blob: d066bafb90f43e43ba8d8ac1151d3a440311a934 [file] [log] [blame]
/*
* 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 <string.h>
#include <cstddef>
#include <cstdint>
#include <iterator>
#include <memory>
#include <utility>
#include <vector>
#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;
} // namespace
RtpPacketizerH264::RtpPacketizerH264(rtc::ArrayView<const uint8_t> 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<const uint8_t> 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<int> 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;
int aggregated_fragments = 0;
size_t fragment_headers_length = 0;
rtc::ArrayView<const uint8_t> fragment = input_fragments_[fragment_index];
RTC_CHECK_GE(payload_size_left, fragment.size());
++num_packets_left_;
const bool has_first_fragment = fragment_index == 0;
auto payload_size_needed = [&] {
size_t fragment_size = fragment.size() + fragment_headers_length;
bool has_last_fragment = fragment_index == input_fragments_.size() - 1;
if (has_first_fragment && has_last_fragment) {
return fragment_size + limits_.single_packet_reduction_len;
} else if (has_first_fragment) {
return fragment_size + limits_.first_packet_reduction_len;
} else if (has_last_fragment) {
return fragment_size + limits_.last_packet_reduction_len;
} else {
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<const uint8_t> 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 & (kH264FBit | kH264NriMask)) | H264::NaluType::kStapA;
size_t index = kNalHeaderSize;
bool is_last_fragment = packet->last_fragment;
while (packet->aggregated) {
rtc::ArrayView<const uint8_t> fragment = packet->source_fragment;
RTC_CHECK_LE(index + kLengthFieldSize + fragment.size(), payload_capacity);
// Add NAL unit length field.
ByteWriter<uint16_t>::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 & (kH264FBit | kH264NriMask)) | H264::NaluType::kFuA;
uint8_t fu_header = 0;
// S | E | R | 5 bit type.
fu_header |= (packet->first_fragment ? kH264SBit : 0);
fu_header |= (packet->last_fragment ? kH264EBit : 0);
uint8_t type = packet->header & kH264TypeMask;
fu_header |= type;
rtc::ArrayView<const uint8_t> 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