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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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| 369 overuse_detector_( | 369 overuse_detector_( |
| 370 Clock::GetRealTimeClock(), | 370 Clock::GetRealTimeClock(), |
| 371 GetCpuOveruseOptions(config.encoder_settings.full_overuse_time), | 371 GetCpuOveruseOptions(config.encoder_settings.full_overuse_time), |
| 372 this, | 372 this, |
| 373 config.post_encode_callback, | 373 config.post_encode_callback, |
| 374 &stats_proxy_), | 374 &stats_proxy_), |
| 375 vie_encoder_(num_cpu_cores, | 375 vie_encoder_(num_cpu_cores, |
| 376 config_.rtp.ssrcs, | 376 config_.rtp.ssrcs, |
| 377 module_process_thread_, | 377 module_process_thread_, |
| 378 &stats_proxy_, | 378 &stats_proxy_, |
| 379 &overuse_detector_), | 379 config.pre_encode_callback, |
| 380 &overuse_detector_, |
| 381 congestion_controller_->pacer()), |
| 380 video_sender_(vie_encoder_.video_sender()), | 382 video_sender_(vie_encoder_.video_sender()), |
| 381 bandwidth_observer_(congestion_controller_->GetBitrateController() | 383 bandwidth_observer_(congestion_controller_->GetBitrateController() |
| 382 ->CreateRtcpBandwidthObserver()), | 384 ->CreateRtcpBandwidthObserver()), |
| 383 rtp_rtcp_modules_(CreateRtpRtcpModules( | 385 rtp_rtcp_modules_(CreateRtpRtcpModules( |
| 384 config.send_transport, | 386 config.send_transport, |
| 385 &encoder_feedback_, | 387 &encoder_feedback_, |
| 386 bandwidth_observer_.get(), | 388 bandwidth_observer_.get(), |
| 387 congestion_controller_->GetTransportFeedbackObserver(), | 389 congestion_controller_->GetTransportFeedbackObserver(), |
| 388 call_stats_->rtcp_rtt_stats(), | 390 call_stats_->rtcp_rtt_stats(), |
| 389 congestion_controller_->pacer(), | 391 congestion_controller_->pacer(), |
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| 569 bitrate_allocator_->EnforceMinBitrate(false); | 571 bitrate_allocator_->EnforceMinBitrate(false); |
| 570 } | 572 } |
| 571 // We might've gotten new settings while configuring the encoder settings, | 573 // We might've gotten new settings while configuring the encoder settings, |
| 572 // restart from the top to see if that's the case before trying to encode | 574 // restart from the top to see if that's the case before trying to encode |
| 573 // a frame (which might correspond to the last frame size). | 575 // a frame (which might correspond to the last frame size). |
| 574 encoder_wakeup_event_.Set(); | 576 encoder_wakeup_event_.Set(); |
| 575 continue; | 577 continue; |
| 576 } | 578 } |
| 577 | 579 |
| 578 VideoFrame frame; | 580 VideoFrame frame; |
| 579 if (input_.GetVideoFrame(&frame)) { | 581 if (input_.GetVideoFrame(&frame)) |
| 580 // TODO(perkj): |pre_encode_callback| is only used by tests. Tests should | |
| 581 // register as a sink to the VideoSource instead. | |
| 582 if (config_.pre_encode_callback) { | |
| 583 config_.pre_encode_callback->OnFrame(frame); | |
| 584 } | |
| 585 vie_encoder_.EncodeVideoFrame(frame); | 582 vie_encoder_.EncodeVideoFrame(frame); |
| 586 } | |
| 587 } | 583 } |
| 588 vie_encoder_.DeRegisterExternalEncoder(config_.encoder_settings.payload_type); | 584 vie_encoder_.DeRegisterExternalEncoder(config_.encoder_settings.payload_type); |
| 589 } | 585 } |
| 590 | 586 |
| 591 void VideoSendStream::ReconfigureVideoEncoder( | 587 void VideoSendStream::ReconfigureVideoEncoder( |
| 592 const VideoEncoderConfig& config) { | 588 const VideoEncoderConfig& config) { |
| 593 TRACE_EVENT0("webrtc", "VideoSendStream::(Re)configureVideoEncoder"); | 589 TRACE_EVENT0("webrtc", "VideoSendStream::(Re)configureVideoEncoder"); |
| 594 LOG(LS_INFO) << "(Re)configureVideoEncoder: " << config.ToString(); | 590 LOG(LS_INFO) << "(Re)configureVideoEncoder: " << config.ToString(); |
| 595 RTC_DCHECK_GE(config_.rtp.ssrcs.size(), config.streams.size()); | 591 RTC_DCHECK_GE(config_.rtp.ssrcs.size(), config.streams.size()); |
| 596 VideoCodec video_codec = VideoEncoderConfigToVideoCodec( | 592 VideoCodec video_codec = VideoEncoderConfigToVideoCodec( |
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| 771 &module_nack_rate); | 767 &module_nack_rate); |
| 772 *sent_video_rate_bps += module_video_rate; | 768 *sent_video_rate_bps += module_video_rate; |
| 773 *sent_nack_rate_bps += module_nack_rate; | 769 *sent_nack_rate_bps += module_nack_rate; |
| 774 *sent_fec_rate_bps += module_fec_rate; | 770 *sent_fec_rate_bps += module_fec_rate; |
| 775 } | 771 } |
| 776 return 0; | 772 return 0; |
| 777 } | 773 } |
| 778 | 774 |
| 779 } // namespace internal | 775 } // namespace internal |
| 780 } // namespace webrtc | 776 } // namespace webrtc |
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