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

Issue 1875713002: New peerconnection test case, for video stream forwarding. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 8 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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349 const Dimensions& dimensions, 349 const Dimensions& dimensions,
350 const VideoCodec& codec) const EXCLUSIVE_LOCKS_REQUIRED(lock_); 350 const VideoCodec& codec) const EXCLUSIVE_LOCKS_REQUIRED(lock_);
351 void SetDimensions(int width, int height) 351 void SetDimensions(int width, int height)
352 EXCLUSIVE_LOCKS_REQUIRED(lock_); 352 EXCLUSIVE_LOCKS_REQUIRED(lock_);
353 bool ValidateRtpParameters(const webrtc::RtpParameters& parameters); 353 bool ValidateRtpParameters(const webrtc::RtpParameters& parameters);
354 354
355 // Calls Start or Stop according to whether or not |sending_| is true, 355 // Calls Start or Stop according to whether or not |sending_| is true,
356 // and whether or not the encoding in |rtp_parameters_| is active. 356 // and whether or not the encoding in |rtp_parameters_| is active.
357 void UpdateSendState() EXCLUSIVE_LOCKS_REQUIRED(lock_); 357 void UpdateSendState() EXCLUSIVE_LOCKS_REQUIRED(lock_);
358 358
359 void OnFrame_w(webrtc::VideoFrame video_frame, int64_t frame_delta_ms);
perkj_webrtc 2016/04/08 14:34:07 no need to add a new method.
nisse-webrtc 2016/04/08 14:57:03 I can't call rtc::Bind with a cricket::VideoFrame.
359 rtc::ThreadChecker thread_checker_; 360 rtc::ThreadChecker thread_checker_;
360 rtc::AsyncInvoker invoker_; 361 rtc::AsyncInvoker invoker_;
361 rtc::Thread* worker_thread_; 362 rtc::Thread* worker_thread_;
362 const std::vector<uint32_t> ssrcs_; 363 const std::vector<uint32_t> ssrcs_;
363 const std::vector<SsrcGroup> ssrc_groups_; 364 const std::vector<SsrcGroup> ssrc_groups_;
364 webrtc::Call* const call_; 365 webrtc::Call* const call_;
365 rtc::VideoSinkWants sink_wants_; 366 rtc::VideoSinkWants sink_wants_;
366 // Counter used for deciding if the video resolution is currently 367 // Counter used for deciding if the video resolution is currently
367 // restricted by CPU usage. It is reset if |source_| is changed. 368 // restricted by CPU usage. It is reset if |source_| is changed.
368 int cpu_restricted_counter_; 369 int cpu_restricted_counter_;
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533 // TODO(deadbeef): Don't duplicate information between 534 // TODO(deadbeef): Don't duplicate information between
534 // send_params/recv_params, rtp_extensions, options, etc. 535 // send_params/recv_params, rtp_extensions, options, etc.
535 VideoSendParameters send_params_; 536 VideoSendParameters send_params_;
536 VideoOptions default_send_options_; 537 VideoOptions default_send_options_;
537 VideoRecvParameters recv_params_; 538 VideoRecvParameters recv_params_;
538 }; 539 };
539 540
540 } // namespace cricket 541 } // namespace cricket
541 542
542 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVIDEOENGINE2_H_ 543 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVIDEOENGINE2_H_
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