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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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| 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_( | 375 vie_encoder_( |
| 376 num_cpu_cores, | 376 num_cpu_cores, |
| 377 config_.rtp.ssrcs, | 377 config_.rtp.ssrcs, |
| 378 module_process_thread_, | 378 module_process_thread_, |
| 379 &stats_proxy_, | 379 &stats_proxy_, |
| 380 config.pre_encode_callback, | |
| 381 &overuse_detector_, | 380 &overuse_detector_, |
| 382 congestion_controller_->pacer(), | |
| 383 &payload_router_, | 381 &payload_router_, |
| 384 config.post_encode_callback ? &encoded_frame_proxy_ : nullptr), | 382 config.post_encode_callback ? &encoded_frame_proxy_ : nullptr), |
| 385 video_sender_(vie_encoder_.video_sender()), | 383 video_sender_(vie_encoder_.video_sender()), |
| 386 bandwidth_observer_(congestion_controller_->GetBitrateController() | 384 bandwidth_observer_(congestion_controller_->GetBitrateController() |
| 387 ->CreateRtcpBandwidthObserver()), | 385 ->CreateRtcpBandwidthObserver()), |
| 388 rtp_rtcp_modules_(CreateRtpRtcpModules( | 386 rtp_rtcp_modules_(CreateRtpRtcpModules( |
| 389 config.send_transport, | 387 config.send_transport, |
| 390 &encoder_feedback_, | 388 &encoder_feedback_, |
| 391 bandwidth_observer_.get(), | 389 bandwidth_observer_.get(), |
| 392 congestion_controller_->GetTransportFeedbackObserver(), | 390 congestion_controller_->GetTransportFeedbackObserver(), |
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| 572 bitrate_allocator_->EnforceMinBitrate(false); | 570 bitrate_allocator_->EnforceMinBitrate(false); |
| 573 } | 571 } |
| 574 // We might've gotten new settings while configuring the encoder settings, | 572 // We might've gotten new settings while configuring the encoder settings, |
| 575 // restart from the top to see if that's the case before trying to encode | 573 // restart from the top to see if that's the case before trying to encode |
| 576 // a frame (which might correspond to the last frame size). | 574 // a frame (which might correspond to the last frame size). |
| 577 encoder_wakeup_event_.Set(); | 575 encoder_wakeup_event_.Set(); |
| 578 continue; | 576 continue; |
| 579 } | 577 } |
| 580 | 578 |
| 581 VideoFrame frame; | 579 VideoFrame frame; |
| 582 if (input_.GetVideoFrame(&frame)) | 580 if (input_.GetVideoFrame(&frame)) { |
| 581 // TODO(perkj): |pre_encode_callback| is only used by tests. Tests should |
| 582 // register as a sink to the VideoSource instead. |
| 583 if (config_.pre_encode_callback) { |
| 584 config_.pre_encode_callback->OnFrame(frame); |
| 585 } |
| 583 vie_encoder_.EncodeVideoFrame(frame); | 586 vie_encoder_.EncodeVideoFrame(frame); |
| 587 } |
| 584 } | 588 } |
| 585 vie_encoder_.DeRegisterExternalEncoder(config_.encoder_settings.payload_type); | 589 vie_encoder_.DeRegisterExternalEncoder(config_.encoder_settings.payload_type); |
| 586 } | 590 } |
| 587 | 591 |
| 588 void VideoSendStream::ReconfigureVideoEncoder( | 592 void VideoSendStream::ReconfigureVideoEncoder( |
| 589 const VideoEncoderConfig& config) { | 593 const VideoEncoderConfig& config) { |
| 590 TRACE_EVENT0("webrtc", "VideoSendStream::(Re)configureVideoEncoder"); | 594 TRACE_EVENT0("webrtc", "VideoSendStream::(Re)configureVideoEncoder"); |
| 591 LOG(LS_INFO) << "(Re)configureVideoEncoder: " << config.ToString(); | 595 LOG(LS_INFO) << "(Re)configureVideoEncoder: " << config.ToString(); |
| 592 RTC_DCHECK_GE(config_.rtp.ssrcs.size(), config.streams.size()); | 596 RTC_DCHECK_GE(config_.rtp.ssrcs.size(), config.streams.size()); |
| 593 VideoCodec video_codec = VideoEncoderConfigToVideoCodec( | 597 VideoCodec video_codec = VideoEncoderConfigToVideoCodec( |
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| 757 &module_nack_rate); | 761 &module_nack_rate); |
| 758 *sent_video_rate_bps += module_video_rate; | 762 *sent_video_rate_bps += module_video_rate; |
| 759 *sent_nack_rate_bps += module_nack_rate; | 763 *sent_nack_rate_bps += module_nack_rate; |
| 760 *sent_fec_rate_bps += module_fec_rate; | 764 *sent_fec_rate_bps += module_fec_rate; |
| 761 } | 765 } |
| 762 return 0; | 766 return 0; |
| 763 } | 767 } |
| 764 | 768 |
| 765 } // namespace internal | 769 } // namespace internal |
| 766 } // namespace webrtc | 770 } // namespace webrtc |
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