| /* |
| * Copyright (c) 2012 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/vie_channel.h" |
| |
| #include <algorithm> |
| #include <map> |
| #include <vector> |
| |
| #include "webrtc/base/checks.h" |
| #include "webrtc/base/logging.h" |
| #include "webrtc/base/platform_thread.h" |
| #include "webrtc/common_video/include/incoming_video_stream.h" |
| #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h" |
| #include "webrtc/frame_callback.h" |
| #include "webrtc/modules/pacing/paced_sender.h" |
| #include "webrtc/modules/pacing/packet_router.h" |
| #include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h" |
| #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" |
| #include "webrtc/modules/utility/include/process_thread.h" |
| #include "webrtc/modules/video_coding/include/video_coding.h" |
| #include "webrtc/modules/video_processing/include/video_processing.h" |
| #include "webrtc/modules/video_render/video_render_defines.h" |
| #include "webrtc/system_wrappers/include/metrics.h" |
| #include "webrtc/video/call_stats.h" |
| #include "webrtc/video/payload_router.h" |
| #include "webrtc/video/receive_statistics_proxy.h" |
| |
| namespace webrtc { |
| |
| static const int kMinSendSidePacketHistorySize = 600; |
| static const int kMaxPacketAgeToNack = 450; |
| static const int kMaxNackListSize = 250; |
| |
| // Helper class receiving statistics callbacks. |
| class ChannelStatsObserver : public CallStatsObserver { |
| public: |
| explicit ChannelStatsObserver(ViEChannel* owner) : owner_(owner) {} |
| virtual ~ChannelStatsObserver() {} |
| |
| // Implements StatsObserver. |
| virtual void OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) { |
| owner_->OnRttUpdate(avg_rtt_ms, max_rtt_ms); |
| } |
| |
| private: |
| ViEChannel* const owner_; |
| }; |
| |
| class ViEChannelProtectionCallback : public VCMProtectionCallback { |
| public: |
| explicit ViEChannelProtectionCallback(ViEChannel* owner) : owner_(owner) {} |
| ~ViEChannelProtectionCallback() {} |
| |
| |
| int ProtectionRequest( |
| const FecProtectionParams* delta_fec_params, |
| const FecProtectionParams* key_fec_params, |
| uint32_t* sent_video_rate_bps, |
| uint32_t* sent_nack_rate_bps, |
| uint32_t* sent_fec_rate_bps) override { |
| return owner_->ProtectionRequest(delta_fec_params, key_fec_params, |
| sent_video_rate_bps, sent_nack_rate_bps, |
| sent_fec_rate_bps); |
| } |
| private: |
| ViEChannel* owner_; |
| }; |
| |
| ViEChannel::ViEChannel(Transport* transport, |
| ProcessThread* module_process_thread, |
| PayloadRouter* send_payload_router, |
| VideoCodingModule* vcm, |
| RtcpIntraFrameObserver* intra_frame_observer, |
| RtcpBandwidthObserver* bandwidth_observer, |
| TransportFeedbackObserver* transport_feedback_observer, |
| RemoteBitrateEstimator* remote_bitrate_estimator, |
| RtcpRttStats* rtt_stats, |
| PacedSender* paced_sender, |
| PacketRouter* packet_router, |
| size_t max_rtp_streams, |
| bool sender) |
| : sender_(sender), |
| module_process_thread_(module_process_thread), |
| send_payload_router_(send_payload_router), |
| vcm_protection_callback_(new ViEChannelProtectionCallback(this)), |
| vcm_(vcm), |
| vie_receiver_(vcm_, remote_bitrate_estimator, this), |
| stats_observer_(new ChannelStatsObserver(this)), |
| receive_stats_callback_(nullptr), |
| incoming_video_stream_(nullptr), |
| intra_frame_observer_(intra_frame_observer), |
| rtt_stats_(rtt_stats), |
| paced_sender_(paced_sender), |
| packet_router_(packet_router), |
| bandwidth_observer_(bandwidth_observer), |
| transport_feedback_observer_(transport_feedback_observer), |
| max_nack_reordering_threshold_(kMaxPacketAgeToNack), |
| pre_render_callback_(NULL), |
| last_rtt_ms_(0), |
| rtp_rtcp_modules_( |
| CreateRtpRtcpModules(!sender, |
| vie_receiver_.GetReceiveStatistics(), |
| transport, |
| intra_frame_observer_, |
| bandwidth_observer_.get(), |
| transport_feedback_observer_, |
| rtt_stats_, |
| &rtcp_packet_type_counter_observer_, |
| remote_bitrate_estimator, |
| paced_sender_, |
| packet_router_, |
| &send_bitrate_observer_, |
| &send_frame_count_observer_, |
| &send_side_delay_observer_, |
| max_rtp_streams)) { |
| vie_receiver_.Init(rtp_rtcp_modules_); |
| if (sender_) { |
| RTC_DCHECK(send_payload_router_); |
| RTC_DCHECK(!vcm_); |
| } else { |
| RTC_DCHECK(!send_payload_router_); |
| RTC_DCHECK(vcm_); |
| vcm_->SetNackSettings(kMaxNackListSize, max_nack_reordering_threshold_, 0); |
| } |
| } |
| |
| int32_t ViEChannel::Init() { |
| static const int kDefaultRenderDelayMs = 10; |
| module_process_thread_->RegisterModule(vie_receiver_.GetReceiveStatistics()); |
| |
| // RTP/RTCP initialization. |
| for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { |
| module_process_thread_->RegisterModule(rtp_rtcp); |
| packet_router_->AddRtpModule(rtp_rtcp); |
| } |
| |
| rtp_rtcp_modules_[0]->SetKeyFrameRequestMethod(kKeyFrameReqPliRtcp); |
| if (paced_sender_) { |
| for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) |
| rtp_rtcp->SetStorePacketsStatus(true, kMinSendSidePacketHistorySize); |
| } |
| if (sender_) { |
| send_payload_router_->SetSendingRtpModules(1); |
| RTC_DCHECK(!send_payload_router_->active()); |
| } else { |
| if (vcm_->RegisterReceiveCallback(this) != 0) { |
| return -1; |
| } |
| vcm_->RegisterFrameTypeCallback(this); |
| vcm_->RegisterReceiveStatisticsCallback(this); |
| vcm_->RegisterDecoderTimingCallback(this); |
| vcm_->SetRenderDelay(kDefaultRenderDelayMs); |
| } |
| return 0; |
| } |
| |
| ViEChannel::~ViEChannel() { |
| // Make sure we don't get more callbacks from the RTP module. |
| module_process_thread_->DeRegisterModule( |
| vie_receiver_.GetReceiveStatistics()); |
| if (sender_) { |
| send_payload_router_->SetSendingRtpModules(0); |
| } |
| for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { |
| packet_router_->RemoveRtpModule(rtp_rtcp); |
| module_process_thread_->DeRegisterModule(rtp_rtcp); |
| delete rtp_rtcp; |
| } |
| } |
| |
| void ViEChannel::SetProtectionMode(bool enable_nack, |
| bool enable_fec, |
| int payload_type_red, |
| int payload_type_fec) { |
| // 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_fec || payload_type_red != -1 || payload_type_fec != -1) { |
| RTC_DCHECK_GE(payload_type_red, 0); |
| RTC_DCHECK_GE(payload_type_fec, 0); |
| RTC_DCHECK_LE(payload_type_red, 127); |
| RTC_DCHECK_LE(payload_type_fec, 127); |
| } else { |
| // Payload types unset. |
| RTC_DCHECK_EQ(payload_type_red, -1); |
| RTC_DCHECK_EQ(payload_type_fec, -1); |
| // Set to valid uint8_ts to be castable later without signed overflows. |
| payload_type_red = 0; |
| payload_type_fec = 0; |
| } |
| |
| VCMVideoProtection protection_method; |
| if (enable_nack) { |
| protection_method = enable_fec ? kProtectionNackFEC : kProtectionNack; |
| } else { |
| protection_method = kProtectionNone; |
| } |
| |
| if (!sender_) |
| vcm_->SetVideoProtection(protection_method, true); |
| |
| // Set NACK. |
| ProcessNACKRequest(enable_nack); |
| |
| // Set FEC. |
| for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { |
| rtp_rtcp->SetGenericFECStatus(enable_fec, |
| static_cast<uint8_t>(payload_type_red), |
| static_cast<uint8_t>(payload_type_fec)); |
| } |
| } |
| |
| void ViEChannel::ProcessNACKRequest(const bool enable) { |
| if (enable) { |
| // Turn on NACK. |
| if (rtp_rtcp_modules_[0]->RTCP() == RtcpMode::kOff) |
| return; |
| vie_receiver_.SetNackStatus(true, max_nack_reordering_threshold_); |
| |
| for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) |
| rtp_rtcp->SetStorePacketsStatus(true, kMinSendSidePacketHistorySize); |
| |
| if (!sender_) { |
| vcm_->RegisterPacketRequestCallback(this); |
| // Don't introduce errors when NACK is enabled. |
| vcm_->SetDecodeErrorMode(kNoErrors); |
| } |
| } else { |
| if (!sender_) { |
| vcm_->RegisterPacketRequestCallback(nullptr); |
| for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) |
| rtp_rtcp->SetStorePacketsStatus(false, 0); |
| // When NACK is off, allow decoding with errors. Otherwise, the video |
| // will freeze, and will only recover with a complete key frame. |
| vcm_->SetDecodeErrorMode(kWithErrors); |
| } |
| vie_receiver_.SetNackStatus(false, max_nack_reordering_threshold_); |
| } |
| } |
| |
| int ViEChannel::GetRequiredNackListSize(int target_delay_ms) { |
| // The max size of the nack list should be large enough to accommodate the |
| // the number of packets (frames) resulting from the increased delay. |
| // Roughly estimating for ~40 packets per frame @ 30fps. |
| return target_delay_ms * 40 * 30 / 1000; |
| } |
| |
| RtpState ViEChannel::GetRtpStateForSsrc(uint32_t ssrc) const { |
| RTC_DCHECK(!rtp_rtcp_modules_[0]->Sending()); |
| RtpState rtp_state; |
| for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { |
| if (rtp_rtcp->GetRtpStateForSsrc(ssrc, &rtp_state)) |
| return rtp_state; |
| } |
| LOG(LS_ERROR) << "Couldn't get RTP state for ssrc: " << ssrc; |
| return rtp_state; |
| } |
| |
| void ViEChannel::RegisterRtcpPacketTypeCounterObserver( |
| RtcpPacketTypeCounterObserver* observer) { |
| rtcp_packet_type_counter_observer_.Set(observer); |
| } |
| |
| void ViEChannel::GetSendStreamDataCounters( |
| StreamDataCounters* rtp_counters, |
| StreamDataCounters* rtx_counters) const { |
| *rtp_counters = StreamDataCounters(); |
| *rtx_counters = StreamDataCounters(); |
| for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { |
| StreamDataCounters rtp_data; |
| StreamDataCounters rtx_data; |
| rtp_rtcp->GetSendStreamDataCounters(&rtp_data, &rtx_data); |
| rtp_counters->Add(rtp_data); |
| rtx_counters->Add(rtx_data); |
| } |
| } |
| |
| void ViEChannel::RegisterSendSideDelayObserver( |
| SendSideDelayObserver* observer) { |
| send_side_delay_observer_.Set(observer); |
| } |
| |
| void ViEChannel::RegisterSendBitrateObserver( |
| BitrateStatisticsObserver* observer) { |
| send_bitrate_observer_.Set(observer); |
| } |
| |
| const std::vector<RtpRtcp*>& ViEChannel::rtp_rtcp() const { |
| return rtp_rtcp_modules_; |
| } |
| |
| ViEReceiver* ViEChannel::vie_receiver() { |
| return &vie_receiver_; |
| } |
| |
| VCMProtectionCallback* ViEChannel::vcm_protection_callback() { |
| return vcm_protection_callback_.get(); |
| } |
| |
| CallStatsObserver* ViEChannel::GetStatsObserver() { |
| return stats_observer_.get(); |
| } |
| |
| // Do not acquire the lock of |vcm_| in this function. Decode callback won't |
| // necessarily be called from the decoding thread. The decoding thread may have |
| // held the lock when calling VideoDecoder::Decode, Reset, or Release. Acquiring |
| // the same lock in the path of decode callback can deadlock. |
| int32_t ViEChannel::FrameToRender(VideoFrame& video_frame) { // NOLINT |
| rtc::CritScope lock(&crit_); |
| |
| if (pre_render_callback_ != NULL) |
| pre_render_callback_->FrameCallback(&video_frame); |
| |
| // TODO(pbos): Remove stream id argument. |
| incoming_video_stream_->RenderFrame(0xFFFFFFFF, video_frame); |
| return 0; |
| } |
| |
| int32_t ViEChannel::ReceivedDecodedReferenceFrame( |
| const uint64_t picture_id) { |
| return rtp_rtcp_modules_[0]->SendRTCPReferencePictureSelection(picture_id); |
| } |
| |
| void ViEChannel::OnIncomingPayloadType(int payload_type) { |
| rtc::CritScope lock(&crit_); |
| if (receive_stats_callback_) |
| receive_stats_callback_->OnIncomingPayloadType(payload_type); |
| } |
| |
| void ViEChannel::OnDecoderImplementationName(const char* implementation_name) { |
| rtc::CritScope lock(&crit_); |
| if (receive_stats_callback_) |
| receive_stats_callback_->OnDecoderImplementationName(implementation_name); |
| } |
| |
| void ViEChannel::OnReceiveRatesUpdated(uint32_t bit_rate, uint32_t frame_rate) { |
| rtc::CritScope lock(&crit_); |
| if (receive_stats_callback_) |
| receive_stats_callback_->OnIncomingRate(frame_rate, bit_rate); |
| } |
| |
| void ViEChannel::OnDiscardedPacketsUpdated(int discarded_packets) { |
| rtc::CritScope lock(&crit_); |
| if (receive_stats_callback_) |
| receive_stats_callback_->OnDiscardedPacketsUpdated(discarded_packets); |
| } |
| |
| void ViEChannel::OnFrameCountsUpdated(const FrameCounts& frame_counts) { |
| rtc::CritScope lock(&crit_); |
| receive_frame_counts_ = frame_counts; |
| if (receive_stats_callback_) |
| receive_stats_callback_->OnFrameCountsUpdated(frame_counts); |
| } |
| |
| void ViEChannel::OnDecoderTiming(int decode_ms, |
| int max_decode_ms, |
| int current_delay_ms, |
| int target_delay_ms, |
| int jitter_buffer_ms, |
| int min_playout_delay_ms, |
| int render_delay_ms) { |
| rtc::CritScope lock(&crit_); |
| if (!receive_stats_callback_) |
| return; |
| receive_stats_callback_->OnDecoderTiming( |
| decode_ms, max_decode_ms, current_delay_ms, target_delay_ms, |
| jitter_buffer_ms, min_playout_delay_ms, render_delay_ms, last_rtt_ms_); |
| } |
| |
| int32_t ViEChannel::RequestKeyFrame() { |
| return rtp_rtcp_modules_[0]->RequestKeyFrame(); |
| } |
| |
| int32_t ViEChannel::SliceLossIndicationRequest( |
| const uint64_t picture_id) { |
| return rtp_rtcp_modules_[0]->SendRTCPSliceLossIndication( |
| static_cast<uint8_t>(picture_id)); |
| } |
| |
| int32_t ViEChannel::ResendPackets(const uint16_t* sequence_numbers, |
| uint16_t length) { |
| return rtp_rtcp_modules_[0]->SendNACK(sequence_numbers, length); |
| } |
| |
| void ViEChannel::OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) { |
| if (!sender_) |
| vcm_->SetReceiveChannelParameters(max_rtt_ms); |
| |
| rtc::CritScope lock(&crit_); |
| last_rtt_ms_ = avg_rtt_ms; |
| } |
| |
| int ViEChannel::ProtectionRequest(const FecProtectionParams* delta_fec_params, |
| const FecProtectionParams* key_fec_params, |
| uint32_t* video_rate_bps, |
| uint32_t* nack_rate_bps, |
| uint32_t* fec_rate_bps) { |
| *video_rate_bps = 0; |
| *nack_rate_bps = 0; |
| *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_fec_params, key_fec_params); |
| rtp_rtcp->BitrateSent(¬_used, &module_video_rate, &module_fec_rate, |
| &module_nack_rate); |
| *video_rate_bps += module_video_rate; |
| *nack_rate_bps += module_nack_rate; |
| *fec_rate_bps += module_fec_rate; |
| } |
| return 0; |
| } |
| |
| std::vector<RtpRtcp*> ViEChannel::CreateRtpRtcpModules( |
| bool receiver_only, |
| ReceiveStatistics* receive_statistics, |
| Transport* outgoing_transport, |
| RtcpIntraFrameObserver* intra_frame_callback, |
| RtcpBandwidthObserver* bandwidth_callback, |
| TransportFeedbackObserver* transport_feedback_callback, |
| RtcpRttStats* rtt_stats, |
| RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer, |
| RemoteBitrateEstimator* remote_bitrate_estimator, |
| RtpPacketSender* paced_sender, |
| TransportSequenceNumberAllocator* transport_sequence_number_allocator, |
| BitrateStatisticsObserver* send_bitrate_observer, |
| FrameCountObserver* send_frame_count_observer, |
| SendSideDelayObserver* send_side_delay_observer, |
| 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 = receiver_only; |
| configuration.receive_statistics = 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 = |
| rtcp_packet_type_counter_observer; |
| configuration.paced_sender = paced_sender; |
| configuration.transport_sequence_number_allocator = |
| transport_sequence_number_allocator; |
| configuration.send_bitrate_observer = send_bitrate_observer; |
| configuration.send_frame_count_observer = send_frame_count_observer; |
| configuration.send_side_delay_observer = send_side_delay_observer; |
| 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); |
| // Receive statistics and remote bitrate estimator should only be set for |
| // the primary (first) module. |
| configuration.receive_statistics = null_receive_statistics; |
| configuration.remote_bitrate_estimator = nullptr; |
| } |
| return modules; |
| } |
| |
| void ViEChannel::RegisterPreRenderCallback( |
| I420FrameCallback* pre_render_callback) { |
| RTC_DCHECK(!sender_); |
| rtc::CritScope lock(&crit_); |
| pre_render_callback_ = pre_render_callback; |
| } |
| |
| // TODO(pbos): Remove as soon as audio can handle a changing payload type |
| // without this callback. |
| int32_t ViEChannel::OnInitializeDecoder( |
| const int8_t payload_type, |
| const char payload_name[RTP_PAYLOAD_NAME_SIZE], |
| const int frequency, |
| const size_t channels, |
| const uint32_t rate) { |
| RTC_NOTREACHED(); |
| return 0; |
| } |
| |
| void ViEChannel::OnIncomingSSRCChanged(const uint32_t ssrc) { |
| rtp_rtcp_modules_[0]->SetRemoteSSRC(ssrc); |
| } |
| |
| void ViEChannel::OnIncomingCSRCChanged(const uint32_t CSRC, const bool added) {} |
| |
| void ViEChannel::RegisterSendFrameCountObserver( |
| FrameCountObserver* observer) { |
| send_frame_count_observer_.Set(observer); |
| } |
| |
| void ViEChannel::RegisterReceiveStatisticsProxy( |
| ReceiveStatisticsProxy* receive_statistics_proxy) { |
| rtc::CritScope lock(&crit_); |
| receive_stats_callback_ = receive_statistics_proxy; |
| } |
| |
| void ViEChannel::SetIncomingVideoStream( |
| IncomingVideoStream* incoming_video_stream) { |
| rtc::CritScope lock(&crit_); |
| incoming_video_stream_ = incoming_video_stream; |
| } |
| } // namespace webrtc |