| /* |
| * 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 "webrtc/video/video_send_stream.h" |
| |
| #include <algorithm> |
| #include <sstream> |
| #include <string> |
| #include <vector> |
| |
| #include "webrtc/base/checks.h" |
| #include "webrtc/base/logging.h" |
| #include "webrtc/base/trace_event.h" |
| #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h" |
| #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h" |
| #include "webrtc/modules/congestion_controller/include/congestion_controller.h" |
| #include "webrtc/modules/pacing/packet_router.h" |
| #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" |
| #include "webrtc/modules/utility/include/process_thread.h" |
| #include "webrtc/modules/video_coding/utility/ivf_file_writer.h" |
| #include "webrtc/video/call_stats.h" |
| #include "webrtc/video/video_capture_input.h" |
| #include "webrtc/video/vie_remb.h" |
| #include "webrtc/video_send_stream.h" |
| |
| namespace webrtc { |
| |
| class RtcpIntraFrameObserver; |
| class TransportFeedbackObserver; |
| |
| static const int kMinSendSidePacketHistorySize = 600; |
| |
| namespace { |
| |
| std::vector<RtpRtcp*> CreateRtpRtcpModules( |
| Transport* outgoing_transport, |
| RtcpIntraFrameObserver* intra_frame_callback, |
| RtcpBandwidthObserver* bandwidth_callback, |
| TransportFeedbackObserver* transport_feedback_callback, |
| RtcpRttStats* rtt_stats, |
| RtpPacketSender* paced_sender, |
| TransportSequenceNumberAllocator* transport_sequence_number_allocator, |
| SendStatisticsProxy* stats_proxy, |
| SendDelayStats* send_delay_stats, |
| size_t num_modules) { |
| RTC_DCHECK_GT(num_modules, 0u); |
| RtpRtcp::Configuration configuration; |
| ReceiveStatistics* null_receive_statistics = configuration.receive_statistics; |
| configuration.audio = false; |
| configuration.receiver_only = false; |
| configuration.receive_statistics = null_receive_statistics; |
| configuration.outgoing_transport = outgoing_transport; |
| configuration.intra_frame_callback = intra_frame_callback; |
| configuration.rtt_stats = rtt_stats; |
| configuration.rtcp_packet_type_counter_observer = stats_proxy; |
| configuration.paced_sender = paced_sender; |
| configuration.transport_sequence_number_allocator = |
| transport_sequence_number_allocator; |
| 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.bandwidth_callback = bandwidth_callback; |
| configuration.transport_feedback_callback = transport_feedback_callback; |
| |
| 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; |
| } |
| |
| } // namespace |
| |
| std::string |
| VideoSendStream::Config::EncoderSettings::ToString() const { |
| std::stringstream ss; |
| ss << "{payload_name: " << payload_name; |
| ss << ", payload_type: " << payload_type; |
| ss << ", encoder: " << (encoder ? "(VideoEncoder)" : "nullptr"); |
| ss << '}'; |
| return ss.str(); |
| } |
| |
| std::string VideoSendStream::Config::Rtp::Rtx::ToString() |
| const { |
| std::stringstream ss; |
| ss << "{ssrcs: ["; |
| for (size_t i = 0; i < ssrcs.size(); ++i) { |
| ss << ssrcs[i]; |
| if (i != ssrcs.size() - 1) |
| ss << ", "; |
| } |
| ss << ']'; |
| |
| ss << ", payload_type: " << payload_type; |
| ss << '}'; |
| return ss.str(); |
| } |
| |
| std::string VideoSendStream::Config::Rtp::ToString() const { |
| std::stringstream ss; |
| ss << "{ssrcs: ["; |
| for (size_t i = 0; i < ssrcs.size(); ++i) { |
| ss << ssrcs[i]; |
| if (i != ssrcs.size() - 1) |
| ss << ", "; |
| } |
| ss << ']'; |
| ss << ", rtcp_mode: " |
| << (rtcp_mode == RtcpMode::kCompound ? "RtcpMode::kCompound" |
| : "RtcpMode::kReducedSize"); |
| ss << ", max_packet_size: " << max_packet_size; |
| ss << ", extensions: ["; |
| for (size_t i = 0; i < extensions.size(); ++i) { |
| ss << extensions[i].ToString(); |
| if (i != extensions.size() - 1) |
| ss << ", "; |
| } |
| ss << ']'; |
| |
| ss << ", nack: {rtp_history_ms: " << nack.rtp_history_ms << '}'; |
| ss << ", fec: " << fec.ToString(); |
| ss << ", rtx: " << rtx.ToString(); |
| ss << ", c_name: " << c_name; |
| ss << '}'; |
| return ss.str(); |
| } |
| |
| std::string VideoSendStream::Config::ToString() const { |
| std::stringstream ss; |
| ss << "{encoder_settings: " << encoder_settings.ToString(); |
| ss << ", rtp: " << rtp.ToString(); |
| ss << ", pre_encode_callback: " |
| << (pre_encode_callback ? "(I420FrameCallback)" : "nullptr"); |
| ss << ", post_encode_callback: " |
| << (post_encode_callback ? "(EncodedFrameObserver)" : "nullptr"); |
| ss << ", local_renderer: " |
| << (local_renderer ? "(VideoRenderer)" : "nullptr"); |
| ss << ", render_delay_ms: " << render_delay_ms; |
| ss << ", target_delay_ms: " << target_delay_ms; |
| ss << ", suspend_below_min_bitrate: " << (suspend_below_min_bitrate ? "on" |
| : "off"); |
| ss << '}'; |
| return ss.str(); |
| } |
| |
| namespace { |
| |
| VideoCodecType PayloadNameToCodecType(const std::string& payload_name) { |
| if (payload_name == "VP8") |
| return kVideoCodecVP8; |
| if (payload_name == "VP9") |
| return kVideoCodecVP9; |
| if (payload_name == "H264") |
| return kVideoCodecH264; |
| return kVideoCodecGeneric; |
| } |
| |
| bool PayloadTypeSupportsSkippingFecPackets(const std::string& payload_name) { |
| switch (PayloadNameToCodecType(payload_name)) { |
| case kVideoCodecVP8: |
| case kVideoCodecVP9: |
| return true; |
| case kVideoCodecH264: |
| case kVideoCodecGeneric: |
| return false; |
| case kVideoCodecI420: |
| case kVideoCodecRED: |
| case kVideoCodecULPFEC: |
| case kVideoCodecUnknown: |
| RTC_NOTREACHED(); |
| return false; |
| } |
| RTC_NOTREACHED(); |
| return false; |
| } |
| |
| // TODO(pbos): Lower these thresholds (to closer to 100%) when we handle |
| // pipelining encoders better (multiple input frames before something comes |
| // out). This should effectively turn off CPU adaptations for systems that |
| // remotely cope with the load right now. |
| CpuOveruseOptions GetCpuOveruseOptions(bool full_overuse_time) { |
| CpuOveruseOptions options; |
| if (full_overuse_time) { |
| options.low_encode_usage_threshold_percent = 150; |
| options.high_encode_usage_threshold_percent = 200; |
| } |
| return options; |
| } |
| |
| VideoCodec VideoEncoderConfigToVideoCodec(const VideoEncoderConfig& config, |
| const std::string& payload_name, |
| int payload_type) { |
| const std::vector<VideoStream>& streams = config.streams; |
| static const int kEncoderMinBitrateKbps = 30; |
| RTC_DCHECK(!streams.empty()); |
| RTC_DCHECK_GE(config.min_transmit_bitrate_bps, 0); |
| |
| VideoCodec video_codec; |
| memset(&video_codec, 0, sizeof(video_codec)); |
| video_codec.codecType = PayloadNameToCodecType(payload_name); |
| |
| switch (config.content_type) { |
| case VideoEncoderConfig::ContentType::kRealtimeVideo: |
| video_codec.mode = kRealtimeVideo; |
| break; |
| case VideoEncoderConfig::ContentType::kScreen: |
| video_codec.mode = kScreensharing; |
| if (config.streams.size() == 1 && |
| config.streams[0].temporal_layer_thresholds_bps.size() == 1) { |
| video_codec.targetBitrate = |
| config.streams[0].temporal_layer_thresholds_bps[0] / 1000; |
| } |
| break; |
| } |
| |
| switch (video_codec.codecType) { |
| case kVideoCodecVP8: { |
| if (config.encoder_specific_settings) { |
| video_codec.codecSpecific.VP8 = *reinterpret_cast<const VideoCodecVP8*>( |
| config.encoder_specific_settings); |
| } else { |
| video_codec.codecSpecific.VP8 = VideoEncoder::GetDefaultVp8Settings(); |
| } |
| video_codec.codecSpecific.VP8.numberOfTemporalLayers = |
| static_cast<unsigned char>( |
| streams.back().temporal_layer_thresholds_bps.size() + 1); |
| break; |
| } |
| case kVideoCodecVP9: { |
| if (config.encoder_specific_settings) { |
| video_codec.codecSpecific.VP9 = *reinterpret_cast<const VideoCodecVP9*>( |
| config.encoder_specific_settings); |
| if (video_codec.mode == kScreensharing) { |
| video_codec.codecSpecific.VP9.flexibleMode = true; |
| // For now VP9 screensharing use 1 temporal and 2 spatial layers. |
| RTC_DCHECK_EQ(video_codec.codecSpecific.VP9.numberOfTemporalLayers, |
| 1); |
| RTC_DCHECK_EQ(video_codec.codecSpecific.VP9.numberOfSpatialLayers, 2); |
| } |
| } else { |
| video_codec.codecSpecific.VP9 = VideoEncoder::GetDefaultVp9Settings(); |
| } |
| video_codec.codecSpecific.VP9.numberOfTemporalLayers = |
| static_cast<unsigned char>( |
| streams.back().temporal_layer_thresholds_bps.size() + 1); |
| break; |
| } |
| case kVideoCodecH264: { |
| if (config.encoder_specific_settings) { |
| video_codec.codecSpecific.H264 = |
| *reinterpret_cast<const VideoCodecH264*>( |
| config.encoder_specific_settings); |
| } else { |
| video_codec.codecSpecific.H264 = VideoEncoder::GetDefaultH264Settings(); |
| } |
| break; |
| } |
| default: |
| // TODO(pbos): Support encoder_settings codec-agnostically. |
| RTC_DCHECK(!config.encoder_specific_settings) |
| << "Encoder-specific settings for codec type not wired up."; |
| break; |
| } |
| |
| strncpy(video_codec.plName, payload_name.c_str(), kPayloadNameSize - 1); |
| video_codec.plName[kPayloadNameSize - 1] = '\0'; |
| video_codec.plType = payload_type; |
| video_codec.numberOfSimulcastStreams = |
| static_cast<unsigned char>(streams.size()); |
| video_codec.minBitrate = streams[0].min_bitrate_bps / 1000; |
| if (video_codec.minBitrate < kEncoderMinBitrateKbps) |
| video_codec.minBitrate = kEncoderMinBitrateKbps; |
| RTC_DCHECK_LE(streams.size(), static_cast<size_t>(kMaxSimulcastStreams)); |
| if (video_codec.codecType == kVideoCodecVP9) { |
| // If the vector is empty, bitrates will be configured automatically. |
| RTC_DCHECK(config.spatial_layers.empty() || |
| config.spatial_layers.size() == |
| video_codec.codecSpecific.VP9.numberOfSpatialLayers); |
| RTC_DCHECK_LE(video_codec.codecSpecific.VP9.numberOfSpatialLayers, |
| kMaxSimulcastStreams); |
| for (size_t i = 0; i < config.spatial_layers.size(); ++i) |
| video_codec.spatialLayers[i] = config.spatial_layers[i]; |
| } |
| for (size_t i = 0; i < streams.size(); ++i) { |
| SimulcastStream* sim_stream = &video_codec.simulcastStream[i]; |
| RTC_DCHECK_GT(streams[i].width, 0u); |
| RTC_DCHECK_GT(streams[i].height, 0u); |
| RTC_DCHECK_GT(streams[i].max_framerate, 0); |
| // Different framerates not supported per stream at the moment. |
| RTC_DCHECK_EQ(streams[i].max_framerate, streams[0].max_framerate); |
| RTC_DCHECK_GE(streams[i].min_bitrate_bps, 0); |
| RTC_DCHECK_GE(streams[i].target_bitrate_bps, streams[i].min_bitrate_bps); |
| RTC_DCHECK_GE(streams[i].max_bitrate_bps, streams[i].target_bitrate_bps); |
| RTC_DCHECK_GE(streams[i].max_qp, 0); |
| |
| sim_stream->width = static_cast<uint16_t>(streams[i].width); |
| sim_stream->height = static_cast<uint16_t>(streams[i].height); |
| sim_stream->minBitrate = streams[i].min_bitrate_bps / 1000; |
| sim_stream->targetBitrate = streams[i].target_bitrate_bps / 1000; |
| sim_stream->maxBitrate = streams[i].max_bitrate_bps / 1000; |
| sim_stream->qpMax = streams[i].max_qp; |
| sim_stream->numberOfTemporalLayers = static_cast<unsigned char>( |
| streams[i].temporal_layer_thresholds_bps.size() + 1); |
| |
| video_codec.width = std::max(video_codec.width, |
| static_cast<uint16_t>(streams[i].width)); |
| video_codec.height = std::max( |
| video_codec.height, static_cast<uint16_t>(streams[i].height)); |
| video_codec.minBitrate = |
| std::min(static_cast<uint16_t>(video_codec.minBitrate), |
| static_cast<uint16_t>(streams[i].min_bitrate_bps / 1000)); |
| video_codec.maxBitrate += streams[i].max_bitrate_bps / 1000; |
| video_codec.qpMax = std::max(video_codec.qpMax, |
| static_cast<unsigned int>(streams[i].max_qp)); |
| } |
| |
| if (video_codec.maxBitrate == 0) { |
| // Unset max bitrate -> cap to one bit per pixel. |
| video_codec.maxBitrate = |
| (video_codec.width * video_codec.height * video_codec.maxFramerate) / |
| 1000; |
| } |
| if (video_codec.maxBitrate < kEncoderMinBitrateKbps) |
| video_codec.maxBitrate = kEncoderMinBitrateKbps; |
| |
| RTC_DCHECK_GT(streams[0].max_framerate, 0); |
| video_codec.maxFramerate = streams[0].max_framerate; |
| |
| return video_codec; |
| } |
| |
| } // namespace |
| |
| namespace internal { |
| VideoSendStream::VideoSendStream( |
| int num_cpu_cores, |
| ProcessThread* module_process_thread, |
| CallStats* call_stats, |
| CongestionController* congestion_controller, |
| BitrateAllocator* bitrate_allocator, |
| SendDelayStats* send_delay_stats, |
| VieRemb* remb, |
| const VideoSendStream::Config& config, |
| const VideoEncoderConfig& encoder_config, |
| const std::map<uint32_t, RtpState>& suspended_ssrcs) |
| : stats_proxy_(Clock::GetRealTimeClock(), |
| config, |
| encoder_config.content_type), |
| encoded_frame_proxy_(config.post_encode_callback), |
| config_(config), |
| suspended_ssrcs_(suspended_ssrcs), |
| module_process_thread_(module_process_thread), |
| call_stats_(call_stats), |
| congestion_controller_(congestion_controller), |
| bitrate_allocator_(bitrate_allocator), |
| remb_(remb), |
| encoder_thread_(EncoderThreadFunction, this, "EncoderThread"), |
| encoder_wakeup_event_(false, false), |
| stop_encoder_thread_(0), |
| overuse_detector_( |
| Clock::GetRealTimeClock(), |
| GetCpuOveruseOptions(config.encoder_settings.full_overuse_time), |
| this, |
| config.post_encode_callback, |
| &stats_proxy_), |
| vie_encoder_(num_cpu_cores, |
| module_process_thread_, |
| &stats_proxy_, |
| &overuse_detector_), |
| encoder_feedback_(Clock::GetRealTimeClock(), |
| config.rtp.ssrcs, |
| &vie_encoder_), |
| video_sender_(vie_encoder_.video_sender()), |
| bandwidth_observer_(congestion_controller_->GetBitrateController() |
| ->CreateRtcpBandwidthObserver()), |
| rtp_rtcp_modules_(CreateRtpRtcpModules( |
| config.send_transport, |
| &encoder_feedback_, |
| bandwidth_observer_.get(), |
| congestion_controller_->GetTransportFeedbackObserver(), |
| call_stats_->rtcp_rtt_stats(), |
| congestion_controller_->pacer(), |
| congestion_controller_->packet_router(), |
| &stats_proxy_, |
| send_delay_stats, |
| config_.rtp.ssrcs.size())), |
| payload_router_(rtp_rtcp_modules_, config.encoder_settings.payload_type), |
| input_(&encoder_wakeup_event_, |
| config_.local_renderer, |
| &stats_proxy_, |
| &overuse_detector_) { |
| LOG(LS_INFO) << "VideoSendStream: " << config_.ToString(); |
| |
| RTC_DCHECK(!config_.rtp.ssrcs.empty()); |
| RTC_DCHECK(module_process_thread_); |
| RTC_DCHECK(call_stats_); |
| RTC_DCHECK(congestion_controller_); |
| RTC_DCHECK(remb_); |
| |
| |
| // RTP/RTCP initialization. |
| for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { |
| module_process_thread_->RegisterModule(rtp_rtcp); |
| congestion_controller_->packet_router()->AddRtpModule(rtp_rtcp); |
| } |
| |
| video_sender_->RegisterProtectionCallback(this); |
| |
| for (size_t i = 0; i < config_.rtp.extensions.size(); ++i) { |
| const std::string& extension = config_.rtp.extensions[i].name; |
| 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)); |
| } |
| } |
| |
| remb_->AddRembSender(rtp_rtcp_modules_[0]); |
| rtp_rtcp_modules_[0]->SetREMBStatus(true); |
| |
| ConfigureProtection(); |
| ConfigureSsrcs(); |
| |
| // TODO(pbos): Should we set CNAME on all RTP modules? |
| rtp_rtcp_modules_.front()->SetCNAME(config_.rtp.c_name.c_str()); |
| // 28 to match packet overhead in ModuleRtpRtcpImpl. |
| static const size_t kRtpPacketSizeOverhead = 28; |
| RTC_DCHECK_LE(config_.rtp.max_packet_size, 0xFFFFu + kRtpPacketSizeOverhead); |
| const uint16_t mtu = static_cast<uint16_t>(config_.rtp.max_packet_size + |
| kRtpPacketSizeOverhead); |
| for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { |
| rtp_rtcp->RegisterRtcpStatisticsCallback(&stats_proxy_); |
| rtp_rtcp->RegisterSendChannelRtpStatisticsCallback(&stats_proxy_); |
| rtp_rtcp->SetMaxTransferUnit(mtu); |
| 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); |
| ReconfigureVideoEncoder(encoder_config); |
| |
| module_process_thread_->RegisterModule(&overuse_detector_); |
| |
| encoder_thread_.Start(); |
| encoder_thread_.SetPriority(rtc::kHighPriority); |
| } |
| |
| VideoSendStream::~VideoSendStream() { |
| LOG(LS_INFO) << "~VideoSendStream: " << config_.ToString(); |
| |
| Stop(); |
| |
| // Stop the encoder thread permanently. |
| rtc::AtomicOps::ReleaseStore(&stop_encoder_thread_, 1); |
| encoder_wakeup_event_.Set(); |
| encoder_thread_.Stop(); |
| |
| // This needs to happen after stopping the encoder thread, |
| // since the encoder thread calls AddObserver. |
| bitrate_allocator_->RemoveObserver(this); |
| |
| module_process_thread_->DeRegisterModule(&overuse_detector_); |
| |
| rtp_rtcp_modules_[0]->SetREMBStatus(false); |
| remb_->RemoveRembSender(rtp_rtcp_modules_[0]); |
| |
| for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { |
| congestion_controller_->packet_router()->RemoveRtpModule(rtp_rtcp); |
| module_process_thread_->DeRegisterModule(rtp_rtcp); |
| delete rtp_rtcp; |
| } |
| } |
| |
| void VideoSendStream::SignalNetworkState(NetworkState state) { |
| // When network goes up, enable RTCP status before setting transmission state. |
| // When it goes down, disable RTCP afterwards. This ensures that any packets |
| // sent due to the network state changed will not be dropped. |
| if (state == kNetworkUp) { |
| for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) |
| rtp_rtcp->SetRTCPStatus(config_.rtp.rtcp_mode); |
| } |
| vie_encoder_.SetNetworkTransmissionState(state == kNetworkUp); |
| if (state == kNetworkDown) { |
| for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) |
| rtp_rtcp->SetRTCPStatus(RtcpMode::kOff); |
| } |
| } |
| |
| bool VideoSendStream::DeliverRtcp(const uint8_t* packet, size_t length) { |
| for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) |
| rtp_rtcp->IncomingRtcpPacket(packet, length); |
| return true; |
| } |
| |
| void VideoSendStream::Start() { |
| if (payload_router_.active()) |
| return; |
| TRACE_EVENT_INSTANT0("webrtc", "VideoSendStream::Start"); |
| payload_router_.set_active(true); |
| // Was not already started, trigger a keyframe. |
| vie_encoder_.SendKeyFrame(); |
| vie_encoder_.Start(); |
| } |
| |
| void VideoSendStream::Stop() { |
| if (!payload_router_.active()) |
| return; |
| TRACE_EVENT_INSTANT0("webrtc", "VideoSendStream::Stop"); |
| vie_encoder_.Pause(); |
| payload_router_.set_active(false); |
| } |
| |
| VideoCaptureInput* VideoSendStream::Input() { |
| return &input_; |
| } |
| |
| bool VideoSendStream::EncoderThreadFunction(void* obj) { |
| static_cast<VideoSendStream*>(obj)->EncoderProcess(); |
| // We're done, return false to abort. |
| return false; |
| } |
| |
| void VideoSendStream::EncoderProcess() { |
| RTC_CHECK_EQ(0, vie_encoder_.RegisterExternalEncoder( |
| config_.encoder_settings.encoder, |
| config_.encoder_settings.payload_type, |
| config_.encoder_settings.internal_source)); |
| |
| while (true) { |
| encoder_wakeup_event_.Wait(rtc::Event::kForever); |
| if (rtc::AtomicOps::AcquireLoad(&stop_encoder_thread_)) |
| break; |
| rtc::Optional<EncoderSettings> encoder_settings; |
| { |
| rtc::CritScope lock(&encoder_settings_crit_); |
| if (pending_encoder_settings_) { |
| encoder_settings = pending_encoder_settings_; |
| pending_encoder_settings_ = rtc::Optional<EncoderSettings>(); |
| } |
| } |
| if (encoder_settings) { |
| encoder_settings->video_codec.startBitrate = |
| bitrate_allocator_->AddObserver( |
| this, encoder_settings->video_codec.minBitrate * 1000, |
| encoder_settings->video_codec.maxBitrate * 1000, |
| !config_.suspend_below_min_bitrate) / |
| 1000; |
| |
| payload_router_.SetSendStreams(encoder_settings->streams); |
| vie_encoder_.SetEncoder(encoder_settings->video_codec, |
| encoder_settings->min_transmit_bitrate_bps, |
| payload_router_.MaxPayloadLength(), this); |
| |
| // vie_encoder_.SetEncoder must be called before this. |
| if (config_.suspend_below_min_bitrate) |
| video_sender_->SuspendBelowMinBitrate(); |
| |
| // Clear stats for disabled layers. |
| for (size_t i = encoder_settings->streams.size(); |
| i < config_.rtp.ssrcs.size(); ++i) { |
| stats_proxy_.OnInactiveSsrc(config_.rtp.ssrcs[i]); |
| } |
| |
| // We might've gotten new settings while configuring the encoder settings, |
| // restart from the top to see if that's the case before trying to encode |
| // a frame (which might correspond to the last frame size). |
| encoder_wakeup_event_.Set(); |
| continue; |
| } |
| |
| VideoFrame frame; |
| if (input_.GetVideoFrame(&frame)) { |
| // TODO(perkj): |pre_encode_callback| is only used by tests. Tests should |
| // register as a sink to the VideoSource instead. |
| if (config_.pre_encode_callback) { |
| config_.pre_encode_callback->OnFrame(frame); |
| } |
| vie_encoder_.EncodeVideoFrame(frame); |
| } |
| } |
| vie_encoder_.DeRegisterExternalEncoder(config_.encoder_settings.payload_type); |
| } |
| |
| void VideoSendStream::ReconfigureVideoEncoder( |
| const VideoEncoderConfig& config) { |
| TRACE_EVENT0("webrtc", "VideoSendStream::(Re)configureVideoEncoder"); |
| LOG(LS_INFO) << "(Re)configureVideoEncoder: " << config.ToString(); |
| RTC_DCHECK_GE(config_.rtp.ssrcs.size(), config.streams.size()); |
| VideoCodec video_codec = VideoEncoderConfigToVideoCodec( |
| config, config_.encoder_settings.payload_name, |
| config_.encoder_settings.payload_type); |
| { |
| rtc::CritScope lock(&encoder_settings_crit_); |
| pending_encoder_settings_ = rtc::Optional<EncoderSettings>( |
| {video_codec, config.min_transmit_bitrate_bps, config.streams}); |
| } |
| encoder_wakeup_event_.Set(); |
| } |
| |
| VideoSendStream::Stats VideoSendStream::GetStats() { |
| return stats_proxy_.GetStats(); |
| } |
| |
| void VideoSendStream::OveruseDetected() { |
| if (config_.overuse_callback) |
| config_.overuse_callback->OnLoadUpdate(LoadObserver::kOveruse); |
| } |
| |
| void VideoSendStream::NormalUsage() { |
| if (config_.overuse_callback) |
| config_.overuse_callback->OnLoadUpdate(LoadObserver::kUnderuse); |
| } |
| |
| int32_t VideoSendStream::Encoded(const EncodedImage& encoded_image, |
| const CodecSpecificInfo* codec_specific_info, |
| const RTPFragmentationHeader* fragmentation) { |
| // |encoded_frame_proxy_| forwards frames to |config_.post_encode_callback|; |
| encoded_frame_proxy_.Encoded(encoded_image, codec_specific_info, |
| fragmentation); |
| int32_t return_value = payload_router_.Encoded( |
| encoded_image, codec_specific_info, fragmentation); |
| |
| if (kEnableFrameRecording) { |
| int layer = codec_specific_info->codecType == kVideoCodecVP8 |
| ? codec_specific_info->codecSpecific.VP8.simulcastIdx |
| : 0; |
| IvfFileWriter* file_writer; |
| { |
| if (file_writers_[layer] == nullptr) { |
| std::ostringstream oss; |
| oss << "send_bitstream_ssrc"; |
| for (uint32_t ssrc : config_.rtp.ssrcs) |
| oss << "_" << ssrc; |
| oss << "_layer" << layer << ".ivf"; |
| file_writers_[layer] = |
| IvfFileWriter::Open(oss.str(), codec_specific_info->codecType); |
| } |
| file_writer = file_writers_[layer].get(); |
| } |
| if (file_writer) { |
| bool ok = file_writer->WriteFrame(encoded_image); |
| RTC_DCHECK(ok); |
| } |
| } |
| |
| return return_value; |
| } |
| |
| void VideoSendStream::ConfigureProtection() { |
| // Enable NACK, FEC or both. |
| const bool enable_protection_nack = config_.rtp.nack.rtp_history_ms > 0; |
| bool enable_protection_fec = config_.rtp.fec.red_payload_type != -1; |
| // Payload types without picture ID cannot determine that a stream is complete |
| // without retransmitting FEC, so using FEC + 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 (enable_protection_nack && |
| !PayloadTypeSupportsSkippingFecPackets( |
| config_.encoder_settings.payload_name)) { |
| LOG(LS_WARNING) << "Transmitting payload type without picture ID using" |
| "NACK+FEC is a waste of bandwidth since FEC packets " |
| "also have to be retransmitted. Disabling FEC."; |
| enable_protection_fec = false; |
| } |
| |
| // Set to valid uint8_ts to be castable later without signed overflows. |
| uint8_t payload_type_red = 0; |
| uint8_t payload_type_fec = 0; |
| // TODO(changbin): Should set RTX for RED mapping in RTP sender in future. |
| // Validate payload types. If either RED or FEC payload types are set then |
| // both should be. If FEC is enabled then they both have to be set. |
| if (enable_protection_fec || config_.rtp.fec.red_payload_type != -1 || |
| config_.rtp.fec.ulpfec_payload_type != -1) { |
| RTC_DCHECK_GE(config_.rtp.fec.red_payload_type, 0); |
| RTC_DCHECK_GE(config_.rtp.fec.ulpfec_payload_type, 0); |
| RTC_DCHECK_LE(config_.rtp.fec.red_payload_type, 127); |
| RTC_DCHECK_LE(config_.rtp.fec.ulpfec_payload_type, 127); |
| payload_type_red = static_cast<uint8_t>(config_.rtp.fec.red_payload_type); |
| payload_type_fec = |
| static_cast<uint8_t>(config_.rtp.fec.ulpfec_payload_type); |
| } else { |
| // Payload types unset. |
| RTC_DCHECK_EQ(config_.rtp.fec.red_payload_type, -1); |
| RTC_DCHECK_EQ(config_.rtp.fec.ulpfec_payload_type, -1); |
| } |
| |
| for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { |
| // Set NACK. |
| rtp_rtcp->SetStorePacketsStatus( |
| enable_protection_nack || congestion_controller_->pacer(), |
| kMinSendSidePacketHistorySize); |
| // Set FEC. |
| for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { |
| rtp_rtcp->SetGenericFECStatus(enable_protection_fec, payload_type_red, |
| payload_type_fec); |
| } |
| } |
| |
| vie_encoder_.SetProtectionMethod(enable_protection_nack, |
| enable_protection_fec); |
| } |
| |
| void VideoSendStream::ConfigureSsrcs() { |
| // 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. |
| 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); |
| 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.fec.red_payload_type != -1 && |
| config_.rtp.fec.red_rtx_payload_type != -1) { |
| for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { |
| rtp_rtcp->SetRtxSendPayloadType(config_.rtp.fec.red_rtx_payload_type, |
| config_.rtp.fec.red_payload_type); |
| } |
| } |
| } |
| |
| std::map<uint32_t, RtpState> VideoSendStream::GetRtpStates() const { |
| 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(); |
| } |
| |
| return rtp_states; |
| } |
| |
| int VideoSendStream::GetPaddingNeededBps() const { |
| return vie_encoder_.GetPaddingNeededBps(); |
| } |
| |
| void VideoSendStream::OnBitrateUpdated(uint32_t bitrate_bps, |
| uint8_t fraction_loss, |
| int64_t rtt) { |
| payload_router_.SetTargetSendBitrate(bitrate_bps); |
| vie_encoder_.OnBitrateUpdated(bitrate_bps, fraction_loss, rtt); |
| } |
| |
| int VideoSendStream::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 (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; |
| } |
| |
| } // namespace internal |
| } // namespace webrtc |
