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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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