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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 |
11 #include "webrtc/video/video_send_stream.h" | 11 #include "webrtc/video/video_send_stream.h" |
12 | 12 |
13 #include <algorithm> | 13 #include <algorithm> |
14 #include <sstream> | 14 #include <sstream> |
15 #include <string> | 15 #include <string> |
16 #include <vector> | 16 #include <vector> |
17 | 17 |
18 #include "webrtc/base/checks.h" | 18 #include "webrtc/base/checks.h" |
19 #include "webrtc/base/logging.h" | 19 #include "webrtc/base/logging.h" |
20 #include "webrtc/base/trace_event.h" | 20 #include "webrtc/base/trace_event.h" |
21 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h" | 21 #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h" |
22 #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h" | 22 #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h" |
23 #include "webrtc/modules/congestion_controller/include/congestion_controller.h" | 23 #include "webrtc/modules/congestion_controller/include/congestion_controller.h" |
24 #include "webrtc/modules/pacing/packet_router.h" | 24 #include "webrtc/modules/pacing/packet_router.h" |
25 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" | 25 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" |
26 #include "webrtc/modules/utility/include/process_thread.h" | 26 #include "webrtc/modules/utility/include/process_thread.h" |
| 27 #include "webrtc/modules/video_coding/utility/ivf_file_writer.h" |
27 #include "webrtc/video/call_stats.h" | 28 #include "webrtc/video/call_stats.h" |
28 #include "webrtc/video/video_capture_input.h" | 29 #include "webrtc/video/video_capture_input.h" |
29 #include "webrtc/video/vie_remb.h" | 30 #include "webrtc/video/vie_remb.h" |
30 #include "webrtc/video_send_stream.h" | 31 #include "webrtc/video_send_stream.h" |
31 | 32 |
32 namespace webrtc { | 33 namespace webrtc { |
33 | 34 |
34 class RtcpIntraFrameObserver; | 35 class RtcpIntraFrameObserver; |
35 class TransportFeedbackObserver; | 36 class TransportFeedbackObserver; |
36 | 37 |
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365 remb_(remb), | 366 remb_(remb), |
366 encoder_thread_(EncoderThreadFunction, this, "EncoderThread"), | 367 encoder_thread_(EncoderThreadFunction, this, "EncoderThread"), |
367 encoder_wakeup_event_(false, false), | 368 encoder_wakeup_event_(false, false), |
368 stop_encoder_thread_(0), | 369 stop_encoder_thread_(0), |
369 overuse_detector_( | 370 overuse_detector_( |
370 Clock::GetRealTimeClock(), | 371 Clock::GetRealTimeClock(), |
371 GetCpuOveruseOptions(config.encoder_settings.full_overuse_time), | 372 GetCpuOveruseOptions(config.encoder_settings.full_overuse_time), |
372 this, | 373 this, |
373 config.post_encode_callback, | 374 config.post_encode_callback, |
374 &stats_proxy_), | 375 &stats_proxy_), |
| 376 encoder_feedback_(Clock::GetRealTimeClock()), |
375 vie_encoder_(num_cpu_cores, | 377 vie_encoder_(num_cpu_cores, |
376 config_.rtp.ssrcs, | |
377 module_process_thread_, | 378 module_process_thread_, |
378 &stats_proxy_, | 379 &stats_proxy_, |
379 config.pre_encode_callback, | 380 config.pre_encode_callback, |
380 &overuse_detector_, | 381 &overuse_detector_, |
381 congestion_controller_->pacer()), | 382 congestion_controller_->pacer()), |
382 video_sender_(vie_encoder_.video_sender()), | 383 video_sender_(vie_encoder_.video_sender()), |
383 bandwidth_observer_(congestion_controller_->GetBitrateController() | 384 bandwidth_observer_(congestion_controller_->GetBitrateController() |
384 ->CreateRtcpBandwidthObserver()), | 385 ->CreateRtcpBandwidthObserver()), |
385 rtp_rtcp_modules_(CreateRtpRtcpModules( | 386 rtp_rtcp_modules_(CreateRtpRtcpModules( |
386 config.send_transport, | 387 config.send_transport, |
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559 encoder_settings->video_codec.startBitrate = | 560 encoder_settings->video_codec.startBitrate = |
560 bitrate_allocator_->AddObserver( | 561 bitrate_allocator_->AddObserver( |
561 this, encoder_settings->video_codec.minBitrate * 1000, | 562 this, encoder_settings->video_codec.minBitrate * 1000, |
562 encoder_settings->video_codec.maxBitrate * 1000) / | 563 encoder_settings->video_codec.maxBitrate * 1000) / |
563 1000; | 564 1000; |
564 | 565 |
565 payload_router_.SetSendStreams(encoder_settings->streams); | 566 payload_router_.SetSendStreams(encoder_settings->streams); |
566 vie_encoder_.SetEncoder(encoder_settings->video_codec, | 567 vie_encoder_.SetEncoder(encoder_settings->video_codec, |
567 encoder_settings->min_transmit_bitrate_bps, | 568 encoder_settings->min_transmit_bitrate_bps, |
568 payload_router_.MaxPayloadLength(), this); | 569 payload_router_.MaxPayloadLength(), this); |
| 570 |
| 571 // Clear stats for disabled layers. |
| 572 for (size_t i = encoder_settings->streams.size(); |
| 573 i < config_.rtp.ssrcs.size(); ++i) { |
| 574 stats_proxy_.OnInactiveSsrc(config_.rtp.ssrcs[i]); |
| 575 } |
| 576 |
569 if (config_.suspend_below_min_bitrate) { | 577 if (config_.suspend_below_min_bitrate) { |
570 video_sender_->SuspendBelowMinBitrate(); | 578 video_sender_->SuspendBelowMinBitrate(); |
571 bitrate_allocator_->EnforceMinBitrate(false); | 579 bitrate_allocator_->EnforceMinBitrate(false); |
572 } | 580 } |
573 // We might've gotten new settings while configuring the encoder settings, | 581 // 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 | 582 // 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). | 583 // a frame (which might correspond to the last frame size). |
576 encoder_wakeup_event_.Set(); | 584 encoder_wakeup_event_.Set(); |
577 continue; | 585 continue; |
578 } | 586 } |
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613 if (config_.overuse_callback) | 621 if (config_.overuse_callback) |
614 config_.overuse_callback->OnLoadUpdate(LoadObserver::kUnderuse); | 622 config_.overuse_callback->OnLoadUpdate(LoadObserver::kUnderuse); |
615 } | 623 } |
616 | 624 |
617 int32_t VideoSendStream::Encoded(const EncodedImage& encoded_image, | 625 int32_t VideoSendStream::Encoded(const EncodedImage& encoded_image, |
618 const CodecSpecificInfo* codec_specific_info, | 626 const CodecSpecificInfo* codec_specific_info, |
619 const RTPFragmentationHeader* fragmentation) { | 627 const RTPFragmentationHeader* fragmentation) { |
620 // |encoded_frame_proxy_| forwards frames to |config_.post_encode_callback|; | 628 // |encoded_frame_proxy_| forwards frames to |config_.post_encode_callback|; |
621 encoded_frame_proxy_.Encoded(encoded_image, codec_specific_info, | 629 encoded_frame_proxy_.Encoded(encoded_image, codec_specific_info, |
622 fragmentation); | 630 fragmentation); |
623 return payload_router_.Encoded(encoded_image, codec_specific_info, | 631 int32_t return_value = payload_router_.Encoded( |
624 fragmentation); | 632 encoded_image, codec_specific_info, fragmentation); |
| 633 |
| 634 if (kEnableFrameRecording) { |
| 635 int layer = codec_specific_info->codecType == kVideoCodecVP8 |
| 636 ? codec_specific_info->codecSpecific.VP8.simulcastIdx |
| 637 : 0; |
| 638 IvfFileWriter* file_writer; |
| 639 { |
| 640 if (file_writers_[layer] == nullptr) { |
| 641 std::ostringstream oss; |
| 642 oss << "send_bitstream_ssrc"; |
| 643 for (uint32_t ssrc : config_.rtp.ssrcs) |
| 644 oss << "_" << ssrc; |
| 645 oss << "_layer" << layer << ".ivf"; |
| 646 file_writers_[layer] = |
| 647 IvfFileWriter::Open(oss.str(), codec_specific_info->codecType); |
| 648 } |
| 649 file_writer = file_writers_[layer].get(); |
| 650 } |
| 651 if (file_writer) { |
| 652 bool ok = file_writer->WriteFrame(encoded_image); |
| 653 RTC_DCHECK(ok); |
| 654 } |
| 655 } |
| 656 |
| 657 return return_value; |
625 } | 658 } |
626 | 659 |
627 void VideoSendStream::ConfigureProtection() { | 660 void VideoSendStream::ConfigureProtection() { |
628 // Enable NACK, FEC or both. | 661 // Enable NACK, FEC or both. |
629 const bool enable_protection_nack = config_.rtp.nack.rtp_history_ms > 0; | 662 const bool enable_protection_nack = config_.rtp.nack.rtp_history_ms > 0; |
630 bool enable_protection_fec = config_.rtp.fec.red_payload_type != -1; | 663 bool enable_protection_fec = config_.rtp.fec.red_payload_type != -1; |
631 // Payload types without picture ID cannot determine that a stream is complete | 664 // Payload types without picture ID cannot determine that a stream is complete |
632 // without retransmitting FEC, so using FEC + NACK for H.264 (for instance) is | 665 // without retransmitting FEC, so using FEC + NACK for H.264 (for instance) is |
633 // a waste of bandwidth since FEC packets still have to be transmitted. Note | 666 // a waste of bandwidth since FEC packets still have to be transmitted. Note |
634 // that this is not the case with FLEXFEC. | 667 // that this is not the case with FLEXFEC. |
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767 &module_nack_rate); | 800 &module_nack_rate); |
768 *sent_video_rate_bps += module_video_rate; | 801 *sent_video_rate_bps += module_video_rate; |
769 *sent_nack_rate_bps += module_nack_rate; | 802 *sent_nack_rate_bps += module_nack_rate; |
770 *sent_fec_rate_bps += module_fec_rate; | 803 *sent_fec_rate_bps += module_fec_rate; |
771 } | 804 } |
772 return 0; | 805 return 0; |
773 } | 806 } |
774 | 807 |
775 } // namespace internal | 808 } // namespace internal |
776 } // namespace webrtc | 809 } // namespace webrtc |
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