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webrtc / src / df578330b8a0b1a003a37ca34253e7344caf17f4 / . / modules / rtp_rtcp / source / rtp_sender.cc
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/*
* 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/rtp_sender.h"
#include <algorithm>
#include <limits>
#include <string>
#include <utility>
#include "absl/memory/memory.h"
#include "absl/strings/match.h"
#include "api/array_view.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/transport/field_trial_based_config.h"
#include "logging/rtc_event_log/events/rtc_event_rtp_packet_outgoing.h"
#include "modules/rtp_rtcp/include/rtp_cvo.h"
#include "modules/rtp_rtcp/source/byte_io.h"
#include "modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension.h"
#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
#include "modules/rtp_rtcp/source/time_util.h"
#include "rtc_base/arraysize.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/safe_minmax.h"
#include "rtc_base/rate_limiter.h"
#include "rtc_base/time_utils.h"
namespace webrtc {
namespace {
// Max in the RFC 3550 is 255 bytes, we limit it to be modulus 32 for SRTP.
constexpr size_t kMaxPaddingLength = 224;
constexpr size_t kMinAudioPaddingLength = 50;
constexpr int kSendSideDelayWindowMs = 1000;
constexpr size_t kRtpHeaderLength = 12;
constexpr uint16_t kMaxInitRtpSeqNumber = 32767; // 2^15 -1.
constexpr uint32_t kTimestampTicksPerMs = 90;
constexpr int kBitrateStatisticsWindowMs = 1000;
// Min size needed to get payload padding from packet history.
constexpr int kMinPayloadPaddingBytes = 50;
template <typename Extension>
constexpr RtpExtensionSize CreateExtensionSize() {
return {Extension::kId, Extension::kValueSizeBytes};
}
template <typename Extension>
constexpr RtpExtensionSize CreateMaxExtensionSize() {
return {Extension::kId, Extension::kMaxValueSizeBytes};
}
// Size info for header extensions that might be used in padding or FEC packets.
constexpr RtpExtensionSize kFecOrPaddingExtensionSizes[] = {
CreateExtensionSize<AbsoluteSendTime>(),
CreateExtensionSize<TransmissionOffset>(),
CreateExtensionSize<TransportSequenceNumber>(),
CreateExtensionSize<PlayoutDelayLimits>(),
CreateMaxExtensionSize<RtpMid>(),
};
// Size info for header extensions that might be used in video packets.
constexpr RtpExtensionSize kVideoExtensionSizes[] = {
CreateExtensionSize<AbsoluteSendTime>(),
CreateExtensionSize<AbsoluteCaptureTimeExtension>(),
CreateExtensionSize<TransmissionOffset>(),
CreateExtensionSize<TransportSequenceNumber>(),
CreateExtensionSize<PlayoutDelayLimits>(),
CreateExtensionSize<VideoOrientation>(),
CreateExtensionSize<VideoContentTypeExtension>(),
CreateExtensionSize<VideoTimingExtension>(),
CreateMaxExtensionSize<RtpStreamId>(),
CreateMaxExtensionSize<RepairedRtpStreamId>(),
CreateMaxExtensionSize<RtpMid>(),
{RtpGenericFrameDescriptorExtension00::kId,
RtpGenericFrameDescriptorExtension00::kMaxSizeBytes},
{RtpGenericFrameDescriptorExtension01::kId,
RtpGenericFrameDescriptorExtension01::kMaxSizeBytes},
};
bool IsEnabled(absl::string_view name,
const WebRtcKeyValueConfig* field_trials) {
FieldTrialBasedConfig default_trials;
auto& trials = field_trials ? *field_trials : default_trials;
return trials.Lookup(name).find("Enabled") == 0;
}
bool HasBweExtension(const RtpHeaderExtensionMap& extensions_map) {
return extensions_map.IsRegistered(kRtpExtensionTransportSequenceNumber) ||
extensions_map.IsRegistered(kRtpExtensionTransportSequenceNumber02) ||
extensions_map.IsRegistered(kRtpExtensionAbsoluteSendTime) ||
extensions_map.IsRegistered(kRtpExtensionTransmissionTimeOffset);
}
} // namespace
RTPSender::RTPSender(const RtpRtcp::Configuration& config)
: clock_(config.clock),
random_(clock_->TimeInMicroseconds()),
audio_configured_(config.audio),
flexfec_ssrc_(config.flexfec_sender
? absl::make_optional(config.flexfec_sender->ssrc())
: absl::nullopt),
paced_sender_(config.paced_sender),
transport_sequence_number_allocator_(
config.transport_sequence_number_allocator),
transport_feedback_observer_(config.transport_feedback_callback),
transport_(config.outgoing_transport),
sending_media_(true), // Default to sending media.
force_part_of_allocation_(false),
max_packet_size_(IP_PACKET_SIZE - 28), // Default is IP-v4/UDP.
last_payload_type_(-1),
rtp_header_extension_map_(config.extmap_allow_mixed),
packet_history_(clock_),
// Statistics
send_delays_(),
max_delay_it_(send_delays_.end()),
sum_delays_ms_(0),
total_packet_send_delay_ms_(0),
rtp_stats_callback_(nullptr),
total_bitrate_sent_(kBitrateStatisticsWindowMs,
RateStatistics::kBpsScale),
nack_bitrate_sent_(kBitrateStatisticsWindowMs, RateStatistics::kBpsScale),
send_side_delay_observer_(config.send_side_delay_observer),
event_log_(config.event_log),
send_packet_observer_(config.send_packet_observer),
bitrate_callback_(config.send_bitrate_observer),
// RTP variables
sequence_number_forced_(false),
ssrc_(config.get_local_media_ssrc()),
ssrc_has_acked_(false),
rtx_ssrc_has_acked_(false),
last_rtp_timestamp_(0),
capture_time_ms_(0),
last_timestamp_time_ms_(0),
media_has_been_sent_(false),
last_packet_marker_bit_(false),
csrcs_(),
rtx_(kRtxOff),
ssrc_rtx_(config.rtx_send_ssrc),
rtp_overhead_bytes_per_packet_(0),
supports_bwe_extension_(false),
retransmission_rate_limiter_(config.retransmission_rate_limiter),
overhead_observer_(config.overhead_observer),
populate_network2_timestamp_(config.populate_network2_timestamp),
send_side_bwe_with_overhead_(
IsEnabled("WebRTC-SendSideBwe-WithOverhead", config.field_trials)) {
// This random initialization is not intended to be cryptographic strong.
timestamp_offset_ = random_.Rand<uint32_t>();
// Random start, 16 bits. Can't be 0.
sequence_number_rtx_ = random_.Rand(1, kMaxInitRtpSeqNumber);
sequence_number_ = random_.Rand(1, kMaxInitRtpSeqNumber);
}
RTPSender::RTPSender(
bool audio,
Clock* clock,
Transport* transport,
RtpPacketSender* paced_sender,
absl::optional<uint32_t> flexfec_ssrc,
TransportSequenceNumberAllocator* sequence_number_allocator,
TransportFeedbackObserver* transport_feedback_observer,
BitrateStatisticsObserver* bitrate_callback,
SendSideDelayObserver* send_side_delay_observer,
RtcEventLog* event_log,
SendPacketObserver* send_packet_observer,
RateLimiter* retransmission_rate_limiter,
OverheadObserver* overhead_observer,
bool populate_network2_timestamp,
FrameEncryptorInterface* frame_encryptor,
bool require_frame_encryption,
bool extmap_allow_mixed,
const WebRtcKeyValueConfig& field_trials)
: clock_(clock),
random_(clock_->TimeInMicroseconds()),
audio_configured_(audio),
flexfec_ssrc_(flexfec_ssrc),
paced_sender_(paced_sender),
transport_sequence_number_allocator_(sequence_number_allocator),
transport_feedback_observer_(transport_feedback_observer),
transport_(transport),
sending_media_(true), // Default to sending media.
force_part_of_allocation_(false),
max_packet_size_(IP_PACKET_SIZE - 28), // Default is IP-v4/UDP.
last_payload_type_(-1),
rtp_header_extension_map_(extmap_allow_mixed),
packet_history_(clock),
// Statistics
send_delays_(),
max_delay_it_(send_delays_.end()),
sum_delays_ms_(0),
total_packet_send_delay_ms_(0),
rtp_stats_callback_(nullptr),
total_bitrate_sent_(kBitrateStatisticsWindowMs,
RateStatistics::kBpsScale),
nack_bitrate_sent_(kBitrateStatisticsWindowMs, RateStatistics::kBpsScale),
send_side_delay_observer_(send_side_delay_observer),
event_log_(event_log),
send_packet_observer_(send_packet_observer),
bitrate_callback_(bitrate_callback),
// RTP variables
sequence_number_forced_(false),
ssrc_has_acked_(false),
rtx_ssrc_has_acked_(false),
last_rtp_timestamp_(0),
capture_time_ms_(0),
last_timestamp_time_ms_(0),
media_has_been_sent_(false),
last_packet_marker_bit_(false),
csrcs_(),
rtx_(kRtxOff),
rtp_overhead_bytes_per_packet_(0),
supports_bwe_extension_(false),
retransmission_rate_limiter_(retransmission_rate_limiter),
overhead_observer_(overhead_observer),
populate_network2_timestamp_(populate_network2_timestamp),
send_side_bwe_with_overhead_(
field_trials.Lookup("WebRTC-SendSideBwe-WithOverhead")
.find("Enabled") == 0) {
// This random initialization is not intended to be cryptographic strong.
timestamp_offset_ = random_.Rand<uint32_t>();
// Random start, 16 bits. Can't be 0.
sequence_number_rtx_ = random_.Rand(1, kMaxInitRtpSeqNumber);
sequence_number_ = random_.Rand(1, kMaxInitRtpSeqNumber);
}
RTPSender::~RTPSender() {
// TODO(tommi): Use a thread checker to ensure the object is created and
// deleted on the same thread. At the moment this isn't possible due to
// voe::ChannelOwner in voice engine. To reproduce, run:
// voe_auto_test --automated --gtest_filter=*MixManyChannelsForStressOpus
// TODO(tommi,holmer): We don't grab locks in the dtor before accessing member
// variables but we grab them in all other methods. (what's the design?)
// Start documenting what thread we're on in what method so that it's easier
// to understand performance attributes and possibly remove locks.
}
rtc::ArrayView<const RtpExtensionSize> RTPSender::FecExtensionSizes() {
return rtc::MakeArrayView(kFecOrPaddingExtensionSizes,
arraysize(kFecOrPaddingExtensionSizes));
}
rtc::ArrayView<const RtpExtensionSize> RTPSender::VideoExtensionSizes() {
return rtc::MakeArrayView(kVideoExtensionSizes,
arraysize(kVideoExtensionSizes));
}
uint16_t RTPSender::ActualSendBitrateKbit() const {
rtc::CritScope cs(&statistics_crit_);
return static_cast<uint16_t>(
total_bitrate_sent_.Rate(clock_->TimeInMilliseconds()).value_or(0) /
1000);
}
uint32_t RTPSender::NackOverheadRate() const {
rtc::CritScope cs(&statistics_crit_);
return nack_bitrate_sent_.Rate(clock_->TimeInMilliseconds()).value_or(0);
}
void RTPSender::SetExtmapAllowMixed(bool extmap_allow_mixed) {
rtc::CritScope lock(&send_critsect_);
rtp_header_extension_map_.SetExtmapAllowMixed(extmap_allow_mixed);
}
int32_t RTPSender::RegisterRtpHeaderExtension(RTPExtensionType type,
uint8_t id) {
rtc::CritScope lock(&send_critsect_);
bool registered = rtp_header_extension_map_.RegisterByType(id, type);
supports_bwe_extension_ = HasBweExtension(rtp_header_extension_map_);
return registered ? 0 : -1;
}
bool RTPSender::RegisterRtpHeaderExtension(const std::string& uri, int id) {
rtc::CritScope lock(&send_critsect_);
bool registered = rtp_header_extension_map_.RegisterByUri(id, uri);
supports_bwe_extension_ = HasBweExtension(rtp_header_extension_map_);
return registered;
}
bool RTPSender::IsRtpHeaderExtensionRegistered(RTPExtensionType type) const {
rtc::CritScope lock(&send_critsect_);
return rtp_header_extension_map_.IsRegistered(type);
}
int32_t RTPSender::DeregisterRtpHeaderExtension(RTPExtensionType type) {
rtc::CritScope lock(&send_critsect_);
int32_t deregistered = rtp_header_extension_map_.Deregister(type);
supports_bwe_extension_ = HasBweExtension(rtp_header_extension_map_);
return deregistered;
}
void RTPSender::SetMaxRtpPacketSize(size_t max_packet_size) {
RTC_DCHECK_GE(max_packet_size, 100);
RTC_DCHECK_LE(max_packet_size, IP_PACKET_SIZE);
rtc::CritScope lock(&send_critsect_);
max_packet_size_ = max_packet_size;
}
size_t RTPSender::MaxRtpPacketSize() const {
return max_packet_size_;
}
void RTPSender::SetRtxStatus(int mode) {
rtc::CritScope lock(&send_critsect_);
rtx_ = mode;
}
int RTPSender::RtxStatus() const {
rtc::CritScope lock(&send_critsect_);
return rtx_;
}
void RTPSender::SetRtxSsrc(uint32_t ssrc) {
rtc::CritScope lock(&send_critsect_);
ssrc_rtx_.emplace(ssrc);
}
uint32_t RTPSender::RtxSsrc() const {
rtc::CritScope lock(&send_critsect_);
RTC_DCHECK(ssrc_rtx_);
return *ssrc_rtx_;
}
void RTPSender::SetRtxPayloadType(int payload_type,
int associated_payload_type) {
rtc::CritScope lock(&send_critsect_);
RTC_DCHECK_LE(payload_type, 127);
RTC_DCHECK_LE(associated_payload_type, 127);
if (payload_type < 0) {
RTC_LOG(LS_ERROR) << "Invalid RTX payload type: " << payload_type << ".";
return;
}
rtx_payload_type_map_[associated_payload_type] = payload_type;
}
void RTPSender::SetStorePacketsStatus(bool enable, uint16_t number_to_store) {
packet_history_.SetStorePacketsStatus(
enable ? RtpPacketHistory::StorageMode::kStoreAndCull
: RtpPacketHistory::StorageMode::kDisabled,
number_to_store);
}
bool RTPSender::StorePackets() const {
return packet_history_.GetStorageMode() !=
RtpPacketHistory::StorageMode::kDisabled;
}
int32_t RTPSender::ReSendPacket(uint16_t packet_id) {
// Try to find packet in RTP packet history. Also verify RTT here, so that we
// don't retransmit too often.
absl::optional<RtpPacketHistory::PacketState> stored_packet =
packet_history_.GetPacketState(packet_id);
if (!stored_packet || stored_packet->pending_transmission) {
// Packet not found or already queued for retransmission, ignore.
return 0;
}
const int32_t packet_size = static_cast<int32_t>(stored_packet->packet_size);
const bool rtx = (RtxStatus() & kRtxRetransmitted) > 0;
if (paced_sender_) {
std::unique_ptr<RtpPacketToSend> packet =
packet_history_.GetPacketAndMarkAsPending(
packet_id, [&](const RtpPacketToSend& stored_packet) {
// Check if we're overusing retransmission bitrate.
// TODO(sprang): Add histograms for nack success or failure
// reasons.
std::unique_ptr<RtpPacketToSend> retransmit_packet;
if (retransmission_rate_limiter_ &&
!retransmission_rate_limiter_->TryUseRate(packet_size)) {
return retransmit_packet;
}
if (rtx) {
retransmit_packet = BuildRtxPacket(stored_packet);
} else {
retransmit_packet =
absl::make_unique<RtpPacketToSend>(stored_packet);
}
if (retransmit_packet) {
retransmit_packet->set_retransmitted_sequence_number(
stored_packet.SequenceNumber());
}
return retransmit_packet;
});
if (!packet) {
return -1;
}
packet->set_packet_type(RtpPacketToSend::Type::kRetransmission);
paced_sender_->EnqueuePacket(std::move(packet));
return packet_size;
}
// TODO(sprang): Replace this whole code-path with a pass-through pacer.
// Check if we're overusing retransmission bitrate.
// TODO(sprang): Add histograms for nack success or failure reasons.
if (retransmission_rate_limiter_ &&
!retransmission_rate_limiter_->TryUseRate(packet_size)) {
return -1;
}
std::unique_ptr<RtpPacketToSend> packet =
packet_history_.GetPacketAndSetSendTime(packet_id);
if (!packet) {
// Packet could theoretically time out between the first check and this one.
return 0;
}
if (!PrepareAndSendPacket(std::move(packet), rtx, true, PacedPacketInfo()))
return -1;
return packet_size;
}
void RTPSender::OnReceivedAckOnSsrc(int64_t extended_highest_sequence_number) {
rtc::CritScope lock(&send_critsect_);
ssrc_has_acked_ = true;
}
void RTPSender::OnReceivedAckOnRtxSsrc(
int64_t extended_highest_sequence_number) {
rtc::CritScope lock(&send_critsect_);
rtx_ssrc_has_acked_ = true;
}
bool RTPSender::SendPacketToNetwork(const RtpPacketToSend& packet,
const PacketOptions& options,
const PacedPacketInfo& pacing_info) {
int bytes_sent = -1;
if (transport_) {
UpdateRtpOverhead(packet);
bytes_sent = transport_->SendRtp(packet.data(), packet.size(), options)
? static_cast<int>(packet.size())
: -1;
if (event_log_ && bytes_sent > 0) {
event_log_->Log(absl::make_unique<RtcEventRtpPacketOutgoing>(
packet, pacing_info.probe_cluster_id));
}
}
// TODO(pwestin): Add a separate bitrate for sent bitrate after pacer.
if (bytes_sent <= 0) {
RTC_LOG(LS_WARNING) << "Transport failed to send packet.";
return false;
}
return true;
}
void RTPSender::OnReceivedNack(
const std::vector<uint16_t>& nack_sequence_numbers,
int64_t avg_rtt) {
packet_history_.SetRtt(5 + avg_rtt);
for (uint16_t seq_no : nack_sequence_numbers) {
const int32_t bytes_sent = ReSendPacket(seq_no);
if (bytes_sent < 0) {
// Failed to send one Sequence number. Give up the rest in this nack.
RTC_LOG(LS_WARNING) << "Failed resending RTP packet " << seq_no
<< ", Discard rest of packets.";
break;
}
}
}
// Called from pacer when we can send the packet.
RtpPacketSendResult RTPSender::TimeToSendPacket(
uint32_t ssrc,
uint16_t sequence_number,
int64_t capture_time_ms,
bool retransmission,
const PacedPacketInfo& pacing_info) {
RTC_NOTREACHED();
return RtpPacketSendResult::kSuccess;
}
// Called from pacer when we can send the packet.
bool RTPSender::TrySendPacket(RtpPacketToSend* packet,
const PacedPacketInfo& pacing_info) {
RTC_DCHECK(packet);
const uint32_t packet_ssrc = packet->Ssrc();
const auto packet_type = packet->packet_type();
RTC_DCHECK(packet_type.has_value());
PacketOptions options;
bool is_media = false;
bool is_rtx = false;
{
rtc::CritScope lock(&send_critsect_);
if (!sending_media_) {
return false;
}
switch (*packet_type) {
case RtpPacketToSend::Type::kAudio:
case RtpPacketToSend::Type::kVideo:
if (packet_ssrc != ssrc_) {
return false;
}
is_media = true;
break;
case RtpPacketToSend::Type::kRetransmission:
case RtpPacketToSend::Type::kPadding:
// Both padding and retransmission must be on either the media or the
// RTX stream.
if (packet_ssrc == ssrc_rtx_) {
is_rtx = true;
} else if (packet_ssrc != ssrc_) {
return false;
}
break;
case RtpPacketToSend::Type::kForwardErrorCorrection:
// FlexFEC is on separate SSRC, ULPFEC uses media SSRC.
if (packet_ssrc != ssrc_ && packet_ssrc != flexfec_ssrc_) {
return false;
}
break;
}
options.included_in_allocation = force_part_of_allocation_;
}
// Bug webrtc:7859. While FEC is invoked from rtp_sender_video, and not after
// the pacer, these modifications of the header below are happening after the
// FEC protection packets are calculated. This will corrupt recovered packets
// at the same place. It's not an issue for extensions, which are present in
// all the packets (their content just may be incorrect on recovered packets).
// In case of VideoTimingExtension, since it's present not in every packet,
// data after rtp header may be corrupted if these packets are protected by
// the FEC.
int64_t now_ms = clock_->TimeInMilliseconds();
int64_t diff_ms = now_ms - packet->capture_time_ms();
if (packet->IsExtensionReserved<TransmissionOffset>()) {
packet->SetExtension<TransmissionOffset>(kTimestampTicksPerMs * diff_ms);
}
if (packet->IsExtensionReserved<AbsoluteSendTime>()) {
packet->SetExtension<AbsoluteSendTime>(
AbsoluteSendTime::MsTo24Bits(now_ms));
}
if (packet->HasExtension<VideoTimingExtension>()) {
if (populate_network2_timestamp_) {
packet->set_network2_time_ms(now_ms);
} else {
packet->set_pacer_exit_time_ms(now_ms);
}
}
// Downstream code actually uses this flag to distinguish between media and
// everything else.
options.is_retransmit = !is_media;
if (auto packet_id = packet->GetExtension<TransportSequenceNumber>()) {
options.packet_id = *packet_id;
options.included_in_feedback = true;
options.included_in_allocation = true;
AddPacketToTransportFeedback(*packet_id, *packet, pacing_info);
}
options.application_data.assign(packet->application_data().begin(),
packet->application_data().end());
if (packet->packet_type() != RtpPacketToSend::Type::kPadding &&
packet->packet_type() != RtpPacketToSend::Type::kRetransmission) {
UpdateDelayStatistics(packet->capture_time_ms(), now_ms, packet_ssrc);
UpdateOnSendPacket(options.packet_id, packet->capture_time_ms(),
packet_ssrc);
}
const bool send_success = SendPacketToNetwork(*packet, options, pacing_info);
// Put packet in retransmission history or update pending status even if
// actual sending fails.
if (is_media && packet->allow_retransmission()) {
packet_history_.PutRtpPacket(absl::make_unique<RtpPacketToSend>(*packet),
StorageType::kAllowRetransmission, now_ms);
} else if (packet->retransmitted_sequence_number()) {
packet_history_.MarkPacketAsSent(*packet->retransmitted_sequence_number());
}
if (send_success) {
UpdateRtpStats(*packet, is_rtx,
packet_type == RtpPacketToSend::Type::kRetransmission);
rtc::CritScope lock(&send_critsect_);
media_has_been_sent_ = true;
}
// Return true even if transport failed (will be handled by retransmissions
// instead in that case), so that PacketRouter does not have to iterate over
// all other RTP modules and fail to send there too.
return true;
}
bool RTPSender::SupportsPadding() const {
rtc::CritScope lock(&send_critsect_);
return sending_media_ && supports_bwe_extension_;
}
bool RTPSender::SupportsRtxPayloadPadding() const {
rtc::CritScope lock(&send_critsect_);
return sending_media_ && supports_bwe_extension_ &&
(rtx_ & kRtxRedundantPayloads);
}
bool RTPSender::PrepareAndSendPacket(std::unique_ptr<RtpPacketToSend> packet,
bool send_over_rtx,
bool is_retransmit,
const PacedPacketInfo& pacing_info) {
RTC_DCHECK(packet);
int64_t capture_time_ms = packet->capture_time_ms();
RtpPacketToSend* packet_to_send = packet.get();
std::unique_ptr<RtpPacketToSend> packet_rtx;
if (send_over_rtx) {
packet_rtx = BuildRtxPacket(*packet);
if (!packet_rtx)
return false;
packet_to_send = packet_rtx.get();
}
// Bug webrtc:7859. While FEC is invoked from rtp_sender_video, and not after
// the pacer, these modifications of the header below are happening after the
// FEC protection packets are calculated. This will corrupt recovered packets
// at the same place. It's not an issue for extensions, which are present in
// all the packets (their content just may be incorrect on recovered packets).
// In case of VideoTimingExtension, since it's present not in every packet,
// data after rtp header may be corrupted if these packets are protected by
// the FEC.
int64_t now_ms = clock_->TimeInMilliseconds();
int64_t diff_ms = now_ms - capture_time_ms;
packet_to_send->SetExtension<TransmissionOffset>(kTimestampTicksPerMs *
diff_ms);
packet_to_send->SetExtension<AbsoluteSendTime>(
AbsoluteSendTime::MsTo24Bits(now_ms));
if (packet_to_send->HasExtension<VideoTimingExtension>()) {
if (populate_network2_timestamp_) {
packet_to_send->set_network2_time_ms(now_ms);
} else {
packet_to_send->set_pacer_exit_time_ms(now_ms);
}
}
PacketOptions options;
// If we are sending over RTX, it also means this is a retransmission.
// E.g. RTPSender::TrySendRedundantPayloads calls PrepareAndSendPacket with
// send_over_rtx = true but is_retransmit = false.
options.is_retransmit = is_retransmit || send_over_rtx;
bool has_transport_seq_num;
{
rtc::CritScope lock(&send_critsect_);
has_transport_seq_num =
UpdateTransportSequenceNumber(packet_to_send, &options.packet_id);
options.included_in_allocation =
has_transport_seq_num || force_part_of_allocation_;
options.included_in_feedback = has_transport_seq_num;
}
if (has_transport_seq_num) {
AddPacketToTransportFeedback(options.packet_id, *packet_to_send,
pacing_info);
}
options.application_data.assign(packet_to_send->application_data().begin(),
packet_to_send->application_data().end());
if (!is_retransmit && !send_over_rtx) {
UpdateDelayStatistics(packet->capture_time_ms(), now_ms, packet->Ssrc());
UpdateOnSendPacket(options.packet_id, packet->capture_time_ms(),
packet->Ssrc());
}
if (!SendPacketToNetwork(*packet_to_send, options, pacing_info))
return false;
{
rtc::CritScope lock(&send_critsect_);
media_has_been_sent_ = true;
}
UpdateRtpStats(*packet_to_send, send_over_rtx, is_retransmit);
return true;
}
void RTPSender::UpdateRtpStats(const RtpPacketToSend& packet,
bool is_rtx,
bool is_retransmit) {
int64_t now_ms = clock_->TimeInMilliseconds();
rtc::CritScope lock(&statistics_crit_);
StreamDataCounters* counters = is_rtx ? &rtx_rtp_stats_ : &rtp_stats_;
total_bitrate_sent_.Update(packet.size(), now_ms);
if (counters->first_packet_time_ms == -1)
counters->first_packet_time_ms = now_ms;
if (packet.packet_type() == RtpPacketToSend::Type::kForwardErrorCorrection) {
counters->fec.AddPacket(packet);
}
if (is_retransmit) {
counters->retransmitted.AddPacket(packet);
nack_bitrate_sent_.Update(packet.size(), now_ms);
}
counters->transmitted.AddPacket(packet);
if (rtp_stats_callback_)
rtp_stats_callback_->DataCountersUpdated(*counters, packet.Ssrc());
}
size_t RTPSender::TimeToSendPadding(size_t bytes,
const PacedPacketInfo& pacing_info) {
// TODO(bugs.webrtc.org/10633): Remove when downstream test usage is gone.
size_t padding_bytes_sent = 0;
for (auto& packet : GeneratePadding(bytes)) {
const size_t packet_size = packet->payload_size() + packet->padding_size();
if (TrySendPacket(packet.get(), pacing_info)) {
padding_bytes_sent += packet_size;
}
}
return padding_bytes_sent;
}
std::vector<std::unique_ptr<RtpPacketToSend>> RTPSender::GeneratePadding(
size_t target_size_bytes) {
// This method does not actually send packets, it just generates
// them and puts them in the pacer queue. Since this should incur
// low overhead, keep the lock for the scope of the method in order
// to make the code more readable.
std::vector<std::unique_ptr<RtpPacketToSend>> padding_packets;
size_t bytes_left = target_size_bytes;
if (SupportsRtxPayloadPadding()) {
while (bytes_left >= kMinPayloadPaddingBytes) {
std::unique_ptr<RtpPacketToSend> packet =
packet_history_.GetPayloadPaddingPacket(
[&](const RtpPacketToSend& packet)
-> std::unique_ptr<RtpPacketToSend> {
return BuildRtxPacket(packet);
});
if (!packet) {
break;
}
bytes_left -= std::min(bytes_left, packet->payload_size());
packet->set_packet_type(RtpPacketToSend::Type::kPadding);
padding_packets.push_back(std::move(packet));
}
}
rtc::CritScope lock(&send_critsect_);
if (!sending_media_) {
return {};
}
size_t padding_bytes_in_packet;
const size_t max_payload_size = max_packet_size_ - RtpHeaderLength();
if (audio_configured_) {
// Allow smaller padding packets for audio.
padding_bytes_in_packet = rtc::SafeClamp<size_t>(
bytes_left, kMinAudioPaddingLength,
rtc::SafeMin(max_payload_size, kMaxPaddingLength));
} else {
// Always send full padding packets. This is accounted for by the
// RtpPacketSender, which will make sure we don't send too much padding even
// if a single packet is larger than requested.
// We do this to avoid frequently sending small packets on higher bitrates.
padding_bytes_in_packet = rtc::SafeMin(max_payload_size, kMaxPaddingLength);
}
while (bytes_left > 0) {
auto padding_packet =
absl::make_unique<RtpPacketToSend>(&rtp_header_extension_map_);
padding_packet->set_packet_type(RtpPacketToSend::Type::kPadding);
padding_packet->SetMarker(false);
padding_packet->SetTimestamp(last_rtp_timestamp_);
padding_packet->set_capture_time_ms(capture_time_ms_);
if (rtx_ == kRtxOff) {
if (last_payload_type_ == -1) {
break;
}
// Without RTX we can't send padding in the middle of frames.
// For audio marker bits doesn't mark the end of a frame and frames
// are usually a single packet, so for now we don't apply this rule
// for audio.
if (!audio_configured_ && !last_packet_marker_bit_) {
break;
}
RTC_DCHECK(ssrc_);
padding_packet->SetSsrc(*ssrc_);
padding_packet->SetPayloadType(last_payload_type_);
padding_packet->SetSequenceNumber(sequence_number_++);
} else {
// Without abs-send-time or transport sequence number a media packet
// must be sent before padding so that the timestamps used for
// estimation are correct.
if (!media_has_been_sent_ &&
!(rtp_header_extension_map_.IsRegistered(AbsoluteSendTime::kId) ||
rtp_header_extension_map_.IsRegistered(
TransportSequenceNumber::kId))) {
break;
}
// Only change the timestamp of padding packets sent over RTX.
// Padding only packets over RTP has to be sent as part of a media
// frame (and therefore the same timestamp).
int64_t now_ms = clock_->TimeInMilliseconds();
if (last_timestamp_time_ms_ > 0) {
padding_packet->SetTimestamp(padding_packet->Timestamp() +
(now_ms - last_timestamp_time_ms_) *
kTimestampTicksPerMs);
padding_packet->set_capture_time_ms(padding_packet->capture_time_ms() +
(now_ms - last_timestamp_time_ms_));
}
RTC_DCHECK(ssrc_rtx_);
padding_packet->SetSsrc(*ssrc_rtx_);
padding_packet->SetSequenceNumber(sequence_number_rtx_++);
padding_packet->SetPayloadType(rtx_payload_type_map_.begin()->second);
}
if (rtp_header_extension_map_.IsRegistered(TransportSequenceNumber::kId)) {
padding_packet->ReserveExtension<TransportSequenceNumber>();
}
if (rtp_header_extension_map_.IsRegistered(TransmissionOffset::kId)) {
padding_packet->ReserveExtension<TransmissionOffset>();
}
if (rtp_header_extension_map_.IsRegistered(AbsoluteSendTime::kId)) {
padding_packet->ReserveExtension<AbsoluteSendTime>();
}
padding_packet->SetPadding(padding_bytes_in_packet);
bytes_left -= std::min(bytes_left, padding_bytes_in_packet);
padding_packets.push_back(std::move(padding_packet));
}
return padding_packets;
}
bool RTPSender::SendToNetwork(std::unique_ptr<RtpPacketToSend> packet,
StorageType storage) {
RTC_DCHECK(packet);
int64_t now_ms = clock_->TimeInMilliseconds();
uint32_t ssrc = packet->Ssrc();
if (paced_sender_) {
auto packet_type = packet->packet_type();
RTC_CHECK(packet_type) << "Packet type must be set before sending.";
if (packet->capture_time_ms() <= 0) {
packet->set_capture_time_ms(now_ms);
}
packet->set_allow_retransmission(storage ==
StorageType::kAllowRetransmission);
paced_sender_->EnqueuePacket(std::move(packet));
return true;
}
PacketOptions options;
options.is_retransmit = false;
// |capture_time_ms| <= 0 is considered invalid.
// TODO(holmer): This should be changed all over Video Engine so that negative
// time is consider invalid, while 0 is considered a valid time.
if (packet->capture_time_ms() > 0) {
packet->SetExtension<TransmissionOffset>(
kTimestampTicksPerMs * (now_ms - packet->capture_time_ms()));
if (populate_network2_timestamp_ &&
packet->HasExtension<VideoTimingExtension>()) {
packet->set_network2_time_ms(now_ms);
}
}
packet->SetExtension<AbsoluteSendTime>(AbsoluteSendTime::MsTo24Bits(now_ms));
bool has_transport_seq_num;
{
rtc::CritScope lock(&send_critsect_);
has_transport_seq_num =
UpdateTransportSequenceNumber(packet.get(), &options.packet_id);
options.included_in_allocation =
has_transport_seq_num || force_part_of_allocation_;
options.included_in_feedback = has_transport_seq_num;
}
if (has_transport_seq_num) {
AddPacketToTransportFeedback(options.packet_id, *packet.get(),
PacedPacketInfo());
}
options.application_data.assign(packet->application_data().begin(),
packet->application_data().end());
UpdateDelayStatistics(packet->capture_time_ms(), now_ms, packet->Ssrc());
UpdateOnSendPacket(options.packet_id, packet->capture_time_ms(),
packet->Ssrc());
bool sent = SendPacketToNetwork(*packet, options, PacedPacketInfo());
if (sent) {
{
rtc::CritScope lock(&send_critsect_);
media_has_been_sent_ = true;
}
UpdateRtpStats(*packet, false, false);
}
// To support retransmissions, we store the media packet as sent in the
// packet history (even if send failed).
if (storage == kAllowRetransmission) {
RTC_DCHECK_EQ(ssrc, SSRC());
packet_history_.PutRtpPacket(std::move(packet), storage, now_ms);
}
return sent;
}
void RTPSender::RecomputeMaxSendDelay() {
max_delay_it_ = send_delays_.begin();
for (auto it = send_delays_.begin(); it != send_delays_.end(); ++it) {
if (it->second >= max_delay_it_->second) {
max_delay_it_ = it;
}
}
}
void RTPSender::UpdateDelayStatistics(int64_t capture_time_ms,
int64_t now_ms,
uint32_t ssrc) {
if (!send_side_delay_observer_ || capture_time_ms <= 0)
return;
int avg_delay_ms = 0;
int max_delay_ms = 0;
uint64_t total_packet_send_delay_ms = 0;
{
rtc::CritScope cs(&statistics_crit_);
// Compute the max and average of the recent capture-to-send delays.
// The time complexity of the current approach depends on the distribution
// of the delay values. This could be done more efficiently.
// Remove elements older than kSendSideDelayWindowMs.
auto lower_bound =
send_delays_.lower_bound(now_ms - kSendSideDelayWindowMs);
for (auto it = send_delays_.begin(); it != lower_bound; ++it) {
if (max_delay_it_ == it) {
max_delay_it_ = send_delays_.end();
}
sum_delays_ms_ -= it->second;
}
send_delays_.erase(send_delays_.begin(), lower_bound);
if (max_delay_it_ == send_delays_.end()) {
// Removed the previous max. Need to recompute.
RecomputeMaxSendDelay();
}
// Add the new element.
RTC_DCHECK_GE(now_ms, static_cast<int64_t>(0));
RTC_DCHECK_LE(now_ms, std::numeric_limits<int64_t>::max() / 2);
RTC_DCHECK_GE(capture_time_ms, static_cast<int64_t>(0));
RTC_DCHECK_LE(capture_time_ms, std::numeric_limits<int64_t>::max() / 2);
int64_t diff_ms = now_ms - capture_time_ms;
RTC_DCHECK_GE(diff_ms, static_cast<int64_t>(0));
RTC_DCHECK_LE(diff_ms,
static_cast<int64_t>(std::numeric_limits<int>::max()));
int new_send_delay = rtc::dchecked_cast<int>(now_ms - capture_time_ms);
SendDelayMap::iterator it;
bool inserted;
std::tie(it, inserted) =
send_delays_.insert(std::make_pair(now_ms, new_send_delay));
if (!inserted) {
// TODO(terelius): If we have multiple delay measurements during the same
// millisecond then we keep the most recent one. It is not clear that this
// is the right decision, but it preserves an earlier behavior.
int previous_send_delay = it->second;
sum_delays_ms_ -= previous_send_delay;
it->second = new_send_delay;
if (max_delay_it_ == it && new_send_delay < previous_send_delay) {
RecomputeMaxSendDelay();
}
}
if (max_delay_it_ == send_delays_.end() ||
it->second >= max_delay_it_->second) {
max_delay_it_ = it;
}
sum_delays_ms_ += new_send_delay;
total_packet_send_delay_ms_ += new_send_delay;
total_packet_send_delay_ms = total_packet_send_delay_ms_;
size_t num_delays = send_delays_.size();
RTC_DCHECK(max_delay_it_ != send_delays_.end());
max_delay_ms = rtc::dchecked_cast<int>(max_delay_it_->second);
int64_t avg_ms = (sum_delays_ms_ + num_delays / 2) / num_delays;
RTC_DCHECK_GE(avg_ms, static_cast<int64_t>(0));
RTC_DCHECK_LE(avg_ms,
static_cast<int64_t>(std::numeric_limits<int>::max()));
avg_delay_ms =
rtc::dchecked_cast<int>((sum_delays_ms_ + num_delays / 2) / num_delays);
}
send_side_delay_observer_->SendSideDelayUpdated(
avg_delay_ms, max_delay_ms, total_packet_send_delay_ms, ssrc);
}
void RTPSender::UpdateOnSendPacket(int packet_id,
int64_t capture_time_ms,
uint32_t ssrc) {
if (!send_packet_observer_ || capture_time_ms <= 0 || packet_id == -1)
return;
send_packet_observer_->OnSendPacket(packet_id, capture_time_ms, ssrc);
}
void RTPSender::ProcessBitrate() {
if (!bitrate_callback_)
return;
int64_t now_ms = clock_->TimeInMilliseconds();
uint32_t ssrc;
{
rtc::CritScope lock(&send_critsect_);
if (!ssrc_)
return;
ssrc = *ssrc_;
}
rtc::CritScope lock(&statistics_crit_);
bitrate_callback_->Notify(total_bitrate_sent_.Rate(now_ms).value_or(0),
nack_bitrate_sent_.Rate(now_ms).value_or(0), ssrc);
}
size_t RTPSender::RtpHeaderLength() const {
rtc::CritScope lock(&send_critsect_);
size_t rtp_header_length = kRtpHeaderLength;
rtp_header_length += sizeof(uint32_t) * csrcs_.size();
rtp_header_length += RtpHeaderExtensionSize(kFecOrPaddingExtensionSizes,
rtp_header_extension_map_);
return rtp_header_length;
}
uint16_t RTPSender::AllocateSequenceNumber(uint16_t packets_to_send) {
rtc::CritScope lock(&send_critsect_);
uint16_t first_allocated_sequence_number = sequence_number_;
sequence_number_ += packets_to_send;
return first_allocated_sequence_number;
}
void RTPSender::GetDataCounters(StreamDataCounters* rtp_stats,
StreamDataCounters* rtx_stats) const {
rtc::CritScope lock(&statistics_crit_);
*rtp_stats = rtp_stats_;
*rtx_stats = rtx_rtp_stats_;
}
std::unique_ptr<RtpPacketToSend> RTPSender::AllocatePacket() const {
rtc::CritScope lock(&send_critsect_);
// TODO(danilchap): Find better motivator and value for extra capacity.
// RtpPacketizer might slightly miscalulate needed size,
// SRTP may benefit from extra space in the buffer and do encryption in place
// saving reallocation.
// While sending slightly oversized packet increase chance of dropped packet,
// it is better than crash on drop packet without trying to send it.
static constexpr int kExtraCapacity = 16;
auto packet = absl::make_unique<RtpPacketToSend>(
&rtp_header_extension_map_, max_packet_size_ + kExtraCapacity);
RTC_DCHECK(ssrc_);
packet->SetSsrc(*ssrc_);
packet->SetCsrcs(csrcs_);
// Reserve extensions, if registered, RtpSender set in SendToNetwork.
packet->ReserveExtension<AbsoluteSendTime>();
packet->ReserveExtension<TransmissionOffset>();
packet->ReserveExtension<TransportSequenceNumber>();
// BUNDLE requires that the receiver "bind" the received SSRC to the values
// in the MID and/or (R)RID header extensions if present. Therefore, the
// sender can reduce overhead by omitting these header extensions once it
// knows that the receiver has "bound" the SSRC.
//
// The algorithm here is fairly simple: Always attach a MID and/or RID (if
// configured) to the outgoing packets until an RTCP receiver report comes
// back for this SSRC. That feedback indicates the receiver must have
// received a packet with the SSRC and header extension(s), so the sender
// then stops attaching the MID and RID.
if (!ssrc_has_acked_) {
// These are no-ops if the corresponding header extension is not registered.
if (!mid_.empty()) {
packet->SetExtension<RtpMid>(mid_);
}
if (!rid_.empty()) {
packet->SetExtension<RtpStreamId>(rid_);
}
}
return packet;
}
bool RTPSender::AssignSequenceNumber(RtpPacketToSend* packet) {
rtc::CritScope lock(&send_critsect_);
if (!sending_media_)
return false;
RTC_DCHECK(packet->Ssrc() == ssrc_);
packet->SetSequenceNumber(sequence_number_++);
// Remember marker bit to determine if padding can be inserted with
// sequence number following |packet|.
last_packet_marker_bit_ = packet->Marker();
// Remember payload type to use in the padding packet if rtx is disabled.
last_payload_type_ = packet->PayloadType();
// Save timestamps to generate timestamp field and extensions for the padding.
last_rtp_timestamp_ = packet->Timestamp();
last_timestamp_time_ms_ = clock_->TimeInMilliseconds();
capture_time_ms_ = packet->capture_time_ms();
return true;
}
bool RTPSender::UpdateTransportSequenceNumber(RtpPacketToSend* packet,
int* packet_id) {
RTC_DCHECK(packet);
RTC_DCHECK(packet_id);
if (!rtp_header_extension_map_.IsRegistered(TransportSequenceNumber::kId))
return false;
if (!transport_sequence_number_allocator_)
return false;
*packet_id = transport_sequence_number_allocator_->AllocateSequenceNumber();
if (!packet->SetExtension<TransportSequenceNumber>(*packet_id))
return false;
return true;
}
void RTPSender::SetSendingMediaStatus(bool enabled) {
rtc::CritScope lock(&send_critsect_);
sending_media_ = enabled;
}
bool RTPSender::SendingMedia() const {
rtc::CritScope lock(&send_critsect_);
return sending_media_;
}
void RTPSender::SetAsPartOfAllocation(bool part_of_allocation) {
rtc::CritScope lock(&send_critsect_);
force_part_of_allocation_ = part_of_allocation;
}
void RTPSender::SetTimestampOffset(uint32_t timestamp) {
rtc::CritScope lock(&send_critsect_);
timestamp_offset_ = timestamp;
}
uint32_t RTPSender::TimestampOffset() const {
rtc::CritScope lock(&send_critsect_);
return timestamp_offset_;
}
void RTPSender::SetSSRC(uint32_t ssrc) {
{
rtc::CritScope lock(&send_critsect_);
if (ssrc_ == ssrc) {
return; // Since it's the same SSRC, don't reset anything.
}
ssrc_.emplace(ssrc);
if (!sequence_number_forced_) {
sequence_number_ = random_.Rand(1, kMaxInitRtpSeqNumber);
}
}
// Clear RTP packet history, since any packets there belong to the old SSRC
// and they may conflict with packets from the new one.
packet_history_.Clear();
}
uint32_t RTPSender::SSRC() const {
rtc::CritScope lock(&send_critsect_);
RTC_DCHECK(ssrc_);
return *ssrc_;
}
void RTPSender::SetRid(const std::string& rid) {
// RID is used in simulcast scenario when multiple layers share the same mid.
rtc::CritScope lock(&send_critsect_);
RTC_DCHECK_LE(rid.length(), RtpStreamId::kMaxValueSizeBytes);
rid_ = rid;
}
void RTPSender::SetMid(const std::string& mid) {
// This is configured via the API.
rtc::CritScope lock(&send_critsect_);
RTC_DCHECK_LE(mid.length(), RtpMid::kMaxValueSizeBytes);
mid_ = mid;
}
absl::optional<uint32_t> RTPSender::FlexfecSsrc() const {
return flexfec_ssrc_;
}
void RTPSender::SetCsrcs(const std::vector<uint32_t>& csrcs) {
RTC_DCHECK_LE(csrcs.size(), kRtpCsrcSize);
rtc::CritScope lock(&send_critsect_);
csrcs_ = csrcs;
}
void RTPSender::SetSequenceNumber(uint16_t seq) {
bool updated_sequence_number = false;
{
rtc::CritScope lock(&send_critsect_);
sequence_number_forced_ = true;
if (sequence_number_ != seq) {
updated_sequence_number = true;
}
sequence_number_ = seq;
}
if (updated_sequence_number) {
// Sequence number series has been reset to a new value, clear RTP packet
// history, since any packets there may conflict with new ones.
packet_history_.Clear();
}
}
uint16_t RTPSender::SequenceNumber() const {
rtc::CritScope lock(&send_critsect_);
return sequence_number_;
}
static void CopyHeaderAndExtensionsToRtxPacket(const RtpPacketToSend& packet,
RtpPacketToSend* rtx_packet) {
// Set the relevant fixed packet headers. The following are not set:
// * Payload type - it is replaced in rtx packets.
// * Sequence number - RTX has a separate sequence numbering.
// * SSRC - RTX stream has its own SSRC.
rtx_packet->SetMarker(packet.Marker());
rtx_packet->SetTimestamp(packet.Timestamp());
// Set the variable fields in the packet header:
// * CSRCs - must be set before header extensions.
// * Header extensions - replace Rid header with RepairedRid header.
const std::vector<uint32_t> csrcs = packet.Csrcs();
rtx_packet->SetCsrcs(csrcs);
for (int extension_num = kRtpExtensionNone + 1;
extension_num < kRtpExtensionNumberOfExtensions; ++extension_num) {
auto extension = static_cast<RTPExtensionType>(extension_num);
// Stream ID header extensions (MID, RSID) are sent per-SSRC. Since RTX
// operates on a different SSRC, the presence and values of these header
// extensions should be determined separately and not blindly copied.
if (extension == kRtpExtensionMid ||
extension == kRtpExtensionRtpStreamId) {
continue;
}
// Empty extensions should be supported, so not checking |source.empty()|.
if (!packet.HasExtension(extension)) {
continue;
}
rtc::ArrayView<const uint8_t> source = packet.FindExtension(extension);
rtc::ArrayView<uint8_t> destination =
rtx_packet->AllocateExtension(extension, source.size());
// Could happen if any:
// 1. Extension has 0 length.
// 2. Extension is not registered in destination.
// 3. Allocating extension in destination failed.
if (destination.empty() || source.size() != destination.size()) {
continue;
}
std::memcpy(destination.begin(), source.begin(), destination.size());
}
}
std::unique_ptr<RtpPacketToSend> RTPSender::BuildRtxPacket(
const RtpPacketToSend& packet) {
std::unique_ptr<RtpPacketToSend> rtx_packet;
// Add original RTP header.
{
rtc::CritScope lock(&send_critsect_);
if (!sending_media_)
return nullptr;
RTC_DCHECK(ssrc_rtx_);
// Replace payload type.
auto kv = rtx_payload_type_map_.find(packet.PayloadType());
if (kv == rtx_payload_type_map_.end())
return nullptr;
rtx_packet = absl::make_unique<RtpPacketToSend>(&rtp_header_extension_map_,
max_packet_size_);
rtx_packet->SetPayloadType(kv->second);
// Replace sequence number.
rtx_packet->SetSequenceNumber(sequence_number_rtx_++);
// Replace SSRC.
rtx_packet->SetSsrc(*ssrc_rtx_);
CopyHeaderAndExtensionsToRtxPacket(packet, rtx_packet.get());
// RTX packets are sent on an SSRC different from the main media, so the
// decision to attach MID and/or RRID header extensions is completely
// separate from that of the main media SSRC.
//
// Note that RTX packets must used the RepairedRtpStreamId (RRID) header
// extension instead of the RtpStreamId (RID) header extension even though
// the payload is identical.
if (!rtx_ssrc_has_acked_) {
// These are no-ops if the corresponding header extension is not
// registered.
if (!mid_.empty()) {
rtx_packet->SetExtension<RtpMid>(mid_);
}
if (!rid_.empty()) {
rtx_packet->SetExtension<RepairedRtpStreamId>(rid_);
}
}
}
RTC_DCHECK(rtx_packet);
uint8_t* rtx_payload =
rtx_packet->AllocatePayload(packet.payload_size() + kRtxHeaderSize);
if (rtx_payload == nullptr)
return nullptr;
// Add OSN (original sequence number).
ByteWriter<uint16_t>::WriteBigEndian(rtx_payload, packet.SequenceNumber());
// Add original payload data.
auto payload = packet.payload();
memcpy(rtx_payload + kRtxHeaderSize, payload.data(), payload.size());
// Add original application data.
rtx_packet->set_application_data(packet.application_data());
// Copy capture time so e.g. TransmissionOffset is correctly set.
rtx_packet->set_capture_time_ms(packet.capture_time_ms());
return rtx_packet;
}
void RTPSender::RegisterRtpStatisticsCallback(
StreamDataCountersCallback* callback) {
rtc::CritScope cs(&statistics_crit_);
rtp_stats_callback_ = callback;
}
StreamDataCountersCallback* RTPSender::GetRtpStatisticsCallback() const {
rtc::CritScope cs(&statistics_crit_);
return rtp_stats_callback_;
}
uint32_t RTPSender::BitrateSent() const {
rtc::CritScope cs(&statistics_crit_);
return total_bitrate_sent_.Rate(clock_->TimeInMilliseconds()).value_or(0);
}
void RTPSender::SetRtpState(const RtpState& rtp_state) {
rtc::CritScope lock(&send_critsect_);
sequence_number_ = rtp_state.sequence_number;
sequence_number_forced_ = true;
timestamp_offset_ = rtp_state.start_timestamp;
last_rtp_timestamp_ = rtp_state.timestamp;
capture_time_ms_ = rtp_state.capture_time_ms;
last_timestamp_time_ms_ = rtp_state.last_timestamp_time_ms;
media_has_been_sent_ = rtp_state.media_has_been_sent;
ssrc_has_acked_ = rtp_state.ssrc_has_acked;
}
RtpState RTPSender::GetRtpState() const {
rtc::CritScope lock(&send_critsect_);
RtpState state;
state.sequence_number = sequence_number_;
state.start_timestamp = timestamp_offset_;
state.timestamp = last_rtp_timestamp_;
state.capture_time_ms = capture_time_ms_;
state.last_timestamp_time_ms = last_timestamp_time_ms_;
state.media_has_been_sent = media_has_been_sent_;
state.ssrc_has_acked = ssrc_has_acked_;
return state;
}
void RTPSender::SetRtxRtpState(const RtpState& rtp_state) {
rtc::CritScope lock(&send_critsect_);
sequence_number_rtx_ = rtp_state.sequence_number;
rtx_ssrc_has_acked_ = rtp_state.ssrc_has_acked;
}
RtpState RTPSender::GetRtxRtpState() const {
rtc::CritScope lock(&send_critsect_);
RtpState state;
state.sequence_number = sequence_number_rtx_;
state.start_timestamp = timestamp_offset_;
state.ssrc_has_acked = rtx_ssrc_has_acked_;
return state;
}
void RTPSender::AddPacketToTransportFeedback(
uint16_t packet_id,
const RtpPacketToSend& packet,
const PacedPacketInfo& pacing_info) {
if (transport_feedback_observer_) {
size_t packet_size = packet.payload_size() + packet.padding_size();
if (send_side_bwe_with_overhead_) {
packet_size = packet.size();
}
RtpPacketSendInfo packet_info;
packet_info.ssrc = SSRC();
packet_info.transport_sequence_number = packet_id;
packet_info.has_rtp_sequence_number = true;
packet_info.rtp_sequence_number = packet.SequenceNumber();
packet_info.length = packet_size;
packet_info.pacing_info = pacing_info;
transport_feedback_observer_->OnAddPacket(packet_info);
}
}
void RTPSender::UpdateRtpOverhead(const RtpPacketToSend& packet) {
if (!overhead_observer_)
return;
size_t overhead_bytes_per_packet;
{
rtc::CritScope lock(&send_critsect_);
if (rtp_overhead_bytes_per_packet_ == packet.headers_size()) {
return;
}
rtp_overhead_bytes_per_packet_ = packet.headers_size();
overhead_bytes_per_packet = rtp_overhead_bytes_per_packet_;
}
overhead_observer_->OnOverheadChanged(overhead_bytes_per_packet);
}
int64_t RTPSender::LastTimestampTimeMs() const {
rtc::CritScope lock(&send_critsect_);
return last_timestamp_time_ms_;
}
void RTPSender::SetRtt(int64_t rtt_ms) {
packet_history_.SetRtt(rtt_ms);
}
void RTPSender::OnPacketsAcknowledged(
rtc::ArrayView<const uint16_t> sequence_numbers) {
packet_history_.CullAcknowledgedPackets(sequence_numbers);
}
} // namespace webrtc