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

Issue 1613643004: Remove mutable from rtc::CriticalSection members. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 11 months ago
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1 /* 1 /*
2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2013 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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124 StatsUpdateTimes() : resolution_update_ms(0), bitrate_update_ms(0) {} 124 StatsUpdateTimes() : resolution_update_ms(0), bitrate_update_ms(0) {}
125 int64_t resolution_update_ms; 125 int64_t resolution_update_ms;
126 int64_t bitrate_update_ms; 126 int64_t bitrate_update_ms;
127 }; 127 };
128 void PurgeOldStats() EXCLUSIVE_LOCKS_REQUIRED(crit_); 128 void PurgeOldStats() EXCLUSIVE_LOCKS_REQUIRED(crit_);
129 VideoSendStream::StreamStats* GetStatsEntry(uint32_t ssrc) 129 VideoSendStream::StreamStats* GetStatsEntry(uint32_t ssrc)
130 EXCLUSIVE_LOCKS_REQUIRED(crit_); 130 EXCLUSIVE_LOCKS_REQUIRED(crit_);
131 131
132 Clock* const clock_; 132 Clock* const clock_;
133 const VideoSendStream::Config config_; 133 const VideoSendStream::Config config_;
134 mutable rtc::CriticalSection crit_; 134 rtc::CriticalSection crit_;
135 VideoEncoderConfig::ContentType content_type_ GUARDED_BY(crit_); 135 VideoEncoderConfig::ContentType content_type_ GUARDED_BY(crit_);
136 VideoSendStream::Stats stats_ GUARDED_BY(crit_); 136 VideoSendStream::Stats stats_ GUARDED_BY(crit_);
137 uint32_t last_sent_frame_timestamp_ GUARDED_BY(crit_); 137 uint32_t last_sent_frame_timestamp_ GUARDED_BY(crit_);
138 std::map<uint32_t, StatsUpdateTimes> update_times_ GUARDED_BY(crit_); 138 std::map<uint32_t, StatsUpdateTimes> update_times_ GUARDED_BY(crit_);
139 rtc::ExpFilter encode_time_ GUARDED_BY(crit_); 139 rtc::ExpFilter encode_time_ GUARDED_BY(crit_);
140 140
141 // Contains stats used for UMA histograms. These stats will be reset if 141 // Contains stats used for UMA histograms. These stats will be reset if
142 // content type changes between real-time video and screenshare, since these 142 // content type changes between real-time video and screenshare, since these
143 // will be reported separately. 143 // will be reported separately.
144 struct UmaSamplesContainer { 144 struct UmaSamplesContainer {
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164 SampleCounter max_delay_counter_; 164 SampleCounter max_delay_counter_;
165 rtc::RateTracker input_frame_rate_tracker_; 165 rtc::RateTracker input_frame_rate_tracker_;
166 rtc::RateTracker sent_frame_rate_tracker_; 166 rtc::RateTracker sent_frame_rate_tracker_;
167 }; 167 };
168 168
169 rtc::scoped_ptr<UmaSamplesContainer> uma_container_ GUARDED_BY(crit_); 169 rtc::scoped_ptr<UmaSamplesContainer> uma_container_ GUARDED_BY(crit_);
170 }; 170 };
171 171
172 } // namespace webrtc 172 } // namespace webrtc
173 #endif // WEBRTC_VIDEO_SEND_STATISTICS_PROXY_H_ 173 #endif // WEBRTC_VIDEO_SEND_STATISTICS_PROXY_H_
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