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Side by Side Diff: webrtc/media/engine/webrtcvideoengine2.h

Issue 2133073002: Add periodic logging of video stats. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: use rtc::TimeMillis Created 4 years, 5 months ago
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1 /* 1 /*
2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2014 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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370 const std::vector<uint32_t> ssrcs_; 370 const std::vector<uint32_t> ssrcs_;
371 const std::vector<SsrcGroup> ssrc_groups_; 371 const std::vector<SsrcGroup> ssrc_groups_;
372 webrtc::Call* const call_; 372 webrtc::Call* const call_;
373 rtc::VideoSinkWants sink_wants_; 373 rtc::VideoSinkWants sink_wants_;
374 // Counter used for deciding if the video resolution is currently 374 // Counter used for deciding if the video resolution is currently
375 // restricted by CPU usage. It is reset if |source_| is changed. 375 // restricted by CPU usage. It is reset if |source_| is changed.
376 int cpu_restricted_counter_; 376 int cpu_restricted_counter_;
377 // Total number of times resolution as been requested to be changed due to 377 // Total number of times resolution as been requested to be changed due to
378 // CPU adaptation. 378 // CPU adaptation.
379 int number_of_cpu_adapt_changes_; 379 int number_of_cpu_adapt_changes_;
380 int64_t last_stats_log_ms_;
380 rtc::VideoSourceInterface<cricket::VideoFrame>* source_; 381 rtc::VideoSourceInterface<cricket::VideoFrame>* source_;
381 WebRtcVideoEncoderFactory* const external_encoder_factory_ 382 WebRtcVideoEncoderFactory* const external_encoder_factory_
382 GUARDED_BY(lock_); 383 GUARDED_BY(lock_);
383 384
384 rtc::CriticalSection lock_; 385 rtc::CriticalSection lock_;
385 webrtc::VideoSendStream* stream_ GUARDED_BY(lock_); 386 webrtc::VideoSendStream* stream_ GUARDED_BY(lock_);
386 // Contains settings that are the same for all streams in the MediaChannel, 387 // Contains settings that are the same for all streams in the MediaChannel,
387 // such as codecs, header extensions, and the global bitrate limit for the 388 // such as codecs, header extensions, and the global bitrate limit for the
388 // entire channel. 389 // entire channel.
389 VideoSendStreamParameters parameters_ GUARDED_BY(lock_); 390 VideoSendStreamParameters parameters_ GUARDED_BY(lock_);
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479 StreamParams stream_params_; 480 StreamParams stream_params_;
480 481
481 webrtc::VideoReceiveStream* stream_; 482 webrtc::VideoReceiveStream* stream_;
482 const bool default_stream_; 483 const bool default_stream_;
483 webrtc::VideoReceiveStream::Config config_; 484 webrtc::VideoReceiveStream::Config config_;
484 bool red_disabled_by_remote_side_; 485 bool red_disabled_by_remote_side_;
485 486
486 WebRtcVideoDecoderFactory* const external_decoder_factory_; 487 WebRtcVideoDecoderFactory* const external_decoder_factory_;
487 std::vector<AllocatedDecoder> allocated_decoders_; 488 std::vector<AllocatedDecoder> allocated_decoders_;
488 489
490 int64_t last_stats_log_ms_;
491
489 rtc::CriticalSection sink_lock_; 492 rtc::CriticalSection sink_lock_;
490 rtc::VideoSinkInterface<cricket::VideoFrame>* sink_ GUARDED_BY(sink_lock_); 493 rtc::VideoSinkInterface<cricket::VideoFrame>* sink_ GUARDED_BY(sink_lock_);
491 int last_width_ GUARDED_BY(sink_lock_); 494 int last_width_ GUARDED_BY(sink_lock_);
492 int last_height_ GUARDED_BY(sink_lock_); 495 int last_height_ GUARDED_BY(sink_lock_);
493 // Expands remote RTP timestamps to int64_t to be able to estimate how long 496 // Expands remote RTP timestamps to int64_t to be able to estimate how long
494 // the stream has been running. 497 // the stream has been running.
495 rtc::TimestampWrapAroundHandler timestamp_wraparound_handler_ 498 rtc::TimestampWrapAroundHandler timestamp_wraparound_handler_
496 GUARDED_BY(sink_lock_); 499 GUARDED_BY(sink_lock_);
497 int64_t first_frame_timestamp_ GUARDED_BY(sink_lock_); 500 int64_t first_frame_timestamp_ GUARDED_BY(sink_lock_);
498 // Start NTP time is estimated as current remote NTP time (estimated from 501 // Start NTP time is estimated as current remote NTP time (estimated from
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546 WebRtcVideoDecoderFactory* const external_decoder_factory_; 549 WebRtcVideoDecoderFactory* const external_decoder_factory_;
547 std::vector<VideoCodecSettings> recv_codecs_; 550 std::vector<VideoCodecSettings> recv_codecs_;
548 std::vector<webrtc::RtpExtension> recv_rtp_extensions_; 551 std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
549 webrtc::Call::Config::BitrateConfig bitrate_config_; 552 webrtc::Call::Config::BitrateConfig bitrate_config_;
550 // TODO(deadbeef): Don't duplicate information between 553 // TODO(deadbeef): Don't duplicate information between
551 // send_params/recv_params, rtp_extensions, options, etc. 554 // send_params/recv_params, rtp_extensions, options, etc.
552 VideoSendParameters send_params_; 555 VideoSendParameters send_params_;
553 VideoOptions default_send_options_; 556 VideoOptions default_send_options_;
554 VideoRecvParameters recv_params_; 557 VideoRecvParameters recv_params_;
555 bool red_disabled_by_remote_side_; 558 bool red_disabled_by_remote_side_;
559 int64_t last_stats_log_ms_;
556 }; 560 };
557 561
558 } // namespace cricket 562 } // namespace cricket
559 563
560 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVIDEOENGINE2_H_ 564 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVIDEOENGINE2_H_
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