blob: ddac5c98d6171310f69dbf7d3b699165ff18332e [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 "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 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),
packet_sender(config, &packet_history),
non_paced_sender(&packet_sender),
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);
RTC_LOG(LS_ERROR)
<< "*********** USING WebRTC INTERNAL IMPLEMENTATION DETAILS ***********";
return std::make_unique<ModuleRtpRtcpImpl>(configuration);
}
ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration)
: rtcp_sender_(configuration),
rtcp_receiver_(configuration, this),
clock_(configuration.clock),
last_bitrate_process_time_(clock_->TimeInMilliseconds()),
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) {
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;
// Returns the number of milliseconds until the module want a worker thread
// to call Process.
int64_t ModuleRtpRtcpImpl::TimeUntilNextProcess() {
return std::max<int64_t>(0,
next_process_time_ - clock_->TimeInMilliseconds());
}
// Process any pending tasks such as timeouts (non time critical events).
void ModuleRtpRtcpImpl::Process() {
const int64_t now = clock_->TimeInMilliseconds();
// TODO(bugs.webrtc.org/11581): Figure out why we need to call Process() 200
// times a second.
next_process_time_ = now + kRtpRtcpMaxIdleTimeProcessMs;
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): Is this a bug? At the top of the function,
// next_process_time_ is incremented by 5ms, here we effectively do a
// std::min() of (now + 5ms, now + 10ms). Seems like this is a no-op?
next_process_time_ =
std::min(next_process_time_, now + kRtpRtcpBitrateProcessTimeMs);
}
}
// 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 &&
rtcp_receiver_.LastReceivedReportBlockMs() > last_rtt_process_time_) {
std::vector<RTCPReportBlock> receive_blocks;
rtcp_receiver_.StatisticsReceived(&receive_blocks);
int64_t max_rtt = 0;
for (std::vector<RTCPReportBlock>::iterator it = receive_blocks.begin();
it != receive_blocks.end(); ++it) {
int64_t rtt = 0;
rtcp_receiver_.RTT(it->sender_ssrc, &rtt, NULL, NULL, NULL);
max_rtt = (rtt > max_rtt) ? rtt : max_rtt;
}
// Report the rtt.
if (rtt_stats_ && max_rtt != 0)
rtt_stats_->OnRttUpdate(max_rtt);
}
// 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.";
}
if (remote_bitrate_ && 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);
}
}
} 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;
// TODO(bugs.webrtc.org/11581): Is this a bug? At the top of the function,
// next_process_time_ is incremented by 5ms, here we effectively do a
// std::min() of (now + 5ms, now + 1000ms). Seems like this is a no-op?
next_process_time_ = std::min(
next_process_time_, last_rtt_process_time_ + kRtpRtcpRttProcessTimeMs);
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 (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(const uint8_t* rtcp_packet,
const size_t length) {
rtcp_receiver_.IncomingPacket(rtcp_packet, length);
}
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 {
return rtp_sender_->packet_generator.SequenceNumber();
}
// Set SequenceNumber, default is a random number.
void ModuleRtpRtcpImpl::SetSequenceNumber(const uint16_t seq_num) {
rtp_sender_->packet_generator.SetSequenceNumber(seq_num);
}
void ModuleRtpRtcpImpl::SetRtpState(const RtpState& rtp_state) {
rtp_sender_->packet_generator.SetRtpState(rtp_state);
rtcp_sender_.SetTimestampOffset(rtp_state.start_timestamp);
}
void ModuleRtpRtcpImpl::SetRtxState(const RtpState& rtp_state) {
rtp_sender_->packet_generator.SetRtxRtpState(rtp_state);
}
RtpState ModuleRtpRtcpImpl::GetRtpState() const {
RtpState state = rtp_sender_->packet_generator.GetRtpState();
return state;
}
RtpState ModuleRtpRtcpImpl::GetRtxState() const {
return rtp_sender_->packet_generator.GetRtxRtpState();
}
void ModuleRtpRtcpImpl::SetRid(const std::string& rid) {
if (rtp_sender_) {
rtp_sender_->packet_generator.SetRid(rid);
}
}
void ModuleRtpRtcpImpl::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 ModuleRtpRtcpImpl::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 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_;
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 ModuleRtpRtcpImpl::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 ModuleRtpRtcpImpl::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 ModuleRtpRtcpImpl::SetSendingMediaStatus(const bool sending) {
if (rtp_sender_) {
rtp_sender_->packet_generator.SetSendingMediaStatus(sending);
} else {
RTC_DCHECK(!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;
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 ModuleRtpRtcpImpl::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 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::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_);
return rtp_sender_->packet_generator.GeneratePadding(
target_size_bytes, rtp_sender_->packet_sender.MediaHasBeenSent());
}
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();
}
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(const char* c_name) {
return rtcp_sender_.SetCNAME(c_name);
}
int32_t ModuleRtpRtcpImpl::AddMixedCNAME(uint32_t ssrc, const char* c_name) {
return rtcp_sender_.AddMixedCNAME(ssrc, c_name);
}
int32_t ModuleRtpRtcpImpl::RemoveMixedCNAME(const uint32_t ssrc) {
return rtcp_sender_.RemoveMixedCNAME(ssrc);
}
int32_t ModuleRtpRtcpImpl::RemoteCNAME(const uint32_t remote_ssrc,
char c_name[RTCP_CNAME_SIZE]) const {
return rtcp_receiver_.CNAME(remote_ssrc, c_name);
}
int32_t ModuleRtpRtcpImpl::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;
}
// Get RoundTripTime.
int32_t ModuleRtpRtcpImpl::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 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 (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 ModuleRtpRtcpImpl::SendRTCP(RTCPPacketType packet_type) {
return rtcp_sender_.SendRTCP(GetFeedbackState(), packet_type);
}
int32_t ModuleRtpRtcpImpl::SetRTCPApplicationSpecificData(
const uint8_t sub_type,
const uint32_t name,
const uint8_t* data,
const uint16_t length) {
RTC_NOTREACHED() << "Not implemented";
return -1;
}
// TODO(asapersson): Replace this method with the one below.
int32_t ModuleRtpRtcpImpl::DataCountersRTP(size_t* bytes_sent,
uint32_t* packets_sent) const {
StreamDataCounters rtp_stats;
StreamDataCounters rtx_stats;
rtp_sender_->packet_sender.GetDataCounters(&rtp_stats, &rtx_stats);
if (bytes_sent) {
// TODO(http://crbug.com/webrtc/10525): Bytes sent should only include
// payload bytes, not header and padding bytes.
*bytes_sent = rtp_stats.transmitted.payload_bytes +
rtp_stats.transmitted.padding_bytes +
rtp_stats.transmitted.header_bytes +
rtx_stats.transmitted.payload_bytes +
rtx_stats.transmitted.padding_bytes +
rtx_stats.transmitted.header_bytes;
}
if (packets_sent) {
*packets_sent =
rtp_stats.transmitted.packets + rtx_stats.transmitted.packets;
}
return 0;
}
void ModuleRtpRtcpImpl::GetSendStreamDataCounters(
StreamDataCounters* rtp_counters,
StreamDataCounters* rtx_counters) const {
rtp_sender_->packet_sender.GetDataCounters(rtp_counters, rtx_counters);
}
// Received RTCP report.
int32_t ModuleRtpRtcpImpl::RemoteRTCPStat(
std::vector<RTCPReportBlock>* receive_blocks) const {
return rtcp_receiver_.StatisticsReceived(receive_blocks);
}
std::vector<ReportBlockData> ModuleRtpRtcpImpl::GetLatestReportBlockData()
const {
return rtcp_receiver_.GetLatestReportBlockData();
}
// (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);
}
int32_t ModuleRtpRtcpImpl::RegisterSendRtpHeaderExtension(
const RTPExtensionType type,
const uint8_t id) {
return rtp_sender_->packet_generator.RegisterRtpHeaderExtension(type, id);
}
void ModuleRtpRtcpImpl::RegisterRtpHeaderExtension(absl::string_view uri,
int id) {
bool registered =
rtp_sender_->packet_generator.RegisterRtpHeaderExtension(uri, id);
RTC_CHECK(registered);
}
int32_t ModuleRtpRtcpImpl::DeregisterSendRtpHeaderExtension(
const RTPExtensionType type) {
return rtp_sender_->packet_generator.DeregisterRtpHeaderExtension(type);
}
void ModuleRtpRtcpImpl::DeregisterSendRtpHeaderExtension(
absl::string_view uri) {
rtp_sender_->packet_generator.DeregisterRtpHeaderExtension(uri);
}
// (TMMBR) Temporary Max Media Bit Rate.
bool ModuleRtpRtcpImpl::TMMBR() const {
return rtcp_sender_.TMMBR();
}
void ModuleRtpRtcpImpl::SetTMMBRStatus(const bool enable) {
rtcp_sender_.SetTMMBRStatus(enable);
}
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) {
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 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);
}
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) {
rtcp_receiver_.RTT(rtcp_receiver_.RemoteSSRC(), NULL, &rtt, NULL, NULL);
}
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);
}
}
}
}
bool ModuleRtpRtcpImpl::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 ModuleRtpRtcpImpl::set_rtt_ms(int64_t rtt_ms) {
{
MutexLock lock(&mutex_rtt_);
rtt_ms_ = rtt_ms;
}
if (rtp_sender_) {
rtp_sender_->packet_history.SetRtt(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;
}
DataRate ModuleRtpRtcpImpl::SendRate() const {
RTC_DCHECK(rtp_sender_);
return rtp_sender_->packet_sender.GetSendRates().Sum();
}
DataRate ModuleRtpRtcpImpl::NackOverheadRate() const {
RTC_DCHECK(rtp_sender_);
return rtp_sender_->packet_sender
.GetSendRates()[RtpPacketMediaType::kRetransmission];
}
} // namespace webrtc