webrtc/modules/rtp_rtcp/source/flexfec_sender.cc
Ilya Nikolaevskiy a5d952f4be Reland "Refactor FEC code to use COW buffers"
Reland with fixes for fuzzer found crashes.

This refactoring helps to reduce unnecessary memcpy calls on the receive side.

This CL replaces |uint8 data[IP_PACKET_SIZE]| with |rtc::CopyOnWriteBuffer data| in Packet class, removes |length| field there, and does necessary changes.

Original Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145332

Bug: webrtc:10750
Change-Id: I6775a701bcb2ae25ec1666e1db90041cd49013b7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151131
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29116}
2019-09-09 16:20:33 +00:00

188 lines
7.2 KiB
C++

/*
* Copyright (c) 2016 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/include/flexfec_sender.h"
#include <string.h>
#include <list>
#include <utility>
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/forward_error_correction.h"
#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
namespace webrtc {
namespace {
// Let first sequence number be in the first half of the interval.
constexpr uint16_t kMaxInitRtpSeqNumber = 0x7fff;
// See breakdown in flexfec_header_reader_writer.cc.
constexpr size_t kFlexfecMaxHeaderSize = 32;
// Since we will mainly use FlexFEC to protect video streams, we use a 90 kHz
// clock for the RTP timestamps. (This is according to the RFC, which states
// that it is RECOMMENDED to use the same clock frequency for FlexFEC as for
// the protected media stream.)
// The constant converts from clock millisecond timestamps to the 90 kHz
// RTP timestamp.
const int kMsToRtpTimestamp = kVideoPayloadTypeFrequency / 1000;
// How often to log the generated FEC packets to the text log.
constexpr int64_t kPacketLogIntervalMs = 10000;
RtpHeaderExtensionMap RegisterSupportedExtensions(
const std::vector<RtpExtension>& rtp_header_extensions) {
RtpHeaderExtensionMap map;
for (const auto& extension : rtp_header_extensions) {
if (extension.uri == TransportSequenceNumber::kUri) {
map.Register<TransportSequenceNumber>(extension.id);
} else if (extension.uri == AbsoluteSendTime::kUri) {
map.Register<AbsoluteSendTime>(extension.id);
} else if (extension.uri == TransmissionOffset::kUri) {
map.Register<TransmissionOffset>(extension.id);
} else if (extension.uri == RtpMid::kUri) {
map.Register<RtpMid>(extension.id);
} else {
RTC_LOG(LS_INFO)
<< "FlexfecSender only supports RTP header extensions for "
<< "BWE and MID, so the extension " << extension.ToString()
<< " will not be used.";
}
}
return map;
}
} // namespace
FlexfecSender::FlexfecSender(
int payload_type,
uint32_t ssrc,
uint32_t protected_media_ssrc,
const std::string& mid,
const std::vector<RtpExtension>& rtp_header_extensions,
rtc::ArrayView<const RtpExtensionSize> extension_sizes,
const RtpState* rtp_state,
Clock* clock)
: clock_(clock),
random_(clock_->TimeInMicroseconds()),
last_generated_packet_ms_(-1),
payload_type_(payload_type),
// Reset RTP state if this is not the first time we are operating.
// Otherwise, randomize the initial timestamp offset and RTP sequence
// numbers. (This is not intended to be cryptographically strong.)
timestamp_offset_(rtp_state ? rtp_state->start_timestamp
: random_.Rand<uint32_t>()),
ssrc_(ssrc),
protected_media_ssrc_(protected_media_ssrc),
mid_(mid),
seq_num_(rtp_state ? rtp_state->sequence_number
: random_.Rand(1, kMaxInitRtpSeqNumber)),
ulpfec_generator_(
ForwardErrorCorrection::CreateFlexfec(ssrc, protected_media_ssrc)),
rtp_header_extension_map_(
RegisterSupportedExtensions(rtp_header_extensions)),
header_extensions_size_(
RtpHeaderExtensionSize(extension_sizes, rtp_header_extension_map_)) {
// This object should not have been instantiated if FlexFEC is disabled.
RTC_DCHECK_GE(payload_type, 0);
RTC_DCHECK_LE(payload_type, 127);
}
FlexfecSender::~FlexfecSender() = default;
// We are reusing the implementation from UlpfecGenerator for SetFecParameters,
// AddRtpPacketAndGenerateFec, and FecAvailable.
void FlexfecSender::SetFecParameters(const FecProtectionParams& params) {
ulpfec_generator_.SetFecParameters(params);
}
bool FlexfecSender::AddRtpPacketAndGenerateFec(const RtpPacketToSend& packet) {
// TODO(brandtr): Generalize this SSRC check when we support multistream
// protection.
RTC_DCHECK_EQ(packet.Ssrc(), protected_media_ssrc_);
return ulpfec_generator_.AddRtpPacketAndGenerateFec(
packet.Buffer(), packet.headers_size()) == 0;
}
bool FlexfecSender::FecAvailable() const {
return ulpfec_generator_.FecAvailable();
}
std::vector<std::unique_ptr<RtpPacketToSend>> FlexfecSender::GetFecPackets() {
std::vector<std::unique_ptr<RtpPacketToSend>> fec_packets_to_send;
fec_packets_to_send.reserve(ulpfec_generator_.generated_fec_packets_.size());
for (const auto* fec_packet : ulpfec_generator_.generated_fec_packets_) {
std::unique_ptr<RtpPacketToSend> fec_packet_to_send(
new RtpPacketToSend(&rtp_header_extension_map_));
fec_packet_to_send->set_packet_type(
RtpPacketToSend::Type::kForwardErrorCorrection);
// RTP header.
fec_packet_to_send->SetMarker(false);
fec_packet_to_send->SetPayloadType(payload_type_);
fec_packet_to_send->SetSequenceNumber(seq_num_++);
fec_packet_to_send->SetTimestamp(
timestamp_offset_ +
static_cast<uint32_t>(kMsToRtpTimestamp *
clock_->TimeInMilliseconds()));
// Set "capture time" so that the TransmissionOffset header extension
// can be set by the RTPSender.
fec_packet_to_send->set_capture_time_ms(clock_->TimeInMilliseconds());
fec_packet_to_send->SetSsrc(ssrc_);
// Reserve extensions, if registered. These will be set by the RTPSender.
fec_packet_to_send->ReserveExtension<AbsoluteSendTime>();
fec_packet_to_send->ReserveExtension<TransmissionOffset>();
fec_packet_to_send->ReserveExtension<TransportSequenceNumber>();
// Possibly include the MID header extension.
if (!mid_.empty()) {
// This is a no-op if the MID header extension is not registered.
fec_packet_to_send->SetExtension<RtpMid>(mid_);
}
// RTP payload.
uint8_t* payload =
fec_packet_to_send->AllocatePayload(fec_packet->data.size());
memcpy(payload, fec_packet->data.cdata(), fec_packet->data.size());
fec_packets_to_send.push_back(std::move(fec_packet_to_send));
}
ulpfec_generator_.ResetState();
int64_t now_ms = clock_->TimeInMilliseconds();
if (!fec_packets_to_send.empty() &&
now_ms - last_generated_packet_ms_ > kPacketLogIntervalMs) {
RTC_LOG(LS_VERBOSE) << "Generated " << fec_packets_to_send.size()
<< " FlexFEC packets with payload type: "
<< payload_type_ << " and SSRC: " << ssrc_ << ".";
last_generated_packet_ms_ = now_ms;
}
return fec_packets_to_send;
}
// The overhead is BWE RTP header extensions and FlexFEC header.
size_t FlexfecSender::MaxPacketOverhead() const {
return header_extensions_size_ + kFlexfecMaxHeaderSize;
}
RtpState FlexfecSender::GetRtpState() {
RtpState rtp_state;
rtp_state.sequence_number = seq_num_;
rtp_state.start_timestamp = timestamp_offset_;
return rtp_state;
}
} // namespace webrtc