| /* |
| * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #include <string.h> |
| #include <algorithm> |
| #include <map> |
| #include <memory> |
| #include <set> |
| #include <utility> |
| #include <vector> |
| |
| #include "webrtc/api/optional.h" |
| #include "webrtc/audio/audio_receive_stream.h" |
| #include "webrtc/audio/audio_send_stream.h" |
| #include "webrtc/audio/audio_state.h" |
| #include "webrtc/audio/scoped_voe_interface.h" |
| #include "webrtc/audio/time_interval.h" |
| #include "webrtc/call/bitrate_allocator.h" |
| #include "webrtc/call/call.h" |
| #include "webrtc/call/flexfec_receive_stream_impl.h" |
| #include "webrtc/call/rtp_stream_receiver_controller.h" |
| #include "webrtc/call/rtp_transport_controller_send.h" |
| #include "webrtc/logging/rtc_event_log/rtc_event_log.h" |
| #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h" |
| #include "webrtc/modules/congestion_controller/include/receive_side_congestion_controller.h" |
| #include "webrtc/modules/rtp_rtcp/include/flexfec_receiver.h" |
| #include "webrtc/modules/rtp_rtcp/include/rtp_header_extension_map.h" |
| #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" |
| #include "webrtc/modules/rtp_rtcp/source/byte_io.h" |
| #include "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h" |
| #include "webrtc/modules/utility/include/process_thread.h" |
| #include "webrtc/rtc_base/basictypes.h" |
| #include "webrtc/rtc_base/checks.h" |
| #include "webrtc/rtc_base/constructormagic.h" |
| #include "webrtc/rtc_base/location.h" |
| #include "webrtc/rtc_base/logging.h" |
| #include "webrtc/rtc_base/ptr_util.h" |
| #include "webrtc/rtc_base/sequenced_task_checker.h" |
| #include "webrtc/rtc_base/task_queue.h" |
| #include "webrtc/rtc_base/thread_annotations.h" |
| #include "webrtc/rtc_base/trace_event.h" |
| #include "webrtc/system_wrappers/include/clock.h" |
| #include "webrtc/system_wrappers/include/cpu_info.h" |
| #include "webrtc/system_wrappers/include/metrics.h" |
| #include "webrtc/system_wrappers/include/rw_lock_wrapper.h" |
| #include "webrtc/system_wrappers/include/trace.h" |
| #include "webrtc/video/call_stats.h" |
| #include "webrtc/video/send_delay_stats.h" |
| #include "webrtc/video/stats_counter.h" |
| #include "webrtc/video/video_receive_stream.h" |
| #include "webrtc/video/video_send_stream.h" |
| |
| namespace webrtc { |
| |
| namespace { |
| |
| // TODO(nisse): This really begs for a shared context struct. |
| bool UseSendSideBwe(const std::vector<RtpExtension>& extensions, |
| bool transport_cc) { |
| if (!transport_cc) |
| return false; |
| for (const auto& extension : extensions) { |
| if (extension.uri == RtpExtension::kTransportSequenceNumberUri) |
| return true; |
| } |
| return false; |
| } |
| |
| bool UseSendSideBwe(const VideoReceiveStream::Config& config) { |
| return UseSendSideBwe(config.rtp.extensions, config.rtp.transport_cc); |
| } |
| |
| bool UseSendSideBwe(const AudioReceiveStream::Config& config) { |
| return UseSendSideBwe(config.rtp.extensions, config.rtp.transport_cc); |
| } |
| |
| bool UseSendSideBwe(const FlexfecReceiveStream::Config& config) { |
| return UseSendSideBwe(config.rtp_header_extensions, config.transport_cc); |
| } |
| |
| const int* FindKeyByValue(const std::map<int, int>& m, int v) { |
| for (const auto& kv : m) { |
| if (kv.second == v) |
| return &kv.first; |
| } |
| return nullptr; |
| } |
| |
| std::unique_ptr<rtclog::StreamConfig> CreateRtcLogStreamConfig( |
| const VideoReceiveStream::Config& config) { |
| auto rtclog_config = rtc::MakeUnique<rtclog::StreamConfig>(); |
| rtclog_config->remote_ssrc = config.rtp.remote_ssrc; |
| rtclog_config->local_ssrc = config.rtp.local_ssrc; |
| rtclog_config->rtx_ssrc = config.rtp.rtx_ssrc; |
| rtclog_config->rtcp_mode = config.rtp.rtcp_mode; |
| rtclog_config->remb = config.rtp.remb; |
| rtclog_config->rtp_extensions = config.rtp.extensions; |
| |
| for (const auto& d : config.decoders) { |
| const int* search = |
| FindKeyByValue(config.rtp.rtx_associated_payload_types, d.payload_type); |
| rtclog_config->codecs.emplace_back(d.payload_name, d.payload_type, |
| search ? *search : 0); |
| } |
| return rtclog_config; |
| } |
| |
| std::unique_ptr<rtclog::StreamConfig> CreateRtcLogStreamConfig( |
| const VideoSendStream::Config& config, |
| size_t ssrc_index) { |
| auto rtclog_config = rtc::MakeUnique<rtclog::StreamConfig>(); |
| rtclog_config->local_ssrc = config.rtp.ssrcs[ssrc_index]; |
| if (ssrc_index < config.rtp.rtx.ssrcs.size()) { |
| rtclog_config->rtx_ssrc = config.rtp.rtx.ssrcs[ssrc_index]; |
| } |
| rtclog_config->rtcp_mode = config.rtp.rtcp_mode; |
| rtclog_config->rtp_extensions = config.rtp.extensions; |
| |
| rtclog_config->codecs.emplace_back(config.encoder_settings.payload_name, |
| config.encoder_settings.payload_type, |
| config.rtp.rtx.payload_type); |
| return rtclog_config; |
| } |
| |
| std::unique_ptr<rtclog::StreamConfig> CreateRtcLogStreamConfig( |
| const AudioReceiveStream::Config& config) { |
| auto rtclog_config = rtc::MakeUnique<rtclog::StreamConfig>(); |
| rtclog_config->remote_ssrc = config.rtp.remote_ssrc; |
| rtclog_config->local_ssrc = config.rtp.local_ssrc; |
| rtclog_config->rtp_extensions = config.rtp.extensions; |
| return rtclog_config; |
| } |
| |
| std::unique_ptr<rtclog::StreamConfig> CreateRtcLogStreamConfig( |
| const AudioSendStream::Config& config) { |
| auto rtclog_config = rtc::MakeUnique<rtclog::StreamConfig>(); |
| rtclog_config->local_ssrc = config.rtp.ssrc; |
| rtclog_config->rtp_extensions = config.rtp.extensions; |
| if (config.send_codec_spec) { |
| rtclog_config->codecs.emplace_back(config.send_codec_spec->format.name, |
| config.send_codec_spec->payload_type, 0); |
| } |
| return rtclog_config; |
| } |
| |
| } // namespace |
| |
| namespace internal { |
| |
| class Call : public webrtc::Call, |
| public PacketReceiver, |
| public RecoveredPacketReceiver, |
| public SendSideCongestionController::Observer, |
| public BitrateAllocator::LimitObserver { |
| public: |
| Call(const Call::Config& config, |
| std::unique_ptr<RtpTransportControllerSendInterface> transport_send); |
| virtual ~Call(); |
| |
| // Implements webrtc::Call. |
| PacketReceiver* Receiver() override; |
| |
| webrtc::AudioSendStream* CreateAudioSendStream( |
| const webrtc::AudioSendStream::Config& config) override; |
| void DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) override; |
| |
| webrtc::AudioReceiveStream* CreateAudioReceiveStream( |
| const webrtc::AudioReceiveStream::Config& config) override; |
| void DestroyAudioReceiveStream( |
| webrtc::AudioReceiveStream* receive_stream) override; |
| |
| webrtc::VideoSendStream* CreateVideoSendStream( |
| webrtc::VideoSendStream::Config config, |
| VideoEncoderConfig encoder_config) override; |
| void DestroyVideoSendStream(webrtc::VideoSendStream* send_stream) override; |
| |
| webrtc::VideoReceiveStream* CreateVideoReceiveStream( |
| webrtc::VideoReceiveStream::Config configuration) override; |
| void DestroyVideoReceiveStream( |
| webrtc::VideoReceiveStream* receive_stream) override; |
| |
| FlexfecReceiveStream* CreateFlexfecReceiveStream( |
| const FlexfecReceiveStream::Config& config) override; |
| void DestroyFlexfecReceiveStream( |
| FlexfecReceiveStream* receive_stream) override; |
| |
| Stats GetStats() const override; |
| |
| // Implements PacketReceiver. |
| DeliveryStatus DeliverPacket(MediaType media_type, |
| const uint8_t* packet, |
| size_t length, |
| const PacketTime& packet_time) override; |
| |
| // Implements RecoveredPacketReceiver. |
| void OnRecoveredPacket(const uint8_t* packet, size_t length) override; |
| |
| void SetBitrateConfig( |
| const webrtc::Call::Config::BitrateConfig& bitrate_config) override; |
| |
| void SetBitrateConfigMask( |
| const webrtc::Call::Config::BitrateConfigMask& bitrate_config) override; |
| |
| void SignalChannelNetworkState(MediaType media, NetworkState state) override; |
| |
| void OnTransportOverheadChanged(MediaType media, |
| int transport_overhead_per_packet) override; |
| |
| void OnNetworkRouteChanged(const std::string& transport_name, |
| const rtc::NetworkRoute& network_route) override; |
| |
| void OnSentPacket(const rtc::SentPacket& sent_packet) override; |
| |
| // Implements BitrateObserver. |
| void OnNetworkChanged(uint32_t bitrate_bps, |
| uint8_t fraction_loss, |
| int64_t rtt_ms, |
| int64_t probing_interval_ms) override; |
| |
| // Implements BitrateAllocator::LimitObserver. |
| void OnAllocationLimitsChanged(uint32_t min_send_bitrate_bps, |
| uint32_t max_padding_bitrate_bps) override; |
| |
| private: |
| DeliveryStatus DeliverRtcp(MediaType media_type, const uint8_t* packet, |
| size_t length); |
| DeliveryStatus DeliverRtp(MediaType media_type, |
| const uint8_t* packet, |
| size_t length, |
| const PacketTime& packet_time); |
| void ConfigureSync(const std::string& sync_group) |
| EXCLUSIVE_LOCKS_REQUIRED(receive_crit_); |
| |
| void NotifyBweOfReceivedPacket(const RtpPacketReceived& packet, |
| MediaType media_type) |
| SHARED_LOCKS_REQUIRED(receive_crit_); |
| |
| rtc::Optional<RtpPacketReceived> ParseRtpPacket( |
| const uint8_t* packet, |
| size_t length, |
| const PacketTime* packet_time) const; |
| |
| void UpdateSendHistograms(int64_t first_sent_packet_ms) |
| EXCLUSIVE_LOCKS_REQUIRED(&bitrate_crit_); |
| void UpdateReceiveHistograms(); |
| void UpdateHistograms(); |
| void UpdateAggregateNetworkState(); |
| |
| // Applies update to the BitrateConfig cached in |config_|, restarting |
| // bandwidth estimation from |new_start| if set. |
| void UpdateCurrentBitrateConfig(const rtc::Optional<int>& new_start); |
| |
| Clock* const clock_; |
| |
| const int num_cpu_cores_; |
| const std::unique_ptr<ProcessThread> module_process_thread_; |
| const std::unique_ptr<ProcessThread> pacer_thread_; |
| const std::unique_ptr<CallStats> call_stats_; |
| const std::unique_ptr<BitrateAllocator> bitrate_allocator_; |
| Call::Config config_; |
| rtc::SequencedTaskChecker configuration_sequence_checker_; |
| |
| NetworkState audio_network_state_; |
| NetworkState video_network_state_; |
| |
| std::unique_ptr<RWLockWrapper> receive_crit_; |
| // Audio, Video, and FlexFEC receive streams are owned by the client that |
| // creates them. |
| std::set<AudioReceiveStream*> audio_receive_streams_ |
| GUARDED_BY(receive_crit_); |
| std::set<VideoReceiveStream*> video_receive_streams_ |
| GUARDED_BY(receive_crit_); |
| |
| std::map<std::string, AudioReceiveStream*> sync_stream_mapping_ |
| GUARDED_BY(receive_crit_); |
| |
| // TODO(nisse): Should eventually be injected at creation, |
| // with a single object in the bundled case. |
| RtpStreamReceiverController audio_receiver_controller_; |
| RtpStreamReceiverController video_receiver_controller_; |
| |
| // This extra map is used for receive processing which is |
| // independent of media type. |
| |
| // TODO(nisse): In the RTP transport refactoring, we should have a |
| // single mapping from ssrc to a more abstract receive stream, with |
| // accessor methods for all configuration we need at this level. |
| struct ReceiveRtpConfig { |
| ReceiveRtpConfig() = default; // Needed by std::map |
| ReceiveRtpConfig(const std::vector<RtpExtension>& extensions, |
| bool use_send_side_bwe) |
| : extensions(extensions), use_send_side_bwe(use_send_side_bwe) {} |
| |
| // Registered RTP header extensions for each stream. Note that RTP header |
| // extensions are negotiated per track ("m= line") in the SDP, but we have |
| // no notion of tracks at the Call level. We therefore store the RTP header |
| // extensions per SSRC instead, which leads to some storage overhead. |
| RtpHeaderExtensionMap extensions; |
| // Set if both RTP extension the RTCP feedback message needed for |
| // send side BWE are negotiated. |
| bool use_send_side_bwe = false; |
| }; |
| std::map<uint32_t, ReceiveRtpConfig> receive_rtp_config_ |
| GUARDED_BY(receive_crit_); |
| |
| std::unique_ptr<RWLockWrapper> send_crit_; |
| // Audio and Video send streams are owned by the client that creates them. |
| std::map<uint32_t, AudioSendStream*> audio_send_ssrcs_ GUARDED_BY(send_crit_); |
| std::map<uint32_t, VideoSendStream*> video_send_ssrcs_ GUARDED_BY(send_crit_); |
| std::set<VideoSendStream*> video_send_streams_ GUARDED_BY(send_crit_); |
| |
| using RtpStateMap = std::map<uint32_t, RtpState>; |
| RtpStateMap suspended_audio_send_ssrcs_ |
| GUARDED_BY(configuration_sequence_checker_); |
| RtpStateMap suspended_video_send_ssrcs_ |
| GUARDED_BY(configuration_sequence_checker_); |
| |
| webrtc::RtcEventLog* event_log_; |
| |
| // The following members are only accessed (exclusively) from one thread and |
| // from the destructor, and therefore doesn't need any explicit |
| // synchronization. |
| RateCounter received_bytes_per_second_counter_; |
| RateCounter received_audio_bytes_per_second_counter_; |
| RateCounter received_video_bytes_per_second_counter_; |
| RateCounter received_rtcp_bytes_per_second_counter_; |
| rtc::Optional<int64_t> first_received_rtp_audio_ms_; |
| rtc::Optional<int64_t> last_received_rtp_audio_ms_; |
| rtc::Optional<int64_t> first_received_rtp_video_ms_; |
| rtc::Optional<int64_t> last_received_rtp_video_ms_; |
| TimeInterval sent_rtp_audio_timer_ms_; |
| |
| // TODO(holmer): Remove this lock once BitrateController no longer calls |
| // OnNetworkChanged from multiple threads. |
| rtc::CriticalSection bitrate_crit_; |
| uint32_t min_allocated_send_bitrate_bps_ GUARDED_BY(&bitrate_crit_); |
| uint32_t configured_max_padding_bitrate_bps_ GUARDED_BY(&bitrate_crit_); |
| AvgCounter estimated_send_bitrate_kbps_counter_ GUARDED_BY(&bitrate_crit_); |
| AvgCounter pacer_bitrate_kbps_counter_ GUARDED_BY(&bitrate_crit_); |
| |
| std::map<std::string, rtc::NetworkRoute> network_routes_; |
| |
| std::unique_ptr<RtpTransportControllerSendInterface> transport_send_; |
| ReceiveSideCongestionController receive_side_cc_; |
| const std::unique_ptr<SendDelayStats> video_send_delay_stats_; |
| const int64_t start_ms_; |
| // TODO(perkj): |worker_queue_| is supposed to replace |
| // |module_process_thread_|. |
| // |worker_queue| is defined last to ensure all pending tasks are cancelled |
| // and deleted before any other members. |
| rtc::TaskQueue worker_queue_; |
| |
| // The config mask set by SetBitrateConfigMask. |
| // 0 <= min <= start <= max |
| Config::BitrateConfigMask bitrate_config_mask_; |
| |
| // The config set by SetBitrateConfig. |
| // min >= 0, start != 0, max == -1 || max > 0 |
| Config::BitrateConfig base_bitrate_config_; |
| |
| RTC_DISALLOW_COPY_AND_ASSIGN(Call); |
| }; |
| } // namespace internal |
| |
| std::string Call::Stats::ToString(int64_t time_ms) const { |
| std::stringstream ss; |
| ss << "Call stats: " << time_ms << ", {"; |
| ss << "send_bw_bps: " << send_bandwidth_bps << ", "; |
| ss << "recv_bw_bps: " << recv_bandwidth_bps << ", "; |
| ss << "max_pad_bps: " << max_padding_bitrate_bps << ", "; |
| ss << "pacer_delay_ms: " << pacer_delay_ms << ", "; |
| ss << "rtt_ms: " << rtt_ms; |
| ss << '}'; |
| return ss.str(); |
| } |
| |
| Call* Call::Create(const Call::Config& config) { |
| return new internal::Call(config, |
| rtc::MakeUnique<RtpTransportControllerSend>( |
| Clock::GetRealTimeClock(), config.event_log)); |
| } |
| |
| Call* Call::Create( |
| const Call::Config& config, |
| std::unique_ptr<RtpTransportControllerSendInterface> transport_send) { |
| return new internal::Call(config, std::move(transport_send)); |
| } |
| |
| namespace internal { |
| |
| Call::Call(const Call::Config& config, |
| std::unique_ptr<RtpTransportControllerSendInterface> transport_send) |
| : clock_(Clock::GetRealTimeClock()), |
| num_cpu_cores_(CpuInfo::DetectNumberOfCores()), |
| module_process_thread_(ProcessThread::Create("ModuleProcessThread")), |
| pacer_thread_(ProcessThread::Create("PacerThread")), |
| call_stats_(new CallStats(clock_)), |
| bitrate_allocator_(new BitrateAllocator(this)), |
| config_(config), |
| audio_network_state_(kNetworkDown), |
| video_network_state_(kNetworkDown), |
| receive_crit_(RWLockWrapper::CreateRWLock()), |
| send_crit_(RWLockWrapper::CreateRWLock()), |
| event_log_(config.event_log), |
| received_bytes_per_second_counter_(clock_, nullptr, true), |
| received_audio_bytes_per_second_counter_(clock_, nullptr, true), |
| received_video_bytes_per_second_counter_(clock_, nullptr, true), |
| received_rtcp_bytes_per_second_counter_(clock_, nullptr, true), |
| min_allocated_send_bitrate_bps_(0), |
| configured_max_padding_bitrate_bps_(0), |
| estimated_send_bitrate_kbps_counter_(clock_, nullptr, true), |
| pacer_bitrate_kbps_counter_(clock_, nullptr, true), |
| receive_side_cc_(clock_, transport_send->packet_router()), |
| video_send_delay_stats_(new SendDelayStats(clock_)), |
| start_ms_(clock_->TimeInMilliseconds()), |
| worker_queue_("call_worker_queue"), |
| base_bitrate_config_(config.bitrate_config) { |
| RTC_DCHECK(config.event_log != nullptr); |
| RTC_DCHECK_GE(config.bitrate_config.min_bitrate_bps, 0); |
| RTC_DCHECK_GE(config.bitrate_config.start_bitrate_bps, |
| config.bitrate_config.min_bitrate_bps); |
| if (config.bitrate_config.max_bitrate_bps != -1) { |
| RTC_DCHECK_GE(config.bitrate_config.max_bitrate_bps, |
| config.bitrate_config.start_bitrate_bps); |
| } |
| Trace::CreateTrace(); |
| transport_send->send_side_cc()->RegisterNetworkObserver(this); |
| transport_send_ = std::move(transport_send); |
| transport_send_->send_side_cc()->SignalNetworkState(kNetworkDown); |
| transport_send_->send_side_cc()->SetBweBitrates( |
| config_.bitrate_config.min_bitrate_bps, |
| config_.bitrate_config.start_bitrate_bps, |
| config_.bitrate_config.max_bitrate_bps); |
| call_stats_->RegisterStatsObserver(&receive_side_cc_); |
| call_stats_->RegisterStatsObserver(transport_send_->send_side_cc()); |
| |
| // We have to attach the pacer to the pacer thread before starting the |
| // module process thread to avoid a race accessing the process thread |
| // both from the process thread and the pacer thread. |
| pacer_thread_->RegisterModule(transport_send_->pacer(), RTC_FROM_HERE); |
| pacer_thread_->RegisterModule( |
| receive_side_cc_.GetRemoteBitrateEstimator(true), RTC_FROM_HERE); |
| pacer_thread_->Start(); |
| |
| module_process_thread_->RegisterModule(call_stats_.get(), RTC_FROM_HERE); |
| module_process_thread_->RegisterModule(&receive_side_cc_, RTC_FROM_HERE); |
| module_process_thread_->RegisterModule(transport_send_->send_side_cc(), |
| RTC_FROM_HERE); |
| module_process_thread_->Start(); |
| } |
| |
| Call::~Call() { |
| RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_); |
| |
| RTC_CHECK(audio_send_ssrcs_.empty()); |
| RTC_CHECK(video_send_ssrcs_.empty()); |
| RTC_CHECK(video_send_streams_.empty()); |
| RTC_CHECK(audio_receive_streams_.empty()); |
| RTC_CHECK(video_receive_streams_.empty()); |
| |
| // The send-side congestion controller must be de-registered prior to |
| // the pacer thread being stopped to avoid a race when accessing the |
| // pacer thread object on the module process thread at the same time as |
| // the pacer thread is stopped. |
| module_process_thread_->DeRegisterModule(transport_send_->send_side_cc()); |
| pacer_thread_->Stop(); |
| pacer_thread_->DeRegisterModule(transport_send_->pacer()); |
| pacer_thread_->DeRegisterModule( |
| receive_side_cc_.GetRemoteBitrateEstimator(true)); |
| module_process_thread_->DeRegisterModule(&receive_side_cc_); |
| module_process_thread_->DeRegisterModule(call_stats_.get()); |
| module_process_thread_->Stop(); |
| call_stats_->DeregisterStatsObserver(&receive_side_cc_); |
| call_stats_->DeregisterStatsObserver(transport_send_->send_side_cc()); |
| |
| int64_t first_sent_packet_ms = |
| transport_send_->send_side_cc()->GetFirstPacketTimeMs(); |
| // Only update histograms after process threads have been shut down, so that |
| // they won't try to concurrently update stats. |
| { |
| rtc::CritScope lock(&bitrate_crit_); |
| UpdateSendHistograms(first_sent_packet_ms); |
| } |
| UpdateReceiveHistograms(); |
| UpdateHistograms(); |
| |
| Trace::ReturnTrace(); |
| } |
| |
| rtc::Optional<RtpPacketReceived> Call::ParseRtpPacket( |
| const uint8_t* packet, |
| size_t length, |
| const PacketTime* packet_time) const { |
| RtpPacketReceived parsed_packet; |
| if (!parsed_packet.Parse(packet, length)) |
| return rtc::Optional<RtpPacketReceived>(); |
| |
| int64_t arrival_time_ms; |
| if (packet_time && packet_time->timestamp != -1) { |
| arrival_time_ms = (packet_time->timestamp + 500) / 1000; |
| } else { |
| arrival_time_ms = clock_->TimeInMilliseconds(); |
| } |
| parsed_packet.set_arrival_time_ms(arrival_time_ms); |
| |
| return rtc::Optional<RtpPacketReceived>(std::move(parsed_packet)); |
| } |
| |
| void Call::UpdateHistograms() { |
| RTC_HISTOGRAM_COUNTS_100000( |
| "WebRTC.Call.LifetimeInSeconds", |
| (clock_->TimeInMilliseconds() - start_ms_) / 1000); |
| } |
| |
| void Call::UpdateSendHistograms(int64_t first_sent_packet_ms) { |
| if (first_sent_packet_ms == -1) |
| return; |
| if (!sent_rtp_audio_timer_ms_.Empty()) { |
| RTC_HISTOGRAM_COUNTS_100000( |
| "WebRTC.Call.TimeSendingAudioRtpPacketsInSeconds", |
| sent_rtp_audio_timer_ms_.Length() / 1000); |
| } |
| int64_t elapsed_sec = |
| (clock_->TimeInMilliseconds() - first_sent_packet_ms) / 1000; |
| if (elapsed_sec < metrics::kMinRunTimeInSeconds) |
| return; |
| const int kMinRequiredPeriodicSamples = 5; |
| AggregatedStats send_bitrate_stats = |
| estimated_send_bitrate_kbps_counter_.ProcessAndGetStats(); |
| if (send_bitrate_stats.num_samples > kMinRequiredPeriodicSamples) { |
| RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.EstimatedSendBitrateInKbps", |
| send_bitrate_stats.average); |
| LOG(LS_INFO) << "WebRTC.Call.EstimatedSendBitrateInKbps, " |
| << send_bitrate_stats.ToString(); |
| } |
| AggregatedStats pacer_bitrate_stats = |
| pacer_bitrate_kbps_counter_.ProcessAndGetStats(); |
| if (pacer_bitrate_stats.num_samples > kMinRequiredPeriodicSamples) { |
| RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.PacerBitrateInKbps", |
| pacer_bitrate_stats.average); |
| LOG(LS_INFO) << "WebRTC.Call.PacerBitrateInKbps, " |
| << pacer_bitrate_stats.ToString(); |
| } |
| } |
| |
| void Call::UpdateReceiveHistograms() { |
| if (first_received_rtp_audio_ms_) { |
| RTC_HISTOGRAM_COUNTS_100000( |
| "WebRTC.Call.TimeReceivingAudioRtpPacketsInSeconds", |
| (*last_received_rtp_audio_ms_ - *first_received_rtp_audio_ms_) / 1000); |
| } |
| if (first_received_rtp_video_ms_) { |
| RTC_HISTOGRAM_COUNTS_100000( |
| "WebRTC.Call.TimeReceivingVideoRtpPacketsInSeconds", |
| (*last_received_rtp_video_ms_ - *first_received_rtp_video_ms_) / 1000); |
| } |
| const int kMinRequiredPeriodicSamples = 5; |
| AggregatedStats video_bytes_per_sec = |
| received_video_bytes_per_second_counter_.GetStats(); |
| if (video_bytes_per_sec.num_samples > kMinRequiredPeriodicSamples) { |
| RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.VideoBitrateReceivedInKbps", |
| video_bytes_per_sec.average * 8 / 1000); |
| LOG(LS_INFO) << "WebRTC.Call.VideoBitrateReceivedInBps, " |
| << video_bytes_per_sec.ToStringWithMultiplier(8); |
| } |
| AggregatedStats audio_bytes_per_sec = |
| received_audio_bytes_per_second_counter_.GetStats(); |
| if (audio_bytes_per_sec.num_samples > kMinRequiredPeriodicSamples) { |
| RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.AudioBitrateReceivedInKbps", |
| audio_bytes_per_sec.average * 8 / 1000); |
| LOG(LS_INFO) << "WebRTC.Call.AudioBitrateReceivedInBps, " |
| << audio_bytes_per_sec.ToStringWithMultiplier(8); |
| } |
| AggregatedStats rtcp_bytes_per_sec = |
| received_rtcp_bytes_per_second_counter_.GetStats(); |
| if (rtcp_bytes_per_sec.num_samples > kMinRequiredPeriodicSamples) { |
| RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.RtcpBitrateReceivedInBps", |
| rtcp_bytes_per_sec.average * 8); |
| LOG(LS_INFO) << "WebRTC.Call.RtcpBitrateReceivedInBps, " |
| << rtcp_bytes_per_sec.ToStringWithMultiplier(8); |
| } |
| AggregatedStats recv_bytes_per_sec = |
| received_bytes_per_second_counter_.GetStats(); |
| if (recv_bytes_per_sec.num_samples > kMinRequiredPeriodicSamples) { |
| RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.BitrateReceivedInKbps", |
| recv_bytes_per_sec.average * 8 / 1000); |
| LOG(LS_INFO) << "WebRTC.Call.BitrateReceivedInBps, " |
| << recv_bytes_per_sec.ToStringWithMultiplier(8); |
| } |
| } |
| |
| PacketReceiver* Call::Receiver() { |
| RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_); |
| return this; |
| } |
| |
| webrtc::AudioSendStream* Call::CreateAudioSendStream( |
| const webrtc::AudioSendStream::Config& config) { |
| TRACE_EVENT0("webrtc", "Call::CreateAudioSendStream"); |
| RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_); |
| event_log_->LogAudioSendStreamConfig(*CreateRtcLogStreamConfig(config)); |
| |
| rtc::Optional<RtpState> suspended_rtp_state; |
| { |
| const auto& iter = suspended_audio_send_ssrcs_.find(config.rtp.ssrc); |
| if (iter != suspended_audio_send_ssrcs_.end()) { |
| suspended_rtp_state.emplace(iter->second); |
| } |
| } |
| |
| AudioSendStream* send_stream = new AudioSendStream( |
| config, config_.audio_state, &worker_queue_, transport_send_.get(), |
| bitrate_allocator_.get(), event_log_, call_stats_->rtcp_rtt_stats(), |
| suspended_rtp_state); |
| { |
| WriteLockScoped write_lock(*send_crit_); |
| RTC_DCHECK(audio_send_ssrcs_.find(config.rtp.ssrc) == |
| audio_send_ssrcs_.end()); |
| audio_send_ssrcs_[config.rtp.ssrc] = send_stream; |
| } |
| { |
| ReadLockScoped read_lock(*receive_crit_); |
| for (AudioReceiveStream* stream : audio_receive_streams_) { |
| if (stream->config().rtp.local_ssrc == config.rtp.ssrc) { |
| stream->AssociateSendStream(send_stream); |
| } |
| } |
| } |
| send_stream->SignalNetworkState(audio_network_state_); |
| UpdateAggregateNetworkState(); |
| return send_stream; |
| } |
| |
| void Call::DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) { |
| TRACE_EVENT0("webrtc", "Call::DestroyAudioSendStream"); |
| RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_); |
| RTC_DCHECK(send_stream != nullptr); |
| |
| send_stream->Stop(); |
| |
| const uint32_t ssrc = send_stream->GetConfig().rtp.ssrc; |
| webrtc::internal::AudioSendStream* audio_send_stream = |
| static_cast<webrtc::internal::AudioSendStream*>(send_stream); |
| suspended_audio_send_ssrcs_[ssrc] = audio_send_stream->GetRtpState(); |
| { |
| WriteLockScoped write_lock(*send_crit_); |
| size_t num_deleted = audio_send_ssrcs_.erase(ssrc); |
| RTC_DCHECK_EQ(1, num_deleted); |
| } |
| { |
| ReadLockScoped read_lock(*receive_crit_); |
| for (AudioReceiveStream* stream : audio_receive_streams_) { |
| if (stream->config().rtp.local_ssrc == ssrc) { |
| stream->AssociateSendStream(nullptr); |
| } |
| } |
| } |
| UpdateAggregateNetworkState(); |
| sent_rtp_audio_timer_ms_.Extend(audio_send_stream->GetActiveLifetime()); |
| delete send_stream; |
| } |
| |
| webrtc::AudioReceiveStream* Call::CreateAudioReceiveStream( |
| const webrtc::AudioReceiveStream::Config& config) { |
| TRACE_EVENT0("webrtc", "Call::CreateAudioReceiveStream"); |
| RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_); |
| event_log_->LogAudioReceiveStreamConfig(*CreateRtcLogStreamConfig(config)); |
| AudioReceiveStream* receive_stream = new AudioReceiveStream( |
| &audio_receiver_controller_, transport_send_->packet_router(), config, |
| config_.audio_state, event_log_); |
| { |
| WriteLockScoped write_lock(*receive_crit_); |
| receive_rtp_config_[config.rtp.remote_ssrc] = |
| ReceiveRtpConfig(config.rtp.extensions, UseSendSideBwe(config)); |
| audio_receive_streams_.insert(receive_stream); |
| |
| ConfigureSync(config.sync_group); |
| } |
| { |
| ReadLockScoped read_lock(*send_crit_); |
| auto it = audio_send_ssrcs_.find(config.rtp.local_ssrc); |
| if (it != audio_send_ssrcs_.end()) { |
| receive_stream->AssociateSendStream(it->second); |
| } |
| } |
| receive_stream->SignalNetworkState(audio_network_state_); |
| UpdateAggregateNetworkState(); |
| return receive_stream; |
| } |
| |
| void Call::DestroyAudioReceiveStream( |
| webrtc::AudioReceiveStream* receive_stream) { |
| TRACE_EVENT0("webrtc", "Call::DestroyAudioReceiveStream"); |
| RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_); |
| RTC_DCHECK(receive_stream != nullptr); |
| webrtc::internal::AudioReceiveStream* audio_receive_stream = |
| static_cast<webrtc::internal::AudioReceiveStream*>(receive_stream); |
| { |
| WriteLockScoped write_lock(*receive_crit_); |
| const AudioReceiveStream::Config& config = audio_receive_stream->config(); |
| uint32_t ssrc = config.rtp.remote_ssrc; |
| receive_side_cc_.GetRemoteBitrateEstimator(UseSendSideBwe(config)) |
| ->RemoveStream(ssrc); |
| audio_receive_streams_.erase(audio_receive_stream); |
| const std::string& sync_group = audio_receive_stream->config().sync_group; |
| const auto it = sync_stream_mapping_.find(sync_group); |
| if (it != sync_stream_mapping_.end() && |
| it->second == audio_receive_stream) { |
| sync_stream_mapping_.erase(it); |
| ConfigureSync(sync_group); |
| } |
| receive_rtp_config_.erase(ssrc); |
| } |
| UpdateAggregateNetworkState(); |
| delete audio_receive_stream; |
| } |
| |
| webrtc::VideoSendStream* Call::CreateVideoSendStream( |
| webrtc::VideoSendStream::Config config, |
| VideoEncoderConfig encoder_config) { |
| TRACE_EVENT0("webrtc", "Call::CreateVideoSendStream"); |
| RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_); |
| |
| video_send_delay_stats_->AddSsrcs(config); |
| for (size_t ssrc_index = 0; ssrc_index < config.rtp.ssrcs.size(); |
| ++ssrc_index) { |
| event_log_->LogVideoSendStreamConfig( |
| *CreateRtcLogStreamConfig(config, ssrc_index)); |
| } |
| |
| // TODO(mflodman): Base the start bitrate on a current bandwidth estimate, if |
| // the call has already started. |
| // Copy ssrcs from |config| since |config| is moved. |
| std::vector<uint32_t> ssrcs = config.rtp.ssrcs; |
| VideoSendStream* send_stream = new VideoSendStream( |
| num_cpu_cores_, module_process_thread_.get(), &worker_queue_, |
| call_stats_.get(), transport_send_.get(), bitrate_allocator_.get(), |
| video_send_delay_stats_.get(), event_log_, std::move(config), |
| std::move(encoder_config), suspended_video_send_ssrcs_); |
| |
| { |
| WriteLockScoped write_lock(*send_crit_); |
| for (uint32_t ssrc : ssrcs) { |
| RTC_DCHECK(video_send_ssrcs_.find(ssrc) == video_send_ssrcs_.end()); |
| video_send_ssrcs_[ssrc] = send_stream; |
| } |
| video_send_streams_.insert(send_stream); |
| } |
| send_stream->SignalNetworkState(video_network_state_); |
| UpdateAggregateNetworkState(); |
| |
| return send_stream; |
| } |
| |
| void Call::DestroyVideoSendStream(webrtc::VideoSendStream* send_stream) { |
| TRACE_EVENT0("webrtc", "Call::DestroyVideoSendStream"); |
| RTC_DCHECK(send_stream != nullptr); |
| RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_); |
| |
| send_stream->Stop(); |
| |
| VideoSendStream* send_stream_impl = nullptr; |
| { |
| WriteLockScoped write_lock(*send_crit_); |
| auto it = video_send_ssrcs_.begin(); |
| while (it != video_send_ssrcs_.end()) { |
| if (it->second == static_cast<VideoSendStream*>(send_stream)) { |
| send_stream_impl = it->second; |
| video_send_ssrcs_.erase(it++); |
| } else { |
| ++it; |
| } |
| } |
| video_send_streams_.erase(send_stream_impl); |
| } |
| RTC_CHECK(send_stream_impl != nullptr); |
| |
| VideoSendStream::RtpStateMap rtp_state = |
| send_stream_impl->StopPermanentlyAndGetRtpStates(); |
| |
| for (VideoSendStream::RtpStateMap::iterator it = rtp_state.begin(); |
| it != rtp_state.end(); ++it) { |
| suspended_video_send_ssrcs_[it->first] = it->second; |
| } |
| |
| UpdateAggregateNetworkState(); |
| delete send_stream_impl; |
| } |
| |
| webrtc::VideoReceiveStream* Call::CreateVideoReceiveStream( |
| webrtc::VideoReceiveStream::Config configuration) { |
| TRACE_EVENT0("webrtc", "Call::CreateVideoReceiveStream"); |
| RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_); |
| |
| VideoReceiveStream* receive_stream = new VideoReceiveStream( |
| &video_receiver_controller_, num_cpu_cores_, |
| transport_send_->packet_router(), std::move(configuration), |
| module_process_thread_.get(), call_stats_.get()); |
| |
| const webrtc::VideoReceiveStream::Config& config = receive_stream->config(); |
| ReceiveRtpConfig receive_config(config.rtp.extensions, |
| UseSendSideBwe(config)); |
| { |
| WriteLockScoped write_lock(*receive_crit_); |
| if (config.rtp.rtx_ssrc) { |
| // We record identical config for the rtx stream as for the main |
| // stream. Since the transport_send_cc negotiation is per payload |
| // type, we may get an incorrect value for the rtx stream, but |
| // that is unlikely to matter in practice. |
| receive_rtp_config_[config.rtp.rtx_ssrc] = receive_config; |
| } |
| receive_rtp_config_[config.rtp.remote_ssrc] = receive_config; |
| video_receive_streams_.insert(receive_stream); |
| ConfigureSync(config.sync_group); |
| } |
| receive_stream->SignalNetworkState(video_network_state_); |
| UpdateAggregateNetworkState(); |
| event_log_->LogVideoReceiveStreamConfig(*CreateRtcLogStreamConfig(config)); |
| return receive_stream; |
| } |
| |
| void Call::DestroyVideoReceiveStream( |
| webrtc::VideoReceiveStream* receive_stream) { |
| TRACE_EVENT0("webrtc", "Call::DestroyVideoReceiveStream"); |
| RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_); |
| RTC_DCHECK(receive_stream != nullptr); |
| VideoReceiveStream* receive_stream_impl = |
| static_cast<VideoReceiveStream*>(receive_stream); |
| const VideoReceiveStream::Config& config = receive_stream_impl->config(); |
| { |
| WriteLockScoped write_lock(*receive_crit_); |
| // Remove all ssrcs pointing to a receive stream. As RTX retransmits on a |
| // separate SSRC there can be either one or two. |
| receive_rtp_config_.erase(config.rtp.remote_ssrc); |
| if (config.rtp.rtx_ssrc) { |
| receive_rtp_config_.erase(config.rtp.rtx_ssrc); |
| } |
| video_receive_streams_.erase(receive_stream_impl); |
| ConfigureSync(config.sync_group); |
| } |
| |
| receive_side_cc_.GetRemoteBitrateEstimator(UseSendSideBwe(config)) |
| ->RemoveStream(config.rtp.remote_ssrc); |
| |
| UpdateAggregateNetworkState(); |
| delete receive_stream_impl; |
| } |
| |
| FlexfecReceiveStream* Call::CreateFlexfecReceiveStream( |
| const FlexfecReceiveStream::Config& config) { |
| TRACE_EVENT0("webrtc", "Call::CreateFlexfecReceiveStream"); |
| RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_); |
| |
| RecoveredPacketReceiver* recovered_packet_receiver = this; |
| |
| FlexfecReceiveStreamImpl* receive_stream; |
| { |
| WriteLockScoped write_lock(*receive_crit_); |
| // Unlike the video and audio receive streams, |
| // FlexfecReceiveStream implements RtpPacketSinkInterface itself, |
| // and hence its constructor passes its |this| pointer to |
| // video_receiver_controller_->CreateStream(). Calling the |
| // constructor while holding |receive_crit_| ensures that we don't |
| // call OnRtpPacket until the constructor is finished and the |
| // object is in a valid state. |
| // TODO(nisse): Fix constructor so that it can be moved outside of |
| // this locked scope. |
| receive_stream = new FlexfecReceiveStreamImpl( |
| &video_receiver_controller_, config, recovered_packet_receiver, |
| call_stats_->rtcp_rtt_stats(), module_process_thread_.get()); |
| |
| RTC_DCHECK(receive_rtp_config_.find(config.remote_ssrc) == |
| receive_rtp_config_.end()); |
| receive_rtp_config_[config.remote_ssrc] = |
| ReceiveRtpConfig(config.rtp_header_extensions, UseSendSideBwe(config)); |
| } |
| |
| // TODO(brandtr): Store config in RtcEventLog here. |
| |
| return receive_stream; |
| } |
| |
| void Call::DestroyFlexfecReceiveStream(FlexfecReceiveStream* receive_stream) { |
| TRACE_EVENT0("webrtc", "Call::DestroyFlexfecReceiveStream"); |
| RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_); |
| |
| RTC_DCHECK(receive_stream != nullptr); |
| { |
| WriteLockScoped write_lock(*receive_crit_); |
| |
| const FlexfecReceiveStream::Config& config = receive_stream->GetConfig(); |
| uint32_t ssrc = config.remote_ssrc; |
| receive_rtp_config_.erase(ssrc); |
| |
| // Remove all SSRCs pointing to the FlexfecReceiveStreamImpl to be |
| // destroyed. |
| receive_side_cc_.GetRemoteBitrateEstimator(UseSendSideBwe(config)) |
| ->RemoveStream(ssrc); |
| } |
| |
| delete receive_stream; |
| } |
| |
| Call::Stats Call::GetStats() const { |
| // TODO(solenberg): Some test cases in EndToEndTest use this from a different |
| // thread. Re-enable once that is fixed. |
| // RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_); |
| Stats stats; |
| // Fetch available send/receive bitrates. |
| uint32_t send_bandwidth = 0; |
| transport_send_->send_side_cc()->GetBitrateController()->AvailableBandwidth( |
| &send_bandwidth); |
| std::vector<unsigned int> ssrcs; |
| uint32_t recv_bandwidth = 0; |
| receive_side_cc_.GetRemoteBitrateEstimator(false)->LatestEstimate( |
| &ssrcs, &recv_bandwidth); |
| stats.send_bandwidth_bps = send_bandwidth; |
| stats.recv_bandwidth_bps = recv_bandwidth; |
| stats.pacer_delay_ms = |
| transport_send_->send_side_cc()->GetPacerQueuingDelayMs(); |
| stats.rtt_ms = call_stats_->rtcp_rtt_stats()->LastProcessedRtt(); |
| { |
| rtc::CritScope cs(&bitrate_crit_); |
| stats.max_padding_bitrate_bps = configured_max_padding_bitrate_bps_; |
| } |
| return stats; |
| } |
| |
| void Call::SetBitrateConfig( |
| const webrtc::Call::Config::BitrateConfig& bitrate_config) { |
| TRACE_EVENT0("webrtc", "Call::SetBitrateConfig"); |
| RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_); |
| RTC_DCHECK_GE(bitrate_config.min_bitrate_bps, 0); |
| RTC_DCHECK_NE(bitrate_config.start_bitrate_bps, 0); |
| if (bitrate_config.max_bitrate_bps != -1) { |
| RTC_DCHECK_GT(bitrate_config.max_bitrate_bps, 0); |
| } |
| |
| rtc::Optional<int> new_start; |
| // Only update the "start" bitrate if it's set, and different from the old |
| // value. In practice, this value comes from the x-google-start-bitrate codec |
| // parameter in SDP, and setting the same remote description twice shouldn't |
| // restart bandwidth estimation. |
| if (bitrate_config.start_bitrate_bps != -1 && |
| bitrate_config.start_bitrate_bps != |
| base_bitrate_config_.start_bitrate_bps) { |
| new_start.emplace(bitrate_config.start_bitrate_bps); |
| } |
| base_bitrate_config_ = bitrate_config; |
| UpdateCurrentBitrateConfig(new_start); |
| } |
| |
| void Call::SetBitrateConfigMask( |
| const webrtc::Call::Config::BitrateConfigMask& mask) { |
| TRACE_EVENT0("webrtc", "Call::SetBitrateConfigMask"); |
| RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_); |
| |
| bitrate_config_mask_ = mask; |
| UpdateCurrentBitrateConfig(mask.start_bitrate_bps); |
| } |
| |
| void Call::UpdateCurrentBitrateConfig(const rtc::Optional<int>& new_start) { |
| Config::BitrateConfig updated; |
| updated.min_bitrate_bps = |
| std::max(bitrate_config_mask_.min_bitrate_bps.value_or(0), |
| base_bitrate_config_.min_bitrate_bps); |
| |
| updated.max_bitrate_bps = |
| MinPositive(bitrate_config_mask_.max_bitrate_bps.value_or(-1), |
| base_bitrate_config_.max_bitrate_bps); |
| |
| // If the combined min ends up greater than the combined max, the max takes |
| // priority. |
| if (updated.max_bitrate_bps != -1 && |
| updated.min_bitrate_bps > updated.max_bitrate_bps) { |
| updated.min_bitrate_bps = updated.max_bitrate_bps; |
| } |
| |
| // If there is nothing to update (min/max unchanged, no new bandwidth |
| // estimation start value), return early. |
| if (updated.min_bitrate_bps == config_.bitrate_config.min_bitrate_bps && |
| updated.max_bitrate_bps == config_.bitrate_config.max_bitrate_bps && |
| !new_start) { |
| LOG(LS_VERBOSE) << "WebRTC.Call.UpdateCurrentBitrateConfig: " |
| << "nothing to update"; |
| return; |
| } |
| |
| if (new_start) { |
| // Clamp start by min and max. |
| updated.start_bitrate_bps = MinPositive( |
| std::max(*new_start, updated.min_bitrate_bps), updated.max_bitrate_bps); |
| } else { |
| updated.start_bitrate_bps = -1; |
| } |
| |
| LOG(INFO) << "WebRTC.Call.UpdateCurrentBitrateConfig: " |
| << "calling SetBweBitrates with args (" << updated.min_bitrate_bps |
| << ", " << updated.start_bitrate_bps << ", " |
| << updated.max_bitrate_bps << ")"; |
| transport_send_->send_side_cc()->SetBweBitrates(updated.min_bitrate_bps, |
| updated.start_bitrate_bps, |
| updated.max_bitrate_bps); |
| if (!new_start) { |
| updated.start_bitrate_bps = config_.bitrate_config.start_bitrate_bps; |
| } |
| config_.bitrate_config = updated; |
| } |
| |
| void Call::SignalChannelNetworkState(MediaType media, NetworkState state) { |
| RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_); |
| switch (media) { |
| case MediaType::AUDIO: |
| audio_network_state_ = state; |
| break; |
| case MediaType::VIDEO: |
| video_network_state_ = state; |
| break; |
| case MediaType::ANY: |
| case MediaType::DATA: |
| RTC_NOTREACHED(); |
| break; |
| } |
| |
| UpdateAggregateNetworkState(); |
| { |
| ReadLockScoped read_lock(*send_crit_); |
| for (auto& kv : audio_send_ssrcs_) { |
| kv.second->SignalNetworkState(audio_network_state_); |
| } |
| for (auto& kv : video_send_ssrcs_) { |
| kv.second->SignalNetworkState(video_network_state_); |
| } |
| } |
| { |
| ReadLockScoped read_lock(*receive_crit_); |
| for (AudioReceiveStream* audio_receive_stream : audio_receive_streams_) { |
| audio_receive_stream->SignalNetworkState(audio_network_state_); |
| } |
| for (VideoReceiveStream* video_receive_stream : video_receive_streams_) { |
| video_receive_stream->SignalNetworkState(video_network_state_); |
| } |
| } |
| } |
| |
| void Call::OnTransportOverheadChanged(MediaType media, |
| int transport_overhead_per_packet) { |
| switch (media) { |
| case MediaType::AUDIO: { |
| ReadLockScoped read_lock(*send_crit_); |
| for (auto& kv : audio_send_ssrcs_) { |
| kv.second->SetTransportOverhead(transport_overhead_per_packet); |
| } |
| break; |
| } |
| case MediaType::VIDEO: { |
| ReadLockScoped read_lock(*send_crit_); |
| for (auto& kv : video_send_ssrcs_) { |
| kv.second->SetTransportOverhead(transport_overhead_per_packet); |
| } |
| break; |
| } |
| case MediaType::ANY: |
| case MediaType::DATA: |
| RTC_NOTREACHED(); |
| break; |
| } |
| } |
| |
| // TODO(honghaiz): Add tests for this method. |
| void Call::OnNetworkRouteChanged(const std::string& transport_name, |
| const rtc::NetworkRoute& network_route) { |
| RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_); |
| // Check if the network route is connected. |
| if (!network_route.connected) { |
| LOG(LS_INFO) << "Transport " << transport_name << " is disconnected"; |
| // TODO(honghaiz): Perhaps handle this in SignalChannelNetworkState and |
| // consider merging these two methods. |
| return; |
| } |
| |
| // Check whether the network route has changed on each transport. |
| auto result = |
| network_routes_.insert(std::make_pair(transport_name, network_route)); |
| auto kv = result.first; |
| bool inserted = result.second; |
| if (inserted) { |
| // No need to reset BWE if this is the first time the network connects. |
| return; |
| } |
| if (kv->second != network_route) { |
| kv->second = network_route; |
| LOG(LS_INFO) << "Network route changed on transport " << transport_name |
| << ": new local network id " << network_route.local_network_id |
| << " new remote network id " << network_route.remote_network_id |
| << " Reset bitrates to min: " |
| << config_.bitrate_config.min_bitrate_bps |
| << " bps, start: " << config_.bitrate_config.start_bitrate_bps |
| << " bps, max: " << config_.bitrate_config.start_bitrate_bps |
| << " bps."; |
| RTC_DCHECK_GT(config_.bitrate_config.start_bitrate_bps, 0); |
| transport_send_->send_side_cc()->OnNetworkRouteChanged( |
| network_route, config_.bitrate_config.start_bitrate_bps, |
| config_.bitrate_config.min_bitrate_bps, |
| config_.bitrate_config.max_bitrate_bps); |
| } |
| } |
| |
| void Call::UpdateAggregateNetworkState() { |
| RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_); |
| |
| bool have_audio = false; |
| bool have_video = false; |
| { |
| ReadLockScoped read_lock(*send_crit_); |
| if (audio_send_ssrcs_.size() > 0) |
| have_audio = true; |
| if (video_send_ssrcs_.size() > 0) |
| have_video = true; |
| } |
| { |
| ReadLockScoped read_lock(*receive_crit_); |
| if (audio_receive_streams_.size() > 0) |
| have_audio = true; |
| if (video_receive_streams_.size() > 0) |
| have_video = true; |
| } |
| |
| NetworkState aggregate_state = kNetworkDown; |
| if ((have_video && video_network_state_ == kNetworkUp) || |
| (have_audio && audio_network_state_ == kNetworkUp)) { |
| aggregate_state = kNetworkUp; |
| } |
| |
| LOG(LS_INFO) << "UpdateAggregateNetworkState: aggregate_state=" |
| << (aggregate_state == kNetworkUp ? "up" : "down"); |
| |
| transport_send_->send_side_cc()->SignalNetworkState(aggregate_state); |
| } |
| |
| void Call::OnSentPacket(const rtc::SentPacket& sent_packet) { |
| video_send_delay_stats_->OnSentPacket(sent_packet.packet_id, |
| clock_->TimeInMilliseconds()); |
| transport_send_->send_side_cc()->OnSentPacket(sent_packet); |
| } |
| |
| void Call::OnNetworkChanged(uint32_t target_bitrate_bps, |
| uint8_t fraction_loss, |
| int64_t rtt_ms, |
| int64_t probing_interval_ms) { |
| // TODO(perkj): Consider making sure CongestionController operates on |
| // |worker_queue_|. |
| if (!worker_queue_.IsCurrent()) { |
| worker_queue_.PostTask( |
| [this, target_bitrate_bps, fraction_loss, rtt_ms, probing_interval_ms] { |
| OnNetworkChanged(target_bitrate_bps, fraction_loss, rtt_ms, |
| probing_interval_ms); |
| }); |
| return; |
| } |
| RTC_DCHECK_RUN_ON(&worker_queue_); |
| // For controlling the rate of feedback messages. |
| receive_side_cc_.OnBitrateChanged(target_bitrate_bps); |
| bitrate_allocator_->OnNetworkChanged(target_bitrate_bps, fraction_loss, |
| rtt_ms, probing_interval_ms); |
| |
| // Ignore updates if bitrate is zero (the aggregate network state is down). |
| if (target_bitrate_bps == 0) { |
| rtc::CritScope lock(&bitrate_crit_); |
| estimated_send_bitrate_kbps_counter_.ProcessAndPause(); |
| pacer_bitrate_kbps_counter_.ProcessAndPause(); |
| return; |
| } |
| |
| bool sending_video; |
| { |
| ReadLockScoped read_lock(*send_crit_); |
| sending_video = !video_send_streams_.empty(); |
| } |
| |
| rtc::CritScope lock(&bitrate_crit_); |
| if (!sending_video) { |
| // Do not update the stats if we are not sending video. |
| estimated_send_bitrate_kbps_counter_.ProcessAndPause(); |
| pacer_bitrate_kbps_counter_.ProcessAndPause(); |
| return; |
| } |
| estimated_send_bitrate_kbps_counter_.Add(target_bitrate_bps / 1000); |
| // Pacer bitrate may be higher than bitrate estimate if enforcing min bitrate. |
| uint32_t pacer_bitrate_bps = |
| std::max(target_bitrate_bps, min_allocated_send_bitrate_bps_); |
| pacer_bitrate_kbps_counter_.Add(pacer_bitrate_bps / 1000); |
| } |
| |
| void Call::OnAllocationLimitsChanged(uint32_t min_send_bitrate_bps, |
| uint32_t max_padding_bitrate_bps) { |
| transport_send_->SetAllocatedSendBitrateLimits(min_send_bitrate_bps, |
| max_padding_bitrate_bps); |
| rtc::CritScope lock(&bitrate_crit_); |
| min_allocated_send_bitrate_bps_ = min_send_bitrate_bps; |
| configured_max_padding_bitrate_bps_ = max_padding_bitrate_bps; |
| } |
| |
| void Call::ConfigureSync(const std::string& sync_group) { |
| // Set sync only if there was no previous one. |
| if (sync_group.empty()) |
| return; |
| |
| AudioReceiveStream* sync_audio_stream = nullptr; |
| // Find existing audio stream. |
| const auto it = sync_stream_mapping_.find(sync_group); |
| if (it != sync_stream_mapping_.end()) { |
| sync_audio_stream = it->second; |
| } else { |
| // No configured audio stream, see if we can find one. |
| for (AudioReceiveStream* stream : audio_receive_streams_) { |
| if (stream->config().sync_group == sync_group) { |
| if (sync_audio_stream != nullptr) { |
| LOG(LS_WARNING) << "Attempting to sync more than one audio stream " |
| "within the same sync group. This is not " |
| "supported in the current implementation."; |
| break; |
| } |
| sync_audio_stream = stream; |
| } |
| } |
| } |
| if (sync_audio_stream) |
| sync_stream_mapping_[sync_group] = sync_audio_stream; |
| size_t num_synced_streams = 0; |
| for (VideoReceiveStream* video_stream : video_receive_streams_) { |
| if (video_stream->config().sync_group != sync_group) |
| continue; |
| ++num_synced_streams; |
| if (num_synced_streams > 1) { |
| // TODO(pbos): Support synchronizing more than one A/V pair. |
| // https://code.google.com/p/webrtc/issues/detail?id=4762 |
| LOG(LS_WARNING) << "Attempting to sync more than one audio/video pair " |
| "within the same sync group. This is not supported in " |
| "the current implementation."; |
| } |
| // Only sync the first A/V pair within this sync group. |
| if (num_synced_streams == 1) { |
| // sync_audio_stream may be null and that's ok. |
| video_stream->SetSync(sync_audio_stream); |
| } else { |
| video_stream->SetSync(nullptr); |
| } |
| } |
| } |
| |
| PacketReceiver::DeliveryStatus Call::DeliverRtcp(MediaType media_type, |
| const uint8_t* packet, |
| size_t length) { |
| TRACE_EVENT0("webrtc", "Call::DeliverRtcp"); |
| // TODO(pbos): Make sure it's a valid packet. |
| // Return DELIVERY_UNKNOWN_SSRC if it can be determined that |
| // there's no receiver of the packet. |
| if (received_bytes_per_second_counter_.HasSample()) { |
| // First RTP packet has been received. |
| received_bytes_per_second_counter_.Add(static_cast<int>(length)); |
| received_rtcp_bytes_per_second_counter_.Add(static_cast<int>(length)); |
| } |
| bool rtcp_delivered = false; |
| if (media_type == MediaType::ANY || media_type == MediaType::VIDEO) { |
| ReadLockScoped read_lock(*receive_crit_); |
| for (VideoReceiveStream* stream : video_receive_streams_) { |
| if (stream->DeliverRtcp(packet, length)) |
| rtcp_delivered = true; |
| } |
| } |
| if (media_type == MediaType::ANY || media_type == MediaType::AUDIO) { |
| ReadLockScoped read_lock(*receive_crit_); |
| for (AudioReceiveStream* stream : audio_receive_streams_) { |
| if (stream->DeliverRtcp(packet, length)) |
| rtcp_delivered = true; |
| } |
| } |
| if (media_type == MediaType::ANY || media_type == MediaType::VIDEO) { |
| ReadLockScoped read_lock(*send_crit_); |
| for (VideoSendStream* stream : video_send_streams_) { |
| if (stream->DeliverRtcp(packet, length)) |
| rtcp_delivered = true; |
| } |
| } |
| if (media_type == MediaType::ANY || media_type == MediaType::AUDIO) { |
| ReadLockScoped read_lock(*send_crit_); |
| for (auto& kv : audio_send_ssrcs_) { |
| if (kv.second->DeliverRtcp(packet, length)) |
| rtcp_delivered = true; |
| } |
| } |
| |
| if (rtcp_delivered) |
| event_log_->LogRtcpPacket(kIncomingPacket, packet, length); |
| |
| return rtcp_delivered ? DELIVERY_OK : DELIVERY_PACKET_ERROR; |
| } |
| |
| PacketReceiver::DeliveryStatus Call::DeliverRtp(MediaType media_type, |
| const uint8_t* packet, |
| size_t length, |
| const PacketTime& packet_time) { |
| TRACE_EVENT0("webrtc", "Call::DeliverRtp"); |
| |
| // TODO(nisse): We should parse the RTP header only here, and pass |
| // on parsed_packet to the receive streams. |
| rtc::Optional<RtpPacketReceived> parsed_packet = |
| ParseRtpPacket(packet, length, &packet_time); |
| |
| // We might get RTP keep-alive packets in accordance with RFC6263 section 4.6. |
| // These are empty (zero length payload) RTP packets with an unsignaled |
| // payload type. |
| const bool is_keep_alive_packet = |
| parsed_packet && parsed_packet->payload_size() == 0; |
| |
| RTC_DCHECK(media_type == MediaType::AUDIO || media_type == MediaType::VIDEO || |
| is_keep_alive_packet); |
| |
| if (!parsed_packet) |
| return DELIVERY_PACKET_ERROR; |
| |
| ReadLockScoped read_lock(*receive_crit_); |
| auto it = receive_rtp_config_.find(parsed_packet->Ssrc()); |
| if (it == receive_rtp_config_.end()) { |
| LOG(LS_ERROR) << "receive_rtp_config_ lookup failed for ssrc " |
| << parsed_packet->Ssrc(); |
| // Destruction of the receive stream, including deregistering from the |
| // RtpDemuxer, is not protected by the |receive_crit_| lock. But |
| // deregistering in the |receive_rtp_config_| map is protected by that lock. |
| // So by not passing the packet on to demuxing in this case, we prevent |
| // incoming packets to be passed on via the demuxer to a receive stream |
| // which is being torned down. |
| return DELIVERY_UNKNOWN_SSRC; |
| } |
| parsed_packet->IdentifyExtensions(it->second.extensions); |
| |
| NotifyBweOfReceivedPacket(*parsed_packet, media_type); |
| |
| if (media_type == MediaType::AUDIO) { |
| if (audio_receiver_controller_.OnRtpPacket(*parsed_packet)) { |
| received_bytes_per_second_counter_.Add(static_cast<int>(length)); |
| received_audio_bytes_per_second_counter_.Add(static_cast<int>(length)); |
| event_log_->LogRtpHeader(kIncomingPacket, packet, length); |
| const int64_t arrival_time_ms = parsed_packet->arrival_time_ms(); |
| if (!first_received_rtp_audio_ms_) { |
| first_received_rtp_audio_ms_.emplace(arrival_time_ms); |
| } |
| last_received_rtp_audio_ms_.emplace(arrival_time_ms); |
| return DELIVERY_OK; |
| } |
| } else if (media_type == MediaType::VIDEO) { |
| if (video_receiver_controller_.OnRtpPacket(*parsed_packet)) { |
| received_bytes_per_second_counter_.Add(static_cast<int>(length)); |
| received_video_bytes_per_second_counter_.Add(static_cast<int>(length)); |
| event_log_->LogRtpHeader(kIncomingPacket, packet, length); |
| const int64_t arrival_time_ms = parsed_packet->arrival_time_ms(); |
| if (!first_received_rtp_video_ms_) { |
| first_received_rtp_video_ms_.emplace(arrival_time_ms); |
| } |
| last_received_rtp_video_ms_.emplace(arrival_time_ms); |
| return DELIVERY_OK; |
| } |
| } |
| return DELIVERY_UNKNOWN_SSRC; |
| } |
| |
| PacketReceiver::DeliveryStatus Call::DeliverPacket( |
| MediaType media_type, |
| const uint8_t* packet, |
| size_t length, |
| const PacketTime& packet_time) { |
| RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_); |
| if (RtpHeaderParser::IsRtcp(packet, length)) |
| return DeliverRtcp(media_type, packet, length); |
| |
| return DeliverRtp(media_type, packet, length, packet_time); |
| } |
| |
| void Call::OnRecoveredPacket(const uint8_t* packet, size_t length) { |
| rtc::Optional<RtpPacketReceived> parsed_packet = |
| ParseRtpPacket(packet, length, nullptr); |
| if (!parsed_packet) |
| return; |
| |
| parsed_packet->set_recovered(true); |
| |
| ReadLockScoped read_lock(*receive_crit_); |
| auto it = receive_rtp_config_.find(parsed_packet->Ssrc()); |
| if (it == receive_rtp_config_.end()) { |
| LOG(LS_ERROR) << "receive_rtp_config_ lookup failed for ssrc " |
| << parsed_packet->Ssrc(); |
| // Destruction of the receive stream, including deregistering from the |
| // RtpDemuxer, is not protected by the |receive_crit_| lock. But |
| // deregistering in the |receive_rtp_config_| map is protected by that lock. |
| // So by not passing the packet on to demuxing in this case, we prevent |
| // incoming packets to be passed on via the demuxer to a receive stream |
| // which is being torned down. |
| return; |
| } |
| parsed_packet->IdentifyExtensions(it->second.extensions); |
| |
| // TODO(brandtr): Update here when we support protecting audio packets too. |
| video_receiver_controller_.OnRtpPacket(*parsed_packet); |
| } |
| |
| void Call::NotifyBweOfReceivedPacket(const RtpPacketReceived& packet, |
| MediaType media_type) { |
| auto it = receive_rtp_config_.find(packet.Ssrc()); |
| bool use_send_side_bwe = |
| (it != receive_rtp_config_.end()) && it->second.use_send_side_bwe; |
| |
| RTPHeader header; |
| packet.GetHeader(&header); |
| |
| if (!use_send_side_bwe && header.extension.hasTransportSequenceNumber) { |
| // Inconsistent configuration of send side BWE. Do nothing. |
| // TODO(nisse): Without this check, we may produce RTCP feedback |
| // packets even when not negotiated. But it would be cleaner to |
| // move the check down to RTCPSender::SendFeedbackPacket, which |
| // would also help the PacketRouter to select an appropriate rtp |
| // module in the case that some, but not all, have RTCP feedback |
| // enabled. |
| return; |
| } |
| // For audio, we only support send side BWE. |
| if (media_type == MediaType::VIDEO || |
| (use_send_side_bwe && header.extension.hasTransportSequenceNumber)) { |
| receive_side_cc_.OnReceivedPacket( |
| packet.arrival_time_ms(), packet.payload_size() + packet.padding_size(), |
| header); |
| } |
| } |
| |
| } // namespace internal |
| |
| } // namespace webrtc |