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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.
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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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