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

Issue 2165523002: Add loss-based BWE estimate to the outgoing bitrate plot. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Use default constructor for POD type. Created 4 years, 5 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
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60 webrtc::MediaType media_type_; 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)
65 : timestamp(timestamp), header(header) {} 65 : timestamp(timestamp), header(header) {}
66 uint64_t timestamp; 66 uint64_t timestamp;
67 RTPHeader header; 67 RTPHeader header;
68 }; 68 };
69 69
70 struct BwePacketLossEvent {
71 uint64_t timestamp;
72 int32_t new_bitrate;
73 uint8_t fraction_loss;
74 int32_t expected_packets;
75 };
76
70 const ParsedRtcEventLog& parsed_log_; 77 const ParsedRtcEventLog& parsed_log_;
71 78
72 // A list of SSRCs we are interested in analysing. 79 // A list of SSRCs we are interested in analysing.
73 // If left empty, all SSRCs will be considered relevant. 80 // If left empty, all SSRCs will be considered relevant.
74 std::vector<uint32_t> desired_ssrc_; 81 std::vector<uint32_t> desired_ssrc_;
75 82
76 // Maps a stream identifier consisting of ssrc, direction and MediaType 83 // Maps a stream identifier consisting of ssrc, direction and MediaType
77 // to the parsed RTP headers in that stream. Header extensions are parsed 84 // to the parsed RTP headers in that stream. Header extensions are parsed
78 // if the stream has been configured. 85 // if the stream has been configured.
79 std::map<StreamId, std::vector<LoggedRtpPacket>> rtp_packets_; 86 std::map<StreamId, std::vector<LoggedRtpPacket>> rtp_packets_;
80 87
88 // A list of all updates from the send-side loss-based bandwidth estimator.
89 std::vector<BwePacketLossEvent> bwe_loss_updates_;
90
81 // Window and step size used for calculating moving averages, e.g. bitrate. 91 // Window and step size used for calculating moving averages, e.g. bitrate.
82 // The generated data points will be |step_| microseconds apart. 92 // The generated data points will be |step_| microseconds apart.
83 // Only events occuring at most |window_duration_| microseconds before the 93 // Only events occuring at most |window_duration_| microseconds before the
84 // current data point will be part of the average. 94 // current data point will be part of the average.
85 uint64_t window_duration_; 95 uint64_t window_duration_;
86 uint64_t step_; 96 uint64_t step_;
87 97
88 // First and last events of the log. 98 // First and last events of the log.
89 uint64_t begin_time_; 99 uint64_t begin_time_;
90 uint64_t end_time_; 100 uint64_t end_time_;
91 }; 101 };
92 102
93 } // namespace plotting 103 } // namespace plotting
94 } // namespace webrtc 104 } // namespace webrtc
95 105
96 #endif // WEBRTC_TOOLS_EVENT_LOG_VISUALIZER_ANALYZER_H_ 106 #endif // WEBRTC_TOOLS_EVENT_LOG_VISUALIZER_ANALYZER_H_
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