/* * Copyright (c) 2014 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_format_h264.h" #include #include #include #include #include #include #include #include "absl/types/optional.h" #include "absl/types/variant.h" #include "common_video/h264/h264_common.h" #include "common_video/h264/pps_parser.h" #include "common_video/h264/sps_parser.h" #include "common_video/h264/sps_vui_rewriter.h" #include "modules/rtp_rtcp/source/byte_io.h" #include "modules/rtp_rtcp/source/rtp_packet_to_send.h" #include "rtc_base/checks.h" #include "rtc_base/logging.h" namespace webrtc { namespace { static const size_t kNalHeaderSize = 1; static const size_t kFuAHeaderSize = 2; static const size_t kLengthFieldSize = 2; // Bit masks for FU (A and B) indicators. enum NalDefs : uint8_t { kFBit = 0x80, kNriMask = 0x60, kTypeMask = 0x1F }; // Bit masks for FU (A and B) headers. enum FuDefs : uint8_t { kSBit = 0x80, kEBit = 0x40, kRBit = 0x20 }; } // namespace RtpPacketizerH264::RtpPacketizerH264(rtc::ArrayView payload, PayloadSizeLimits limits, H264PacketizationMode packetization_mode) : limits_(limits), num_packets_left_(0) { // Guard against uninitialized memory in packetization_mode. RTC_CHECK(packetization_mode == H264PacketizationMode::NonInterleaved || packetization_mode == H264PacketizationMode::SingleNalUnit); for (const auto& nalu : H264::FindNaluIndices(payload.data(), payload.size())) { input_fragments_.push_back( payload.subview(nalu.payload_start_offset, nalu.payload_size)); } if (!GeneratePackets(packetization_mode)) { // If failed to generate all the packets, discard already generated // packets in case the caller would ignore return value and still try to // call NextPacket(). num_packets_left_ = 0; while (!packets_.empty()) { packets_.pop(); } } } RtpPacketizerH264::~RtpPacketizerH264() = default; size_t RtpPacketizerH264::NumPackets() const { return num_packets_left_; } bool RtpPacketizerH264::GeneratePackets( H264PacketizationMode packetization_mode) { for (size_t i = 0; i < input_fragments_.size();) { switch (packetization_mode) { case H264PacketizationMode::SingleNalUnit: if (!PacketizeSingleNalu(i)) return false; ++i; break; case H264PacketizationMode::NonInterleaved: int fragment_len = input_fragments_[i].size(); int single_packet_capacity = limits_.max_payload_len; if (input_fragments_.size() == 1) single_packet_capacity -= limits_.single_packet_reduction_len; else if (i == 0) single_packet_capacity -= limits_.first_packet_reduction_len; else if (i + 1 == input_fragments_.size()) single_packet_capacity -= limits_.last_packet_reduction_len; if (fragment_len > single_packet_capacity) { if (!PacketizeFuA(i)) return false; ++i; } else { i = PacketizeStapA(i); } break; } } return true; } bool RtpPacketizerH264::PacketizeFuA(size_t fragment_index) { // Fragment payload into packets (FU-A). rtc::ArrayView fragment = input_fragments_[fragment_index]; PayloadSizeLimits limits = limits_; // Leave room for the FU-A header. limits.max_payload_len -= kFuAHeaderSize; // Update single/first/last packet reductions unless it is single/first/last // fragment. if (input_fragments_.size() != 1) { // if this fragment is put into a single packet, it might still be the // first or the last packet in the whole sequence of packets. if (fragment_index == input_fragments_.size() - 1) { limits.single_packet_reduction_len = limits_.last_packet_reduction_len; } else if (fragment_index == 0) { limits.single_packet_reduction_len = limits_.first_packet_reduction_len; } else { limits.single_packet_reduction_len = 0; } } if (fragment_index != 0) limits.first_packet_reduction_len = 0; if (fragment_index != input_fragments_.size() - 1) limits.last_packet_reduction_len = 0; // Strip out the original header. size_t payload_left = fragment.size() - kNalHeaderSize; int offset = kNalHeaderSize; std::vector payload_sizes = SplitAboutEqually(payload_left, limits); if (payload_sizes.empty()) return false; for (size_t i = 0; i < payload_sizes.size(); ++i) { int packet_length = payload_sizes[i]; RTC_CHECK_GT(packet_length, 0); packets_.push(PacketUnit(fragment.subview(offset, packet_length), /*first_fragment=*/i == 0, /*last_fragment=*/i == payload_sizes.size() - 1, false, fragment[0])); offset += packet_length; payload_left -= packet_length; } num_packets_left_ += payload_sizes.size(); RTC_CHECK_EQ(0, payload_left); return true; } size_t RtpPacketizerH264::PacketizeStapA(size_t fragment_index) { // Aggregate fragments into one packet (STAP-A). size_t payload_size_left = limits_.max_payload_len; if (input_fragments_.size() == 1) payload_size_left -= limits_.single_packet_reduction_len; else if (fragment_index == 0) payload_size_left -= limits_.first_packet_reduction_len; int aggregated_fragments = 0; size_t fragment_headers_length = 0; rtc::ArrayView fragment = input_fragments_[fragment_index]; RTC_CHECK_GE(payload_size_left, fragment.size()); ++num_packets_left_; auto payload_size_needed = [&] { size_t fragment_size = fragment.size() + fragment_headers_length; if (input_fragments_.size() == 1) { // Single fragment, single packet, payload_size_left already adjusted // with limits_.single_packet_reduction_len. return fragment_size; } if (fragment_index == input_fragments_.size() - 1) { // Last fragment, so STAP-A might be the last packet. return fragment_size + limits_.last_packet_reduction_len; } return fragment_size; }; while (payload_size_left >= payload_size_needed()) { RTC_CHECK_GT(fragment.size(), 0); packets_.push(PacketUnit(fragment, aggregated_fragments == 0, false, true, fragment[0])); payload_size_left -= fragment.size(); payload_size_left -= fragment_headers_length; fragment_headers_length = kLengthFieldSize; // If we are going to try to aggregate more fragments into this packet // we need to add the STAP-A NALU header and a length field for the first // NALU of this packet. if (aggregated_fragments == 0) fragment_headers_length += kNalHeaderSize + kLengthFieldSize; ++aggregated_fragments; // Next fragment. ++fragment_index; if (fragment_index == input_fragments_.size()) break; fragment = input_fragments_[fragment_index]; } RTC_CHECK_GT(aggregated_fragments, 0); packets_.back().last_fragment = true; return fragment_index; } bool RtpPacketizerH264::PacketizeSingleNalu(size_t fragment_index) { // Add a single NALU to the queue, no aggregation. size_t payload_size_left = limits_.max_payload_len; if (input_fragments_.size() == 1) payload_size_left -= limits_.single_packet_reduction_len; else if (fragment_index == 0) payload_size_left -= limits_.first_packet_reduction_len; else if (fragment_index + 1 == input_fragments_.size()) payload_size_left -= limits_.last_packet_reduction_len; rtc::ArrayView fragment = input_fragments_[fragment_index]; if (payload_size_left < fragment.size()) { RTC_LOG(LS_ERROR) << "Failed to fit a fragment to packet in SingleNalu " "packetization mode. Payload size left " << payload_size_left << ", fragment length " << fragment.size() << ", packet capacity " << limits_.max_payload_len; return false; } RTC_CHECK_GT(fragment.size(), 0u); packets_.push(PacketUnit(fragment, true /* first */, true /* last */, false /* aggregated */, fragment[0])); ++num_packets_left_; return true; } bool RtpPacketizerH264::NextPacket(RtpPacketToSend* rtp_packet) { RTC_DCHECK(rtp_packet); if (packets_.empty()) { return false; } PacketUnit packet = packets_.front(); if (packet.first_fragment && packet.last_fragment) { // Single NAL unit packet. size_t bytes_to_send = packet.source_fragment.size(); uint8_t* buffer = rtp_packet->AllocatePayload(bytes_to_send); memcpy(buffer, packet.source_fragment.data(), bytes_to_send); packets_.pop(); input_fragments_.pop_front(); } else if (packet.aggregated) { NextAggregatePacket(rtp_packet); } else { NextFragmentPacket(rtp_packet); } rtp_packet->SetMarker(packets_.empty()); --num_packets_left_; return true; } void RtpPacketizerH264::NextAggregatePacket(RtpPacketToSend* rtp_packet) { // Reserve maximum available payload, set actual payload size later. size_t payload_capacity = rtp_packet->FreeCapacity(); RTC_CHECK_GE(payload_capacity, kNalHeaderSize); uint8_t* buffer = rtp_packet->AllocatePayload(payload_capacity); RTC_DCHECK(buffer); PacketUnit* packet = &packets_.front(); RTC_CHECK(packet->first_fragment); // STAP-A NALU header. buffer[0] = (packet->header & (kFBit | kNriMask)) | H264::NaluType::kStapA; size_t index = kNalHeaderSize; bool is_last_fragment = packet->last_fragment; while (packet->aggregated) { rtc::ArrayView fragment = packet->source_fragment; RTC_CHECK_LE(index + kLengthFieldSize + fragment.size(), payload_capacity); // Add NAL unit length field. ByteWriter::WriteBigEndian(&buffer[index], fragment.size()); index += kLengthFieldSize; // Add NAL unit. memcpy(&buffer[index], fragment.data(), fragment.size()); index += fragment.size(); packets_.pop(); input_fragments_.pop_front(); if (is_last_fragment) break; packet = &packets_.front(); is_last_fragment = packet->last_fragment; } RTC_CHECK(is_last_fragment); rtp_packet->SetPayloadSize(index); } void RtpPacketizerH264::NextFragmentPacket(RtpPacketToSend* rtp_packet) { PacketUnit* packet = &packets_.front(); // NAL unit fragmented over multiple packets (FU-A). // We do not send original NALU header, so it will be replaced by the // FU indicator header of the first packet. uint8_t fu_indicator = (packet->header & (kFBit | kNriMask)) | H264::NaluType::kFuA; uint8_t fu_header = 0; // S | E | R | 5 bit type. fu_header |= (packet->first_fragment ? kSBit : 0); fu_header |= (packet->last_fragment ? kEBit : 0); uint8_t type = packet->header & kTypeMask; fu_header |= type; rtc::ArrayView fragment = packet->source_fragment; uint8_t* buffer = rtp_packet->AllocatePayload(kFuAHeaderSize + fragment.size()); buffer[0] = fu_indicator; buffer[1] = fu_header; memcpy(buffer + kFuAHeaderSize, fragment.data(), fragment.size()); if (packet->last_fragment) input_fragments_.pop_front(); packets_.pop(); } } // namespace webrtc