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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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