| OLD | NEW |
| 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 |
| (...skipping 332 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
| 369 encoder_thread_(EncoderThreadFunction, this, "EncoderThread"), | 370 encoder_thread_(EncoderThreadFunction, this, "EncoderThread"), |
| 370 encoder_wakeup_event_(false, false), | 371 encoder_wakeup_event_(false, false), |
| 371 stop_encoder_thread_(0), | 372 stop_encoder_thread_(0), |
| 372 overuse_detector_( | 373 overuse_detector_( |
| 373 Clock::GetRealTimeClock(), | 374 Clock::GetRealTimeClock(), |
| 374 GetCpuOveruseOptions(config.encoder_settings.full_overuse_time), | 375 GetCpuOveruseOptions(config.encoder_settings.full_overuse_time), |
| 375 this, | 376 this, |
| 376 config.post_encode_callback, | 377 config.post_encode_callback, |
| 377 &stats_proxy_), | 378 &stats_proxy_), |
| 378 vie_encoder_(num_cpu_cores, | 379 vie_encoder_(num_cpu_cores, |
| 379 config_.rtp.ssrcs, | |
| 380 module_process_thread_, | 380 module_process_thread_, |
| 381 &stats_proxy_, | 381 &stats_proxy_, |
| 382 &overuse_detector_), | 382 &overuse_detector_), |
| 383 encoder_feedback_(Clock::GetRealTimeClock(), |
| 384 config.rtp.ssrcs, |
| 385 &vie_encoder_), |
| 383 video_sender_(vie_encoder_.video_sender()), | 386 video_sender_(vie_encoder_.video_sender()), |
| 384 bandwidth_observer_(congestion_controller_->GetBitrateController() | 387 bandwidth_observer_(congestion_controller_->GetBitrateController() |
| 385 ->CreateRtcpBandwidthObserver()), | 388 ->CreateRtcpBandwidthObserver()), |
| 386 rtp_rtcp_modules_(CreateRtpRtcpModules( | 389 rtp_rtcp_modules_(CreateRtpRtcpModules( |
| 387 config.send_transport, | 390 config.send_transport, |
| 388 &encoder_feedback_, | 391 &encoder_feedback_, |
| 389 bandwidth_observer_.get(), | 392 bandwidth_observer_.get(), |
| 390 congestion_controller_->GetTransportFeedbackObserver(), | 393 congestion_controller_->GetTransportFeedbackObserver(), |
| 391 call_stats_->rtcp_rtt_stats(), | 394 call_stats_->rtcp_rtt_stats(), |
| 392 congestion_controller_->pacer(), | 395 congestion_controller_->pacer(), |
| 393 congestion_controller_->packet_router(), | 396 congestion_controller_->packet_router(), |
| 394 &stats_proxy_, | 397 &stats_proxy_, |
| 395 send_delay_stats, | 398 send_delay_stats, |
| 396 config_.rtp.ssrcs.size())), | 399 config_.rtp.ssrcs.size())), |
| 397 payload_router_(rtp_rtcp_modules_, config.encoder_settings.payload_type), | 400 payload_router_(rtp_rtcp_modules_, config.encoder_settings.payload_type), |
| 398 input_(&encoder_wakeup_event_, | 401 input_(&encoder_wakeup_event_, |
| 399 config_.local_renderer, | 402 config_.local_renderer, |
| 400 &stats_proxy_, | 403 &stats_proxy_, |
| 401 &overuse_detector_) { | 404 &overuse_detector_) { |
| 402 LOG(LS_INFO) << "VideoSendStream: " << config_.ToString(); | 405 LOG(LS_INFO) << "VideoSendStream: " << config_.ToString(); |
| 403 | 406 |
| 404 RTC_DCHECK(!config_.rtp.ssrcs.empty()); | 407 RTC_DCHECK(!config_.rtp.ssrcs.empty()); |
| 405 RTC_DCHECK(module_process_thread_); | 408 RTC_DCHECK(module_process_thread_); |
| 406 RTC_DCHECK(call_stats_); | 409 RTC_DCHECK(call_stats_); |
| 407 RTC_DCHECK(congestion_controller_); | 410 RTC_DCHECK(congestion_controller_); |
| 408 RTC_DCHECK(remb_); | 411 RTC_DCHECK(remb_); |
| 409 | 412 |
| 410 encoder_feedback_.Init(config_.rtp.ssrcs, &vie_encoder_); | |
| 411 | 413 |
| 412 // RTP/RTCP initialization. | 414 // RTP/RTCP initialization. |
| 413 for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { | 415 for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { |
| 414 module_process_thread_->RegisterModule(rtp_rtcp); | 416 module_process_thread_->RegisterModule(rtp_rtcp); |
| 415 congestion_controller_->packet_router()->AddRtpModule(rtp_rtcp); | 417 congestion_controller_->packet_router()->AddRtpModule(rtp_rtcp); |
| 416 } | 418 } |
| 417 | 419 |
| 418 video_sender_->RegisterProtectionCallback(this); | 420 video_sender_->RegisterProtectionCallback(this); |
| 419 | 421 |
| 420 for (size_t i = 0; i < config_.rtp.extensions.size(); ++i) { | 422 for (size_t i = 0; i < config_.rtp.extensions.size(); ++i) { |
| (...skipping 139 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
| 560 encoder_settings->video_codec.startBitrate = | 562 encoder_settings->video_codec.startBitrate = |
| 561 bitrate_allocator_->AddObserver( | 563 bitrate_allocator_->AddObserver( |
| 562 this, encoder_settings->video_codec.minBitrate * 1000, | 564 this, encoder_settings->video_codec.minBitrate * 1000, |
| 563 encoder_settings->video_codec.maxBitrate * 1000) / | 565 encoder_settings->video_codec.maxBitrate * 1000) / |
| 564 1000; | 566 1000; |
| 565 | 567 |
| 566 payload_router_.SetSendStreams(encoder_settings->streams); | 568 payload_router_.SetSendStreams(encoder_settings->streams); |
| 567 vie_encoder_.SetEncoder(encoder_settings->video_codec, | 569 vie_encoder_.SetEncoder(encoder_settings->video_codec, |
| 568 encoder_settings->min_transmit_bitrate_bps, | 570 encoder_settings->min_transmit_bitrate_bps, |
| 569 payload_router_.MaxPayloadLength(), this); | 571 payload_router_.MaxPayloadLength(), this); |
| 572 |
| 573 // Clear stats for disabled layers. |
| 574 for (size_t i = encoder_settings->streams.size(); |
| 575 i < config_.rtp.ssrcs.size(); ++i) { |
| 576 stats_proxy_.OnInactiveSsrc(config_.rtp.ssrcs[i]); |
| 577 } |
| 578 |
| 570 if (config_.suspend_below_min_bitrate) { | 579 if (config_.suspend_below_min_bitrate) { |
| 571 video_sender_->SuspendBelowMinBitrate(); | 580 video_sender_->SuspendBelowMinBitrate(); |
| 572 bitrate_allocator_->EnforceMinBitrate(false); | 581 bitrate_allocator_->EnforceMinBitrate(false); |
| 573 } | 582 } |
| 574 // We might've gotten new settings while configuring the encoder settings, | 583 // 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 | 584 // 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). | 585 // a frame (which might correspond to the last frame size). |
| 577 encoder_wakeup_event_.Set(); | 586 encoder_wakeup_event_.Set(); |
| 578 continue; | 587 continue; |
| 579 } | 588 } |
| (...skipping 40 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
| 620 if (config_.overuse_callback) | 629 if (config_.overuse_callback) |
| 621 config_.overuse_callback->OnLoadUpdate(LoadObserver::kUnderuse); | 630 config_.overuse_callback->OnLoadUpdate(LoadObserver::kUnderuse); |
| 622 } | 631 } |
| 623 | 632 |
| 624 int32_t VideoSendStream::Encoded(const EncodedImage& encoded_image, | 633 int32_t VideoSendStream::Encoded(const EncodedImage& encoded_image, |
| 625 const CodecSpecificInfo* codec_specific_info, | 634 const CodecSpecificInfo* codec_specific_info, |
| 626 const RTPFragmentationHeader* fragmentation) { | 635 const RTPFragmentationHeader* fragmentation) { |
| 627 // |encoded_frame_proxy_| forwards frames to |config_.post_encode_callback|; | 636 // |encoded_frame_proxy_| forwards frames to |config_.post_encode_callback|; |
| 628 encoded_frame_proxy_.Encoded(encoded_image, codec_specific_info, | 637 encoded_frame_proxy_.Encoded(encoded_image, codec_specific_info, |
| 629 fragmentation); | 638 fragmentation); |
| 630 return payload_router_.Encoded(encoded_image, codec_specific_info, | 639 int32_t return_value = payload_router_.Encoded( |
| 631 fragmentation); | 640 encoded_image, codec_specific_info, fragmentation); |
| 641 |
| 642 if (kEnableFrameRecording) { |
| 643 int layer = codec_specific_info->codecType == kVideoCodecVP8 |
| 644 ? codec_specific_info->codecSpecific.VP8.simulcastIdx |
| 645 : 0; |
| 646 IvfFileWriter* file_writer; |
| 647 { |
| 648 if (file_writers_[layer] == nullptr) { |
| 649 std::ostringstream oss; |
| 650 oss << "send_bitstream_ssrc"; |
| 651 for (uint32_t ssrc : config_.rtp.ssrcs) |
| 652 oss << "_" << ssrc; |
| 653 oss << "_layer" << layer << ".ivf"; |
| 654 file_writers_[layer] = |
| 655 IvfFileWriter::Open(oss.str(), codec_specific_info->codecType); |
| 656 } |
| 657 file_writer = file_writers_[layer].get(); |
| 658 } |
| 659 if (file_writer) { |
| 660 bool ok = file_writer->WriteFrame(encoded_image); |
| 661 RTC_DCHECK(ok); |
| 662 } |
| 663 } |
| 664 |
| 665 return return_value; |
| 632 } | 666 } |
| 633 | 667 |
| 634 void VideoSendStream::ConfigureProtection() { | 668 void VideoSendStream::ConfigureProtection() { |
| 635 // Enable NACK, FEC or both. | 669 // Enable NACK, FEC or both. |
| 636 const bool enable_protection_nack = config_.rtp.nack.rtp_history_ms > 0; | 670 const bool enable_protection_nack = config_.rtp.nack.rtp_history_ms > 0; |
| 637 bool enable_protection_fec = config_.rtp.fec.red_payload_type != -1; | 671 bool enable_protection_fec = config_.rtp.fec.red_payload_type != -1; |
| 638 // Payload types without picture ID cannot determine that a stream is complete | 672 // Payload types without picture ID cannot determine that a stream is complete |
| 639 // without retransmitting FEC, so using FEC + NACK for H.264 (for instance) is | 673 // without retransmitting FEC, so using FEC + NACK for H.264 (for instance) is |
| 640 // a waste of bandwidth since FEC packets still have to be transmitted. Note | 674 // a waste of bandwidth since FEC packets still have to be transmitted. Note |
| 641 // that this is not the case with FLEXFEC. | 675 // that this is not the case with FLEXFEC. |
| (...skipping 132 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
| 774 &module_nack_rate); | 808 &module_nack_rate); |
| 775 *sent_video_rate_bps += module_video_rate; | 809 *sent_video_rate_bps += module_video_rate; |
| 776 *sent_nack_rate_bps += module_nack_rate; | 810 *sent_nack_rate_bps += module_nack_rate; |
| 777 *sent_fec_rate_bps += module_fec_rate; | 811 *sent_fec_rate_bps += module_fec_rate; |
| 778 } | 812 } |
| 779 return 0; | 813 return 0; |
| 780 } | 814 } |
| 781 | 815 |
| 782 } // namespace internal | 816 } // namespace internal |
| 783 } // namespace webrtc | 817 } // namespace webrtc |
| OLD | NEW |