Line data Source code
1 : /*
2 : * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
3 : *
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
6 : * tree. An additional intellectual property rights grant can be found
7 : * in the file PATENTS. All contributing project authors may
8 : * be found in the AUTHORS file in the root of the source tree.
9 : */
10 :
11 : #ifndef WEBRTC_VIDEO_SEND_DELAY_STATS_H_
12 : #define WEBRTC_VIDEO_SEND_DELAY_STATS_H_
13 :
14 : #include <map>
15 : #include <memory>
16 : #include <set>
17 :
18 : #include "webrtc/base/criticalsection.h"
19 : #include "webrtc/base/thread_annotations.h"
20 : #include "webrtc/common_types.h"
21 : #include "webrtc/modules/include/module_common_types.h"
22 : #include "webrtc/system_wrappers/include/clock.h"
23 : #include "webrtc/video/stats_counter.h"
24 : #include "webrtc/video_send_stream.h"
25 :
26 : namespace webrtc {
27 :
28 : class SendDelayStats : public SendPacketObserver {
29 : public:
30 : explicit SendDelayStats(Clock* clock);
31 : virtual ~SendDelayStats();
32 :
33 : // Adds the configured ssrcs for the rtp streams.
34 : // Stats will be calculated for these streams.
35 : void AddSsrcs(const VideoSendStream::Config& config);
36 :
37 : // Called when a packet is sent (leaving socket).
38 : bool OnSentPacket(int packet_id, int64_t time_ms);
39 :
40 : protected:
41 : // From SendPacketObserver.
42 : // Called when a packet is sent to the transport.
43 : void OnSendPacket(uint16_t packet_id,
44 : int64_t capture_time_ms,
45 : uint32_t ssrc) override;
46 :
47 : private:
48 : // Map holding sent packets (mapped by sequence number).
49 : struct SequenceNumberOlderThan {
50 0 : bool operator()(uint16_t seq1, uint16_t seq2) const {
51 0 : return IsNewerSequenceNumber(seq2, seq1);
52 : }
53 : };
54 : struct Packet {
55 0 : Packet(uint32_t ssrc, int64_t capture_time_ms, int64_t send_time_ms)
56 0 : : ssrc(ssrc),
57 : capture_time_ms(capture_time_ms),
58 0 : send_time_ms(send_time_ms) {}
59 : uint32_t ssrc;
60 : int64_t capture_time_ms;
61 : int64_t send_time_ms;
62 : };
63 : typedef std::map<uint16_t, Packet, SequenceNumberOlderThan> PacketMap;
64 :
65 : void UpdateHistograms();
66 : void RemoveOld(int64_t now, PacketMap* packets)
67 : EXCLUSIVE_LOCKS_REQUIRED(crit_);
68 : AvgCounter* GetSendDelayCounter(uint32_t ssrc)
69 : EXCLUSIVE_LOCKS_REQUIRED(crit_);
70 :
71 : Clock* const clock_;
72 : rtc::CriticalSection crit_;
73 :
74 : PacketMap packets_ GUARDED_BY(crit_);
75 : size_t num_old_packets_ GUARDED_BY(crit_);
76 : size_t num_skipped_packets_ GUARDED_BY(crit_);
77 :
78 : std::set<uint32_t> ssrcs_ GUARDED_BY(crit_);
79 :
80 : // Mapped by SSRC.
81 : std::map<uint32_t, std::unique_ptr<AvgCounter>> send_delay_counters_
82 : GUARDED_BY(crit_);
83 : };
84 :
85 : } // namespace webrtc
86 : #endif // WEBRTC_VIDEO_SEND_DELAY_STATS_H_
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