| /* |
| * Copyright (c) 2015 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 "call/rtp_video_sender.h" |
| |
| #include <algorithm> |
| #include <memory> |
| #include <string> |
| #include <utility> |
| |
| #include "absl/algorithm/container.h" |
| #include "absl/memory/memory.h" |
| #include "api/array_view.h" |
| #include "api/transport/field_trial_based_config.h" |
| #include "call/rtp_transport_controller_send_interface.h" |
| #include "modules/pacing/packet_router.h" |
| #include "modules/rtp_rtcp/include/rtp_rtcp.h" |
| #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" |
| #include "modules/rtp_rtcp/source/playout_delay_oracle.h" |
| #include "modules/rtp_rtcp/source/rtp_sender.h" |
| #include "modules/utility/include/process_thread.h" |
| #include "modules/video_coding/include/video_codec_interface.h" |
| #include "rtc_base/checks.h" |
| #include "rtc_base/location.h" |
| #include "rtc_base/logging.h" |
| #include "system_wrappers/include/field_trial.h" |
| |
| namespace webrtc { |
| |
| namespace webrtc_internal_rtp_video_sender { |
| |
| RtpStreamSender::RtpStreamSender( |
| std::unique_ptr<PlayoutDelayOracle> playout_delay_oracle, |
| std::unique_ptr<RtpRtcp> rtp_rtcp, |
| std::unique_ptr<RTPSenderVideo> sender_video) |
| : playout_delay_oracle(std::move(playout_delay_oracle)), |
| rtp_rtcp(std::move(rtp_rtcp)), |
| sender_video(std::move(sender_video)) {} |
| |
| RtpStreamSender::~RtpStreamSender() = default; |
| |
| } // namespace webrtc_internal_rtp_video_sender |
| |
| namespace { |
| static const int kMinSendSidePacketHistorySize = 600; |
| // We don't do MTU discovery, so assume that we have the standard ethernet MTU. |
| static const size_t kPathMTU = 1500; |
| |
| using webrtc_internal_rtp_video_sender::RtpStreamSender; |
| |
| std::vector<RtpStreamSender> CreateRtpStreamSenders( |
| Clock* clock, |
| const RtpConfig& rtp_config, |
| int rtcp_report_interval_ms, |
| Transport* send_transport, |
| RtcpIntraFrameObserver* intra_frame_callback, |
| RtcpLossNotificationObserver* rtcp_loss_notification_observer, |
| RtcpBandwidthObserver* bandwidth_callback, |
| RtpTransportControllerSendInterface* transport, |
| RtcpRttStats* rtt_stats, |
| FlexfecSender* flexfec_sender, |
| BitrateStatisticsObserver* bitrate_observer, |
| RtcpPacketTypeCounterObserver* rtcp_type_observer, |
| SendSideDelayObserver* send_delay_observer, |
| SendPacketObserver* send_packet_observer, |
| RtcEventLog* event_log, |
| RateLimiter* retransmission_rate_limiter, |
| OverheadObserver* overhead_observer, |
| FrameEncryptorInterface* frame_encryptor, |
| const CryptoOptions& crypto_options) { |
| RTC_DCHECK_GT(rtp_config.ssrcs.size(), 0); |
| |
| RtpRtcp::Configuration configuration; |
| configuration.clock = clock; |
| configuration.audio = false; |
| configuration.receiver_only = false; |
| configuration.outgoing_transport = send_transport; |
| configuration.intra_frame_callback = intra_frame_callback; |
| configuration.rtcp_loss_notification_observer = |
| rtcp_loss_notification_observer; |
| configuration.bandwidth_callback = bandwidth_callback; |
| configuration.transport_feedback_callback = |
| transport->transport_feedback_observer(); |
| configuration.rtt_stats = rtt_stats; |
| configuration.rtcp_packet_type_counter_observer = rtcp_type_observer; |
| configuration.paced_sender = transport->packet_sender(); |
| configuration.transport_sequence_number_allocator = |
| transport->packet_router(); |
| configuration.send_bitrate_observer = bitrate_observer; |
| configuration.send_side_delay_observer = send_delay_observer; |
| configuration.send_packet_observer = send_packet_observer; |
| configuration.event_log = event_log; |
| configuration.retransmission_rate_limiter = retransmission_rate_limiter; |
| configuration.overhead_observer = overhead_observer; |
| configuration.frame_encryptor = frame_encryptor; |
| configuration.require_frame_encryption = |
| crypto_options.sframe.require_frame_encryption; |
| configuration.extmap_allow_mixed = rtp_config.extmap_allow_mixed; |
| configuration.rtcp_report_interval_ms = rtcp_report_interval_ms; |
| |
| std::vector<RtpStreamSender> rtp_streams; |
| const std::vector<uint32_t>& flexfec_protected_ssrcs = |
| rtp_config.flexfec.protected_media_ssrcs; |
| for (uint32_t ssrc : rtp_config.ssrcs) { |
| bool enable_flexfec = flexfec_sender != nullptr && |
| std::find(flexfec_protected_ssrcs.begin(), |
| flexfec_protected_ssrcs.end(), |
| ssrc) != flexfec_protected_ssrcs.end(); |
| configuration.flexfec_sender = enable_flexfec ? flexfec_sender : nullptr; |
| auto playout_delay_oracle = absl::make_unique<PlayoutDelayOracle>(); |
| |
| configuration.ack_observer = playout_delay_oracle.get(); |
| auto rtp_rtcp = RtpRtcp::Create(configuration); |
| rtp_rtcp->SetSendingStatus(false); |
| rtp_rtcp->SetSendingMediaStatus(false); |
| rtp_rtcp->SetRTCPStatus(RtcpMode::kCompound); |
| |
| auto sender_video = absl::make_unique<RTPSenderVideo>( |
| configuration.clock, rtp_rtcp->RtpSender(), |
| configuration.flexfec_sender, playout_delay_oracle.get(), |
| frame_encryptor, crypto_options.sframe.require_frame_encryption, |
| rtp_config.lntf.enabled, FieldTrialBasedConfig()); |
| rtp_streams.emplace_back(std::move(playout_delay_oracle), |
| std::move(rtp_rtcp), std::move(sender_video)); |
| } |
| return rtp_streams; |
| } |
| |
| bool PayloadTypeSupportsSkippingFecPackets(const std::string& payload_name) { |
| const VideoCodecType codecType = PayloadStringToCodecType(payload_name); |
| if (codecType == kVideoCodecVP8 || codecType == kVideoCodecVP9) { |
| return true; |
| } |
| if (codecType == kVideoCodecGeneric && |
| field_trial::IsEnabled("WebRTC-GenericPictureId")) { |
| return true; |
| } |
| return false; |
| } |
| |
| // TODO(brandtr): Update this function when we support multistream protection. |
| std::unique_ptr<FlexfecSender> MaybeCreateFlexfecSender( |
| Clock* clock, |
| const RtpConfig& rtp, |
| const std::map<uint32_t, RtpState>& suspended_ssrcs) { |
| if (rtp.flexfec.payload_type < 0) { |
| return nullptr; |
| } |
| RTC_DCHECK_GE(rtp.flexfec.payload_type, 0); |
| RTC_DCHECK_LE(rtp.flexfec.payload_type, 127); |
| if (rtp.flexfec.ssrc == 0) { |
| RTC_LOG(LS_WARNING) << "FlexFEC is enabled, but no FlexFEC SSRC given. " |
| "Therefore disabling FlexFEC."; |
| return nullptr; |
| } |
| if (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 (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(rtp.flexfec.ssrc); |
| if (it != suspended_ssrcs.end()) { |
| rtp_state = &it->second; |
| } |
| |
| RTC_DCHECK_EQ(1U, rtp.flexfec.protected_media_ssrcs.size()); |
| return absl::make_unique<FlexfecSender>( |
| rtp.flexfec.payload_type, rtp.flexfec.ssrc, |
| rtp.flexfec.protected_media_ssrcs[0], rtp.mid, rtp.extensions, |
| RTPSender::FecExtensionSizes(), rtp_state, clock); |
| } |
| |
| DataRate CalculateOverheadRate(DataRate data_rate, |
| DataSize packet_size, |
| DataSize overhead_per_packet) { |
| Frequency packet_rate = data_rate / packet_size; |
| // TOSO(srte): We should not need to round to nearest whole packet per second |
| // rate here. |
| return packet_rate.RoundUpTo(Frequency::hertz(1)) * overhead_per_packet; |
| } |
| } // namespace |
| |
| RtpVideoSender::RtpVideoSender( |
| Clock* clock, |
| std::map<uint32_t, RtpState> suspended_ssrcs, |
| const std::map<uint32_t, RtpPayloadState>& states, |
| const RtpConfig& rtp_config, |
| int rtcp_report_interval_ms, |
| Transport* send_transport, |
| const RtpSenderObservers& observers, |
| RtpTransportControllerSendInterface* transport, |
| RtcEventLog* event_log, |
| RateLimiter* retransmission_limiter, |
| std::unique_ptr<FecController> fec_controller, |
| FrameEncryptorInterface* frame_encryptor, |
| const CryptoOptions& crypto_options) |
| : send_side_bwe_with_overhead_( |
| webrtc::field_trial::IsEnabled("WebRTC-SendSideBwe-WithOverhead")), |
| account_for_packetization_overhead_(!webrtc::field_trial::IsDisabled( |
| "WebRTC-SubtractPacketizationOverhead")), |
| use_early_loss_detection_( |
| !webrtc::field_trial::IsDisabled("WebRTC-UseEarlyLossDetection")), |
| active_(false), |
| module_process_thread_(nullptr), |
| suspended_ssrcs_(std::move(suspended_ssrcs)), |
| flexfec_sender_( |
| MaybeCreateFlexfecSender(clock, rtp_config, suspended_ssrcs_)), |
| fec_controller_(std::move(fec_controller)), |
| fec_allowed_(true), |
| rtp_streams_( |
| CreateRtpStreamSenders(clock, |
| rtp_config, |
| rtcp_report_interval_ms, |
| send_transport, |
| observers.intra_frame_callback, |
| observers.rtcp_loss_notification_observer, |
| transport->GetBandwidthObserver(), |
| transport, |
| observers.rtcp_rtt_stats, |
| flexfec_sender_.get(), |
| observers.bitrate_observer, |
| observers.rtcp_type_observer, |
| observers.send_delay_observer, |
| observers.send_packet_observer, |
| event_log, |
| retransmission_limiter, |
| this, |
| frame_encryptor, |
| crypto_options)), |
| rtp_config_(rtp_config), |
| transport_(transport), |
| transport_overhead_bytes_per_packet_(0), |
| overhead_bytes_per_packet_(0), |
| encoder_target_rate_bps_(0), |
| frame_counts_(rtp_config.ssrcs.size()), |
| frame_count_observer_(observers.frame_count_observer) { |
| RTC_DCHECK_EQ(rtp_config_.ssrcs.size(), rtp_streams_.size()); |
| module_process_thread_checker_.Detach(); |
| // SSRCs are assumed to be sorted in the same order as |rtp_modules|. |
| for (uint32_t ssrc : rtp_config_.ssrcs) { |
| // Restore state if it previously existed. |
| const RtpPayloadState* state = nullptr; |
| auto it = states.find(ssrc); |
| if (it != states.end()) { |
| state = &it->second; |
| shared_frame_id_ = std::max(shared_frame_id_, state->shared_frame_id); |
| } |
| params_.push_back(RtpPayloadParams(ssrc, state)); |
| } |
| |
| // 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. |
| |
| // TODO(nisse): Consider moving registration with PacketRouter last, after the |
| // modules are fully configured. |
| for (const RtpStreamSender& stream : rtp_streams_) { |
| constexpr bool remb_candidate = true; |
| transport->packet_router()->AddSendRtpModule(stream.rtp_rtcp.get(), |
| remb_candidate); |
| } |
| |
| for (size_t i = 0; i < rtp_config_.extensions.size(); ++i) { |
| const std::string& extension = rtp_config_.extensions[i].uri; |
| int id = rtp_config_.extensions[i].id; |
| RTC_DCHECK(RtpExtension::IsSupportedForVideo(extension)); |
| for (const RtpStreamSender& stream : rtp_streams_) { |
| RTC_CHECK(stream.rtp_rtcp->RegisterRtpHeaderExtension(extension, id)); |
| } |
| } |
| |
| ConfigureProtection(); |
| ConfigureSsrcs(); |
| ConfigureRids(); |
| |
| if (!rtp_config_.mid.empty()) { |
| for (const RtpStreamSender& stream : rtp_streams_) { |
| stream.rtp_rtcp->SetMid(rtp_config_.mid); |
| } |
| } |
| |
| for (const RtpStreamSender& stream : rtp_streams_) { |
| // Simulcast has one module for each layer. Set the CNAME on all modules. |
| stream.rtp_rtcp->SetCNAME(rtp_config_.c_name.c_str()); |
| stream.rtp_rtcp->RegisterRtcpStatisticsCallback(observers.rtcp_stats); |
| stream.rtp_rtcp->SetReportBlockDataObserver( |
| observers.report_block_data_observer); |
| stream.rtp_rtcp->RegisterSendChannelRtpStatisticsCallback( |
| observers.rtp_stats); |
| stream.rtp_rtcp->SetMaxRtpPacketSize(rtp_config_.max_packet_size); |
| stream.rtp_rtcp->RegisterSendPayloadFrequency(rtp_config_.payload_type, |
| kVideoPayloadTypeFrequency); |
| stream.sender_video->RegisterPayloadType(rtp_config_.payload_type, |
| rtp_config_.payload_name, |
| rtp_config_.raw_payload); |
| } |
| // Currently, both ULPFEC and FlexFEC use the same FEC rate calculation logic, |
| // so enable that logic if either of those FEC schemes are enabled. |
| fec_controller_->SetProtectionMethod(FecEnabled(), NackEnabled()); |
| |
| fec_controller_->SetProtectionCallback(this); |
| // Signal congestion controller this object is ready for OnPacket* callbacks. |
| transport_->RegisterPacketFeedbackObserver(this); |
| } |
| |
| RtpVideoSender::~RtpVideoSender() { |
| for (const RtpStreamSender& stream : rtp_streams_) { |
| transport_->packet_router()->RemoveSendRtpModule(stream.rtp_rtcp.get()); |
| } |
| transport_->DeRegisterPacketFeedbackObserver(this); |
| } |
| |
| void RtpVideoSender::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 (const RtpStreamSender& stream : rtp_streams_) { |
| module_process_thread_->RegisterModule(stream.rtp_rtcp.get(), |
| RTC_FROM_HERE); |
| } |
| } |
| |
| void RtpVideoSender::DeRegisterProcessThread() { |
| RTC_DCHECK_RUN_ON(&module_process_thread_checker_); |
| for (const RtpStreamSender& stream : rtp_streams_) |
| module_process_thread_->DeRegisterModule(stream.rtp_rtcp.get()); |
| } |
| |
| void RtpVideoSender::SetActive(bool active) { |
| rtc::CritScope lock(&crit_); |
| if (active_ == active) |
| return; |
| const std::vector<bool> active_modules(rtp_streams_.size(), active); |
| SetActiveModules(active_modules); |
| } |
| |
| void RtpVideoSender::SetActiveModules(const std::vector<bool> active_modules) { |
| rtc::CritScope lock(&crit_); |
| RTC_DCHECK_EQ(rtp_streams_.size(), active_modules.size()); |
| active_ = false; |
| for (size_t i = 0; i < active_modules.size(); ++i) { |
| if (active_modules[i]) { |
| active_ = true; |
| } |
| // Sends a kRtcpByeCode when going from true to false. |
| rtp_streams_[i].rtp_rtcp->SetSendingStatus(active_modules[i]); |
| // If set to false this module won't send media. |
| rtp_streams_[i].rtp_rtcp->SetSendingMediaStatus(active_modules[i]); |
| } |
| } |
| |
| bool RtpVideoSender::IsActive() { |
| rtc::CritScope lock(&crit_); |
| return active_ && !rtp_streams_.empty(); |
| } |
| |
| EncodedImageCallback::Result RtpVideoSender::OnEncodedImage( |
| const EncodedImage& encoded_image, |
| const CodecSpecificInfo* codec_specific_info, |
| const RTPFragmentationHeader* fragmentation) { |
| fec_controller_->UpdateWithEncodedData(encoded_image.size(), |
| encoded_image._frameType); |
| rtc::CritScope lock(&crit_); |
| RTC_DCHECK(!rtp_streams_.empty()); |
| if (!active_) |
| return Result(Result::ERROR_SEND_FAILED); |
| |
| shared_frame_id_++; |
| size_t stream_index = 0; |
| if (codec_specific_info && |
| (codec_specific_info->codecType == kVideoCodecVP8 || |
| codec_specific_info->codecType == kVideoCodecH264 || |
| codec_specific_info->codecType == kVideoCodecGeneric)) { |
| // Map spatial index to simulcast. |
| stream_index = encoded_image.SpatialIndex().value_or(0); |
| } |
| RTC_DCHECK_LT(stream_index, rtp_streams_.size()); |
| RTPVideoHeader rtp_video_header = params_[stream_index].GetRtpVideoHeader( |
| encoded_image, codec_specific_info, shared_frame_id_); |
| |
| uint32_t rtp_timestamp = |
| encoded_image.Timestamp() + |
| rtp_streams_[stream_index].rtp_rtcp->StartTimestamp(); |
| |
| // RTCPSender has it's own copy of the timestamp offset, added in |
| // RTCPSender::BuildSR, hence we must not add the in the offset for this call. |
| // TODO(nisse): Delete RTCPSender:timestamp_offset_, and see if we can confine |
| // knowledge of the offset to a single place. |
| if (!rtp_streams_[stream_index].rtp_rtcp->OnSendingRtpFrame( |
| encoded_image.Timestamp(), encoded_image.capture_time_ms_, |
| rtp_config_.payload_type, |
| encoded_image._frameType == VideoFrameType::kVideoFrameKey)) { |
| // The payload router could be active but this module isn't sending. |
| return Result(Result::ERROR_SEND_FAILED); |
| } |
| |
| absl::optional<int64_t> expected_retransmission_time_ms; |
| if (encoded_image.RetransmissionAllowed()) { |
| expected_retransmission_time_ms = |
| rtp_streams_[stream_index].rtp_rtcp->ExpectedRetransmissionTimeMs(); |
| } |
| |
| bool send_result = rtp_streams_[stream_index].sender_video->SendVideo( |
| encoded_image._frameType, rtp_config_.payload_type, rtp_timestamp, |
| encoded_image.capture_time_ms_, encoded_image.data(), |
| encoded_image.size(), fragmentation, &rtp_video_header, |
| expected_retransmission_time_ms); |
| if (frame_count_observer_) { |
| FrameCounts& counts = frame_counts_[stream_index]; |
| if (encoded_image._frameType == VideoFrameType::kVideoFrameKey) { |
| ++counts.key_frames; |
| } else if (encoded_image._frameType == VideoFrameType::kVideoFrameDelta) { |
| ++counts.delta_frames; |
| } else { |
| RTC_DCHECK(encoded_image._frameType == VideoFrameType::kEmptyFrame); |
| } |
| frame_count_observer_->FrameCountUpdated(counts, |
| rtp_config_.ssrcs[stream_index]); |
| } |
| if (!send_result) |
| return Result(Result::ERROR_SEND_FAILED); |
| |
| return Result(Result::OK, rtp_timestamp); |
| } |
| |
| void RtpVideoSender::OnBitrateAllocationUpdated( |
| const VideoBitrateAllocation& bitrate) { |
| rtc::CritScope lock(&crit_); |
| if (IsActive()) { |
| if (rtp_streams_.size() == 1) { |
| // If spatial scalability is enabled, it is covered by a single stream. |
| rtp_streams_[0].rtp_rtcp->SetVideoBitrateAllocation(bitrate); |
| } else { |
| std::vector<absl::optional<VideoBitrateAllocation>> layer_bitrates = |
| bitrate.GetSimulcastAllocations(); |
| // Simulcast is in use, split the VideoBitrateAllocation into one struct |
| // per rtp stream, moving over the temporal layer allocation. |
| for (size_t i = 0; i < rtp_streams_.size(); ++i) { |
| // The next spatial layer could be used if the current one is |
| // inactive. |
| if (layer_bitrates[i]) { |
| rtp_streams_[i].rtp_rtcp->SetVideoBitrateAllocation( |
| *layer_bitrates[i]); |
| } else { |
| // Signal a 0 bitrate on a simulcast stream. |
| rtp_streams_[i].rtp_rtcp->SetVideoBitrateAllocation( |
| VideoBitrateAllocation()); |
| } |
| } |
| } |
| } |
| } |
| |
| void RtpVideoSender::ConfigureProtection() { |
| // 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 = rtp_config_.nack.rtp_history_ms > 0; |
| int red_payload_type = rtp_config_.ulpfec.red_payload_type; |
| int ulpfec_payload_type = rtp_config_.ulpfec.ulpfec_payload_type; |
| |
| // Shorthands. |
| auto IsRedEnabled = [&]() { return red_payload_type >= 0; }; |
| auto IsUlpfecEnabled = [&]() { return ulpfec_payload_type >= 0; }; |
| auto DisableRedAndUlpfec = [&]() { |
| red_payload_type = -1; |
| ulpfec_payload_type = -1; |
| }; |
| |
| if (webrtc::field_trial::IsEnabled("WebRTC-DisableUlpFecExperiment")) { |
| RTC_LOG(LS_INFO) << "Experiment to disable sending ULPFEC is enabled."; |
| DisableRedAndUlpfec(); |
| } |
| |
| // If enabled, FlexFEC takes priority over RED+ULPFEC. |
| if (flexfec_enabled) { |
| if (IsUlpfecEnabled()) { |
| RTC_LOG(LS_INFO) |
| << "Both FlexFEC and ULPFEC are configured. Disabling ULPFEC."; |
| } |
| DisableRedAndUlpfec(); |
| } |
| |
| // 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(rtp_config_.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."; |
| DisableRedAndUlpfec(); |
| } |
| |
| // Verify payload types. |
| if (IsUlpfecEnabled() ^ IsRedEnabled()) { |
| RTC_LOG(LS_WARNING) |
| << "Only RED or only ULPFEC enabled, but not both. Disabling both."; |
| DisableRedAndUlpfec(); |
| } |
| |
| for (const RtpStreamSender& stream : rtp_streams_) { |
| // Set NACK. |
| stream.rtp_rtcp->SetStorePacketsStatus(true, kMinSendSidePacketHistorySize); |
| // Set RED/ULPFEC information. |
| stream.sender_video->SetUlpfecConfig(red_payload_type, ulpfec_payload_type); |
| } |
| } |
| |
| bool RtpVideoSender::FecEnabled() const { |
| const bool flexfec_enabled = (flexfec_sender_ != nullptr); |
| const bool ulpfec_enabled = |
| !webrtc::field_trial::IsEnabled("WebRTC-DisableUlpFecExperiment") && |
| (rtp_config_.ulpfec.ulpfec_payload_type >= 0); |
| return flexfec_enabled || ulpfec_enabled; |
| } |
| |
| bool RtpVideoSender::NackEnabled() const { |
| const bool nack_enabled = rtp_config_.nack.rtp_history_ms > 0; |
| return nack_enabled; |
| } |
| |
| uint32_t RtpVideoSender::GetPacketizationOverheadRate() const { |
| uint32_t packetization_overhead_bps = 0; |
| for (size_t i = 0; i < rtp_streams_.size(); ++i) { |
| if (rtp_streams_[i].rtp_rtcp->SendingMedia()) { |
| packetization_overhead_bps += |
| rtp_streams_[i].sender_video->PacketizationOverheadBps(); |
| } |
| } |
| return packetization_overhead_bps; |
| } |
| |
| void RtpVideoSender::DeliverRtcp(const uint8_t* packet, size_t length) { |
| // Runs on a network thread. |
| for (const RtpStreamSender& stream : rtp_streams_) |
| stream.rtp_rtcp->IncomingRtcpPacket(packet, length); |
| } |
| |
| void RtpVideoSender::ConfigureSsrcs() { |
| // Configure regular SSRCs. |
| RTC_CHECK(ssrc_to_rtp_sender_.empty()); |
| for (size_t i = 0; i < rtp_config_.ssrcs.size(); ++i) { |
| uint32_t ssrc = rtp_config_.ssrcs[i]; |
| RtpRtcp* const rtp_rtcp = rtp_streams_[i].rtp_rtcp.get(); |
| rtp_rtcp->SetSSRC(ssrc); |
| |
| // Restore RTP state if previous existed. |
| auto it = suspended_ssrcs_.find(ssrc); |
| if (it != suspended_ssrcs_.end()) |
| rtp_rtcp->SetRtpState(it->second); |
| |
| RTPSender* rtp_sender = rtp_rtcp->RtpSender(); |
| RTC_DCHECK(rtp_sender != nullptr); |
| ssrc_to_rtp_sender_[ssrc] = rtp_sender; |
| } |
| |
| // Set up RTX if available. |
| if (rtp_config_.rtx.ssrcs.empty()) |
| return; |
| |
| // Configure RTX SSRCs. |
| RTC_DCHECK_EQ(rtp_config_.rtx.ssrcs.size(), rtp_config_.ssrcs.size()); |
| for (size_t i = 0; i < rtp_config_.rtx.ssrcs.size(); ++i) { |
| uint32_t ssrc = rtp_config_.rtx.ssrcs[i]; |
| RtpRtcp* const rtp_rtcp = rtp_streams_[i].rtp_rtcp.get(); |
| rtp_rtcp->SetRtxSsrc(ssrc); |
| auto it = suspended_ssrcs_.find(ssrc); |
| if (it != suspended_ssrcs_.end()) |
| rtp_rtcp->SetRtxState(it->second); |
| } |
| |
| // Configure RTX payload types. |
| RTC_DCHECK_GE(rtp_config_.rtx.payload_type, 0); |
| for (const RtpStreamSender& stream : rtp_streams_) { |
| stream.rtp_rtcp->SetRtxSendPayloadType(rtp_config_.rtx.payload_type, |
| rtp_config_.payload_type); |
| stream.rtp_rtcp->SetRtxSendStatus(kRtxRetransmitted | |
| kRtxRedundantPayloads); |
| } |
| if (rtp_config_.ulpfec.red_payload_type != -1 && |
| rtp_config_.ulpfec.red_rtx_payload_type != -1) { |
| for (const RtpStreamSender& stream : rtp_streams_) { |
| stream.rtp_rtcp->SetRtxSendPayloadType( |
| rtp_config_.ulpfec.red_rtx_payload_type, |
| rtp_config_.ulpfec.red_payload_type); |
| } |
| } |
| } |
| |
| void RtpVideoSender::ConfigureRids() { |
| RTC_DCHECK(rtp_config_.rids.empty() || |
| rtp_config_.rids.size() == rtp_config_.ssrcs.size()); |
| RTC_DCHECK(rtp_config_.rids.empty() || |
| rtp_config_.rids.size() == rtp_streams_.size()); |
| for (size_t i = 0; i < rtp_config_.rids.size(); ++i) { |
| const std::string& rid = rtp_config_.rids[i]; |
| rtp_streams_[i].rtp_rtcp->SetRid(rid); |
| } |
| } |
| |
| void RtpVideoSender::OnNetworkAvailability(bool network_available) { |
| for (const RtpStreamSender& stream : rtp_streams_) { |
| stream.rtp_rtcp->SetRTCPStatus(network_available ? rtp_config_.rtcp_mode |
| : RtcpMode::kOff); |
| } |
| } |
| |
| std::map<uint32_t, RtpState> RtpVideoSender::GetRtpStates() const { |
| std::map<uint32_t, RtpState> rtp_states; |
| |
| for (size_t i = 0; i < rtp_config_.ssrcs.size(); ++i) { |
| uint32_t ssrc = rtp_config_.ssrcs[i]; |
| RTC_DCHECK_EQ(ssrc, rtp_streams_[i].rtp_rtcp->SSRC()); |
| rtp_states[ssrc] = rtp_streams_[i].rtp_rtcp->GetRtpState(); |
| } |
| |
| for (size_t i = 0; i < rtp_config_.rtx.ssrcs.size(); ++i) { |
| uint32_t ssrc = rtp_config_.rtx.ssrcs[i]; |
| rtp_states[ssrc] = rtp_streams_[i].rtp_rtcp->GetRtxState(); |
| } |
| |
| if (flexfec_sender_) { |
| uint32_t ssrc = rtp_config_.flexfec.ssrc; |
| rtp_states[ssrc] = flexfec_sender_->GetRtpState(); |
| } |
| |
| return rtp_states; |
| } |
| |
| std::map<uint32_t, RtpPayloadState> RtpVideoSender::GetRtpPayloadStates() |
| const { |
| rtc::CritScope lock(&crit_); |
| std::map<uint32_t, RtpPayloadState> payload_states; |
| for (const auto& param : params_) { |
| payload_states[param.ssrc()] = param.state(); |
| payload_states[param.ssrc()].shared_frame_id = shared_frame_id_; |
| } |
| return payload_states; |
| } |
| |
| void RtpVideoSender::OnTransportOverheadChanged( |
| size_t transport_overhead_bytes_per_packet) { |
| rtc::CritScope lock(&crit_); |
| transport_overhead_bytes_per_packet_ = transport_overhead_bytes_per_packet; |
| |
| size_t max_rtp_packet_size = |
| std::min(rtp_config_.max_packet_size, |
| kPathMTU - transport_overhead_bytes_per_packet_); |
| for (const RtpStreamSender& stream : rtp_streams_) { |
| stream.rtp_rtcp->SetMaxRtpPacketSize(max_rtp_packet_size); |
| } |
| } |
| |
| void RtpVideoSender::OnOverheadChanged(size_t overhead_bytes_per_packet) { |
| rtc::CritScope lock(&crit_); |
| overhead_bytes_per_packet_ = overhead_bytes_per_packet; |
| } |
| |
| void RtpVideoSender::OnBitrateUpdated(uint32_t bitrate_bps, |
| uint8_t fraction_loss, |
| int64_t rtt, |
| int framerate) { |
| // Substract overhead from bitrate. |
| rtc::CritScope lock(&crit_); |
| DataSize packet_overhead = DataSize::bytes( |
| overhead_bytes_per_packet_ + transport_overhead_bytes_per_packet_); |
| DataSize max_total_packet_size = DataSize::bytes( |
| rtp_config_.max_packet_size + transport_overhead_bytes_per_packet_); |
| uint32_t payload_bitrate_bps = bitrate_bps; |
| if (send_side_bwe_with_overhead_) { |
| DataRate overhead_rate = CalculateOverheadRate( |
| DataRate::bps(bitrate_bps), max_total_packet_size, packet_overhead); |
| // TODO(srte): We probably should not accept 0 payload bitrate here. |
| payload_bitrate_bps = |
| rtc::saturated_cast<uint32_t>(bitrate_bps - overhead_rate.bps()); |
| } |
| |
| // Get the encoder target rate. It is the estimated network rate - |
| // protection overhead. |
| encoder_target_rate_bps_ = fec_controller_->UpdateFecRates( |
| payload_bitrate_bps, framerate, fraction_loss, loss_mask_vector_, rtt); |
| if (!fec_allowed_) { |
| encoder_target_rate_bps_ = payload_bitrate_bps; |
| // fec_controller_->UpdateFecRates() was still called so as to allow |
| // |fec_controller_| to update whatever internal state it might have, |
| // since |fec_allowed_| may be toggled back on at any moment. |
| } |
| |
| uint32_t packetization_rate_bps = 0; |
| if (account_for_packetization_overhead_) { |
| // Subtract packetization overhead from the encoder target. If target rate |
| // is really low, cap the overhead at 50%. This also avoids the case where |
| // |encoder_target_rate_bps_| is 0 due to encoder pause event while the |
| // packetization rate is positive since packets are still flowing. |
| packetization_rate_bps = |
| std::min(GetPacketizationOverheadRate(), encoder_target_rate_bps_ / 2); |
| encoder_target_rate_bps_ -= packetization_rate_bps; |
| } |
| |
| loss_mask_vector_.clear(); |
| |
| uint32_t encoder_overhead_rate_bps = 0; |
| if (send_side_bwe_with_overhead_) { |
| // TODO(srte): The packet size should probably be the same as in the |
| // CalculateOverheadRate call above (just max_total_packet_size), it doesn't |
| // make sense to use different packet rates for different overhead |
| // calculations. |
| DataRate encoder_overhead_rate = CalculateOverheadRate( |
| DataRate::bps(encoder_target_rate_bps_), |
| max_total_packet_size - DataSize::bytes(overhead_bytes_per_packet_), |
| packet_overhead); |
| encoder_overhead_rate_bps = |
| std::min(encoder_overhead_rate.bps<uint32_t>(), |
| bitrate_bps - encoder_target_rate_bps_); |
| } |
| // When the field trial "WebRTC-SendSideBwe-WithOverhead" is enabled |
| // protection_bitrate includes overhead. |
| const uint32_t media_rate = encoder_target_rate_bps_ + |
| encoder_overhead_rate_bps + |
| packetization_rate_bps; |
| RTC_DCHECK_GE(bitrate_bps, media_rate); |
| protection_bitrate_bps_ = bitrate_bps - media_rate; |
| } |
| |
| uint32_t RtpVideoSender::GetPayloadBitrateBps() const { |
| return encoder_target_rate_bps_; |
| } |
| |
| uint32_t RtpVideoSender::GetProtectionBitrateBps() const { |
| return protection_bitrate_bps_; |
| } |
| |
| std::vector<RtpSequenceNumberMap::Info> RtpVideoSender::GetSentRtpPacketInfos( |
| uint32_t ssrc, |
| rtc::ArrayView<const uint16_t> sequence_numbers) const { |
| for (const auto& rtp_stream : rtp_streams_) { |
| if (ssrc == rtp_stream.rtp_rtcp->SSRC()) { |
| return rtp_stream.sender_video->GetSentRtpPacketInfos(sequence_numbers); |
| } |
| } |
| return std::vector<RtpSequenceNumberMap::Info>(); |
| } |
| |
| int RtpVideoSender::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) { |
| *sent_video_rate_bps = 0; |
| *sent_nack_rate_bps = 0; |
| *sent_fec_rate_bps = 0; |
| for (const RtpStreamSender& stream : rtp_streams_) { |
| uint32_t not_used = 0; |
| uint32_t module_nack_rate = 0; |
| stream.sender_video->SetFecParameters(*delta_params, *key_params); |
| *sent_video_rate_bps += stream.sender_video->VideoBitrateSent(); |
| *sent_fec_rate_bps += stream.sender_video->FecOverheadRate(); |
| stream.rtp_rtcp->BitrateSent(¬_used, /*video_rate=*/nullptr, |
| /*fec_rate=*/nullptr, &module_nack_rate); |
| *sent_nack_rate_bps += module_nack_rate; |
| } |
| return 0; |
| } |
| |
| void RtpVideoSender::SetFecAllowed(bool fec_allowed) { |
| rtc::CritScope cs(&crit_); |
| fec_allowed_ = fec_allowed; |
| } |
| |
| void RtpVideoSender::OnPacketFeedbackVector( |
| const std::vector<PacketFeedback>& packet_feedback_vector) { |
| if (fec_controller_->UseLossVectorMask()) { |
| rtc::CritScope cs(&crit_); |
| for (const PacketFeedback& packet : packet_feedback_vector) { |
| if (packet.send_time_ms == PacketFeedback::kNoSendTime || !packet.ssrc || |
| absl::c_find(rtp_config_.ssrcs, *packet.ssrc) == |
| rtp_config_.ssrcs.end()) { |
| // If packet send time is missing, the feedback for this packet has |
| // probably already been processed, so ignore it. |
| // If packet does not belong to a registered media ssrc, we are also |
| // not interested in it. |
| continue; |
| } |
| loss_mask_vector_.push_back(packet.arrival_time_ms == |
| PacketFeedback::kNotReceived); |
| } |
| } |
| |
| // Map from SSRC to all acked packets for that RTP module. |
| std::map<uint32_t, std::vector<uint16_t>> acked_packets_per_ssrc; |
| for (const PacketFeedback& packet : packet_feedback_vector) { |
| if (packet.ssrc && packet.arrival_time_ms != PacketFeedback::kNotReceived) { |
| acked_packets_per_ssrc[*packet.ssrc].push_back( |
| packet.rtp_sequence_number); |
| } |
| } |
| |
| if (use_early_loss_detection_) { |
| // Map from SSRC to vector of RTP sequence numbers that are indicated as |
| // lost by feedback, without being trailed by any received packets. |
| std::map<uint32_t, std::vector<uint16_t>> early_loss_detected_per_ssrc; |
| |
| for (const PacketFeedback& packet : packet_feedback_vector) { |
| if (packet.send_time_ms == PacketFeedback::kNoSendTime || !packet.ssrc || |
| absl::c_find(rtp_config_.ssrcs, *packet.ssrc) == |
| rtp_config_.ssrcs.end()) { |
| // If packet send time is missing, the feedback for this packet has |
| // probably already been processed, so ignore it. |
| // If packet does not belong to a registered media ssrc, we are also |
| // not interested in it. |
| continue; |
| } |
| |
| if (packet.arrival_time_ms == PacketFeedback::kNotReceived) { |
| // Last known lost packet, might not be detectable as lost by remote |
| // jitter buffer. |
| early_loss_detected_per_ssrc[*packet.ssrc].push_back( |
| packet.rtp_sequence_number); |
| } else { |
| // Packet received, so any loss prior to this is already detectable. |
| early_loss_detected_per_ssrc.erase(*packet.ssrc); |
| } |
| } |
| |
| for (const auto& kv : early_loss_detected_per_ssrc) { |
| const uint32_t ssrc = kv.first; |
| auto it = ssrc_to_rtp_sender_.find(ssrc); |
| RTC_DCHECK(it != ssrc_to_rtp_sender_.end()); |
| RTPSender* rtp_sender = it->second; |
| for (uint16_t sequence_number : kv.second) { |
| rtp_sender->ReSendPacket(sequence_number); |
| } |
| } |
| } |
| |
| for (const auto& kv : acked_packets_per_ssrc) { |
| const uint32_t ssrc = kv.first; |
| auto it = ssrc_to_rtp_sender_.find(ssrc); |
| if (it == ssrc_to_rtp_sender_.end()) { |
| // Packets not for a media SSRC, so likely RTX or FEC. If so, ignore |
| // since there's no RTP history to clean up anyway. |
| continue; |
| } |
| rtc::ArrayView<const uint16_t> rtp_sequence_numbers(kv.second); |
| it->second->OnPacketsAcknowledged(rtp_sequence_numbers); |
| } |
| } |
| |
| void RtpVideoSender::SetEncodingData(size_t width, |
| size_t height, |
| size_t num_temporal_layers) { |
| fec_controller_->SetEncodingData(width, height, num_temporal_layers, |
| rtp_config_.max_packet_size); |
| } |
| } // namespace webrtc |
