blob: 31e4854839c84668aedbd664362a85afd7368db3 [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_impl2.h"
#include <string.h>
#include <algorithm>
#include <cstdint>
#include <memory>
#include <set>
#include <string>
#include <utility>
#include "api/transport/field_trial_based_config.h"
#include "modules/rtp_rtcp/source/rtcp_packet/dlrr.h"
#include "modules/rtp_rtcp/source/rtp_rtcp_config.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#ifdef _WIN32
// Disable warning C4355: 'this' : used in base member initializer list.
#pragma warning(disable : 4355)
#endif
namespace webrtc {
namespace {
const int64_t kRtpRtcpMaxIdleTimeProcessMs = 5;
const int64_t kDefaultExpectedRetransmissionTimeMs = 125;
constexpr TimeDelta kRttUpdateInterval = TimeDelta::Millis(1000);
} // namespace
ModuleRtpRtcpImpl2::RtpSenderContext::RtpSenderContext(
const RtpRtcpInterface::Configuration& config)
: packet_history(config.clock, config.enable_rtx_padding_prioritization),
packet_sender(config, &packet_history),
non_paced_sender(&packet_sender, this),
packet_generator(
config,
&packet_history,
config.paced_sender ? config.paced_sender : &non_paced_sender) {}
void ModuleRtpRtcpImpl2::RtpSenderContext::AssignSequenceNumber(
RtpPacketToSend* packet) {
packet_generator.AssignSequenceNumber(packet);
}
ModuleRtpRtcpImpl2::ModuleRtpRtcpImpl2(const Configuration& configuration)
: worker_queue_(TaskQueueBase::Current()),
rtcp_sender_(configuration),
rtcp_receiver_(configuration, this),
clock_(configuration.clock),
last_rtt_process_time_(clock_->TimeInMilliseconds()),
next_process_time_(clock_->TimeInMilliseconds() +
kRtpRtcpMaxIdleTimeProcessMs),
packet_overhead_(28), // IPV4 UDP.
nack_last_time_sent_full_ms_(0),
nack_last_seq_number_sent_(0),
remote_bitrate_(configuration.remote_bitrate_estimator),
rtt_stats_(configuration.rtt_stats),
rtt_ms_(0) {
RTC_DCHECK(worker_queue_);
process_thread_checker_.Detach();
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);
if (rtt_stats_) {
rtt_update_task_ = RepeatingTaskHandle::DelayedStart(
worker_queue_, kRttUpdateInterval, [this]() {
PeriodicUpdate();
return kRttUpdateInterval;
});
}
}
ModuleRtpRtcpImpl2::~ModuleRtpRtcpImpl2() {
RTC_DCHECK_RUN_ON(worker_queue_);
rtt_update_task_.Stop();
}
// static
std::unique_ptr<ModuleRtpRtcpImpl2> ModuleRtpRtcpImpl2::Create(
const Configuration& configuration) {
RTC_DCHECK(configuration.clock);
RTC_DCHECK(TaskQueueBase::Current());
return std::make_unique<ModuleRtpRtcpImpl2>(configuration);
}
// Returns the number of milliseconds until the module want a worker thread
// to call Process.
int64_t ModuleRtpRtcpImpl2::TimeUntilNextProcess() {
RTC_DCHECK_RUN_ON(&process_thread_checker_);
return std::max<int64_t>(0,
next_process_time_ - clock_->TimeInMilliseconds());
}
// Process any pending tasks such as timeouts (non time critical events).
void ModuleRtpRtcpImpl2::Process() {
RTC_DCHECK_RUN_ON(&process_thread_checker_);
const Timestamp now = clock_->CurrentTime();
// TODO(bugs.webrtc.org/11581): Figure out why we need to call Process() 200
// times a second.
next_process_time_ = now.ms() + kRtpRtcpMaxIdleTimeProcessMs;
// TODO(bugs.webrtc.org/11581): once we don't use Process() to trigger
// calls to SendRTCP(), the only remaining timer will require remote_bitrate_
// to be not null. In that case, we can disable the timer when it is null.
if (remote_bitrate_ && rtcp_sender_.Sending() && rtcp_sender_.TMMBR()) {
unsigned int target_bitrate = 0;
std::vector<unsigned int> ssrcs;
if (remote_bitrate_->LatestEstimate(&ssrcs, &target_bitrate)) {
if (!ssrcs.empty()) {
target_bitrate = target_bitrate / ssrcs.size();
}
rtcp_sender_.SetTargetBitrate(target_bitrate);
}
}
// TODO(bugs.webrtc.org/11581): Run this on a separate set of delayed tasks
// based off of next_time_to_send_rtcp_ in RTCPSender.
if (rtcp_sender_.TimeToSendRTCPReport())
rtcp_sender_.SendRTCP(GetFeedbackState(), kRtcpReport);
}
void ModuleRtpRtcpImpl2::SetRtxSendStatus(int mode) {
rtp_sender_->packet_generator.SetRtxStatus(mode);
}
int ModuleRtpRtcpImpl2::RtxSendStatus() const {
return rtp_sender_ ? rtp_sender_->packet_generator.RtxStatus() : kRtxOff;
}
void ModuleRtpRtcpImpl2::SetRtxSendPayloadType(int payload_type,
int associated_payload_type) {
rtp_sender_->packet_generator.SetRtxPayloadType(payload_type,
associated_payload_type);
}
absl::optional<uint32_t> ModuleRtpRtcpImpl2::RtxSsrc() const {
return rtp_sender_ ? rtp_sender_->packet_generator.RtxSsrc() : absl::nullopt;
}
absl::optional<uint32_t> ModuleRtpRtcpImpl2::FlexfecSsrc() const {
if (rtp_sender_) {
return rtp_sender_->packet_generator.FlexfecSsrc();
}
return absl::nullopt;
}
void ModuleRtpRtcpImpl2::IncomingRtcpPacket(const uint8_t* rtcp_packet,
const size_t length) {
rtcp_receiver_.IncomingPacket(rtcp_packet, length);
}
void ModuleRtpRtcpImpl2::RegisterSendPayloadFrequency(int payload_type,
int payload_frequency) {
rtcp_sender_.SetRtpClockRate(payload_type, payload_frequency);
}
int32_t ModuleRtpRtcpImpl2::DeRegisterSendPayload(const int8_t payload_type) {
return 0;
}
uint32_t ModuleRtpRtcpImpl2::StartTimestamp() const {
return rtp_sender_->packet_generator.TimestampOffset();
}
// Configure start timestamp, default is a random number.
void ModuleRtpRtcpImpl2::SetStartTimestamp(const uint32_t timestamp) {
rtcp_sender_.SetTimestampOffset(timestamp);
rtp_sender_->packet_generator.SetTimestampOffset(timestamp);
rtp_sender_->packet_sender.SetTimestampOffset(timestamp);
}
uint16_t ModuleRtpRtcpImpl2::SequenceNumber() const {
return rtp_sender_->packet_generator.SequenceNumber();
}
// Set SequenceNumber, default is a random number.
void ModuleRtpRtcpImpl2::SetSequenceNumber(const uint16_t seq_num) {
rtp_sender_->packet_generator.SetSequenceNumber(seq_num);
}
void ModuleRtpRtcpImpl2::SetRtpState(const RtpState& rtp_state) {
rtp_sender_->packet_generator.SetRtpState(rtp_state);
rtcp_sender_.SetTimestampOffset(rtp_state.start_timestamp);
}
void ModuleRtpRtcpImpl2::SetRtxState(const RtpState& rtp_state) {
rtp_sender_->packet_generator.SetRtxRtpState(rtp_state);
}
RtpState ModuleRtpRtcpImpl2::GetRtpState() const {
RtpState state = rtp_sender_->packet_generator.GetRtpState();
return state;
}
RtpState ModuleRtpRtcpImpl2::GetRtxState() const {
return rtp_sender_->packet_generator.GetRtxRtpState();
}
void ModuleRtpRtcpImpl2::SetRid(const std::string& rid) {
if (rtp_sender_) {
rtp_sender_->packet_generator.SetRid(rid);
}
}
void ModuleRtpRtcpImpl2::SetMid(const std::string& 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.
}
void ModuleRtpRtcpImpl2::SetCsrcs(const std::vector<uint32_t>& csrcs) {
rtcp_sender_.SetCsrcs(csrcs);
rtp_sender_->packet_generator.SetCsrcs(csrcs);
}
// TODO(pbos): Handle media and RTX streams separately (separate RTCP
// feedbacks).
RTCPSender::FeedbackState ModuleRtpRtcpImpl2::GetFeedbackState() {
// TODO(bugs.webrtc.org/11581): Called by potentially multiple threads.
// Mostly "Send*" methods. Make sure it's only called on the
// construction thread.
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_;
LastReceivedNTP(&state.last_rr_ntp_secs, &state.last_rr_ntp_frac,
&state.remote_sr);
state.last_xr_rtis = rtcp_receiver_.ConsumeReceivedXrReferenceTimeInfo();
return state;
}
// TODO(nisse): This method shouldn't be called for a receive-only
// stream. Delete rtp_sender_ check as soon as all applications are
// updated.
int32_t ModuleRtpRtcpImpl2::SetSendingStatus(const bool sending) {
if (rtcp_sender_.Sending() != sending) {
// Sends RTCP BYE when going from true to false
if (rtcp_sender_.SetSendingStatus(GetFeedbackState(), sending) != 0) {
RTC_LOG(LS_WARNING) << "Failed to send RTCP BYE";
}
}
return 0;
}
bool ModuleRtpRtcpImpl2::Sending() const {
return rtcp_sender_.Sending();
}
// TODO(nisse): This method shouldn't be called for a receive-only
// stream. Delete rtp_sender_ check as soon as all applications are
// updated.
void ModuleRtpRtcpImpl2::SetSendingMediaStatus(const bool sending) {
if (rtp_sender_) {
rtp_sender_->packet_generator.SetSendingMediaStatus(sending);
} else {
RTC_DCHECK(!sending);
}
}
bool ModuleRtpRtcpImpl2::SendingMedia() const {
return rtp_sender_ ? rtp_sender_->packet_generator.SendingMedia() : false;
}
bool ModuleRtpRtcpImpl2::IsAudioConfigured() const {
return rtp_sender_ ? rtp_sender_->packet_generator.IsAudioConfigured()
: false;
}
void ModuleRtpRtcpImpl2::SetAsPartOfAllocation(bool part_of_allocation) {
RTC_CHECK(rtp_sender_);
rtp_sender_->packet_sender.ForceIncludeSendPacketsInAllocation(
part_of_allocation);
}
bool ModuleRtpRtcpImpl2::OnSendingRtpFrame(uint32_t timestamp,
int64_t capture_time_ms,
int payload_type,
bool force_sender_report) {
if (!Sending())
return false;
rtcp_sender_.SetLastRtpTime(timestamp, capture_time_ms, payload_type);
// 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 ModuleRtpRtcpImpl2::TrySendPacket(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;
}
rtp_sender_->packet_sender.SendPacket(packet, pacing_info);
return true;
}
void ModuleRtpRtcpImpl2::SetFecProtectionParams(
const FecProtectionParams& delta_params,
const FecProtectionParams& key_params) {
RTC_DCHECK(rtp_sender_);
rtp_sender_->packet_sender.SetFecProtectionParameters(delta_params,
key_params);
}
std::vector<std::unique_ptr<RtpPacketToSend>>
ModuleRtpRtcpImpl2::FetchFecPackets() {
RTC_DCHECK(rtp_sender_);
auto fec_packets = rtp_sender_->packet_sender.FetchFecPackets();
if (!fec_packets.empty()) {
// Don't assign sequence numbers for FlexFEC packets.
const bool generate_sequence_numbers =
!rtp_sender_->packet_sender.FlexFecSsrc().has_value();
if (generate_sequence_numbers) {
for (auto& fec_packet : fec_packets) {
rtp_sender_->packet_generator.AssignSequenceNumber(fec_packet.get());
}
}
}
return fec_packets;
}
void ModuleRtpRtcpImpl2::OnPacketsAcknowledged(
rtc::ArrayView<const uint16_t> sequence_numbers) {
RTC_DCHECK(rtp_sender_);
rtp_sender_->packet_history.CullAcknowledgedPackets(sequence_numbers);
}
bool ModuleRtpRtcpImpl2::SupportsPadding() const {
RTC_DCHECK(rtp_sender_);
return rtp_sender_->packet_generator.SupportsPadding();
}
bool ModuleRtpRtcpImpl2::SupportsRtxPayloadPadding() const {
RTC_DCHECK(rtp_sender_);
return rtp_sender_->packet_generator.SupportsRtxPayloadPadding();
}
std::vector<std::unique_ptr<RtpPacketToSend>>
ModuleRtpRtcpImpl2::GeneratePadding(size_t target_size_bytes) {
RTC_DCHECK(rtp_sender_);
return rtp_sender_->packet_generator.GeneratePadding(
target_size_bytes, rtp_sender_->packet_sender.MediaHasBeenSent());
}
std::vector<RtpSequenceNumberMap::Info>
ModuleRtpRtcpImpl2::GetSentRtpPacketInfos(
rtc::ArrayView<const uint16_t> sequence_numbers) const {
RTC_DCHECK(rtp_sender_);
return rtp_sender_->packet_sender.GetSentRtpPacketInfos(sequence_numbers);
}
size_t ModuleRtpRtcpImpl2::ExpectedPerPacketOverhead() const {
if (!rtp_sender_) {
return 0;
}
return rtp_sender_->packet_generator.ExpectedPerPacketOverhead();
}
size_t ModuleRtpRtcpImpl2::MaxRtpPacketSize() const {
RTC_DCHECK(rtp_sender_);
return rtp_sender_->packet_generator.MaxRtpPacketSize();
}
void ModuleRtpRtcpImpl2::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 ModuleRtpRtcpImpl2::RTCP() const {
return rtcp_sender_.Status();
}
// Configure RTCP status i.e on/off.
void ModuleRtpRtcpImpl2::SetRTCPStatus(const RtcpMode method) {
rtcp_sender_.SetRTCPStatus(method);
}
int32_t ModuleRtpRtcpImpl2::SetCNAME(const char* c_name) {
return rtcp_sender_.SetCNAME(c_name);
}
int32_t ModuleRtpRtcpImpl2::RemoteNTP(uint32_t* received_ntpsecs,
uint32_t* received_ntpfrac,
uint32_t* rtcp_arrival_time_secs,
uint32_t* rtcp_arrival_time_frac,
uint32_t* rtcp_timestamp) const {
return rtcp_receiver_.NTP(received_ntpsecs, received_ntpfrac,
rtcp_arrival_time_secs, rtcp_arrival_time_frac,
rtcp_timestamp)
? 0
: -1;
}
// TODO(tommi): Check if |avg_rtt_ms|, |min_rtt_ms|, |max_rtt_ms| params are
// actually used in practice (some callers ask for it but don't use it). It
// could be that only |rtt| is needed and if so, then the fast path could be to
// just call rtt_ms() and rely on the calculation being done periodically.
int32_t ModuleRtpRtcpImpl2::RTT(const uint32_t remote_ssrc,
int64_t* rtt,
int64_t* avg_rtt,
int64_t* min_rtt,
int64_t* max_rtt) const {
int32_t ret = rtcp_receiver_.RTT(remote_ssrc, rtt, avg_rtt, min_rtt, max_rtt);
if (rtt && *rtt == 0) {
// Try to get RTT from RtcpRttStats class.
*rtt = rtt_ms();
}
return ret;
}
int64_t ModuleRtpRtcpImpl2::ExpectedRetransmissionTimeMs() const {
int64_t expected_retransmission_time_ms = rtt_ms();
if (expected_retransmission_time_ms > 0) {
return expected_retransmission_time_ms;
}
// No rtt available (|kRttUpdateInterval| not yet passed?), so try to
// poll avg_rtt_ms directly from rtcp receiver.
if (rtcp_receiver_.RTT(rtcp_receiver_.RemoteSSRC(), nullptr,
&expected_retransmission_time_ms, nullptr,
nullptr) == 0) {
return expected_retransmission_time_ms;
}
return kDefaultExpectedRetransmissionTimeMs;
}
// Force a send of an RTCP packet.
// Normal SR and RR are triggered via the process function.
int32_t ModuleRtpRtcpImpl2::SendRTCP(RTCPPacketType packet_type) {
return rtcp_sender_.SendRTCP(GetFeedbackState(), packet_type);
}
void ModuleRtpRtcpImpl2::SetRtcpXrRrtrStatus(bool enable) {
rtcp_receiver_.SetRtcpXrRrtrStatus(enable);
rtcp_sender_.SendRtcpXrReceiverReferenceTime(enable);
}
bool ModuleRtpRtcpImpl2::RtcpXrRrtrStatus() const {
return rtcp_sender_.RtcpXrReceiverReferenceTime();
}
void ModuleRtpRtcpImpl2::GetSendStreamDataCounters(
StreamDataCounters* rtp_counters,
StreamDataCounters* rtx_counters) const {
rtp_sender_->packet_sender.GetDataCounters(rtp_counters, rtx_counters);
}
// Received RTCP report.
int32_t ModuleRtpRtcpImpl2::RemoteRTCPStat(
std::vector<RTCPReportBlock>* receive_blocks) const {
return rtcp_receiver_.StatisticsReceived(receive_blocks);
}
std::vector<ReportBlockData> ModuleRtpRtcpImpl2::GetLatestReportBlockData()
const {
return rtcp_receiver_.GetLatestReportBlockData();
}
// (REMB) Receiver Estimated Max Bitrate.
void ModuleRtpRtcpImpl2::SetRemb(int64_t bitrate_bps,
std::vector<uint32_t> ssrcs) {
rtcp_sender_.SetRemb(bitrate_bps, std::move(ssrcs));
}
void ModuleRtpRtcpImpl2::UnsetRemb() {
rtcp_sender_.UnsetRemb();
}
void ModuleRtpRtcpImpl2::SetExtmapAllowMixed(bool extmap_allow_mixed) {
rtp_sender_->packet_generator.SetExtmapAllowMixed(extmap_allow_mixed);
}
void ModuleRtpRtcpImpl2::RegisterRtpHeaderExtension(absl::string_view uri,
int id) {
bool registered =
rtp_sender_->packet_generator.RegisterRtpHeaderExtension(uri, id);
RTC_CHECK(registered);
}
int32_t ModuleRtpRtcpImpl2::DeregisterSendRtpHeaderExtension(
const RTPExtensionType type) {
return rtp_sender_->packet_generator.DeregisterRtpHeaderExtension(type);
}
void ModuleRtpRtcpImpl2::DeregisterSendRtpHeaderExtension(
absl::string_view uri) {
rtp_sender_->packet_generator.DeregisterRtpHeaderExtension(uri);
}
void ModuleRtpRtcpImpl2::SetTmmbn(std::vector<rtcp::TmmbItem> bounding_set) {
rtcp_sender_.SetTmmbn(std::move(bounding_set));
}
// Send a Negative acknowledgment packet.
int32_t ModuleRtpRtcpImpl2::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 ModuleRtpRtcpImpl2::SendNack(
const std::vector<uint16_t>& sequence_numbers) {
rtcp_sender_.SendRTCP(GetFeedbackState(), kRtcpNack, sequence_numbers.size(),
sequence_numbers.data());
}
bool ModuleRtpRtcpImpl2::TimeToSendFullNackList(int64_t now) const {
// Use RTT from RtcpRttStats class if provided.
int64_t rtt = rtt_ms();
if (rtt == 0) {
rtcp_receiver_.RTT(rtcp_receiver_.RemoteSSRC(), NULL, &rtt, NULL, NULL);
}
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 ModuleRtpRtcpImpl2::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 ModuleRtpRtcpImpl2::StorePackets() const {
return rtp_sender_->packet_history.GetStorageMode() !=
RtpPacketHistory::StorageMode::kDisabled;
}
void ModuleRtpRtcpImpl2::SendCombinedRtcpPacket(
std::vector<std::unique_ptr<rtcp::RtcpPacket>> rtcp_packets) {
rtcp_sender_.SendCombinedRtcpPacket(std::move(rtcp_packets));
}
int32_t ModuleRtpRtcpImpl2::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 ModuleRtpRtcpImpl2::SetRemoteSSRC(const uint32_t ssrc) {
// Inform about the incoming SSRC.
rtcp_sender_.SetRemoteSSRC(ssrc);
rtcp_receiver_.SetRemoteSSRC(ssrc);
}
// TODO(nisse): Delete video_rate amd fec_rate arguments.
void ModuleRtpRtcpImpl2::BitrateSent(uint32_t* total_rate,
uint32_t* video_rate,
uint32_t* fec_rate,
uint32_t* nack_rate) const {
RTC_DCHECK_RUN_ON(worker_queue_);
RtpSendRates send_rates = rtp_sender_->packet_sender.GetSendRates();
*total_rate = send_rates.Sum().bps<uint32_t>();
if (video_rate)
*video_rate = 0;
if (fec_rate)
*fec_rate = 0;
*nack_rate = send_rates[RtpPacketMediaType::kRetransmission].bps<uint32_t>();
}
RtpSendRates ModuleRtpRtcpImpl2::GetSendRates() const {
RTC_DCHECK_RUN_ON(worker_queue_);
return rtp_sender_->packet_sender.GetSendRates();
}
void ModuleRtpRtcpImpl2::OnRequestSendReport() {
SendRTCP(kRtcpSr);
}
void ModuleRtpRtcpImpl2::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) {
rtcp_receiver_.RTT(rtcp_receiver_.RemoteSSRC(), NULL, &rtt, NULL, NULL);
}
rtp_sender_->packet_generator.OnReceivedNack(nack_sequence_numbers, rtt);
}
void ModuleRtpRtcpImpl2::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);
}
}
}
}
bool ModuleRtpRtcpImpl2::LastReceivedNTP(
uint32_t* rtcp_arrival_time_secs, // When we got the last report.
uint32_t* rtcp_arrival_time_frac,
uint32_t* remote_sr) const {
// Remote SR: NTP inside the last received (mid 16 bits from sec and frac).
uint32_t ntp_secs = 0;
uint32_t ntp_frac = 0;
if (!rtcp_receiver_.NTP(&ntp_secs, &ntp_frac, rtcp_arrival_time_secs,
rtcp_arrival_time_frac, NULL)) {
return false;
}
*remote_sr =
((ntp_secs & 0x0000ffff) << 16) + ((ntp_frac & 0xffff0000) >> 16);
return true;
}
void ModuleRtpRtcpImpl2::set_rtt_ms(int64_t rtt_ms) {
RTC_DCHECK_RUN_ON(worker_queue_);
{
MutexLock lock(&mutex_rtt_);
rtt_ms_ = rtt_ms;
}
if (rtp_sender_) {
rtp_sender_->packet_history.SetRtt(rtt_ms);
}
}
int64_t ModuleRtpRtcpImpl2::rtt_ms() const {
MutexLock lock(&mutex_rtt_);
return rtt_ms_;
}
void ModuleRtpRtcpImpl2::SetVideoBitrateAllocation(
const VideoBitrateAllocation& bitrate) {
rtcp_sender_.SetVideoBitrateAllocation(bitrate);
}
RTPSender* ModuleRtpRtcpImpl2::RtpSender() {
return rtp_sender_ ? &rtp_sender_->packet_generator : nullptr;
}
const RTPSender* ModuleRtpRtcpImpl2::RtpSender() const {
return rtp_sender_ ? &rtp_sender_->packet_generator : nullptr;
}
void ModuleRtpRtcpImpl2::PeriodicUpdate() {
RTC_DCHECK_RUN_ON(worker_queue_);
Timestamp check_since = clock_->CurrentTime() - kRttUpdateInterval;
absl::optional<TimeDelta> rtt =
rtcp_receiver_.OnPeriodicRttUpdate(check_since, rtcp_sender_.Sending());
if (rtt) {
rtt_stats_->OnRttUpdate(rtt->ms());
set_rtt_ms(rtt->ms());
}
// kTmmbrTimeoutIntervalMs is 25 seconds, so an order of seconds.
// Instead of this polling approach, consider having an optional timer in the
// RTCPReceiver class that is started/stopped based on the state of
// rtcp_sender_.TMMBR().
if (rtcp_sender_.TMMBR() && rtcp_receiver_.UpdateTmmbrTimers())
rtcp_receiver_.NotifyTmmbrUpdated();
}
} // namespace webrtc