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Side by Side Diff: webrtc/video/video_send_stream.cc

Issue 1917793002: Remove SendPacer from ViEEncoder (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Renamed SenderDelegate to PacketSender. Created 4 years, 7 months ago
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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 config.pre_encode_callback, 379 &overuse_detector_),
380 &overuse_detector_,
381 congestion_controller_->pacer()),
382 video_sender_(vie_encoder_.video_sender()), 380 video_sender_(vie_encoder_.video_sender()),
383 bandwidth_observer_(congestion_controller_->GetBitrateController() 381 bandwidth_observer_(congestion_controller_->GetBitrateController()
384 ->CreateRtcpBandwidthObserver()), 382 ->CreateRtcpBandwidthObserver()),
385 rtp_rtcp_modules_(CreateRtpRtcpModules( 383 rtp_rtcp_modules_(CreateRtpRtcpModules(
386 config.send_transport, 384 config.send_transport,
387 &encoder_feedback_, 385 &encoder_feedback_,
388 bandwidth_observer_.get(), 386 bandwidth_observer_.get(),
389 congestion_controller_->GetTransportFeedbackObserver(), 387 congestion_controller_->GetTransportFeedbackObserver(),
390 call_stats_->rtcp_rtt_stats(), 388 call_stats_->rtcp_rtt_stats(),
391 congestion_controller_->pacer(), 389 congestion_controller_->pacer(),
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571 bitrate_allocator_->EnforceMinBitrate(false); 569 bitrate_allocator_->EnforceMinBitrate(false);
572 } 570 }
573 // We might've gotten new settings while configuring the encoder settings, 571 // We might've gotten new settings while configuring the encoder settings,
574 // restart from the top to see if that's the case before trying to encode 572 // restart from the top to see if that's the case before trying to encode
575 // a frame (which might correspond to the last frame size). 573 // a frame (which might correspond to the last frame size).
576 encoder_wakeup_event_.Set(); 574 encoder_wakeup_event_.Set();
577 continue; 575 continue;
578 } 576 }
579 577
580 VideoFrame frame; 578 VideoFrame frame;
581 if (input_.GetVideoFrame(&frame)) 579 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 }
582 vie_encoder_.EncodeVideoFrame(frame); 585 vie_encoder_.EncodeVideoFrame(frame);
586 }
583 } 587 }
584 vie_encoder_.DeRegisterExternalEncoder(config_.encoder_settings.payload_type); 588 vie_encoder_.DeRegisterExternalEncoder(config_.encoder_settings.payload_type);
585 } 589 }
586 590
587 void VideoSendStream::ReconfigureVideoEncoder( 591 void VideoSendStream::ReconfigureVideoEncoder(
588 const VideoEncoderConfig& config) { 592 const VideoEncoderConfig& config) {
589 TRACE_EVENT0("webrtc", "VideoSendStream::(Re)configureVideoEncoder"); 593 TRACE_EVENT0("webrtc", "VideoSendStream::(Re)configureVideoEncoder");
590 LOG(LS_INFO) << "(Re)configureVideoEncoder: " << config.ToString(); 594 LOG(LS_INFO) << "(Re)configureVideoEncoder: " << config.ToString();
591 RTC_DCHECK_GE(config_.rtp.ssrcs.size(), config.streams.size()); 595 RTC_DCHECK_GE(config_.rtp.ssrcs.size(), config.streams.size());
592 VideoCodec video_codec = VideoEncoderConfigToVideoCodec( 596 VideoCodec video_codec = VideoEncoderConfigToVideoCodec(
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767 &module_nack_rate); 771 &module_nack_rate);
768 *sent_video_rate_bps += module_video_rate; 772 *sent_video_rate_bps += module_video_rate;
769 *sent_nack_rate_bps += module_nack_rate; 773 *sent_nack_rate_bps += module_nack_rate;
770 *sent_fec_rate_bps += module_fec_rate; 774 *sent_fec_rate_bps += module_fec_rate;
771 } 775 }
772 return 0; 776 return 0;
773 } 777 }
774 778
775 } // namespace internal 779 } // namespace internal
776 } // namespace webrtc 780 } // namespace webrtc
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