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

Issue 2122743003: Fix stats for encoder target bitrate when target rate is zero. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Removed bad logging. Created 4 years, 5 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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131 static const bool kEnableFrameRecording = false; 131 static const bool kEnableFrameRecording = false;
132 static const int kMaxLayers = 3; 132 static const int kMaxLayers = 3;
133 std::unique_ptr<IvfFileWriter> file_writers_[kMaxLayers]; 133 std::unique_ptr<IvfFileWriter> file_writers_[kMaxLayers];
134 134
135 rtc::PlatformThread encoder_thread_; 135 rtc::PlatformThread encoder_thread_;
136 rtc::Event encoder_wakeup_event_; 136 rtc::Event encoder_wakeup_event_;
137 volatile int stop_encoder_thread_; 137 volatile int stop_encoder_thread_;
138 rtc::CriticalSection encoder_settings_crit_; 138 rtc::CriticalSection encoder_settings_crit_;
139 std::unique_ptr<EncoderSettings> pending_encoder_settings_ 139 std::unique_ptr<EncoderSettings> pending_encoder_settings_
140 GUARDED_BY(encoder_settings_crit_); 140 GUARDED_BY(encoder_settings_crit_);
141 uint32_t encoder_max_bitrate_bps_ GUARDED_BY(encoder_settings_crit_);
141 142
142 enum class State { 143 enum class State {
143 kStopped, // VideoSendStream::Start has not yet been called. 144 kStopped, // VideoSendStream::Start has not yet been called.
144 kStarted, // VideoSendStream::Start has been called. 145 kStarted, // VideoSendStream::Start has been called.
145 // VideoSendStream::Start has been called but the encoder have timed out. 146 // VideoSendStream::Start has been called but the encoder have timed out.
146 kEncoderTimedOut, 147 kEncoderTimedOut,
147 }; 148 };
148 rtc::Optional<State> pending_state_change_ GUARDED_BY(encoder_settings_crit_); 149 rtc::Optional<State> pending_state_change_ GUARDED_BY(encoder_settings_crit_);
149 150
150 // Only used on the encoder thread. 151 // Only used on the encoder thread.
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163 const std::unique_ptr<RtcpBandwidthObserver> bandwidth_observer_; 164 const std::unique_ptr<RtcpBandwidthObserver> bandwidth_observer_;
164 // RtpRtcp modules, declared here as they use other members on construction. 165 // RtpRtcp modules, declared here as they use other members on construction.
165 const std::vector<RtpRtcp*> rtp_rtcp_modules_; 166 const std::vector<RtpRtcp*> rtp_rtcp_modules_;
166 PayloadRouter payload_router_; 167 PayloadRouter payload_router_;
167 VideoCaptureInput input_; 168 VideoCaptureInput input_;
168 }; 169 };
169 } // namespace internal 170 } // namespace internal
170 } // namespace webrtc 171 } // namespace webrtc
171 172
172 #endif // WEBRTC_VIDEO_VIDEO_SEND_STREAM_H_ 173 #endif // WEBRTC_VIDEO_VIDEO_SEND_STREAM_H_
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