webrtc_m130/modules/rtp_rtcp/source/ulpfec_receiver_impl.cc
Mirko Bonadei 317a1f09ed Use std::make_unique instead of absl::make_unique.
WebRTC is now using C++14 so there is no need to use the Abseil version
of std::make_unique.

This CL has been created with the following steps:

git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt
git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt
git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt

diff --new-line-format="" --unchanged-line-format="" \
  /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \
  uniq > /tmp/only_make_unique.txt
diff --new-line-format="" --unchanged-line-format="" \
  /tmp/only_make_unique.txt /tmp/memory.txt | \
  xargs grep -l "absl/memory" > /tmp/add-memory.txt

git grep -l "\babsl::make_unique\b" | \
  xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g"

git checkout PRESUBMIT.py abseil-in-webrtc.md

cat /tmp/add-memory.txt | \
  xargs sed -i \
  's/#include "absl\/memory\/memory.h"/#include <memory>/g'
git cl format
# Manual fix order of the new inserted #include <memory>

cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \
  xargs sed -i '/#include "absl\/memory\/memory.h"/d'

git ls-files | grep BUILD.gn | \
  xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d'

python tools_webrtc/gn_check_autofix.py \
  -m tryserver.webrtc -b linux_rel

# Repead the gn_check_autofix step for other platforms

git ls-files | grep BUILD.gn | \
  xargs sed -i 's/absl\/memory:memory/absl\/memory/g'
git cl format

Bug: webrtc:10945
Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 15:47:29 +00:00

222 lines
8.2 KiB
C++

/*
* 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 <string.h>
#include <memory>
#include <utility>
#include "api/scoped_refptr.h"
#include "modules/rtp_rtcp/source/byte_io.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> UlpfecReceiver::Create(
uint32_t ssrc,
RecoveredPacketReceiver* callback,
rtc::ArrayView<const RtpExtension> extensions) {
return std::make_unique<UlpfecReceiverImpl>(ssrc, callback, extensions);
}
UlpfecReceiverImpl::UlpfecReceiverImpl(
uint32_t ssrc,
RecoveredPacketReceiver* callback,
rtc::ArrayView<const RtpExtension> extensions)
: ssrc_(ssrc),
extensions_(extensions),
recovered_packet_callback_(callback),
fec_(ForwardErrorCorrection::CreateUlpfec(ssrc_)) {}
UlpfecReceiverImpl::~UlpfecReceiverImpl() {
received_packets_.clear();
fec_->ResetState(&recovered_packets_);
}
FecPacketCounter UlpfecReceiverImpl::GetPacketCounter() const {
rtc::CritScope cs(&crit_sect_);
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.
int32_t UlpfecReceiverImpl::AddReceivedRedPacket(
const RTPHeader& header,
const uint8_t* incoming_rtp_packet,
size_t packet_length,
uint8_t ulpfec_payload_type) {
if (header.ssrc != ssrc_) {
RTC_LOG(LS_WARNING)
<< "Received RED packet with different SSRC than expected; dropping.";
return -1;
}
if (packet_length > IP_PACKET_SIZE) {
RTC_LOG(LS_WARNING) << "Received RED packet with length exceeds maximum IP "
"packet size; dropping.";
return -1;
}
rtc::CritScope cs(&crit_sect_);
uint8_t red_header_length = 1;
size_t payload_data_length = packet_length - header.headerLength;
if (payload_data_length == 0) {
RTC_LOG(LS_WARNING) << "Corrupt/truncated FEC packet.";
return -1;
}
// Remove RED header of incoming packet and store as a virtual RTP packet.
std::unique_ptr<ForwardErrorCorrection::ReceivedPacket> received_packet(
new ForwardErrorCorrection::ReceivedPacket());
received_packet->pkt = new ForwardErrorCorrection::Packet();
// Get payload type from RED header and sequence number from RTP header.
uint8_t payload_type = incoming_rtp_packet[header.headerLength] & 0x7f;
received_packet->is_fec = payload_type == ulpfec_payload_type;
received_packet->ssrc = header.ssrc;
received_packet->seq_num = header.sequenceNumber;
if (incoming_rtp_packet[header.headerLength] & 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 -1;
}
++packet_counter_.num_packets;
packet_counter_.num_bytes += packet_length;
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.SetData(
incoming_rtp_packet + header.headerLength + red_header_length,
payload_data_length - red_header_length);
received_packet->ssrc =
ByteReader<uint32_t>::ReadBigEndian(&incoming_rtp_packet[8]);
} else {
received_packet->pkt->data.SetSize(header.headerLength +
payload_data_length - red_header_length);
// Copy RTP header.
memcpy(received_packet->pkt->data.data(), incoming_rtp_packet,
header.headerLength);
// Set payload type.
received_packet->pkt->data[1] &= 0x80; // Reset RED payload type.
received_packet->pkt->data[1] += payload_type; // Set media payload type.
// Copy payload data.
if (payload_data_length > red_header_length) {
memcpy(received_packet->pkt->data.data() + header.headerLength,
incoming_rtp_packet + header.headerLength + red_header_length,
payload_data_length - red_header_length);
}
}
if (received_packet->pkt->data.size() == 0) {
return 0;
}
received_packets_.push_back(std::move(received_packet));
return 0;
}
// TODO(nisse): Drop always-zero return value.
int32_t UlpfecReceiverImpl::ProcessReceivedFec() {
crit_sect_.Enter();
// 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<std::unique_ptr<ForwardErrorCorrection::ReceivedPacket>>
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;
crit_sect_.Leave();
recovered_packet_callback_->OnRecoveredPacket(packet->data.data(),
packet->data.size());
crit_sect_.Enter();
// Create a packet with the buffer to modify it.
RtpPacketReceived rtp_packet;
rtp_packet.Parse(packet->data);
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();
}
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;
crit_sect_.Leave();
recovered_packet_callback_->OnRecoveredPacket(packet->data.data(),
packet->data.size());
crit_sect_.Enter();
}
crit_sect_.Leave();
return 0;
}
} // namespace webrtc