| /* |
| * 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_engine/vie_channel.h" |
| |
| #include <algorithm> |
| #include <vector> |
| |
| #include "webrtc/base/checks.h" |
| #include "webrtc/common.h" |
| #include "webrtc/common_video/interface/incoming_video_stream.h" |
| #include "webrtc/common_video/libyuv/include/webrtc_libyuv.h" |
| #include "webrtc/frame_callback.h" |
| #include "webrtc/modules/pacing/include/paced_sender.h" |
| #include "webrtc/modules/pacing/include/packet_router.h" |
| #include "webrtc/modules/rtp_rtcp/interface/rtp_receiver.h" |
| #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h" |
| #include "webrtc/modules/utility/interface/process_thread.h" |
| #include "webrtc/modules/video_coding/main/interface/video_coding.h" |
| #include "webrtc/modules/video_processing/main/interface/video_processing.h" |
| #include "webrtc/modules/video_render/include/video_render_defines.h" |
| #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" |
| #include "webrtc/system_wrappers/interface/logging.h" |
| #include "webrtc/system_wrappers/interface/metrics.h" |
| #include "webrtc/system_wrappers/interface/thread_wrapper.h" |
| #include "webrtc/video/receive_statistics_proxy.h" |
| #include "webrtc/video_engine/call_stats.h" |
| #include "webrtc/video_engine/payload_router.h" |
| #include "webrtc/video_engine/report_block_stats.h" |
| #include "webrtc/video_engine/vie_defines.h" |
| |
| namespace webrtc { |
| |
| const int kMaxDecodeWaitTimeMs = 50; |
| static const int kMaxTargetDelayMs = 10000; |
| static const float kMaxIncompleteTimeMultiplier = 3.5f; |
| |
| // 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: |
| 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(uint32_t number_of_cores, |
| Transport* transport, |
| ProcessThread* module_process_thread, |
| 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) |
| : number_of_cores_(number_of_cores), |
| sender_(sender), |
| module_process_thread_(module_process_thread), |
| crit_(CriticalSectionWrapper::CreateCriticalSection()), |
| send_payload_router_(new PayloadRouter()), |
| vcm_protection_callback_(new ViEChannelProtectionCallback(this)), |
| vcm_(VideoCodingModule::Create(Clock::GetRealTimeClock(), |
| nullptr, |
| nullptr)), |
| vie_receiver_(vcm_, remote_bitrate_estimator, this), |
| vie_sync_(vcm_), |
| 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), |
| nack_history_size_sender_(kSendSidePacketHistorySize), |
| max_nack_reordering_threshold_(kMaxPacketAgeToNack), |
| pre_render_callback_(NULL), |
| report_block_stats_sender_(new ReportBlockStats()), |
| time_of_first_rtt_ms_(-1), |
| rtt_sum_ms_(0), |
| last_rtt_ms_(0), |
| num_rtts_(0), |
| rtp_rtcp_modules_( |
| CreateRtpRtcpModules(!sender, |
| vie_receiver_.GetReceiveStatistics(), |
| transport, |
| sender ? intra_frame_observer_ : nullptr, |
| sender ? bandwidth_observer_.get() : nullptr, |
| 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)), |
| num_active_rtp_rtcp_modules_(1) { |
| vie_receiver_.SetRtpRtcpModule(rtp_rtcp_modules_[0]); |
| vcm_->SetNackSettings(kMaxNackListSize, max_nack_reordering_threshold_, 0); |
| } |
| |
| int32_t ViEChannel::Init() { |
| module_process_thread_->RegisterModule(vie_receiver_.GetReceiveStatistics()); |
| |
| // RTP/RTCP initialization. |
| module_process_thread_->RegisterModule(rtp_rtcp_modules_[0]); |
| |
| rtp_rtcp_modules_[0]->SetKeyFrameRequestMethod(kKeyFrameReqPliRtcp); |
| if (paced_sender_) { |
| for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) |
| rtp_rtcp->SetStorePacketsStatus(true, nack_history_size_sender_); |
| } |
| packet_router_->AddRtpModule(rtp_rtcp_modules_[0]); |
| if (sender_) { |
| std::list<RtpRtcp*> send_rtp_modules(1, rtp_rtcp_modules_[0]); |
| send_payload_router_->SetSendingRtpModules(send_rtp_modules); |
| RTC_DCHECK(!send_payload_router_->active()); |
| } |
| if (vcm_->RegisterReceiveCallback(this) != 0) { |
| return -1; |
| } |
| vcm_->RegisterFrameTypeCallback(this); |
| vcm_->RegisterReceiveStatisticsCallback(this); |
| vcm_->RegisterDecoderTimingCallback(this); |
| vcm_->SetRenderDelay(kViEDefaultRenderDelayMs); |
| |
| module_process_thread_->RegisterModule(vcm_); |
| module_process_thread_->RegisterModule(&vie_sync_); |
| |
| return 0; |
| } |
| |
| ViEChannel::~ViEChannel() { |
| UpdateHistograms(); |
| // Make sure we don't get more callbacks from the RTP module. |
| module_process_thread_->DeRegisterModule( |
| vie_receiver_.GetReceiveStatistics()); |
| module_process_thread_->DeRegisterModule(vcm_); |
| module_process_thread_->DeRegisterModule(&vie_sync_); |
| send_payload_router_->SetSendingRtpModules(std::list<RtpRtcp*>()); |
| for (size_t i = 0; i < num_active_rtp_rtcp_modules_; ++i) |
| packet_router_->RemoveRtpModule(rtp_rtcp_modules_[i]); |
| for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { |
| module_process_thread_->DeRegisterModule(rtp_rtcp); |
| delete rtp_rtcp; |
| } |
| if (decode_thread_) { |
| StopDecodeThread(); |
| } |
| // Release modules. |
| VideoCodingModule::Destroy(vcm_); |
| } |
| |
| void ViEChannel::UpdateHistograms() { |
| int64_t now = Clock::GetRealTimeClock()->TimeInMilliseconds(); |
| |
| { |
| CriticalSectionScoped cs(crit_.get()); |
| int64_t elapsed_sec = (now - time_of_first_rtt_ms_) / 1000; |
| if (time_of_first_rtt_ms_ != -1 && num_rtts_ > 0 && |
| elapsed_sec > metrics::kMinRunTimeInSeconds) { |
| int64_t avg_rtt_ms = (rtt_sum_ms_ + num_rtts_ / 2) / num_rtts_; |
| RTC_HISTOGRAM_COUNTS_10000( |
| "WebRTC.Video.AverageRoundTripTimeInMilliseconds", avg_rtt_ms); |
| } |
| } |
| |
| if (sender_) { |
| RtcpPacketTypeCounter rtcp_counter; |
| GetSendRtcpPacketTypeCounter(&rtcp_counter); |
| int64_t elapsed_sec = rtcp_counter.TimeSinceFirstPacketInMs(now) / 1000; |
| if (elapsed_sec > metrics::kMinRunTimeInSeconds) { |
| RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.NackPacketsReceivedPerMinute", |
| rtcp_counter.nack_packets * 60 / elapsed_sec); |
| RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.FirPacketsReceivedPerMinute", |
| rtcp_counter.fir_packets * 60 / elapsed_sec); |
| RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.PliPacketsReceivedPerMinute", |
| rtcp_counter.pli_packets * 60 / elapsed_sec); |
| if (rtcp_counter.nack_requests > 0) { |
| RTC_HISTOGRAM_PERCENTAGE( |
| "WebRTC.Video.UniqueNackRequestsReceivedInPercent", |
| rtcp_counter.UniqueNackRequestsInPercent()); |
| } |
| int fraction_lost = report_block_stats_sender_->FractionLostInPercent(); |
| if (fraction_lost != -1) { |
| RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.SentPacketsLostInPercent", |
| fraction_lost); |
| } |
| } |
| |
| StreamDataCounters rtp; |
| StreamDataCounters rtx; |
| GetSendStreamDataCounters(&rtp, &rtx); |
| StreamDataCounters rtp_rtx = rtp; |
| rtp_rtx.Add(rtx); |
| elapsed_sec = rtp_rtx.TimeSinceFirstPacketInMs( |
| Clock::GetRealTimeClock()->TimeInMilliseconds()) / |
| 1000; |
| if (elapsed_sec > metrics::kMinRunTimeInSeconds) { |
| RTC_HISTOGRAM_COUNTS_100000( |
| "WebRTC.Video.BitrateSentInKbps", |
| static_cast<int>(rtp_rtx.transmitted.TotalBytes() * 8 / elapsed_sec / |
| 1000)); |
| RTC_HISTOGRAM_COUNTS_10000( |
| "WebRTC.Video.MediaBitrateSentInKbps", |
| static_cast<int>(rtp.MediaPayloadBytes() * 8 / elapsed_sec / 1000)); |
| RTC_HISTOGRAM_COUNTS_10000( |
| "WebRTC.Video.PaddingBitrateSentInKbps", |
| static_cast<int>(rtp_rtx.transmitted.padding_bytes * 8 / elapsed_sec / |
| 1000)); |
| RTC_HISTOGRAM_COUNTS_10000( |
| "WebRTC.Video.RetransmittedBitrateSentInKbps", |
| static_cast<int>(rtp_rtx.retransmitted.TotalBytes() * 8 / |
| elapsed_sec / 1000)); |
| if (rtp_rtcp_modules_[0]->RtxSendStatus() != kRtxOff) { |
| RTC_HISTOGRAM_COUNTS_10000( |
| "WebRTC.Video.RtxBitrateSentInKbps", |
| static_cast<int>(rtx.transmitted.TotalBytes() * 8 / elapsed_sec / |
| 1000)); |
| } |
| bool fec_enabled = false; |
| uint8_t pltype_red; |
| uint8_t pltype_fec; |
| rtp_rtcp_modules_[0]->GenericFECStatus(fec_enabled, pltype_red, |
| pltype_fec); |
| if (fec_enabled) { |
| RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.FecBitrateSentInKbps", |
| static_cast<int>(rtp_rtx.fec.TotalBytes() * |
| 8 / elapsed_sec / 1000)); |
| } |
| } |
| } else if (vie_receiver_.GetRemoteSsrc() > 0) { |
| // Get receive stats if we are receiving packets, i.e. there is a remote |
| // ssrc. |
| RtcpPacketTypeCounter rtcp_counter; |
| GetReceiveRtcpPacketTypeCounter(&rtcp_counter); |
| int64_t elapsed_sec = rtcp_counter.TimeSinceFirstPacketInMs(now) / 1000; |
| if (elapsed_sec > metrics::kMinRunTimeInSeconds) { |
| RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.NackPacketsSentPerMinute", |
| rtcp_counter.nack_packets * 60 / elapsed_sec); |
| RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.FirPacketsSentPerMinute", |
| rtcp_counter.fir_packets * 60 / elapsed_sec); |
| RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.PliPacketsSentPerMinute", |
| rtcp_counter.pli_packets * 60 / elapsed_sec); |
| if (rtcp_counter.nack_requests > 0) { |
| RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.UniqueNackRequestsSentInPercent", |
| rtcp_counter.UniqueNackRequestsInPercent()); |
| } |
| } |
| |
| StreamDataCounters rtp; |
| StreamDataCounters rtx; |
| GetReceiveStreamDataCounters(&rtp, &rtx); |
| StreamDataCounters rtp_rtx = rtp; |
| rtp_rtx.Add(rtx); |
| elapsed_sec = rtp_rtx.TimeSinceFirstPacketInMs(now) / 1000; |
| if (elapsed_sec > metrics::kMinRunTimeInSeconds) { |
| RTC_HISTOGRAM_COUNTS_10000( |
| "WebRTC.Video.BitrateReceivedInKbps", |
| static_cast<int>(rtp_rtx.transmitted.TotalBytes() * 8 / elapsed_sec / |
| 1000)); |
| RTC_HISTOGRAM_COUNTS_10000( |
| "WebRTC.Video.MediaBitrateReceivedInKbps", |
| static_cast<int>(rtp.MediaPayloadBytes() * 8 / elapsed_sec / 1000)); |
| RTC_HISTOGRAM_COUNTS_10000( |
| "WebRTC.Video.PaddingBitrateReceivedInKbps", |
| static_cast<int>(rtp_rtx.transmitted.padding_bytes * 8 / elapsed_sec / |
| 1000)); |
| RTC_HISTOGRAM_COUNTS_10000( |
| "WebRTC.Video.RetransmittedBitrateReceivedInKbps", |
| static_cast<int>(rtp_rtx.retransmitted.TotalBytes() * 8 / |
| elapsed_sec / 1000)); |
| uint32_t ssrc = 0; |
| if (vie_receiver_.GetRtxSsrc(&ssrc)) { |
| RTC_HISTOGRAM_COUNTS_10000( |
| "WebRTC.Video.RtxBitrateReceivedInKbps", |
| static_cast<int>(rtx.transmitted.TotalBytes() * 8 / elapsed_sec / |
| 1000)); |
| } |
| if (vie_receiver_.IsFecEnabled()) { |
| RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.FecBitrateReceivedInKbps", |
| static_cast<int>(rtp_rtx.fec.TotalBytes() * |
| 8 / elapsed_sec / 1000)); |
| } |
| } |
| } |
| } |
| |
| int32_t ViEChannel::SetSendCodec(const VideoCodec& video_codec, |
| bool new_stream) { |
| RTC_DCHECK(sender_); |
| if (video_codec.codecType == kVideoCodecRED || |
| video_codec.codecType == kVideoCodecULPFEC) { |
| LOG_F(LS_ERROR) << "Not a valid send codec " << video_codec.codecType; |
| return -1; |
| } |
| if (kMaxSimulcastStreams < video_codec.numberOfSimulcastStreams) { |
| LOG_F(LS_ERROR) << "Incorrect config " |
| << video_codec.numberOfSimulcastStreams; |
| return -1; |
| } |
| // Update the RTP module with the settings. |
| // Stop and Start the RTP module -> trigger new SSRC, if an SSRC hasn't been |
| // set explicitly. |
| // The first layer is always active, so the first module can be checked for |
| // sending status. |
| bool is_sending = rtp_rtcp_modules_[0]->Sending(); |
| bool router_was_active = send_payload_router_->active(); |
| send_payload_router_->set_active(false); |
| send_payload_router_->SetSendingRtpModules(std::list<RtpRtcp*>()); |
| |
| std::vector<RtpRtcp*> registered_modules; |
| std::vector<RtpRtcp*> deregistered_modules; |
| size_t num_active_modules = video_codec.numberOfSimulcastStreams > 0 |
| ? video_codec.numberOfSimulcastStreams |
| : 1; |
| size_t num_prev_active_modules; |
| { |
| // Cache which modules are active so StartSend can know which ones to start. |
| CriticalSectionScoped cs(crit_.get()); |
| num_prev_active_modules = num_active_rtp_rtcp_modules_; |
| num_active_rtp_rtcp_modules_ = num_active_modules; |
| } |
| for (size_t i = 0; i < num_active_modules; ++i) |
| registered_modules.push_back(rtp_rtcp_modules_[i]); |
| |
| for (size_t i = num_active_modules; i < rtp_rtcp_modules_.size(); ++i) |
| deregistered_modules.push_back(rtp_rtcp_modules_[i]); |
| |
| // Disable inactive modules. |
| for (RtpRtcp* rtp_rtcp : deregistered_modules) { |
| rtp_rtcp->SetSendingStatus(false); |
| rtp_rtcp->SetSendingMediaStatus(false); |
| } |
| |
| // Configure active modules. |
| for (RtpRtcp* rtp_rtcp : registered_modules) { |
| rtp_rtcp->DeRegisterSendPayload(video_codec.plType); |
| if (rtp_rtcp->RegisterSendPayload(video_codec) != 0) { |
| return -1; |
| } |
| rtp_rtcp->SetSendingStatus(is_sending); |
| rtp_rtcp->SetSendingMediaStatus(is_sending); |
| } |
| |
| // |RegisterSimulcastRtpRtcpModules| resets all old weak pointers and old |
| // modules can be deleted after this step. |
| vie_receiver_.RegisterRtpRtcpModules(registered_modules); |
| |
| // Update the packet and payload routers with the sending RtpRtcp modules. |
| if (sender_) { |
| std::list<RtpRtcp*> active_send_modules; |
| for (RtpRtcp* rtp_rtcp : registered_modules) |
| active_send_modules.push_back(rtp_rtcp); |
| send_payload_router_->SetSendingRtpModules(active_send_modules); |
| } |
| |
| if (router_was_active) |
| send_payload_router_->set_active(true); |
| |
| // Deregister previously registered modules. |
| for (size_t i = num_active_modules; i < num_prev_active_modules; ++i) { |
| module_process_thread_->DeRegisterModule(rtp_rtcp_modules_[i]); |
| packet_router_->RemoveRtpModule(rtp_rtcp_modules_[i]); |
| } |
| // Register new active modules. |
| for (size_t i = num_prev_active_modules; i < num_active_modules; ++i) { |
| module_process_thread_->RegisterModule(rtp_rtcp_modules_[i]); |
| packet_router_->AddRtpModule(rtp_rtcp_modules_[i]); |
| } |
| return 0; |
| } |
| |
| int32_t ViEChannel::SetReceiveCodec(const VideoCodec& video_codec) { |
| RTC_DCHECK(!sender_); |
| if (!vie_receiver_.SetReceiveCodec(video_codec)) { |
| return -1; |
| } |
| |
| if (video_codec.codecType != kVideoCodecRED && |
| video_codec.codecType != kVideoCodecULPFEC) { |
| // Register codec type with VCM, but do not register RED or ULPFEC. |
| if (vcm_->RegisterReceiveCodec(&video_codec, number_of_cores_, false) != |
| VCM_OK) { |
| return -1; |
| } |
| } |
| return 0; |
| } |
| |
| |
| int32_t ViEChannel::RegisterExternalDecoder(const uint8_t pl_type, |
| VideoDecoder* decoder, |
| bool buffered_rendering, |
| int32_t render_delay) { |
| RTC_DCHECK(!sender_); |
| int32_t result; |
| result = vcm_->RegisterExternalDecoder(decoder, pl_type, buffered_rendering); |
| if (result != VCM_OK) { |
| return result; |
| } |
| return vcm_->SetRenderDelay(render_delay); |
| } |
| |
| int32_t ViEChannel::DeRegisterExternalDecoder(const uint8_t pl_type) { |
| RTC_DCHECK(!sender_); |
| VideoCodec current_receive_codec; |
| int32_t result = 0; |
| result = vcm_->ReceiveCodec(¤t_receive_codec); |
| if (vcm_->RegisterExternalDecoder(NULL, pl_type, false) != VCM_OK) { |
| return -1; |
| } |
| |
| if (result == 0 && current_receive_codec.plType == pl_type) { |
| result = vcm_->RegisterReceiveCodec(¤t_receive_codec, |
| number_of_cores_, false); |
| } |
| return result; |
| } |
| |
| int32_t ViEChannel::ReceiveCodecStatistics(uint32_t* num_key_frames, |
| uint32_t* num_delta_frames) { |
| CriticalSectionScoped cs(crit_.get()); |
| *num_key_frames = receive_frame_counts_.key_frames; |
| *num_delta_frames = receive_frame_counts_.delta_frames; |
| return 0; |
| } |
| |
| uint32_t ViEChannel::DiscardedPackets() const { |
| return vcm_->DiscardedPackets(); |
| } |
| |
| int ViEChannel::ReceiveDelay() const { |
| return vcm_->Delay(); |
| } |
| |
| void ViEChannel::SetRTCPMode(const RtcpMode rtcp_mode) { |
| for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) |
| rtp_rtcp->SetRTCPStatus(rtcp_mode); |
| } |
| |
| void ViEChannel::SetProtectionMode(bool enable_nack, |
| bool enable_fec, |
| int payload_type_red, |
| int payload_type_fec) { |
| // Validate payload types. |
| if (enable_fec) { |
| 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 { |
| 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; |
| } |
| |
| 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, nack_history_size_sender_); |
| |
| vcm_->RegisterPacketRequestCallback(this); |
| // Don't introduce errors when NACK is enabled. |
| vcm_->SetDecodeErrorMode(kNoErrors); |
| } else { |
| vcm_->RegisterPacketRequestCallback(NULL); |
| if (paced_sender_ == nullptr) { |
| for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) |
| rtp_rtcp->SetStorePacketsStatus(false, 0); |
| } |
| vie_receiver_.SetNackStatus(false, max_nack_reordering_threshold_); |
| // 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); |
| } |
| } |
| |
| bool ViEChannel::IsSendingFecEnabled() { |
| bool fec_enabled = false; |
| uint8_t pltype_red = 0; |
| uint8_t pltype_fec = 0; |
| |
| for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { |
| rtp_rtcp->GenericFECStatus(fec_enabled, pltype_red, pltype_fec); |
| if (fec_enabled) |
| return true; |
| } |
| return false; |
| } |
| |
| int ViEChannel::SetSenderBufferingMode(int target_delay_ms) { |
| if ((target_delay_ms < 0) || (target_delay_ms > kMaxTargetDelayMs)) { |
| LOG(LS_ERROR) << "Invalid send buffer value."; |
| return -1; |
| } |
| if (target_delay_ms == 0) { |
| // Real-time mode. |
| nack_history_size_sender_ = kSendSidePacketHistorySize; |
| } else { |
| nack_history_size_sender_ = GetRequiredNackListSize(target_delay_ms); |
| // Don't allow a number lower than the default value. |
| if (nack_history_size_sender_ < kSendSidePacketHistorySize) { |
| nack_history_size_sender_ = kSendSidePacketHistorySize; |
| } |
| } |
| for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) |
| rtp_rtcp->SetStorePacketsStatus(true, nack_history_size_sender_); |
| return 0; |
| } |
| |
| int ViEChannel::SetReceiverBufferingMode(int target_delay_ms) { |
| if ((target_delay_ms < 0) || (target_delay_ms > kMaxTargetDelayMs)) { |
| LOG(LS_ERROR) << "Invalid receive buffer delay value."; |
| return -1; |
| } |
| int max_nack_list_size; |
| int max_incomplete_time_ms; |
| if (target_delay_ms == 0) { |
| // Real-time mode - restore default settings. |
| max_nack_reordering_threshold_ = kMaxPacketAgeToNack; |
| max_nack_list_size = kMaxNackListSize; |
| max_incomplete_time_ms = 0; |
| } else { |
| max_nack_list_size = 3 * GetRequiredNackListSize(target_delay_ms) / 4; |
| max_nack_reordering_threshold_ = max_nack_list_size; |
| // Calculate the max incomplete time and round to int. |
| max_incomplete_time_ms = static_cast<int>(kMaxIncompleteTimeMultiplier * |
| target_delay_ms + 0.5f); |
| } |
| vcm_->SetNackSettings(max_nack_list_size, max_nack_reordering_threshold_, |
| max_incomplete_time_ms); |
| vcm_->SetMinReceiverDelay(target_delay_ms); |
| if (vie_sync_.SetTargetBufferingDelay(target_delay_ms) < 0) |
| return -1; |
| return 0; |
| } |
| |
| 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; |
| } |
| |
| int ViEChannel::SetSendTimestampOffsetStatus(bool enable, int id) { |
| // Disable any previous registrations of this extension to avoid errors. |
| for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { |
| rtp_rtcp->DeregisterSendRtpHeaderExtension( |
| kRtpExtensionTransmissionTimeOffset); |
| } |
| if (!enable) |
| return 0; |
| // Enable the extension. |
| int error = 0; |
| for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { |
| error |= rtp_rtcp->RegisterSendRtpHeaderExtension( |
| kRtpExtensionTransmissionTimeOffset, id); |
| } |
| return error; |
| } |
| |
| int ViEChannel::SetReceiveTimestampOffsetStatus(bool enable, int id) { |
| return vie_receiver_.SetReceiveTimestampOffsetStatus(enable, id) ? 0 : -1; |
| } |
| |
| int ViEChannel::SetSendAbsoluteSendTimeStatus(bool enable, int id) { |
| // Disable any previous registrations of this extension to avoid errors. |
| for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) |
| rtp_rtcp->DeregisterSendRtpHeaderExtension(kRtpExtensionAbsoluteSendTime); |
| if (!enable) |
| return 0; |
| // Enable the extension. |
| int error = 0; |
| for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { |
| error |= rtp_rtcp->RegisterSendRtpHeaderExtension( |
| kRtpExtensionAbsoluteSendTime, id); |
| } |
| return error; |
| } |
| |
| int ViEChannel::SetReceiveAbsoluteSendTimeStatus(bool enable, int id) { |
| return vie_receiver_.SetReceiveAbsoluteSendTimeStatus(enable, id) ? 0 : -1; |
| } |
| |
| int ViEChannel::SetSendVideoRotationStatus(bool enable, int id) { |
| // Disable any previous registrations of this extension to avoid errors. |
| for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) |
| rtp_rtcp->DeregisterSendRtpHeaderExtension(kRtpExtensionVideoRotation); |
| if (!enable) |
| return 0; |
| // Enable the extension. |
| int error = 0; |
| for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { |
| error |= rtp_rtcp->RegisterSendRtpHeaderExtension( |
| kRtpExtensionVideoRotation, id); |
| } |
| return error; |
| } |
| |
| int ViEChannel::SetReceiveVideoRotationStatus(bool enable, int id) { |
| return vie_receiver_.SetReceiveVideoRotationStatus(enable, id) ? 0 : -1; |
| } |
| |
| int ViEChannel::SetSendTransportSequenceNumber(bool enable, int id) { |
| // Disable any previous registrations of this extension to avoid errors. |
| for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { |
| rtp_rtcp->DeregisterSendRtpHeaderExtension( |
| kRtpExtensionTransportSequenceNumber); |
| } |
| if (!enable) |
| return 0; |
| // Enable the extension. |
| int error = 0; |
| for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { |
| error |= rtp_rtcp->RegisterSendRtpHeaderExtension( |
| kRtpExtensionTransportSequenceNumber, id); |
| } |
| return error; |
| } |
| |
| int ViEChannel::SetReceiveTransportSequenceNumber(bool enable, int id) { |
| return vie_receiver_.SetReceiveTransportSequenceNumber(enable, id) ? 0 : -1; |
| } |
| |
| void ViEChannel::SetRtcpXrRrtrStatus(bool enable) { |
| rtp_rtcp_modules_[0]->SetRtcpXrRrtrStatus(enable); |
| } |
| |
| void ViEChannel::EnableTMMBR(bool enable) { |
| rtp_rtcp_modules_[0]->SetTMMBRStatus(enable); |
| } |
| |
| int32_t ViEChannel::SetSSRC(const uint32_t SSRC, |
| const StreamType usage, |
| const uint8_t simulcast_idx) { |
| RtpRtcp* rtp_rtcp = rtp_rtcp_modules_[simulcast_idx]; |
| if (usage == kViEStreamTypeRtx) { |
| rtp_rtcp->SetRtxSsrc(SSRC); |
| } else { |
| rtp_rtcp->SetSSRC(SSRC); |
| } |
| return 0; |
| } |
| |
| int32_t ViEChannel::SetRemoteSSRCType(const StreamType usage, |
| const uint32_t SSRC) { |
| vie_receiver_.SetRtxSsrc(SSRC); |
| return 0; |
| } |
| |
| int32_t ViEChannel::GetLocalSSRC(uint8_t idx, unsigned int* ssrc) { |
| RTC_DCHECK_LE(idx, rtp_rtcp_modules_.size()); |
| *ssrc = rtp_rtcp_modules_[idx]->SSRC(); |
| return 0; |
| } |
| |
| uint32_t ViEChannel::GetRemoteSSRC() { |
| return vie_receiver_.GetRemoteSsrc(); |
| } |
| |
| int ViEChannel::SetRtxSendPayloadType(int payload_type, |
| int associated_payload_type) { |
| for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) |
| rtp_rtcp->SetRtxSendPayloadType(payload_type, associated_payload_type); |
| SetRtxSendStatus(true); |
| return 0; |
| } |
| |
| void ViEChannel::SetRtxSendStatus(bool enable) { |
| int rtx_settings = |
| enable ? kRtxRetransmitted | kRtxRedundantPayloads : kRtxOff; |
| for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) |
| rtp_rtcp->SetRtxSendStatus(rtx_settings); |
| } |
| |
| void ViEChannel::SetRtxReceivePayloadType(int payload_type, |
| int associated_payload_type) { |
| vie_receiver_.SetRtxPayloadType(payload_type, associated_payload_type); |
| } |
| |
| void ViEChannel::SetUseRtxPayloadMappingOnRestore(bool val) { |
| vie_receiver_.SetUseRtxPayloadMappingOnRestore(val); |
| } |
| |
| void ViEChannel::SetRtpStateForSsrc(uint32_t ssrc, const RtpState& rtp_state) { |
| RTC_DCHECK(!rtp_rtcp_modules_[0]->Sending()); |
| for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { |
| if (rtp_rtcp->SetRtpStateForSsrc(ssrc, rtp_state)) |
| return; |
| } |
| } |
| |
| RtpState ViEChannel::GetRtpStateForSsrc(uint32_t ssrc) { |
| 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; |
| } |
| |
| // TODO(pbos): Set CNAME on all modules. |
| int32_t ViEChannel::SetRTCPCName(const char* rtcp_cname) { |
| RTC_DCHECK(!rtp_rtcp_modules_[0]->Sending()); |
| return rtp_rtcp_modules_[0]->SetCNAME(rtcp_cname); |
| } |
| |
| int32_t ViEChannel::GetRemoteRTCPCName(char rtcp_cname[]) { |
| uint32_t remoteSSRC = vie_receiver_.GetRemoteSsrc(); |
| return rtp_rtcp_modules_[0]->RemoteCNAME(remoteSSRC, rtcp_cname); |
| } |
| |
| int32_t ViEChannel::GetSendRtcpStatistics(uint16_t* fraction_lost, |
| uint32_t* cumulative_lost, |
| uint32_t* extended_max, |
| uint32_t* jitter_samples, |
| int64_t* rtt_ms) { |
| // Aggregate the report blocks associated with streams sent on this channel. |
| std::vector<RTCPReportBlock> report_blocks; |
| for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) |
| rtp_rtcp->RemoteRTCPStat(&report_blocks); |
| |
| if (report_blocks.empty()) |
| return -1; |
| |
| uint32_t remote_ssrc = vie_receiver_.GetRemoteSsrc(); |
| std::vector<RTCPReportBlock>::const_iterator it = report_blocks.begin(); |
| for (; it != report_blocks.end(); ++it) { |
| if (it->remoteSSRC == remote_ssrc) |
| break; |
| } |
| if (it == report_blocks.end()) { |
| // We have not received packets with an SSRC matching the report blocks. To |
| // have a chance of calculating an RTT we will try with the SSRC of the |
| // first report block received. |
| // This is very important for send-only channels where we don't know the |
| // SSRC of the other end. |
| remote_ssrc = report_blocks[0].remoteSSRC; |
| } |
| |
| // TODO(asapersson): Change report_block_stats to not rely on |
| // GetSendRtcpStatistics to be called. |
| RTCPReportBlock report = |
| report_block_stats_sender_->AggregateAndStore(report_blocks); |
| *fraction_lost = report.fractionLost; |
| *cumulative_lost = report.cumulativeLost; |
| *extended_max = report.extendedHighSeqNum; |
| *jitter_samples = report.jitter; |
| |
| int64_t dummy; |
| int64_t rtt = 0; |
| if (rtp_rtcp_modules_[0]->RTT(remote_ssrc, &rtt, &dummy, &dummy, &dummy) != |
| 0) { |
| return -1; |
| } |
| *rtt_ms = rtt; |
| return 0; |
| } |
| |
| void ViEChannel::RegisterSendChannelRtcpStatisticsCallback( |
| RtcpStatisticsCallback* callback) { |
| for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) |
| rtp_rtcp->RegisterRtcpStatisticsCallback(callback); |
| } |
| |
| void ViEChannel::RegisterReceiveChannelRtcpStatisticsCallback( |
| RtcpStatisticsCallback* callback) { |
| vie_receiver_.GetReceiveStatistics()->RegisterRtcpStatisticsCallback( |
| callback); |
| rtp_rtcp_modules_[0]->RegisterRtcpStatisticsCallback(callback); |
| } |
| |
| 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::GetReceiveStreamDataCounters( |
| StreamDataCounters* rtp_counters, |
| StreamDataCounters* rtx_counters) const { |
| StreamStatistician* statistician = vie_receiver_.GetReceiveStatistics()-> |
| GetStatistician(vie_receiver_.GetRemoteSsrc()); |
| if (statistician) { |
| statistician->GetReceiveStreamDataCounters(rtp_counters); |
| } |
| uint32_t rtx_ssrc = 0; |
| if (vie_receiver_.GetRtxSsrc(&rtx_ssrc)) { |
| StreamStatistician* statistician = |
| vie_receiver_.GetReceiveStatistics()->GetStatistician(rtx_ssrc); |
| if (statistician) { |
| statistician->GetReceiveStreamDataCounters(rtx_counters); |
| } |
| } |
| } |
| |
| void ViEChannel::RegisterSendChannelRtpStatisticsCallback( |
| StreamDataCountersCallback* callback) { |
| for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) |
| rtp_rtcp->RegisterSendChannelRtpStatisticsCallback(callback); |
| } |
| |
| void ViEChannel::RegisterReceiveChannelRtpStatisticsCallback( |
| StreamDataCountersCallback* callback) { |
| vie_receiver_.GetReceiveStatistics()->RegisterRtpStatisticsCallback(callback); |
| } |
| |
| void ViEChannel::GetSendRtcpPacketTypeCounter( |
| RtcpPacketTypeCounter* packet_counter) const { |
| std::map<uint32_t, RtcpPacketTypeCounter> counter_map = |
| rtcp_packet_type_counter_observer_.GetPacketTypeCounterMap(); |
| |
| RtcpPacketTypeCounter counter; |
| for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) |
| counter.Add(counter_map[rtp_rtcp->SSRC()]); |
| *packet_counter = counter; |
| } |
| |
| void ViEChannel::GetReceiveRtcpPacketTypeCounter( |
| RtcpPacketTypeCounter* packet_counter) const { |
| std::map<uint32_t, RtcpPacketTypeCounter> counter_map = |
| rtcp_packet_type_counter_observer_.GetPacketTypeCounterMap(); |
| |
| RtcpPacketTypeCounter counter; |
| counter.Add(counter_map[vie_receiver_.GetRemoteSsrc()]); |
| |
| *packet_counter = counter; |
| } |
| |
| void ViEChannel::RegisterSendSideDelayObserver( |
| SendSideDelayObserver* observer) { |
| send_side_delay_observer_.Set(observer); |
| } |
| |
| void ViEChannel::RegisterSendBitrateObserver( |
| BitrateStatisticsObserver* observer) { |
| send_bitrate_observer_.Set(observer); |
| } |
| |
| int32_t ViEChannel::StartSend() { |
| CriticalSectionScoped cs(crit_.get()); |
| |
| if (rtp_rtcp_modules_[0]->Sending()) |
| return -1; |
| |
| for (size_t i = 0; i < num_active_rtp_rtcp_modules_; ++i) { |
| RtpRtcp* rtp_rtcp = rtp_rtcp_modules_[i]; |
| rtp_rtcp->SetSendingMediaStatus(true); |
| rtp_rtcp->SetSendingStatus(true); |
| } |
| send_payload_router_->set_active(true); |
| return 0; |
| } |
| |
| int32_t ViEChannel::StopSend() { |
| send_payload_router_->set_active(false); |
| for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) |
| rtp_rtcp->SetSendingMediaStatus(false); |
| |
| if (!rtp_rtcp_modules_[0]->Sending()) { |
| return -1; |
| } |
| |
| for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { |
| rtp_rtcp->SetSendingStatus(false); |
| } |
| return 0; |
| } |
| |
| bool ViEChannel::Sending() { |
| return rtp_rtcp_modules_[0]->Sending(); |
| } |
| |
| void ViEChannel::StartReceive() { |
| if (!sender_) |
| StartDecodeThread(); |
| vie_receiver_.StartReceive(); |
| } |
| |
| void ViEChannel::StopReceive() { |
| vie_receiver_.StopReceive(); |
| if (!sender_) { |
| StopDecodeThread(); |
| vcm_->ResetDecoder(); |
| } |
| } |
| |
| int32_t ViEChannel::ReceivedRTPPacket(const void* rtp_packet, |
| size_t rtp_packet_length, |
| const PacketTime& packet_time) { |
| return vie_receiver_.ReceivedRTPPacket( |
| rtp_packet, rtp_packet_length, packet_time); |
| } |
| |
| int32_t ViEChannel::ReceivedRTCPPacket(const void* rtcp_packet, |
| size_t rtcp_packet_length) { |
| return vie_receiver_.ReceivedRTCPPacket(rtcp_packet, rtcp_packet_length); |
| } |
| |
| int32_t ViEChannel::SetMTU(uint16_t mtu) { |
| for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) |
| rtp_rtcp->SetMaxTransferUnit(mtu); |
| return 0; |
| } |
| |
| RtpRtcp* ViEChannel::rtp_rtcp() { |
| return rtp_rtcp_modules_[0]; |
| } |
| |
| rtc::scoped_refptr<PayloadRouter> ViEChannel::send_payload_router() { |
| return send_payload_router_; |
| } |
| |
| 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 |
| CriticalSectionScoped cs(crit_.get()); |
| |
| 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) { |
| CriticalSectionScoped cs(crit_.get()); |
| if (receive_stats_callback_) |
| receive_stats_callback_->OnIncomingPayloadType(payload_type); |
| } |
| |
| void ViEChannel::OnReceiveRatesUpdated(uint32_t bit_rate, uint32_t frame_rate) { |
| CriticalSectionScoped cs(crit_.get()); |
| if (receive_stats_callback_) |
| receive_stats_callback_->OnIncomingRate(frame_rate, bit_rate); |
| } |
| |
| void ViEChannel::OnDiscardedPacketsUpdated(int discarded_packets) { |
| CriticalSectionScoped cs(crit_.get()); |
| if (receive_stats_callback_) |
| receive_stats_callback_->OnDiscardedPacketsUpdated(discarded_packets); |
| } |
| |
| void ViEChannel::OnFrameCountsUpdated(const FrameCounts& frame_counts) { |
| CriticalSectionScoped cs(crit_.get()); |
| 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) { |
| CriticalSectionScoped cs(crit_.get()); |
| 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); |
| } |
| |
| bool ViEChannel::ChannelDecodeThreadFunction(void* obj) { |
| return static_cast<ViEChannel*>(obj)->ChannelDecodeProcess(); |
| } |
| |
| bool ViEChannel::ChannelDecodeProcess() { |
| vcm_->Decode(kMaxDecodeWaitTimeMs); |
| return true; |
| } |
| |
| void ViEChannel::OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) { |
| vcm_->SetReceiveChannelParameters(max_rtt_ms); |
| |
| CriticalSectionScoped cs(crit_.get()); |
| if (time_of_first_rtt_ms_ == -1) |
| time_of_first_rtt_ms_ = Clock::GetRealTimeClock()->TimeInMilliseconds(); |
| rtt_sum_ms_ += avg_rtt_ms; |
| last_rtt_ms_ = avg_rtt_ms; |
| ++num_rtts_; |
| } |
| |
| 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::StartDecodeThread() { |
| RTC_DCHECK(!sender_); |
| // Start the decode thread |
| if (decode_thread_) |
| return; |
| decode_thread_ = ThreadWrapper::CreateThread(ChannelDecodeThreadFunction, |
| this, "DecodingThread"); |
| decode_thread_->Start(); |
| decode_thread_->SetPriority(kHighestPriority); |
| } |
| |
| void ViEChannel::StopDecodeThread() { |
| if (!decode_thread_) |
| return; |
| |
| vcm_->TriggerDecoderShutdown(); |
| |
| decode_thread_->Stop(); |
| decode_thread_.reset(); |
| } |
| |
| int32_t ViEChannel::SetVoiceChannel(int32_t ve_channel_id, |
| VoEVideoSync* ve_sync_interface) { |
| return vie_sync_.ConfigureSync(ve_channel_id, ve_sync_interface, |
| rtp_rtcp_modules_[0], |
| vie_receiver_.GetRtpReceiver()); |
| } |
| |
| int32_t ViEChannel::VoiceChannel() { |
| return vie_sync_.VoiceChannel(); |
| } |
| |
| void ViEChannel::RegisterPreRenderCallback( |
| I420FrameCallback* pre_render_callback) { |
| CriticalSectionScoped cs(crit_.get()); |
| pre_render_callback_ = pre_render_callback; |
| } |
| |
| void ViEChannel::RegisterPreDecodeImageCallback( |
| EncodedImageCallback* pre_decode_callback) { |
| vcm_->RegisterPreDecodeImageCallback(pre_decode_callback); |
| } |
| |
| // TODO(pbos): Remove OnInitializeDecoder which is called from the RTP module, |
| // any decoder resetting should be handled internally within the VCM. |
| int32_t ViEChannel::OnInitializeDecoder( |
| const int8_t payload_type, |
| const char payload_name[RTP_PAYLOAD_NAME_SIZE], |
| const int frequency, |
| const uint8_t channels, |
| const uint32_t rate) { |
| LOG(LS_INFO) << "OnInitializeDecoder " << static_cast<int>(payload_type) |
| << " " << payload_name; |
| vcm_->ResetDecoder(); |
| |
| 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) { |
| CriticalSectionScoped cs(crit_.get()); |
| receive_stats_callback_ = receive_statistics_proxy; |
| } |
| |
| void ViEChannel::SetIncomingVideoStream( |
| IncomingVideoStream* incoming_video_stream) { |
| CriticalSectionScoped cs(crit_.get()); |
| incoming_video_stream_ = incoming_video_stream; |
| } |
| } // namespace webrtc |
