blob: d5d4e2f32adbe736dc32d0f68237128df5671bab [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 "webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h"
#include <string.h>
#include <set>
#include <string>
#include "webrtc/base/checks.h"
#include "webrtc/base/logging.h"
#include "webrtc/common_types.h"
#include "webrtc/config.h"
#include "webrtc/system_wrappers/include/trace.h"
#ifdef _WIN32
// Disable warning C4355: 'this' : used in base member initializer list.
#pragma warning(disable : 4355)
#endif
namespace webrtc {
RTPExtensionType StringToRtpExtensionType(const std::string& extension) {
if (extension == RtpExtension::kTimestampOffsetUri)
return kRtpExtensionTransmissionTimeOffset;
if (extension == RtpExtension::kAudioLevelUri)
return kRtpExtensionAudioLevel;
if (extension == RtpExtension::kAbsSendTimeUri)
return kRtpExtensionAbsoluteSendTime;
if (extension == RtpExtension::kVideoRotationUri)
return kRtpExtensionVideoRotation;
if (extension == RtpExtension::kTransportSequenceNumberUri)
return kRtpExtensionTransportSequenceNumber;
if (extension == RtpExtension::kPlayoutDelayUri)
return kRtpExtensionPlayoutDelay;
RTC_NOTREACHED() << "Looking up unsupported RTP extension.";
return kRtpExtensionNone;
}
RtpRtcp::Configuration::Configuration()
: receive_statistics(NullObjectReceiveStatistics()) {}
RtpRtcp* RtpRtcp::CreateRtpRtcp(const RtpRtcp::Configuration& configuration) {
if (configuration.clock) {
return new ModuleRtpRtcpImpl(configuration);
} else {
// No clock implementation provided, use default clock.
RtpRtcp::Configuration configuration_copy;
memcpy(&configuration_copy, &configuration,
sizeof(RtpRtcp::Configuration));
configuration_copy.clock = Clock::GetRealTimeClock();
return new ModuleRtpRtcpImpl(configuration_copy);
}
}
// Deprecated.
int32_t RtpRtcp::SetFecParameters(const FecProtectionParams* delta_params,
const FecProtectionParams* key_params) {
RTC_DCHECK(delta_params);
RTC_DCHECK(key_params);
return SetFecParameters(*delta_params, *key_params) ? 0 : -1;
}
ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration)
: rtcp_sender_(configuration.audio,
configuration.clock,
configuration.receive_statistics,
configuration.rtcp_packet_type_counter_observer,
configuration.event_log,
configuration.outgoing_transport),
rtcp_receiver_(configuration.clock,
configuration.receiver_only,
configuration.rtcp_packet_type_counter_observer,
configuration.bandwidth_callback,
configuration.intra_frame_callback,
configuration.transport_feedback_callback,
configuration.bitrate_allocation_observer,
this),
clock_(configuration.clock),
audio_(configuration.audio),
last_process_time_(configuration.clock->TimeInMilliseconds()),
last_bitrate_process_time_(configuration.clock->TimeInMilliseconds()),
last_rtt_process_time_(configuration.clock->TimeInMilliseconds()),
packet_overhead_(28), // IPV4 UDP.
nack_last_time_sent_full_(0),
nack_last_time_sent_full_prev_(0),
nack_last_seq_number_sent_(0),
key_frame_req_method_(kKeyFrameReqPliRtcp),
remote_bitrate_(configuration.remote_bitrate_estimator),
rtt_stats_(configuration.rtt_stats),
rtt_ms_(0) {
if (!configuration.receiver_only) {
rtp_sender_.reset(new RTPSender(
configuration.audio,
configuration.clock,
configuration.outgoing_transport,
configuration.paced_sender,
configuration.flexfec_sender,
configuration.transport_sequence_number_allocator,
configuration.transport_feedback_callback,
configuration.send_bitrate_observer,
configuration.send_frame_count_observer,
configuration.send_side_delay_observer,
configuration.event_log,
configuration.send_packet_observer,
configuration.retransmission_rate_limiter,
configuration.overhead_observer));
// Make sure rtcp sender use same timestamp offset as rtp sender.
rtcp_sender_.SetTimestampOffset(rtp_sender_->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);
}
// Returns the number of milliseconds until the module want a worker thread
// to call Process.
int64_t ModuleRtpRtcpImpl::TimeUntilNextProcess() {
const int64_t now = clock_->TimeInMilliseconds();
const int64_t kRtpRtcpMaxIdleTimeProcessMs = 5;
return kRtpRtcpMaxIdleTimeProcessMs - (now - last_process_time_);
}
// Process any pending tasks such as timeouts (non time critical events).
void ModuleRtpRtcpImpl::Process() {
const int64_t now = clock_->TimeInMilliseconds();
last_process_time_ = now;
if (rtp_sender_) {
const int64_t kRtpRtcpBitrateProcessTimeMs = 10;
if (now >= last_bitrate_process_time_ + kRtpRtcpBitrateProcessTimeMs) {
rtp_sender_->ProcessBitrate();
last_bitrate_process_time_ = now;
}
}
const int64_t kRtpRtcpRttProcessTimeMs = 1000;
bool process_rtt = now >= last_rtt_process_time_ + kRtpRtcpRttProcessTimeMs;
if (rtcp_sender_.Sending()) {
// Process RTT if we have received a receiver report and we haven't
// processed RTT for at least |kRtpRtcpRttProcessTimeMs| milliseconds.
if (rtcp_receiver_.LastReceivedReceiverReport() >
last_rtt_process_time_ && process_rtt) {
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->remoteSSRC, &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.
int64_t rtcp_interval = RtcpReportInterval();
if (rtcp_receiver_.RtcpRrTimeout(rtcp_interval)) {
LOG_F(LS_WARNING) << "Timeout: No RTCP RR received.";
} else if (rtcp_receiver_.RtcpRrSequenceNumberTimeout(rtcp_interval)) {
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;
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_->SetRtxStatus(mode);
}
int ModuleRtpRtcpImpl::RtxSendStatus() const {
return rtp_sender_ ? rtp_sender_->RtxStatus() : kRtxOff;
}
void ModuleRtpRtcpImpl::SetRtxSsrc(uint32_t ssrc) {
rtp_sender_->SetRtxSsrc(ssrc);
}
void ModuleRtpRtcpImpl::SetRtxSendPayloadType(int payload_type,
int associated_payload_type) {
rtp_sender_->SetRtxPayloadType(payload_type, associated_payload_type);
}
rtc::Optional<uint32_t> ModuleRtpRtcpImpl::FlexfecSsrc() const {
return rtp_sender_->FlexfecSsrc();
}
int32_t ModuleRtpRtcpImpl::IncomingRtcpPacket(
const uint8_t* rtcp_packet,
const size_t length) {
return rtcp_receiver_.IncomingPacket(rtcp_packet, length) ? 0 : -1;
}
int32_t ModuleRtpRtcpImpl::RegisterSendPayload(
const CodecInst& voice_codec) {
return rtp_sender_->RegisterPayload(
voice_codec.plname, voice_codec.pltype, voice_codec.plfreq,
voice_codec.channels, (voice_codec.rate < 0) ? 0 : voice_codec.rate);
}
int32_t ModuleRtpRtcpImpl::RegisterSendPayload(const VideoCodec& video_codec) {
return rtp_sender_->RegisterPayload(video_codec.plName, video_codec.plType,
90000, 0, 0);
}
void ModuleRtpRtcpImpl::RegisterVideoSendPayload(int payload_type,
const char* payload_name) {
RTC_CHECK_EQ(
0, rtp_sender_->RegisterPayload(payload_name, payload_type, 90000, 0, 0));
}
int32_t ModuleRtpRtcpImpl::DeRegisterSendPayload(const int8_t payload_type) {
return rtp_sender_->DeRegisterSendPayload(payload_type);
}
uint32_t ModuleRtpRtcpImpl::StartTimestamp() const {
return rtp_sender_->TimestampOffset();
}
// Configure start timestamp, default is a random number.
void ModuleRtpRtcpImpl::SetStartTimestamp(const uint32_t timestamp) {
rtcp_sender_.SetTimestampOffset(timestamp);
rtp_sender_->SetTimestampOffset(timestamp);
}
uint16_t ModuleRtpRtcpImpl::SequenceNumber() const {
return rtp_sender_->SequenceNumber();
}
// Set SequenceNumber, default is a random number.
void ModuleRtpRtcpImpl::SetSequenceNumber(const uint16_t seq_num) {
rtp_sender_->SetSequenceNumber(seq_num);
}
void ModuleRtpRtcpImpl::SetRtpState(const RtpState& rtp_state) {
rtp_sender_->SetRtpState(rtp_state);
rtcp_sender_.SetTimestampOffset(rtp_state.start_timestamp);
}
void ModuleRtpRtcpImpl::SetRtxState(const RtpState& rtp_state) {
rtp_sender_->SetRtxRtpState(rtp_state);
}
RtpState ModuleRtpRtcpImpl::GetRtpState() const {
return rtp_sender_->GetRtpState();
}
RtpState ModuleRtpRtcpImpl::GetRtxState() const {
return rtp_sender_->GetRtxRtpState();
}
uint32_t ModuleRtpRtcpImpl::SSRC() const {
return rtcp_sender_.SSRC();
}
void ModuleRtpRtcpImpl::SetSSRC(const uint32_t ssrc) {
if (rtp_sender_) {
rtp_sender_->SetSSRC(ssrc);
}
rtcp_sender_.SetSSRC(ssrc);
SetRtcpReceiverSsrcs(ssrc);
}
void ModuleRtpRtcpImpl::SetCsrcs(const std::vector<uint32_t>& csrcs) {
rtcp_sender_.SetCsrcs(csrcs);
rtp_sender_->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_->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_->BitrateSent();
}
state.module = this;
LastReceivedNTP(&state.last_rr_ntp_secs,
&state.last_rr_ntp_frac,
&state.remote_sr);
state.has_last_xr_rr =
rtcp_receiver_.LastReceivedXrReferenceTimeInfo(&state.last_xr_rr);
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) {
LOG(LS_WARNING) << "Failed to send RTCP BYE";
}
if (sending && rtp_sender_) {
// Update Rtcp receiver config, to track Rtx config changes from
// the SetRtxStatus and SetRtxSsrc methods.
SetRtcpReceiverSsrcs(rtp_sender_->SSRC());
}
}
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_->SetSendingMediaStatus(sending);
} else {
RTC_DCHECK(!sending);
}
}
bool ModuleRtpRtcpImpl::SendingMedia() const {
return rtp_sender_ ? rtp_sender_->SendingMedia() : false;
}
bool ModuleRtpRtcpImpl::SendOutgoingData(
FrameType frame_type,
int8_t payload_type,
uint32_t time_stamp,
int64_t capture_time_ms,
const uint8_t* payload_data,
size_t payload_size,
const RTPFragmentationHeader* fragmentation,
const RTPVideoHeader* rtp_video_header,
uint32_t* transport_frame_id_out) {
rtcp_sender_.SetLastRtpTime(time_stamp, capture_time_ms);
// Make sure an RTCP report isn't queued behind a key frame.
if (rtcp_sender_.TimeToSendRTCPReport(kVideoFrameKey == frame_type)) {
rtcp_sender_.SendRTCP(GetFeedbackState(), kRtcpReport);
}
return rtp_sender_->SendOutgoingData(
frame_type, payload_type, time_stamp, capture_time_ms, payload_data,
payload_size, fragmentation, rtp_video_header, transport_frame_id_out);
}
bool ModuleRtpRtcpImpl::TimeToSendPacket(uint32_t ssrc,
uint16_t sequence_number,
int64_t capture_time_ms,
bool retransmission,
const PacedPacketInfo& pacing_info) {
return rtp_sender_->TimeToSendPacket(ssrc, sequence_number, capture_time_ms,
retransmission, pacing_info);
}
size_t ModuleRtpRtcpImpl::TimeToSendPadding(
size_t bytes,
const PacedPacketInfo& pacing_info) {
return rtp_sender_->TimeToSendPadding(bytes, pacing_info);
}
size_t ModuleRtpRtcpImpl::MaxPayloadSize() const {
return rtp_sender_->MaxPayloadSize();
}
size_t ModuleRtpRtcpImpl::MaxRtpPacketSize() const {
return rtp_sender_->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_->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;
}
// 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);
}
// Force a send of an RTCP packet.
// Normal SR and RR are triggered via the process function.
int32_t ModuleRtpRtcpImpl::SendCompoundRTCP(
const std::set<RTCPPacketType>& packet_types) {
return rtcp_sender_.SendCompoundRTCP(GetFeedbackState(), packet_types);
}
int32_t ModuleRtpRtcpImpl::SetRTCPApplicationSpecificData(
const uint8_t sub_type,
const uint32_t name,
const uint8_t* data,
const uint16_t length) {
return rtcp_sender_.SetApplicationSpecificData(sub_type, name, data, length);
}
// (XR) VOIP metric.
int32_t ModuleRtpRtcpImpl::SetRTCPVoIPMetrics(
const RTCPVoIPMetric* voip_metric) {
return rtcp_sender_.SetRTCPVoIPMetrics(voip_metric);
}
void ModuleRtpRtcpImpl::SetRtcpXrRrtrStatus(bool enable) {
rtcp_receiver_.SetRtcpXrRrtrStatus(enable);
rtcp_sender_.SendRtcpXrReceiverReferenceTime(enable);
}
bool ModuleRtpRtcpImpl::RtcpXrRrtrStatus() const {
return rtcp_sender_.RtcpXrReceiverReferenceTime();
}
// 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_->GetDataCounters(&rtp_stats, &rtx_stats);
if (bytes_sent) {
*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_->GetDataCounters(rtp_counters, rtx_counters);
}
void ModuleRtpRtcpImpl::GetRtpPacketLossStats(
bool outgoing,
uint32_t ssrc,
struct RtpPacketLossStats* loss_stats) const {
if (!loss_stats) return;
const PacketLossStats* stats_source = NULL;
if (outgoing) {
if (SSRC() == ssrc) {
stats_source = &send_loss_stats_;
}
} else {
if (rtcp_receiver_.RemoteSSRC() == ssrc) {
stats_source = &receive_loss_stats_;
}
}
if (stats_source) {
loss_stats->single_packet_loss_count =
stats_source->GetSingleLossCount();
loss_stats->multiple_packet_loss_event_count =
stats_source->GetMultipleLossEventCount();
loss_stats->multiple_packet_loss_packet_count =
stats_source->GetMultipleLossPacketCount();
}
}
int32_t ModuleRtpRtcpImpl::RemoteRTCPStat(RTCPSenderInfo* sender_info) {
return rtcp_receiver_.SenderInfoReceived(sender_info);
}
// Received RTCP report.
int32_t ModuleRtpRtcpImpl::RemoteRTCPStat(
std::vector<RTCPReportBlock>* receive_blocks) const {
return rtcp_receiver_.StatisticsReceived(receive_blocks);
}
// (REMB) Receiver Estimated Max Bitrate.
bool ModuleRtpRtcpImpl::REMB() const {
return rtcp_sender_.REMB();
}
void ModuleRtpRtcpImpl::SetREMBStatus(const bool enable) {
rtcp_sender_.SetREMBStatus(enable);
}
void ModuleRtpRtcpImpl::SetREMBData(const uint32_t bitrate,
const std::vector<uint32_t>& ssrcs) {
rtcp_sender_.SetREMBData(bitrate, ssrcs);
}
int32_t ModuleRtpRtcpImpl::RegisterSendRtpHeaderExtension(
const RTPExtensionType type,
const uint8_t id) {
return rtp_sender_->RegisterRtpHeaderExtension(type, id);
}
int32_t ModuleRtpRtcpImpl::DeregisterSendRtpHeaderExtension(
const RTPExtensionType type) {
return rtp_sender_->DeregisterRtpHeaderExtension(type);
}
bool ModuleRtpRtcpImpl::HasBweExtensions() const {
return rtp_sender_->IsRtpHeaderExtensionRegistered(
kRtpExtensionTransportSequenceNumber) ||
rtp_sender_->IsRtpHeaderExtensionRegistered(
kRtpExtensionAbsoluteSendTime) ||
rtp_sender_->IsRtpHeaderExtensionRegistered(
kRtpExtensionTransmissionTimeOffset);
}
// (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));
}
// Returns the currently configured retransmission mode.
int ModuleRtpRtcpImpl::SelectiveRetransmissions() const {
return rtp_sender_->SelectiveRetransmissions();
}
// Enable or disable a retransmission mode, which decides which packets will
// be retransmitted if NACKed.
int ModuleRtpRtcpImpl::SetSelectiveRetransmissions(uint8_t settings) {
return rtp_sender_->SetSelectiveRetransmissions(settings);
}
// Send a Negative acknowledgment packet.
int32_t ModuleRtpRtcpImpl::SendNACK(const uint16_t* nack_list,
const uint16_t size) {
for (int i = 0; i < size; ++i) {
receive_loss_stats_.AddLostPacket(nack_list[i]);
}
uint16_t nack_length = size;
uint16_t start_id = 0;
int64_t now = clock_->TimeInMilliseconds();
if (TimeToSendFullNackList(now)) {
nack_last_time_sent_full_ = now;
nack_last_time_sent_full_prev_ = now;
} 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|.
if (rtt_stats_) {
return now - nack_last_time_sent_full_ > wait_time;
}
return now - nack_last_time_sent_full_prev_ > 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_->SetStorePacketsStatus(enable, number_to_store);
}
bool ModuleRtpRtcpImpl::StorePackets() const {
return rtp_sender_->StorePackets();
}
void ModuleRtpRtcpImpl::RegisterRtcpStatisticsCallback(
RtcpStatisticsCallback* callback) {
rtcp_receiver_.RegisterRtcpStatisticsCallback(callback);
}
RtcpStatisticsCallback* ModuleRtpRtcpImpl::GetRtcpStatisticsCallback() {
return rtcp_receiver_.GetRtcpStatisticsCallback();
}
bool ModuleRtpRtcpImpl::SendFeedbackPacket(
const rtcp::TransportFeedback& packet) {
return rtcp_sender_.SendFeedbackPacket(packet);
}
// Send a TelephoneEvent tone using RFC 2833 (4733).
int32_t ModuleRtpRtcpImpl::SendTelephoneEventOutband(
const uint8_t key,
const uint16_t time_ms,
const uint8_t level) {
return rtp_sender_->SendTelephoneEvent(key, time_ms, level);
}
int32_t ModuleRtpRtcpImpl::SetAudioPacketSize(
const uint16_t packet_size_samples) {
return audio_ ? 0 : -1;
}
int32_t ModuleRtpRtcpImpl::SetAudioLevel(
const uint8_t level_d_bov) {
return rtp_sender_->SetAudioLevel(level_d_bov);
}
int32_t ModuleRtpRtcpImpl::SetKeyFrameRequestMethod(
const KeyFrameRequestMethod method) {
key_frame_req_method_ = method;
return 0;
}
int32_t ModuleRtpRtcpImpl::RequestKeyFrame() {
switch (key_frame_req_method_) {
case kKeyFrameReqPliRtcp:
return SendRTCP(kRtcpPli);
case kKeyFrameReqFirRtcp:
return SendRTCP(kRtcpFir);
}
return -1;
}
void ModuleRtpRtcpImpl::SetUlpfecConfig(int red_payload_type,
int ulpfec_payload_type) {
rtp_sender_->SetUlpfecConfig(red_payload_type, ulpfec_payload_type);
}
bool ModuleRtpRtcpImpl::SetFecParameters(
const FecProtectionParams& delta_params,
const FecProtectionParams& key_params) {
return rtp_sender_->SetFecParameters(delta_params, key_params);
}
void ModuleRtpRtcpImpl::SetRemoteSSRC(const uint32_t ssrc) {
// Inform about the incoming SSRC.
rtcp_sender_.SetRemoteSSRC(ssrc);
rtcp_receiver_.SetRemoteSSRC(ssrc);
}
void ModuleRtpRtcpImpl::BitrateSent(uint32_t* total_rate,
uint32_t* video_rate,
uint32_t* fec_rate,
uint32_t* nack_rate) const {
*total_rate = rtp_sender_->BitrateSent();
*video_rate = rtp_sender_->VideoBitrateSent();
*fec_rate = rtp_sender_->FecOverheadRate();
*nack_rate = rtp_sender_->NackOverheadRate();
}
void ModuleRtpRtcpImpl::OnRequestSendReport() {
SendRTCP(kRtcpSr);
}
void ModuleRtpRtcpImpl::OnReceivedNack(
const std::vector<uint16_t>& nack_sequence_numbers) {
if (!rtp_sender_)
return;
for (uint16_t nack_sequence_number : nack_sequence_numbers) {
send_loss_stats_.AddLostPacket(nack_sequence_number);
}
if (!rtp_sender_->StorePackets() ||
nack_sequence_numbers.size() == 0) {
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_->OnReceivedNack(nack_sequence_numbers, rtt);
}
void ModuleRtpRtcpImpl::OnReceivedRtcpReportBlocks(
const ReportBlockList& report_blocks) {
if (rtp_sender_)
rtp_sender_->OnReceivedRtcpReportBlocks(report_blocks);
}
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;
}
// Called from RTCPsender.
std::vector<rtcp::TmmbItem> ModuleRtpRtcpImpl::BoundingSet(bool* tmmbr_owner) {
return rtcp_receiver_.BoundingSet(tmmbr_owner);
}
int64_t ModuleRtpRtcpImpl::RtcpReportInterval() {
if (audio_)
return RTCP_INTERVAL_AUDIO_MS;
else
return RTCP_INTERVAL_VIDEO_MS;
}
void ModuleRtpRtcpImpl::SetRtcpReceiverSsrcs(uint32_t main_ssrc) {
std::set<uint32_t> ssrcs;
ssrcs.insert(main_ssrc);
if (RtxSendStatus() != kRtxOff)
ssrcs.insert(rtp_sender_->RtxSsrc());
rtcp_receiver_.SetSsrcs(main_ssrc, ssrcs);
}
void ModuleRtpRtcpImpl::set_rtt_ms(int64_t rtt_ms) {
rtc::CritScope cs(&critical_section_rtt_);
rtt_ms_ = rtt_ms;
}
int64_t ModuleRtpRtcpImpl::rtt_ms() const {
rtc::CritScope cs(&critical_section_rtt_);
return rtt_ms_;
}
void ModuleRtpRtcpImpl::RegisterSendChannelRtpStatisticsCallback(
StreamDataCountersCallback* callback) {
rtp_sender_->RegisterRtpStatisticsCallback(callback);
}
StreamDataCountersCallback*
ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const {
return rtp_sender_->GetRtpStatisticsCallback();
}
void ModuleRtpRtcpImpl::SetVideoBitrateAllocation(
const BitrateAllocation& bitrate) {
rtcp_sender_.SetVideoBitrateAllocation(bitrate);
}
} // namespace webrtc