Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(458)

Side by Side Diff: webrtc/video/send_delay_stats.h

Issue 2013403002: Start integrating StatsCounter class. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 6 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
« no previous file with comments | « no previous file | webrtc/video/send_delay_stats.cc » ('j') | no next file with comments »
Toggle Intra-line Diffs ('i') | Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
OLDNEW
1 /* 1 /*
2 * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2016 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 10
11 #ifndef WEBRTC_VIDEO_SEND_DELAY_STATS_H_ 11 #ifndef WEBRTC_VIDEO_SEND_DELAY_STATS_H_
12 #define WEBRTC_VIDEO_SEND_DELAY_STATS_H_ 12 #define WEBRTC_VIDEO_SEND_DELAY_STATS_H_
13 13
14 #include <map> 14 #include <map>
15 #include <memory> 15 #include <memory>
16 #include <set> 16 #include <set>
17 17
18 #include "webrtc/base/criticalsection.h" 18 #include "webrtc/base/criticalsection.h"
19 #include "webrtc/base/thread_annotations.h" 19 #include "webrtc/base/thread_annotations.h"
20 #include "webrtc/common_types.h" 20 #include "webrtc/common_types.h"
21 #include "webrtc/modules/include/module_common_types.h" 21 #include "webrtc/modules/include/module_common_types.h"
22 #include "webrtc/system_wrappers/include/clock.h" 22 #include "webrtc/system_wrappers/include/clock.h"
23 #include "webrtc/video/stats_counter.h"
23 #include "webrtc/video_send_stream.h" 24 #include "webrtc/video_send_stream.h"
24 25
25 namespace webrtc { 26 namespace webrtc {
26 27
27 class SendDelayStats : public SendPacketObserver { 28 class SendDelayStats : public SendPacketObserver {
28 public: 29 public:
29 explicit SendDelayStats(Clock* clock); 30 explicit SendDelayStats(Clock* clock);
30 virtual ~SendDelayStats(); 31 virtual ~SendDelayStats();
31 32
32 // Adds the configured ssrcs for the rtp streams. 33 // Adds the configured ssrcs for the rtp streams.
(...skipping 21 matching lines...) Expand all
54 Packet(uint32_t ssrc, int64_t capture_time_ms, int64_t send_time_ms) 55 Packet(uint32_t ssrc, int64_t capture_time_ms, int64_t send_time_ms)
55 : ssrc(ssrc), 56 : ssrc(ssrc),
56 capture_time_ms(capture_time_ms), 57 capture_time_ms(capture_time_ms),
57 send_time_ms(send_time_ms) {} 58 send_time_ms(send_time_ms) {}
58 uint32_t ssrc; 59 uint32_t ssrc;
59 int64_t capture_time_ms; 60 int64_t capture_time_ms;
60 int64_t send_time_ms; 61 int64_t send_time_ms;
61 }; 62 };
62 typedef std::map<uint16_t, Packet, SequenceNumberOlderThan> PacketMap; 63 typedef std::map<uint16_t, Packet, SequenceNumberOlderThan> PacketMap;
63 64
64 class SampleCounter {
65 public:
66 SampleCounter() : sum(0), num_samples(0) {}
67 ~SampleCounter() {}
68 void Add(int sample);
69 int Avg(int min_required_samples) const;
70
71 private:
72 int sum;
73 int num_samples;
74 };
75
76 void UpdateHistograms(); 65 void UpdateHistograms();
77 void RemoveOld(int64_t now, PacketMap* packets) 66 void RemoveOld(int64_t now, PacketMap* packets)
78 EXCLUSIVE_LOCKS_REQUIRED(crit_); 67 EXCLUSIVE_LOCKS_REQUIRED(crit_);
68 AvgCounter* GetSendDelayCounter(uint32_t ssrc)
69 EXCLUSIVE_LOCKS_REQUIRED(crit_);
79 70
80 Clock* const clock_; 71 Clock* const clock_;
81 rtc::CriticalSection crit_; 72 rtc::CriticalSection crit_;
82 73
83 PacketMap packets_ GUARDED_BY(crit_); 74 PacketMap packets_ GUARDED_BY(crit_);
84 size_t num_old_packets_ GUARDED_BY(crit_); 75 size_t num_old_packets_ GUARDED_BY(crit_);
85 size_t num_skipped_packets_ GUARDED_BY(crit_); 76 size_t num_skipped_packets_ GUARDED_BY(crit_);
86 77
87 std::set<uint32_t> ssrcs_ GUARDED_BY(crit_); 78 std::set<uint32_t> ssrcs_ GUARDED_BY(crit_);
88 std::map<uint32_t, SampleCounter> send_delay_counters_ 79
89 GUARDED_BY(crit_); // Mapped by SSRC. 80 // Mapped by SSRC.
81 std::map<uint32_t, std::unique_ptr<AvgCounter>> send_delay_counters_
82 GUARDED_BY(crit_);
90 }; 83 };
91 84
92 } // namespace webrtc 85 } // namespace webrtc
93 #endif // WEBRTC_VIDEO_SEND_DELAY_STATS_H_ 86 #endif // WEBRTC_VIDEO_SEND_DELAY_STATS_H_
OLDNEW
« no previous file with comments | « no previous file | webrtc/video/send_delay_stats.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698