| /* |
| * 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 "modules/rtp_rtcp/source/rtp_rtcp_impl.h" |
| |
| #include <string.h> |
| |
| #include <algorithm> |
| #include <set> |
| #include <string> |
| |
| #include "api/rtpparameters.h" |
| #include "rtc_base/checks.h" |
| #include "rtc_base/logging.h" |
| |
| #ifdef _WIN32 |
| // Disable warning C4355: 'this' : used in base member initializer list. |
| #pragma warning(disable : 4355) |
| #endif |
| |
| namespace webrtc { |
| namespace { |
| const int64_t kRtpRtcpMaxIdleTimeProcessMs = 5; |
| const int64_t kRtpRtcpRttProcessTimeMs = 1000; |
| const int64_t kRtpRtcpBitrateProcessTimeMs = 10; |
| const int64_t kDefaultExpectedRetransmissionTimeMs = 125; |
| } // namespace |
| |
| RTPExtensionType StringToRtpExtensionType(const std::string& extension) { |
| if (extension == RtpExtension::kTimestampOffsetUri) |
| return kRtpExtensionTransmissionTimeOffset; |
| if (extension == RtpExtension::kAudioLevelUri) |
| return kRtpExtensionAudioLevel; |
| if (extension == RtpExtension::kAbsSendTimeUri) |
| return kRtpExtensionAbsoluteSendTime; |
| if (extension == RtpExtension::kVideoRotationUri) |
| return kRtpExtensionVideoRotation; |
| if (extension == RtpExtension::kTransportSequenceNumberUri) |
| return kRtpExtensionTransportSequenceNumber; |
| if (extension == RtpExtension::kPlayoutDelayUri) |
| return kRtpExtensionPlayoutDelay; |
| if (extension == RtpExtension::kVideoContentTypeUri) |
| return kRtpExtensionVideoContentType; |
| if (extension == RtpExtension::kVideoTimingUri) |
| return kRtpExtensionVideoTiming; |
| if (extension == RtpExtension::kMidUri) |
| return kRtpExtensionMid; |
| RTC_NOTREACHED() << "Looking up unsupported RTP extension."; |
| return kRtpExtensionNone; |
| } |
| |
| RtpRtcp::Configuration::Configuration() = default; |
| |
| RtpRtcp* RtpRtcp::CreateRtpRtcp(const RtpRtcp::Configuration& configuration) { |
| if (configuration.clock) { |
| return new ModuleRtpRtcpImpl(configuration); |
| } else { |
| // No clock implementation provided, use default clock. |
| RtpRtcp::Configuration configuration_copy; |
| memcpy(&configuration_copy, &configuration, |
| sizeof(RtpRtcp::Configuration)); |
| configuration_copy.clock = Clock::GetRealTimeClock(); |
| return new ModuleRtpRtcpImpl(configuration_copy); |
| } |
| } |
| |
| // Deprecated. |
| int32_t RtpRtcp::SetFecParameters(const FecProtectionParams* delta_params, |
| const FecProtectionParams* key_params) { |
| RTC_DCHECK(delta_params); |
| RTC_DCHECK(key_params); |
| return SetFecParameters(*delta_params, *key_params) ? 0 : -1; |
| } |
| |
| ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration) |
| : rtcp_sender_(configuration.audio, |
| configuration.clock, |
| configuration.receive_statistics, |
| configuration.rtcp_packet_type_counter_observer, |
| configuration.event_log, |
| configuration.outgoing_transport, |
| configuration.rtcp_interval_config), |
| rtcp_receiver_(configuration.clock, |
| configuration.receiver_only, |
| configuration.rtcp_packet_type_counter_observer, |
| configuration.bandwidth_callback, |
| configuration.intra_frame_callback, |
| configuration.transport_feedback_callback, |
| configuration.bitrate_allocation_observer, |
| this), |
| clock_(configuration.clock), |
| audio_(configuration.audio), |
| keepalive_config_(configuration.keepalive_config), |
| last_bitrate_process_time_(clock_->TimeInMilliseconds()), |
| last_rtt_process_time_(clock_->TimeInMilliseconds()), |
| next_process_time_(clock_->TimeInMilliseconds() + |
| kRtpRtcpMaxIdleTimeProcessMs), |
| next_keepalive_time_(-1), |
| packet_overhead_(28), // IPV4 UDP. |
| nack_last_time_sent_full_(0), |
| nack_last_time_sent_full_prev_(0), |
| nack_last_seq_number_sent_(0), |
| key_frame_req_method_(kKeyFrameReqPliRtcp), |
| remote_bitrate_(configuration.remote_bitrate_estimator), |
| rtt_stats_(configuration.rtt_stats), |
| rtt_ms_(0) { |
| if (!configuration.receiver_only) { |
| rtp_sender_.reset(new RTPSender( |
| configuration.audio, configuration.clock, |
| configuration.outgoing_transport, configuration.paced_sender, |
| configuration.flexfec_sender, |
| configuration.transport_sequence_number_allocator, |
| configuration.transport_feedback_callback, |
| configuration.send_bitrate_observer, |
| configuration.send_frame_count_observer, |
| configuration.send_side_delay_observer, configuration.event_log, |
| configuration.send_packet_observer, |
| configuration.retransmission_rate_limiter, |
| configuration.overhead_observer, |
| configuration.populate_network2_timestamp)); |
| // Make sure rtcp sender use same timestamp offset as rtp sender. |
| rtcp_sender_.SetTimestampOffset(rtp_sender_->TimestampOffset()); |
| |
| if (keepalive_config_.timeout_interval_ms != -1) { |
| next_keepalive_time_ = |
| clock_->TimeInMilliseconds() + keepalive_config_.timeout_interval_ms; |
| } |
| } |
| |
| // Set default packet size limit. |
| // TODO(nisse): Kind-of duplicates |
| // webrtc::VideoSendStream::Config::Rtp::kDefaultMaxPacketSize. |
| const size_t kTcpOverIpv4HeaderSize = 40; |
| SetMaxRtpPacketSize(IP_PACKET_SIZE - kTcpOverIpv4HeaderSize); |
| } |
| |
| ModuleRtpRtcpImpl::~ModuleRtpRtcpImpl() = default; |
| |
| // Returns the number of milliseconds until the module want a worker thread |
| // to call Process. |
| int64_t ModuleRtpRtcpImpl::TimeUntilNextProcess() { |
| return std::max<int64_t>(0, |
| next_process_time_ - clock_->TimeInMilliseconds()); |
| } |
| |
| // Process any pending tasks such as timeouts (non time critical events). |
| void ModuleRtpRtcpImpl::Process() { |
| const int64_t now = clock_->TimeInMilliseconds(); |
| next_process_time_ = now + kRtpRtcpMaxIdleTimeProcessMs; |
| |
| if (rtp_sender_) { |
| if (now >= last_bitrate_process_time_ + kRtpRtcpBitrateProcessTimeMs) { |
| rtp_sender_->ProcessBitrate(); |
| last_bitrate_process_time_ = now; |
| next_process_time_ = |
| std::min(next_process_time_, now + kRtpRtcpBitrateProcessTimeMs); |
| } |
| if (keepalive_config_.timeout_interval_ms > 0 && |
| now >= next_keepalive_time_) { |
| int64_t last_send_time_ms = rtp_sender_->LastTimestampTimeMs(); |
| // If no packet has been sent, |last_send_time_ms| will be 0, and so the |
| // keep-alive will be triggered as expected. |
| if (now >= last_send_time_ms + keepalive_config_.timeout_interval_ms) { |
| rtp_sender_->SendKeepAlive(keepalive_config_.payload_type); |
| next_keepalive_time_ = now + keepalive_config_.timeout_interval_ms; |
| } else { |
| next_keepalive_time_ = |
| last_send_time_ms + keepalive_config_.timeout_interval_ms; |
| } |
| next_process_time_ = std::min(next_process_time_, next_keepalive_time_); |
| } |
| } |
| |
| bool process_rtt = now >= last_rtt_process_time_ + kRtpRtcpRttProcessTimeMs; |
| if (rtcp_sender_.Sending()) { |
| // Process RTT if we have received a report block and we haven't |
| // processed RTT for at least |kRtpRtcpRttProcessTimeMs| milliseconds. |
| if (rtcp_receiver_.LastReceivedReportBlockMs() > last_rtt_process_time_ && |
| process_rtt) { |
| std::vector<RTCPReportBlock> receive_blocks; |
| rtcp_receiver_.StatisticsReceived(&receive_blocks); |
| int64_t max_rtt = 0; |
| for (std::vector<RTCPReportBlock>::iterator it = receive_blocks.begin(); |
| it != receive_blocks.end(); ++it) { |
| int64_t rtt = 0; |
| rtcp_receiver_.RTT(it->sender_ssrc, &rtt, NULL, NULL, NULL); |
| max_rtt = (rtt > max_rtt) ? rtt : max_rtt; |
| } |
| // Report the rtt. |
| if (rtt_stats_ && max_rtt != 0) |
| rtt_stats_->OnRttUpdate(max_rtt); |
| } |
| |
| // Verify receiver reports are delivered and the reported sequence number |
| // is increasing. |
| int64_t rtcp_interval = RtcpReportInterval(); |
| if (rtcp_receiver_.RtcpRrTimeout(rtcp_interval)) { |
| RTC_LOG_F(LS_WARNING) << "Timeout: No RTCP RR received."; |
| } else if (rtcp_receiver_.RtcpRrSequenceNumberTimeout(rtcp_interval)) { |
| RTC_LOG_F(LS_WARNING) << "Timeout: No increase in RTCP RR extended " |
| "highest sequence number."; |
| } |
| |
| if (remote_bitrate_ && rtcp_sender_.TMMBR()) { |
| unsigned int target_bitrate = 0; |
| std::vector<unsigned int> ssrcs; |
| if (remote_bitrate_->LatestEstimate(&ssrcs, &target_bitrate)) { |
| if (!ssrcs.empty()) { |
| target_bitrate = target_bitrate / ssrcs.size(); |
| } |
| rtcp_sender_.SetTargetBitrate(target_bitrate); |
| } |
| } |
| } else { |
| // Report rtt from receiver. |
| if (process_rtt) { |
| int64_t rtt_ms; |
| if (rtt_stats_ && rtcp_receiver_.GetAndResetXrRrRtt(&rtt_ms)) { |
| rtt_stats_->OnRttUpdate(rtt_ms); |
| } |
| } |
| } |
| |
| // Get processed rtt. |
| if (process_rtt) { |
| last_rtt_process_time_ = now; |
| next_process_time_ = std::min( |
| next_process_time_, last_rtt_process_time_ + kRtpRtcpRttProcessTimeMs); |
| if (rtt_stats_) { |
| // Make sure we have a valid RTT before setting. |
| int64_t last_rtt = rtt_stats_->LastProcessedRtt(); |
| if (last_rtt >= 0) |
| set_rtt_ms(last_rtt); |
| } |
| } |
| |
| if (rtcp_sender_.TimeToSendRTCPReport()) |
| rtcp_sender_.SendRTCP(GetFeedbackState(), kRtcpReport); |
| |
| if (TMMBR() && rtcp_receiver_.UpdateTmmbrTimers()) { |
| rtcp_receiver_.NotifyTmmbrUpdated(); |
| } |
| } |
| |
| void ModuleRtpRtcpImpl::SetRtxSendStatus(int mode) { |
| rtp_sender_->SetRtxStatus(mode); |
| } |
| |
| int ModuleRtpRtcpImpl::RtxSendStatus() const { |
| return rtp_sender_ ? rtp_sender_->RtxStatus() : kRtxOff; |
| } |
| |
| void ModuleRtpRtcpImpl::SetRtxSsrc(uint32_t ssrc) { |
| rtp_sender_->SetRtxSsrc(ssrc); |
| } |
| |
| void ModuleRtpRtcpImpl::SetRtxSendPayloadType(int payload_type, |
| int associated_payload_type) { |
| rtp_sender_->SetRtxPayloadType(payload_type, associated_payload_type); |
| } |
| |
| absl::optional<uint32_t> ModuleRtpRtcpImpl::FlexfecSsrc() const { |
| if (rtp_sender_) |
| return rtp_sender_->FlexfecSsrc(); |
| return absl::nullopt; |
| } |
| |
| void ModuleRtpRtcpImpl::IncomingRtcpPacket(const uint8_t* rtcp_packet, |
| const size_t length) { |
| rtcp_receiver_.IncomingPacket(rtcp_packet, length); |
| } |
| |
| int32_t ModuleRtpRtcpImpl::RegisterSendPayload( |
| const CodecInst& voice_codec) { |
| return rtp_sender_->RegisterPayload( |
| voice_codec.plname, voice_codec.pltype, voice_codec.plfreq, |
| voice_codec.channels, (voice_codec.rate < 0) ? 0 : voice_codec.rate); |
| } |
| |
| void ModuleRtpRtcpImpl::RegisterVideoSendPayload(int payload_type, |
| const char* payload_name) { |
| RTC_CHECK_EQ( |
| 0, rtp_sender_->RegisterPayload(payload_name, payload_type, 90000, 0, 0)); |
| } |
| |
| int32_t ModuleRtpRtcpImpl::DeRegisterSendPayload(const int8_t payload_type) { |
| return rtp_sender_->DeRegisterSendPayload(payload_type); |
| } |
| |
| uint32_t ModuleRtpRtcpImpl::StartTimestamp() const { |
| return rtp_sender_->TimestampOffset(); |
| } |
| |
| // Configure start timestamp, default is a random number. |
| void ModuleRtpRtcpImpl::SetStartTimestamp(const uint32_t timestamp) { |
| rtcp_sender_.SetTimestampOffset(timestamp); |
| rtp_sender_->SetTimestampOffset(timestamp); |
| } |
| |
| uint16_t ModuleRtpRtcpImpl::SequenceNumber() const { |
| return rtp_sender_->SequenceNumber(); |
| } |
| |
| // Set SequenceNumber, default is a random number. |
| void ModuleRtpRtcpImpl::SetSequenceNumber(const uint16_t seq_num) { |
| rtp_sender_->SetSequenceNumber(seq_num); |
| } |
| |
| void ModuleRtpRtcpImpl::SetRtpState(const RtpState& rtp_state) { |
| rtp_sender_->SetRtpState(rtp_state); |
| rtcp_sender_.SetTimestampOffset(rtp_state.start_timestamp); |
| } |
| |
| void ModuleRtpRtcpImpl::SetRtxState(const RtpState& rtp_state) { |
| rtp_sender_->SetRtxRtpState(rtp_state); |
| } |
| |
| RtpState ModuleRtpRtcpImpl::GetRtpState() const { |
| return rtp_sender_->GetRtpState(); |
| } |
| |
| RtpState ModuleRtpRtcpImpl::GetRtxState() const { |
| return rtp_sender_->GetRtxRtpState(); |
| } |
| |
| uint32_t ModuleRtpRtcpImpl::SSRC() const { |
| return rtcp_sender_.SSRC(); |
| } |
| |
| void ModuleRtpRtcpImpl::SetSSRC(const uint32_t ssrc) { |
| if (rtp_sender_) { |
| rtp_sender_->SetSSRC(ssrc); |
| } |
| rtcp_sender_.SetSSRC(ssrc); |
| SetRtcpReceiverSsrcs(ssrc); |
| } |
| |
| void ModuleRtpRtcpImpl::SetMid(const std::string& mid) { |
| if (rtp_sender_) { |
| rtp_sender_->SetMid(mid); |
| } |
| // TODO(bugs.webrtc.org/4050): If we end up supporting the MID SDES item for |
| // RTCP, this will need to be passed down to the RTCPSender also. |
| } |
| |
| void ModuleRtpRtcpImpl::SetCsrcs(const std::vector<uint32_t>& csrcs) { |
| rtcp_sender_.SetCsrcs(csrcs); |
| rtp_sender_->SetCsrcs(csrcs); |
| } |
| |
| // TODO(pbos): Handle media and RTX streams separately (separate RTCP |
| // feedbacks). |
| RTCPSender::FeedbackState ModuleRtpRtcpImpl::GetFeedbackState() { |
| RTCPSender::FeedbackState state; |
| // This is called also when receiver_only is true. Hence below |
| // checks that rtp_sender_ exists. |
| if (rtp_sender_) { |
| StreamDataCounters rtp_stats; |
| StreamDataCounters rtx_stats; |
| rtp_sender_->GetDataCounters(&rtp_stats, &rtx_stats); |
| state.packets_sent = rtp_stats.transmitted.packets + |
| rtx_stats.transmitted.packets; |
| state.media_bytes_sent = rtp_stats.transmitted.payload_bytes + |
| rtx_stats.transmitted.payload_bytes; |
| state.send_bitrate = rtp_sender_->BitrateSent(); |
| } |
| state.module = this; |
| |
| LastReceivedNTP(&state.last_rr_ntp_secs, |
| &state.last_rr_ntp_frac, |
| &state.remote_sr); |
| |
| state.last_xr_rtis = rtcp_receiver_.ConsumeReceivedXrReferenceTimeInfo(); |
| |
| return state; |
| } |
| |
| // TODO(nisse): This method shouldn't be called for a receive-only |
| // stream. Delete rtp_sender_ check as soon as all applications are |
| // updated. |
| int32_t ModuleRtpRtcpImpl::SetSendingStatus(const bool sending) { |
| if (rtcp_sender_.Sending() != sending) { |
| // Sends RTCP BYE when going from true to false |
| if (rtcp_sender_.SetSendingStatus(GetFeedbackState(), sending) != 0) { |
| RTC_LOG(LS_WARNING) << "Failed to send RTCP BYE"; |
| } |
| if (sending && rtp_sender_) { |
| // Update Rtcp receiver config, to track Rtx config changes from |
| // the SetRtxStatus and SetRtxSsrc methods. |
| SetRtcpReceiverSsrcs(rtp_sender_->SSRC()); |
| } |
| } |
| return 0; |
| } |
| |
| bool ModuleRtpRtcpImpl::Sending() const { |
| return rtcp_sender_.Sending(); |
| } |
| |
| // TODO(nisse): This method shouldn't be called for a receive-only |
| // stream. Delete rtp_sender_ check as soon as all applications are |
| // updated. |
| void ModuleRtpRtcpImpl::SetSendingMediaStatus(const bool sending) { |
| if (rtp_sender_) { |
| rtp_sender_->SetSendingMediaStatus(sending); |
| } else { |
| RTC_DCHECK(!sending); |
| } |
| } |
| |
| bool ModuleRtpRtcpImpl::SendingMedia() const { |
| return rtp_sender_ ? rtp_sender_->SendingMedia() : false; |
| } |
| |
| bool ModuleRtpRtcpImpl::SendOutgoingData( |
| FrameType frame_type, |
| int8_t payload_type, |
| uint32_t time_stamp, |
| int64_t capture_time_ms, |
| const uint8_t* payload_data, |
| size_t payload_size, |
| const RTPFragmentationHeader* fragmentation, |
| const RTPVideoHeader* rtp_video_header, |
| uint32_t* transport_frame_id_out) { |
| rtcp_sender_.SetLastRtpTime(time_stamp, capture_time_ms); |
| // Make sure an RTCP report isn't queued behind a key frame. |
| if (rtcp_sender_.TimeToSendRTCPReport(kVideoFrameKey == frame_type)) { |
| rtcp_sender_.SendRTCP(GetFeedbackState(), kRtcpReport); |
| } |
| int64_t expected_retransmission_time_ms = rtt_ms(); |
| if (expected_retransmission_time_ms == 0) { |
| // No rtt available (|kRtpRtcpRttProcessTimeMs| not yet passed?), so try to |
| // poll avg_rtt_ms directly from rtcp receiver. |
| if (rtcp_receiver_.RTT(rtcp_receiver_.RemoteSSRC(), nullptr, |
| &expected_retransmission_time_ms, nullptr, |
| nullptr) == -1) { |
| expected_retransmission_time_ms = kDefaultExpectedRetransmissionTimeMs; |
| } |
| } |
| return rtp_sender_->SendOutgoingData( |
| frame_type, payload_type, time_stamp, capture_time_ms, payload_data, |
| payload_size, fragmentation, rtp_video_header, transport_frame_id_out, |
| expected_retransmission_time_ms); |
| } |
| |
| bool ModuleRtpRtcpImpl::TimeToSendPacket(uint32_t ssrc, |
| uint16_t sequence_number, |
| int64_t capture_time_ms, |
| bool retransmission, |
| const PacedPacketInfo& pacing_info) { |
| return rtp_sender_->TimeToSendPacket(ssrc, sequence_number, capture_time_ms, |
| retransmission, pacing_info); |
| } |
| |
| size_t ModuleRtpRtcpImpl::TimeToSendPadding( |
| size_t bytes, |
| const PacedPacketInfo& pacing_info) { |
| return rtp_sender_->TimeToSendPadding(bytes, pacing_info); |
| } |
| |
| size_t ModuleRtpRtcpImpl::MaxRtpPacketSize() const { |
| return rtp_sender_->MaxRtpPacketSize(); |
| } |
| |
| void ModuleRtpRtcpImpl::SetMaxRtpPacketSize(size_t rtp_packet_size) { |
| RTC_DCHECK_LE(rtp_packet_size, IP_PACKET_SIZE) |
| << "rtp packet size too large: " << rtp_packet_size; |
| RTC_DCHECK_GT(rtp_packet_size, packet_overhead_) |
| << "rtp packet size too small: " << rtp_packet_size; |
| |
| rtcp_sender_.SetMaxRtpPacketSize(rtp_packet_size); |
| if (rtp_sender_) |
| rtp_sender_->SetMaxRtpPacketSize(rtp_packet_size); |
| } |
| |
| RtcpMode ModuleRtpRtcpImpl::RTCP() const { |
| return rtcp_sender_.Status(); |
| } |
| |
| // Configure RTCP status i.e on/off. |
| void ModuleRtpRtcpImpl::SetRTCPStatus(const RtcpMode method) { |
| rtcp_sender_.SetRTCPStatus(method); |
| } |
| |
| int32_t ModuleRtpRtcpImpl::SetCNAME(const char* c_name) { |
| return rtcp_sender_.SetCNAME(c_name); |
| } |
| |
| int32_t ModuleRtpRtcpImpl::AddMixedCNAME(uint32_t ssrc, const char* c_name) { |
| return rtcp_sender_.AddMixedCNAME(ssrc, c_name); |
| } |
| |
| int32_t ModuleRtpRtcpImpl::RemoveMixedCNAME(const uint32_t ssrc) { |
| return rtcp_sender_.RemoveMixedCNAME(ssrc); |
| } |
| |
| int32_t ModuleRtpRtcpImpl::RemoteCNAME( |
| const uint32_t remote_ssrc, |
| char c_name[RTCP_CNAME_SIZE]) const { |
| return rtcp_receiver_.CNAME(remote_ssrc, c_name); |
| } |
| |
| int32_t ModuleRtpRtcpImpl::RemoteNTP( |
| uint32_t* received_ntpsecs, |
| uint32_t* received_ntpfrac, |
| uint32_t* rtcp_arrival_time_secs, |
| uint32_t* rtcp_arrival_time_frac, |
| uint32_t* rtcp_timestamp) const { |
| return rtcp_receiver_.NTP(received_ntpsecs, |
| received_ntpfrac, |
| rtcp_arrival_time_secs, |
| rtcp_arrival_time_frac, |
| rtcp_timestamp) |
| ? 0 |
| : -1; |
| } |
| |
| // Get RoundTripTime. |
| int32_t ModuleRtpRtcpImpl::RTT(const uint32_t remote_ssrc, |
| int64_t* rtt, |
| int64_t* avg_rtt, |
| int64_t* min_rtt, |
| int64_t* max_rtt) const { |
| int32_t ret = rtcp_receiver_.RTT(remote_ssrc, rtt, avg_rtt, min_rtt, max_rtt); |
| if (rtt && *rtt == 0) { |
| // Try to get RTT from RtcpRttStats class. |
| *rtt = rtt_ms(); |
| } |
| return ret; |
| } |
| |
| // Force a send of an RTCP packet. |
| // Normal SR and RR are triggered via the process function. |
| int32_t ModuleRtpRtcpImpl::SendRTCP(RTCPPacketType packet_type) { |
| return rtcp_sender_.SendRTCP(GetFeedbackState(), packet_type); |
| } |
| |
| // Force a send of an RTCP packet. |
| // Normal SR and RR are triggered via the process function. |
| int32_t ModuleRtpRtcpImpl::SendCompoundRTCP( |
| const std::set<RTCPPacketType>& packet_types) { |
| return rtcp_sender_.SendCompoundRTCP(GetFeedbackState(), packet_types); |
| } |
| |
| int32_t ModuleRtpRtcpImpl::SetRTCPApplicationSpecificData( |
| const uint8_t sub_type, |
| const uint32_t name, |
| const uint8_t* data, |
| const uint16_t length) { |
| return rtcp_sender_.SetApplicationSpecificData(sub_type, name, data, length); |
| } |
| |
| // (XR) VOIP metric. |
| int32_t ModuleRtpRtcpImpl::SetRTCPVoIPMetrics( |
| const RTCPVoIPMetric* voip_metric) { |
| return rtcp_sender_.SetRTCPVoIPMetrics(voip_metric); |
| } |
| |
| void ModuleRtpRtcpImpl::SetRtcpXrRrtrStatus(bool enable) { |
| rtcp_receiver_.SetRtcpXrRrtrStatus(enable); |
| rtcp_sender_.SendRtcpXrReceiverReferenceTime(enable); |
| } |
| |
| bool ModuleRtpRtcpImpl::RtcpXrRrtrStatus() const { |
| return rtcp_sender_.RtcpXrReceiverReferenceTime(); |
| } |
| |
| // TODO(asapersson): Replace this method with the one below. |
| int32_t ModuleRtpRtcpImpl::DataCountersRTP( |
| size_t* bytes_sent, |
| uint32_t* packets_sent) const { |
| StreamDataCounters rtp_stats; |
| StreamDataCounters rtx_stats; |
| rtp_sender_->GetDataCounters(&rtp_stats, &rtx_stats); |
| |
| if (bytes_sent) { |
| *bytes_sent = rtp_stats.transmitted.payload_bytes + |
| rtp_stats.transmitted.padding_bytes + |
| rtp_stats.transmitted.header_bytes + |
| rtx_stats.transmitted.payload_bytes + |
| rtx_stats.transmitted.padding_bytes + |
| rtx_stats.transmitted.header_bytes; |
| } |
| if (packets_sent) { |
| *packets_sent = rtp_stats.transmitted.packets + |
| rtx_stats.transmitted.packets; |
| } |
| return 0; |
| } |
| |
| void ModuleRtpRtcpImpl::GetSendStreamDataCounters( |
| StreamDataCounters* rtp_counters, |
| StreamDataCounters* rtx_counters) const { |
| rtp_sender_->GetDataCounters(rtp_counters, rtx_counters); |
| } |
| |
| void ModuleRtpRtcpImpl::GetRtpPacketLossStats( |
| bool outgoing, |
| uint32_t ssrc, |
| struct RtpPacketLossStats* loss_stats) const { |
| if (!loss_stats) return; |
| const PacketLossStats* stats_source = NULL; |
| if (outgoing) { |
| if (SSRC() == ssrc) { |
| stats_source = &send_loss_stats_; |
| } |
| } else { |
| if (rtcp_receiver_.RemoteSSRC() == ssrc) { |
| stats_source = &receive_loss_stats_; |
| } |
| } |
| if (stats_source) { |
| loss_stats->single_packet_loss_count = |
| stats_source->GetSingleLossCount(); |
| loss_stats->multiple_packet_loss_event_count = |
| stats_source->GetMultipleLossEventCount(); |
| loss_stats->multiple_packet_loss_packet_count = |
| stats_source->GetMultipleLossPacketCount(); |
| } |
| } |
| |
| // Received RTCP report. |
| int32_t ModuleRtpRtcpImpl::RemoteRTCPStat( |
| std::vector<RTCPReportBlock>* receive_blocks) const { |
| return rtcp_receiver_.StatisticsReceived(receive_blocks); |
| } |
| |
| // (REMB) Receiver Estimated Max Bitrate. |
| void ModuleRtpRtcpImpl::SetRemb(int64_t bitrate_bps, |
| std::vector<uint32_t> ssrcs) { |
| rtcp_sender_.SetRemb(bitrate_bps, std::move(ssrcs)); |
| } |
| |
| void ModuleRtpRtcpImpl::UnsetRemb() { |
| rtcp_sender_.UnsetRemb(); |
| } |
| |
| int32_t ModuleRtpRtcpImpl::RegisterSendRtpHeaderExtension( |
| const RTPExtensionType type, |
| const uint8_t id) { |
| return rtp_sender_->RegisterRtpHeaderExtension(type, id); |
| } |
| |
| int32_t ModuleRtpRtcpImpl::DeregisterSendRtpHeaderExtension( |
| const RTPExtensionType type) { |
| return rtp_sender_->DeregisterRtpHeaderExtension(type); |
| } |
| |
| bool ModuleRtpRtcpImpl::HasBweExtensions() const { |
| return rtp_sender_->IsRtpHeaderExtensionRegistered( |
| kRtpExtensionTransportSequenceNumber) || |
| rtp_sender_->IsRtpHeaderExtensionRegistered( |
| kRtpExtensionAbsoluteSendTime) || |
| rtp_sender_->IsRtpHeaderExtensionRegistered( |
| kRtpExtensionTransmissionTimeOffset); |
| } |
| |
| // (TMMBR) Temporary Max Media Bit Rate. |
| bool ModuleRtpRtcpImpl::TMMBR() const { |
| return rtcp_sender_.TMMBR(); |
| } |
| |
| void ModuleRtpRtcpImpl::SetTMMBRStatus(const bool enable) { |
| rtcp_sender_.SetTMMBRStatus(enable); |
| } |
| |
| void ModuleRtpRtcpImpl::SetTmmbn(std::vector<rtcp::TmmbItem> bounding_set) { |
| rtcp_sender_.SetTmmbn(std::move(bounding_set)); |
| } |
| |
| // Returns the currently configured retransmission mode. |
| int ModuleRtpRtcpImpl::SelectiveRetransmissions() const { |
| return rtp_sender_->SelectiveRetransmissions(); |
| } |
| |
| // Enable or disable a retransmission mode, which decides which packets will |
| // be retransmitted if NACKed. |
| int ModuleRtpRtcpImpl::SetSelectiveRetransmissions(uint8_t settings) { |
| return rtp_sender_->SetSelectiveRetransmissions(settings); |
| } |
| |
| // Send a Negative acknowledgment packet. |
| int32_t ModuleRtpRtcpImpl::SendNACK(const uint16_t* nack_list, |
| const uint16_t size) { |
| for (int i = 0; i < size; ++i) { |
| receive_loss_stats_.AddLostPacket(nack_list[i]); |
| } |
| uint16_t nack_length = size; |
| uint16_t start_id = 0; |
| int64_t now = clock_->TimeInMilliseconds(); |
| if (TimeToSendFullNackList(now)) { |
| nack_last_time_sent_full_ = now; |
| nack_last_time_sent_full_prev_ = now; |
| } else { |
| // Only send extended list. |
| if (nack_last_seq_number_sent_ == nack_list[size - 1]) { |
| // Last sequence number is the same, do not send list. |
| return 0; |
| } |
| // Send new sequence numbers. |
| for (int i = 0; i < size; ++i) { |
| if (nack_last_seq_number_sent_ == nack_list[i]) { |
| start_id = i + 1; |
| break; |
| } |
| } |
| nack_length = size - start_id; |
| } |
| |
| // Our RTCP NACK implementation is limited to kRtcpMaxNackFields sequence |
| // numbers per RTCP packet. |
| if (nack_length > kRtcpMaxNackFields) { |
| nack_length = kRtcpMaxNackFields; |
| } |
| nack_last_seq_number_sent_ = nack_list[start_id + nack_length - 1]; |
| |
| return rtcp_sender_.SendRTCP(GetFeedbackState(), kRtcpNack, nack_length, |
| &nack_list[start_id]); |
| } |
| |
| void ModuleRtpRtcpImpl::SendNack( |
| const std::vector<uint16_t>& sequence_numbers) { |
| rtcp_sender_.SendRTCP(GetFeedbackState(), kRtcpNack, sequence_numbers.size(), |
| sequence_numbers.data()); |
| } |
| |
| bool ModuleRtpRtcpImpl::TimeToSendFullNackList(int64_t now) const { |
| // Use RTT from RtcpRttStats class if provided. |
| int64_t rtt = rtt_ms(); |
| if (rtt == 0) { |
| rtcp_receiver_.RTT(rtcp_receiver_.RemoteSSRC(), NULL, &rtt, NULL, NULL); |
| } |
| |
| const int64_t kStartUpRttMs = 100; |
| int64_t wait_time = 5 + ((rtt * 3) >> 1); // 5 + RTT * 1.5. |
| if (rtt == 0) { |
| wait_time = kStartUpRttMs; |
| } |
| |
| // Send a full NACK list once within every |wait_time|. |
| if (rtt_stats_) { |
| return now - nack_last_time_sent_full_ > wait_time; |
| } |
| return now - nack_last_time_sent_full_prev_ > wait_time; |
| } |
| |
| // Store the sent packets, needed to answer to Negative acknowledgment requests. |
| void ModuleRtpRtcpImpl::SetStorePacketsStatus(const bool enable, |
| const uint16_t number_to_store) { |
| rtp_sender_->SetStorePacketsStatus(enable, number_to_store); |
| } |
| |
| bool ModuleRtpRtcpImpl::StorePackets() const { |
| return rtp_sender_->StorePackets(); |
| } |
| |
| void ModuleRtpRtcpImpl::RegisterRtcpStatisticsCallback( |
| RtcpStatisticsCallback* callback) { |
| rtcp_receiver_.RegisterRtcpStatisticsCallback(callback); |
| } |
| |
| RtcpStatisticsCallback* ModuleRtpRtcpImpl::GetRtcpStatisticsCallback() { |
| return rtcp_receiver_.GetRtcpStatisticsCallback(); |
| } |
| |
| bool ModuleRtpRtcpImpl::SendFeedbackPacket( |
| const rtcp::TransportFeedback& packet) { |
| return rtcp_sender_.SendFeedbackPacket(packet); |
| } |
| |
| // Send a TelephoneEvent tone using RFC 2833 (4733). |
| int32_t ModuleRtpRtcpImpl::SendTelephoneEventOutband( |
| const uint8_t key, |
| const uint16_t time_ms, |
| const uint8_t level) { |
| return rtp_sender_->SendTelephoneEvent(key, time_ms, level); |
| } |
| |
| int32_t ModuleRtpRtcpImpl::SetAudioLevel( |
| const uint8_t level_d_bov) { |
| return rtp_sender_->SetAudioLevel(level_d_bov); |
| } |
| |
| int32_t ModuleRtpRtcpImpl::SetKeyFrameRequestMethod( |
| const KeyFrameRequestMethod method) { |
| key_frame_req_method_ = method; |
| return 0; |
| } |
| |
| int32_t ModuleRtpRtcpImpl::RequestKeyFrame() { |
| switch (key_frame_req_method_) { |
| case kKeyFrameReqPliRtcp: |
| return SendRTCP(kRtcpPli); |
| case kKeyFrameReqFirRtcp: |
| return SendRTCP(kRtcpFir); |
| } |
| return -1; |
| } |
| |
| void ModuleRtpRtcpImpl::SetUlpfecConfig(int red_payload_type, |
| int ulpfec_payload_type) { |
| rtp_sender_->SetUlpfecConfig(red_payload_type, ulpfec_payload_type); |
| } |
| |
| bool ModuleRtpRtcpImpl::SetFecParameters( |
| const FecProtectionParams& delta_params, |
| const FecProtectionParams& key_params) { |
| return rtp_sender_->SetFecParameters(delta_params, key_params); |
| } |
| |
| void ModuleRtpRtcpImpl::SetRemoteSSRC(const uint32_t ssrc) { |
| // Inform about the incoming SSRC. |
| rtcp_sender_.SetRemoteSSRC(ssrc); |
| rtcp_receiver_.SetRemoteSSRC(ssrc); |
| } |
| |
| void ModuleRtpRtcpImpl::BitrateSent(uint32_t* total_rate, |
| uint32_t* video_rate, |
| uint32_t* fec_rate, |
| uint32_t* nack_rate) const { |
| *total_rate = rtp_sender_->BitrateSent(); |
| *video_rate = rtp_sender_->VideoBitrateSent(); |
| *fec_rate = rtp_sender_->FecOverheadRate(); |
| *nack_rate = rtp_sender_->NackOverheadRate(); |
| } |
| |
| void ModuleRtpRtcpImpl::OnRequestSendReport() { |
| SendRTCP(kRtcpSr); |
| } |
| |
| void ModuleRtpRtcpImpl::OnReceivedNack( |
| const std::vector<uint16_t>& nack_sequence_numbers) { |
| if (!rtp_sender_) |
| return; |
| |
| for (uint16_t nack_sequence_number : nack_sequence_numbers) { |
| send_loss_stats_.AddLostPacket(nack_sequence_number); |
| } |
| if (!rtp_sender_->StorePackets() || |
| nack_sequence_numbers.size() == 0) { |
| return; |
| } |
| // Use RTT from RtcpRttStats class if provided. |
| int64_t rtt = rtt_ms(); |
| if (rtt == 0) { |
| rtcp_receiver_.RTT(rtcp_receiver_.RemoteSSRC(), NULL, &rtt, NULL, NULL); |
| } |
| rtp_sender_->OnReceivedNack(nack_sequence_numbers, rtt); |
| } |
| |
| void ModuleRtpRtcpImpl::OnReceivedRtcpReportBlocks( |
| const ReportBlockList& report_blocks) { |
| if (rtp_sender_) |
| rtp_sender_->OnReceivedRtcpReportBlocks(report_blocks); |
| } |
| |
| bool ModuleRtpRtcpImpl::LastReceivedNTP( |
| uint32_t* rtcp_arrival_time_secs, // When we got the last report. |
| uint32_t* rtcp_arrival_time_frac, |
| uint32_t* remote_sr) const { |
| // Remote SR: NTP inside the last received (mid 16 bits from sec and frac). |
| uint32_t ntp_secs = 0; |
| uint32_t ntp_frac = 0; |
| |
| if (!rtcp_receiver_.NTP(&ntp_secs, |
| &ntp_frac, |
| rtcp_arrival_time_secs, |
| rtcp_arrival_time_frac, |
| NULL)) { |
| return false; |
| } |
| *remote_sr = |
| ((ntp_secs & 0x0000ffff) << 16) + ((ntp_frac & 0xffff0000) >> 16); |
| return true; |
| } |
| |
| // Called from RTCPsender. |
| std::vector<rtcp::TmmbItem> ModuleRtpRtcpImpl::BoundingSet(bool* tmmbr_owner) { |
| return rtcp_receiver_.BoundingSet(tmmbr_owner); |
| } |
| |
| int64_t ModuleRtpRtcpImpl::RtcpReportInterval() { |
| if (audio_) |
| return rtcp_sender_.RtcpAudioReportInverval(); |
| else |
| return rtcp_sender_.RtcpVideoReportInverval(); |
| } |
| |
| void ModuleRtpRtcpImpl::SetRtcpReceiverSsrcs(uint32_t main_ssrc) { |
| std::set<uint32_t> ssrcs; |
| ssrcs.insert(main_ssrc); |
| if (RtxSendStatus() != kRtxOff) |
| ssrcs.insert(rtp_sender_->RtxSsrc()); |
| absl::optional<uint32_t> flexfec_ssrc = FlexfecSsrc(); |
| if (flexfec_ssrc) |
| ssrcs.insert(*flexfec_ssrc); |
| rtcp_receiver_.SetSsrcs(main_ssrc, ssrcs); |
| } |
| |
| void ModuleRtpRtcpImpl::set_rtt_ms(int64_t rtt_ms) { |
| rtc::CritScope cs(&critical_section_rtt_); |
| rtt_ms_ = rtt_ms; |
| if (rtp_sender_) |
| rtp_sender_->SetRtt(rtt_ms); |
| } |
| |
| int64_t ModuleRtpRtcpImpl::rtt_ms() const { |
| rtc::CritScope cs(&critical_section_rtt_); |
| return rtt_ms_; |
| } |
| |
| void ModuleRtpRtcpImpl::RegisterSendChannelRtpStatisticsCallback( |
| StreamDataCountersCallback* callback) { |
| rtp_sender_->RegisterRtpStatisticsCallback(callback); |
| } |
| |
| StreamDataCountersCallback* |
| ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const { |
| return rtp_sender_->GetRtpStatisticsCallback(); |
| } |
| |
| void ModuleRtpRtcpImpl::SetVideoBitrateAllocation( |
| const VideoBitrateAllocation& bitrate) { |
| rtcp_sender_.SetVideoBitrateAllocation(bitrate); |
| } |
| } // namespace webrtc |
