc/base/thread_annotations.h" 20 #include "webrtc/base/thread_annotations.h"
20 #include "webrtc/modules/include/module_common_types.h" 21 #include "webrtc/modules/include/module_common_types.h"
21 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" 22 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
22 #include "webrtc/typedefs.h" 23 #include "webrtc/typedefs.h"
23 24
24 namespace webrtc { 25 namespace webrtc {
25 26
26 class Clock; 27 class Clock;
28 class RtpPacketToSend;
27 29
28 static const size_t kMaxHistoryCapacity = 9600; 30 static const size_t kMaxHistoryCapacity = 9600;
29 31
30 class RTPPacketHistory { 32 class RTPPacketHistory {
31 public: 33 public:
32 explicit RTPPacketHistory(Clock* clock); 34 explicit RTPPacketHistory(Clock* clock);
33 ~RTPPacketHistory(); 35 ~RTPPacketHistory();
34 36
35 void SetStorePacketsStatus(bool enable, uint16_t number_to_store); 37 void SetStoreSize(uint16_t number_to_store);
36
37 bool StorePackets() const; 38 bool StorePackets() const;
38 39
39 // Stores RTP packet. 40 // Stores RTP packet.
40 int32_t PutRTPPacket(const uint8_t* packet, 41 RtpPacketToSend* PutRtpPacket(std::unique_ptr<RtpPacketToSend>* packet,
41 size_t packet_length, 42 StorageType type);
42 int64_t capture_time_ms,
43 StorageType type);
44 43
45 // Gets stored RTP packet corresponding to the input sequence number. 44 // Gets stored RTP packet corresponding to the input sequence number.
46 // The packet is copied to the buffer pointed to by ptr_rtp_packet. 45 // The packet is copied to the buffer pointed to by ptr_rtp_packet.
47 // The rtp_packet_length should show the available buffer size. 46 // The rtp_packet_length should show the available buffer size.
48 // Returns true if packet is found. 47 // Returns true if packet is found.
49 // packet_length: returns the copied packet length on success. 48 // packet_length: returns the copied packet length on success.
50 // min_elapsed_time_ms: the minimum time that must have elapsed since the last 49 // min_elapsed_time_ms: the minimum time that must have elapsed since the last
51 // time the packet was resent (parameter is ignored if set to zero). 50 // time the packet was resent (parameter is ignored if set to zero).
52 // If the packet is found but the minimum time has not elapsed, no bytes are 51 // If the packet is found but the minimum time has not elapsed, no bytes are
53 // copied. 52 // copied.
54 // stored_time_ms: returns the time when the packet was stored. 53 // stored_time_ms: returns the time when the packet was stored.
55 bool GetPacketAndSetSendTime(uint16_t sequence_number, 54 RtpPacketToSend* GetPacket(uint16_t sequence_number,
56 int64_t min_elapsed_time_ms, 55 int64_t min_elapsed_time_ms,
57 bool retransmit, 56 bool retransmit);
58 uint8_t* packet,
59 size_t* packet_length,
60 int64_t* stored_time_ms);
61 57
62 bool GetBestFittingPacket(uint8_t* packet, size_t* packet_length, 58 RtpPacketToSend* GetBestFittingPacket(size_t packet_length);
63 int64_t* stored_time_ms);
64 59
65 bool HasRTPPacket(uint16_t sequence_number) const; 60 bool HasRTPPacket(uint16_t sequence_number) const;
66 61
67 bool SetSent(uint16_t sequence_number); 62 private:
63 struct StoredPacket {
64 std::unique_ptr<RtpPacketToSend> packet;
65 uint16_t sequence_number = 0;
66 StorageType storage_type = kDontRetransmit;
67 bool has_been_retransmitted = false;
68 };
68 69
69 private: 70 RtpPacketToSend* StorePacket(std::unique_ptr<RtpPacketToSend> packet,
70 void GetPacket(int index, 71 StorageType type)
71 uint8_t* packet,
72 size_t* packet_length,
73 int64_t* stored_time_ms) const
74 EXCLUSIVE_LOCKS_REQUIRED(critsect_); 72 EXCLUSIVE_LOCKS_REQUIRED(critsect_);
75 void Allocate(size_t number_to_store) EXCLUSIVE_LOCKS_REQUIRED(critsect_); 73 void Allocate(size_t number_to_store) EXCLUSIVE_LOCKS_REQUIRED(critsect_);
76 void Free() EXCLUSIVE_LOCKS_REQUIRED(critsect_); 74 void Free() EXCLUSIVE_LOCKS_REQUIRED(critsect_);
77 void VerifyAndAllocatePacketLength(size_t packet_length, uint32_t start_index) 75 void VerifyAndAllocatePacketLength(size_t packet_length, uint32_t start_index)
78 EXCLUSIVE_LOCKS_REQUIRED(critsect_); 76 EXCLUSIVE_LOCKS_REQUIRED(critsect_);
79 bool FindSeqNum(uint16_t sequence_number, int32_t* index) const 77 bool FindSeqNum(uint16_t sequence_number, size_t* index) const
80 EXCLUSIVE_LOCKS_REQUIRED(critsect_); 78 EXCLUSIVE_LOCKS_REQUIRED(critsect_);
81 int FindBestFittingPacket(size_t size) const 79 int FindBestFittingPacket(size_t size) const
82 EXCLUSIVE_LOCKS_REQUIRED(critsect_); 80 EXCLUSIVE_LOCKS_REQUIRED(critsect_);
83 81
84 private:
85 Clock* clock_; 82 Clock* clock_;
86 rtc::CriticalSection critsect_; 83 rtc::CriticalSection critsect_;
87 bool store_ GUARDED_BY(critsect_);
88 uint32_t prev_index_ GUARDED_BY(critsect_); 84 uint32_t prev_index_ GUARDED_BY(critsect_);
89 85
90 struct StoredPacket {
91 StoredPacket();
92 uint16_t sequence_number = 0;
93 int64_t time_ms = 0;
94 int64_t send_time = 0;
95 StorageType storage_type = kDontRetransmit;
96 bool has_been_retransmitted = false;
97
98 uint8_t data[IP_PACKET_SIZE];
99 size_t length = 0;
100 };
101 std::vector<StoredPacket> stored_packets_ GUARDED_BY(critsect_); 86 std::vector<StoredPacket> stored_packets_ GUARDED_BY(critsect_);
102 }; 87 };
103 } // namespace webrtc 88 } // namespace webrtc
104 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_PACKET_HISTORY_H_ 89 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_PACKET_HISTORY_H_
OLDNEW
« no previous file with comments | « webrtc/modules/rtp_rtcp/source/rtp_packet.cc ('k') | webrtc/modules/rtp_rtcp/source/rtp_packet_history.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698