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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/video_coding/utility/simulcast_state.h" |
| 26 #include "webrtc/modules/utility/include/process_thread.h" | 27 #include "webrtc/modules/utility/include/process_thread.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; |
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| 597 LOG(LS_INFO) << "(Re)configureVideoEncoder: " << config.ToString(); | 598 LOG(LS_INFO) << "(Re)configureVideoEncoder: " << config.ToString(); |
| 598 RTC_DCHECK_GE(config_.rtp.ssrcs.size(), config.streams.size()); | 599 RTC_DCHECK_GE(config_.rtp.ssrcs.size(), config.streams.size()); |
| 599 VideoCodec video_codec = VideoEncoderConfigToVideoCodec( | 600 VideoCodec video_codec = VideoEncoderConfigToVideoCodec( |
| 600 config, config_.encoder_settings.payload_name, | 601 config, config_.encoder_settings.payload_name, |
| 601 config_.encoder_settings.payload_type); | 602 config_.encoder_settings.payload_type); |
| 602 { | 603 { |
| 603 rtc::CritScope lock(&encoder_settings_crit_); | 604 rtc::CritScope lock(&encoder_settings_crit_); |
| 604 pending_encoder_settings_ = rtc::Optional<EncoderSettings>( | 605 pending_encoder_settings_ = rtc::Optional<EncoderSettings>( |
| 605 {video_codec, config.min_transmit_bitrate_bps, config.streams}); | 606 {video_codec, config.min_transmit_bitrate_bps, config.streams}); |
| 606 } | 607 } |
| 608 payload_router_.UpdateSimulcastState(SimulcastState(video_codec)); |
| 607 encoder_wakeup_event_.Set(); | 609 encoder_wakeup_event_.Set(); |
| 608 } | 610 } |
| 609 | 611 |
| 610 VideoSendStream::Stats VideoSendStream::GetStats() { | 612 VideoSendStream::Stats VideoSendStream::GetStats() { |
| 611 return stats_proxy_.GetStats(); | 613 return stats_proxy_.GetStats(); |
| 612 } | 614 } |
| 613 | 615 |
| 614 void VideoSendStream::OveruseDetected() { | 616 void VideoSendStream::OveruseDetected() { |
| 615 if (config_.overuse_callback) | 617 if (config_.overuse_callback) |
| 616 config_.overuse_callback->OnLoadUpdate(LoadObserver::kOveruse); | 618 config_.overuse_callback->OnLoadUpdate(LoadObserver::kOveruse); |
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| 774 &module_nack_rate); | 776 &module_nack_rate); |
| 775 *sent_video_rate_bps += module_video_rate; | 777 *sent_video_rate_bps += module_video_rate; |
| 776 *sent_nack_rate_bps += module_nack_rate; | 778 *sent_nack_rate_bps += module_nack_rate; |
| 777 *sent_fec_rate_bps += module_fec_rate; | 779 *sent_fec_rate_bps += module_fec_rate; |
| 778 } | 780 } |
| 779 return 0; | 781 return 0; |
| 780 } | 782 } |
| 781 | 783 |
| 782 } // namespace internal | 784 } // namespace internal |
| 783 } // namespace webrtc | 785 } // namespace webrtc |
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