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Side by Side Diff: webrtc/tools/event_log_visualizer/analyzer.h

Issue 2188033004: Add BWE plot to event log analyzer. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 4 months ago
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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_TOOLS_EVENT_LOG_VISUALIZER_ANALYZER_H_ 11 #ifndef WEBRTC_TOOLS_EVENT_LOG_VISUALIZER_ANALYZER_H_
12 #define WEBRTC_TOOLS_EVENT_LOG_VISUALIZER_ANALYZER_H_ 12 #define WEBRTC_TOOLS_EVENT_LOG_VISUALIZER_ANALYZER_H_
13 13
14 #include <vector> 14 #include <vector>
15 #include <map> 15 #include <map>
16 16
17 #include "webrtc/call/rtc_event_log_parser.h" 17 #include "webrtc/call/rtc_event_log_parser.h"
18 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
19 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
18 #include "webrtc/tools/event_log_visualizer/plot_base.h" 20 #include "webrtc/tools/event_log_visualizer/plot_base.h"
19 21
20 namespace webrtc { 22 namespace webrtc {
21 namespace plotting { 23 namespace plotting {
22 24
23 class EventLogAnalyzer { 25 class EventLogAnalyzer {
24 public: 26 public:
25 // The EventLogAnalyzer keeps a reference to the ParsedRtcEventLog for the 27 // The EventLogAnalyzer keeps a reference to the ParsedRtcEventLog for the
26 // duration of its lifetime. The ParsedRtcEventLog must not be destroyed or 28 // duration of its lifetime. The ParsedRtcEventLog must not be destroyed or
27 // modified while the EventLogAnalyzer is being used. 29 // modified while the EventLogAnalyzer is being used.
28 explicit EventLogAnalyzer(const ParsedRtcEventLog& log); 30 explicit EventLogAnalyzer(const ParsedRtcEventLog& log);
29 31
30 void CreatePacketGraph(PacketDirection desired_direction, Plot* plot); 32 void CreatePacketGraph(PacketDirection desired_direction, Plot* plot);
31 33
32 void CreatePlayoutGraph(Plot* plot); 34 void CreatePlayoutGraph(Plot* plot);
33 35
34 void CreateSequenceNumberGraph(Plot* plot); 36 void CreateSequenceNumberGraph(Plot* plot);
35 37
36 void CreateDelayChangeGraph(Plot* plot); 38 void CreateDelayChangeGraph(Plot* plot);
37 39
38 void CreateAccumulatedDelayChangeGraph(Plot* plot); 40 void CreateAccumulatedDelayChangeGraph(Plot* plot);
39 41
40 void CreateTotalBitrateGraph(PacketDirection desired_direction, Plot* plot); 42 void CreateTotalBitrateGraph(PacketDirection desired_direction, Plot* plot);
41 43
42 void CreateStreamBitrateGraph(PacketDirection desired_direction, Plot* plot); 44 void CreateStreamBitrateGraph(PacketDirection desired_direction, Plot* plot);
43 45
46 void CreateBweGraph(Plot* plot);
47
44 private: 48 private:
45 class StreamId { 49 class StreamId {
46 public: 50 public:
47 StreamId(uint32_t ssrc, 51 StreamId(uint32_t ssrc, webrtc::PacketDirection direction)
48 webrtc::PacketDirection direction, 52 : ssrc_(ssrc), direction_(direction) {}
49 webrtc::MediaType media_type)
50 : ssrc_(ssrc), direction_(direction), media_type_(media_type) {}
51 bool operator<(const StreamId& other) const; 53 bool operator<(const StreamId& other) const;
52 bool operator==(const StreamId& other) const; 54 bool operator==(const StreamId& other) const;
53 uint32_t GetSsrc() const { return ssrc_; } 55 uint32_t GetSsrc() const { return ssrc_; }
54 webrtc::PacketDirection GetDirection() const { return direction_; } 56 webrtc::PacketDirection GetDirection() const { return direction_; }
55 webrtc::MediaType GetMediaType() const { return media_type_; }
56 57
57 private: 58 private:
58 uint32_t ssrc_; 59 uint32_t ssrc_;
59 webrtc::PacketDirection direction_; 60 webrtc::PacketDirection direction_;
60 webrtc::MediaType media_type_;
61 }; 61 };
62 62
63 struct LoggedRtpPacket { 63 struct LoggedRtpPacket {
64 LoggedRtpPacket(uint64_t timestamp, RTPHeader header) 64 LoggedRtpPacket(uint64_t timestamp, RTPHeader header, size_t total_length)
65 : timestamp(timestamp), header(header) {} 65 : timestamp(timestamp), header(header), total_length(total_length) {}
66 uint64_t timestamp; 66 uint64_t timestamp;
67 RTPHeader header; 67 RTPHeader header;
68 size_t total_length;
69 };
70
71 struct LoggedRtcpPacket {
72 LoggedRtcpPacket(uint64_t timestamp,
73 RTCPPacketType rtcp_type,
74 std::unique_ptr<rtcp::RtcpPacket> rtcp_packet)
75 : timestamp(timestamp),
76 type(rtcp_type),
77 packet(std::move(rtcp_packet)) {}
78 uint64_t timestamp;
79 RTCPPacketType type;
80 std::unique_ptr<rtcp::RtcpPacket> packet;
68 }; 81 };
69 82
70 struct BwePacketLossEvent { 83 struct BwePacketLossEvent {
71 uint64_t timestamp; 84 uint64_t timestamp;
72 int32_t new_bitrate; 85 int32_t new_bitrate;
73 uint8_t fraction_loss; 86 uint8_t fraction_loss;
74 int32_t expected_packets; 87 int32_t expected_packets;
75 }; 88 };
76 89
77 const ParsedRtcEventLog& parsed_log_; 90 const ParsedRtcEventLog& parsed_log_;
78 91
79 // A list of SSRCs we are interested in analysing. 92 // A list of SSRCs we are interested in analysing.
80 // If left empty, all SSRCs will be considered relevant. 93 // If left empty, all SSRCs will be considered relevant.
81 std::vector<uint32_t> desired_ssrc_; 94 std::vector<uint32_t> desired_ssrc_;
82 95
83 // Maps a stream identifier consisting of ssrc, direction and MediaType 96 // Maps a stream identifier consisting of ssrc, direction and MediaType
84 // to the parsed RTP headers in that stream. Header extensions are parsed 97 // to the parsed RTP headers in that stream. Header extensions are parsed
85 // if the stream has been configured. 98 // if the stream has been configured.
86 std::map<StreamId, std::vector<LoggedRtpPacket>> rtp_packets_; 99 std::map<StreamId, std::vector<LoggedRtpPacket>> rtp_packets_;
87 100
101 std::map<StreamId, std::vector<LoggedRtcpPacket>> rtcp_packets_;
102
88 // A list of all updates from the send-side loss-based bandwidth estimator. 103 // A list of all updates from the send-side loss-based bandwidth estimator.
89 std::vector<BwePacketLossEvent> bwe_loss_updates_; 104 std::vector<BwePacketLossEvent> bwe_loss_updates_;
90 105
91 // Window and step size used for calculating moving averages, e.g. bitrate. 106 // Window and step size used for calculating moving averages, e.g. bitrate.
92 // The generated data points will be |step_| microseconds apart. 107 // The generated data points will be |step_| microseconds apart.
93 // Only events occuring at most |window_duration_| microseconds before the 108 // Only events occuring at most |window_duration_| microseconds before the
94 // current data point will be part of the average. 109 // current data point will be part of the average.
95 uint64_t window_duration_; 110 uint64_t window_duration_;
96 uint64_t step_; 111 uint64_t step_;
97 112
98 // First and last events of the log. 113 // First and last events of the log.
99 uint64_t begin_time_; 114 uint64_t begin_time_;
100 uint64_t end_time_; 115 uint64_t end_time_;
101 }; 116 };
102 117
103 } // namespace plotting 118 } // namespace plotting
104 } // namespace webrtc 119 } // namespace webrtc
105 120
106 #endif // WEBRTC_TOOLS_EVENT_LOG_VISUALIZER_ANALYZER_H_ 121 #endif // WEBRTC_TOOLS_EVENT_LOG_VISUALIZER_ANALYZER_H_
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