| /* |
| * 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 "video/video_send_stream.h" |
| |
| #include <algorithm> |
| #include <cmath> |
| #include <sstream> |
| #include <string> |
| #include <utility> |
| #include <vector> |
| |
| #include "call/rtp_transport_controller_send_interface.h" |
| #include "common_types.h" // NOLINT(build/include) |
| #include "common_video/include/video_bitrate_allocator.h" |
| #include "modules/bitrate_controller/include/bitrate_controller.h" |
| #include "modules/congestion_controller/include/send_side_congestion_controller.h" |
| #include "modules/pacing/alr_detector.h" |
| #include "modules/pacing/packet_router.h" |
| #include "modules/rtp_rtcp/include/rtp_rtcp.h" |
| #include "modules/rtp_rtcp/source/rtp_sender.h" |
| #include "modules/utility/include/process_thread.h" |
| #include "modules/video_coding/utility/ivf_file_writer.h" |
| #include "rtc_base/checks.h" |
| #include "rtc_base/file.h" |
| #include "rtc_base/location.h" |
| #include "rtc_base/logging.h" |
| #include "rtc_base/trace_event.h" |
| #include "rtc_base/weak_ptr.h" |
| #include "system_wrappers/include/field_trial.h" |
| #include "video/call_stats.h" |
| #include "video/payload_router.h" |
| #include "call/video_send_stream.h" |
| |
| namespace webrtc { |
| |
| static const int kMinSendSidePacketHistorySize = 600; |
| namespace { |
| |
| // We don't do MTU discovery, so assume that we have the standard ethernet MTU. |
| const size_t kPathMTU = 1500; |
| |
| std::vector<RtpRtcp*> CreateRtpRtcpModules( |
| Transport* outgoing_transport, |
| RtcpIntraFrameObserver* intra_frame_callback, |
| RtcpBandwidthObserver* bandwidth_callback, |
| RtpTransportControllerSendInterface* transport, |
| RtcpRttStats* rtt_stats, |
| FlexfecSender* flexfec_sender, |
| SendStatisticsProxy* stats_proxy, |
| SendDelayStats* send_delay_stats, |
| RtcEventLog* event_log, |
| RateLimiter* retransmission_rate_limiter, |
| OverheadObserver* overhead_observer, |
| size_t num_modules, |
| RtpKeepAliveConfig keepalive_config) { |
| RTC_DCHECK_GT(num_modules, 0); |
| RtpRtcp::Configuration configuration; |
| configuration.audio = false; |
| configuration.receiver_only = false; |
| configuration.flexfec_sender = flexfec_sender; |
| configuration.outgoing_transport = outgoing_transport; |
| configuration.intra_frame_callback = intra_frame_callback; |
| configuration.bandwidth_callback = bandwidth_callback; |
| configuration.transport_feedback_callback = |
| transport->transport_feedback_observer(); |
| configuration.rtt_stats = rtt_stats; |
| configuration.rtcp_packet_type_counter_observer = stats_proxy; |
| configuration.paced_sender = transport->packet_sender(); |
| configuration.transport_sequence_number_allocator = |
| transport->packet_router(); |
| configuration.send_bitrate_observer = stats_proxy; |
| configuration.send_frame_count_observer = stats_proxy; |
| configuration.send_side_delay_observer = stats_proxy; |
| configuration.send_packet_observer = send_delay_stats; |
| configuration.event_log = event_log; |
| configuration.retransmission_rate_limiter = retransmission_rate_limiter; |
| configuration.overhead_observer = overhead_observer; |
| configuration.keepalive_config = keepalive_config; |
| std::vector<RtpRtcp*> modules; |
| for (size_t i = 0; i < num_modules; ++i) { |
| RtpRtcp* rtp_rtcp = RtpRtcp::CreateRtpRtcp(configuration); |
| rtp_rtcp->SetSendingStatus(false); |
| rtp_rtcp->SetSendingMediaStatus(false); |
| rtp_rtcp->SetRTCPStatus(RtcpMode::kCompound); |
| modules.push_back(rtp_rtcp); |
| } |
| return modules; |
| } |
| |
| // TODO(brandtr): Update this function when we support multistream protection. |
| std::unique_ptr<FlexfecSender> MaybeCreateFlexfecSender( |
| const VideoSendStream::Config& config, |
| const std::map<uint32_t, RtpState>& suspended_ssrcs) { |
| if (config.rtp.flexfec.payload_type < 0) { |
| return nullptr; |
| } |
| RTC_DCHECK_GE(config.rtp.flexfec.payload_type, 0); |
| RTC_DCHECK_LE(config.rtp.flexfec.payload_type, 127); |
| if (config.rtp.flexfec.ssrc == 0) { |
| RTC_LOG(LS_WARNING) << "FlexFEC is enabled, but no FlexFEC SSRC given. " |
| "Therefore disabling FlexFEC."; |
| return nullptr; |
| } |
| if (config.rtp.flexfec.protected_media_ssrcs.empty()) { |
| RTC_LOG(LS_WARNING) |
| << "FlexFEC is enabled, but no protected media SSRC given. " |
| "Therefore disabling FlexFEC."; |
| return nullptr; |
| } |
| |
| if (config.rtp.ssrcs.size() > 1) { |
| RTC_LOG(LS_WARNING) |
| << "Both FlexFEC and simulcast are enabled. This " |
| "combination is however not supported by our current " |
| "FlexFEC implementation. Therefore disabling FlexFEC."; |
| return nullptr; |
| } |
| |
| if (config.rtp.flexfec.protected_media_ssrcs.size() > 1) { |
| RTC_LOG(LS_WARNING) |
| << "The supplied FlexfecConfig contained multiple protected " |
| "media streams, but our implementation currently only " |
| "supports protecting a single media stream. " |
| "To avoid confusion, disabling FlexFEC completely."; |
| return nullptr; |
| } |
| |
| const RtpState* rtp_state = nullptr; |
| auto it = suspended_ssrcs.find(config.rtp.flexfec.ssrc); |
| if (it != suspended_ssrcs.end()) { |
| rtp_state = &it->second; |
| } |
| |
| RTC_DCHECK_EQ(1U, config.rtp.flexfec.protected_media_ssrcs.size()); |
| return std::unique_ptr<FlexfecSender>(new FlexfecSender( |
| config.rtp.flexfec.payload_type, config.rtp.flexfec.ssrc, |
| config.rtp.flexfec.protected_media_ssrcs[0], config.rtp.extensions, |
| RTPSender::FecExtensionSizes(), rtp_state, Clock::GetRealTimeClock())); |
| } |
| |
| bool TransportSeqNumExtensionConfigured(const VideoSendStream::Config& config) { |
| const std::vector<RtpExtension>& extensions = config.rtp.extensions; |
| return std::find_if( |
| extensions.begin(), extensions.end(), [](const RtpExtension& ext) { |
| return ext.uri == RtpExtension::kTransportSequenceNumberUri; |
| }) != extensions.end(); |
| } |
| |
| const char kForcedFallbackFieldTrial[] = |
| "WebRTC-VP8-Forced-Fallback-Encoder-v2"; |
| |
| rtc::Optional<int> GetFallbackMinBpsFromFieldTrial() { |
| if (!webrtc::field_trial::IsEnabled(kForcedFallbackFieldTrial)) |
| return rtc::nullopt; |
| |
| std::string group = |
| webrtc::field_trial::FindFullName(kForcedFallbackFieldTrial); |
| if (group.empty()) |
| return rtc::nullopt; |
| |
| int min_pixels; |
| int max_pixels; |
| int min_bps; |
| if (sscanf(group.c_str(), "Enabled-%d,%d,%d", &min_pixels, &max_pixels, |
| &min_bps) != 3) { |
| return rtc::nullopt; |
| } |
| |
| if (min_bps <= 0) |
| return rtc::nullopt; |
| |
| return min_bps; |
| } |
| |
| int GetEncoderMinBitrateBps() { |
| const int kDefaultEncoderMinBitrateBps = 30000; |
| return GetFallbackMinBpsFromFieldTrial().value_or( |
| kDefaultEncoderMinBitrateBps); |
| } |
| |
| bool PayloadTypeSupportsSkippingFecPackets(const std::string& payload_name) { |
| const VideoCodecType codecType = PayloadStringToCodecType(payload_name); |
| if (codecType == kVideoCodecVP8 || codecType == kVideoCodecVP9) { |
| return true; |
| } |
| return false; |
| } |
| |
| int CalculateMaxPadBitrateBps(std::vector<VideoStream> streams, |
| int min_transmit_bitrate_bps, |
| bool pad_to_min_bitrate) { |
| int pad_up_to_bitrate_bps = 0; |
| // Calculate max padding bitrate for a multi layer codec. |
| if (streams.size() > 1) { |
| // Pad to min bitrate of the highest layer. |
| pad_up_to_bitrate_bps = streams[streams.size() - 1].min_bitrate_bps; |
| // Add target_bitrate_bps of the lower layers. |
| for (size_t i = 0; i < streams.size() - 1; ++i) |
| pad_up_to_bitrate_bps += streams[i].target_bitrate_bps; |
| } else if (pad_to_min_bitrate) { |
| pad_up_to_bitrate_bps = streams[0].min_bitrate_bps; |
| } |
| |
| pad_up_to_bitrate_bps = |
| std::max(pad_up_to_bitrate_bps, min_transmit_bitrate_bps); |
| |
| return pad_up_to_bitrate_bps; |
| } |
| |
| uint32_t CalculateOverheadRateBps(int packets_per_second, |
| size_t overhead_bytes_per_packet, |
| uint32_t max_overhead_bps) { |
| uint32_t overhead_bps = |
| static_cast<uint32_t>(8 * overhead_bytes_per_packet * packets_per_second); |
| return std::min(overhead_bps, max_overhead_bps); |
| } |
| |
| int CalculatePacketRate(uint32_t bitrate_bps, size_t packet_size_bytes) { |
| size_t packet_size_bits = 8 * packet_size_bytes; |
| // Ceil for int value of bitrate_bps / packet_size_bits. |
| return static_cast<int>((bitrate_bps + packet_size_bits - 1) / |
| packet_size_bits); |
| } |
| |
| } // namespace |
| |
| namespace internal { |
| |
| // VideoSendStreamImpl implements internal::VideoSendStream. |
| // It is created and destroyed on |worker_queue|. The intent is to decrease the |
| // need for locking and to ensure methods are called in sequence. |
| // Public methods except |DeliverRtcp| must be called on |worker_queue|. |
| // DeliverRtcp is called on the libjingle worker thread or a network thread. |
| // An encoder may deliver frames through the EncodedImageCallback on an |
| // arbitrary thread. |
| class VideoSendStreamImpl : public webrtc::BitrateAllocatorObserver, |
| public webrtc::OverheadObserver, |
| public webrtc::VCMProtectionCallback, |
| public VideoStreamEncoder::EncoderSink, |
| public VideoBitrateAllocationObserver { |
| public: |
| VideoSendStreamImpl( |
| SendStatisticsProxy* stats_proxy, |
| rtc::TaskQueue* worker_queue, |
| CallStats* call_stats, |
| RtpTransportControllerSendInterface* transport, |
| BitrateAllocator* bitrate_allocator, |
| SendDelayStats* send_delay_stats, |
| VideoStreamEncoder* video_stream_encoder, |
| RtcEventLog* event_log, |
| const VideoSendStream::Config* config, |
| int initial_encoder_max_bitrate, |
| double initial_encoder_bitrate_priority, |
| std::map<uint32_t, RtpState> suspended_ssrcs, |
| std::map<uint32_t, RtpPayloadState> suspended_payload_states, |
| VideoEncoderConfig::ContentType content_type); |
| ~VideoSendStreamImpl() override; |
| |
| // RegisterProcessThread register |module_process_thread| with those objects |
| // that use it. Registration has to happen on the thread were |
| // |module_process_thread| was created (libjingle's worker thread). |
| // TODO(perkj): Replace the use of |module_process_thread| with a TaskQueue, |
| // maybe |worker_queue|. |
| void RegisterProcessThread(ProcessThread* module_process_thread); |
| void DeRegisterProcessThread(); |
| |
| void SignalNetworkState(NetworkState state); |
| bool DeliverRtcp(const uint8_t* packet, size_t length); |
| void Start(); |
| void Stop(); |
| |
| VideoSendStream::RtpStateMap GetRtpStates() const; |
| VideoSendStream::RtpPayloadStateMap GetRtpPayloadStates() const; |
| |
| void EnableEncodedFrameRecording(const std::vector<rtc::PlatformFile>& files, |
| size_t byte_limit); |
| |
| void SetTransportOverhead(size_t transport_overhead_per_packet); |
| |
| private: |
| class CheckEncoderActivityTask; |
| class EncoderReconfiguredTask; |
| |
| // Implements BitrateAllocatorObserver. |
| uint32_t OnBitrateUpdated(uint32_t bitrate_bps, |
| uint8_t fraction_loss, |
| int64_t rtt, |
| int64_t probing_interval_ms) override; |
| |
| // Implements webrtc::VCMProtectionCallback. |
| int ProtectionRequest(const FecProtectionParams* delta_params, |
| const FecProtectionParams* key_params, |
| uint32_t* sent_video_rate_bps, |
| uint32_t* sent_nack_rate_bps, |
| uint32_t* sent_fec_rate_bps) override; |
| |
| // Implements OverheadObserver. |
| void OnOverheadChanged(size_t overhead_bytes_per_packet) override; |
| |
| void OnEncoderConfigurationChanged(std::vector<VideoStream> streams, |
| int min_transmit_bitrate_bps) override; |
| |
| // Implements EncodedImageCallback. The implementation routes encoded frames |
| // to the |payload_router_| and |config.pre_encode_callback| if set. |
| // Called on an arbitrary encoder callback thread. |
| EncodedImageCallback::Result OnEncodedImage( |
| const EncodedImage& encoded_image, |
| const CodecSpecificInfo* codec_specific_info, |
| const RTPFragmentationHeader* fragmentation) override; |
| |
| // Implements VideoBitrateAllocationObserver. |
| void OnBitrateAllocationUpdated(const BitrateAllocation& allocation) override; |
| |
| void ConfigureProtection(); |
| void ConfigureSsrcs(); |
| void SignalEncoderTimedOut(); |
| void SignalEncoderActive(); |
| |
| const bool send_side_bwe_with_overhead_; |
| |
| SendStatisticsProxy* const stats_proxy_; |
| const VideoSendStream::Config* const config_; |
| std::map<uint32_t, RtpState> suspended_ssrcs_; |
| |
| ProcessThread* module_process_thread_; |
| rtc::ThreadChecker module_process_thread_checker_; |
| rtc::TaskQueue* const worker_queue_; |
| |
| rtc::CriticalSection encoder_activity_crit_sect_; |
| CheckEncoderActivityTask* check_encoder_activity_task_ |
| RTC_GUARDED_BY(encoder_activity_crit_sect_); |
| |
| CallStats* const call_stats_; |
| RtpTransportControllerSendInterface* const transport_; |
| BitrateAllocator* const bitrate_allocator_; |
| |
| // TODO(brandtr): Move ownership to PayloadRouter. |
| std::unique_ptr<FlexfecSender> flexfec_sender_; |
| |
| rtc::CriticalSection ivf_writers_crit_; |
| std::unique_ptr<IvfFileWriter> |
| file_writers_[kMaxSimulcastStreams] RTC_GUARDED_BY(ivf_writers_crit_); |
| |
| int max_padding_bitrate_; |
| int encoder_min_bitrate_bps_; |
| uint32_t encoder_max_bitrate_bps_; |
| uint32_t encoder_target_rate_bps_; |
| double encoder_bitrate_priority_; |
| |
| VideoStreamEncoder* const video_stream_encoder_; |
| EncoderRtcpFeedback encoder_feedback_; |
| ProtectionBitrateCalculator protection_bitrate_calculator_; |
| |
| RtcpBandwidthObserver* const bandwidth_observer_; |
| // RtpRtcp modules, declared here as they use other members on construction. |
| const std::vector<RtpRtcp*> rtp_rtcp_modules_; |
| PayloadRouter payload_router_; |
| |
| // |weak_ptr_| to our self. This is used since we can not call |
| // |weak_ptr_factory_.GetWeakPtr| from multiple sequences but it is ok to copy |
| // an existing WeakPtr. |
| rtc::WeakPtr<VideoSendStreamImpl> weak_ptr_; |
| // |weak_ptr_factory_| must be declared last to make sure all WeakPtr's are |
| // invalidated before any other members are destroyed. |
| rtc::WeakPtrFactory<VideoSendStreamImpl> weak_ptr_factory_; |
| |
| rtc::CriticalSection overhead_bytes_per_packet_crit_; |
| size_t overhead_bytes_per_packet_ |
| RTC_GUARDED_BY(overhead_bytes_per_packet_crit_); |
| size_t transport_overhead_bytes_per_packet_; |
| }; |
| |
| // TODO(tommi): See if there's a more elegant way to create a task that creates |
| // an object on the correct task queue. |
| class VideoSendStream::ConstructionTask : public rtc::QueuedTask { |
| public: |
| ConstructionTask( |
| std::unique_ptr<VideoSendStreamImpl>* send_stream, |
| rtc::Event* done_event, |
| SendStatisticsProxy* stats_proxy, |
| VideoStreamEncoder* video_stream_encoder, |
| ProcessThread* module_process_thread, |
| CallStats* call_stats, |
| RtpTransportControllerSendInterface* transport, |
| BitrateAllocator* bitrate_allocator, |
| SendDelayStats* send_delay_stats, |
| RtcEventLog* event_log, |
| const VideoSendStream::Config* config, |
| int initial_encoder_max_bitrate, |
| double initial_encoder_bitrate_priority, |
| const std::map<uint32_t, RtpState>& suspended_ssrcs, |
| const std::map<uint32_t, RtpPayloadState>& suspended_payload_states, |
| VideoEncoderConfig::ContentType content_type) |
| : send_stream_(send_stream), |
| done_event_(done_event), |
| stats_proxy_(stats_proxy), |
| video_stream_encoder_(video_stream_encoder), |
| call_stats_(call_stats), |
| transport_(transport), |
| bitrate_allocator_(bitrate_allocator), |
| send_delay_stats_(send_delay_stats), |
| event_log_(event_log), |
| config_(config), |
| initial_encoder_max_bitrate_(initial_encoder_max_bitrate), |
| initial_encoder_bitrate_priority_(initial_encoder_bitrate_priority), |
| suspended_ssrcs_(suspended_ssrcs), |
| suspended_payload_states_(suspended_payload_states), |
| content_type_(content_type) {} |
| |
| ~ConstructionTask() override { done_event_->Set(); } |
| |
| private: |
| bool Run() override { |
| send_stream_->reset(new VideoSendStreamImpl( |
| stats_proxy_, rtc::TaskQueue::Current(), call_stats_, transport_, |
| bitrate_allocator_, send_delay_stats_, video_stream_encoder_, |
| event_log_, config_, initial_encoder_max_bitrate_, |
| initial_encoder_bitrate_priority_, std::move(suspended_ssrcs_), |
| std::move(suspended_payload_states_), content_type_)); |
| return true; |
| } |
| |
| std::unique_ptr<VideoSendStreamImpl>* const send_stream_; |
| rtc::Event* const done_event_; |
| SendStatisticsProxy* const stats_proxy_; |
| VideoStreamEncoder* const video_stream_encoder_; |
| CallStats* const call_stats_; |
| RtpTransportControllerSendInterface* const transport_; |
| BitrateAllocator* const bitrate_allocator_; |
| SendDelayStats* const send_delay_stats_; |
| RtcEventLog* const event_log_; |
| const VideoSendStream::Config* config_; |
| int initial_encoder_max_bitrate_; |
| double initial_encoder_bitrate_priority_; |
| std::map<uint32_t, RtpState> suspended_ssrcs_; |
| std::map<uint32_t, RtpPayloadState> suspended_payload_states_; |
| const VideoEncoderConfig::ContentType content_type_; |
| }; |
| |
| class VideoSendStream::DestructAndGetRtpStateTask : public rtc::QueuedTask { |
| public: |
| DestructAndGetRtpStateTask( |
| VideoSendStream::RtpStateMap* state_map, |
| VideoSendStream::RtpPayloadStateMap* payload_state_map, |
| std::unique_ptr<VideoSendStreamImpl> send_stream, |
| rtc::Event* done_event) |
| : state_map_(state_map), |
| payload_state_map_(payload_state_map), |
| send_stream_(std::move(send_stream)), |
| done_event_(done_event) {} |
| |
| ~DestructAndGetRtpStateTask() override { RTC_CHECK(!send_stream_); } |
| |
| private: |
| bool Run() override { |
| send_stream_->Stop(); |
| *state_map_ = send_stream_->GetRtpStates(); |
| *payload_state_map_ = send_stream_->GetRtpPayloadStates(); |
| send_stream_.reset(); |
| done_event_->Set(); |
| return true; |
| } |
| |
| VideoSendStream::RtpStateMap* state_map_; |
| VideoSendStream::RtpPayloadStateMap* payload_state_map_; |
| std::unique_ptr<VideoSendStreamImpl> send_stream_; |
| rtc::Event* done_event_; |
| }; |
| |
| // CheckEncoderActivityTask is used for tracking when the encoder last produced |
| // and encoded video frame. If the encoder has not produced anything the last |
| // kEncoderTimeOutMs we also want to stop sending padding. |
| class VideoSendStreamImpl::CheckEncoderActivityTask : public rtc::QueuedTask { |
| public: |
| static const int kEncoderTimeOutMs = 2000; |
| explicit CheckEncoderActivityTask( |
| const rtc::WeakPtr<VideoSendStreamImpl>& send_stream) |
| : activity_(0), send_stream_(std::move(send_stream)), timed_out_(false) {} |
| |
| void Stop() { |
| RTC_CHECK(task_checker_.CalledSequentially()); |
| send_stream_.reset(); |
| } |
| |
| void UpdateEncoderActivity() { |
| // UpdateEncoderActivity is called from VideoSendStreamImpl::Encoded on |
| // whatever thread the real encoder implementation run on. In the case of |
| // hardware encoders, there might be several encoders |
| // running in parallel on different threads. |
| rtc::AtomicOps::ReleaseStore(&activity_, 1); |
| } |
| |
| private: |
| bool Run() override { |
| RTC_CHECK(task_checker_.CalledSequentially()); |
| if (!send_stream_) |
| return true; |
| if (!rtc::AtomicOps::AcquireLoad(&activity_)) { |
| if (!timed_out_) { |
| send_stream_->SignalEncoderTimedOut(); |
| } |
| timed_out_ = true; |
| } else if (timed_out_) { |
| send_stream_->SignalEncoderActive(); |
| timed_out_ = false; |
| } |
| rtc::AtomicOps::ReleaseStore(&activity_, 0); |
| |
| rtc::TaskQueue::Current()->PostDelayedTask( |
| std::unique_ptr<rtc::QueuedTask>(this), kEncoderTimeOutMs); |
| // Return false to prevent this task from being deleted. Ownership has been |
| // transferred to the task queue when PostDelayedTask was called. |
| return false; |
| } |
| volatile int activity_; |
| |
| rtc::SequencedTaskChecker task_checker_; |
| rtc::WeakPtr<VideoSendStreamImpl> send_stream_; |
| bool timed_out_; |
| }; |
| |
| class VideoSendStreamImpl::EncoderReconfiguredTask : public rtc::QueuedTask { |
| public: |
| EncoderReconfiguredTask(const rtc::WeakPtr<VideoSendStreamImpl>& send_stream, |
| std::vector<VideoStream> streams, |
| int min_transmit_bitrate_bps) |
| : send_stream_(std::move(send_stream)), |
| streams_(std::move(streams)), |
| min_transmit_bitrate_bps_(min_transmit_bitrate_bps) {} |
| |
| private: |
| bool Run() override { |
| if (send_stream_) |
| send_stream_->OnEncoderConfigurationChanged(std::move(streams_), |
| min_transmit_bitrate_bps_); |
| return true; |
| } |
| |
| rtc::WeakPtr<VideoSendStreamImpl> send_stream_; |
| std::vector<VideoStream> streams_; |
| int min_transmit_bitrate_bps_; |
| }; |
| |
| VideoSendStream::VideoSendStream( |
| int num_cpu_cores, |
| ProcessThread* module_process_thread, |
| rtc::TaskQueue* worker_queue, |
| CallStats* call_stats, |
| RtpTransportControllerSendInterface* transport, |
| BitrateAllocator* bitrate_allocator, |
| SendDelayStats* send_delay_stats, |
| RtcEventLog* event_log, |
| VideoSendStream::Config config, |
| VideoEncoderConfig encoder_config, |
| const std::map<uint32_t, RtpState>& suspended_ssrcs, |
| const std::map<uint32_t, RtpPayloadState>& suspended_payload_states) |
| : worker_queue_(worker_queue), |
| thread_sync_event_(false /* manual_reset */, false), |
| stats_proxy_(Clock::GetRealTimeClock(), |
| config, |
| encoder_config.content_type), |
| config_(std::move(config)), |
| content_type_(encoder_config.content_type) { |
| video_stream_encoder_.reset( |
| new VideoStreamEncoder(num_cpu_cores, &stats_proxy_, |
| config_.encoder_settings, |
| config_.pre_encode_callback, |
| std::unique_ptr<OveruseFrameDetector>())); |
| worker_queue_->PostTask(std::unique_ptr<rtc::QueuedTask>(new ConstructionTask( |
| &send_stream_, &thread_sync_event_, &stats_proxy_, |
| video_stream_encoder_.get(), module_process_thread, call_stats, transport, |
| bitrate_allocator, send_delay_stats, event_log, &config_, |
| encoder_config.max_bitrate_bps, encoder_config.bitrate_priority, |
| suspended_ssrcs, suspended_payload_states, encoder_config.content_type))); |
| |
| // Wait for ConstructionTask to complete so that |send_stream_| can be used. |
| // |module_process_thread| must be registered and deregistered on the thread |
| // it was created on. |
| thread_sync_event_.Wait(rtc::Event::kForever); |
| send_stream_->RegisterProcessThread(module_process_thread); |
| // TODO(sprang): Enable this also for regular video calls if it works well. |
| if (encoder_config.content_type == VideoEncoderConfig::ContentType::kScreen) { |
| // Only signal target bitrate for screenshare streams, for now. |
| video_stream_encoder_->SetBitrateObserver(send_stream_.get()); |
| } |
| |
| ReconfigureVideoEncoder(std::move(encoder_config)); |
| } |
| |
| VideoSendStream::~VideoSendStream() { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| RTC_DCHECK(!send_stream_); |
| } |
| |
| void VideoSendStream::Start() { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| RTC_LOG(LS_INFO) << "VideoSendStream::Start"; |
| VideoSendStreamImpl* send_stream = send_stream_.get(); |
| worker_queue_->PostTask([this, send_stream] { |
| send_stream->Start(); |
| thread_sync_event_.Set(); |
| }); |
| |
| // It is expected that after VideoSendStream::Start has been called, incoming |
| // frames are not dropped in VideoStreamEncoder. To ensure this, Start has to |
| // be synchronized. |
| thread_sync_event_.Wait(rtc::Event::kForever); |
| } |
| |
| void VideoSendStream::Stop() { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| RTC_LOG(LS_INFO) << "VideoSendStream::Stop"; |
| VideoSendStreamImpl* send_stream = send_stream_.get(); |
| worker_queue_->PostTask([send_stream] { send_stream->Stop(); }); |
| } |
| |
| void VideoSendStream::SetSource( |
| rtc::VideoSourceInterface<webrtc::VideoFrame>* source, |
| const DegradationPreference& degradation_preference) { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| video_stream_encoder_->SetSource(source, degradation_preference); |
| } |
| |
| void VideoSendStream::ReconfigureVideoEncoder(VideoEncoderConfig config) { |
| // TODO(perkj): Some test cases in VideoSendStreamTest call |
| // ReconfigureVideoEncoder from the network thread. |
| // RTC_DCHECK_RUN_ON(&thread_checker_); |
| RTC_DCHECK(content_type_ == config.content_type); |
| video_stream_encoder_->ConfigureEncoder(std::move(config), |
| config_.rtp.max_packet_size, |
| config_.rtp.nack.rtp_history_ms > 0); |
| } |
| |
| VideoSendStream::Stats VideoSendStream::GetStats() { |
| // TODO(perkj, solenberg): Some test cases in EndToEndTest call GetStats from |
| // a network thread. See comment in Call::GetStats(). |
| // RTC_DCHECK_RUN_ON(&thread_checker_); |
| return stats_proxy_.GetStats(); |
| } |
| |
| void VideoSendStream::SignalNetworkState(NetworkState state) { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| VideoSendStreamImpl* send_stream = send_stream_.get(); |
| worker_queue_->PostTask( |
| [send_stream, state] { send_stream->SignalNetworkState(state); }); |
| } |
| |
| void VideoSendStream::StopPermanentlyAndGetRtpStates( |
| VideoSendStream::RtpStateMap* rtp_state_map, |
| VideoSendStream::RtpPayloadStateMap* payload_state_map) { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| video_stream_encoder_->Stop(); |
| send_stream_->DeRegisterProcessThread(); |
| worker_queue_->PostTask( |
| std::unique_ptr<rtc::QueuedTask>(new DestructAndGetRtpStateTask( |
| rtp_state_map, payload_state_map, std::move(send_stream_), |
| &thread_sync_event_))); |
| thread_sync_event_.Wait(rtc::Event::kForever); |
| } |
| |
| void VideoSendStream::SetTransportOverhead( |
| size_t transport_overhead_per_packet) { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| VideoSendStreamImpl* send_stream = send_stream_.get(); |
| worker_queue_->PostTask([send_stream, transport_overhead_per_packet] { |
| send_stream->SetTransportOverhead(transport_overhead_per_packet); |
| }); |
| } |
| |
| bool VideoSendStream::DeliverRtcp(const uint8_t* packet, size_t length) { |
| // Called on a network thread. |
| return send_stream_->DeliverRtcp(packet, length); |
| } |
| |
| void VideoSendStream::EnableEncodedFrameRecording( |
| const std::vector<rtc::PlatformFile>& files, |
| size_t byte_limit) { |
| send_stream_->EnableEncodedFrameRecording(files, byte_limit); |
| } |
| |
| VideoSendStreamImpl::VideoSendStreamImpl( |
| SendStatisticsProxy* stats_proxy, |
| rtc::TaskQueue* worker_queue, |
| CallStats* call_stats, |
| RtpTransportControllerSendInterface* transport, |
| BitrateAllocator* bitrate_allocator, |
| SendDelayStats* send_delay_stats, |
| VideoStreamEncoder* video_stream_encoder, |
| RtcEventLog* event_log, |
| const VideoSendStream::Config* config, |
| int initial_encoder_max_bitrate, |
| double initial_encoder_bitrate_priority, |
| std::map<uint32_t, RtpState> suspended_ssrcs, |
| std::map<uint32_t, RtpPayloadState> suspended_payload_states, |
| VideoEncoderConfig::ContentType content_type) |
| : send_side_bwe_with_overhead_( |
| webrtc::field_trial::IsEnabled("WebRTC-SendSideBwe-WithOverhead")), |
| stats_proxy_(stats_proxy), |
| config_(config), |
| suspended_ssrcs_(std::move(suspended_ssrcs)), |
| module_process_thread_(nullptr), |
| worker_queue_(worker_queue), |
| check_encoder_activity_task_(nullptr), |
| call_stats_(call_stats), |
| transport_(transport), |
| bitrate_allocator_(bitrate_allocator), |
| flexfec_sender_(MaybeCreateFlexfecSender(*config_, suspended_ssrcs_)), |
| max_padding_bitrate_(0), |
| encoder_min_bitrate_bps_(0), |
| encoder_max_bitrate_bps_(initial_encoder_max_bitrate), |
| encoder_target_rate_bps_(0), |
| encoder_bitrate_priority_(initial_encoder_bitrate_priority), |
| video_stream_encoder_(video_stream_encoder), |
| encoder_feedback_(Clock::GetRealTimeClock(), |
| config_->rtp.ssrcs, |
| video_stream_encoder), |
| protection_bitrate_calculator_(Clock::GetRealTimeClock(), this), |
| bandwidth_observer_(transport->send_side_cc()->GetBandwidthObserver()), |
| rtp_rtcp_modules_(CreateRtpRtcpModules( |
| config_->send_transport, |
| &encoder_feedback_, |
| bandwidth_observer_, |
| transport, |
| call_stats_->rtcp_rtt_stats(), |
| flexfec_sender_.get(), |
| stats_proxy_, |
| send_delay_stats, |
| event_log, |
| transport->send_side_cc()->GetRetransmissionRateLimiter(), |
| this, |
| config_->rtp.ssrcs.size(), |
| transport->keepalive_config())), |
| payload_router_(rtp_rtcp_modules_, |
| config_->rtp.ssrcs, |
| config_->encoder_settings.payload_type, |
| suspended_payload_states), |
| weak_ptr_factory_(this), |
| overhead_bytes_per_packet_(0), |
| transport_overhead_bytes_per_packet_(0) { |
| RTC_DCHECK_RUN_ON(worker_queue_); |
| RTC_LOG(LS_INFO) << "VideoSendStreamInternal: " << config_->ToString(); |
| weak_ptr_ = weak_ptr_factory_.GetWeakPtr(); |
| module_process_thread_checker_.DetachFromThread(); |
| |
| RTC_DCHECK(!config_->rtp.ssrcs.empty()); |
| RTC_DCHECK(call_stats_); |
| RTC_DCHECK(transport_); |
| RTC_DCHECK(transport_->send_side_cc()); |
| RTC_DCHECK_GT(encoder_max_bitrate_bps_, 0); |
| RTC_CHECK(field_trial::FindFullName( |
| AlrDetector::kStrictPacingAndProbingExperimentName) |
| .empty() || |
| field_trial::FindFullName( |
| AlrDetector::kScreenshareProbingBweExperimentName) |
| .empty()); |
| // If send-side BWE is enabled, check if we should apply updated probing and |
| // pacing settings. |
| if (TransportSeqNumExtensionConfigured(*config_)) { |
| rtc::Optional<AlrDetector::AlrExperimentSettings> alr_settings; |
| if (content_type == VideoEncoderConfig::ContentType::kScreen) { |
| alr_settings = AlrDetector::ParseAlrSettingsFromFieldTrial( |
| AlrDetector::kScreenshareProbingBweExperimentName); |
| } else { |
| alr_settings = AlrDetector::ParseAlrSettingsFromFieldTrial( |
| AlrDetector::kStrictPacingAndProbingExperimentName); |
| } |
| if (alr_settings) { |
| transport->send_side_cc()->EnablePeriodicAlrProbing(true); |
| transport->pacer()->SetPacingFactor(alr_settings->pacing_factor); |
| transport->pacer()->SetQueueTimeLimit(alr_settings->max_paced_queue_time); |
| } |
| } |
| |
| if (config_->periodic_alr_bandwidth_probing) { |
| transport->send_side_cc()->EnablePeriodicAlrProbing(true); |
| } |
| |
| // RTP/RTCP initialization. |
| |
| // We add the highest spatial layer first to ensure it'll be prioritized |
| // when sending padding, with the hope that the packet rate will be smaller, |
| // and that it's more important to protect than the lower layers. |
| for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { |
| constexpr bool remb_candidate = true; |
| transport->packet_router()->AddSendRtpModule(rtp_rtcp, remb_candidate); |
| } |
| |
| for (size_t i = 0; i < config_->rtp.extensions.size(); ++i) { |
| const std::string& extension = config_->rtp.extensions[i].uri; |
| int id = config_->rtp.extensions[i].id; |
| // One-byte-extension local identifiers are in the range 1-14 inclusive. |
| RTC_DCHECK_GE(id, 1); |
| RTC_DCHECK_LE(id, 14); |
| RTC_DCHECK(RtpExtension::IsSupportedForVideo(extension)); |
| for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { |
| RTC_CHECK_EQ(0, rtp_rtcp->RegisterSendRtpHeaderExtension( |
| StringToRtpExtensionType(extension), id)); |
| } |
| } |
| |
| ConfigureProtection(); |
| ConfigureSsrcs(); |
| |
| // TODO(pbos): Should we set CNAME on all RTP modules? |
| rtp_rtcp_modules_.front()->SetCNAME(config_->rtp.c_name.c_str()); |
| |
| for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { |
| rtp_rtcp->RegisterRtcpStatisticsCallback(stats_proxy_); |
| rtp_rtcp->RegisterSendChannelRtpStatisticsCallback(stats_proxy_); |
| rtp_rtcp->SetMaxRtpPacketSize(config_->rtp.max_packet_size); |
| rtp_rtcp->RegisterVideoSendPayload( |
| config_->encoder_settings.payload_type, |
| config_->encoder_settings.payload_name.c_str()); |
| } |
| |
| RTC_DCHECK(config_->encoder_settings.encoder); |
| RTC_DCHECK_GE(config_->encoder_settings.payload_type, 0); |
| RTC_DCHECK_LE(config_->encoder_settings.payload_type, 127); |
| |
| video_stream_encoder_->SetStartBitrate( |
| bitrate_allocator_->GetStartBitrate(this)); |
| |
| // Only request rotation at the source when we positively know that the remote |
| // side doesn't support the rotation extension. This allows us to prepare the |
| // encoder in the expectation that rotation is supported - which is the common |
| // case. |
| bool rotation_applied = |
| std::find_if(config_->rtp.extensions.begin(), |
| config_->rtp.extensions.end(), |
| [](const RtpExtension& extension) { |
| return extension.uri == RtpExtension::kVideoRotationUri; |
| }) == config_->rtp.extensions.end(); |
| |
| video_stream_encoder_->SetSink(this, rotation_applied); |
| } |
| |
| void VideoSendStreamImpl::RegisterProcessThread( |
| ProcessThread* module_process_thread) { |
| RTC_DCHECK_RUN_ON(&module_process_thread_checker_); |
| RTC_DCHECK(!module_process_thread_); |
| module_process_thread_ = module_process_thread; |
| |
| for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) |
| module_process_thread_->RegisterModule(rtp_rtcp, RTC_FROM_HERE); |
| } |
| |
| void VideoSendStreamImpl::DeRegisterProcessThread() { |
| RTC_DCHECK_RUN_ON(&module_process_thread_checker_); |
| for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) |
| module_process_thread_->DeRegisterModule(rtp_rtcp); |
| } |
| |
| VideoSendStreamImpl::~VideoSendStreamImpl() { |
| RTC_DCHECK_RUN_ON(worker_queue_); |
| RTC_DCHECK(!payload_router_.IsActive()) |
| << "VideoSendStreamImpl::Stop not called"; |
| RTC_LOG(LS_INFO) << "~VideoSendStreamInternal: " << config_->ToString(); |
| |
| for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { |
| transport_->packet_router()->RemoveSendRtpModule(rtp_rtcp); |
| delete rtp_rtcp; |
| } |
| } |
| |
| bool VideoSendStreamImpl::DeliverRtcp(const uint8_t* packet, size_t length) { |
| // Runs on a network thread. |
| RTC_DCHECK(!worker_queue_->IsCurrent()); |
| for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) |
| rtp_rtcp->IncomingRtcpPacket(packet, length); |
| return true; |
| } |
| |
| void VideoSendStreamImpl::Start() { |
| RTC_DCHECK_RUN_ON(worker_queue_); |
| RTC_LOG(LS_INFO) << "VideoSendStream::Start"; |
| if (payload_router_.IsActive()) |
| return; |
| TRACE_EVENT_INSTANT0("webrtc", "VideoSendStream::Start"); |
| payload_router_.SetActive(true); |
| |
| bitrate_allocator_->AddObserver( |
| this, encoder_min_bitrate_bps_, encoder_max_bitrate_bps_, |
| max_padding_bitrate_, !config_->suspend_below_min_bitrate, |
| config_->track_id, encoder_bitrate_priority_); |
| |
| // Start monitoring encoder activity. |
| { |
| rtc::CritScope lock(&encoder_activity_crit_sect_); |
| RTC_DCHECK(!check_encoder_activity_task_); |
| check_encoder_activity_task_ = new CheckEncoderActivityTask(weak_ptr_); |
| worker_queue_->PostDelayedTask( |
| std::unique_ptr<rtc::QueuedTask>(check_encoder_activity_task_), |
| CheckEncoderActivityTask::kEncoderTimeOutMs); |
| } |
| |
| video_stream_encoder_->SendKeyFrame(); |
| } |
| |
| void VideoSendStreamImpl::Stop() { |
| RTC_DCHECK_RUN_ON(worker_queue_); |
| RTC_LOG(LS_INFO) << "VideoSendStream::Stop"; |
| if (!payload_router_.IsActive()) |
| return; |
| TRACE_EVENT_INSTANT0("webrtc", "VideoSendStream::Stop"); |
| payload_router_.SetActive(false); |
| bitrate_allocator_->RemoveObserver(this); |
| { |
| rtc::CritScope lock(&encoder_activity_crit_sect_); |
| check_encoder_activity_task_->Stop(); |
| check_encoder_activity_task_ = nullptr; |
| } |
| video_stream_encoder_->OnBitrateUpdated(0, 0, 0); |
| stats_proxy_->OnSetEncoderTargetRate(0); |
| } |
| |
| void VideoSendStreamImpl::SignalEncoderTimedOut() { |
| RTC_DCHECK_RUN_ON(worker_queue_); |
| // If the encoder has not produced anything the last kEncoderTimeOutMs and it |
| // is supposed to, deregister as BitrateAllocatorObserver. This can happen |
| // if a camera stops producing frames. |
| if (encoder_target_rate_bps_ > 0) { |
| RTC_LOG(LS_INFO) << "SignalEncoderTimedOut, Encoder timed out."; |
| bitrate_allocator_->RemoveObserver(this); |
| } |
| } |
| |
| void VideoSendStreamImpl::OnBitrateAllocationUpdated( |
| const BitrateAllocation& allocation) { |
| payload_router_.OnBitrateAllocationUpdated(allocation); |
| } |
| |
| void VideoSendStreamImpl::SignalEncoderActive() { |
| RTC_DCHECK_RUN_ON(worker_queue_); |
| RTC_LOG(LS_INFO) << "SignalEncoderActive, Encoder is active."; |
| bitrate_allocator_->AddObserver( |
| this, encoder_min_bitrate_bps_, encoder_max_bitrate_bps_, |
| max_padding_bitrate_, !config_->suspend_below_min_bitrate, |
| config_->track_id, encoder_bitrate_priority_); |
| } |
| |
| void VideoSendStreamImpl::OnEncoderConfigurationChanged( |
| std::vector<VideoStream> streams, |
| int min_transmit_bitrate_bps) { |
| if (!worker_queue_->IsCurrent()) { |
| worker_queue_->PostTask( |
| std::unique_ptr<rtc::QueuedTask>(new EncoderReconfiguredTask( |
| weak_ptr_, std::move(streams), min_transmit_bitrate_bps))); |
| return; |
| } |
| RTC_DCHECK_GE(config_->rtp.ssrcs.size(), streams.size()); |
| TRACE_EVENT0("webrtc", "VideoSendStream::OnEncoderConfigurationChanged"); |
| RTC_DCHECK_GE(config_->rtp.ssrcs.size(), streams.size()); |
| RTC_DCHECK_RUN_ON(worker_queue_); |
| |
| encoder_min_bitrate_bps_ = |
| std::max(streams[0].min_bitrate_bps, GetEncoderMinBitrateBps()); |
| encoder_max_bitrate_bps_ = 0; |
| double stream_bitrate_priority_sum = 0; |
| for (const auto& stream : streams) { |
| encoder_max_bitrate_bps_ += stream.max_bitrate_bps; |
| if (stream.bitrate_priority) { |
| RTC_DCHECK_GT(*stream.bitrate_priority, 0); |
| stream_bitrate_priority_sum += *stream.bitrate_priority; |
| } |
| } |
| RTC_DCHECK_GT(stream_bitrate_priority_sum, 0); |
| encoder_bitrate_priority_ = stream_bitrate_priority_sum; |
| max_padding_bitrate_ = CalculateMaxPadBitrateBps( |
| streams, min_transmit_bitrate_bps, config_->suspend_below_min_bitrate); |
| |
| // Clear stats for disabled layers. |
| for (size_t i = streams.size(); i < config_->rtp.ssrcs.size(); ++i) { |
| stats_proxy_->OnInactiveSsrc(config_->rtp.ssrcs[i]); |
| } |
| |
| size_t number_of_temporal_layers = |
| streams.back().temporal_layer_thresholds_bps.size() + 1; |
| protection_bitrate_calculator_.SetEncodingData( |
| streams[0].width, streams[0].height, number_of_temporal_layers, |
| config_->rtp.max_packet_size); |
| |
| if (payload_router_.IsActive()) { |
| // The send stream is started already. Update the allocator with new bitrate |
| // limits. |
| bitrate_allocator_->AddObserver( |
| this, encoder_min_bitrate_bps_, encoder_max_bitrate_bps_, |
| max_padding_bitrate_, !config_->suspend_below_min_bitrate, |
| config_->track_id, encoder_bitrate_priority_); |
| } |
| } |
| |
| EncodedImageCallback::Result VideoSendStreamImpl::OnEncodedImage( |
| const EncodedImage& encoded_image, |
| const CodecSpecificInfo* codec_specific_info, |
| const RTPFragmentationHeader* fragmentation) { |
| // Encoded is called on whatever thread the real encoder implementation run |
| // on. In the case of hardware encoders, there might be several encoders |
| // running in parallel on different threads. |
| size_t simulcast_idx = 0; |
| if (codec_specific_info->codecType == kVideoCodecVP8) { |
| simulcast_idx = codec_specific_info->codecSpecific.VP8.simulcastIdx; |
| } |
| if (config_->post_encode_callback) { |
| config_->post_encode_callback->EncodedFrameCallback(EncodedFrame( |
| encoded_image._buffer, encoded_image._length, encoded_image._frameType, |
| simulcast_idx, encoded_image._timeStamp)); |
| } |
| { |
| rtc::CritScope lock(&encoder_activity_crit_sect_); |
| if (check_encoder_activity_task_) |
| check_encoder_activity_task_->UpdateEncoderActivity(); |
| } |
| |
| protection_bitrate_calculator_.UpdateWithEncodedData(encoded_image); |
| EncodedImageCallback::Result result = payload_router_.OnEncodedImage( |
| encoded_image, codec_specific_info, fragmentation); |
| |
| RTC_DCHECK(codec_specific_info); |
| |
| int layer = codec_specific_info->codecType == kVideoCodecVP8 |
| ? codec_specific_info->codecSpecific.VP8.simulcastIdx |
| : 0; |
| { |
| rtc::CritScope lock(&ivf_writers_crit_); |
| if (file_writers_[layer].get()) { |
| bool ok = file_writers_[layer]->WriteFrame( |
| encoded_image, codec_specific_info->codecType); |
| RTC_DCHECK(ok); |
| } |
| } |
| |
| return result; |
| } |
| |
| void VideoSendStreamImpl::ConfigureProtection() { |
| RTC_DCHECK_RUN_ON(worker_queue_); |
| |
| // Consistency of FlexFEC parameters is checked in MaybeCreateFlexfecSender. |
| const bool flexfec_enabled = (flexfec_sender_ != nullptr); |
| |
| // Consistency of NACK and RED+ULPFEC parameters is checked in this function. |
| const bool nack_enabled = config_->rtp.nack.rtp_history_ms > 0; |
| int red_payload_type = config_->rtp.ulpfec.red_payload_type; |
| int ulpfec_payload_type = config_->rtp.ulpfec.ulpfec_payload_type; |
| |
| // Shorthands. |
| auto IsRedEnabled = [&]() { return red_payload_type >= 0; }; |
| auto DisableRed = [&]() { red_payload_type = -1; }; |
| auto IsUlpfecEnabled = [&]() { return ulpfec_payload_type >= 0; }; |
| auto DisableUlpfec = [&]() { ulpfec_payload_type = -1; }; |
| |
| if (webrtc::field_trial::IsEnabled("WebRTC-DisableUlpFecExperiment")) { |
| RTC_LOG(LS_INFO) << "Experiment to disable sending ULPFEC is enabled."; |
| DisableUlpfec(); |
| } |
| |
| // If enabled, FlexFEC takes priority over RED+ULPFEC. |
| if (flexfec_enabled) { |
| // We can safely disable RED here, because if the remote supports FlexFEC, |
| // we know that it has a receiver without the RED/RTX workaround. |
| // See http://crbug.com/webrtc/6650 for more information. |
| if (IsRedEnabled()) { |
| RTC_LOG(LS_INFO) << "Both FlexFEC and RED are configured. Disabling RED."; |
| DisableRed(); |
| } |
| if (IsUlpfecEnabled()) { |
| RTC_LOG(LS_INFO) |
| << "Both FlexFEC and ULPFEC are configured. Disabling ULPFEC."; |
| DisableUlpfec(); |
| } |
| } |
| |
| // Payload types without picture ID cannot determine that a stream is complete |
| // without retransmitting FEC, so using ULPFEC + NACK for H.264 (for instance) |
| // is a waste of bandwidth since FEC packets still have to be transmitted. |
| // Note that this is not the case with FlexFEC. |
| if (nack_enabled && IsUlpfecEnabled() && |
| !PayloadTypeSupportsSkippingFecPackets( |
| config_->encoder_settings.payload_name)) { |
| RTC_LOG(LS_WARNING) |
| << "Transmitting payload type without picture ID using " |
| "NACK+ULPFEC is a waste of bandwidth since ULPFEC packets " |
| "also have to be retransmitted. Disabling ULPFEC."; |
| DisableUlpfec(); |
| } |
| |
| // Verify payload types. |
| // |
| // Due to how old receivers work, we need to always send RED if it has been |
| // negotiated. This is a remnant of an old RED/RTX workaround, see |
| // https://codereview.webrtc.org/2469093003. |
| // TODO(brandtr): This change went into M56, so we can remove it in ~M59. |
| // At that time, we can disable RED whenever ULPFEC is disabled, as there is |
| // no point in using RED without ULPFEC. |
| if (IsRedEnabled()) { |
| RTC_DCHECK_GE(red_payload_type, 0); |
| RTC_DCHECK_LE(red_payload_type, 127); |
| } |
| if (IsUlpfecEnabled()) { |
| RTC_DCHECK_GE(ulpfec_payload_type, 0); |
| RTC_DCHECK_LE(ulpfec_payload_type, 127); |
| if (!IsRedEnabled()) { |
| RTC_LOG(LS_WARNING) |
| << "ULPFEC is enabled but RED is disabled. Disabling ULPFEC."; |
| DisableUlpfec(); |
| } |
| } |
| |
| for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { |
| // Set NACK. |
| rtp_rtcp->SetStorePacketsStatus( |
| true, |
| kMinSendSidePacketHistorySize); |
| // Set RED/ULPFEC information. |
| rtp_rtcp->SetUlpfecConfig(red_payload_type, ulpfec_payload_type); |
| } |
| |
| // Currently, both ULPFEC and FlexFEC use the same FEC rate calculation logic, |
| // so enable that logic if either of those FEC schemes are enabled. |
| protection_bitrate_calculator_.SetProtectionMethod( |
| flexfec_enabled || IsUlpfecEnabled(), nack_enabled); |
| } |
| |
| void VideoSendStreamImpl::ConfigureSsrcs() { |
| RTC_DCHECK_RUN_ON(worker_queue_); |
| // Configure regular SSRCs. |
| for (size_t i = 0; i < config_->rtp.ssrcs.size(); ++i) { |
| uint32_t ssrc = config_->rtp.ssrcs[i]; |
| RtpRtcp* const rtp_rtcp = rtp_rtcp_modules_[i]; |
| rtp_rtcp->SetSSRC(ssrc); |
| |
| // Restore RTP state if previous existed. |
| VideoSendStream::RtpStateMap::iterator it = suspended_ssrcs_.find(ssrc); |
| if (it != suspended_ssrcs_.end()) |
| rtp_rtcp->SetRtpState(it->second); |
| } |
| |
| // Set up RTX if available. |
| if (config_->rtp.rtx.ssrcs.empty()) |
| return; |
| |
| // Configure RTX SSRCs. |
| RTC_DCHECK_EQ(config_->rtp.rtx.ssrcs.size(), config_->rtp.ssrcs.size()); |
| for (size_t i = 0; i < config_->rtp.rtx.ssrcs.size(); ++i) { |
| uint32_t ssrc = config_->rtp.rtx.ssrcs[i]; |
| RtpRtcp* const rtp_rtcp = rtp_rtcp_modules_[i]; |
| rtp_rtcp->SetRtxSsrc(ssrc); |
| VideoSendStream::RtpStateMap::iterator it = suspended_ssrcs_.find(ssrc); |
| if (it != suspended_ssrcs_.end()) |
| rtp_rtcp->SetRtxState(it->second); |
| } |
| |
| // Configure RTX payload types. |
| RTC_DCHECK_GE(config_->rtp.rtx.payload_type, 0); |
| for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { |
| rtp_rtcp->SetRtxSendPayloadType(config_->rtp.rtx.payload_type, |
| config_->encoder_settings.payload_type); |
| rtp_rtcp->SetRtxSendStatus(kRtxRetransmitted | kRtxRedundantPayloads); |
| } |
| if (config_->rtp.ulpfec.red_payload_type != -1 && |
| config_->rtp.ulpfec.red_rtx_payload_type != -1) { |
| for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { |
| rtp_rtcp->SetRtxSendPayloadType(config_->rtp.ulpfec.red_rtx_payload_type, |
| config_->rtp.ulpfec.red_payload_type); |
| } |
| } |
| } |
| |
| std::map<uint32_t, RtpState> VideoSendStreamImpl::GetRtpStates() const { |
| RTC_DCHECK_RUN_ON(worker_queue_); |
| std::map<uint32_t, RtpState> rtp_states; |
| |
| for (size_t i = 0; i < config_->rtp.ssrcs.size(); ++i) { |
| uint32_t ssrc = config_->rtp.ssrcs[i]; |
| RTC_DCHECK_EQ(ssrc, rtp_rtcp_modules_[i]->SSRC()); |
| rtp_states[ssrc] = rtp_rtcp_modules_[i]->GetRtpState(); |
| } |
| |
| for (size_t i = 0; i < config_->rtp.rtx.ssrcs.size(); ++i) { |
| uint32_t ssrc = config_->rtp.rtx.ssrcs[i]; |
| rtp_states[ssrc] = rtp_rtcp_modules_[i]->GetRtxState(); |
| } |
| |
| if (flexfec_sender_) { |
| uint32_t ssrc = config_->rtp.flexfec.ssrc; |
| rtp_states[ssrc] = flexfec_sender_->GetRtpState(); |
| } |
| |
| return rtp_states; |
| } |
| |
| std::map<uint32_t, RtpPayloadState> VideoSendStreamImpl::GetRtpPayloadStates() |
| const { |
| RTC_DCHECK_RUN_ON(worker_queue_); |
| return payload_router_.GetRtpPayloadStates(); |
| } |
| |
| void VideoSendStreamImpl::SignalNetworkState(NetworkState state) { |
| RTC_DCHECK_RUN_ON(worker_queue_); |
| for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { |
| rtp_rtcp->SetRTCPStatus(state == kNetworkUp ? config_->rtp.rtcp_mode |
| : RtcpMode::kOff); |
| } |
| } |
| |
| uint32_t VideoSendStreamImpl::OnBitrateUpdated(uint32_t bitrate_bps, |
| uint8_t fraction_loss, |
| int64_t rtt, |
| int64_t probing_interval_ms) { |
| RTC_DCHECK_RUN_ON(worker_queue_); |
| RTC_DCHECK(payload_router_.IsActive()) |
| << "VideoSendStream::Start has not been called."; |
| |
| // Substract overhead from bitrate. |
| rtc::CritScope lock(&overhead_bytes_per_packet_crit_); |
| uint32_t payload_bitrate_bps = bitrate_bps; |
| if (send_side_bwe_with_overhead_) { |
| payload_bitrate_bps -= CalculateOverheadRateBps( |
| CalculatePacketRate(bitrate_bps, |
| config_->rtp.max_packet_size + |
| transport_overhead_bytes_per_packet_), |
| overhead_bytes_per_packet_ + transport_overhead_bytes_per_packet_, |
| bitrate_bps); |
| } |
| |
| // Get the encoder target rate. It is the estimated network rate - |
| // protection overhead. |
| encoder_target_rate_bps_ = protection_bitrate_calculator_.SetTargetRates( |
| payload_bitrate_bps, stats_proxy_->GetSendFrameRate(), fraction_loss, |
| rtt); |
| |
| uint32_t encoder_overhead_rate_bps = |
| send_side_bwe_with_overhead_ |
| ? CalculateOverheadRateBps( |
| CalculatePacketRate(encoder_target_rate_bps_, |
| config_->rtp.max_packet_size + |
| transport_overhead_bytes_per_packet_ - |
| overhead_bytes_per_packet_), |
| overhead_bytes_per_packet_ + |
| transport_overhead_bytes_per_packet_, |
| bitrate_bps - encoder_target_rate_bps_) |
| : 0; |
| |
| // When the field trial "WebRTC-SendSideBwe-WithOverhead" is enabled |
| // protection_bitrate includes overhead. |
| uint32_t protection_bitrate = |
| bitrate_bps - (encoder_target_rate_bps_ + encoder_overhead_rate_bps); |
| |
| encoder_target_rate_bps_ = |
| std::min(encoder_max_bitrate_bps_, encoder_target_rate_bps_); |
| video_stream_encoder_->OnBitrateUpdated(encoder_target_rate_bps_, |
| fraction_loss, rtt); |
| stats_proxy_->OnSetEncoderTargetRate(encoder_target_rate_bps_); |
| return protection_bitrate; |
| } |
| |
| void VideoSendStreamImpl::EnableEncodedFrameRecording( |
| const std::vector<rtc::PlatformFile>& files, |
| size_t byte_limit) { |
| { |
| rtc::CritScope lock(&ivf_writers_crit_); |
| for (unsigned int i = 0; i < kMaxSimulcastStreams; ++i) { |
| if (i < files.size()) { |
| file_writers_[i] = IvfFileWriter::Wrap(rtc::File(files[i]), byte_limit); |
| } else { |
| file_writers_[i].reset(); |
| } |
| } |
| } |
| |
| if (!files.empty()) { |
| // Make a keyframe appear as early as possible in the logs, to give actually |
| // decodable output. |
| video_stream_encoder_->SendKeyFrame(); |
| } |
| } |
| |
| int VideoSendStreamImpl::ProtectionRequest( |
| const FecProtectionParams* delta_params, |
| const FecProtectionParams* key_params, |
| uint32_t* sent_video_rate_bps, |
| uint32_t* sent_nack_rate_bps, |
| uint32_t* sent_fec_rate_bps) { |
| RTC_DCHECK_RUN_ON(worker_queue_); |
| *sent_video_rate_bps = 0; |
| *sent_nack_rate_bps = 0; |
| *sent_fec_rate_bps = 0; |
| for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { |
| uint32_t not_used = 0; |
| uint32_t module_video_rate = 0; |
| uint32_t module_fec_rate = 0; |
| uint32_t module_nack_rate = 0; |
| rtp_rtcp->SetFecParameters(*delta_params, *key_params); |
| rtp_rtcp->BitrateSent(¬_used, &module_video_rate, &module_fec_rate, |
| &module_nack_rate); |
| *sent_video_rate_bps += module_video_rate; |
| *sent_nack_rate_bps += module_nack_rate; |
| *sent_fec_rate_bps += module_fec_rate; |
| } |
| return 0; |
| } |
| |
| void VideoSendStreamImpl::OnOverheadChanged(size_t overhead_bytes_per_packet) { |
| rtc::CritScope lock(&overhead_bytes_per_packet_crit_); |
| overhead_bytes_per_packet_ = overhead_bytes_per_packet; |
| } |
| |
| void VideoSendStreamImpl::SetTransportOverhead( |
| size_t transport_overhead_bytes_per_packet) { |
| if (transport_overhead_bytes_per_packet >= static_cast<int>(kPathMTU)) { |
| RTC_LOG(LS_ERROR) << "Transport overhead exceeds size of ethernet frame"; |
| return; |
| } |
| |
| transport_overhead_bytes_per_packet_ = transport_overhead_bytes_per_packet; |
| |
| transport_->send_side_cc()->SetTransportOverhead( |
| transport_overhead_bytes_per_packet_); |
| |
| size_t rtp_packet_size = |
| std::min(config_->rtp.max_packet_size, |
| kPathMTU - transport_overhead_bytes_per_packet_); |
| |
| for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { |
| rtp_rtcp->SetMaxRtpPacketSize(rtp_packet_size); |
| } |
| } |
| |
| } // namespace internal |
| } // namespace webrtc |