Google Git
Sign in
webrtc / src / webrtc / afa6c6b1e8e89cb18ab427776639b86b613a0584 / . / modules / rtp_rtcp / source / rtp_sender.cc
blob: cda776bf22b2cdcef3510233fe94acda11286aa4 [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 <stdlib.h> // srand
#include <algorithm>
#include <utility>
#include "webrtc/base/checks.h"
#include "webrtc/base/logging.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/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 {
// Max in the RFC 3550 is 255 bytes, we limit it to be modulus 32 for SRTP.
static const size_t kMaxPaddingLength = 224;
static const int kSendSideDelayWindowMs = 1000;
static const uint32_t kAbsSendTimeFraction = 18;
namespace {
const size_t kRtpHeaderLength = 12;
const uint16_t kMaxInitRtpSeqNumber = 32767; // 2^15 -1.
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 "";
}
// TODO(holmer): Merge this with the implementation in
// remote_bitrate_estimator_abs_send_time.cc.
uint32_t ConvertMsTo24Bits(int64_t time_ms) {
uint32_t time_24_bits =
static_cast<uint32_t>(
((static_cast<uint64_t>(time_ms) << kAbsSendTimeFraction) + 500) /
1000) &
0x00FFFFFF;
return time_24_bits;
}
} // namespace
RTPSender::BitrateAggregator::BitrateAggregator(
BitrateStatisticsObserver* bitrate_callback)
: callback_(bitrate_callback),
total_bitrate_observer_(*this),
retransmit_bitrate_observer_(*this),
ssrc_(0) {}
void RTPSender::BitrateAggregator::OnStatsUpdated() const {
if (callback_) {
callback_->Notify(total_bitrate_observer_.statistics(),
retransmit_bitrate_observer_.statistics(), ssrc_);
}
}
Bitrate::Observer* RTPSender::BitrateAggregator::total_bitrate_observer() {
return &total_bitrate_observer_;
}
Bitrate::Observer* RTPSender::BitrateAggregator::retransmit_bitrate_observer() {
return &retransmit_bitrate_observer_;
}
void RTPSender::BitrateAggregator::set_ssrc(uint32_t ssrc) {
ssrc_ = ssrc;
}
RTPSender::BitrateAggregator::BitrateObserver::BitrateObserver(
const BitrateAggregator& aggregator)
: aggregator_(aggregator) {}
// Implements Bitrate::Observer.
void RTPSender::BitrateAggregator::BitrateObserver::BitrateUpdated(
const BitrateStatistics& stats) {
statistics_ = stats;
aggregator_.OnStatsUpdated();
}
const BitrateStatistics&
RTPSender::BitrateAggregator::BitrateObserver::statistics() const {
return statistics_;
}
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)
: clock_(clock),
// TODO(holmer): Remove this conversion?
clock_delta_ms_(clock_->TimeInMilliseconds() - rtc::TimeMillis()),
random_(clock_->TimeInMicroseconds()),
bitrates_(bitrate_callback),
total_bitrate_sent_(clock, bitrates_.total_bitrate_observer()),
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),
cvo_mode_(kCVONone),
transport_sequence_number_(0),
// NACK.
nack_byte_count_times_(),
nack_byte_count_(),
nack_bitrate_(clock, bitrates_.retransmit_bitrate_observer()),
packet_history_(clock),
// Statistics
rtp_stats_callback_(NULL),
frame_count_observer_(frame_count_observer),
send_side_delay_observer_(send_side_delay_observer),
event_log_(event_log),
send_packet_observer_(send_packet_observer),
// RTP variables
start_timestamp_forced_(false),
start_timestamp_(0),
ssrc_db_(SSRCDatabase::GetSSRCDatabase()),
remote_ssrc_(0),
sequence_number_forced_(false),
ssrc_forced_(false),
timestamp_(0),
capture_time_ms_(0),
last_timestamp_time_ms_(0),
media_has_been_sent_(false),
last_packet_marker_bit_(false),
csrcs_(),
rtx_(kRtxOff),
target_bitrate_(0) {
memset(nack_byte_count_times_, 0, sizeof(nack_byte_count_times_));
memset(nack_byte_count_, 0, sizeof(nack_byte_count_));
// We need to seed the random generator for BuildPaddingPacket() below.
// TODO(holmer,tommi): Note that TimeInMilliseconds might return 0 on Mac
// early on in the process.
srand(static_cast<uint32_t>(clock_->TimeInMilliseconds()));
ssrc_ = ssrc_db_->CreateSSRC();
RTC_DCHECK(ssrc_ != 0);
ssrc_rtx_ = ssrc_db_->CreateSSRC();
RTC_DCHECK(ssrc_rtx_ != 0);
bitrates_.set_ssrc(ssrc_);
// 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);
}
}
void RTPSender::SetTargetBitrate(uint32_t bitrate) {
rtc::CritScope cs(&target_bitrate_critsect_);
target_bitrate_ = bitrate;
}
uint32_t RTPSender::GetTargetBitrate() {
rtc::CritScope cs(&target_bitrate_critsect_);
return target_bitrate_;
}
uint16_t RTPSender::ActualSendBitrateKbit() const {
return (uint16_t)(total_bitrate_sent_.BitrateNow() / 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 {
return nack_bitrate_.BitrateLast();
}
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_);
if (type == kRtpExtensionVideoRotation) {
cvo_mode_ = kCVOInactive;
return rtp_header_extension_map_.RegisterInactive(type, id);
}
return rtp_header_extension_map_.Register(type, id);
}
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::RtpHeaderExtensionTotalLength() 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 {
int rtx;
{
rtc::CritScope lock(&send_critsect_);
rtx = rtx_;
}
if (audio_configured_) {
return max_payload_length_ - RTPHeaderLength();
} else {
return max_payload_length_ - RTPHeaderLength() // RTP overhead.
- video_->FECPacketOverhead() // FEC/ULP/RED overhead.
- ((rtx) ? 2 : 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 (audio_configured_) {
int8_t red_pl_type = -1;
if (audio_->RED(&red_pl_type) == 0) {
// We have configured RED.
if (red_pl_type == payload_type) {
// And it's a match...
return 0;
}
}
}
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;
}
RTPSenderInterface::CVOMode RTPSender::ActivateCVORtpHeaderExtension() {
if (cvo_mode_ == kCVOInactive) {
rtc::CritScope lock(&send_critsect_);
if (rtp_header_extension_map_.SetActive(kRtpExtensionVideoRotation, true)) {
cvo_mode_ = kCVOActivated;
}
}
return cvo_mode_;
}
int32_t 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_hdr) {
uint32_t ssrc;
{
// Drop this packet if we're not sending media packets.
rtc::CritScope lock(&send_critsect_);
ssrc = ssrc_;
if (!sending_media_) {
return 0;
}
}
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 -1;
}
int32_t ret_val;
if (audio_configured_) {
TRACE_EVENT_ASYNC_STEP1("webrtc", "Audio", capture_timestamp,
"Send", "type", FrameTypeToString(frame_type));
assert(frame_type == kAudioFrameSpeech || frame_type == kAudioFrameCN ||
frame_type == kEmptyFrame);
ret_val = audio_->SendAudio(frame_type, payload_type, capture_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 0;
ret_val =
video_->SendVideo(video_type, frame_type, payload_type,
capture_timestamp, capture_time_ms, payload_data,
payload_size, fragmentation, rtp_hdr);
}
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 ret_val;
}
size_t RTPSender::TrySendRedundantPayloads(size_t bytes_to_send) {
{
rtc::CritScope lock(&send_critsect_);
if (!sending_media_)
return 0;
if ((rtx_ & kRtxRedundantPayloads) == 0)
return 0;
}
uint8_t buffer[IP_PACKET_SIZE];
int bytes_left = static_cast<int>(bytes_to_send);
while (bytes_left > 0) {
size_t length = bytes_left;
int64_t capture_time_ms;
if (!packet_history_.GetBestFittingPacket(buffer, &length,
&capture_time_ms)) {
break;
}
if (!PrepareAndSendPacket(buffer, length, capture_time_ms, true, false))
break;
RtpUtility::RtpHeaderParser rtp_parser(buffer, length);
RTPHeader rtp_header;
rtp_parser.Parse(&rtp_header);
bytes_left -= static_cast<int>(length - rtp_header.headerLength);
}
return bytes_to_send - bytes_left;
}
void RTPSender::BuildPaddingPacket(uint8_t* packet,
size_t header_length,
size_t padding_length) {
packet[0] |= 0x20; // Set padding bit.
int32_t* data = reinterpret_cast<int32_t*>(&(packet[header_length]));
// Fill data buffer with random data.
for (size_t j = 0; j < (padding_length >> 2); ++j) {
data[j] = rand(); // NOLINT
}
// Set number of padding bytes in the last byte of the packet.
packet[header_length + padding_length - 1] =
static_cast<uint8_t>(padding_length);
}
size_t RTPSender::SendPadData(size_t bytes,
bool timestamp_provided,
uint32_t timestamp,
int64_t capture_time_ms) {
// 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 = rtp_header_extension_map_.IsRegistered(
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 = 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;
}
}
uint8_t padding_packet[IP_PACKET_SIZE];
size_t header_length =
CreateRtpHeader(padding_packet, payload_type, ssrc, false, timestamp,
sequence_number, std::vector<uint32_t>());
BuildPaddingPacket(padding_packet, header_length, padding_bytes_in_packet);
size_t length = padding_bytes_in_packet + header_length;
int64_t now_ms = clock_->TimeInMilliseconds();
RtpUtility::RtpHeaderParser rtp_parser(padding_packet, length);
RTPHeader rtp_header;
rtp_parser.Parse(&rtp_header);
if (capture_time_ms > 0) {
UpdateTransmissionTimeOffset(
padding_packet, length, rtp_header, now_ms - capture_time_ms);
}
UpdateAbsoluteSendTime(padding_packet, length, rtp_header, now_ms);
PacketOptions options;
if (AllocateTransportSequenceNumber(&options.packet_id)) {
if (UpdateTransportSequenceNumber(options.packet_id, padding_packet,
length, rtp_header)) {
if (transport_feedback_observer_)
transport_feedback_observer_->AddPacket(options.packet_id, length,
true);
}
}
if (!SendPacketToNetwork(padding_packet, length, options))
break;
bytes_sent += padding_bytes_in_packet;
UpdateRtpStats(padding_packet, length, rtp_header, 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) {
size_t length = IP_PACKET_SIZE;
uint8_t data_buffer[IP_PACKET_SIZE];
int64_t capture_time_ms;
if (!packet_history_.GetPacketAndSetSendTime(packet_id, min_resend_time, true,
data_buffer, &length,
&capture_time_ms)) {
// Packet not found.
return 0;
}
if (paced_sender_) {
RtpUtility::RtpHeaderParser rtp_parser(data_buffer, length);
RTPHeader header;
if (!rtp_parser.Parse(&header)) {
assert(false);
return -1;
}
// Convert from TickTime to Clock since capture_time_ms is based on
// TickTime.
int64_t corrected_capture_tims_ms = capture_time_ms + clock_delta_ms_;
paced_sender_->InsertPacket(
RtpPacketSender::kNormalPriority, header.ssrc, header.sequenceNumber,
corrected_capture_tims_ms, length - header.headerLength, true);
return length;
}
int rtx = kRtxOff;
{
rtc::CritScope lock(&send_critsect_);
rtx = rtx_;
}
if (!PrepareAndSendPacket(data_buffer, length, capture_time_ms,
(rtx & kRtxRetransmitted) > 0, true)) {
return -1;
}
return static_cast<int32_t>(length);
}
bool RTPSender::SendPacketToNetwork(const uint8_t* packet,
size_t size,
const PacketOptions& options) {
int bytes_sent = -1;
if (transport_) {
bytes_sent = transport_->SendRtp(packet, size, options)
? static_cast<int>(size)
: -1;
if (event_log_ && bytes_sent > 0) {
event_log_->LogRtpHeader(kOutgoingPacket, MediaType::ANY, packet, size);
}
}
TRACE_EVENT_INSTANT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
"RTPSender::SendPacketToNetwork", "size", 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::list<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);
const int64_t now = clock_->TimeInMilliseconds();
uint32_t bytes_re_sent = 0;
uint32_t target_bitrate = GetTargetBitrate();
// Enough bandwidth to send NACK?
if (!ProcessNACKBitRate(now)) {
LOG(LS_INFO) << "NACK bitrate reached. Skip sending NACK response. Target "
<< target_bitrate;
return;
}
for (std::list<uint16_t>::const_iterator it = nack_sequence_numbers.begin();
it != nack_sequence_numbers.end(); ++it) {
const int32_t bytes_sent = ReSendPacket(*it, 5 + avg_rtt);
if (bytes_sent > 0) {
bytes_re_sent += bytes_sent;
} else if (bytes_sent == 0) {
// The packet has previously been resent.
// Try resending next packet in the list.
continue;
} else {
// Failed to send one Sequence number. Give up the rest in this nack.
LOG(LS_WARNING) << "Failed resending RTP packet " << *it
<< ", Discard rest of packets";
break;
}
// Delay bandwidth estimate (RTT * BW).
if (target_bitrate != 0 && avg_rtt) {
// kbits/s * ms = bits => bits/8 = bytes
size_t target_bytes =
(static_cast<size_t>(target_bitrate / 1000) * avg_rtt) >> 3;
if (bytes_re_sent > target_bytes) {
break; // Ignore the rest of the packets in the list.
}
}
}
if (bytes_re_sent > 0) {
UpdateNACKBitRate(bytes_re_sent, now);
}
}
bool RTPSender::ProcessNACKBitRate(uint32_t now) {
uint32_t num = 0;
size_t byte_count = 0;
const uint32_t kAvgIntervalMs = 1000;
uint32_t target_bitrate = GetTargetBitrate();
rtc::CritScope lock(&send_critsect_);
if (target_bitrate == 0) {
return true;
}
for (num = 0; num < NACK_BYTECOUNT_SIZE; ++num) {
if ((now - nack_byte_count_times_[num]) > kAvgIntervalMs) {
// Don't use data older than 1sec.
break;
} else {
byte_count += nack_byte_count_[num];
}
}
uint32_t time_interval = kAvgIntervalMs;
if (num == NACK_BYTECOUNT_SIZE) {
// More than NACK_BYTECOUNT_SIZE nack messages has been received
// during the last msg_interval.
if (nack_byte_count_times_[num - 1] <= now) {
time_interval = now - nack_byte_count_times_[num - 1];
}
}
return (byte_count * 8) < (target_bitrate / 1000 * time_interval);
}
void RTPSender::UpdateNACKBitRate(uint32_t bytes, int64_t now) {
rtc::CritScope lock(&send_critsect_);
if (bytes == 0)
return;
nack_bitrate_.Update(bytes);
// Save bitrate statistics.
// Shift all but first time.
for (int i = NACK_BYTECOUNT_SIZE - 2; i >= 0; i--) {
nack_byte_count_[i + 1] = nack_byte_count_[i];
nack_byte_count_times_[i + 1] = nack_byte_count_times_[i];
}
nack_byte_count_[0] = bytes;
nack_byte_count_times_[0] = now;
}
// Called from pacer when we can send the packet.
bool RTPSender::TimeToSendPacket(uint16_t sequence_number,
int64_t capture_time_ms,
bool retransmission) {
size_t length = IP_PACKET_SIZE;
uint8_t data_buffer[IP_PACKET_SIZE];
int64_t stored_time_ms;
if (!packet_history_.GetPacketAndSetSendTime(sequence_number,
0,
retransmission,
data_buffer,
&length,
&stored_time_ms)) {
// Packet cannot be found. Allow sending to continue.
return true;
}
int rtx;
{
rtc::CritScope lock(&send_critsect_);
rtx = rtx_;
}
return PrepareAndSendPacket(data_buffer,
length,
capture_time_ms,
retransmission && (rtx & kRtxRetransmitted) > 0,
retransmission);
}
bool RTPSender::PrepareAndSendPacket(uint8_t* buffer,
size_t length,
int64_t capture_time_ms,
bool send_over_rtx,
bool is_retransmit) {
uint8_t* buffer_to_send_ptr = buffer;
RtpUtility::RtpHeaderParser rtp_parser(buffer, length);
RTPHeader rtp_header;
rtp_parser.Parse(&rtp_header);
if (!is_retransmit && rtp_header.markerBit) {
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", rtp_header.timestamp, "seqnum", rtp_header.sequenceNumber);
uint8_t data_buffer_rtx[IP_PACKET_SIZE];
if (send_over_rtx) {
BuildRtxPacket(buffer, &length, data_buffer_rtx);
buffer_to_send_ptr = data_buffer_rtx;
}
int64_t now_ms = clock_->TimeInMilliseconds();
int64_t diff_ms = now_ms - capture_time_ms;
UpdateTransmissionTimeOffset(buffer_to_send_ptr, length, rtp_header,
diff_ms);
UpdateAbsoluteSendTime(buffer_to_send_ptr, length, rtp_header, now_ms);
PacketOptions options;
if (AllocateTransportSequenceNumber(&options.packet_id)) {
if (UpdateTransportSequenceNumber(options.packet_id, buffer_to_send_ptr,
length, rtp_header)) {
if (transport_feedback_observer_)
transport_feedback_observer_->AddPacket(options.packet_id, length,
true);
}
}
if (!is_retransmit && !send_over_rtx) {
UpdateDelayStatistics(capture_time_ms, now_ms);
UpdateOnSendPacket(options.packet_id, capture_time_ms, rtp_header.ssrc);
}
bool ret = SendPacketToNetwork(buffer_to_send_ptr, length, options);
if (ret) {
rtc::CritScope lock(&send_critsect_);
media_has_been_sent_ = true;
}
UpdateRtpStats(buffer_to_send_ptr, length, rtp_header, send_over_rtx,
is_retransmit);
return ret;
}
void RTPSender::UpdateRtpStats(const uint8_t* buffer,
size_t packet_length,
const RTPHeader& header,
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();
rtc::CritScope lock(&statistics_crit_);
if (is_rtx) {
counters = &rtx_rtp_stats_;
} else {
counters = &rtp_stats_;
}
total_bitrate_sent_.Update(packet_length);
if (counters->first_packet_time_ms == -1) {
counters->first_packet_time_ms = clock_->TimeInMilliseconds();
}
if (IsFecPacket(buffer, header)) {
counters->fec.AddPacket(packet_length, header);
}
if (is_retransmit) {
counters->retransmitted.AddPacket(packet_length, header);
}
counters->transmitted.AddPacket(packet_length, header);
if (rtp_stats_callback_) {
rtp_stats_callback_->DataCountersUpdated(*counters, ssrc);
}
}
bool RTPSender::IsFecPacket(const uint8_t* buffer,
const RTPHeader& header) 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 &&
header.payloadType == pt_red &&
buffer[header.headerLength] == pt_fec;
}
size_t RTPSender::TimeToSendPadding(size_t bytes) {
if (audio_configured_ || bytes == 0)
return 0;
size_t bytes_sent = TrySendRedundantPayloads(bytes);
if (bytes_sent < bytes)
bytes_sent += SendPadData(bytes - bytes_sent, false, 0, 0);
return bytes_sent;
}
// TODO(pwestin): send in the RtpHeaderParser to avoid parsing it again.
int32_t 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;
RtpUtility::RtpHeaderParser rtp_parser(buffer, length);
RTPHeader rtp_header;
rtp_parser.Parse(&rtp_header);
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 (capture_time_ms > 0) {
UpdateTransmissionTimeOffset(buffer, length, rtp_header,
now_ms - capture_time_ms);
}
UpdateAbsoluteSendTime(buffer, length, rtp_header, now_ms);
// Used for NACK and to spread out the transmission of packets.
if (packet_history_.PutRTPPacket(buffer, length, capture_time_ms, storage) !=
0) {
return -1;
}
if (paced_sender_) {
// Correct offset between implementations of millisecond time stamps in
// TickTime and Clock.
int64_t corrected_time_ms = capture_time_ms + clock_delta_ms_;
paced_sender_->InsertPacket(priority, rtp_header.ssrc,
rtp_header.sequenceNumber, 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 0;
}
PacketOptions options;
if (AllocateTransportSequenceNumber(&options.packet_id)) {
if (UpdateTransportSequenceNumber(options.packet_id, buffer, length,
rtp_header)) {
if (transport_feedback_observer_)
transport_feedback_observer_->AddPacket(options.packet_id, length,
true);
}
}
UpdateDelayStatistics(capture_time_ms, now_ms);
UpdateOnSendPacket(options.packet_id, capture_time_ms, rtp_header.ssrc);
bool sent = SendPacketToNetwork(buffer, length, options);
// 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_.SetSent(rtp_header.sequenceNumber);
if (!sent)
return -1;
{
rtc::CritScope lock(&send_critsect_);
media_has_been_sent_ = true;
}
UpdateRtpStats(buffer, length, rtp_header, false, false);
return 0;
}
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() {
rtc::CritScope lock(&send_critsect_);
total_bitrate_sent_.Process();
nack_bitrate_.Process();
if (audio_configured_) {
return;
}
video_->ProcessBitrate();
}
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 += RtpHeaderExtensionTotalLength();
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;
}
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) {
assert(payload_type >= 0);
rtc::CritScope lock(&send_critsect_);
if (timestamp_provided) {
timestamp_ = start_timestamp_ + capture_timestamp;
} else {
// Make a unique time stamp.
// We can't inc by the actual time, since then we increase the risk of back
// timing.
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,
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;
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;
}
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;
}
void RTPSender::UpdateTransmissionTimeOffset(uint8_t* rtp_packet,
size_t rtp_packet_length,
const RTPHeader& rtp_header,
int64_t time_diff_ms) const {
size_t offset;
rtc::CritScope lock(&send_critsect_);
switch (VerifyExtension(kRtpExtensionTransmissionTimeOffset, rtp_packet,
rtp_packet_length, rtp_header,
kTransmissionTimeOffsetLength, &offset)) {
case ExtensionStatus::kNotRegistered:
return;
case ExtensionStatus::kError:
LOG(LS_WARNING) << "Failed to update transmission time offset.";
return;
case ExtensionStatus::kOk:
break;
default:
RTC_NOTREACHED();
}
// Update transmission offset field (converting to a 90 kHz timestamp).
ByteWriter<int32_t, 3>::WriteBigEndian(rtp_packet + offset + 1,
time_diff_ms * 90); // RTP timestamp.
}
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;
}
void RTPSender::UpdateAbsoluteSendTime(uint8_t* rtp_packet,
size_t rtp_packet_length,
const RTPHeader& rtp_header,
int64_t now_ms) const {
size_t offset;
rtc::CritScope lock(&send_critsect_);
switch (VerifyExtension(kRtpExtensionAbsoluteSendTime, rtp_packet,
rtp_packet_length, rtp_header,
kAbsoluteSendTimeLength, &offset)) {
case ExtensionStatus::kNotRegistered:
return;
case ExtensionStatus::kError:
LOG(LS_WARNING) << "Failed to update absolute send time";
return;
case ExtensionStatus::kOk:
break;
default:
RTC_NOTREACHED();
}
// Update absolute send time field (convert ms to 24-bit unsigned with 18 bit
// fractional part).
ByteWriter<uint32_t, 3>::WriteBigEndian(rtp_packet + offset + 1,
ConvertMsTo24Bits(now_ms));
}
bool RTPSender::UpdateTransportSequenceNumber(
uint16_t sequence_number,
uint8_t* rtp_packet,
size_t rtp_packet_length,
const RTPHeader& rtp_header) const {
size_t offset;
rtc::CritScope lock(&send_critsect_);
switch (VerifyExtension(kRtpExtensionTransportSequenceNumber, rtp_packet,
rtp_packet_length, rtp_header,
kTransportSequenceNumberLength, &offset)) {
case ExtensionStatus::kNotRegistered:
return false;
case ExtensionStatus::kError:
LOG(LS_WARNING) << "Failed to update transport sequence number";
return false;
case ExtensionStatus::kOk:
break;
default:
RTC_NOTREACHED();
}
BuildTransportSequenceNumberExtension(rtp_packet + offset, sequence_number);
return true;
}
bool RTPSender::AllocateTransportSequenceNumber(int* packet_id) const {
if (!transport_sequence_number_allocator_)
return false;
*packet_id = transport_sequence_number_allocator_->AllocateSequenceNumber();
return true;
}
void RTPSender::SetSendingStatus(bool enabled) {
if (enabled) {
uint32_t frequency_hz = SendPayloadFrequency();
uint32_t RTPtime = CurrentRtp(*clock_, frequency_hz);
// Will be ignored if it's already configured via API.
SetStartTimestamp(RTPtime, false);
} else {
rtc::CritScope lock(&send_critsect_);
if (!ssrc_forced_) {
// Generate a new SSRC.
ssrc_db_->ReturnSSRC(ssrc_);
ssrc_ = ssrc_db_->CreateSSRC();
RTC_DCHECK(ssrc_ != 0);
bitrates_.set_ssrc(ssrc_);
}
// 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_;
}
uint32_t RTPSender::Timestamp() const {
rtc::CritScope lock(&send_critsect_);
return timestamp_;
}
void RTPSender::SetStartTimestamp(uint32_t timestamp, bool force) {
rtc::CritScope lock(&send_critsect_);
if (force) {
start_timestamp_forced_ = true;
start_timestamp_ = timestamp;
} else {
if (!start_timestamp_forced_) {
start_timestamp_ = timestamp;
}
}
}
uint32_t RTPSender::StartTimestamp() const {
rtc::CritScope lock(&send_critsect_);
return start_timestamp_;
}
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);
bitrates_.set_ssrc(ssrc_);
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;
bitrates_.set_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);
}
int32_t RTPSender::SetRED(int8_t payload_type) {
if (!audio_configured_) {
return -1;
}
return audio_->SetRED(payload_type);
}
int32_t RTPSender::RED(int8_t *payload_type) const {
if (!audio_configured_) {
return -1;
}
return audio_->RED(payload_type);
}
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;
}
void RTPSender::BuildRtxPacket(uint8_t* buffer, size_t* length,
uint8_t* buffer_rtx) {
rtc::CritScope lock(&send_critsect_);
uint8_t* data_buffer_rtx = buffer_rtx;
// Add RTX header.
RtpUtility::RtpHeaderParser rtp_parser(
reinterpret_cast<const uint8_t*>(buffer), *length);
RTPHeader rtp_header;
rtp_parser.Parse(&rtp_header);
// Add original RTP header.
memcpy(data_buffer_rtx, buffer, rtp_header.headerLength);
// Replace payload type, if a specific type is set for RTX.
auto kv = rtx_payload_type_map_.find(rtp_header.payloadType);
// Use rtx mapping associated with media codec if we can't find one, assuming
// 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())
data_buffer_rtx[1] = kv->second;
if (rtp_header.markerBit)
data_buffer_rtx[1] |= kRtpMarkerBitMask;
// Replace sequence number.
uint8_t* ptr = data_buffer_rtx + 2;
ByteWriter<uint16_t>::WriteBigEndian(ptr, sequence_number_rtx_++);
// Replace SSRC.
ptr += 6;
ByteWriter<uint32_t>::WriteBigEndian(ptr, ssrc_rtx_);
// Add OSN (original sequence number).
ptr = data_buffer_rtx + rtp_header.headerLength;
ByteWriter<uint16_t>::WriteBigEndian(ptr, rtp_header.sequenceNumber);
ptr += 2;
// Add original payload data.
memcpy(ptr, buffer + rtp_header.headerLength,
*length - rtp_header.headerLength);
*length += 2;
}
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 {
return total_bitrate_sent_.BitrateLast();
}
void RTPSender::SetRtpState(const RtpState& rtp_state) {
rtc::CritScope lock(&send_critsect_);
sequence_number_ = rtp_state.sequence_number;
sequence_number_forced_ = true;
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 = start_timestamp_;
state.timestamp = 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 = start_timestamp_;
return state;
}
} // namespace webrtc