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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 <string.h> | 11 #include <string.h> |
12 #include <algorithm> | 12 #include <algorithm> |
13 #include <map> | 13 #include <map> |
14 #include <memory> | 14 #include <memory> |
15 #include <set> | 15 #include <set> |
16 #include <utility> | 16 #include <utility> |
17 #include <vector> | 17 #include <vector> |
18 | 18 |
19 #include "webrtc/api/optional.h" | 19 #include "webrtc/api/optional.h" |
20 #include "webrtc/audio/audio_receive_stream.h" | 20 #include "webrtc/audio/audio_receive_stream.h" |
21 #include "webrtc/audio/audio_send_stream.h" | 21 #include "webrtc/audio/audio_send_stream.h" |
22 #include "webrtc/audio/audio_state.h" | 22 #include "webrtc/audio/audio_state.h" |
23 #include "webrtc/audio/scoped_voe_interface.h" | 23 #include "webrtc/audio/scoped_voe_interface.h" |
24 #include "webrtc/audio/time_interval.h" | 24 #include "webrtc/audio/time_interval.h" |
25 #include "webrtc/call/bitrate_allocator.h" | 25 #include "webrtc/call/bitrate_allocator.h" |
26 #include "webrtc/call/call.h" | 26 #include "webrtc/call/call.h" |
27 #include "webrtc/call/flexfec_receive_stream_impl.h" | 27 #include "webrtc/call/flexfec_receive_stream_impl.h" |
28 #include "webrtc/call/rtp_stream_receiver_controller.h" | 28 #include "webrtc/call/rtp_stream_receiver_controller.h" |
29 #include "webrtc/call/rtp_transport_controller_send.h" | 29 #include "webrtc/call/rtp_transport_controller_send.h" |
| 30 #include "webrtc/logging/rtc_event_log/events/rtc_event_audio_receive_stream_con
fig.h" |
| 31 #include "webrtc/logging/rtc_event_log/events/rtc_event_audio_send_stream_config
.h" |
| 32 #include "webrtc/logging/rtc_event_log/events/rtc_event_rtcp_packet_incoming.h" |
| 33 #include "webrtc/logging/rtc_event_log/events/rtc_event_rtp_packet_incoming.h" |
| 34 #include "webrtc/logging/rtc_event_log/events/rtc_event_video_receive_stream_con
fig.h" |
| 35 #include "webrtc/logging/rtc_event_log/events/rtc_event_video_send_stream_config
.h" |
30 #include "webrtc/logging/rtc_event_log/rtc_event_log.h" | 36 #include "webrtc/logging/rtc_event_log/rtc_event_log.h" |
31 #include "webrtc/logging/rtc_event_log/rtc_stream_config.h" | 37 #include "webrtc/logging/rtc_event_log/rtc_stream_config.h" |
32 #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h" | 38 #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h" |
33 #include "webrtc/modules/congestion_controller/include/receive_side_congestion_c
ontroller.h" | 39 #include "webrtc/modules/congestion_controller/include/receive_side_congestion_c
ontroller.h" |
34 #include "webrtc/modules/rtp_rtcp/include/flexfec_receiver.h" | 40 #include "webrtc/modules/rtp_rtcp/include/flexfec_receiver.h" |
35 #include "webrtc/modules/rtp_rtcp/include/rtp_header_extension_map.h" | 41 #include "webrtc/modules/rtp_rtcp/include/rtp_header_extension_map.h" |
36 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" | 42 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" |
37 #include "webrtc/modules/rtp_rtcp/source/byte_io.h" | 43 #include "webrtc/modules/rtp_rtcp/source/byte_io.h" |
38 #include "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h" | 44 #include "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h" |
39 #include "webrtc/modules/utility/include/process_thread.h" | 45 #include "webrtc/modules/utility/include/process_thread.h" |
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602 | 608 |
603 PacketReceiver* Call::Receiver() { | 609 PacketReceiver* Call::Receiver() { |
604 RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_); | 610 RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_); |
605 return this; | 611 return this; |
606 } | 612 } |
607 | 613 |
608 webrtc::AudioSendStream* Call::CreateAudioSendStream( | 614 webrtc::AudioSendStream* Call::CreateAudioSendStream( |
609 const webrtc::AudioSendStream::Config& config) { | 615 const webrtc::AudioSendStream::Config& config) { |
610 TRACE_EVENT0("webrtc", "Call::CreateAudioSendStream"); | 616 TRACE_EVENT0("webrtc", "Call::CreateAudioSendStream"); |
611 RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_); | 617 RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_); |
612 event_log_->LogAudioSendStreamConfig(*CreateRtcLogStreamConfig(config)); | 618 event_log_->Log(rtc::MakeUnique<RtcEventAudioSendStreamConfig>( |
| 619 CreateRtcLogStreamConfig(config))); |
613 | 620 |
614 rtc::Optional<RtpState> suspended_rtp_state; | 621 rtc::Optional<RtpState> suspended_rtp_state; |
615 { | 622 { |
616 const auto& iter = suspended_audio_send_ssrcs_.find(config.rtp.ssrc); | 623 const auto& iter = suspended_audio_send_ssrcs_.find(config.rtp.ssrc); |
617 if (iter != suspended_audio_send_ssrcs_.end()) { | 624 if (iter != suspended_audio_send_ssrcs_.end()) { |
618 suspended_rtp_state.emplace(iter->second); | 625 suspended_rtp_state.emplace(iter->second); |
619 } | 626 } |
620 } | 627 } |
621 | 628 |
622 AudioSendStream* send_stream = new AudioSendStream( | 629 AudioSendStream* send_stream = new AudioSendStream( |
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668 } | 675 } |
669 UpdateAggregateNetworkState(); | 676 UpdateAggregateNetworkState(); |
670 sent_rtp_audio_timer_ms_.Extend(audio_send_stream->GetActiveLifetime()); | 677 sent_rtp_audio_timer_ms_.Extend(audio_send_stream->GetActiveLifetime()); |
671 delete send_stream; | 678 delete send_stream; |
672 } | 679 } |
673 | 680 |
674 webrtc::AudioReceiveStream* Call::CreateAudioReceiveStream( | 681 webrtc::AudioReceiveStream* Call::CreateAudioReceiveStream( |
675 const webrtc::AudioReceiveStream::Config& config) { | 682 const webrtc::AudioReceiveStream::Config& config) { |
676 TRACE_EVENT0("webrtc", "Call::CreateAudioReceiveStream"); | 683 TRACE_EVENT0("webrtc", "Call::CreateAudioReceiveStream"); |
677 RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_); | 684 RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_); |
678 event_log_->LogAudioReceiveStreamConfig(*CreateRtcLogStreamConfig(config)); | 685 event_log_->Log(rtc::MakeUnique<RtcEventAudioReceiveStreamConfig>( |
| 686 CreateRtcLogStreamConfig(config))); |
679 AudioReceiveStream* receive_stream = new AudioReceiveStream( | 687 AudioReceiveStream* receive_stream = new AudioReceiveStream( |
680 &audio_receiver_controller_, transport_send_->packet_router(), config, | 688 &audio_receiver_controller_, transport_send_->packet_router(), config, |
681 config_.audio_state, event_log_); | 689 config_.audio_state, event_log_); |
682 { | 690 { |
683 WriteLockScoped write_lock(*receive_crit_); | 691 WriteLockScoped write_lock(*receive_crit_); |
684 receive_rtp_config_[config.rtp.remote_ssrc] = | 692 receive_rtp_config_[config.rtp.remote_ssrc] = |
685 ReceiveRtpConfig(config.rtp.extensions, UseSendSideBwe(config)); | 693 ReceiveRtpConfig(config.rtp.extensions, UseSendSideBwe(config)); |
686 audio_receive_streams_.insert(receive_stream); | 694 audio_receive_streams_.insert(receive_stream); |
687 | 695 |
688 ConfigureSync(config.sync_group); | 696 ConfigureSync(config.sync_group); |
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728 | 736 |
729 webrtc::VideoSendStream* Call::CreateVideoSendStream( | 737 webrtc::VideoSendStream* Call::CreateVideoSendStream( |
730 webrtc::VideoSendStream::Config config, | 738 webrtc::VideoSendStream::Config config, |
731 VideoEncoderConfig encoder_config) { | 739 VideoEncoderConfig encoder_config) { |
732 TRACE_EVENT0("webrtc", "Call::CreateVideoSendStream"); | 740 TRACE_EVENT0("webrtc", "Call::CreateVideoSendStream"); |
733 RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_); | 741 RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_); |
734 | 742 |
735 video_send_delay_stats_->AddSsrcs(config); | 743 video_send_delay_stats_->AddSsrcs(config); |
736 for (size_t ssrc_index = 0; ssrc_index < config.rtp.ssrcs.size(); | 744 for (size_t ssrc_index = 0; ssrc_index < config.rtp.ssrcs.size(); |
737 ++ssrc_index) { | 745 ++ssrc_index) { |
738 event_log_->LogVideoSendStreamConfig( | 746 event_log_->Log(rtc::MakeUnique<RtcEventVideoSendStreamConfig>( |
739 *CreateRtcLogStreamConfig(config, ssrc_index)); | 747 CreateRtcLogStreamConfig(config, ssrc_index))); |
740 } | 748 } |
741 | 749 |
742 // TODO(mflodman): Base the start bitrate on a current bandwidth estimate, if | 750 // TODO(mflodman): Base the start bitrate on a current bandwidth estimate, if |
743 // the call has already started. | 751 // the call has already started. |
744 // Copy ssrcs from |config| since |config| is moved. | 752 // Copy ssrcs from |config| since |config| is moved. |
745 std::vector<uint32_t> ssrcs = config.rtp.ssrcs; | 753 std::vector<uint32_t> ssrcs = config.rtp.ssrcs; |
746 VideoSendStream* send_stream = new VideoSendStream( | 754 VideoSendStream* send_stream = new VideoSendStream( |
747 num_cpu_cores_, module_process_thread_.get(), &worker_queue_, | 755 num_cpu_cores_, module_process_thread_.get(), &worker_queue_, |
748 call_stats_.get(), transport_send_.get(), bitrate_allocator_.get(), | 756 call_stats_.get(), transport_send_.get(), bitrate_allocator_.get(), |
749 video_send_delay_stats_.get(), event_log_, std::move(config), | 757 video_send_delay_stats_.get(), event_log_, std::move(config), |
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819 // type, we may get an incorrect value for the rtx stream, but | 827 // type, we may get an incorrect value for the rtx stream, but |
820 // that is unlikely to matter in practice. | 828 // that is unlikely to matter in practice. |
821 receive_rtp_config_[config.rtp.rtx_ssrc] = receive_config; | 829 receive_rtp_config_[config.rtp.rtx_ssrc] = receive_config; |
822 } | 830 } |
823 receive_rtp_config_[config.rtp.remote_ssrc] = receive_config; | 831 receive_rtp_config_[config.rtp.remote_ssrc] = receive_config; |
824 video_receive_streams_.insert(receive_stream); | 832 video_receive_streams_.insert(receive_stream); |
825 ConfigureSync(config.sync_group); | 833 ConfigureSync(config.sync_group); |
826 } | 834 } |
827 receive_stream->SignalNetworkState(video_network_state_); | 835 receive_stream->SignalNetworkState(video_network_state_); |
828 UpdateAggregateNetworkState(); | 836 UpdateAggregateNetworkState(); |
829 event_log_->LogVideoReceiveStreamConfig(*CreateRtcLogStreamConfig(config)); | 837 event_log_->Log(rtc::MakeUnique<RtcEventVideoReceiveStreamConfig>( |
| 838 CreateRtcLogStreamConfig(config))); |
830 return receive_stream; | 839 return receive_stream; |
831 } | 840 } |
832 | 841 |
833 void Call::DestroyVideoReceiveStream( | 842 void Call::DestroyVideoReceiveStream( |
834 webrtc::VideoReceiveStream* receive_stream) { | 843 webrtc::VideoReceiveStream* receive_stream) { |
835 TRACE_EVENT0("webrtc", "Call::DestroyVideoReceiveStream"); | 844 TRACE_EVENT0("webrtc", "Call::DestroyVideoReceiveStream"); |
836 RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_); | 845 RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_); |
837 RTC_DCHECK(receive_stream != nullptr); | 846 RTC_DCHECK(receive_stream != nullptr); |
838 VideoReceiveStream* receive_stream_impl = | 847 VideoReceiveStream* receive_stream_impl = |
839 static_cast<VideoReceiveStream*>(receive_stream); | 848 static_cast<VideoReceiveStream*>(receive_stream); |
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1295 } | 1304 } |
1296 } | 1305 } |
1297 if (media_type == MediaType::ANY || media_type == MediaType::AUDIO) { | 1306 if (media_type == MediaType::ANY || media_type == MediaType::AUDIO) { |
1298 ReadLockScoped read_lock(*send_crit_); | 1307 ReadLockScoped read_lock(*send_crit_); |
1299 for (auto& kv : audio_send_ssrcs_) { | 1308 for (auto& kv : audio_send_ssrcs_) { |
1300 if (kv.second->DeliverRtcp(packet, length)) | 1309 if (kv.second->DeliverRtcp(packet, length)) |
1301 rtcp_delivered = true; | 1310 rtcp_delivered = true; |
1302 } | 1311 } |
1303 } | 1312 } |
1304 | 1313 |
1305 if (rtcp_delivered) | 1314 if (rtcp_delivered) { |
1306 event_log_->LogIncomingRtcpPacket(rtc::MakeArrayView(packet, length)); | 1315 event_log_->Log(rtc::MakeUnique<RtcEventRtcpPacketIncoming>( |
| 1316 rtc::MakeArrayView(packet, length))); |
| 1317 } |
1307 | 1318 |
1308 return rtcp_delivered ? DELIVERY_OK : DELIVERY_PACKET_ERROR; | 1319 return rtcp_delivered ? DELIVERY_OK : DELIVERY_PACKET_ERROR; |
1309 } | 1320 } |
1310 | 1321 |
1311 PacketReceiver::DeliveryStatus Call::DeliverRtp(MediaType media_type, | 1322 PacketReceiver::DeliveryStatus Call::DeliverRtp(MediaType media_type, |
1312 const uint8_t* packet, | 1323 const uint8_t* packet, |
1313 size_t length, | 1324 size_t length, |
1314 const PacketTime& packet_time) { | 1325 const PacketTime& packet_time) { |
1315 TRACE_EVENT0("webrtc", "Call::DeliverRtp"); | 1326 TRACE_EVENT0("webrtc", "Call::DeliverRtp"); |
1316 | 1327 |
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1345 return DELIVERY_UNKNOWN_SSRC; | 1356 return DELIVERY_UNKNOWN_SSRC; |
1346 } | 1357 } |
1347 parsed_packet->IdentifyExtensions(it->second.extensions); | 1358 parsed_packet->IdentifyExtensions(it->second.extensions); |
1348 | 1359 |
1349 NotifyBweOfReceivedPacket(*parsed_packet, media_type); | 1360 NotifyBweOfReceivedPacket(*parsed_packet, media_type); |
1350 | 1361 |
1351 if (media_type == MediaType::AUDIO) { | 1362 if (media_type == MediaType::AUDIO) { |
1352 if (audio_receiver_controller_.OnRtpPacket(*parsed_packet)) { | 1363 if (audio_receiver_controller_.OnRtpPacket(*parsed_packet)) { |
1353 received_bytes_per_second_counter_.Add(static_cast<int>(length)); | 1364 received_bytes_per_second_counter_.Add(static_cast<int>(length)); |
1354 received_audio_bytes_per_second_counter_.Add(static_cast<int>(length)); | 1365 received_audio_bytes_per_second_counter_.Add(static_cast<int>(length)); |
1355 event_log_->LogIncomingRtpHeader(*parsed_packet); | 1366 event_log_->Log( |
| 1367 rtc::MakeUnique<RtcEventRtpPacketIncoming>(*parsed_packet)); |
1356 const int64_t arrival_time_ms = parsed_packet->arrival_time_ms(); | 1368 const int64_t arrival_time_ms = parsed_packet->arrival_time_ms(); |
1357 if (!first_received_rtp_audio_ms_) { | 1369 if (!first_received_rtp_audio_ms_) { |
1358 first_received_rtp_audio_ms_.emplace(arrival_time_ms); | 1370 first_received_rtp_audio_ms_.emplace(arrival_time_ms); |
1359 } | 1371 } |
1360 last_received_rtp_audio_ms_.emplace(arrival_time_ms); | 1372 last_received_rtp_audio_ms_.emplace(arrival_time_ms); |
1361 return DELIVERY_OK; | 1373 return DELIVERY_OK; |
1362 } | 1374 } |
1363 } else if (media_type == MediaType::VIDEO) { | 1375 } else if (media_type == MediaType::VIDEO) { |
1364 if (video_receiver_controller_.OnRtpPacket(*parsed_packet)) { | 1376 if (video_receiver_controller_.OnRtpPacket(*parsed_packet)) { |
1365 received_bytes_per_second_counter_.Add(static_cast<int>(length)); | 1377 received_bytes_per_second_counter_.Add(static_cast<int>(length)); |
1366 received_video_bytes_per_second_counter_.Add(static_cast<int>(length)); | 1378 received_video_bytes_per_second_counter_.Add(static_cast<int>(length)); |
1367 event_log_->LogIncomingRtpHeader(*parsed_packet); | 1379 event_log_->Log( |
| 1380 rtc::MakeUnique<RtcEventRtpPacketIncoming>(*parsed_packet)); |
1368 const int64_t arrival_time_ms = parsed_packet->arrival_time_ms(); | 1381 const int64_t arrival_time_ms = parsed_packet->arrival_time_ms(); |
1369 if (!first_received_rtp_video_ms_) { | 1382 if (!first_received_rtp_video_ms_) { |
1370 first_received_rtp_video_ms_.emplace(arrival_time_ms); | 1383 first_received_rtp_video_ms_.emplace(arrival_time_ms); |
1371 } | 1384 } |
1372 last_received_rtp_video_ms_.emplace(arrival_time_ms); | 1385 last_received_rtp_video_ms_.emplace(arrival_time_ms); |
1373 return DELIVERY_OK; | 1386 return DELIVERY_OK; |
1374 } | 1387 } |
1375 } | 1388 } |
1376 return DELIVERY_UNKNOWN_SSRC; | 1389 return DELIVERY_UNKNOWN_SSRC; |
1377 } | 1390 } |
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1439 (use_send_side_bwe && header.extension.hasTransportSequenceNumber)) { | 1452 (use_send_side_bwe && header.extension.hasTransportSequenceNumber)) { |
1440 receive_side_cc_.OnReceivedPacket( | 1453 receive_side_cc_.OnReceivedPacket( |
1441 packet.arrival_time_ms(), packet.payload_size() + packet.padding_size(), | 1454 packet.arrival_time_ms(), packet.payload_size() + packet.padding_size(), |
1442 header); | 1455 header); |
1443 } | 1456 } |
1444 } | 1457 } |
1445 | 1458 |
1446 } // namespace internal | 1459 } // namespace internal |
1447 | 1460 |
1448 } // namespace webrtc | 1461 } // namespace webrtc |
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