| /* |
| * 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 <utility> |
| |
| #include "api/array_view.h" |
| #include "api/video/video_stream_encoder_settings.h" |
| #include "modules/rtp_rtcp/include/rtp_header_extension_map.h" |
| #include "modules/rtp_rtcp/source/rtp_header_extension_size.h" |
| #include "modules/rtp_rtcp/source/rtp_sender.h" |
| #include "rtc_base/checks.h" |
| #include "rtc_base/logging.h" |
| #include "rtc_base/strings/string_builder.h" |
| #include "rtc_base/task_utils/to_queued_task.h" |
| #include "system_wrappers/include/clock.h" |
| #include "system_wrappers/include/field_trial.h" |
| #include "video/adaptation/overuse_frame_detector.h" |
| #include "video/video_stream_encoder.h" |
| |
| namespace webrtc { |
| |
| namespace { |
| |
| size_t CalculateMaxHeaderSize(const RtpConfig& config) { |
| size_t header_size = kRtpHeaderSize; |
| size_t extensions_size = 0; |
| size_t fec_extensions_size = 0; |
| if (!config.extensions.empty()) { |
| RtpHeaderExtensionMap extensions_map(config.extensions); |
| extensions_size = RtpHeaderExtensionSize(RTPSender::VideoExtensionSizes(), |
| extensions_map); |
| fec_extensions_size = |
| RtpHeaderExtensionSize(RTPSender::FecExtensionSizes(), extensions_map); |
| } |
| header_size += extensions_size; |
| if (config.flexfec.payload_type >= 0) { |
| // All FEC extensions again plus maximum FlexFec overhead. |
| header_size += fec_extensions_size + 32; |
| } else { |
| if (config.ulpfec.ulpfec_payload_type >= 0) { |
| // Header with all the FEC extensions will be repeated plus maximum |
| // UlpFec overhead. |
| header_size += fec_extensions_size + 18; |
| } |
| if (config.ulpfec.red_payload_type >= 0) { |
| header_size += 1; // RED header. |
| } |
| } |
| // Additional room for Rtx. |
| if (config.rtx.payload_type >= 0) |
| header_size += kRtxHeaderSize; |
| return header_size; |
| } |
| |
| VideoStreamEncoder::BitrateAllocationCallbackType |
| GetBitrateAllocationCallbackType(const VideoSendStream::Config& config) { |
| if (webrtc::RtpExtension::FindHeaderExtensionByUri( |
| config.rtp.extensions, |
| webrtc::RtpExtension::kVideoLayersAllocationUri, |
| config.crypto_options.srtp.enable_encrypted_rtp_header_extensions |
| ? RtpExtension::Filter::kPreferEncryptedExtension |
| : RtpExtension::Filter::kDiscardEncryptedExtension)) { |
| return VideoStreamEncoder::BitrateAllocationCallbackType:: |
| kVideoLayersAllocation; |
| } |
| if (field_trial::IsEnabled("WebRTC-Target-Bitrate-Rtcp")) { |
| return VideoStreamEncoder::BitrateAllocationCallbackType:: |
| kVideoBitrateAllocation; |
| } |
| return VideoStreamEncoder::BitrateAllocationCallbackType:: |
| kVideoBitrateAllocationWhenScreenSharing; |
| } |
| |
| RtpSenderFrameEncryptionConfig CreateFrameEncryptionConfig( |
| const VideoSendStream::Config* config) { |
| RtpSenderFrameEncryptionConfig frame_encryption_config; |
| frame_encryption_config.frame_encryptor = config->frame_encryptor; |
| frame_encryption_config.crypto_options = config->crypto_options; |
| return frame_encryption_config; |
| } |
| |
| RtpSenderObservers CreateObservers(RtcpRttStats* call_stats, |
| EncoderRtcpFeedback* encoder_feedback, |
| SendStatisticsProxy* stats_proxy, |
| SendDelayStats* send_delay_stats) { |
| RtpSenderObservers observers; |
| observers.rtcp_rtt_stats = call_stats; |
| observers.intra_frame_callback = encoder_feedback; |
| observers.rtcp_loss_notification_observer = encoder_feedback; |
| observers.report_block_data_observer = stats_proxy; |
| observers.rtp_stats = stats_proxy; |
| observers.bitrate_observer = stats_proxy; |
| observers.frame_count_observer = stats_proxy; |
| observers.rtcp_type_observer = stats_proxy; |
| observers.send_delay_observer = stats_proxy; |
| observers.send_packet_observer = send_delay_stats; |
| return observers; |
| } |
| |
| } // namespace |
| |
| namespace internal { |
| |
| VideoSendStream::VideoSendStream( |
| Clock* clock, |
| int num_cpu_cores, |
| TaskQueueFactory* task_queue_factory, |
| RtcpRttStats* call_stats, |
| RtpTransportControllerSendInterface* transport, |
| BitrateAllocatorInterface* 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, |
| std::unique_ptr<FecController> fec_controller) |
| : rtp_transport_queue_(transport->GetWorkerQueue()), |
| transport_(transport), |
| stats_proxy_(clock, config, encoder_config.content_type), |
| config_(std::move(config)), |
| content_type_(encoder_config.content_type), |
| video_stream_encoder_(std::make_unique<VideoStreamEncoder>( |
| clock, |
| num_cpu_cores, |
| &stats_proxy_, |
| config_.encoder_settings, |
| std::make_unique<OveruseFrameDetector>(&stats_proxy_), |
| task_queue_factory, |
| GetBitrateAllocationCallbackType(config_))), |
| encoder_feedback_( |
| clock, |
| config_.rtp.ssrcs, |
| video_stream_encoder_.get(), |
| [this](uint32_t ssrc, const std::vector<uint16_t>& seq_nums) { |
| return rtp_video_sender_->GetSentRtpPacketInfos(ssrc, seq_nums); |
| }), |
| rtp_video_sender_( |
| transport->CreateRtpVideoSender(suspended_ssrcs, |
| suspended_payload_states, |
| config_.rtp, |
| config_.rtcp_report_interval_ms, |
| config_.send_transport, |
| CreateObservers(call_stats, |
| &encoder_feedback_, |
| &stats_proxy_, |
| send_delay_stats), |
| event_log, |
| std::move(fec_controller), |
| CreateFrameEncryptionConfig(&config_), |
| config_.frame_transformer)), |
| send_stream_(clock, |
| &stats_proxy_, |
| rtp_transport_queue_, |
| transport, |
| bitrate_allocator, |
| video_stream_encoder_.get(), |
| &config_, |
| encoder_config.max_bitrate_bps, |
| encoder_config.bitrate_priority, |
| encoder_config.content_type, |
| rtp_video_sender_) { |
| RTC_DCHECK(config_.encoder_settings.encoder_factory); |
| RTC_DCHECK(config_.encoder_settings.bitrate_allocator_factory); |
| |
| video_stream_encoder_->SetFecControllerOverride(rtp_video_sender_); |
| |
| ReconfigureVideoEncoder(std::move(encoder_config)); |
| } |
| |
| VideoSendStream::~VideoSendStream() { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| RTC_DCHECK(!running_); |
| transport_->DestroyRtpVideoSender(rtp_video_sender_); |
| } |
| |
| void VideoSendStream::UpdateActiveSimulcastLayers( |
| const std::vector<bool> active_layers) { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| |
| // Keep our `running_` flag expected state in sync with active layers since |
| // the `send_stream_` will be implicitly stopped/started depending on the |
| // state of the layers. |
| bool running = false; |
| |
| rtc::StringBuilder active_layers_string; |
| active_layers_string << "{"; |
| for (size_t i = 0; i < active_layers.size(); ++i) { |
| if (active_layers[i]) { |
| running = true; |
| active_layers_string << "1"; |
| } else { |
| active_layers_string << "0"; |
| } |
| if (i < active_layers.size() - 1) { |
| active_layers_string << ", "; |
| } |
| } |
| active_layers_string << "}"; |
| RTC_LOG(LS_INFO) << "UpdateActiveSimulcastLayers: " |
| << active_layers_string.str(); |
| |
| rtp_transport_queue_->PostTask( |
| ToQueuedTask(transport_queue_safety_, [this, active_layers] { |
| send_stream_.UpdateActiveSimulcastLayers(active_layers); |
| })); |
| |
| running_ = running; |
| } |
| |
| void VideoSendStream::Start() { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| RTC_DLOG(LS_INFO) << "VideoSendStream::Start"; |
| if (running_) |
| return; |
| |
| running_ = true; |
| |
| rtp_transport_queue_->PostTask(ToQueuedTask([this] { |
| transport_queue_safety_->SetAlive(); |
| 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. |
| // TODO(tommi): ^^^ Validate if this still holds. |
| thread_sync_event_.Wait(rtc::Event::kForever); |
| } |
| |
| void VideoSendStream::Stop() { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| if (!running_) |
| return; |
| RTC_DLOG(LS_INFO) << "VideoSendStream::Stop"; |
| running_ = false; |
| rtp_transport_queue_->PostTask(ToQueuedTask(transport_queue_safety_, [this] { |
| transport_queue_safety_->SetNotAlive(); |
| send_stream_.Stop(); |
| })); |
| } |
| |
| bool VideoSendStream::started() { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| return running_; |
| } |
| |
| void VideoSendStream::AddAdaptationResource( |
| rtc::scoped_refptr<Resource> resource) { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| video_stream_encoder_->AddAdaptationResource(resource); |
| } |
| |
| std::vector<rtc::scoped_refptr<Resource>> |
| VideoSendStream::GetAdaptationResources() { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| return video_stream_encoder_->GetAdaptationResources(); |
| } |
| |
| 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) { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| RTC_DCHECK_EQ(content_type_, config.content_type); |
| video_stream_encoder_->ConfigureEncoder( |
| std::move(config), |
| config_.rtp.max_packet_size - CalculateMaxHeaderSize(config_.rtp)); |
| } |
| |
| 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(); |
| } |
| |
| absl::optional<float> VideoSendStream::GetPacingFactorOverride() const { |
| return send_stream_.configured_pacing_factor(); |
| } |
| |
| void VideoSendStream::StopPermanentlyAndGetRtpStates( |
| VideoSendStream::RtpStateMap* rtp_state_map, |
| VideoSendStream::RtpPayloadStateMap* payload_state_map) { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| video_stream_encoder_->Stop(); |
| |
| running_ = false; |
| // Always run these cleanup steps regardless of whether running_ was set |
| // or not. This will unregister callbacks before destruction. |
| // See `VideoSendStreamImpl::StopVideoSendStream` for more. |
| rtp_transport_queue_->PostTask([this, rtp_state_map, payload_state_map]() { |
| transport_queue_safety_->SetNotAlive(); |
| send_stream_.Stop(); |
| *rtp_state_map = send_stream_.GetRtpStates(); |
| *payload_state_map = send_stream_.GetRtpPayloadStates(); |
| thread_sync_event_.Set(); |
| }); |
| thread_sync_event_.Wait(rtc::Event::kForever); |
| } |
| |
| void VideoSendStream::DeliverRtcp(const uint8_t* packet, size_t length) { |
| // Called on a network thread. |
| send_stream_.DeliverRtcp(packet, length); |
| } |
| |
| } // namespace internal |
| } // namespace webrtc |
