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Inverse of H264RtpPacketizer. Takes incoming H264 packets and emits H264 NALUs. Co-authored-by: Paul-Louis Ageneau <[email protected]>
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/** | ||
* Copyright (c) 2020 Staz Modrzynski | ||
* Copyright (c) 2020 Paul-Louis Ageneau | ||
* | ||
* This Source Code Form is subject to the terms of the Mozilla Public | ||
* License, v. 2.0. If a copy of the MPL was not distributed with this | ||
* file, You can obtain one at https://mozilla.org/MPL/2.0/. | ||
*/ | ||
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#ifndef RTC_H264_RTP_DEPACKETIZER_H | ||
#define RTC_H264_RTP_DEPACKETIZER_H | ||
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#if RTC_ENABLE_MEDIA | ||
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#include "common.hpp" | ||
#include "mediahandler.hpp" | ||
#include "message.hpp" | ||
#include "rtp.hpp" | ||
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#include <iterator> | ||
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namespace rtc { | ||
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/// RTP depacketization for H264 | ||
class RTC_CPP_EXPORT H264RtpDepacketizer : public MediaHandler { | ||
public: | ||
H264RtpDepacketizer() = default; | ||
virtual ~H264RtpDepacketizer() = default; | ||
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void incoming(message_vector &messages, const message_callback &send) override; | ||
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private: | ||
std::vector<message_ptr> mRtpBuffer; | ||
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message_vector buildFrame(message_vector::iterator firstPkt, message_vector::iterator lastPkt); | ||
}; | ||
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} // namespace rtc | ||
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#endif // RTC_ENABLE_MEDIA | ||
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#endif /* RTC_H264_RTP_DEPACKETIZER_H */ |
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/** | ||
* Copyright (c) 2023 Paul-Louis Ageneau | ||
* | ||
* This Source Code Form is subject to the terms of the Mozilla Public | ||
* License, v. 2.0. If a copy of the MPL was not distributed with this | ||
* file, You can obtain one at https://mozilla.org/MPL/2.0/. | ||
*/ | ||
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#if RTC_ENABLE_MEDIA | ||
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#include "h264rtpdepacketizer.hpp" | ||
#include "track.hpp" | ||
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#include "impl/logcounter.hpp" | ||
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#include <cmath> | ||
#include <utility> | ||
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#ifdef _WIN32 | ||
#include <winsock2.h> | ||
#else | ||
#include <arpa/inet.h> | ||
#endif | ||
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namespace rtc { | ||
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const unsigned long stapaHeaderSize = 1; | ||
const auto fuaHeaderSize = 2; | ||
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const uint8_t naluTypeBitmask = 0x1F; | ||
const uint8_t naluTypeSTAPA = 24; | ||
const uint8_t naluTypeFUA = 28; | ||
const uint8_t fuaEndBitmask = 0x40; | ||
const uint8_t naluRefIdcBitmask = 0x60; | ||
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message_vector H264RtpDepacketizer::buildFrame(message_vector::iterator first, | ||
message_vector::iterator last) { | ||
message_vector out = {}; | ||
auto fua_buffer = std::vector<std::byte>{}; | ||
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for (auto it = first; (it - 1) != last; it++) { | ||
auto pkt = it->get(); | ||
auto pktParsed = reinterpret_cast<const rtc::RtpHeader *>(pkt->data()); | ||
auto headerSize = | ||
sizeof(rtc::RtpHeader) + pktParsed->csrcCount() + pktParsed->getExtensionHeaderSize(); | ||
auto firstByte = std::to_integer<uint8_t>(pkt->at(headerSize)); | ||
auto secondByte = std::to_integer<uint8_t>(pkt->at(headerSize + 1)); | ||
auto naluType = firstByte & naluTypeBitmask; | ||
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if (fua_buffer.size() != 0 || naluType == naluTypeFUA) { | ||
if (fua_buffer.size() == 0) { | ||
fua_buffer.push_back(std::byte(0)); | ||
} | ||
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std::copy(pkt->begin() + headerSize + fuaHeaderSize, pkt->end(), | ||
std::back_inserter(fua_buffer)); | ||
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if ((secondByte & fuaEndBitmask) != 0) { | ||
auto naluRefIdc = firstByte & naluRefIdcBitmask; | ||
auto fragmentedNaluType = secondByte & naluTypeBitmask; | ||
fua_buffer.at(0) = std::byte(naluRefIdc | fragmentedNaluType); | ||
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out.push_back(make_message(std::move(fua_buffer))); | ||
fua_buffer.clear(); | ||
} | ||
} else if (naluType > 0 && naluType < 24) { | ||
out.push_back(make_message(pkt->begin() + headerSize, pkt->end())); | ||
} else if (naluType == naluTypeSTAPA) { | ||
auto currOffset = stapaHeaderSize + headerSize; | ||
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while (currOffset < pkt->size()) { | ||
auto naluSize = | ||
uint16_t(pkt->at(currOffset)) << 8 | uint8_t(pkt->at(currOffset + 1)); | ||
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currOffset += 2; | ||
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if (pkt->size() < currOffset + naluSize) { | ||
throw std::runtime_error("STAP-A declared size is larger then buffer"); | ||
} | ||
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out.push_back( | ||
make_message(pkt->begin() + currOffset, pkt->begin() + currOffset + naluSize)); | ||
currOffset += naluSize; | ||
} | ||
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} else { | ||
throw std::runtime_error("Unknown H264 RTP Packetization"); | ||
} | ||
} | ||
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return out; | ||
} | ||
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void H264RtpDepacketizer::incoming(message_vector &messages, const message_callback &) { | ||
for (auto message : messages) { | ||
if (message->type == Message::Control) { | ||
continue; // RTCP | ||
} | ||
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if (message->size() < sizeof(RtpHeader)) { | ||
PLOG_VERBOSE << "RTP packet is too small, size=" << message->size(); | ||
continue; | ||
} | ||
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mRtpBuffer.push_back(make_message(message->begin(), message->end())); | ||
} | ||
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while (mRtpBuffer.size() != 0) { | ||
uint32_t current_timestamp = 0; | ||
size_t packets_in_timestamp = 0; | ||
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for (const auto &pkt : mRtpBuffer) { | ||
auto p = reinterpret_cast<const rtc::RtpHeader *>(pkt->data()); | ||
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if (current_timestamp == 0) { | ||
current_timestamp = p->timestamp(); | ||
} else if (current_timestamp != p->timestamp()) { | ||
break; | ||
} | ||
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packets_in_timestamp++; | ||
} | ||
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if (packets_in_timestamp == mRtpBuffer.size()) { | ||
break; | ||
} | ||
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auto first = mRtpBuffer.begin(); | ||
auto last = mRtpBuffer.begin() + (packets_in_timestamp - 1); | ||
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messages = buildFrame(first, last); | ||
mRtpBuffer.erase(mRtpBuffer.begin(), mRtpBuffer.begin() + packets_in_timestamp); | ||
} | ||
} | ||
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} // namespace rtc | ||
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#endif // RTC_ENABLE_MEDIA |