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 <string.h> | 11 #include <string.h> |
12 | |
13 #include <algorithm> | 12 #include <algorithm> |
14 #include <map> | 13 #include <map> |
15 #include <memory> | 14 #include <memory> |
16 #include <vector> | 15 #include <vector> |
17 | 16 |
18 #include "webrtc/audio/audio_receive_stream.h" | 17 #include "webrtc/audio/audio_receive_stream.h" |
19 #include "webrtc/audio/audio_send_stream.h" | 18 #include "webrtc/audio/audio_send_stream.h" |
20 #include "webrtc/audio/audio_state.h" | 19 #include "webrtc/audio/audio_state.h" |
21 #include "webrtc/audio/scoped_voe_interface.h" | 20 #include "webrtc/audio/scoped_voe_interface.h" |
22 #include "webrtc/base/checks.h" | 21 #include "webrtc/base/checks.h" |
23 #include "webrtc/base/constructormagic.h" | 22 #include "webrtc/base/constructormagic.h" |
24 #include "webrtc/base/logging.h" | 23 #include "webrtc/base/logging.h" |
24 #include "webrtc/base/task_queue.h" | |
25 #include "webrtc/base/thread_annotations.h" | 25 #include "webrtc/base/thread_annotations.h" |
26 #include "webrtc/base/thread_checker.h" | 26 #include "webrtc/base/thread_checker.h" |
27 #include "webrtc/base/trace_event.h" | 27 #include "webrtc/base/trace_event.h" |
28 #include "webrtc/call.h" | 28 #include "webrtc/call.h" |
29 #include "webrtc/call/bitrate_allocator.h" | 29 #include "webrtc/call/bitrate_allocator.h" |
30 #include "webrtc/call/rtc_event_log.h" | 30 #include "webrtc/call/rtc_event_log.h" |
31 #include "webrtc/config.h" | 31 #include "webrtc/config.h" |
32 #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h" | 32 #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h" |
33 #include "webrtc/modules/congestion_controller/include/congestion_controller.h" | 33 #include "webrtc/modules/congestion_controller/include/congestion_controller.h" |
34 #include "webrtc/modules/pacing/paced_sender.h" | 34 #include "webrtc/modules/pacing/paced_sender.h" |
(...skipping 32 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... | |
67 webrtc::AudioSendStream* CreateAudioSendStream( | 67 webrtc::AudioSendStream* CreateAudioSendStream( |
68 const webrtc::AudioSendStream::Config& config) override; | 68 const webrtc::AudioSendStream::Config& config) override; |
69 void DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) override; | 69 void DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) override; |
70 | 70 |
71 webrtc::AudioReceiveStream* CreateAudioReceiveStream( | 71 webrtc::AudioReceiveStream* CreateAudioReceiveStream( |
72 const webrtc::AudioReceiveStream::Config& config) override; | 72 const webrtc::AudioReceiveStream::Config& config) override; |
73 void DestroyAudioReceiveStream( | 73 void DestroyAudioReceiveStream( |
74 webrtc::AudioReceiveStream* receive_stream) override; | 74 webrtc::AudioReceiveStream* receive_stream) override; |
75 | 75 |
76 webrtc::VideoSendStream* CreateVideoSendStream( | 76 webrtc::VideoSendStream* CreateVideoSendStream( |
77 const webrtc::VideoSendStream::Config& config, | 77 webrtc::VideoSendStream::Config config, |
78 const VideoEncoderConfig& encoder_config) override; | 78 VideoEncoderConfig encoder_config) override; |
79 void DestroyVideoSendStream(webrtc::VideoSendStream* send_stream) override; | 79 void DestroyVideoSendStream(webrtc::VideoSendStream* send_stream) override; |
80 | 80 |
81 webrtc::VideoReceiveStream* CreateVideoReceiveStream( | 81 webrtc::VideoReceiveStream* CreateVideoReceiveStream( |
82 webrtc::VideoReceiveStream::Config configuration) override; | 82 webrtc::VideoReceiveStream::Config configuration) override; |
83 void DestroyVideoReceiveStream( | 83 void DestroyVideoReceiveStream( |
84 webrtc::VideoReceiveStream* receive_stream) override; | 84 webrtc::VideoReceiveStream* receive_stream) override; |
85 | 85 |
86 Stats GetStats() const override; | 86 Stats GetStats() const override; |
87 | 87 |
88 DeliveryStatus DeliverPacket(MediaType media_type, | 88 DeliveryStatus DeliverPacket(MediaType media_type, |
(...skipping 100 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... | |
189 int64_t pacer_bitrate_sum_kbits_ GUARDED_BY(&bitrate_crit_); | 189 int64_t pacer_bitrate_sum_kbits_ GUARDED_BY(&bitrate_crit_); |
190 uint32_t min_allocated_send_bitrate_bps_ GUARDED_BY(&bitrate_crit_); | 190 uint32_t min_allocated_send_bitrate_bps_ GUARDED_BY(&bitrate_crit_); |
191 int64_t num_bitrate_updates_ GUARDED_BY(&bitrate_crit_); | 191 int64_t num_bitrate_updates_ GUARDED_BY(&bitrate_crit_); |
192 uint32_t configured_max_padding_bitrate_bps_ GUARDED_BY(&bitrate_crit_); | 192 uint32_t configured_max_padding_bitrate_bps_ GUARDED_BY(&bitrate_crit_); |
193 | 193 |
194 std::map<std::string, rtc::NetworkRoute> network_routes_; | 194 std::map<std::string, rtc::NetworkRoute> network_routes_; |
195 | 195 |
196 VieRemb remb_; | 196 VieRemb remb_; |
197 const std::unique_ptr<CongestionController> congestion_controller_; | 197 const std::unique_ptr<CongestionController> congestion_controller_; |
198 const std::unique_ptr<SendDelayStats> video_send_delay_stats_; | 198 const std::unique_ptr<SendDelayStats> video_send_delay_stats_; |
199 // TODO(perkj): |worker_queue_| is supposed to replace | |
200 // |module_process_thread_|. | |
201 // |worker_queue| is defined last to ensure all pending tasks are cancelled | |
202 // and deleted before any other members. | |
203 rtc::TaskQueue worker_queue_; | |
199 | 204 |
200 RTC_DISALLOW_COPY_AND_ASSIGN(Call); | 205 RTC_DISALLOW_COPY_AND_ASSIGN(Call); |
201 }; | 206 }; |
202 } // namespace internal | 207 } // namespace internal |
203 | 208 |
204 Call* Call::Create(const Call::Config& config) { | 209 Call* Call::Create(const Call::Config& config) { |
205 return new internal::Call(config); | 210 return new internal::Call(config); |
206 } | 211 } |
207 | 212 |
208 namespace internal { | 213 namespace internal { |
(...skipping 19 matching lines...) Expand all Loading... | |
228 first_packet_sent_ms_(-1), | 233 first_packet_sent_ms_(-1), |
229 estimated_send_bitrate_sum_kbits_(0), | 234 estimated_send_bitrate_sum_kbits_(0), |
230 pacer_bitrate_sum_kbits_(0), | 235 pacer_bitrate_sum_kbits_(0), |
231 min_allocated_send_bitrate_bps_(0), | 236 min_allocated_send_bitrate_bps_(0), |
232 num_bitrate_updates_(0), | 237 num_bitrate_updates_(0), |
233 configured_max_padding_bitrate_bps_(0), | 238 configured_max_padding_bitrate_bps_(0), |
234 | 239 |
235 remb_(clock_), | 240 remb_(clock_), |
236 congestion_controller_( | 241 congestion_controller_( |
237 new CongestionController(clock_, this, &remb_, event_log_.get())), | 242 new CongestionController(clock_, this, &remb_, event_log_.get())), |
238 video_send_delay_stats_(new SendDelayStats(clock_)) { | 243 video_send_delay_stats_(new SendDelayStats(clock_)), |
244 worker_queue_("worker_queue") { | |
pbos-webrtc
2016/07/13 12:35:38
call_worker_queue or something tying it to call?
perkj_webrtc
2016/07/14 10:11:27
Done.
| |
239 RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread()); | 245 RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread()); |
240 RTC_DCHECK_GE(config.bitrate_config.min_bitrate_bps, 0); | 246 RTC_DCHECK_GE(config.bitrate_config.min_bitrate_bps, 0); |
241 RTC_DCHECK_GE(config.bitrate_config.start_bitrate_bps, | 247 RTC_DCHECK_GE(config.bitrate_config.start_bitrate_bps, |
242 config.bitrate_config.min_bitrate_bps); | 248 config.bitrate_config.min_bitrate_bps); |
243 if (config.bitrate_config.max_bitrate_bps != -1) { | 249 if (config.bitrate_config.max_bitrate_bps != -1) { |
244 RTC_DCHECK_GE(config.bitrate_config.max_bitrate_bps, | 250 RTC_DCHECK_GE(config.bitrate_config.max_bitrate_bps, |
245 config.bitrate_config.start_bitrate_bps); | 251 config.bitrate_config.start_bitrate_bps); |
246 } | 252 } |
247 | 253 |
248 Trace::CreateTrace(); | 254 Trace::CreateTrace(); |
249 call_stats_->RegisterStatsObserver(congestion_controller_.get()); | 255 call_stats_->RegisterStatsObserver(congestion_controller_.get()); |
250 | 256 |
251 congestion_controller_->SetBweBitrates( | 257 congestion_controller_->SetBweBitrates( |
252 config_.bitrate_config.min_bitrate_bps, | 258 config_.bitrate_config.min_bitrate_bps, |
253 config_.bitrate_config.start_bitrate_bps, | 259 config_.bitrate_config.start_bitrate_bps, |
254 config_.bitrate_config.max_bitrate_bps); | 260 config_.bitrate_config.max_bitrate_bps); |
255 | 261 |
256 module_process_thread_->Start(); | 262 module_process_thread_->Start(); |
257 module_process_thread_->RegisterModule(call_stats_.get()); | 263 module_process_thread_->RegisterModule(call_stats_.get()); |
258 module_process_thread_->RegisterModule(congestion_controller_.get()); | 264 module_process_thread_->RegisterModule(congestion_controller_.get()); |
259 pacer_thread_->RegisterModule(congestion_controller_->pacer()); | 265 pacer_thread_->RegisterModule(congestion_controller_->pacer()); |
260 pacer_thread_->RegisterModule( | 266 pacer_thread_->RegisterModule( |
261 congestion_controller_->GetRemoteBitrateEstimator(true)); | 267 congestion_controller_->GetRemoteBitrateEstimator(true)); |
262 pacer_thread_->Start(); | 268 pacer_thread_->Start(); |
263 } | 269 } |
264 | 270 |
265 Call::~Call() { | 271 Call::~Call() { |
266 RTC_DCHECK(!remb_.InUse()); | 272 RTC_DCHECK(!remb_.InUse()); |
267 RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread()); | 273 RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread()); |
268 UpdateSendHistograms(); | 274 { |
275 rtc::CritScope lock(&bitrate_crit_); | |
276 UpdateSendHistograms(); | |
277 } | |
269 UpdateReceiveHistograms(); | 278 UpdateReceiveHistograms(); |
270 RTC_CHECK(audio_send_ssrcs_.empty()); | 279 RTC_CHECK(audio_send_ssrcs_.empty()); |
271 RTC_CHECK(video_send_ssrcs_.empty()); | 280 RTC_CHECK(video_send_ssrcs_.empty()); |
272 RTC_CHECK(video_send_streams_.empty()); | 281 RTC_CHECK(video_send_streams_.empty()); |
273 RTC_CHECK(audio_receive_ssrcs_.empty()); | 282 RTC_CHECK(audio_receive_ssrcs_.empty()); |
274 RTC_CHECK(video_receive_ssrcs_.empty()); | 283 RTC_CHECK(video_receive_ssrcs_.empty()); |
275 RTC_CHECK(video_receive_streams_.empty()); | 284 RTC_CHECK(video_receive_streams_.empty()); |
276 | 285 |
277 pacer_thread_->Stop(); | 286 pacer_thread_->Stop(); |
278 pacer_thread_->DeRegisterModule(congestion_controller_->pacer()); | 287 pacer_thread_->DeRegisterModule(congestion_controller_->pacer()); |
(...skipping 133 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... | |
412 it->second == audio_receive_stream) { | 421 it->second == audio_receive_stream) { |
413 sync_stream_mapping_.erase(it); | 422 sync_stream_mapping_.erase(it); |
414 ConfigureSync(sync_group); | 423 ConfigureSync(sync_group); |
415 } | 424 } |
416 } | 425 } |
417 UpdateAggregateNetworkState(); | 426 UpdateAggregateNetworkState(); |
418 delete audio_receive_stream; | 427 delete audio_receive_stream; |
419 } | 428 } |
420 | 429 |
421 webrtc::VideoSendStream* Call::CreateVideoSendStream( | 430 webrtc::VideoSendStream* Call::CreateVideoSendStream( |
422 const webrtc::VideoSendStream::Config& config, | 431 webrtc::VideoSendStream::Config config, |
423 const VideoEncoderConfig& encoder_config) { | 432 VideoEncoderConfig encoder_config) { |
424 TRACE_EVENT0("webrtc", "Call::CreateVideoSendStream"); | 433 TRACE_EVENT0("webrtc", "Call::CreateVideoSendStream"); |
425 RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread()); | 434 RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread()); |
426 | 435 |
427 video_send_delay_stats_->AddSsrcs(config); | 436 video_send_delay_stats_->AddSsrcs(config); |
437 event_log_->LogVideoSendStreamConfig(config); | |
438 | |
428 // TODO(mflodman): Base the start bitrate on a current bandwidth estimate, if | 439 // TODO(mflodman): Base the start bitrate on a current bandwidth estimate, if |
429 // the call has already started. | 440 // the call has already started. |
441 // Copy ssrcs from |config| since |config| is moved. | |
442 std::vector<uint32_t> ssrcs = config.rtp.ssrcs; | |
430 VideoSendStream* send_stream = new VideoSendStream( | 443 VideoSendStream* send_stream = new VideoSendStream( |
431 num_cpu_cores_, module_process_thread_.get(), call_stats_.get(), | 444 num_cpu_cores_, module_process_thread_.get(), &worker_queue_, |
432 congestion_controller_.get(), bitrate_allocator_.get(), | 445 call_stats_.get(), congestion_controller_.get(), bitrate_allocator_.get(), |
433 video_send_delay_stats_.get(), &remb_, event_log_.get(), config, | 446 video_send_delay_stats_.get(), &remb_, event_log_.get(), |
434 encoder_config, suspended_video_send_ssrcs_); | 447 std::move(config), std::move(encoder_config), |
448 suspended_video_send_ssrcs_); | |
449 | |
435 { | 450 { |
436 WriteLockScoped write_lock(*send_crit_); | 451 WriteLockScoped write_lock(*send_crit_); |
437 for (uint32_t ssrc : config.rtp.ssrcs) { | 452 for (uint32_t ssrc : ssrcs) { |
438 RTC_DCHECK(video_send_ssrcs_.find(ssrc) == video_send_ssrcs_.end()); | 453 RTC_DCHECK(video_send_ssrcs_.find(ssrc) == video_send_ssrcs_.end()); |
439 video_send_ssrcs_[ssrc] = send_stream; | 454 video_send_ssrcs_[ssrc] = send_stream; |
440 } | 455 } |
441 video_send_streams_.insert(send_stream); | 456 video_send_streams_.insert(send_stream); |
442 } | 457 } |
443 send_stream->SignalNetworkState(video_network_state_); | 458 send_stream->SignalNetworkState(video_network_state_); |
444 UpdateAggregateNetworkState(); | 459 UpdateAggregateNetworkState(); |
445 event_log_->LogVideoSendStreamConfig(config); | 460 |
446 return send_stream; | 461 return send_stream; |
447 } | 462 } |
448 | 463 |
449 void Call::DestroyVideoSendStream(webrtc::VideoSendStream* send_stream) { | 464 void Call::DestroyVideoSendStream(webrtc::VideoSendStream* send_stream) { |
450 TRACE_EVENT0("webrtc", "Call::DestroyVideoSendStream"); | 465 TRACE_EVENT0("webrtc", "Call::DestroyVideoSendStream"); |
451 RTC_DCHECK(send_stream != nullptr); | 466 RTC_DCHECK(send_stream != nullptr); |
452 RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread()); | 467 RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread()); |
453 | 468 |
454 send_stream->Stop(); | 469 send_stream->Stop(); |
455 | 470 |
456 VideoSendStream* send_stream_impl = nullptr; | 471 VideoSendStream* send_stream_impl = nullptr; |
457 { | 472 { |
458 WriteLockScoped write_lock(*send_crit_); | 473 WriteLockScoped write_lock(*send_crit_); |
459 auto it = video_send_ssrcs_.begin(); | 474 auto it = video_send_ssrcs_.begin(); |
460 while (it != video_send_ssrcs_.end()) { | 475 while (it != video_send_ssrcs_.end()) { |
461 if (it->second == static_cast<VideoSendStream*>(send_stream)) { | 476 if (it->second == static_cast<VideoSendStream*>(send_stream)) { |
462 send_stream_impl = it->second; | 477 send_stream_impl = it->second; |
463 video_send_ssrcs_.erase(it++); | 478 video_send_ssrcs_.erase(it++); |
464 } else { | 479 } else { |
465 ++it; | 480 ++it; |
466 } | 481 } |
467 } | 482 } |
468 video_send_streams_.erase(send_stream_impl); | 483 video_send_streams_.erase(send_stream_impl); |
469 } | 484 } |
470 RTC_CHECK(send_stream_impl != nullptr); | 485 RTC_CHECK(send_stream_impl != nullptr); |
471 | 486 |
472 VideoSendStream::RtpStateMap rtp_state = send_stream_impl->GetRtpStates(); | 487 VideoSendStream::RtpStateMap rtp_state = |
488 send_stream_impl->StopPermanentlyAndGetRtpStates(); | |
473 | 489 |
474 for (VideoSendStream::RtpStateMap::iterator it = rtp_state.begin(); | 490 for (VideoSendStream::RtpStateMap::iterator it = rtp_state.begin(); |
475 it != rtp_state.end(); | 491 it != rtp_state.end(); ++it) { |
476 ++it) { | |
477 suspended_video_send_ssrcs_[it->first] = it->second; | 492 suspended_video_send_ssrcs_[it->first] = it->second; |
478 } | 493 } |
479 | 494 |
480 UpdateAggregateNetworkState(); | 495 UpdateAggregateNetworkState(); |
481 delete send_stream_impl; | 496 delete send_stream_impl; |
482 } | 497 } |
483 | 498 |
484 webrtc::VideoReceiveStream* Call::CreateVideoReceiveStream( | 499 webrtc::VideoReceiveStream* Call::CreateVideoReceiveStream( |
485 webrtc::VideoReceiveStream::Config configuration) { | 500 webrtc::VideoReceiveStream::Config configuration) { |
486 TRACE_EVENT0("webrtc", "Call::CreateVideoReceiveStream"); | 501 TRACE_EVENT0("webrtc", "Call::CreateVideoReceiveStream"); |
(...skipping 209 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... | |
696 void Call::OnSentPacket(const rtc::SentPacket& sent_packet) { | 711 void Call::OnSentPacket(const rtc::SentPacket& sent_packet) { |
697 if (first_packet_sent_ms_ == -1) | 712 if (first_packet_sent_ms_ == -1) |
698 first_packet_sent_ms_ = clock_->TimeInMilliseconds(); | 713 first_packet_sent_ms_ = clock_->TimeInMilliseconds(); |
699 video_send_delay_stats_->OnSentPacket(sent_packet.packet_id, | 714 video_send_delay_stats_->OnSentPacket(sent_packet.packet_id, |
700 clock_->TimeInMilliseconds()); | 715 clock_->TimeInMilliseconds()); |
701 congestion_controller_->OnSentPacket(sent_packet); | 716 congestion_controller_->OnSentPacket(sent_packet); |
702 } | 717 } |
703 | 718 |
704 void Call::OnNetworkChanged(uint32_t target_bitrate_bps, uint8_t fraction_loss, | 719 void Call::OnNetworkChanged(uint32_t target_bitrate_bps, uint8_t fraction_loss, |
705 int64_t rtt_ms) { | 720 int64_t rtt_ms) { |
721 // TODO(perkj): Consider making sure CongestionController operates on | |
722 // |worker_queue_|. | |
723 if (!worker_queue_.IsCurrent()) { | |
724 worker_queue_.PostTask([this, target_bitrate_bps, fraction_loss, rtt_ms] { | |
725 OnNetworkChanged(target_bitrate_bps, fraction_loss, rtt_ms); | |
726 }); | |
727 return; | |
728 } | |
729 RTC_DCHECK_RUN_ON(&worker_queue_); | |
706 bitrate_allocator_->OnNetworkChanged(target_bitrate_bps, fraction_loss, | 730 bitrate_allocator_->OnNetworkChanged(target_bitrate_bps, fraction_loss, |
707 rtt_ms); | 731 rtt_ms); |
708 | 732 |
709 { | 733 { |
710 rtc::CritScope lock(&bitrate_crit_); | 734 rtc::CritScope lock(&bitrate_crit_); |
711 // We only update these stats if we have send streams, and assume that | 735 // We only update these stats if we have send streams, and assume that |
712 // OnNetworkChanged is called roughly with a fixed frequency. | 736 // OnNetworkChanged is called roughly with a fixed frequency. |
713 estimated_send_bitrate_sum_kbits_ += target_bitrate_bps / 1000; | 737 estimated_send_bitrate_sum_kbits_ += target_bitrate_bps / 1000; |
714 // Pacer bitrate might be higher than bitrate estimate if enforcing min | 738 // Pacer bitrate might be higher than bitrate estimate if enforcing min |
715 // bitrate. | 739 // bitrate. |
(...skipping 157 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... | |
873 // thread. Then this check can be enabled. | 897 // thread. Then this check can be enabled. |
874 // RTC_DCHECK(!configuration_thread_checker_.CalledOnValidThread()); | 898 // RTC_DCHECK(!configuration_thread_checker_.CalledOnValidThread()); |
875 if (RtpHeaderParser::IsRtcp(packet, length)) | 899 if (RtpHeaderParser::IsRtcp(packet, length)) |
876 return DeliverRtcp(media_type, packet, length); | 900 return DeliverRtcp(media_type, packet, length); |
877 | 901 |
878 return DeliverRtp(media_type, packet, length, packet_time); | 902 return DeliverRtp(media_type, packet, length, packet_time); |
879 } | 903 } |
880 | 904 |
881 } // namespace internal | 905 } // namespace internal |
882 } // namespace webrtc | 906 } // namespace webrtc |
OLD | NEW |