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586b19bdb615dde34cdcf107272d8857fe2f5631
platform-external-webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
Stefan Holmer 586b19bdb6 Enable probing with repeated payload packets by default. 
To make this possible padding only packets will have the same timestamp
as the previously sent media packet, as long as RTX is not enabled. This
has the side effect that if we send only padding for a long time without
sending media, a receive-side jitter buffer could potentially overflow.

In practice this shouldn't be an issue, partly because RTX is recommended and
used by default, but also because padding typically is terminated before being
received by a client. It is also not an issue for bandwidth estimation as long
as abs-send-time is used instead of toffset.

BUG=chromium:425925
R=mflodman@webrtc.org, sprang@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1327933003 .

Cr-Commit-Position: refs/heads/master@{#9984}
2015-09-18 09:14:42 +00:00

1905 lines
63 KiB
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/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
#include <stdlib.h> // srand
#include <utility>
#include "webrtc/base/checks.h"
#include "webrtc/modules/rtp_rtcp/interface/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/system_wrappers/interface/critical_section_wrapper.h"
#include "webrtc/system_wrappers/interface/logging.h"
#include "webrtc/system_wrappers/interface/tick_util.h"
#include "webrtc/system_wrappers/interface/trace_event.h"
namespace webrtc {
// Max in the RFC 3550 is 255 bytes, we limit it to be modulus 32 for SRTP.
const size_t kMaxPaddingLength = 224;
const int kSendSideDelayWindowMs = 1000;
namespace {
const size_t kRtpHeaderLength = 12;
const char* FrameTypeToString(FrameType frame_type) {
switch (frame_type) {
case kFrameEmpty: return "empty";
case kAudioFrameSpeech: return "audio_speech";
case kAudioFrameCN: return "audio_cn";
case kVideoFrameKey: return "video_key";
case kVideoFrameDelta: return "video_delta";
}
return "";
}
} // namespace
class BitrateAggregator {
public:
explicit BitrateAggregator(BitrateStatisticsObserver* bitrate_callback)
: callback_(bitrate_callback),
total_bitrate_observer_(*this),
retransmit_bitrate_observer_(*this),
ssrc_(0) {}
void OnStatsUpdated() const {
if (callback_)
callback_->Notify(total_bitrate_observer_.statistics(),
retransmit_bitrate_observer_.statistics(),
ssrc_);
}
Bitrate::Observer* total_bitrate_observer() {
return &total_bitrate_observer_;
}
Bitrate::Observer* retransmit_bitrate_observer() {
return &retransmit_bitrate_observer_;
}
void set_ssrc(uint32_t ssrc) { ssrc_ = ssrc; }
private:
// We assume that these observers are called on the same thread, which is
// true for RtpSender as they are called on the Process thread.
class BitrateObserver : public Bitrate::Observer {
public:
explicit BitrateObserver(const BitrateAggregator& aggregator)
: aggregator_(aggregator) {}
// Implements Bitrate::Observer.
void BitrateUpdated(const BitrateStatistics& stats) override {
statistics_ = stats;
aggregator_.OnStatsUpdated();
}
BitrateStatistics statistics() const { return statistics_; }
private:
BitrateStatistics statistics_;
const BitrateAggregator& aggregator_;
};
BitrateStatisticsObserver* const callback_;
BitrateObserver total_bitrate_observer_;
BitrateObserver retransmit_bitrate_observer_;
uint32_t ssrc_;
};
RTPSender::RTPSender(bool audio,
Clock* clock,
Transport* transport,
RtpAudioFeedback* audio_feedback,
PacedSender* paced_sender,
PacketRouter* packet_router,
TransportFeedbackObserver* transport_feedback_observer,
BitrateStatisticsObserver* bitrate_callback,
FrameCountObserver* frame_count_observer,
SendSideDelayObserver* send_side_delay_observer)
: clock_(clock),
// TODO(holmer): Remove this conversion when we remove the use of
// TickTime.
clock_delta_ms_(clock_->TimeInMilliseconds() -
TickTime::MillisecondTimestamp()),
bitrates_(new BitrateAggregator(bitrate_callback)),
total_bitrate_sent_(clock, bitrates_->total_bitrate_observer()),
audio_configured_(audio),
audio_(audio ? new RTPSenderAudio(clock, this, audio_feedback) : nullptr),
video_(audio ? nullptr : new RTPSenderVideo(clock, this)),
paced_sender_(paced_sender),
packet_router_(packet_router),
transport_feedback_observer_(transport_feedback_observer),
last_capture_time_ms_sent_(0),
send_critsect_(CriticalSectionWrapper::CreateCriticalSection()),
transport_(transport),
sending_media_(true), // Default to sending media.
max_payload_length_(IP_PACKET_SIZE - 28), // Default is IP-v4/UDP.
packet_over_head_(28),
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
statistics_crit_(CriticalSectionWrapper::CreateCriticalSection()),
rtp_stats_callback_(NULL),
frame_count_observer_(frame_count_observer),
send_side_delay_observer_(send_side_delay_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),
rtx_payload_type_(-1),
target_bitrate_critsect_(CriticalSectionWrapper::CreateCriticalSection()),
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.
srand(static_cast<uint32_t>(clock_->TimeInMilliseconds()));
ssrc_ = ssrc_db_.CreateSSRC(); // Can't be 0.
ssrc_rtx_ = ssrc_db_.CreateSSRC(); // Can't be 0.
bitrates_->set_ssrc(ssrc_);
// Random start, 16 bits. Can't be 0.
sequence_number_rtx_ = static_cast<uint16_t>(rand() + 1) & 0x7FFF;
sequence_number_ = static_cast<uint16_t>(rand() + 1) & 0x7FFF;
}
RTPSender::~RTPSender() {
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) {
CriticalSectionScoped cs(target_bitrate_critsect_.get());
target_bitrate_ = bitrate;
}
uint32_t RTPSender::GetTargetBitrate() {
CriticalSectionScoped cs(target_bitrate_critsect_.get());
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;
}
CriticalSectionScoped cs(send_critsect_.get());
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;
}
CriticalSectionScoped cs(send_critsect_.get());
absolute_send_time_ = absolute_send_time;
return 0;
}
void RTPSender::SetVideoRotation(VideoRotation rotation) {
CriticalSectionScoped cs(send_critsect_.get());
rotation_ = rotation;
}
int32_t RTPSender::SetTransportSequenceNumber(uint16_t sequence_number) {
CriticalSectionScoped cs(send_critsect_.get());
transport_sequence_number_ = sequence_number;
return 0;
}
int32_t RTPSender::RegisterRtpHeaderExtension(RTPExtensionType type,
uint8_t id) {
CriticalSectionScoped cs(send_critsect_.get());
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) {
CriticalSectionScoped cs(send_critsect_.get());
return rtp_header_extension_map_.IsRegistered(type);
}
int32_t RTPSender::DeregisterRtpHeaderExtension(RTPExtensionType type) {
CriticalSectionScoped cs(send_critsect_.get());
return rtp_header_extension_map_.Deregister(type);
}
size_t RTPSender::RtpHeaderExtensionTotalLength() const {
CriticalSectionScoped cs(send_critsect_.get());
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,
uint8_t channels,
uint32_t rate) {
assert(payload_name);
CriticalSectionScoped cs(send_critsect_.get());
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 = NULL;
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, rate);
}
if (payload) {
payload_type_map_[payload_number] = payload;
}
return ret_val;
}
int32_t RTPSender::DeRegisterSendPayload(int8_t payload_type) {
CriticalSectionScoped lock(send_critsect_.get());
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) {
CriticalSectionScoped cs(send_critsect_.get());
payload_type_ = payload_type;
}
int8_t RTPSender::SendPayloadType() const {
CriticalSectionScoped cs(send_critsect_.get());
return payload_type_;
}
int RTPSender::SendPayloadFrequency() const {
return audio_ != NULL ? audio_->AudioFrequency() : kVideoPayloadTypeFrequency;
}
int32_t RTPSender::SetMaxPayloadLength(size_t max_payload_length,
uint16_t packet_over_head) {
// Sanity check.
RTC_DCHECK(max_payload_length >= 100 && max_payload_length <= IP_PACKET_SIZE)
<< "Invalid max payload length: " << max_payload_length;
CriticalSectionScoped cs(send_critsect_.get());
max_payload_length_ = max_payload_length;
packet_over_head_ = packet_over_head;
return 0;
}
size_t RTPSender::MaxDataPayloadLength() const {
int rtx;
{
CriticalSectionScoped rtx_lock(send_critsect_.get());
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_;
}
uint16_t RTPSender::PacketOverHead() const { return packet_over_head_; }
void RTPSender::SetRtxStatus(int mode) {
CriticalSectionScoped cs(send_critsect_.get());
rtx_ = mode;
}
int RTPSender::RtxStatus() const {
CriticalSectionScoped cs(send_critsect_.get());
return rtx_;
}
void RTPSender::SetRtxSsrc(uint32_t ssrc) {
CriticalSectionScoped cs(send_critsect_.get());
ssrc_rtx_ = ssrc;
}
uint32_t RTPSender::RtxSsrc() const {
CriticalSectionScoped cs(send_critsect_.get());
return ssrc_rtx_;
}
void RTPSender::SetRtxPayloadType(int payload_type,
int associated_payload_type) {
CriticalSectionScoped cs(send_critsect_.get());
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;
rtx_payload_type_ = payload_type;
}
std::pair<int, int> RTPSender::RtxPayloadType() const {
CriticalSectionScoped cs(send_critsect_.get());
for (const auto& kv : rtx_payload_type_map_) {
if (kv.second == rtx_payload_type_) {
return std::make_pair(rtx_payload_type_, kv.first);
}
}
return std::make_pair(-1, -1);
}
int32_t RTPSender::CheckPayloadType(int8_t payload_type,
RtpVideoCodecTypes* video_type) {
CriticalSectionScoped cs(send_critsect_.get());
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 " << 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;
video_->SetMaxConfiguredBitrateVideo(payload->typeSpecific.Video.maxRate);
}
return 0;
}
RTPSenderInterface::CVOMode RTPSender::ActivateCVORtpHeaderExtension() {
if (cvo_mode_ == kCVOInactive) {
CriticalSectionScoped cs(send_critsect_.get());
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.
CriticalSectionScoped cs(send_critsect_.get());
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.";
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 == kFrameEmpty);
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 == kFrameEmpty)
return 0;
ret_val =
video_->SendVideo(video_type, frame_type, payload_type,
capture_timestamp, capture_time_ms, payload_data,
payload_size, fragmentation, rtp_hdr);
}
CriticalSectionScoped cs(statistics_crit_.get());
// 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) {
{
CriticalSectionScoped cs(send_critsect_.get());
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 PacedSender,
// 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) &&
packet_router_;
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;
{
CriticalSectionScoped cs(send_critsect_.get());
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 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))
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_;
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);
uint16_t transport_seq = 0;
if (using_transport_seq) {
transport_seq =
UpdateTransportSequenceNumber(padding_packet, length, rtp_header);
}
if (!SendPacketToNetwork(padding_packet, length))
break;
if (using_transport_seq && transport_feedback_observer_) {
transport_feedback_observer_->OnPacketSent(
PacketInfo(0, now_ms, transport_seq, length, true));
}
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_;
if (!paced_sender_->SendPacket(
PacedSender::kHighPriority, header.ssrc, header.sequenceNumber,
corrected_capture_tims_ms, length - header.headerLength, true)) {
// We can't send the packet right now.
// We will be called when it is time.
return length;
}
}
int rtx = kRtxOff;
{
CriticalSectionScoped lock(send_critsect_.get());
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) {
int bytes_sent = -1;
if (transport_) {
bytes_sent = transport_->SendPacket(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();
CriticalSectionScoped cs(send_critsect_.get());
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) {
CriticalSectionScoped cs(send_critsect_.get());
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;
}
if (!retransmission && capture_time_ms > 0) {
UpdateDelayStatistics(capture_time_ms, clock_->TimeInMilliseconds());
}
int rtx;
{
CriticalSectionScoped lock(send_critsect_.get());
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);
uint16_t transport_seq = 0;
// TODO(sprang): Potentially too much overhead in IsRegistered()?
bool using_transport_seq = rtp_header_extension_map_.IsRegistered(
kRtpExtensionTransportSequenceNumber) &&
packet_router_ && !is_retransmit;
if (using_transport_seq) {
transport_seq =
UpdateTransportSequenceNumber(buffer_to_send_ptr, length, rtp_header);
}
bool ret = SendPacketToNetwork(buffer_to_send_ptr, length);
if (ret) {
CriticalSectionScoped lock(send_critsect_.get());
media_has_been_sent_ = true;
}
if (using_transport_seq && transport_feedback_observer_) {
transport_feedback_observer_->OnPacketSent(
PacketInfo(0, now_ms, transport_seq, length, 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();
CriticalSectionScoped lock(statistics_crit_.get());
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 (bytes == 0)
return 0;
{
CriticalSectionScoped cs(send_critsect_.get());
if (!sending_media_)
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,
PacedSender::Priority priority) {
RtpUtility::RtpHeaderParser rtp_parser(buffer,
payload_length + rtp_header_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, payload_length + rtp_header_length,
rtp_header, now_ms - capture_time_ms);
}
UpdateAbsoluteSendTime(buffer, payload_length + rtp_header_length,
rtp_header, now_ms);
// Used for NACK and to spread out the transmission of packets.
if (packet_history_.PutRTPPacket(buffer, rtp_header_length + payload_length,
capture_time_ms, storage) != 0) {
return -1;
}
if (paced_sender_ && storage != kDontStore) {
// Correct offset between implementations of millisecond time stamps in
// TickTime and Clock.
int64_t corrected_time_ms = capture_time_ms + clock_delta_ms_;
if (!paced_sender_->SendPacket(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);
}
// We can't send the packet right now.
// We will be called when it is time.
return 0;
}
}
if (capture_time_ms > 0) {
UpdateDelayStatistics(capture_time_ms, now_ms);
}
size_t length = payload_length + rtp_header_length;
bool sent = SendPacketToNetwork(buffer, length);
if (storage != kDontStore) {
// 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;
{
CriticalSectionScoped lock(send_critsect_.get());
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_)
return;
uint32_t ssrc;
int avg_delay_ms = 0;
int max_delay_ms = 0;
{
CriticalSectionScoped lock(send_critsect_.get());
ssrc = ssrc_;
}
{
CriticalSectionScoped cs(statistics_crit_.get());
// 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::ProcessBitrate() {
CriticalSectionScoped cs(send_critsect_.get());
total_bitrate_sent_.Process();
nack_bitrate_.Process();
if (audio_configured_) {
return;
}
video_->ProcessBitrate();
}
size_t RTPSender::RTPHeaderLength() const {
CriticalSectionScoped lock(send_critsect_.get());
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) {
CriticalSectionScoped cs(send_critsect_.get());
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 {
CriticalSectionScoped lock(statistics_crit_.get());
*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);
CriticalSectionScoped cs(send_critsect_.get());
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 block_pos =
kRtpHeaderLength + rtp_header.numCSRCs + 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[kRtpHeaderLength + rtp_header.numCSRCs] == 0xBE) &&
(rtp_packet[kRtpHeaderLength + rtp_header.numCSRCs + 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 that header contains extension.
if (!((rtp_packet[kRtpHeaderLength + rtp_header.numCSRCs] == 0xBE) &&
(rtp_packet[kRtpHeaderLength + rtp_header.numCSRCs + 1] == 0xDE))) {
LOG(LS_WARNING)
<< "Failed to update absolute send time, hdr extension not found.";
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;
CriticalSectionScoped cs(send_critsect_.get());
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;
CriticalSectionScoped cs(send_critsect_.get());
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;
CriticalSectionScoped cs(send_critsect_.get());
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;
CriticalSectionScoped cs(send_critsect_.get());
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,
((now_ms << 18) / 1000) & 0x00ffffff);
}
uint16_t RTPSender::UpdateTransportSequenceNumber(
uint8_t* rtp_packet,
size_t rtp_packet_length,
const RTPHeader& rtp_header) const {
size_t offset;
CriticalSectionScoped cs(send_critsect_.get());
switch (VerifyExtension(kRtpExtensionTransportSequenceNumber, rtp_packet,
rtp_packet_length, rtp_header,
kTransportSequenceNumberLength, &offset)) {
case ExtensionStatus::kNotRegistered:
return 0;
case ExtensionStatus::kError:
LOG(LS_WARNING) << "Failed to update transport sequence number";
return 0;
case ExtensionStatus::kOk:
break;
default:
RTC_NOTREACHED();
}
uint16_t seq = packet_router_->AllocateSequenceNumber();
BuildTransportSequenceNumberExtension(rtp_packet + offset, seq);
return seq;
}
void RTPSender::SetSendingStatus(bool enabled) {
if (enabled) {
uint32_t frequency_hz = SendPayloadFrequency();
uint32_t RTPtime = RtpUtility::GetCurrentRTP(clock_, frequency_hz);
// Will be ignored if it's already configured via API.
SetStartTimestamp(RTPtime, false);
} else {
CriticalSectionScoped lock(send_critsect_.get());
if (!ssrc_forced_) {
// Generate a new SSRC.
ssrc_db_.ReturnSSRC(ssrc_);
ssrc_ = ssrc_db_.CreateSSRC(); // Can't be 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_ =
rand() / (RAND_MAX / MAX_INIT_RTP_SEQ_NUMBER); // NOLINT
}
}
}
void RTPSender::SetSendingMediaStatus(bool enabled) {
CriticalSectionScoped cs(send_critsect_.get());
sending_media_ = enabled;
}
bool RTPSender::SendingMedia() const {
CriticalSectionScoped cs(send_critsect_.get());
return sending_media_;
}
uint32_t RTPSender::Timestamp() const {
CriticalSectionScoped cs(send_critsect_.get());
return timestamp_;
}
void RTPSender::SetStartTimestamp(uint32_t timestamp, bool force) {
CriticalSectionScoped cs(send_critsect_.get());
if (force) {
start_timestamp_forced_ = true;
start_timestamp_ = timestamp;
} else {
if (!start_timestamp_forced_) {
start_timestamp_ = timestamp;
}
}
}
uint32_t RTPSender::StartTimestamp() const {
CriticalSectionScoped cs(send_critsect_.get());
return start_timestamp_;
}
uint32_t RTPSender::GenerateNewSSRC() {
// If configured via API, return 0.
CriticalSectionScoped cs(send_critsect_.get());
if (ssrc_forced_) {
return 0;
}
ssrc_ = ssrc_db_.CreateSSRC(); // Can't be 0.
bitrates_->set_ssrc(ssrc_);
return ssrc_;
}
void RTPSender::SetSSRC(uint32_t ssrc) {
// This is configured via the API.
CriticalSectionScoped cs(send_critsect_.get());
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_ =
rand() / (RAND_MAX / MAX_INIT_RTP_SEQ_NUMBER); // NOLINT
}
}
uint32_t RTPSender::SSRC() const {
CriticalSectionScoped cs(send_critsect_.get());
return ssrc_;
}
void RTPSender::SetCsrcs(const std::vector<uint32_t>& csrcs) {
assert(csrcs.size() <= kRtpCsrcSize);
CriticalSectionScoped cs(send_critsect_.get());
csrcs_ = csrcs;
}
void RTPSender::SetSequenceNumber(uint16_t seq) {
CriticalSectionScoped cs(send_critsect_.get());
sequence_number_forced_ = true;
sequence_number_ = seq;
}
uint16_t RTPSender::SequenceNumber() const {
CriticalSectionScoped cs(send_critsect_.get());
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();
}
uint32_t RTPSender::MaxConfiguredBitrateVideo() const {
if (audio_configured_) {
return 0;
}
return video_->MaxConfiguredBitrateVideo();
}
int32_t RTPSender::SendRTPIntraRequest() {
if (audio_configured_) {
return -1;
}
return video_->SendRTPIntraRequest();
}
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) {
CriticalSectionScoped cs(send_critsect_.get());
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.
if (rtx_payload_type_ != -1) {
data_buffer_rtx[1] = static_cast<uint8_t>(rtx_payload_type_);
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) {
CriticalSectionScoped cs(statistics_crit_.get());
rtp_stats_callback_ = callback;
}
StreamDataCountersCallback* RTPSender::GetRtpStatisticsCallback() const {
CriticalSectionScoped cs(statistics_crit_.get());
return rtp_stats_callback_;
}
uint32_t RTPSender::BitrateSent() const {
return total_bitrate_sent_.BitrateLast();
}
void RTPSender::SetRtpState(const RtpState& rtp_state) {
SetStartTimestamp(rtp_state.start_timestamp, true);
CriticalSectionScoped lock(send_critsect_.get());
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 {
CriticalSectionScoped lock(send_critsect_.get());
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) {
CriticalSectionScoped lock(send_critsect_.get());
sequence_number_rtx_ = rtp_state.sequence_number;
}
RtpState RTPSender::GetRtxRtpState() const {
CriticalSectionScoped lock(send_critsect_.get());
RtpState state;
state.sequence_number = sequence_number_rtx_;
state.start_timestamp = start_timestamp_;
return state;
}
} // namespace webrtc
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