/* * 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/ulpfec_receiver_impl.h" #include #include #include "api/scoped_refptr.h" #include "modules/rtp_rtcp/source/rtp_packet_received.h" #include "rtc_base/logging.h" #include "rtc_base/time_utils.h" namespace webrtc { std::unique_ptr UlpfecReceiver::Create( uint32_t ssrc, RecoveredPacketReceiver* callback, rtc::ArrayView extensions) { return std::make_unique(ssrc, callback, extensions); } UlpfecReceiverImpl::UlpfecReceiverImpl( uint32_t ssrc, RecoveredPacketReceiver* callback, rtc::ArrayView extensions) : ssrc_(ssrc), extensions_(extensions), recovered_packet_callback_(callback), fec_(ForwardErrorCorrection::CreateUlpfec(ssrc_)) {} UlpfecReceiverImpl::~UlpfecReceiverImpl() { RTC_DCHECK_RUN_ON(&sequence_checker_); received_packets_.clear(); fec_->ResetState(&recovered_packets_); } FecPacketCounter UlpfecReceiverImpl::GetPacketCounter() const { RTC_DCHECK_RUN_ON(&sequence_checker_); return packet_counter_; } // 0 1 2 3 // 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ // |F| block PT | timestamp offset | block length | // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ // // // RFC 2198 RTP Payload for Redundant Audio Data September 1997 // // The bits in the header are specified as follows: // // F: 1 bit First bit in header indicates whether another header block // follows. If 1 further header blocks follow, if 0 this is the // last header block. // If 0 there is only 1 byte RED header // // block PT: 7 bits RTP payload type for this block. // // timestamp offset: 14 bits Unsigned offset of timestamp of this block // relative to timestamp given in RTP header. The use of an unsigned // offset implies that redundant data must be sent after the primary // data, and is hence a time to be subtracted from the current // timestamp to determine the timestamp of the data for which this // block is the redundancy. // // block length: 10 bits Length in bytes of the corresponding data // block excluding header. bool UlpfecReceiverImpl::AddReceivedRedPacket( const RtpPacketReceived& rtp_packet, uint8_t ulpfec_payload_type) { RTC_DCHECK_RUN_ON(&sequence_checker_); // TODO(bugs.webrtc.org/11993): We get here via Call::DeliverRtp, so should be // moved to the network thread. if (rtp_packet.Ssrc() != ssrc_) { RTC_LOG(LS_WARNING) << "Received RED packet with different SSRC than expected; dropping."; return false; } if (rtp_packet.size() > IP_PACKET_SIZE) { RTC_LOG(LS_WARNING) << "Received RED packet with length exceeds maximum IP " "packet size; dropping."; return false; } static constexpr uint8_t kRedHeaderLength = 1; if (rtp_packet.payload_size() == 0) { RTC_LOG(LS_WARNING) << "Corrupt/truncated FEC packet."; return false; } // Remove RED header of incoming packet and store as a virtual RTP packet. auto received_packet = std::make_unique(); received_packet->pkt = new ForwardErrorCorrection::Packet(); // Get payload type from RED header and sequence number from RTP header. uint8_t payload_type = rtp_packet.payload()[0] & 0x7f; received_packet->is_fec = payload_type == ulpfec_payload_type; received_packet->is_recovered = rtp_packet.recovered(); received_packet->ssrc = rtp_packet.Ssrc(); received_packet->seq_num = rtp_packet.SequenceNumber(); if (rtp_packet.payload()[0] & 0x80) { // f bit set in RED header, i.e. there are more than one RED header blocks. // WebRTC never generates multiple blocks in a RED packet for FEC. RTC_LOG(LS_WARNING) << "More than 1 block in RED packet is not supported."; return false; } ++packet_counter_.num_packets; packet_counter_.num_bytes += rtp_packet.size(); if (packet_counter_.first_packet_time_ms == -1) { packet_counter_.first_packet_time_ms = rtc::TimeMillis(); } if (received_packet->is_fec) { ++packet_counter_.num_fec_packets; // everything behind the RED header received_packet->pkt->data = rtp_packet.Buffer().Slice(rtp_packet.headers_size() + kRedHeaderLength, rtp_packet.payload_size() - kRedHeaderLength); } else { received_packet->pkt->data.EnsureCapacity(rtp_packet.size() - kRedHeaderLength); // Copy RTP header. received_packet->pkt->data.SetData(rtp_packet.data(), rtp_packet.headers_size()); // Set payload type. uint8_t& payload_type_byte = received_packet->pkt->data.MutableData()[1]; payload_type_byte &= 0x80; // Reset RED payload type. payload_type_byte += payload_type; // Set media payload type. // Copy payload and padding data, after the RED header. received_packet->pkt->data.AppendData( rtp_packet.data() + rtp_packet.headers_size() + kRedHeaderLength, rtp_packet.size() - rtp_packet.headers_size() - kRedHeaderLength); } if (received_packet->pkt->data.size() > 0) { received_packets_.push_back(std::move(received_packet)); } return true; } // TODO(nisse): Drop always-zero return value. int32_t UlpfecReceiverImpl::ProcessReceivedFec() { RTC_DCHECK_RUN_ON(&sequence_checker_); // If we iterate over |received_packets_| and it contains a packet that cause // us to recurse back to this function (for example a RED packet encapsulating // a RED packet), then we will recurse forever. To avoid this we swap // |received_packets_| with an empty vector so that the next recursive call // wont iterate over the same packet again. This also solves the problem of // not modifying the vector we are currently iterating over (packets are added // in AddReceivedRedPacket). std::vector> received_packets; received_packets.swap(received_packets_); for (const auto& received_packet : received_packets) { // Send received media packet to VCM. if (!received_packet->is_fec) { ForwardErrorCorrection::Packet* packet = received_packet->pkt; recovered_packet_callback_->OnRecoveredPacket(packet->data.data(), packet->data.size()); // Create a packet with the buffer to modify it. RtpPacketReceived rtp_packet; const uint8_t* const original_data = packet->data.cdata(); if (!rtp_packet.Parse(packet->data)) { RTC_LOG(LS_WARNING) << "Corrupted media packet"; } else { rtp_packet.IdentifyExtensions(extensions_); // Reset buffer reference, so zeroing would work on a buffer with a // single reference. packet->data = rtc::CopyOnWriteBuffer(0); rtp_packet.ZeroMutableExtensions(); packet->data = rtp_packet.Buffer(); // Ensure that zeroing of extensions was done in place. RTC_DCHECK_EQ(packet->data.cdata(), original_data); } } if (!received_packet->is_recovered) { // Do not pass recovered packets to FEC. Recovered packet might have // different set of the RTP header extensions and thus different byte // representation than the original packet, That will corrupt // FEC calculation. fec_->DecodeFec(*received_packet, &recovered_packets_); } } // Send any recovered media packets to VCM. for (const auto& recovered_packet : recovered_packets_) { if (recovered_packet->returned) { // Already sent to the VCM and the jitter buffer. continue; } ForwardErrorCorrection::Packet* packet = recovered_packet->pkt; ++packet_counter_.num_recovered_packets; // Set this flag first; in case the recovered packet carries a RED // header, OnRecoveredPacket will recurse back here. recovered_packet->returned = true; recovered_packet_callback_->OnRecoveredPacket(packet->data.data(), packet->data.size()); } return 0; } } // namespace webrtc