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webrtc_m130/modules/rtp_rtcp/source/rtp_sender.cc
Steve Anton 296a0ce4c7 Add MID sending to RTPSender 
This CL adds the ability to configure RTPSender to include the
MID header extension when sending packets. The MID will be
included on every packet at the start of the stream until an RTCP
acknoledgment is received for that SSRC at which point it will
stop being included. The MID will be included on regular RTP
streams as well as RTX streams.

Bug: webrtc:4050
Change-Id: Ie27ebee1cd00a67f2b931f5363788f523e3e684f
Reviewed-on: https://webrtc-review.googlesource.com/60582
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22574}
2018-03-23 01:50:45 +00:00

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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "modules/rtp_rtcp/source/rtp_sender.h"
#include <algorithm>
#include <string>
#include <utility>
#include "logging/rtc_event_log/events/rtc_event_rtp_packet_outgoing.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "modules/remote_bitrate_estimator/test/bwe_test_logging.h"
#include "modules/rtp_rtcp/include/rtp_cvo.h"
#include "modules/rtp_rtcp/source/byte_io.h"
#include "modules/rtp_rtcp/source/playout_delay_oracle.h"
#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
#include "modules/rtp_rtcp/source/rtp_sender_audio.h"
#include "modules/rtp_rtcp/source/rtp_sender_video.h"
#include "modules/rtp_rtcp/source/time_util.h"
#include "rtc_base/arraysize.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/safe_minmax.h"
#include "rtc_base/ptr_util.h"
#include "rtc_base/rate_limiter.h"
#include "rtc_base/timeutils.h"
#include "rtc_base/trace_event.h"
#include "system_wrappers/include/field_trial.h"
namespace webrtc {
namespace {
// Max in the RFC 3550 is 255 bytes, we limit it to be modulus 32 for SRTP.
constexpr size_t kMaxPaddingLength = 224;
constexpr size_t kMinAudioPaddingLength = 50;
constexpr int kSendSideDelayWindowMs = 1000;
constexpr size_t kRtpHeaderLength = 12;
constexpr uint16_t kMaxInitRtpSeqNumber = 32767; // 2^15 -1.
constexpr uint32_t kTimestampTicksPerMs = 90;
constexpr int kBitrateStatisticsWindowMs = 1000;
constexpr size_t kMinFlexfecPacketsToStoreForPacing = 50;
template <typename Extension>
constexpr RtpExtensionSize CreateExtensionSize() {
return {Extension::kId, Extension::kValueSizeBytes};
}
// Size info for header extensions that might be used in padding or FEC packets.
constexpr RtpExtensionSize kFecOrPaddingExtensionSizes[] = {
CreateExtensionSize<AbsoluteSendTime>(),
CreateExtensionSize<TransmissionOffset>(),
CreateExtensionSize<TransportSequenceNumber>(),
CreateExtensionSize<PlayoutDelayLimits>(),
};
// Size info for header extensions that might be used in video packets.
constexpr RtpExtensionSize kVideoExtensionSizes[] = {
CreateExtensionSize<AbsoluteSendTime>(),
CreateExtensionSize<TransmissionOffset>(),
CreateExtensionSize<TransportSequenceNumber>(),
CreateExtensionSize<PlayoutDelayLimits>(),
CreateExtensionSize<VideoOrientation>(),
CreateExtensionSize<VideoContentTypeExtension>(),
CreateExtensionSize<VideoTimingExtension>(),
};
const char* FrameTypeToString(FrameType frame_type) {
switch (frame_type) {
case kEmptyFrame:
return "empty";
case kAudioFrameSpeech: return "audio_speech";
case kAudioFrameCN: return "audio_cn";
case kVideoFrameKey: return "video_key";
case kVideoFrameDelta: return "video_delta";
}
return "";
}
void CountPacket(RtpPacketCounter* counter, const RtpPacketToSend& packet) {
++counter->packets;
counter->header_bytes += packet.headers_size();
counter->padding_bytes += packet.padding_size();
counter->payload_bytes += packet.payload_size();
}
} // namespace
RTPSender::RTPSender(
bool audio,
Clock* clock,
Transport* transport,
RtpPacketSender* paced_sender,
FlexfecSender* flexfec_sender,
TransportSequenceNumberAllocator* sequence_number_allocator,
TransportFeedbackObserver* transport_feedback_observer,
BitrateStatisticsObserver* bitrate_callback,
FrameCountObserver* frame_count_observer,
SendSideDelayObserver* send_side_delay_observer,
RtcEventLog* event_log,
SendPacketObserver* send_packet_observer,
RateLimiter* retransmission_rate_limiter,
OverheadObserver* overhead_observer,
bool populate_network2_timestamp)
: clock_(clock),
// TODO(holmer): Remove this conversion?
clock_delta_ms_(clock_->TimeInMilliseconds() - rtc::TimeMillis()),
random_(clock_->TimeInMicroseconds()),
audio_configured_(audio),
audio_(audio ? new RTPSenderAudio(clock, this) : nullptr),
video_(audio ? nullptr : new RTPSenderVideo(clock, this, flexfec_sender)),
paced_sender_(paced_sender),
transport_sequence_number_allocator_(sequence_number_allocator),
transport_feedback_observer_(transport_feedback_observer),
last_capture_time_ms_sent_(0),
transport_(transport),
sending_media_(true), // Default to sending media.
max_packet_size_(IP_PACKET_SIZE - 28), // Default is IP-v4/UDP.
last_payload_type_(-1),
payload_type_map_(),
rtp_header_extension_map_(),
packet_history_(clock),
flexfec_packet_history_(clock),
// Statistics
rtp_stats_callback_(nullptr),
total_bitrate_sent_(kBitrateStatisticsWindowMs,
RateStatistics::kBpsScale),
nack_bitrate_sent_(kBitrateStatisticsWindowMs, RateStatistics::kBpsScale),
frame_count_observer_(frame_count_observer),
send_side_delay_observer_(send_side_delay_observer),
event_log_(event_log),
send_packet_observer_(send_packet_observer),
bitrate_callback_(bitrate_callback),
// RTP variables
remote_ssrc_(0),
sequence_number_forced_(false),
last_rtp_timestamp_(0),
capture_time_ms_(0),
last_timestamp_time_ms_(0),
media_has_been_sent_(false),
last_packet_marker_bit_(false),
csrcs_(),
rtx_(kRtxOff),
rtp_overhead_bytes_per_packet_(0),
retransmission_rate_limiter_(retransmission_rate_limiter),
overhead_observer_(overhead_observer),
populate_network2_timestamp_(populate_network2_timestamp),
send_side_bwe_with_overhead_(
webrtc::field_trial::IsEnabled("WebRTC-SendSideBwe-WithOverhead")) {
// This random initialization is not intended to be cryptographic strong.
timestamp_offset_ = random_.Rand<uint32_t>();
// Random start, 16 bits. Can't be 0.
sequence_number_rtx_ = random_.Rand(1, kMaxInitRtpSeqNumber);
sequence_number_ = random_.Rand(1, kMaxInitRtpSeqNumber);
// Store FlexFEC packets in the packet history data structure, so they can
// be found when paced.
if (flexfec_sender) {
flexfec_packet_history_.SetStorePacketsStatus(
RtpPacketHistory::StorageMode::kStore,
kMinFlexfecPacketsToStoreForPacing);
}
}
RTPSender::~RTPSender() {
// TODO(tommi): Use a thread checker to ensure the object is created and
// deleted on the same thread. At the moment this isn't possible due to
// voe::ChannelOwner in voice engine. To reproduce, run:
// voe_auto_test --automated --gtest_filter=*MixManyChannelsForStressOpus
// TODO(tommi,holmer): We don't grab locks in the dtor before accessing member
// variables but we grab them in all other methods. (what's the design?)
// Start documenting what thread we're on in what method so that it's easier
// to understand performance attributes and possibly remove locks.
while (!payload_type_map_.empty()) {
std::map<int8_t, RtpUtility::Payload*>::iterator it =
payload_type_map_.begin();
delete it->second;
payload_type_map_.erase(it);
}
}
rtc::ArrayView<const RtpExtensionSize> RTPSender::FecExtensionSizes() {
return rtc::MakeArrayView(kFecOrPaddingExtensionSizes,
arraysize(kFecOrPaddingExtensionSizes));
}
rtc::ArrayView<const RtpExtensionSize> RTPSender::VideoExtensionSizes() {
return rtc::MakeArrayView(kVideoExtensionSizes,
arraysize(kVideoExtensionSizes));
}
uint16_t RTPSender::ActualSendBitrateKbit() const {
rtc::CritScope cs(&statistics_crit_);
return static_cast<uint16_t>(
total_bitrate_sent_.Rate(clock_->TimeInMilliseconds()).value_or(0) /
1000);
}
uint32_t RTPSender::VideoBitrateSent() const {
if (video_) {
return video_->VideoBitrateSent();
}
return 0;
}
uint32_t RTPSender::FecOverheadRate() const {
if (video_) {
return video_->FecOverheadRate();
}
return 0;
}
uint32_t RTPSender::NackOverheadRate() const {
rtc::CritScope cs(&statistics_crit_);
return nack_bitrate_sent_.Rate(clock_->TimeInMilliseconds()).value_or(0);
}
int32_t RTPSender::RegisterRtpHeaderExtension(RTPExtensionType type,
uint8_t id) {
rtc::CritScope lock(&send_critsect_);
return rtp_header_extension_map_.RegisterByType(id, type) ? 0 : -1;
}
bool RTPSender::IsRtpHeaderExtensionRegistered(RTPExtensionType type) const {
rtc::CritScope lock(&send_critsect_);
return rtp_header_extension_map_.IsRegistered(type);
}
int32_t RTPSender::DeregisterRtpHeaderExtension(RTPExtensionType type) {
rtc::CritScope lock(&send_critsect_);
return rtp_header_extension_map_.Deregister(type);
}
int32_t RTPSender::RegisterPayload(
const char payload_name[RTP_PAYLOAD_NAME_SIZE],
int8_t payload_number,
uint32_t frequency,
size_t channels,
uint32_t rate) {
RTC_DCHECK_LT(strlen(payload_name), RTP_PAYLOAD_NAME_SIZE);
rtc::CritScope lock(&send_critsect_);
std::map<int8_t, RtpUtility::Payload*>::iterator it =
payload_type_map_.find(payload_number);
if (payload_type_map_.end() != it) {
// We already use this payload type.
RtpUtility::Payload* payload = it->second;
RTC_DCHECK(payload);
// Check if it's the same as we already have.
if (RtpUtility::StringCompare(
payload->name, payload_name, RTP_PAYLOAD_NAME_SIZE - 1)) {
if (audio_configured_ && payload->typeSpecific.is_audio()) {
auto& p = payload->typeSpecific.audio_payload();
if (rtc::SafeEq(p.format.clockrate_hz, frequency) &&
(p.rate == rate || p.rate == 0 || rate == 0)) {
p.rate = rate;
// Ensure that we update the rate if new or old is zero.
return 0;
}
}
if (!audio_configured_ && !payload->typeSpecific.is_audio()) {
return 0;
}
}
return -1;
}
int32_t ret_val = 0;
RtpUtility::Payload* payload = nullptr;
if (audio_configured_) {
// TODO(mflodman): Change to CreateAudioPayload and make static.
ret_val = audio_->RegisterAudioPayload(payload_name, payload_number,
frequency, channels, rate, &payload);
} else {
payload = video_->CreateVideoPayload(payload_name, payload_number);
}
if (payload) {
payload_type_map_[payload_number] = payload;
}
return ret_val;
}
int32_t RTPSender::DeRegisterSendPayload(int8_t payload_type) {
rtc::CritScope lock(&send_critsect_);
std::map<int8_t, RtpUtility::Payload*>::iterator it =
payload_type_map_.find(payload_type);
if (payload_type_map_.end() == it) {
return -1;
}
RtpUtility::Payload* payload = it->second;
delete payload;
payload_type_map_.erase(it);
return 0;
}
void RTPSender::SetMaxRtpPacketSize(size_t max_packet_size) {
RTC_DCHECK_GE(max_packet_size, 100);
RTC_DCHECK_LE(max_packet_size, IP_PACKET_SIZE);
rtc::CritScope lock(&send_critsect_);
max_packet_size_ = max_packet_size;
}
size_t RTPSender::MaxRtpPacketSize() const {
return max_packet_size_;
}
void RTPSender::SetRtxStatus(int mode) {
rtc::CritScope lock(&send_critsect_);
rtx_ = mode;
}
int RTPSender::RtxStatus() const {
rtc::CritScope lock(&send_critsect_);
return rtx_;
}
void RTPSender::SetRtxSsrc(uint32_t ssrc) {
rtc::CritScope lock(&send_critsect_);
ssrc_rtx_.emplace(ssrc);
if (mid_oracle_rtx_) {
mid_oracle_rtx_->SetSsrc(ssrc);
}
}
uint32_t RTPSender::RtxSsrc() const {
rtc::CritScope lock(&send_critsect_);
RTC_DCHECK(ssrc_rtx_);
return *ssrc_rtx_;
}
void RTPSender::SetRtxPayloadType(int payload_type,
int associated_payload_type) {
rtc::CritScope lock(&send_critsect_);
RTC_DCHECK_LE(payload_type, 127);
RTC_DCHECK_LE(associated_payload_type, 127);
if (payload_type < 0) {
RTC_LOG(LS_ERROR) << "Invalid RTX payload type: " << payload_type << ".";
return;
}
rtx_payload_type_map_[associated_payload_type] = payload_type;
}
int32_t RTPSender::CheckPayloadType(int8_t payload_type,
RtpVideoCodecTypes* video_type) {
rtc::CritScope lock(&send_critsect_);
if (payload_type < 0) {
RTC_LOG(LS_ERROR) << "Invalid payload_type " << payload_type << ".";
return -1;
}
if (last_payload_type_ == payload_type) {
if (!audio_configured_) {
*video_type = video_->VideoCodecType();
}
return 0;
}
std::map<int8_t, RtpUtility::Payload*>::iterator it =
payload_type_map_.find(payload_type);
if (it == payload_type_map_.end()) {
RTC_LOG(LS_WARNING) << "Payload type " << static_cast<int>(payload_type)
<< " not registered.";
return -1;
}
RtpUtility::Payload* payload = it->second;
RTC_DCHECK(payload);
if (payload->typeSpecific.is_video() && !audio_configured_) {
video_->SetVideoCodecType(
payload->typeSpecific.video_payload().videoCodecType);
*video_type = payload->typeSpecific.video_payload().videoCodecType;
}
return 0;
}
bool RTPSender::SendOutgoingData(FrameType frame_type,
int8_t payload_type,
uint32_t capture_timestamp,
int64_t capture_time_ms,
const uint8_t* payload_data,
size_t payload_size,
const RTPFragmentationHeader* fragmentation,
const RTPVideoHeader* rtp_header,
uint32_t* transport_frame_id_out,
int64_t expected_retransmission_time_ms) {
uint32_t ssrc;
uint16_t sequence_number;
uint32_t rtp_timestamp;
{
// Drop this packet if we're not sending media packets.
rtc::CritScope lock(&send_critsect_);
RTC_DCHECK(ssrc_);
ssrc = *ssrc_;
sequence_number = sequence_number_;
rtp_timestamp = timestamp_offset_ + capture_timestamp;
if (transport_frame_id_out)
*transport_frame_id_out = rtp_timestamp;
if (!sending_media_)
return true;
}
RtpVideoCodecTypes video_type = kRtpVideoGeneric;
if (CheckPayloadType(payload_type, &video_type) != 0) {
RTC_LOG(LS_ERROR) << "Don't send data with unknown payload type: "
<< static_cast<int>(payload_type) << ".";
return false;
}
switch (frame_type) {
case kAudioFrameSpeech:
case kAudioFrameCN:
RTC_CHECK(audio_configured_);
break;
case kVideoFrameKey:
case kVideoFrameDelta:
RTC_CHECK(!audio_configured_);
break;
case kEmptyFrame:
break;
}
bool result;
if (audio_configured_) {
TRACE_EVENT_ASYNC_STEP1("webrtc", "Audio", rtp_timestamp, "Send", "type",
FrameTypeToString(frame_type));
// The only known way to produce of RTPFragmentationHeader for audio is
// to use the AudioCodingModule directly.
RTC_DCHECK(fragmentation == nullptr);
result = audio_->SendAudio(frame_type, payload_type, rtp_timestamp,
payload_data, payload_size);
} else {
TRACE_EVENT_ASYNC_STEP1("webrtc", "Video", capture_time_ms,
"Send", "type", FrameTypeToString(frame_type));
if (frame_type == kEmptyFrame)
return true;
if (rtp_header) {
playout_delay_oracle_.UpdateRequest(ssrc, rtp_header->playout_delay,
sequence_number);
}
result = video_->SendVideo(video_type, frame_type, payload_type,
rtp_timestamp, capture_time_ms, payload_data,
payload_size, fragmentation, rtp_header,
expected_retransmission_time_ms);
}
rtc::CritScope cs(&statistics_crit_);
// Note: This is currently only counting for video.
if (frame_type == kVideoFrameKey) {
++frame_counts_.key_frames;
} else if (frame_type == kVideoFrameDelta) {
++frame_counts_.delta_frames;
}
if (frame_count_observer_) {
frame_count_observer_->FrameCountUpdated(frame_counts_, ssrc);
}
return result;
}
size_t RTPSender::TrySendRedundantPayloads(size_t bytes_to_send,
const PacedPacketInfo& pacing_info) {
{
rtc::CritScope lock(&send_critsect_);
if (!sending_media_)
return 0;
if ((rtx_ & kRtxRedundantPayloads) == 0)
return 0;
}
int bytes_left = static_cast<int>(bytes_to_send);
while (bytes_left > 0) {
std::unique_ptr<RtpPacketToSend> packet =
packet_history_.GetBestFittingPacket(bytes_left);
if (!packet)
break;
size_t payload_size = packet->payload_size();
if (!PrepareAndSendPacket(std::move(packet), true, false, pacing_info))
break;
bytes_left -= payload_size;
}
return bytes_to_send - bytes_left;
}
size_t RTPSender::SendPadData(size_t bytes,
const PacedPacketInfo& pacing_info) {
size_t padding_bytes_in_packet;
size_t max_payload_size = max_packet_size_ - RtpHeaderLength();
if (audio_configured_) {
// Allow smaller padding packets for audio.
padding_bytes_in_packet = rtc::SafeClamp<size_t>(
bytes, kMinAudioPaddingLength,
rtc::SafeMin(max_payload_size, kMaxPaddingLength));
} else {
// Always send full padding packets. This is accounted for by the
// RtpPacketSender, which will make sure we don't send too much padding even
// if a single packet is larger than requested.
// We do this to avoid frequently sending small packets on higher bitrates.
padding_bytes_in_packet =
rtc::SafeMin<size_t>(max_payload_size, kMaxPaddingLength);
}
size_t bytes_sent = 0;
while (bytes_sent < bytes) {
int64_t now_ms = clock_->TimeInMilliseconds();
uint32_t ssrc;
uint32_t timestamp;
int64_t capture_time_ms;
uint16_t sequence_number;
int payload_type;
bool over_rtx;
{
rtc::CritScope lock(&send_critsect_);
if (!sending_media_)
break;
timestamp = last_rtp_timestamp_;
capture_time_ms = capture_time_ms_;
if (rtx_ == kRtxOff) {
if (last_payload_type_ == -1)
break;
// Without RTX we can't send padding in the middle of frames.
// For audio marker bits doesn't mark the end of a frame and frames
// are usually a single packet, so for now we don't apply this rule
// for audio.
if (!audio_configured_ && !last_packet_marker_bit_) {
break;
}
if (!ssrc_) {
RTC_LOG(LS_ERROR) << "SSRC unset.";
return 0;
}
RTC_DCHECK(ssrc_);
ssrc = *ssrc_;
sequence_number = sequence_number_;
++sequence_number_;
payload_type = last_payload_type_;
over_rtx = false;
} else {
// Without abs-send-time or transport sequence number a media packet
// must be sent before padding so that the timestamps used for
// estimation are correct.
if (!media_has_been_sent_ &&
!(rtp_header_extension_map_.IsRegistered(AbsoluteSendTime::kId) ||
(rtp_header_extension_map_.IsRegistered(
TransportSequenceNumber::kId) &&
transport_sequence_number_allocator_))) {
break;
}
// Only change change the timestamp of padding packets sent over RTX.
// Padding only packets over RTP has to be sent as part of a media
// frame (and therefore the same timestamp).
if (last_timestamp_time_ms_ > 0) {
timestamp +=
(now_ms - last_timestamp_time_ms_) * kTimestampTicksPerMs;
capture_time_ms += (now_ms - last_timestamp_time_ms_);
}
if (!ssrc_rtx_) {
RTC_LOG(LS_ERROR) << "RTX SSRC unset.";
return 0;
}
RTC_DCHECK(ssrc_rtx_);
ssrc = *ssrc_rtx_;
sequence_number = sequence_number_rtx_;
++sequence_number_rtx_;
payload_type = rtx_payload_type_map_.begin()->second;
over_rtx = true;
}
}
RtpPacketToSend padding_packet(&rtp_header_extension_map_);
padding_packet.SetPayloadType(payload_type);
padding_packet.SetMarker(false);
padding_packet.SetSequenceNumber(sequence_number);
padding_packet.SetTimestamp(timestamp);
padding_packet.SetSsrc(ssrc);
if (capture_time_ms > 0) {
padding_packet.SetExtension<TransmissionOffset>(
(now_ms - capture_time_ms) * kTimestampTicksPerMs);
}
padding_packet.SetExtension<AbsoluteSendTime>(
AbsoluteSendTime::MsTo24Bits(now_ms));
PacketOptions options;
bool has_transport_seq_num =
UpdateTransportSequenceNumber(&padding_packet, &options.packet_id);
padding_packet.SetPadding(padding_bytes_in_packet, &random_);
if (has_transport_seq_num) {
AddPacketToTransportFeedback(options.packet_id, padding_packet,
pacing_info);
}
if (!SendPacketToNetwork(padding_packet, options, pacing_info))
break;
bytes_sent += padding_bytes_in_packet;
UpdateRtpStats(padding_packet, over_rtx, false);
}
return bytes_sent;
}
void RTPSender::SetStorePacketsStatus(bool enable, uint16_t number_to_store) {
RtpPacketHistory::StorageMode mode =
enable ? RtpPacketHistory::StorageMode::kStore
: RtpPacketHistory::StorageMode::kDisabled;
packet_history_.SetStorePacketsStatus(mode, number_to_store);
}
bool RTPSender::StorePackets() const {
return packet_history_.GetStorageMode() !=
RtpPacketHistory::StorageMode::kDisabled;
}
int32_t RTPSender::ReSendPacket(uint16_t packet_id) {
// Try to find packet in RTP packet history. Also verify RTT here, so that we
// don't retransmit too often.
rtc::Optional<RtpPacketHistory::PacketState> stored_packet =
packet_history_.GetPacketState(packet_id, true);
if (!stored_packet) {
// Packet not found.
return 0;
}
const int32_t packet_size = static_cast<int32_t>(stored_packet->payload_size);
RTC_DCHECK(retransmission_rate_limiter_);
// Check if we're overusing retransmission bitrate.
// TODO(sprang): Add histograms for nack success or failure reasons.
if (!retransmission_rate_limiter_->TryUseRate(packet_size)) {
return -1;
}
if (paced_sender_) {
// Convert from TickTime to Clock since capture_time_ms is based on
// TickTime.
int64_t corrected_capture_tims_ms =
stored_packet->capture_time_ms + clock_delta_ms_;
paced_sender_->InsertPacket(
RtpPacketSender::kNormalPriority, stored_packet->ssrc,
stored_packet->rtp_sequence_number, corrected_capture_tims_ms,
stored_packet->payload_size, true);
return packet_size;
}
std::unique_ptr<RtpPacketToSend> packet =
packet_history_.GetPacketAndSetSendTime(packet_id, true);
if (!packet) {
// Packet could theoretically time out between the first check and this one.
return 0;
}
const bool rtx = (RtxStatus() & kRtxRetransmitted) > 0;
if (!PrepareAndSendPacket(std::move(packet), rtx, true, PacedPacketInfo()))
return -1;
return packet_size;
}
bool RTPSender::SendPacketToNetwork(const RtpPacketToSend& packet,
const PacketOptions& options,
const PacedPacketInfo& pacing_info) {
int bytes_sent = -1;
if (transport_) {
UpdateRtpOverhead(packet);
bytes_sent = transport_->SendRtp(packet.data(), packet.size(), options)
? static_cast<int>(packet.size())
: -1;
if (event_log_ && bytes_sent > 0) {
event_log_->Log(rtc::MakeUnique<RtcEventRtpPacketOutgoing>(
packet, pacing_info.probe_cluster_id));
}
}
TRACE_EVENT_INSTANT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
"RTPSender::SendPacketToNetwork", "size", packet.size(),
"sent", bytes_sent);
// TODO(pwestin): Add a separate bitrate for sent bitrate after pacer.
if (bytes_sent <= 0) {
RTC_LOG(LS_WARNING) << "Transport failed to send packet.";
return false;
}
return true;
}
int RTPSender::SelectiveRetransmissions() const {
if (!video_)
return -1;
return video_->SelectiveRetransmissions();
}
int RTPSender::SetSelectiveRetransmissions(uint8_t settings) {
if (!video_)
return -1;
video_->SetSelectiveRetransmissions(settings);
return 0;
}
void RTPSender::OnReceivedNack(
const std::vector<uint16_t>& nack_sequence_numbers,
int64_t avg_rtt) {
TRACE_EVENT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
"RTPSender::OnReceivedNACK", "num_seqnum",
nack_sequence_numbers.size(), "avg_rtt", avg_rtt);
packet_history_.SetRtt(5 + avg_rtt);
for (uint16_t seq_no : nack_sequence_numbers) {
const int32_t bytes_sent = ReSendPacket(seq_no);
if (bytes_sent < 0) {
// Failed to send one Sequence number. Give up the rest in this nack.
RTC_LOG(LS_WARNING) << "Failed resending RTP packet " << seq_no
<< ", Discard rest of packets.";
break;
}
}
}
void RTPSender::OnReceivedRtcpReportBlocks(
const ReportBlockList& report_blocks) {
playout_delay_oracle_.OnReceivedRtcpReportBlocks(report_blocks);
{
rtc::CritScope lock(&send_critsect_);
if (mid_oracle_) {
mid_oracle_->OnReceivedRtcpReportBlocks(report_blocks);
}
if (mid_oracle_rtx_) {
mid_oracle_rtx_->OnReceivedRtcpReportBlocks(report_blocks);
}
}
}
// Called from pacer when we can send the packet.
bool RTPSender::TimeToSendPacket(uint32_t ssrc,
uint16_t sequence_number,
int64_t capture_time_ms,
bool retransmission,
const PacedPacketInfo& pacing_info) {
if (!SendingMedia())
return true;
std::unique_ptr<RtpPacketToSend> packet;
// No need to verify RTT here, it has already been checked before putting the
// packet into the pacer. But _do_ update the send time.
if (ssrc == SSRC()) {
packet = packet_history_.GetPacketAndSetSendTime(sequence_number, false);
} else if (ssrc == FlexfecSsrc()) {
packet =
flexfec_packet_history_.GetPacketAndSetSendTime(sequence_number, false);
}
if (!packet) {
// Packet cannot be found.
return true;
}
return PrepareAndSendPacket(
std::move(packet),
retransmission && (RtxStatus() & kRtxRetransmitted) > 0, retransmission,
pacing_info);
}
bool RTPSender::PrepareAndSendPacket(std::unique_ptr<RtpPacketToSend> packet,
bool send_over_rtx,
bool is_retransmit,
const PacedPacketInfo& pacing_info) {
RTC_DCHECK(packet);
int64_t capture_time_ms = packet->capture_time_ms();
RtpPacketToSend* packet_to_send = packet.get();
if (!is_retransmit && packet->Marker()) {
TRACE_EVENT_ASYNC_END0(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), "PacedSend",
capture_time_ms);
}
TRACE_EVENT_INSTANT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
"PrepareAndSendPacket", "timestamp", packet->Timestamp(),
"seqnum", packet->SequenceNumber());
std::unique_ptr<RtpPacketToSend> packet_rtx;
if (send_over_rtx) {
packet_rtx = BuildRtxPacket(*packet);
if (!packet_rtx)
return false;
packet_to_send = packet_rtx.get();
}
// Bug webrtc:7859. While FEC is invoked from rtp_sender_video, and not after
// the pacer, these modifications of the header below are happening after the
// FEC protection packets are calculated. This will corrupt recovered packets
// at the same place. It's not an issue for extensions, which are present in
// all the packets (their content just may be incorrect on recovered packets).
// In case of VideoTimingExtension, since it's present not in every packet,
// data after rtp header may be corrupted if these packets are protected by
// the FEC.
int64_t now_ms = clock_->TimeInMilliseconds();
int64_t diff_ms = now_ms - capture_time_ms;
packet_to_send->SetExtension<TransmissionOffset>(kTimestampTicksPerMs *
diff_ms);
packet_to_send->SetExtension<AbsoluteSendTime>(
AbsoluteSendTime::MsTo24Bits(now_ms));
if (packet_to_send->HasExtension<VideoTimingExtension>()) {
if (populate_network2_timestamp_) {
packet_to_send->set_network2_time_ms(now_ms);
} else {
packet_to_send->set_pacer_exit_time_ms(now_ms);
}
}
PacketOptions options;
if (UpdateTransportSequenceNumber(packet_to_send, &options.packet_id)) {
AddPacketToTransportFeedback(options.packet_id, *packet_to_send,
pacing_info);
}
options.application_data.assign(packet_to_send->application_data().begin(),
packet_to_send->application_data().end());
if (!is_retransmit && !send_over_rtx) {
UpdateDelayStatistics(packet->capture_time_ms(), now_ms);
UpdateOnSendPacket(options.packet_id, packet->capture_time_ms(),
packet->Ssrc());
}
if (!SendPacketToNetwork(*packet_to_send, options, pacing_info))
return false;
{
rtc::CritScope lock(&send_critsect_);
media_has_been_sent_ = true;
}
UpdateRtpStats(*packet_to_send, send_over_rtx, is_retransmit);
return true;
}
void RTPSender::UpdateRtpStats(const RtpPacketToSend& packet,
bool is_rtx,
bool is_retransmit) {
int64_t now_ms = clock_->TimeInMilliseconds();
rtc::CritScope lock(&statistics_crit_);
StreamDataCounters* counters = is_rtx ? &rtx_rtp_stats_ : &rtp_stats_;
total_bitrate_sent_.Update(packet.size(), now_ms);
if (counters->first_packet_time_ms == -1)
counters->first_packet_time_ms = now_ms;
if (IsFecPacket(packet))
CountPacket(&counters->fec, packet);
if (is_retransmit) {
CountPacket(&counters->retransmitted, packet);
nack_bitrate_sent_.Update(packet.size(), now_ms);
}
CountPacket(&counters->transmitted, packet);
if (rtp_stats_callback_)
rtp_stats_callback_->DataCountersUpdated(*counters, packet.Ssrc());
}
bool RTPSender::IsFecPacket(const RtpPacketToSend& packet) const {
if (!video_)
return false;
// FlexFEC.
if (packet.Ssrc() == FlexfecSsrc())
return true;
// RED+ULPFEC.
int pt_red;
int pt_fec;
video_->GetUlpfecConfig(&pt_red, &pt_fec);
return static_cast<int>(packet.PayloadType()) == pt_red &&
static_cast<int>(packet.payload()[0]) == pt_fec;
}
size_t RTPSender::TimeToSendPadding(size_t bytes,
const PacedPacketInfo& pacing_info) {
if (bytes == 0)
return 0;
size_t bytes_sent = TrySendRedundantPayloads(bytes, pacing_info);
if (bytes_sent < bytes)
bytes_sent += SendPadData(bytes - bytes_sent, pacing_info);
return bytes_sent;
}
bool RTPSender::SendToNetwork(std::unique_ptr<RtpPacketToSend> packet,
StorageType storage,
RtpPacketSender::Priority priority) {
RTC_DCHECK(packet);
int64_t now_ms = clock_->TimeInMilliseconds();
// |capture_time_ms| <= 0 is considered invalid.
// TODO(holmer): This should be changed all over Video Engine so that negative
// time is consider invalid, while 0 is considered a valid time.
if (packet->capture_time_ms() > 0) {
packet->SetExtension<TransmissionOffset>(
kTimestampTicksPerMs * (now_ms - packet->capture_time_ms()));
}
packet->SetExtension<AbsoluteSendTime>(AbsoluteSendTime::MsTo24Bits(now_ms));
if (video_) {
BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "VideoTotBitrate_kbps", now_ms,
ActualSendBitrateKbit(), packet->Ssrc());
BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "VideoFecBitrate_kbps", now_ms,
FecOverheadRate() / 1000, packet->Ssrc());
BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "VideoNackBitrate_kbps", now_ms,
NackOverheadRate() / 1000, packet->Ssrc());
} else {
BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "AudioTotBitrate_kbps", now_ms,
ActualSendBitrateKbit(), packet->Ssrc());
BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "AudioNackBitrate_kbps", now_ms,
NackOverheadRate() / 1000, packet->Ssrc());
}
uint32_t ssrc = packet->Ssrc();
rtc::Optional<uint32_t> flexfec_ssrc = FlexfecSsrc();
if (paced_sender_) {
uint16_t seq_no = packet->SequenceNumber();
// Correct offset between implementations of millisecond time stamps in
// TickTime and Clock.
int64_t corrected_time_ms = packet->capture_time_ms() + clock_delta_ms_;
size_t payload_length = packet->payload_size();
if (ssrc == flexfec_ssrc) {
// Store FlexFEC packets in the history here, so they can be found
// when the pacer calls TimeToSendPacket.
flexfec_packet_history_.PutRtpPacket(std::move(packet), storage,
rtc::nullopt);
} else {
packet_history_.PutRtpPacket(std::move(packet), storage, rtc::nullopt);
}
paced_sender_->InsertPacket(priority, ssrc, seq_no, corrected_time_ms,
payload_length, false);
if (last_capture_time_ms_sent_ == 0 ||
corrected_time_ms > last_capture_time_ms_sent_) {
last_capture_time_ms_sent_ = corrected_time_ms;
TRACE_EVENT_ASYNC_BEGIN1(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
"PacedSend", corrected_time_ms,
"capture_time_ms", corrected_time_ms);
}
return true;
}
PacketOptions options;
if (UpdateTransportSequenceNumber(packet.get(), &options.packet_id)) {
AddPacketToTransportFeedback(options.packet_id, *packet.get(),
PacedPacketInfo());
}
options.application_data.assign(packet->application_data().begin(),
packet->application_data().end());
UpdateDelayStatistics(packet->capture_time_ms(), now_ms);
UpdateOnSendPacket(options.packet_id, packet->capture_time_ms(),
packet->Ssrc());
bool sent = SendPacketToNetwork(*packet, options, PacedPacketInfo());
if (sent) {
{
rtc::CritScope lock(&send_critsect_);
media_has_been_sent_ = true;
}
UpdateRtpStats(*packet, false, false);
}
// To support retransmissions, we store the media packet as sent in the
// packet history (even if send failed).
if (storage == kAllowRetransmission) {
RTC_DCHECK_EQ(ssrc, SSRC());
packet_history_.PutRtpPacket(std::move(packet), storage, now_ms);
}
return sent;
}
void RTPSender::UpdateDelayStatistics(int64_t capture_time_ms, int64_t now_ms) {
if (!send_side_delay_observer_ || capture_time_ms <= 0)
return;
uint32_t ssrc;
int64_t avg_delay_ms = 0;
int max_delay_ms = 0;
{
rtc::CritScope lock(&send_critsect_);
if (!ssrc_)
return;
ssrc = *ssrc_;
}
{
rtc::CritScope cs(&statistics_crit_);
// TODO(holmer): Compute this iteratively instead.
send_delays_[now_ms] = now_ms - capture_time_ms;
send_delays_.erase(send_delays_.begin(),
send_delays_.lower_bound(now_ms -
kSendSideDelayWindowMs));
int num_delays = 0;
for (auto it = send_delays_.upper_bound(now_ms - kSendSideDelayWindowMs);
it != send_delays_.end(); ++it) {
max_delay_ms = std::max(max_delay_ms, it->second);
avg_delay_ms += it->second;
++num_delays;
}
if (num_delays == 0)
return;
avg_delay_ms = (avg_delay_ms + num_delays / 2) / num_delays;
}
send_side_delay_observer_->SendSideDelayUpdated(
rtc::dchecked_cast<int>(avg_delay_ms), max_delay_ms, ssrc);
}
void RTPSender::UpdateOnSendPacket(int packet_id,
int64_t capture_time_ms,
uint32_t ssrc) {
if (!send_packet_observer_ || capture_time_ms <= 0 || packet_id == -1)
return;
send_packet_observer_->OnSendPacket(packet_id, capture_time_ms, ssrc);
}
void RTPSender::ProcessBitrate() {
if (!bitrate_callback_)
return;
int64_t now_ms = clock_->TimeInMilliseconds();
uint32_t ssrc;
{
rtc::CritScope lock(&send_critsect_);
if (!ssrc_)
return;
ssrc = *ssrc_;
}
rtc::CritScope lock(&statistics_crit_);
bitrate_callback_->Notify(total_bitrate_sent_.Rate(now_ms).value_or(0),
nack_bitrate_sent_.Rate(now_ms).value_or(0), ssrc);
}
size_t RTPSender::RtpHeaderLength() const {
rtc::CritScope lock(&send_critsect_);
size_t rtp_header_length = kRtpHeaderLength;
rtp_header_length += sizeof(uint32_t) * csrcs_.size();
rtp_header_length += rtp_header_extension_map_.GetTotalLengthInBytes(
kFecOrPaddingExtensionSizes);
return rtp_header_length;
}
uint16_t RTPSender::AllocateSequenceNumber(uint16_t packets_to_send) {
rtc::CritScope lock(&send_critsect_);
uint16_t first_allocated_sequence_number = sequence_number_;
sequence_number_ += packets_to_send;
return first_allocated_sequence_number;
}
void RTPSender::GetDataCounters(StreamDataCounters* rtp_stats,
StreamDataCounters* rtx_stats) const {
rtc::CritScope lock(&statistics_crit_);
*rtp_stats = rtp_stats_;
*rtx_stats = rtx_rtp_stats_;
}
std::unique_ptr<RtpPacketToSend> RTPSender::AllocatePacket() const {
rtc::CritScope lock(&send_critsect_);
std::unique_ptr<RtpPacketToSend> packet(
new RtpPacketToSend(&rtp_header_extension_map_, max_packet_size_));
RTC_DCHECK(ssrc_);
packet->SetSsrc(*ssrc_);
packet->SetCsrcs(csrcs_);
// Reserve extensions, if registered, RtpSender set in SendToNetwork.
packet->ReserveExtension<AbsoluteSendTime>();
packet->ReserveExtension<TransmissionOffset>();
packet->ReserveExtension<TransportSequenceNumber>();
if (playout_delay_oracle_.send_playout_delay()) {
packet->SetExtension<PlayoutDelayLimits>(
playout_delay_oracle_.playout_delay());
}
if (mid_oracle_ && mid_oracle_->send_mid()) {
// This is a no-op if the MID header extension is not registered.
packet->SetExtension<RtpMid>(mid_oracle_->mid());
}
return packet;
}
bool RTPSender::AssignSequenceNumber(RtpPacketToSend* packet) {
rtc::CritScope lock(&send_critsect_);
if (!sending_media_)
return false;
RTC_DCHECK(packet->Ssrc() == ssrc_);
packet->SetSequenceNumber(sequence_number_++);
// Remember marker bit to determine if padding can be inserted with
// sequence number following |packet|.
last_packet_marker_bit_ = packet->Marker();
// Remember payload type to use in the padding packet if rtx is disabled.
last_payload_type_ = packet->PayloadType();
// Save timestamps to generate timestamp field and extensions for the padding.
last_rtp_timestamp_ = packet->Timestamp();
last_timestamp_time_ms_ = clock_->TimeInMilliseconds();
capture_time_ms_ = packet->capture_time_ms();
return true;
}
bool RTPSender::UpdateTransportSequenceNumber(RtpPacketToSend* packet,
int* packet_id) const {
RTC_DCHECK(packet);
RTC_DCHECK(packet_id);
rtc::CritScope lock(&send_critsect_);
if (!rtp_header_extension_map_.IsRegistered(TransportSequenceNumber::kId))
return false;
if (!transport_sequence_number_allocator_)
return false;
*packet_id = transport_sequence_number_allocator_->AllocateSequenceNumber();
if (!packet->SetExtension<TransportSequenceNumber>(*packet_id))
return false;
return true;
}
void RTPSender::SetSendingMediaStatus(bool enabled) {
rtc::CritScope lock(&send_critsect_);
sending_media_ = enabled;
}
bool RTPSender::SendingMedia() const {
rtc::CritScope lock(&send_critsect_);
return sending_media_;
}
void RTPSender::SetTimestampOffset(uint32_t timestamp) {
rtc::CritScope lock(&send_critsect_);
timestamp_offset_ = timestamp;
}
uint32_t RTPSender::TimestampOffset() const {
rtc::CritScope lock(&send_critsect_);
return timestamp_offset_;
}
void RTPSender::SetSSRC(uint32_t ssrc) {
// This is configured via the API.
rtc::CritScope lock(&send_critsect_);
if (ssrc_ == ssrc) {
return; // Since it's same ssrc, don't reset anything.
}
ssrc_.emplace(ssrc);
if (!sequence_number_forced_) {
sequence_number_ = random_.Rand(1, kMaxInitRtpSeqNumber);
}
if (mid_oracle_) {
mid_oracle_->SetSsrc(ssrc);
}
}
uint32_t RTPSender::SSRC() const {
rtc::CritScope lock(&send_critsect_);
RTC_DCHECK(ssrc_);
return *ssrc_;
}
void RTPSender::SetMid(const std::string& mid) {
// This is configured via the API.
rtc::CritScope lock(&send_critsect_);
// Cannot change MID once sending.
RTC_DCHECK(!sending_media_);
// Cannot change the MID if it is already set.
if (mid_oracle_) {
RTC_DCHECK_EQ(mid_oracle_->mid(), mid);
return;
}
if (mid_oracle_rtx_) {
RTC_DCHECK_EQ(mid_oracle_rtx_->mid(), mid);
return;
}
mid_oracle_ = rtc::MakeUnique<MidOracle>(mid);
if (ssrc_) {
mid_oracle_->SetSsrc(*ssrc_);
}
mid_oracle_rtx_ = rtc::MakeUnique<MidOracle>(mid);
if (ssrc_rtx_) {
mid_oracle_rtx_->SetSsrc(*ssrc_rtx_);
}
}
rtc::Optional<uint32_t> RTPSender::FlexfecSsrc() const {
if (video_) {
return video_->FlexfecSsrc();
}
return rtc::nullopt;
}
void RTPSender::SetCsrcs(const std::vector<uint32_t>& csrcs) {
RTC_DCHECK_LE(csrcs.size(), kRtpCsrcSize);
rtc::CritScope lock(&send_critsect_);
csrcs_ = csrcs;
}
void RTPSender::SetSequenceNumber(uint16_t seq) {
rtc::CritScope lock(&send_critsect_);
sequence_number_forced_ = true;
sequence_number_ = seq;
}
uint16_t RTPSender::SequenceNumber() const {
rtc::CritScope lock(&send_critsect_);
return sequence_number_;
}
// Audio.
int32_t RTPSender::SendTelephoneEvent(uint8_t key,
uint16_t time_ms,
uint8_t level) {
if (!audio_configured_) {
return -1;
}
return audio_->SendTelephoneEvent(key, time_ms, level);
}
int32_t RTPSender::SetAudioLevel(uint8_t level_d_bov) {
return audio_->SetAudioLevel(level_d_bov);
}
RtpVideoCodecTypes RTPSender::VideoCodecType() const {
RTC_DCHECK(!audio_configured_) << "Sender is an audio stream!";
return video_->VideoCodecType();
}
void RTPSender::SetUlpfecConfig(int red_payload_type, int ulpfec_payload_type) {
RTC_DCHECK(!audio_configured_);
video_->SetUlpfecConfig(red_payload_type, ulpfec_payload_type);
}
bool RTPSender::SetFecParameters(const FecProtectionParams& delta_params,
const FecProtectionParams& key_params) {
if (audio_configured_) {
return false;
}
video_->SetFecParameters(delta_params, key_params);
return true;
}
std::unique_ptr<RtpPacketToSend> RTPSender::BuildRtxPacket(
const RtpPacketToSend& packet) {
// TODO(danilchap): Create rtx packet with extra capacity for SRTP
// when transport interface would be updated to take buffer class.
std::unique_ptr<RtpPacketToSend> rtx_packet(new RtpPacketToSend(
&rtp_header_extension_map_, packet.size() + kRtxHeaderSize));
// Add original RTP header.
rtx_packet->CopyHeaderFrom(packet);
{
rtc::CritScope lock(&send_critsect_);
if (!sending_media_)
return nullptr;
RTC_DCHECK(ssrc_rtx_);
// Replace payload type.
auto kv = rtx_payload_type_map_.find(packet.PayloadType());
if (kv == rtx_payload_type_map_.end())
return nullptr;
rtx_packet->SetPayloadType(kv->second);
// Replace sequence number.
rtx_packet->SetSequenceNumber(sequence_number_rtx_++);
// Replace SSRC.
rtx_packet->SetSsrc(*ssrc_rtx_);
// Possibly include the MID header extension.
if (mid_oracle_rtx_ && mid_oracle_rtx_->send_mid()) {
// This is a no-op if the MID header extension is not registered.
rtx_packet->SetExtension<RtpMid>(mid_oracle_rtx_->mid());
}
}
uint8_t* rtx_payload =
rtx_packet->AllocatePayload(packet.payload_size() + kRtxHeaderSize);
RTC_DCHECK(rtx_payload);
// Add OSN (original sequence number).
ByteWriter<uint16_t>::WriteBigEndian(rtx_payload, packet.SequenceNumber());
// Add original payload data.
auto payload = packet.payload();
memcpy(rtx_payload + kRtxHeaderSize, payload.data(), payload.size());
// Add original application data.
rtx_packet->set_application_data(packet.application_data());
return rtx_packet;
}
void RTPSender::RegisterRtpStatisticsCallback(
StreamDataCountersCallback* callback) {
rtc::CritScope cs(&statistics_crit_);
rtp_stats_callback_ = callback;
}
StreamDataCountersCallback* RTPSender::GetRtpStatisticsCallback() const {
rtc::CritScope cs(&statistics_crit_);
return rtp_stats_callback_;
}
uint32_t RTPSender::BitrateSent() const {
rtc::CritScope cs(&statistics_crit_);
return total_bitrate_sent_.Rate(clock_->TimeInMilliseconds()).value_or(0);
}
void RTPSender::SetRtpState(const RtpState& rtp_state) {
rtc::CritScope lock(&send_critsect_);
sequence_number_ = rtp_state.sequence_number;
sequence_number_forced_ = true;
timestamp_offset_ = rtp_state.start_timestamp;
last_rtp_timestamp_ = rtp_state.timestamp;
capture_time_ms_ = rtp_state.capture_time_ms;
last_timestamp_time_ms_ = rtp_state.last_timestamp_time_ms;
media_has_been_sent_ = rtp_state.media_has_been_sent;
}
RtpState RTPSender::GetRtpState() const {
rtc::CritScope lock(&send_critsect_);
RtpState state;
state.sequence_number = sequence_number_;
state.start_timestamp = timestamp_offset_;
state.timestamp = last_rtp_timestamp_;
state.capture_time_ms = capture_time_ms_;
state.last_timestamp_time_ms = last_timestamp_time_ms_;
state.media_has_been_sent = media_has_been_sent_;
return state;
}
void RTPSender::SetRtxRtpState(const RtpState& rtp_state) {
rtc::CritScope lock(&send_critsect_);
sequence_number_rtx_ = rtp_state.sequence_number;
}
RtpState RTPSender::GetRtxRtpState() const {
rtc::CritScope lock(&send_critsect_);
RtpState state;
state.sequence_number = sequence_number_rtx_;
state.start_timestamp = timestamp_offset_;
return state;
}
void RTPSender::AddPacketToTransportFeedback(
uint16_t packet_id,
const RtpPacketToSend& packet,
const PacedPacketInfo& pacing_info) {
size_t packet_size = packet.payload_size() + packet.padding_size();
if (send_side_bwe_with_overhead_) {
packet_size = packet.size();
}
if (transport_feedback_observer_) {
transport_feedback_observer_->AddPacket(SSRC(), packet_id, packet_size,
pacing_info);
}
}
void RTPSender::UpdateRtpOverhead(const RtpPacketToSend& packet) {
if (!overhead_observer_)
return;
size_t overhead_bytes_per_packet;
{
rtc::CritScope lock(&send_critsect_);
if (rtp_overhead_bytes_per_packet_ == packet.headers_size()) {
return;
}
rtp_overhead_bytes_per_packet_ = packet.headers_size();
overhead_bytes_per_packet = rtp_overhead_bytes_per_packet_;
}
overhead_observer_->OnOverheadChanged(overhead_bytes_per_packet);
}
int64_t RTPSender::LastTimestampTimeMs() const {
rtc::CritScope lock(&send_critsect_);
return last_timestamp_time_ms_;
}
void RTPSender::SendKeepAlive(uint8_t payload_type) {
std::unique_ptr<RtpPacketToSend> packet = AllocatePacket();
packet->SetPayloadType(payload_type);
// Set marker bit and timestamps in the same manner as plain padding packets.
packet->SetMarker(false);
{
rtc::CritScope lock(&send_critsect_);
packet->SetTimestamp(last_rtp_timestamp_);
packet->set_capture_time_ms(capture_time_ms_);
}
AssignSequenceNumber(packet.get());
SendToNetwork(std::move(packet), StorageType::kDontRetransmit,
RtpPacketSender::Priority::kLowPriority);
}
void RTPSender::SetRtt(int64_t rtt_ms) {
packet_history_.SetRtt(rtt_ms);
flexfec_packet_history_.SetRtt(rtt_ms);
}
} // namespace webrtc
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