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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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