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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 #include "webrtc/video/video_send_stream.h" | 10 #include "webrtc/video/video_send_stream.h" |
11 | 11 |
12 #include <algorithm> | 12 #include <algorithm> |
| 13 #include <cmath> |
13 #include <sstream> | 14 #include <sstream> |
14 #include <string> | 15 #include <string> |
15 #include <utility> | 16 #include <utility> |
16 #include <vector> | 17 #include <vector> |
17 | 18 |
18 #include "webrtc/common_types.h" | 19 #include "webrtc/common_types.h" |
19 #include "webrtc/base/checks.h" | 20 #include "webrtc/base/checks.h" |
20 #include "webrtc/base/file.h" | 21 #include "webrtc/base/file.h" |
21 #include "webrtc/base/logging.h" | 22 #include "webrtc/base/logging.h" |
22 #include "webrtc/base/trace_event.h" | 23 #include "webrtc/base/trace_event.h" |
23 #include "webrtc/base/weak_ptr.h" | 24 #include "webrtc/base/weak_ptr.h" |
24 #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h" | 25 #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h" |
25 #include "webrtc/modules/congestion_controller/include/congestion_controller.h" | 26 #include "webrtc/modules/congestion_controller/include/congestion_controller.h" |
26 #include "webrtc/modules/pacing/packet_router.h" | 27 #include "webrtc/modules/pacing/packet_router.h" |
27 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" | 28 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" |
28 #include "webrtc/modules/utility/include/process_thread.h" | 29 #include "webrtc/modules/utility/include/process_thread.h" |
29 #include "webrtc/modules/video_coding/utility/ivf_file_writer.h" | 30 #include "webrtc/modules/video_coding/utility/ivf_file_writer.h" |
| 31 #include "webrtc/system_wrappers/include/field_trial.h" |
30 #include "webrtc/video/call_stats.h" | 32 #include "webrtc/video/call_stats.h" |
31 #include "webrtc/video/vie_remb.h" | 33 #include "webrtc/video/vie_remb.h" |
32 #include "webrtc/video_send_stream.h" | 34 #include "webrtc/video_send_stream.h" |
33 | 35 |
34 namespace webrtc { | 36 namespace webrtc { |
35 | 37 |
36 static const int kMinSendSidePacketHistorySize = 600; | 38 static const int kMinSendSidePacketHistorySize = 600; |
37 namespace { | 39 namespace { |
38 | 40 |
39 std::vector<RtpRtcp*> CreateRtpRtcpModules( | 41 std::vector<RtpRtcp*> CreateRtpRtcpModules( |
40 Transport* outgoing_transport, | 42 Transport* outgoing_transport, |
41 RtcpIntraFrameObserver* intra_frame_callback, | 43 RtcpIntraFrameObserver* intra_frame_callback, |
42 RtcpBandwidthObserver* bandwidth_callback, | 44 RtcpBandwidthObserver* bandwidth_callback, |
43 TransportFeedbackObserver* transport_feedback_callback, | 45 TransportFeedbackObserver* transport_feedback_callback, |
44 RtcpRttStats* rtt_stats, | 46 RtcpRttStats* rtt_stats, |
45 RtpPacketSender* paced_sender, | 47 RtpPacketSender* paced_sender, |
46 TransportSequenceNumberAllocator* transport_sequence_number_allocator, | 48 TransportSequenceNumberAllocator* transport_sequence_number_allocator, |
47 FlexfecSender* flexfec_sender, | 49 FlexfecSender* flexfec_sender, |
48 SendStatisticsProxy* stats_proxy, | 50 SendStatisticsProxy* stats_proxy, |
49 SendDelayStats* send_delay_stats, | 51 SendDelayStats* send_delay_stats, |
50 RtcEventLog* event_log, | 52 RtcEventLog* event_log, |
51 RateLimiter* retransmission_rate_limiter, | 53 RateLimiter* retransmission_rate_limiter, |
| 54 OverheadObserver* overhead_observer, |
52 size_t num_modules) { | 55 size_t num_modules) { |
53 RTC_DCHECK_GT(num_modules, 0); | 56 RTC_DCHECK_GT(num_modules, 0); |
54 RtpRtcp::Configuration configuration; | 57 RtpRtcp::Configuration configuration; |
55 ReceiveStatistics* null_receive_statistics = configuration.receive_statistics; | 58 ReceiveStatistics* null_receive_statistics = configuration.receive_statistics; |
56 configuration.audio = false; | 59 configuration.audio = false; |
57 configuration.receiver_only = false; | 60 configuration.receiver_only = false; |
58 configuration.flexfec_sender = flexfec_sender; | 61 configuration.flexfec_sender = flexfec_sender; |
59 configuration.receive_statistics = null_receive_statistics; | 62 configuration.receive_statistics = null_receive_statistics; |
60 configuration.outgoing_transport = outgoing_transport; | 63 configuration.outgoing_transport = outgoing_transport; |
61 configuration.intra_frame_callback = intra_frame_callback; | 64 configuration.intra_frame_callback = intra_frame_callback; |
62 configuration.bandwidth_callback = bandwidth_callback; | 65 configuration.bandwidth_callback = bandwidth_callback; |
63 configuration.transport_feedback_callback = transport_feedback_callback; | 66 configuration.transport_feedback_callback = transport_feedback_callback; |
64 configuration.rtt_stats = rtt_stats; | 67 configuration.rtt_stats = rtt_stats; |
65 configuration.rtcp_packet_type_counter_observer = stats_proxy; | 68 configuration.rtcp_packet_type_counter_observer = stats_proxy; |
66 configuration.paced_sender = paced_sender; | 69 configuration.paced_sender = paced_sender; |
67 configuration.transport_sequence_number_allocator = | 70 configuration.transport_sequence_number_allocator = |
68 transport_sequence_number_allocator; | 71 transport_sequence_number_allocator; |
69 configuration.send_bitrate_observer = stats_proxy; | 72 configuration.send_bitrate_observer = stats_proxy; |
70 configuration.send_frame_count_observer = stats_proxy; | 73 configuration.send_frame_count_observer = stats_proxy; |
71 configuration.send_side_delay_observer = stats_proxy; | 74 configuration.send_side_delay_observer = stats_proxy; |
72 configuration.send_packet_observer = send_delay_stats; | 75 configuration.send_packet_observer = send_delay_stats; |
73 configuration.event_log = event_log; | 76 configuration.event_log = event_log; |
74 configuration.retransmission_rate_limiter = retransmission_rate_limiter; | 77 configuration.retransmission_rate_limiter = retransmission_rate_limiter; |
75 | 78 configuration.overhead_observer = overhead_observer; |
76 std::vector<RtpRtcp*> modules; | 79 std::vector<RtpRtcp*> modules; |
77 for (size_t i = 0; i < num_modules; ++i) { | 80 for (size_t i = 0; i < num_modules; ++i) { |
78 RtpRtcp* rtp_rtcp = RtpRtcp::CreateRtpRtcp(configuration); | 81 RtpRtcp* rtp_rtcp = RtpRtcp::CreateRtpRtcp(configuration); |
79 rtp_rtcp->SetSendingStatus(false); | 82 rtp_rtcp->SetSendingStatus(false); |
80 rtp_rtcp->SetSendingMediaStatus(false); | 83 rtp_rtcp->SetSendingMediaStatus(false); |
81 rtp_rtcp->SetRTCPStatus(RtcpMode::kCompound); | 84 rtp_rtcp->SetRTCPStatus(RtcpMode::kCompound); |
82 modules.push_back(rtp_rtcp); | 85 modules.push_back(rtp_rtcp); |
83 } | 86 } |
84 return modules; | 87 return modules; |
85 } | 88 } |
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277 namespace internal { | 280 namespace internal { |
278 | 281 |
279 // VideoSendStreamImpl implements internal::VideoSendStream. | 282 // VideoSendStreamImpl implements internal::VideoSendStream. |
280 // It is created and destroyed on |worker_queue|. The intent is to decrease the | 283 // It is created and destroyed on |worker_queue|. The intent is to decrease the |
281 // need for locking and to ensure methods are called in sequence. | 284 // need for locking and to ensure methods are called in sequence. |
282 // Public methods except |DeliverRtcp| must be called on |worker_queue|. | 285 // Public methods except |DeliverRtcp| must be called on |worker_queue|. |
283 // DeliverRtcp is called on the libjingle worker thread or a network thread. | 286 // DeliverRtcp is called on the libjingle worker thread or a network thread. |
284 // An encoder may deliver frames through the EncodedImageCallback on an | 287 // An encoder may deliver frames through the EncodedImageCallback on an |
285 // arbitrary thread. | 288 // arbitrary thread. |
286 class VideoSendStreamImpl : public webrtc::BitrateAllocatorObserver, | 289 class VideoSendStreamImpl : public webrtc::BitrateAllocatorObserver, |
| 290 public webrtc::OverheadObserver, |
287 public webrtc::VCMProtectionCallback, | 291 public webrtc::VCMProtectionCallback, |
288 public ViEEncoder::EncoderSink { | 292 public ViEEncoder::EncoderSink { |
289 public: | 293 public: |
290 VideoSendStreamImpl(SendStatisticsProxy* stats_proxy, | 294 VideoSendStreamImpl(SendStatisticsProxy* stats_proxy, |
291 rtc::TaskQueue* worker_queue, | 295 rtc::TaskQueue* worker_queue, |
292 CallStats* call_stats, | 296 CallStats* call_stats, |
293 CongestionController* congestion_controller, | 297 CongestionController* congestion_controller, |
294 BitrateAllocator* bitrate_allocator, | 298 BitrateAllocator* bitrate_allocator, |
295 SendDelayStats* send_delay_stats, | 299 SendDelayStats* send_delay_stats, |
296 VieRemb* remb, | 300 VieRemb* remb, |
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330 uint8_t fraction_loss, | 334 uint8_t fraction_loss, |
331 int64_t rtt) override; | 335 int64_t rtt) override; |
332 | 336 |
333 // Implements webrtc::VCMProtectionCallback. | 337 // Implements webrtc::VCMProtectionCallback. |
334 int ProtectionRequest(const FecProtectionParams* delta_params, | 338 int ProtectionRequest(const FecProtectionParams* delta_params, |
335 const FecProtectionParams* key_params, | 339 const FecProtectionParams* key_params, |
336 uint32_t* sent_video_rate_bps, | 340 uint32_t* sent_video_rate_bps, |
337 uint32_t* sent_nack_rate_bps, | 341 uint32_t* sent_nack_rate_bps, |
338 uint32_t* sent_fec_rate_bps) override; | 342 uint32_t* sent_fec_rate_bps) override; |
339 | 343 |
| 344 // Implements OverheadObserver. |
| 345 void OnOverheadChanged(size_t overhead_bytes_per_packet) override; |
| 346 |
340 void OnEncoderConfigurationChanged(std::vector<VideoStream> streams, | 347 void OnEncoderConfigurationChanged(std::vector<VideoStream> streams, |
341 int min_transmit_bitrate_bps) override; | 348 int min_transmit_bitrate_bps) override; |
342 | 349 |
343 // Implements EncodedImageCallback. The implementation routes encoded frames | 350 // Implements EncodedImageCallback. The implementation routes encoded frames |
344 // to the |payload_router_| and |config.pre_encode_callback| if set. | 351 // to the |payload_router_| and |config.pre_encode_callback| if set. |
345 // Called on an arbitrary encoder callback thread. | 352 // Called on an arbitrary encoder callback thread. |
346 EncodedImageCallback::Result OnEncodedImage( | 353 EncodedImageCallback::Result OnEncodedImage( |
347 const EncodedImage& encoded_image, | 354 const EncodedImage& encoded_image, |
348 const CodecSpecificInfo* codec_specific_info, | 355 const CodecSpecificInfo* codec_specific_info, |
349 const RTPFragmentationHeader* fragmentation) override; | 356 const RTPFragmentationHeader* fragmentation) override; |
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391 const std::vector<RtpRtcp*> rtp_rtcp_modules_; | 398 const std::vector<RtpRtcp*> rtp_rtcp_modules_; |
392 PayloadRouter payload_router_; | 399 PayloadRouter payload_router_; |
393 | 400 |
394 // |weak_ptr_| to our self. This is used since we can not call | 401 // |weak_ptr_| to our self. This is used since we can not call |
395 // |weak_ptr_factory_.GetWeakPtr| from multiple sequences but it is ok to copy | 402 // |weak_ptr_factory_.GetWeakPtr| from multiple sequences but it is ok to copy |
396 // an existing WeakPtr. | 403 // an existing WeakPtr. |
397 rtc::WeakPtr<VideoSendStreamImpl> weak_ptr_; | 404 rtc::WeakPtr<VideoSendStreamImpl> weak_ptr_; |
398 // |weak_ptr_factory_| must be declared last to make sure all WeakPtr's are | 405 // |weak_ptr_factory_| must be declared last to make sure all WeakPtr's are |
399 // invalidated before any other members are destroyed. | 406 // invalidated before any other members are destroyed. |
400 rtc::WeakPtrFactory<VideoSendStreamImpl> weak_ptr_factory_; | 407 rtc::WeakPtrFactory<VideoSendStreamImpl> weak_ptr_factory_; |
| 408 |
| 409 rtc::CriticalSection overhead_bytes_per_packet_crit_; |
| 410 size_t overhead_bytes_per_packet_ GUARDED_BY(overhead_bytes_per_packet_crit_); |
| 411 size_t transport_overhead_bytes_per_packet_; |
401 }; | 412 }; |
402 | 413 |
403 // TODO(tommi): See if there's a more elegant way to create a task that creates | 414 // TODO(tommi): See if there's a more elegant way to create a task that creates |
404 // an object on the correct task queue. | 415 // an object on the correct task queue. |
405 class VideoSendStream::ConstructionTask : public rtc::QueuedTask { | 416 class VideoSendStream::ConstructionTask : public rtc::QueuedTask { |
406 public: | 417 public: |
407 ConstructionTask(std::unique_ptr<VideoSendStreamImpl>* send_stream, | 418 ConstructionTask(std::unique_ptr<VideoSendStreamImpl>* send_stream, |
408 rtc::Event* done_event, | 419 rtc::Event* done_event, |
409 SendStatisticsProxy* stats_proxy, | 420 SendStatisticsProxy* stats_proxy, |
410 ViEEncoder* vie_encoder, | 421 ViEEncoder* vie_encoder, |
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727 bandwidth_observer_.get(), | 738 bandwidth_observer_.get(), |
728 congestion_controller_->GetTransportFeedbackObserver(), | 739 congestion_controller_->GetTransportFeedbackObserver(), |
729 call_stats_->rtcp_rtt_stats(), | 740 call_stats_->rtcp_rtt_stats(), |
730 congestion_controller_->pacer(), | 741 congestion_controller_->pacer(), |
731 congestion_controller_->packet_router(), | 742 congestion_controller_->packet_router(), |
732 flexfec_sender_.get(), | 743 flexfec_sender_.get(), |
733 stats_proxy_, | 744 stats_proxy_, |
734 send_delay_stats, | 745 send_delay_stats, |
735 event_log, | 746 event_log, |
736 congestion_controller_->GetRetransmissionRateLimiter(), | 747 congestion_controller_->GetRetransmissionRateLimiter(), |
| 748 this, |
737 config_->rtp.ssrcs.size())), | 749 config_->rtp.ssrcs.size())), |
738 payload_router_(rtp_rtcp_modules_, | 750 payload_router_(rtp_rtcp_modules_, |
739 config_->encoder_settings.payload_type), | 751 config_->encoder_settings.payload_type), |
740 weak_ptr_factory_(this) { | 752 weak_ptr_factory_(this), |
| 753 overhead_bytes_per_packet_(0), |
| 754 transport_overhead_bytes_per_packet_(0) { |
741 RTC_DCHECK_RUN_ON(worker_queue_); | 755 RTC_DCHECK_RUN_ON(worker_queue_); |
742 LOG(LS_INFO) << "VideoSendStreamInternal: " << config_->ToString(); | 756 LOG(LS_INFO) << "VideoSendStreamInternal: " << config_->ToString(); |
743 weak_ptr_ = weak_ptr_factory_.GetWeakPtr(); | 757 weak_ptr_ = weak_ptr_factory_.GetWeakPtr(); |
744 module_process_thread_checker_.DetachFromThread(); | 758 module_process_thread_checker_.DetachFromThread(); |
745 | 759 |
746 RTC_DCHECK(!config_->rtp.ssrcs.empty()); | 760 RTC_DCHECK(!config_->rtp.ssrcs.empty()); |
747 RTC_DCHECK(call_stats_); | 761 RTC_DCHECK(call_stats_); |
748 RTC_DCHECK(congestion_controller_); | 762 RTC_DCHECK(congestion_controller_); |
749 RTC_DCHECK(remb_); | 763 RTC_DCHECK(remb_); |
750 | 764 |
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770 } | 784 } |
771 | 785 |
772 remb_->AddRembSender(rtp_rtcp_modules_[0]); | 786 remb_->AddRembSender(rtp_rtcp_modules_[0]); |
773 rtp_rtcp_modules_[0]->SetREMBStatus(true); | 787 rtp_rtcp_modules_[0]->SetREMBStatus(true); |
774 | 788 |
775 ConfigureProtection(); | 789 ConfigureProtection(); |
776 ConfigureSsrcs(); | 790 ConfigureSsrcs(); |
777 | 791 |
778 // TODO(pbos): Should we set CNAME on all RTP modules? | 792 // TODO(pbos): Should we set CNAME on all RTP modules? |
779 rtp_rtcp_modules_.front()->SetCNAME(config_->rtp.c_name.c_str()); | 793 rtp_rtcp_modules_.front()->SetCNAME(config_->rtp.c_name.c_str()); |
780 // 28 to match packet overhead in ModuleRtpRtcpImpl. | 794 |
781 static const size_t kRtpPacketSizeOverhead = 28; | |
782 RTC_DCHECK_LE(config_->rtp.max_packet_size, 0xFFFFu + kRtpPacketSizeOverhead); | |
783 const uint16_t mtu = static_cast<uint16_t>(config_->rtp.max_packet_size + | |
784 kRtpPacketSizeOverhead); | |
785 for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { | 795 for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { |
786 rtp_rtcp->RegisterRtcpStatisticsCallback(stats_proxy_); | 796 rtp_rtcp->RegisterRtcpStatisticsCallback(stats_proxy_); |
787 rtp_rtcp->RegisterSendChannelRtpStatisticsCallback(stats_proxy_); | 797 rtp_rtcp->RegisterSendChannelRtpStatisticsCallback(stats_proxy_); |
788 rtp_rtcp->SetMaxTransferUnit(mtu); | 798 rtp_rtcp->SetMaxTransferUnit( |
| 799 static_cast<uint16_t>(config_->rtp.max_packet_size)); |
789 rtp_rtcp->RegisterVideoSendPayload( | 800 rtp_rtcp->RegisterVideoSendPayload( |
790 config_->encoder_settings.payload_type, | 801 config_->encoder_settings.payload_type, |
791 config_->encoder_settings.payload_name.c_str()); | 802 config_->encoder_settings.payload_name.c_str()); |
792 } | 803 } |
793 | 804 |
794 RTC_DCHECK(config_->encoder_settings.encoder); | 805 RTC_DCHECK(config_->encoder_settings.encoder); |
795 RTC_DCHECK_GE(config_->encoder_settings.payload_type, 0); | 806 RTC_DCHECK_GE(config_->encoder_settings.payload_type, 0); |
796 RTC_DCHECK_LE(config_->encoder_settings.payload_type, 127); | 807 RTC_DCHECK_LE(config_->encoder_settings.payload_type, 127); |
797 | 808 |
798 vie_encoder_->SetStartBitrate(bitrate_allocator_->GetStartBitrate(this)); | 809 vie_encoder_->SetStartBitrate(bitrate_allocator_->GetStartBitrate(this)); |
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1148 : RtcpMode::kOff); | 1159 : RtcpMode::kOff); |
1149 } | 1160 } |
1150 } | 1161 } |
1151 | 1162 |
1152 uint32_t VideoSendStreamImpl::OnBitrateUpdated(uint32_t bitrate_bps, | 1163 uint32_t VideoSendStreamImpl::OnBitrateUpdated(uint32_t bitrate_bps, |
1153 uint8_t fraction_loss, | 1164 uint8_t fraction_loss, |
1154 int64_t rtt) { | 1165 int64_t rtt) { |
1155 RTC_DCHECK_RUN_ON(worker_queue_); | 1166 RTC_DCHECK_RUN_ON(worker_queue_); |
1156 RTC_DCHECK(payload_router_.active()) | 1167 RTC_DCHECK(payload_router_.active()) |
1157 << "VideoSendStream::Start has not been called."; | 1168 << "VideoSendStream::Start has not been called."; |
| 1169 |
| 1170 if (webrtc::field_trial::FindFullName("WebRTC-SendSideBwe-WithOverhead") == |
| 1171 "Enabled") { |
| 1172 // Substract overhead from bitrate. |
| 1173 rtc::CritScope lock(&overhead_bytes_per_packet_crit_); |
| 1174 int packets_per_second = |
| 1175 std::ceil(bitrate_bps / (8 * (config_->rtp.max_packet_size + |
| 1176 transport_overhead_bytes_per_packet_))); |
| 1177 uint32_t overhead_bps = static_cast<uint32_t>( |
| 1178 8 * overhead_bytes_per_packet_ * packets_per_second); |
| 1179 bitrate_bps = overhead_bps > bitrate_bps ? 0u : bitrate_bps - overhead_bps; |
| 1180 } |
| 1181 |
1158 // Get the encoder target rate. It is the estimated network rate - | 1182 // Get the encoder target rate. It is the estimated network rate - |
1159 // protection overhead. | 1183 // protection overhead. |
1160 encoder_target_rate_bps_ = protection_bitrate_calculator_.SetTargetRates( | 1184 encoder_target_rate_bps_ = protection_bitrate_calculator_.SetTargetRates( |
1161 bitrate_bps, stats_proxy_->GetSendFrameRate(), fraction_loss, rtt); | 1185 bitrate_bps, stats_proxy_->GetSendFrameRate(), fraction_loss, rtt); |
1162 uint32_t protection_bitrate = bitrate_bps - encoder_target_rate_bps_; | 1186 uint32_t protection_bitrate = bitrate_bps - encoder_target_rate_bps_; |
1163 | 1187 |
1164 encoder_target_rate_bps_ = | 1188 encoder_target_rate_bps_ = |
1165 std::min(encoder_max_bitrate_bps_, encoder_target_rate_bps_); | 1189 std::min(encoder_max_bitrate_bps_, encoder_target_rate_bps_); |
1166 vie_encoder_->OnBitrateUpdated(encoder_target_rate_bps_, fraction_loss, rtt); | 1190 vie_encoder_->OnBitrateUpdated(encoder_target_rate_bps_, fraction_loss, rtt); |
1167 stats_proxy_->OnSetEncoderTargetRate(encoder_target_rate_bps_); | 1191 stats_proxy_->OnSetEncoderTargetRate(encoder_target_rate_bps_); |
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1207 rtp_rtcp->SetFecParameters(*delta_params, *key_params); | 1231 rtp_rtcp->SetFecParameters(*delta_params, *key_params); |
1208 rtp_rtcp->BitrateSent(¬_used, &module_video_rate, &module_fec_rate, | 1232 rtp_rtcp->BitrateSent(¬_used, &module_video_rate, &module_fec_rate, |
1209 &module_nack_rate); | 1233 &module_nack_rate); |
1210 *sent_video_rate_bps += module_video_rate; | 1234 *sent_video_rate_bps += module_video_rate; |
1211 *sent_nack_rate_bps += module_nack_rate; | 1235 *sent_nack_rate_bps += module_nack_rate; |
1212 *sent_fec_rate_bps += module_fec_rate; | 1236 *sent_fec_rate_bps += module_fec_rate; |
1213 } | 1237 } |
1214 return 0; | 1238 return 0; |
1215 } | 1239 } |
1216 | 1240 |
| 1241 void VideoSendStreamImpl::OnOverheadChanged(size_t overhead_bytes_per_packet) { |
| 1242 rtc::CritScope lock(&overhead_bytes_per_packet_crit_); |
| 1243 overhead_bytes_per_packet_ = overhead_bytes_per_packet; |
| 1244 } |
| 1245 |
1217 void VideoSendStreamImpl::SetTransportOverhead( | 1246 void VideoSendStreamImpl::SetTransportOverhead( |
1218 int transport_overhead_per_packet) { | 1247 int transport_overhead_bytes_per_packet) { |
1219 for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) | 1248 transport_overhead_bytes_per_packet_ = transport_overhead_bytes_per_packet; |
1220 rtp_rtcp->SetTransportOverhead(transport_overhead_per_packet); | 1249 RTC_DCHECK_LE(config_->rtp.max_packet_size, |
| 1250 0xFFFFu + transport_overhead_bytes_per_packet); |
| 1251 const uint16_t mtu = static_cast<uint16_t>( |
| 1252 config_->rtp.max_packet_size + transport_overhead_bytes_per_packet); |
| 1253 for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { |
| 1254 rtp_rtcp->SetTransportOverhead(transport_overhead_bytes_per_packet); |
| 1255 rtp_rtcp->SetMaxTransferUnit(mtu); |
| 1256 } |
1221 } | 1257 } |
1222 | 1258 |
1223 } // namespace internal | 1259 } // namespace internal |
1224 } // namespace webrtc | 1260 } // namespace webrtc |
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