Google Git
Sign in
webrtc / src / 291cd8fac37a2efa026713d198d653bc083881db / . / webrtc / modules / rtp_rtcp / source / rtp_sender.cc
blob: 9622f8b82653837f746df858dcc237f06223de5c [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_sender.h"
#include <algorithm>
#include <utility>
#include "webrtc/base/checks.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/rate_limiter.h"
#include "webrtc/base/trace_event.h"
#include "webrtc/base/timeutils.h"
#include "webrtc/call.h"
#include "webrtc/call/rtc_event_log.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_cvo.h"
#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
#include "webrtc/modules/rtp_rtcp/source/playout_delay_oracle.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_sender_video.h"
#include "webrtc/modules/rtp_rtcp/source/time_util.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 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;
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,
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)
: 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)),
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_payload_length_(IP_PACKET_SIZE - 28), // Default is IP-v4/UDP.
payload_type_(-1),
payload_type_map_(),
rtp_header_extension_map_(),
transmission_time_offset_(0),
absolute_send_time_(0),
rotation_(kVideoRotation_0),
video_rotation_active_(false),
transport_sequence_number_(0),
playout_delay_active_(false),
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
ssrc_db_(SSRCDatabase::GetSSRCDatabase()),
remote_ssrc_(0),
sequence_number_forced_(false),
ssrc_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),
retransmission_rate_limiter_(retransmission_rate_limiter) {
ssrc_ = ssrc_db_->CreateSSRC();
RTC_DCHECK(ssrc_ != 0);
ssrc_rtx_ = ssrc_db_->CreateSSRC();
RTC_DCHECK(ssrc_rtx_ != 0);
// 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);
}
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.
if (remote_ssrc_ != 0) {
ssrc_db_->ReturnSSRC(remote_ssrc_);
}
ssrc_db_->ReturnSSRC(ssrc_);
SSRCDatabase::ReturnSSRCDatabase();
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);
}
}
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::SetTransmissionTimeOffset(int32_t transmission_time_offset) {
if (transmission_time_offset > (0x800000 - 1) ||
transmission_time_offset < -(0x800000 - 1)) { // Word24.
return -1;
}
rtc::CritScope lock(&send_critsect_);
transmission_time_offset_ = transmission_time_offset;
return 0;
}
int32_t RTPSender::SetAbsoluteSendTime(uint32_t absolute_send_time) {
if (absolute_send_time > 0xffffff) { // UWord24.
return -1;
}
rtc::CritScope lock(&send_critsect_);
absolute_send_time_ = absolute_send_time;
return 0;
}
void RTPSender::SetVideoRotation(VideoRotation rotation) {
rtc::CritScope lock(&send_critsect_);
rotation_ = rotation;
}
int32_t RTPSender::SetTransportSequenceNumber(uint16_t sequence_number) {
rtc::CritScope lock(&send_critsect_);
transport_sequence_number_ = sequence_number;
return 0;
}
int32_t RTPSender::RegisterRtpHeaderExtension(RTPExtensionType type,
uint8_t id) {
rtc::CritScope lock(&send_critsect_);
switch (type) {
case kRtpExtensionVideoRotation:
video_rotation_active_ = false;
return rtp_header_extension_map_.RegisterInactive(type, id);
case kRtpExtensionPlayoutDelay:
playout_delay_active_ = false;
return rtp_header_extension_map_.RegisterInactive(type, id);
case kRtpExtensionTransmissionTimeOffset:
case kRtpExtensionAbsoluteSendTime:
case kRtpExtensionAudioLevel:
case kRtpExtensionTransportSequenceNumber:
return rtp_header_extension_map_.Register(type, id);
case kRtpExtensionNone:
case kRtpExtensionNumberOfExtensions:
LOG(LS_ERROR) << "Invalid RTP extension type for registration";
return -1;
}
return -1;
}
bool RTPSender::IsRtpHeaderExtensionRegistered(RTPExtensionType type) {
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);
}
size_t RTPSender::RtpHeaderExtensionLength() const {
rtc::CritScope lock(&send_critsect_);
return rtp_header_extension_map_.GetTotalLengthInBytes();
}
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;
assert(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->audio &&
payload->typeSpecific.Audio.frequency == frequency &&
(payload->typeSpecific.Audio.rate == rate ||
payload->typeSpecific.Audio.rate == 0 || rate == 0)) {
payload->typeSpecific.Audio.rate = rate;
// Ensure that we update the rate if new or old is zero.
return 0;
}
if (!audio_configured_ && !payload->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::SetSendPayloadType(int8_t payload_type) {
rtc::CritScope lock(&send_critsect_);
payload_type_ = payload_type;
}
int8_t RTPSender::SendPayloadType() const {
rtc::CritScope lock(&send_critsect_);
return payload_type_;
}
int RTPSender::SendPayloadFrequency() const {
return audio_ != NULL ? audio_->AudioFrequency() : kVideoPayloadTypeFrequency;
}
void RTPSender::SetMaxPayloadLength(size_t max_payload_length) {
// Sanity check.
RTC_DCHECK(max_payload_length >= 100 && max_payload_length <= IP_PACKET_SIZE)
<< "Invalid max payload length: " << max_payload_length;
rtc::CritScope lock(&send_critsect_);
max_payload_length_ = max_payload_length;
}
size_t RTPSender::MaxDataPayloadLength() const {
if (audio_configured_) {
return max_payload_length_ - RtpHeaderLength();
} else {
return max_payload_length_ - RtpHeaderLength() // RTP overhead.
- video_->FECPacketOverhead() // FEC/ULP/RED overhead.
- (RtxStatus() ? kRtxHeaderSize : 0); // RTX overhead.
}
}
size_t RTPSender::MaxPayloadLength() const {
return max_payload_length_;
}
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_ = ssrc;
}
uint32_t RTPSender::RtxSsrc() const {
rtc::CritScope lock(&send_critsect_);
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) {
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) {
LOG(LS_ERROR) << "Invalid payload_type " << payload_type;
return -1;
}
if (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()) {
LOG(LS_WARNING) << "Payload type " << static_cast<int>(payload_type)
<< " not registered.";
return -1;
}
SetSendPayloadType(payload_type);
RtpUtility::Payload* payload = it->second;
assert(payload);
if (!payload->audio && !audio_configured_) {
video_->SetVideoCodecType(payload->typeSpecific.Video.videoCodecType);
*video_type = payload->typeSpecific.Video.videoCodecType;
}
return 0;
}
bool RTPSender::ActivateCVORtpHeaderExtension() {
if (!video_rotation_active_) {
rtc::CritScope lock(&send_critsect_);
if (rtp_header_extension_map_.SetActive(kRtpExtensionVideoRotation, true)) {
video_rotation_active_ = true;
}
}
return video_rotation_active_;
}
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) {
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_);
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) {
LOG(LS_ERROR) << "Don't send data with unknown payload type: "
<< static_cast<int>(payload_type) << ".";
return false;
}
bool result;
if (audio_configured_) {
TRACE_EVENT_ASYNC_STEP1("webrtc", "Audio", rtp_timestamp, "Send", "type",
FrameTypeToString(frame_type));
assert(frame_type == kAudioFrameSpeech || frame_type == kAudioFrameCN ||
frame_type == kEmptyFrame);
result = audio_->SendAudio(frame_type, payload_type, rtp_timestamp,
payload_data, payload_size, fragmentation);
} else {
TRACE_EVENT_ASYNC_STEP1("webrtc", "Video", capture_time_ms,
"Send", "type", FrameTypeToString(frame_type));
assert(frame_type != kAudioFrameSpeech && frame_type != kAudioFrameCN);
if (frame_type == kEmptyFrame)
return true;
if (rtp_header) {
playout_delay_oracle_.UpdateRequest(ssrc, rtp_header->playout_delay,
sequence_number);
}
// Update the active/inactive status of playout delay extension based
// on what the oracle indicates.
{
rtc::CritScope lock(&send_critsect_);
bool send_playout_delay = playout_delay_oracle_.send_playout_delay();
if (playout_delay_active_ != send_playout_delay) {
playout_delay_active_ = send_playout_delay;
rtp_header_extension_map_.SetActive(kRtpExtensionPlayoutDelay,
playout_delay_active_);
}
}
result = video_->SendVideo(video_type, frame_type, payload_type,
rtp_timestamp, capture_time_ms, payload_data,
payload_size, fragmentation, rtp_header);
}
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,
int probe_cluster_id) {
{
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, probe_cluster_id))
break;
bytes_left -= payload_size;
}
return bytes_to_send - bytes_left;
}
size_t RTPSender::SendPadData(size_t bytes,
bool timestamp_provided,
uint32_t timestamp,
int64_t capture_time_ms) {
return SendPadData(bytes, timestamp_provided, timestamp, capture_time_ms,
PacketInfo::kNotAProbe);
}
size_t RTPSender::SendPadData(size_t bytes,
bool timestamp_provided,
uint32_t timestamp,
int64_t capture_time_ms,
int probe_cluster_id) {
// 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.
size_t padding_bytes_in_packet =
std::min(MaxDataPayloadLength(), kMaxPaddingLength);
size_t bytes_sent = 0;
bool using_transport_seq =
IsRtpHeaderExtensionRegistered(kRtpExtensionTransportSequenceNumber) &&
transport_sequence_number_allocator_;
for (; bytes > 0; bytes -= padding_bytes_in_packet) {
if (bytes < padding_bytes_in_packet)
bytes = padding_bytes_in_packet;
uint32_t ssrc;
uint16_t sequence_number;
int payload_type;
bool over_rtx;
{
rtc::CritScope lock(&send_critsect_);
if (!sending_media_)
return bytes_sent;
if (!timestamp_provided) {
timestamp = last_rtp_timestamp_;
capture_time_ms = capture_time_ms_;
}
if (rtx_ == kRtxOff) {
// Without RTX we can't send padding in the middle of frames.
if (!last_packet_marker_bit_)
return 0;
ssrc = ssrc_;
sequence_number = sequence_number_;
++sequence_number_;
payload_type = 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(
kRtpExtensionAbsoluteSendTime) ||
using_transport_seq)) {
return 0;
}
// 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 +=
(clock_->TimeInMilliseconds() - last_timestamp_time_ms_) * 90;
capture_time_ms +=
(clock_->TimeInMilliseconds() - last_timestamp_time_ms_);
}
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_, IP_PACKET_SIZE);
padding_packet.SetPayloadType(payload_type);
padding_packet.SetMarker(false);
padding_packet.SetSequenceNumber(sequence_number);
padding_packet.SetTimestamp(timestamp);
padding_packet.SetSsrc(ssrc);
int64_t now_ms = clock_->TimeInMilliseconds();
if (capture_time_ms > 0) {
padding_packet.SetExtension<TransmissionOffset>(
kTimestampTicksPerMs * (now_ms - capture_time_ms));
}
padding_packet.SetExtension<AbsoluteSendTime>(now_ms);
PacketOptions options;
bool has_transport_seq_no =
UpdateTransportSequenceNumber(&padding_packet, &options.packet_id);
padding_packet.SetPadding(padding_bytes_in_packet, &random_);
if (has_transport_seq_no && transport_feedback_observer_)
transport_feedback_observer_->AddPacket(
options.packet_id,
padding_packet.payload_size() + padding_packet.padding_size(),
probe_cluster_id);
if (!SendPacketToNetwork(padding_packet, options))
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) {
packet_history_.SetStorePacketsStatus(enable, number_to_store);
}
bool RTPSender::StorePackets() const {
return packet_history_.StorePackets();
}
int32_t RTPSender::ReSendPacket(uint16_t packet_id, int64_t min_resend_time) {
std::unique_ptr<RtpPacketToSend> packet =
packet_history_.GetPacketAndSetSendTime(packet_id, min_resend_time, true);
if (!packet) {
// Packet not found.
return 0;
}
// Check if we're overusing retransmission bitrate.
// TODO(sprang): Add histograms for nack success or failure reasons.
RTC_DCHECK(retransmission_rate_limiter_);
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 =
packet->capture_time_ms() + clock_delta_ms_;
paced_sender_->InsertPacket(RtpPacketSender::kNormalPriority,
packet->Ssrc(), packet->SequenceNumber(),
corrected_capture_tims_ms,
packet->payload_size(), true);
return packet->size();
}
bool rtx = (RtxStatus() & kRtxRetransmitted) > 0;
int32_t packet_size = static_cast<int32_t>(packet->size());
if (!PrepareAndSendPacket(std::move(packet), rtx, true,
PacketInfo::kNotAProbe))
return -1;
return packet_size;
}
bool RTPSender::SendPacketToNetwork(const RtpPacketToSend& packet,
const PacketOptions& options) {
int bytes_sent = -1;
if (transport_) {
bytes_sent = transport_->SendRtp(packet.data(), packet.size(), options)
? static_cast<int>(packet.size())
: -1;
if (event_log_ && bytes_sent > 0) {
event_log_->LogRtpHeader(kOutgoingPacket, MediaType::ANY, packet.data(),
packet.size());
}
}
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) {
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);
for (uint16_t seq_no : nack_sequence_numbers) {
const int32_t bytes_sent = ReSendPacket(seq_no, 5 + avg_rtt);
if (bytes_sent < 0) {
// Failed to send one Sequence number. Give up the rest in this nack.
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);
}
// Called from pacer when we can send the packet.
bool RTPSender::TimeToSendPacket(uint16_t sequence_number,
int64_t capture_time_ms,
bool retransmission,
int probe_cluster_id) {
std::unique_ptr<RtpPacketToSend> packet =
packet_history_.GetPacketAndSetSendTime(sequence_number, 0,
retransmission);
if (!packet) {
// Packet cannot be found. Allow sending to continue.
return true;
}
return PrepareAndSendPacket(
std::move(packet),
retransmission && (RtxStatus() & kRtxRetransmitted) > 0, retransmission,
probe_cluster_id);
}
bool RTPSender::PrepareAndSendPacket(std::unique_ptr<RtpPacketToSend> packet,
bool send_over_rtx,
bool is_retransmit,
int probe_cluster_id) {
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();
}
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>(now_ms);
PacketOptions options;
if (UpdateTransportSequenceNumber(packet_to_send, &options.packet_id) &&
transport_feedback_observer_) {
transport_feedback_observer_->AddPacket(
options.packet_id,
packet_to_send->payload_size() + packet_to_send->padding_size(),
probe_cluster_id);
}
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))
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) {
StreamDataCounters* counters;
// Get ssrc before taking statistics_crit_ to avoid possible deadlock.
uint32_t ssrc = is_rtx ? RtxSsrc() : SSRC();
int64_t now_ms = clock_->TimeInMilliseconds();
rtc::CritScope lock(&statistics_crit_);
if (is_rtx) {
counters = &rtx_rtp_stats_;
} else {
counters = &rtp_stats_;
}
total_bitrate_sent_.Update(packet.size(), now_ms);
if (counters->first_packet_time_ms == -1) {
counters->first_packet_time_ms = clock_->TimeInMilliseconds();
}
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, ssrc);
}
}
bool RTPSender::IsFecPacket(const RtpPacketToSend& packet) const {
if (!video_) {
return false;
}
bool fec_enabled;
uint8_t pt_red;
uint8_t pt_fec;
video_->GenericFECStatus(&fec_enabled, &pt_red, &pt_fec);
return fec_enabled && packet.PayloadType() == pt_red &&
packet.payload()[0] == pt_fec;
}
size_t RTPSender::TimeToSendPadding(size_t bytes, int probe_cluster_id) {
if (audio_configured_ || bytes == 0)
return 0;
size_t bytes_sent = TrySendRedundantPayloads(bytes, probe_cluster_id);
if (bytes_sent < bytes)
bytes_sent +=
SendPadData(bytes - bytes_sent, false, 0, 0, probe_cluster_id);
return bytes_sent;
}
bool RTPSender::SendToNetwork(uint8_t* buffer,
size_t payload_length,
size_t rtp_header_length,
int64_t capture_time_ms,
StorageType storage,
RtpPacketSender::Priority priority) {
size_t length = payload_length + rtp_header_length;
std::unique_ptr<RtpPacketToSend> packet(
new RtpPacketToSend(&rtp_header_extension_map_, length));
RTC_CHECK(packet->Parse(buffer, length));
packet->set_capture_time_ms(capture_time_ms);
return SendToNetwork(std::move(packet), storage, priority);
}
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>(now_ms);
if (paced_sender_) {
uint16_t seq_no = packet->SequenceNumber();
uint32_t ssrc = packet->Ssrc();
// 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();
packet_history_.PutRtpPacket(std::move(packet), storage, false);
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) &&
transport_feedback_observer_) {
transport_feedback_observer_->AddPacket(
options.packet_id, packet->payload_size() + packet->padding_size(),
PacketInfo::kNotAProbe);
}
UpdateDelayStatistics(packet->capture_time_ms(), now_ms);
UpdateOnSendPacket(options.packet_id, packet->capture_time_ms(),
packet->Ssrc());
bool sent = SendPacketToNetwork(*packet, options);
if (sent) {
{
rtc::CritScope lock(&send_critsect_);
media_has_been_sent_ = true;
}
UpdateRtpStats(*packet, false, false);
}
// Mark the packet as sent in the history even if send failed. Dropping a
// packet here should be treated as any other packet drop so we should be
// ready for a retransmission.
packet_history_.PutRtpPacket(std::move(packet), storage, true);
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;
int avg_delay_ms = 0;
int max_delay_ms = 0;
{
rtc::CritScope lock(&send_critsect_);
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(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_);
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 += RtpHeaderExtensionLength();
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_;
}
size_t RTPSender::CreateRtpHeader(uint8_t* header,
int8_t payload_type,
uint32_t ssrc,
bool marker_bit,
uint32_t timestamp,
uint16_t sequence_number,
const std::vector<uint32_t>& csrcs) const {
header[0] = 0x80; // version 2.
header[1] = static_cast<uint8_t>(payload_type);
if (marker_bit) {
header[1] |= kRtpMarkerBitMask; // Marker bit is set.
}
ByteWriter<uint16_t>::WriteBigEndian(header + 2, sequence_number);
ByteWriter<uint32_t>::WriteBigEndian(header + 4, timestamp);
ByteWriter<uint32_t>::WriteBigEndian(header + 8, ssrc);
int32_t rtp_header_length = kRtpHeaderLength;
if (csrcs.size() > 0) {
uint8_t* ptr = &header[rtp_header_length];
for (size_t i = 0; i < csrcs.size(); ++i) {
ByteWriter<uint32_t>::WriteBigEndian(ptr, csrcs[i]);
ptr += 4;
}
header[0] = (header[0] & 0xf0) | csrcs.size();
// Update length of header.
rtp_header_length += sizeof(uint32_t) * csrcs.size();
}
uint16_t len =
BuildRtpHeaderExtension(header + rtp_header_length, marker_bit);
if (len > 0) {
header[0] |= 0x10; // Set extension bit.
rtp_header_length += len;
}
return rtp_header_length;
}
std::unique_ptr<RtpPacketToSend> RTPSender::AllocatePacket() const {
rtc::CritScope lock(&send_critsect_);
std::unique_ptr<RtpPacketToSend> packet(
new RtpPacketToSend(&rtp_header_extension_map_, max_payload_length_));
packet->SetSsrc(ssrc_);
packet->SetCsrcs(csrcs_);
// Reserve extensions, if registered, RtpSender set in SendToNetwork.
packet->ReserveExtension<AbsoluteSendTime>();
packet->ReserveExtension<TransmissionOffset>();
packet->ReserveExtension<TransportSequenceNumber>();
return packet;
}
bool RTPSender::AssignSequenceNumber(RtpPacketToSend* packet) {
rtc::CritScope lock(&send_critsect_);
if (!sending_media_)
return false;
RTC_DCHECK_EQ(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();
// 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;
}
int32_t RTPSender::BuildRTPheader(uint8_t* data_buffer,
int8_t payload_type,
bool marker_bit,
uint32_t capture_timestamp,
int64_t capture_time_ms,
bool timestamp_provided,
bool inc_sequence_number) {
return BuildRtpHeader(data_buffer, payload_type, marker_bit,
capture_timestamp, capture_time_ms);
}
int32_t RTPSender::BuildRtpHeader(uint8_t* data_buffer,
int8_t payload_type,
bool marker_bit,
uint32_t rtp_timestamp,
int64_t capture_time_ms) {
assert(payload_type >= 0);
rtc::CritScope lock(&send_critsect_);
if (!sending_media_)
return -1;
last_rtp_timestamp_ = rtp_timestamp;
last_timestamp_time_ms_ = clock_->TimeInMilliseconds();
uint32_t sequence_number = sequence_number_++;
capture_time_ms_ = capture_time_ms;
last_packet_marker_bit_ = marker_bit;
return CreateRtpHeader(data_buffer, payload_type, ssrc_, marker_bit,
rtp_timestamp, sequence_number, csrcs_);
}
uint16_t RTPSender::BuildRtpHeaderExtension(uint8_t* data_buffer,
bool marker_bit) const {
if (rtp_header_extension_map_.Size() <= 0) {
return 0;
}
// RTP header extension, RFC 3550.
// 0 1 2 3
// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
// | defined by profile | length |
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
// | header extension |
// | .... |
//
const uint32_t kPosLength = 2;
const uint32_t kHeaderLength = kRtpOneByteHeaderLength;
// Add extension ID (0xBEDE).
ByteWriter<uint16_t>::WriteBigEndian(data_buffer,
kRtpOneByteHeaderExtensionId);
// Add extensions.
uint16_t total_block_length = 0;
RTPExtensionType type = rtp_header_extension_map_.First();
while (type != kRtpExtensionNone) {
uint8_t block_length = 0;
uint8_t* extension_data = &data_buffer[kHeaderLength + total_block_length];
switch (type) {
case kRtpExtensionTransmissionTimeOffset:
block_length = BuildTransmissionTimeOffsetExtension(extension_data);
break;
case kRtpExtensionAudioLevel:
block_length = BuildAudioLevelExtension(extension_data);
break;
case kRtpExtensionAbsoluteSendTime:
block_length = BuildAbsoluteSendTimeExtension(extension_data);
break;
case kRtpExtensionVideoRotation:
block_length = BuildVideoRotationExtension(extension_data);
break;
case kRtpExtensionTransportSequenceNumber:
block_length = BuildTransportSequenceNumberExtension(
extension_data, transport_sequence_number_);
break;
case kRtpExtensionPlayoutDelay: {
PlayoutDelay playout_delay = playout_delay_oracle_.playout_delay();
block_length = BuildPlayoutDelayExtension(
extension_data, playout_delay.min_ms, playout_delay.max_ms);
break;
}
default:
assert(false);
}
total_block_length += block_length;
type = rtp_header_extension_map_.Next(type);
}
if (total_block_length == 0) {
// No extension added.
return 0;
}
// Add padding elements until we've filled a 32 bit block.
size_t padding_bytes =
RtpUtility::Word32Align(total_block_length) - total_block_length;
if (padding_bytes > 0) {
memset(&data_buffer[kHeaderLength + total_block_length], 0, padding_bytes);
total_block_length += padding_bytes;
}
// Set header length (in number of Word32, header excluded).
ByteWriter<uint16_t>::WriteBigEndian(data_buffer + kPosLength,
total_block_length / 4);
// Total added length.
return kHeaderLength + total_block_length;
}
uint8_t RTPSender::BuildTransmissionTimeOffsetExtension(
uint8_t* data_buffer) const {
// From RFC 5450: Transmission Time Offsets in RTP Streams.
//
// The transmission time is signaled to the receiver in-band using the
// general mechanism for RTP header extensions [RFC5285]. The payload
// of this extension (the transmitted value) is a 24-bit signed integer.
// When added to the RTP timestamp of the packet, it represents the
// "effective" RTP transmission time of the packet, on the RTP
// timescale.
//
// The form of the transmission offset extension block:
//
// 0 1 2 3
// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
// | ID | len=2 | transmission offset |
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
// Get id defined by user.
uint8_t id;
if (rtp_header_extension_map_.GetId(kRtpExtensionTransmissionTimeOffset,
&id) != 0) {
// Not registered.
return 0;
}
size_t pos = 0;
const uint8_t len = 2;
data_buffer[pos++] = (id << 4) + len;
ByteWriter<int32_t, 3>::WriteBigEndian(data_buffer + pos,
transmission_time_offset_);
pos += 3;
assert(pos == kTransmissionTimeOffsetLength);
return kTransmissionTimeOffsetLength;
}
uint8_t RTPSender::BuildAudioLevelExtension(uint8_t* data_buffer) const {
// An RTP Header Extension for Client-to-Mixer Audio Level Indication
//
// https://datatracker.ietf.org/doc/draft-lennox-avt-rtp-audio-level-exthdr/
//
// The form of the audio level extension block:
//
// 0 1
// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
// | ID | len=0 |V| level |
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
//
// Get id defined by user.
uint8_t id;
if (rtp_header_extension_map_.GetId(kRtpExtensionAudioLevel, &id) != 0) {
// Not registered.
return 0;
}
size_t pos = 0;
const uint8_t len = 0;
data_buffer[pos++] = (id << 4) + len;
data_buffer[pos++] = (1 << 7) + 0; // Voice, 0 dBov.
assert(pos == kAudioLevelLength);
return kAudioLevelLength;
}
uint8_t RTPSender::BuildAbsoluteSendTimeExtension(uint8_t* data_buffer) const {
// Absolute send time in RTP streams.
//
// The absolute send time is signaled to the receiver in-band using the
// general mechanism for RTP header extensions [RFC5285]. The payload
// of this extension (the transmitted value) is a 24-bit unsigned integer
// containing the sender's current time in seconds as a fixed point number
// with 18 bits fractional part.
//
// The form of the absolute send time extension block:
//
// 0 1 2 3
// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
// | ID | len=2 | absolute send time |
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
// Get id defined by user.
uint8_t id;
if (rtp_header_extension_map_.GetId(kRtpExtensionAbsoluteSendTime,
&id) != 0) {
// Not registered.
return 0;
}
size_t pos = 0;
const uint8_t len = 2;
data_buffer[pos++] = (id << 4) + len;
ByteWriter<uint32_t, 3>::WriteBigEndian(data_buffer + pos,
absolute_send_time_);
pos += 3;
assert(pos == kAbsoluteSendTimeLength);
return kAbsoluteSendTimeLength;
}
uint8_t RTPSender::BuildVideoRotationExtension(uint8_t* data_buffer) const {
// Coordination of Video Orientation in RTP streams.
//
// Coordination of Video Orientation consists in signaling of the current
// orientation of the image captured on the sender side to the receiver for
// appropriate rendering and displaying.
//
// 0 1
// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
// | ID | len=0 |0 0 0 0 C F R R|
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
//
// Get id defined by user.
uint8_t id;
if (rtp_header_extension_map_.GetId(kRtpExtensionVideoRotation, &id) != 0) {
// Not registered.
return 0;
}
size_t pos = 0;
const uint8_t len = 0;
data_buffer[pos++] = (id << 4) + len;
data_buffer[pos++] = ConvertVideoRotationToCVOByte(rotation_);
assert(pos == kVideoRotationLength);
return kVideoRotationLength;
}
uint8_t RTPSender::BuildTransportSequenceNumberExtension(
uint8_t* data_buffer,
uint16_t sequence_number) const {
// 0 1 2
// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
// | ID | L=1 |transport wide sequence number |
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
// Get id defined by user.
uint8_t id;
if (rtp_header_extension_map_.GetId(kRtpExtensionTransportSequenceNumber,
&id) != 0) {
// Not registered.
return 0;
}
size_t pos = 0;
const uint8_t len = 1;
data_buffer[pos++] = (id << 4) + len;
ByteWriter<uint16_t>::WriteBigEndian(data_buffer + pos, sequence_number);
pos += 2;
assert(pos == kTransportSequenceNumberLength);
return kTransportSequenceNumberLength;
}
uint8_t RTPSender::BuildPlayoutDelayExtension(
uint8_t* data_buffer,
uint16_t min_playout_delay_ms,
uint16_t max_playout_delay_ms) const {
RTC_DCHECK_LE(min_playout_delay_ms, kPlayoutDelayMaxMs);
RTC_DCHECK_LE(max_playout_delay_ms, kPlayoutDelayMaxMs);
RTC_DCHECK_LE(min_playout_delay_ms, max_playout_delay_ms);
// 0 1 2 3
// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
// | ID | len=2 | MIN delay | MAX delay |
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
uint8_t id;
if (rtp_header_extension_map_.GetId(kRtpExtensionPlayoutDelay, &id) != 0) {
// Not registered.
return 0;
}
size_t pos = 0;
const uint8_t len = 2;
// Convert MS to value to be sent on extension header.
uint16_t min_playout = min_playout_delay_ms / kPlayoutDelayGranularityMs;
uint16_t max_playout = max_playout_delay_ms / kPlayoutDelayGranularityMs;
data_buffer[pos++] = (id << 4) + len;
data_buffer[pos++] = min_playout >> 4;
data_buffer[pos++] = ((min_playout & 0xf) << 4) | (max_playout >> 8);
data_buffer[pos++] = max_playout & 0xff;
assert(pos == kPlayoutDelayLength);
return kPlayoutDelayLength;
}
bool RTPSender::FindHeaderExtensionPosition(RTPExtensionType type,
const uint8_t* rtp_packet,
size_t rtp_packet_length,
const RTPHeader& rtp_header,
size_t* position) const {
// Get length until start of header extension block.
int extension_block_pos =
rtp_header_extension_map_.GetLengthUntilBlockStartInBytes(type);
if (extension_block_pos < 0) {
LOG(LS_WARNING) << "Failed to find extension position for " << type
<< " as it is not registered.";
return false;
}
HeaderExtension header_extension(type);
size_t extension_pos =
kRtpHeaderLength + rtp_header.numCSRCs * sizeof(uint32_t);
size_t block_pos = extension_pos + extension_block_pos;
if (rtp_packet_length < block_pos + header_extension.length ||
rtp_header.headerLength < block_pos + header_extension.length) {
LOG(LS_WARNING) << "Failed to find extension position for " << type
<< " as the length is invalid.";
return false;
}
// Verify that header contains extension.
if (!(rtp_packet[extension_pos] == 0xBE &&
rtp_packet[extension_pos + 1] == 0xDE)) {
LOG(LS_WARNING) << "Failed to find extension position for " << type
<< "as hdr extension not found.";
return false;
}
*position = block_pos;
return true;
}
RTPSender::ExtensionStatus RTPSender::VerifyExtension(
RTPExtensionType extension_type,
uint8_t* rtp_packet,
size_t rtp_packet_length,
const RTPHeader& rtp_header,
size_t extension_length_bytes,
size_t* extension_offset) const {
// Get id.
uint8_t id = 0;
if (rtp_header_extension_map_.GetId(extension_type, &id) != 0)
return ExtensionStatus::kNotRegistered;
size_t block_pos = 0;
if (!FindHeaderExtensionPosition(extension_type, rtp_packet,
rtp_packet_length, rtp_header, &block_pos))
return ExtensionStatus::kError;
// Verify first byte in block.
const uint8_t first_block_byte = (id << 4) + (extension_length_bytes - 2);
if (rtp_packet[block_pos] != first_block_byte)
return ExtensionStatus::kError;
*extension_offset = block_pos;
return ExtensionStatus::kOk;
}
bool RTPSender::UpdateAudioLevel(uint8_t* rtp_packet,
size_t rtp_packet_length,
const RTPHeader& rtp_header,
bool is_voiced,
uint8_t dBov) const {
size_t offset;
rtc::CritScope lock(&send_critsect_);
switch (VerifyExtension(kRtpExtensionAudioLevel, rtp_packet,
rtp_packet_length, rtp_header, kAudioLevelLength,
&offset)) {
case ExtensionStatus::kNotRegistered:
return false;
case ExtensionStatus::kError:
LOG(LS_WARNING) << "Failed to update audio level.";
return false;
case ExtensionStatus::kOk:
break;
default:
RTC_NOTREACHED();
}
rtp_packet[offset + 1] = (is_voiced ? 0x80 : 0x00) + (dBov & 0x7f);
return true;
}
bool RTPSender::UpdateVideoRotation(uint8_t* rtp_packet,
size_t rtp_packet_length,
const RTPHeader& rtp_header,
VideoRotation rotation) const {
size_t offset;
rtc::CritScope lock(&send_critsect_);
switch (VerifyExtension(kRtpExtensionVideoRotation, rtp_packet,
rtp_packet_length, rtp_header, kVideoRotationLength,
&offset)) {
case ExtensionStatus::kNotRegistered:
return false;
case ExtensionStatus::kError:
LOG(LS_WARNING) << "Failed to update CVO.";
return false;
case ExtensionStatus::kOk:
break;
default:
RTC_NOTREACHED();
}
rtp_packet[offset + 1] = ConvertVideoRotationToCVOByte(rotation);
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::SetSendingStatus(bool enabled) {
if (!enabled) {
rtc::CritScope lock(&send_critsect_);
if (!ssrc_forced_) {
// Generate a new SSRC.
ssrc_db_->ReturnSSRC(ssrc_);
ssrc_ = ssrc_db_->CreateSSRC();
RTC_DCHECK(ssrc_ != 0);
}
// Don't initialize seq number if SSRC passed externally.
if (!sequence_number_forced_ && !ssrc_forced_) {
// Generate a new sequence number.
sequence_number_ = random_.Rand(1, kMaxInitRtpSeqNumber);
}
}
}
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_;
}
uint32_t RTPSender::GenerateNewSSRC() {
// If configured via API, return 0.
rtc::CritScope lock(&send_critsect_);
if (ssrc_forced_) {
return 0;
}
ssrc_ = ssrc_db_->CreateSSRC();
RTC_DCHECK(ssrc_ != 0);
return ssrc_;
}
void RTPSender::SetSSRC(uint32_t ssrc) {
// This is configured via the API.
rtc::CritScope lock(&send_critsect_);
if (ssrc_ == ssrc && ssrc_forced_) {
return; // Since it's same ssrc, don't reset anything.
}
ssrc_forced_ = true;
ssrc_db_->ReturnSSRC(ssrc_);
ssrc_db_->RegisterSSRC(ssrc);
ssrc_ = ssrc;
if (!sequence_number_forced_) {
sequence_number_ = random_.Rand(1, kMaxInitRtpSeqNumber);
}
}
uint32_t RTPSender::SSRC() const {
rtc::CritScope lock(&send_critsect_);
return ssrc_;
}
void RTPSender::SetCsrcs(const std::vector<uint32_t>& csrcs) {
assert(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::SetAudioPacketSize(uint16_t packet_size_samples) {
if (!audio_configured_) {
return -1;
}
return audio_->SetAudioPacketSize(packet_size_samples);
}
int32_t RTPSender::SetAudioLevel(uint8_t level_d_bov) {
return audio_->SetAudioLevel(level_d_bov);
}
RtpVideoCodecTypes RTPSender::VideoCodecType() const {
assert(!audio_configured_ && "Sender is an audio stream!");
return video_->VideoCodecType();
}
void RTPSender::SetGenericFECStatus(bool enable,
uint8_t payload_type_red,
uint8_t payload_type_fec) {
RTC_DCHECK(!audio_configured_);
video_->SetGenericFECStatus(enable, payload_type_red, payload_type_fec);
}
void RTPSender::GenericFECStatus(bool* enable,
uint8_t* payload_type_red,
uint8_t* payload_type_fec) const {
RTC_DCHECK(!audio_configured_);
video_->GenericFECStatus(enable, payload_type_red, payload_type_fec);
}
int32_t RTPSender::SetFecParameters(
const FecProtectionParams *delta_params,
const FecProtectionParams *key_params) {
if (audio_configured_) {
return -1;
}
video_->SetFecParameters(delta_params, key_params);
return 0;
}
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;
// Replace payload type, if a specific type is set for RTX.
auto kv = rtx_payload_type_map_.find(packet.PayloadType());
// Use rtx mapping associated with media codec if we can't find one,
// assume it's red.
// TODO(holmer): Remove once old Chrome versions don't rely on this.
if (kv == rtx_payload_type_map_.end())
kv = rtx_payload_type_map_.find(payload_type_);
if (kv != rtx_payload_type_map_.end())
rtx_packet->SetPayloadType(kv->second);
// Replace sequence number.
rtx_packet->SetSequenceNumber(sequence_number_rtx_++);
// Replace SSRC.
rtx_packet->SetSsrc(ssrc_rtx_);
}
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.
memcpy(rtx_payload + kRtxHeaderSize, packet.payload(), packet.payload_size());
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;
}
} // namespace webrtc