blob: a43f788445835fc130be2d12ed61adbf1394f323 [file] [log] [blame]
/*
* 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_rtcp_impl.h"
#include <string.h>
#include <algorithm>
#include <cstdint>
#include <memory>
#include <set>
#include <string>
#include <utility>
#include "absl/strings/string_view.h"
#include "absl/types/optional.h"
#include "modules/rtp_rtcp/source/rtcp_packet/dlrr.h"
#include "modules/rtp_rtcp/source/rtcp_sender.h"
#include "modules/rtp_rtcp/source/rtp_rtcp_config.h"
#include "modules/rtp_rtcp/source/rtp_rtcp_interface.h"
#include "modules/rtp_rtcp/source/time_util.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "system_wrappers/include/ntp_time.h"
#ifdef _WIN32
// Disable warning C4355: 'this' : used in base member initializer list.
#pragma warning(disable : 4355)
#endif
namespace webrtc {
namespace {
const int64_t kRtpRtcpRttProcessTimeMs = 1000;
const int64_t kRtpRtcpBitrateProcessTimeMs = 10;
const int64_t kDefaultExpectedRetransmissionTimeMs = 125;
} // namespace
ModuleRtpRtcpImpl::RtpSenderContext::RtpSenderContext(
const RtpRtcpInterface::Configuration& config)
: packet_history(config.clock, config.enable_rtx_padding_prioritization),
sequencer_(config.local_media_ssrc,
config.rtx_send_ssrc,
/*require_marker_before_media_padding=*/!config.audio,
config.clock),
packet_sender(config, &packet_history),
non_paced_sender(&packet_sender, &sequencer_),
packet_generator(
config,
&packet_history,
config.paced_sender ? config.paced_sender : &non_paced_sender) {}
std::unique_ptr<RtpRtcp> RtpRtcp::DEPRECATED_Create(
const Configuration& configuration) {
RTC_DCHECK(configuration.clock);
return std::make_unique<ModuleRtpRtcpImpl>(configuration);
}
ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration)
: rtcp_sender_(
RTCPSender::Configuration::FromRtpRtcpConfiguration(configuration)),
rtcp_receiver_(configuration, this),
clock_(configuration.clock),
last_bitrate_process_time_(clock_->TimeInMilliseconds()),
last_rtt_process_time_(clock_->TimeInMilliseconds()),
packet_overhead_(28), // IPV4 UDP.
nack_last_time_sent_full_ms_(0),
nack_last_seq_number_sent_(0),
rtt_stats_(configuration.rtt_stats),
rtt_ms_(0) {
if (!configuration.receiver_only) {
rtp_sender_ = std::make_unique<RtpSenderContext>(configuration);
// Make sure rtcp sender use same timestamp offset as rtp sender.
rtcp_sender_.SetTimestampOffset(
rtp_sender_->packet_generator.TimestampOffset());
}
// Set default packet size limit.
// TODO(nisse): Kind-of duplicates
// webrtc::VideoSendStream::Config::Rtp::kDefaultMaxPacketSize.
const size_t kTcpOverIpv4HeaderSize = 40;
SetMaxRtpPacketSize(IP_PACKET_SIZE - kTcpOverIpv4HeaderSize);
}
ModuleRtpRtcpImpl::~ModuleRtpRtcpImpl() = default;
// Process any pending tasks such as timeouts (non time critical events).
void ModuleRtpRtcpImpl::Process() {
const int64_t now = clock_->TimeInMilliseconds();
if (rtp_sender_) {
if (now >= last_bitrate_process_time_ + kRtpRtcpBitrateProcessTimeMs) {
rtp_sender_->packet_sender.ProcessBitrateAndNotifyObservers();
last_bitrate_process_time_ = now;
}
}
// TODO(bugs.webrtc.org/11581): We update the RTT once a second, whereas other
// things that run in this method are updated much more frequently. Move the
// RTT checking over to the worker thread, which matches better with where the
// stats are maintained.
bool process_rtt = now >= last_rtt_process_time_ + kRtpRtcpRttProcessTimeMs;
if (rtcp_sender_.Sending()) {
// Process RTT if we have received a report block and we haven't
// processed RTT for at least `kRtpRtcpRttProcessTimeMs` milliseconds.
// Note that LastReceivedReportBlockMs() grabs a lock, so check
// `process_rtt` first.
if (process_rtt && rtt_stats_ != nullptr &&
rtcp_receiver_.LastReceivedReportBlockMs() > last_rtt_process_time_) {
TimeDelta max_rtt = TimeDelta::Zero();
for (const auto& block : rtcp_receiver_.GetLatestReportBlockData()) {
if (block.last_rtt() > max_rtt) {
max_rtt = block.last_rtt();
}
}
// Report the rtt.
if (max_rtt > TimeDelta::Zero()) {
rtt_stats_->OnRttUpdate(max_rtt.ms());
}
}
// Verify receiver reports are delivered and the reported sequence number
// is increasing.
// TODO(bugs.webrtc.org/11581): The timeout value needs to be checked every
// few seconds (see internals of RtcpRrTimeout). Here, we may be polling it
// a couple of hundred times a second, which isn't great since it grabs a
// lock. Note also that LastReceivedReportBlockMs() (called above) and
// RtcpRrTimeout() both grab the same lock and check the same timer, so
// it should be possible to consolidate that work somehow.
if (rtcp_receiver_.RtcpRrTimeout()) {
RTC_LOG_F(LS_WARNING) << "Timeout: No RTCP RR received.";
} else if (rtcp_receiver_.RtcpRrSequenceNumberTimeout()) {
RTC_LOG_F(LS_WARNING) << "Timeout: No increase in RTCP RR extended "
"highest sequence number.";
}
} else {
// Report rtt from receiver.
if (process_rtt) {
int64_t rtt_ms;
if (rtt_stats_ && rtcp_receiver_.GetAndResetXrRrRtt(&rtt_ms)) {
rtt_stats_->OnRttUpdate(rtt_ms);
}
}
}
// Get processed rtt.
if (process_rtt) {
last_rtt_process_time_ = now;
if (rtt_stats_) {
// Make sure we have a valid RTT before setting.
int64_t last_rtt = rtt_stats_->LastProcessedRtt();
if (last_rtt >= 0)
set_rtt_ms(last_rtt);
}
}
if (rtcp_sender_.TimeToSendRTCPReport())
rtcp_sender_.SendRTCP(GetFeedbackState(), kRtcpReport);
if (rtcp_sender_.TMMBR() && rtcp_receiver_.UpdateTmmbrTimers()) {
rtcp_receiver_.NotifyTmmbrUpdated();
}
}
void ModuleRtpRtcpImpl::SetRtxSendStatus(int mode) {
rtp_sender_->packet_generator.SetRtxStatus(mode);
}
int ModuleRtpRtcpImpl::RtxSendStatus() const {
return rtp_sender_ ? rtp_sender_->packet_generator.RtxStatus() : kRtxOff;
}
void ModuleRtpRtcpImpl::SetRtxSendPayloadType(int payload_type,
int associated_payload_type) {
rtp_sender_->packet_generator.SetRtxPayloadType(payload_type,
associated_payload_type);
}
absl::optional<uint32_t> ModuleRtpRtcpImpl::RtxSsrc() const {
return rtp_sender_ ? rtp_sender_->packet_generator.RtxSsrc() : absl::nullopt;
}
absl::optional<uint32_t> ModuleRtpRtcpImpl::FlexfecSsrc() const {
if (rtp_sender_) {
return rtp_sender_->packet_generator.FlexfecSsrc();
}
return absl::nullopt;
}
void ModuleRtpRtcpImpl::IncomingRtcpPacket(
rtc::ArrayView<const uint8_t> rtcp_packet) {
rtcp_receiver_.IncomingPacket(rtcp_packet);
}
void ModuleRtpRtcpImpl::RegisterSendPayloadFrequency(int payload_type,
int payload_frequency) {
rtcp_sender_.SetRtpClockRate(payload_type, payload_frequency);
}
int32_t ModuleRtpRtcpImpl::DeRegisterSendPayload(const int8_t payload_type) {
return 0;
}
uint32_t ModuleRtpRtcpImpl::StartTimestamp() const {
return rtp_sender_->packet_generator.TimestampOffset();
}
// Configure start timestamp, default is a random number.
void ModuleRtpRtcpImpl::SetStartTimestamp(const uint32_t timestamp) {
rtcp_sender_.SetTimestampOffset(timestamp);
rtp_sender_->packet_generator.SetTimestampOffset(timestamp);
rtp_sender_->packet_sender.SetTimestampOffset(timestamp);
}
uint16_t ModuleRtpRtcpImpl::SequenceNumber() const {
MutexLock lock(&rtp_sender_->sequencer_mutex);
return rtp_sender_->sequencer_.media_sequence_number();
}
// Set SequenceNumber, default is a random number.
void ModuleRtpRtcpImpl::SetSequenceNumber(const uint16_t seq_num) {
MutexLock lock(&rtp_sender_->sequencer_mutex);
rtp_sender_->sequencer_.set_media_sequence_number(seq_num);
}
void ModuleRtpRtcpImpl::SetRtpState(const RtpState& rtp_state) {
MutexLock lock(&rtp_sender_->sequencer_mutex);
rtp_sender_->packet_generator.SetRtpState(rtp_state);
rtp_sender_->sequencer_.SetRtpState(rtp_state);
rtcp_sender_.SetTimestampOffset(rtp_state.start_timestamp);
}
void ModuleRtpRtcpImpl::SetRtxState(const RtpState& rtp_state) {
MutexLock lock(&rtp_sender_->sequencer_mutex);
rtp_sender_->packet_generator.SetRtxRtpState(rtp_state);
rtp_sender_->sequencer_.set_rtx_sequence_number(rtp_state.sequence_number);
}
RtpState ModuleRtpRtcpImpl::GetRtpState() const {
MutexLock lock(&rtp_sender_->sequencer_mutex);
RtpState state = rtp_sender_->packet_generator.GetRtpState();
rtp_sender_->sequencer_.PopulateRtpState(state);
return state;
}
RtpState ModuleRtpRtcpImpl::GetRtxState() const {
MutexLock lock(&rtp_sender_->sequencer_mutex);
RtpState state = rtp_sender_->packet_generator.GetRtxRtpState();
state.sequence_number = rtp_sender_->sequencer_.rtx_sequence_number();
return state;
}
void ModuleRtpRtcpImpl::SetMid(absl::string_view mid) {
if (rtp_sender_) {
rtp_sender_->packet_generator.SetMid(mid);
}
// TODO(bugs.webrtc.org/4050): If we end up supporting the MID SDES item for
// RTCP, this will need to be passed down to the RTCPSender also.
}
// TODO(pbos): Handle media and RTX streams separately (separate RTCP
// feedbacks).
RTCPSender::FeedbackState ModuleRtpRtcpImpl::GetFeedbackState() {
RTCPSender::FeedbackState state;
// This is called also when receiver_only is true. Hence below
// checks that rtp_sender_ exists.
if (rtp_sender_) {
StreamDataCounters rtp_stats;
StreamDataCounters rtx_stats;
rtp_sender_->packet_sender.GetDataCounters(&rtp_stats, &rtx_stats);
state.packets_sent =
rtp_stats.transmitted.packets + rtx_stats.transmitted.packets;
state.media_bytes_sent = rtp_stats.transmitted.payload_bytes +
rtx_stats.transmitted.payload_bytes;
state.send_bitrate =
rtp_sender_->packet_sender.GetSendRates().Sum().bps<uint32_t>();
}
state.receiver = &rtcp_receiver_;
if (absl::optional<RtpRtcpInterface::SenderReportStats> last_sr =
rtcp_receiver_.GetSenderReportStats();
last_sr.has_value()) {
state.remote_sr = CompactNtp(last_sr->last_remote_timestamp);
state.last_rr = last_sr->last_arrival_timestamp;
}
state.last_xr_rtis = rtcp_receiver_.ConsumeReceivedXrReferenceTimeInfo();
return state;
}
int32_t ModuleRtpRtcpImpl::SetSendingStatus(const bool sending) {
if (rtcp_sender_.Sending() != sending) {
// Sends RTCP BYE when going from true to false
rtcp_sender_.SetSendingStatus(GetFeedbackState(), sending);
}
return 0;
}
bool ModuleRtpRtcpImpl::Sending() const {
return rtcp_sender_.Sending();
}
void ModuleRtpRtcpImpl::SetSendingMediaStatus(const bool sending) {
rtp_sender_->packet_generator.SetSendingMediaStatus(sending);
}
bool ModuleRtpRtcpImpl::SendingMedia() const {
return rtp_sender_ ? rtp_sender_->packet_generator.SendingMedia() : false;
}
bool ModuleRtpRtcpImpl::IsAudioConfigured() const {
return rtp_sender_ ? rtp_sender_->packet_generator.IsAudioConfigured()
: false;
}
void ModuleRtpRtcpImpl::SetAsPartOfAllocation(bool part_of_allocation) {
RTC_CHECK(rtp_sender_);
rtp_sender_->packet_sender.ForceIncludeSendPacketsInAllocation(
part_of_allocation);
}
bool ModuleRtpRtcpImpl::OnSendingRtpFrame(uint32_t timestamp,
int64_t capture_time_ms,
int payload_type,
bool force_sender_report) {
if (!Sending())
return false;
// TODO(bugs.webrtc.org/12873): Migrate this method and it's users to use
// optional Timestamps.
absl::optional<Timestamp> capture_time;
if (capture_time_ms > 0) {
capture_time = Timestamp::Millis(capture_time_ms);
}
absl::optional<int> payload_type_optional;
if (payload_type >= 0)
payload_type_optional = payload_type;
rtcp_sender_.SetLastRtpTime(timestamp, capture_time, payload_type_optional);
// Make sure an RTCP report isn't queued behind a key frame.
if (rtcp_sender_.TimeToSendRTCPReport(force_sender_report))
rtcp_sender_.SendRTCP(GetFeedbackState(), kRtcpReport);
return true;
}
bool ModuleRtpRtcpImpl::TrySendPacket(std::unique_ptr<RtpPacketToSend> packet,
const PacedPacketInfo& pacing_info) {
RTC_DCHECK(rtp_sender_);
// TODO(sprang): Consider if we can remove this check.
if (!rtp_sender_->packet_generator.SendingMedia()) {
return false;
}
{
MutexLock lock(&rtp_sender_->sequencer_mutex);
if (packet->packet_type() == RtpPacketMediaType::kPadding &&
packet->Ssrc() == rtp_sender_->packet_generator.SSRC() &&
!rtp_sender_->sequencer_.CanSendPaddingOnMediaSsrc()) {
// New media packet preempted this generated padding packet, discard it.
return false;
}
bool is_flexfec =
packet->packet_type() == RtpPacketMediaType::kForwardErrorCorrection &&
packet->Ssrc() == rtp_sender_->packet_generator.FlexfecSsrc();
if (!is_flexfec) {
rtp_sender_->sequencer_.Sequence(*packet);
}
}
rtp_sender_->packet_sender.SendPacket(packet.get(), pacing_info);
return true;
}
void ModuleRtpRtcpImpl::SetFecProtectionParams(const FecProtectionParams&,
const FecProtectionParams&) {
// Deferred FEC not supported in deprecated RTP module.
}
std::vector<std::unique_ptr<RtpPacketToSend>>
ModuleRtpRtcpImpl::FetchFecPackets() {
// Deferred FEC not supported in deprecated RTP module.
return {};
}
void ModuleRtpRtcpImpl::OnAbortedRetransmissions(
rtc::ArrayView<const uint16_t> sequence_numbers) {
RTC_DCHECK_NOTREACHED()
<< "Stream flushing not supported with legacy rtp modules.";
}
void ModuleRtpRtcpImpl::OnPacketsAcknowledged(
rtc::ArrayView<const uint16_t> sequence_numbers) {
RTC_DCHECK(rtp_sender_);
rtp_sender_->packet_history.CullAcknowledgedPackets(sequence_numbers);
}
bool ModuleRtpRtcpImpl::SupportsPadding() const {
RTC_DCHECK(rtp_sender_);
return rtp_sender_->packet_generator.SupportsPadding();
}
bool ModuleRtpRtcpImpl::SupportsRtxPayloadPadding() const {
RTC_DCHECK(rtp_sender_);
return rtp_sender_->packet_generator.SupportsRtxPayloadPadding();
}
std::vector<std::unique_ptr<RtpPacketToSend>>
ModuleRtpRtcpImpl::GeneratePadding(size_t target_size_bytes) {
RTC_DCHECK(rtp_sender_);
MutexLock lock(&rtp_sender_->sequencer_mutex);
return rtp_sender_->packet_generator.GeneratePadding(
target_size_bytes, rtp_sender_->packet_sender.MediaHasBeenSent(),
rtp_sender_->sequencer_.CanSendPaddingOnMediaSsrc());
}
std::vector<RtpSequenceNumberMap::Info>
ModuleRtpRtcpImpl::GetSentRtpPacketInfos(
rtc::ArrayView<const uint16_t> sequence_numbers) const {
RTC_DCHECK(rtp_sender_);
return rtp_sender_->packet_sender.GetSentRtpPacketInfos(sequence_numbers);
}
size_t ModuleRtpRtcpImpl::ExpectedPerPacketOverhead() const {
if (!rtp_sender_) {
return 0;
}
return rtp_sender_->packet_generator.ExpectedPerPacketOverhead();
}
void ModuleRtpRtcpImpl::OnPacketSendingThreadSwitched() {}
size_t ModuleRtpRtcpImpl::MaxRtpPacketSize() const {
RTC_DCHECK(rtp_sender_);
return rtp_sender_->packet_generator.MaxRtpPacketSize();
}
void ModuleRtpRtcpImpl::SetMaxRtpPacketSize(size_t rtp_packet_size) {
RTC_DCHECK_LE(rtp_packet_size, IP_PACKET_SIZE)
<< "rtp packet size too large: " << rtp_packet_size;
RTC_DCHECK_GT(rtp_packet_size, packet_overhead_)
<< "rtp packet size too small: " << rtp_packet_size;
rtcp_sender_.SetMaxRtpPacketSize(rtp_packet_size);
if (rtp_sender_) {
rtp_sender_->packet_generator.SetMaxRtpPacketSize(rtp_packet_size);
}
}
RtcpMode ModuleRtpRtcpImpl::RTCP() const {
return rtcp_sender_.Status();
}
// Configure RTCP status i.e on/off.
void ModuleRtpRtcpImpl::SetRTCPStatus(const RtcpMode method) {
rtcp_sender_.SetRTCPStatus(method);
}
int32_t ModuleRtpRtcpImpl::SetCNAME(absl::string_view c_name) {
return rtcp_sender_.SetCNAME(c_name);
}
absl::optional<TimeDelta> ModuleRtpRtcpImpl::LastRtt() const {
absl::optional<TimeDelta> rtt = rtcp_receiver_.LastRtt();
if (!rtt.has_value()) {
MutexLock lock(&mutex_rtt_);
if (rtt_ms_ > 0) {
rtt = TimeDelta::Millis(rtt_ms_);
}
}
return rtt;
}
int64_t ModuleRtpRtcpImpl::ExpectedRetransmissionTimeMs() const {
int64_t expected_retransmission_time_ms = rtt_ms();
if (expected_retransmission_time_ms > 0) {
return expected_retransmission_time_ms;
}
// No rtt available (`kRtpRtcpRttProcessTimeMs` not yet passed?), so try to
// poll avg_rtt_ms directly from rtcp receiver.
if (absl::optional<TimeDelta> rtt = rtcp_receiver_.AverageRtt()) {
return rtt->ms();
}
return kDefaultExpectedRetransmissionTimeMs;
}
// Force a send of an RTCP packet.
// Normal SR and RR are triggered via the process function.
int32_t ModuleRtpRtcpImpl::SendRTCP(RTCPPacketType packet_type) {
return rtcp_sender_.SendRTCP(GetFeedbackState(), packet_type);
}
void ModuleRtpRtcpImpl::GetSendStreamDataCounters(
StreamDataCounters* rtp_counters,
StreamDataCounters* rtx_counters) const {
rtp_sender_->packet_sender.GetDataCounters(rtp_counters, rtx_counters);
}
// Received RTCP report.
std::vector<ReportBlockData> ModuleRtpRtcpImpl::GetLatestReportBlockData()
const {
return rtcp_receiver_.GetLatestReportBlockData();
}
absl::optional<RtpRtcpInterface::SenderReportStats>
ModuleRtpRtcpImpl::GetSenderReportStats() const {
return rtcp_receiver_.GetSenderReportStats();
}
absl::optional<RtpRtcpInterface::NonSenderRttStats>
ModuleRtpRtcpImpl::GetNonSenderRttStats() const {
// This is not implemented for this legacy class.
return absl::nullopt;
}
// (REMB) Receiver Estimated Max Bitrate.
void ModuleRtpRtcpImpl::SetRemb(int64_t bitrate_bps,
std::vector<uint32_t> ssrcs) {
rtcp_sender_.SetRemb(bitrate_bps, std::move(ssrcs));
}
void ModuleRtpRtcpImpl::UnsetRemb() {
rtcp_sender_.UnsetRemb();
}
void ModuleRtpRtcpImpl::SetExtmapAllowMixed(bool extmap_allow_mixed) {
rtp_sender_->packet_generator.SetExtmapAllowMixed(extmap_allow_mixed);
}
void ModuleRtpRtcpImpl::RegisterRtpHeaderExtension(absl::string_view uri,
int id) {
bool registered =
rtp_sender_->packet_generator.RegisterRtpHeaderExtension(uri, id);
RTC_CHECK(registered);
}
void ModuleRtpRtcpImpl::DeregisterSendRtpHeaderExtension(
absl::string_view uri) {
rtp_sender_->packet_generator.DeregisterRtpHeaderExtension(uri);
}
void ModuleRtpRtcpImpl::SetTmmbn(std::vector<rtcp::TmmbItem> bounding_set) {
rtcp_sender_.SetTmmbn(std::move(bounding_set));
}
// Send a Negative acknowledgment packet.
int32_t ModuleRtpRtcpImpl::SendNACK(const uint16_t* nack_list,
const uint16_t size) {
uint16_t nack_length = size;
uint16_t start_id = 0;
int64_t now_ms = clock_->TimeInMilliseconds();
if (TimeToSendFullNackList(now_ms)) {
nack_last_time_sent_full_ms_ = now_ms;
} else {
// Only send extended list.
if (nack_last_seq_number_sent_ == nack_list[size - 1]) {
// Last sequence number is the same, do not send list.
return 0;
}
// Send new sequence numbers.
for (int i = 0; i < size; ++i) {
if (nack_last_seq_number_sent_ == nack_list[i]) {
start_id = i + 1;
break;
}
}
nack_length = size - start_id;
}
// Our RTCP NACK implementation is limited to kRtcpMaxNackFields sequence
// numbers per RTCP packet.
if (nack_length > kRtcpMaxNackFields) {
nack_length = kRtcpMaxNackFields;
}
nack_last_seq_number_sent_ = nack_list[start_id + nack_length - 1];
return rtcp_sender_.SendRTCP(GetFeedbackState(), kRtcpNack, nack_length,
&nack_list[start_id]);
}
void ModuleRtpRtcpImpl::SendNack(
const std::vector<uint16_t>& sequence_numbers) {
rtcp_sender_.SendRTCP(GetFeedbackState(), kRtcpNack, sequence_numbers.size(),
sequence_numbers.data());
}
bool ModuleRtpRtcpImpl::TimeToSendFullNackList(int64_t now) const {
// Use RTT from RtcpRttStats class if provided.
int64_t rtt = rtt_ms();
if (rtt == 0) {
if (absl::optional<TimeDelta> average_rtt = rtcp_receiver_.AverageRtt()) {
rtt = average_rtt->ms();
}
}
const int64_t kStartUpRttMs = 100;
int64_t wait_time = 5 + ((rtt * 3) >> 1); // 5 + RTT * 1.5.
if (rtt == 0) {
wait_time = kStartUpRttMs;
}
// Send a full NACK list once within every `wait_time`.
return now - nack_last_time_sent_full_ms_ > wait_time;
}
// Store the sent packets, needed to answer to Negative acknowledgment requests.
void ModuleRtpRtcpImpl::SetStorePacketsStatus(const bool enable,
const uint16_t number_to_store) {
rtp_sender_->packet_history.SetStorePacketsStatus(
enable ? RtpPacketHistory::StorageMode::kStoreAndCull
: RtpPacketHistory::StorageMode::kDisabled,
number_to_store);
}
bool ModuleRtpRtcpImpl::StorePackets() const {
return rtp_sender_->packet_history.GetStorageMode() !=
RtpPacketHistory::StorageMode::kDisabled;
}
void ModuleRtpRtcpImpl::SendCombinedRtcpPacket(
std::vector<std::unique_ptr<rtcp::RtcpPacket>> rtcp_packets) {
rtcp_sender_.SendCombinedRtcpPacket(std::move(rtcp_packets));
}
int32_t ModuleRtpRtcpImpl::SendLossNotification(uint16_t last_decoded_seq_num,
uint16_t last_received_seq_num,
bool decodability_flag,
bool buffering_allowed) {
return rtcp_sender_.SendLossNotification(
GetFeedbackState(), last_decoded_seq_num, last_received_seq_num,
decodability_flag, buffering_allowed);
}
void ModuleRtpRtcpImpl::SetRemoteSSRC(const uint32_t ssrc) {
// Inform about the incoming SSRC.
rtcp_sender_.SetRemoteSSRC(ssrc);
rtcp_receiver_.SetRemoteSSRC(ssrc);
}
void ModuleRtpRtcpImpl::SetLocalSsrc(uint32_t local_ssrc) {
rtcp_receiver_.set_local_media_ssrc(local_ssrc);
rtcp_sender_.SetSsrc(local_ssrc);
}
RtpSendRates ModuleRtpRtcpImpl::GetSendRates() const {
return rtp_sender_->packet_sender.GetSendRates();
}
void ModuleRtpRtcpImpl::OnRequestSendReport() {
SendRTCP(kRtcpSr);
}
void ModuleRtpRtcpImpl::OnReceivedNack(
const std::vector<uint16_t>& nack_sequence_numbers) {
if (!rtp_sender_)
return;
if (!StorePackets() || nack_sequence_numbers.empty()) {
return;
}
// Use RTT from RtcpRttStats class if provided.
int64_t rtt = rtt_ms();
if (rtt == 0) {
if (absl::optional<TimeDelta> average_rtt = rtcp_receiver_.AverageRtt()) {
rtt = average_rtt->ms();
}
}
rtp_sender_->packet_generator.OnReceivedNack(nack_sequence_numbers, rtt);
}
void ModuleRtpRtcpImpl::OnReceivedRtcpReportBlocks(
const ReportBlockList& report_blocks) {
if (rtp_sender_) {
uint32_t ssrc = SSRC();
absl::optional<uint32_t> rtx_ssrc;
if (rtp_sender_->packet_generator.RtxStatus() != kRtxOff) {
rtx_ssrc = rtp_sender_->packet_generator.RtxSsrc();
}
for (const RTCPReportBlock& report_block : report_blocks) {
if (ssrc == report_block.source_ssrc) {
rtp_sender_->packet_generator.OnReceivedAckOnSsrc(
report_block.extended_highest_sequence_number);
} else if (rtx_ssrc && *rtx_ssrc == report_block.source_ssrc) {
rtp_sender_->packet_generator.OnReceivedAckOnRtxSsrc(
report_block.extended_highest_sequence_number);
}
}
}
}
void ModuleRtpRtcpImpl::set_rtt_ms(int64_t rtt_ms) {
{
MutexLock lock(&mutex_rtt_);
rtt_ms_ = rtt_ms;
}
if (rtp_sender_) {
rtp_sender_->packet_history.SetRtt(TimeDelta::Millis(rtt_ms));
}
}
int64_t ModuleRtpRtcpImpl::rtt_ms() const {
MutexLock lock(&mutex_rtt_);
return rtt_ms_;
}
void ModuleRtpRtcpImpl::SetVideoBitrateAllocation(
const VideoBitrateAllocation& bitrate) {
rtcp_sender_.SetVideoBitrateAllocation(bitrate);
}
RTPSender* ModuleRtpRtcpImpl::RtpSender() {
return rtp_sender_ ? &rtp_sender_->packet_generator : nullptr;
}
const RTPSender* ModuleRtpRtcpImpl::RtpSender() const {
return rtp_sender_ ? &rtp_sender_->packet_generator : nullptr;
}
} // namespace webrtc