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webrtc / src / 4591fbd09f9cb6e83433c49a12dd8524c2806502 / . / webrtc / modules / rtp_rtcp / source / rtp_sender.cc
blob: 38ac859fd749f4e6de4db3d974d058e33be461da [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 "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 char* FrameTypeToString(const 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.
virtual 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(const int32_t id,
const bool audio,
Clock* clock,
Transport* transport,
RtpAudioFeedback* audio_feedback,
PacedSender* paced_sender,
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()),
id_(id),
audio_configured_(audio),
audio_(NULL),
video_(NULL),
paced_sender_(paced_sender),
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),
// 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),
num_csrcs_(0),
csrcs_(),
include_csrcs_(true),
rtx_(kRtxOff),
payload_type_rtx_(-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_));
memset(csrcs_, 0, sizeof(csrcs_));
// 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;
if (audio) {
audio_ = new RTPSenderAudio(id, clock_, this);
audio_->RegisterAudioCallback(audio_feedback);
} else {
video_ = new RTPSenderVideo(clock_, this);
}
}
RTPSender::~RTPSender() {
if (remote_ssrc_ != 0) {
ssrc_db_.ReturnSSRC(remote_ssrc_);
}
ssrc_db_.ReturnSSRC(ssrc_);
SSRCDatabase::ReturnSSRCDatabase();
delete send_critsect_;
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);
}
delete audio_;
delete video_;
}
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();
}
bool RTPSender::GetSendSideDelay(int* avg_send_delay_ms,
int* max_send_delay_ms) const {
CriticalSectionScoped lock(statistics_crit_.get());
SendDelayMap::const_iterator it = send_delays_.upper_bound(
clock_->TimeInMilliseconds() - kSendSideDelayWindowMs);
if (it == send_delays_.end())
return false;
int num_delays = 0;
for (; it != send_delays_.end(); ++it) {
*max_send_delay_ms = std::max(*max_send_delay_ms, it->second);
*avg_send_delay_ms += it->second;
++num_delays;
}
*avg_send_delay_ms = (*avg_send_delay_ms + num_delays / 2) / num_delays;
return true;
}
int32_t RTPSender::SetTransmissionTimeOffset(
const int32_t transmission_time_offset) {
if (transmission_time_offset > (0x800000 - 1) ||
transmission_time_offset < -(0x800000 - 1)) { // Word24.
return -1;
}
CriticalSectionScoped cs(send_critsect_);
transmission_time_offset_ = transmission_time_offset;
return 0;
}
int32_t RTPSender::SetAbsoluteSendTime(
const uint32_t absolute_send_time) {
if (absolute_send_time > 0xffffff) { // UWord24.
return -1;
}
CriticalSectionScoped cs(send_critsect_);
absolute_send_time_ = absolute_send_time;
return 0;
}
int32_t RTPSender::RegisterRtpHeaderExtension(const RTPExtensionType type,
const uint8_t id) {
CriticalSectionScoped cs(send_critsect_);
return rtp_header_extension_map_.Register(type, id);
}
int32_t RTPSender::DeregisterRtpHeaderExtension(
const RTPExtensionType type) {
CriticalSectionScoped cs(send_critsect_);
return rtp_header_extension_map_.Deregister(type);
}
size_t RTPSender::RtpHeaderExtensionTotalLength() const {
CriticalSectionScoped cs(send_critsect_);
return rtp_header_extension_map_.GetTotalLengthInBytes();
}
int32_t RTPSender::RegisterPayload(
const char payload_name[RTP_PAYLOAD_NAME_SIZE],
const int8_t payload_number, const uint32_t frequency,
const uint8_t channels, const uint32_t rate) {
assert(payload_name);
CriticalSectionScoped cs(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 = -1;
RtpUtility::Payload* payload = NULL;
if (audio_configured_) {
ret_val = audio_->RegisterAudioPayload(payload_name, payload_number,
frequency, channels, rate, payload);
} else {
ret_val = video_->RegisterVideoPayload(payload_name, payload_number, rate,
payload);
}
if (payload) {
payload_type_map_[payload_number] = payload;
}
return ret_val;
}
int32_t RTPSender::DeRegisterSendPayload(
const int8_t payload_type) {
CriticalSectionScoped 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) {
CriticalSectionScoped cs(send_critsect_);
payload_type_ = payload_type;
}
int8_t RTPSender::SendPayloadType() const {
CriticalSectionScoped cs(send_critsect_);
return payload_type_;
}
int RTPSender::SendPayloadFrequency() const {
return audio_ != NULL ? audio_->AudioFrequency() : kVideoPayloadTypeFrequency;
}
int32_t RTPSender::SetMaxPayloadLength(
const size_t max_payload_length,
const uint16_t packet_over_head) {
// Sanity check.
if (max_payload_length < 100 || max_payload_length > IP_PACKET_SIZE) {
LOG(LS_ERROR) << "Invalid max payload length: " << max_payload_length;
return -1;
}
CriticalSectionScoped cs(send_critsect_);
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_);
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_);
rtx_ = mode;
}
void RTPSender::SetRtxSsrc(uint32_t ssrc) {
CriticalSectionScoped cs(send_critsect_);
ssrc_rtx_ = ssrc;
}
uint32_t RTPSender::RtxSsrc() const {
CriticalSectionScoped cs(send_critsect_);
return ssrc_rtx_;
}
void RTPSender::RTXStatus(int* mode, uint32_t* ssrc,
int* payload_type) const {
CriticalSectionScoped cs(send_critsect_);
*mode = rtx_;
*ssrc = ssrc_rtx_;
*payload_type = payload_type_rtx_;
}
void RTPSender::SetRtxPayloadType(int payload_type) {
CriticalSectionScoped cs(send_critsect_);
payload_type_rtx_ = payload_type;
}
int32_t RTPSender::CheckPayloadType(const int8_t payload_type,
RtpVideoCodecTypes *video_type) {
CriticalSectionScoped cs(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 " << 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;
}
int32_t RTPSender::SendOutgoingData(
const FrameType frame_type, const int8_t payload_type,
const uint32_t capture_timestamp, int64_t capture_time_ms,
const uint8_t *payload_data, const size_t payload_size,
const RTPFragmentationHeader *fragmentation,
VideoCodecInformation *codec_info, const RTPVideoTypeHeader *rtp_type_hdr) {
uint32_t ssrc;
{
// Drop this packet if we're not sending media packets.
CriticalSectionScoped cs(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.";
return -1;
}
uint32_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, codec_info,
rtp_type_hdr);
}
CriticalSectionScoped cs(statistics_crit_.get());
uint32_t frame_count = ++frame_counts_[frame_type];
if (frame_count_observer_) {
frame_count_observer_->FrameCountUpdated(frame_type, frame_count, ssrc);
}
return ret_val;
}
size_t RTPSender::TrySendRedundantPayloads(size_t bytes_to_send) {
{
CriticalSectionScoped cs(send_critsect_);
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;
}
size_t RTPSender::BuildPaddingPacket(uint8_t* packet, size_t header_length) {
size_t padding_bytes_in_packet = kMaxPaddingLength;
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_bytes_in_packet >> 2); ++j) {
data[j] = rand(); // NOLINT
}
// Set number of padding bytes in the last byte of the packet.
packet[header_length + padding_bytes_in_packet - 1] =
static_cast<uint8_t>(padding_bytes_in_packet);
return padding_bytes_in_packet;
}
size_t RTPSender::TrySendPadData(size_t bytes) {
int64_t capture_time_ms;
uint32_t timestamp;
{
CriticalSectionScoped cs(send_critsect_);
timestamp = timestamp_;
capture_time_ms = capture_time_ms_;
if (last_timestamp_time_ms_ > 0) {
timestamp +=
(clock_->TimeInMilliseconds() - last_timestamp_time_ms_) * 90;
capture_time_ms +=
(clock_->TimeInMilliseconds() - last_timestamp_time_ms_);
}
}
return SendPadData(timestamp, capture_time_ms, bytes);
}
size_t RTPSender::SendPadData(uint32_t timestamp,
int64_t capture_time_ms,
size_t bytes) {
size_t padding_bytes_in_packet = 0;
size_t bytes_sent = 0;
for (; bytes > 0; bytes -= padding_bytes_in_packet) {
// Always send full padding packets.
if (bytes < kMaxPaddingLength)
bytes = kMaxPaddingLength;
uint32_t ssrc;
uint16_t sequence_number;
int payload_type;
bool over_rtx;
{
CriticalSectionScoped cs(send_critsect_);
// Only send padding packets following the last packet of a frame,
// indicated by the marker bit.
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;
ssrc = ssrc_rtx_;
sequence_number = sequence_number_rtx_;
++sequence_number_rtx_;
payload_type = ((rtx_ & kRtxRedundantPayloads) > 0) ? 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,
NULL,
0);
assert(header_length != static_cast<size_t>(-1));
padding_bytes_in_packet = BuildPaddingPacket(padding_packet, header_length);
assert(padding_bytes_in_packet <= bytes);
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);
if (!SendPacketToNetwork(padding_packet, length))
break;
bytes_sent += padding_bytes_in_packet;
UpdateRtpStats(padding_packet, length, rtp_header, over_rtx, false);
}
return bytes_sent;
}
void RTPSender::SetStorePacketsStatus(const bool enable,
const 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, uint32_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_);
rtx = rtx_;
}
return PrepareAndSendPacket(data_buffer, length, capture_time_ms,
(rtx & kRtxRetransmitted) > 0, true) ?
static_cast<int32_t>(length) : -1;
}
bool RTPSender::SendPacketToNetwork(const uint8_t *packet, size_t size) {
int bytes_sent = -1;
if (transport_) {
bytes_sent = transport_->SendPacket(id_, packet, size);
}
TRACE_EVENT_INSTANT2("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;
return video_->SetSelectiveRetransmissions(settings);
}
void RTPSender::OnReceivedNACK(
const std::list<uint16_t>& nack_sequence_numbers,
const uint16_t avg_rtt) {
TRACE_EVENT2("webrtc_rtp", "RTPSender::OnReceivedNACK",
"num_seqnum", nack_sequence_numbers.size(), "avg_rtt", avg_rtt);
const int64_t now = clock_->TimeInMilliseconds();
size_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 if (bytes_sent < 0) {
// 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) {
// TODO(pwestin) consolidate these two methods.
UpdateNACKBitRate(bytes_re_sent, now);
nack_bitrate_.Update(bytes_re_sent);
}
}
bool RTPSender::ProcessNACKBitRate(const 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_);
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(const size_t bytes,
const uint32_t now) {
CriticalSectionScoped cs(send_critsect_);
// Save bitrate statistics.
if (bytes > 0) {
if (now == 0) {
// Add padding length.
nack_byte_count_[0] += bytes;
} else {
if (nack_byte_count_times_[0] == 0) {
// First no shift.
} else {
// Shift.
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_);
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("webrtc_rtp", "PacedSend", capture_time_ms);
}
TRACE_EVENT_INSTANT2("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);
bool ret = SendPacketToNetwork(buffer_to_send_ptr, length);
if (ret) {
CriticalSectionScoped 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 size,
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(size);
++counters->packets;
if (IsFecPacket(buffer, header)) {
++counters->fec_packets;
}
if (is_retransmit) {
++counters->retransmitted_packets;
} else {
counters->bytes += size - (header.headerLength + header.paddingLength);
counters->header_bytes += header.headerLength;
counters->padding_bytes += header.paddingLength;
}
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) {
{
CriticalSectionScoped cs(send_critsect_);
if (!sending_media_) return 0;
}
if (bytes == 0)
return 0;
size_t bytes_sent = TrySendRedundantPayloads(bytes);
if (bytes_sent < bytes)
bytes_sent += TrySendPadData(bytes - bytes_sent);
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,
max_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("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;
if (!SendPacketToNetwork(buffer, length))
return -1;
{
CriticalSectionScoped 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) {
uint32_t ssrc;
int avg_delay_ms = 0;
int max_delay_ms = 0;
{
CriticalSectionScoped lock(send_critsect_);
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));
}
if (send_side_delay_observer_ &&
GetSendSideDelay(&avg_delay_ms, &max_delay_ms)) {
send_side_delay_observer_->SendSideDelayUpdated(avg_delay_ms,
max_delay_ms, ssrc);
}
}
void RTPSender::ProcessBitrate() {
CriticalSectionScoped cs(send_critsect_);
total_bitrate_sent_.Process();
nack_bitrate_.Process();
if (audio_configured_) {
return;
}
video_->ProcessBitrate();
}
size_t RTPSender::RTPHeaderLength() const {
CriticalSectionScoped lock(send_critsect_);
size_t rtp_header_length = 12;
if (include_csrcs_) {
rtp_header_length += sizeof(uint32_t) * num_csrcs_;
}
rtp_header_length += RtpHeaderExtensionTotalLength();
return rtp_header_length;
}
uint16_t RTPSender::IncrementSequenceNumber() {
CriticalSectionScoped cs(send_critsect_);
return sequence_number_++;
}
void RTPSender::ResetDataCounters() {
uint32_t ssrc;
uint32_t ssrc_rtx;
{
CriticalSectionScoped ssrc_lock(send_critsect_);
ssrc = ssrc_;
ssrc_rtx = ssrc_rtx_;
}
CriticalSectionScoped lock(statistics_crit_.get());
rtp_stats_ = StreamDataCounters();
rtx_rtp_stats_ = StreamDataCounters();
if (rtp_stats_callback_) {
rtp_stats_callback_->DataCountersUpdated(rtp_stats_, ssrc);
rtp_stats_callback_->DataCountersUpdated(rtx_rtp_stats_, ssrc_rtx);
}
}
void RTPSender::GetDataCounters(StreamDataCounters* rtp_stats,
StreamDataCounters* rtx_stats) const {
CriticalSectionScoped lock(statistics_crit_.get());
*rtp_stats = rtp_stats_;
*rtx_stats = rtx_rtp_stats_;
}
int RTPSender::CreateRTPHeader(
uint8_t* header, int8_t payload_type, uint32_t ssrc, bool marker_bit,
uint32_t timestamp, uint16_t sequence_number, const uint32_t* csrcs,
uint8_t num_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.
}
RtpUtility::AssignUWord16ToBuffer(header + 2, sequence_number);
RtpUtility::AssignUWord32ToBuffer(header + 4, timestamp);
RtpUtility::AssignUWord32ToBuffer(header + 8, ssrc);
int32_t rtp_header_length = 12;
// Add the CSRCs if any.
if (num_csrcs > 0) {
if (num_csrcs > kRtpCsrcSize) {
// error
assert(false);
return -1;
}
uint8_t *ptr = &header[rtp_header_length];
for (int i = 0; i < num_csrcs; ++i) {
RtpUtility::AssignUWord32ToBuffer(ptr, csrcs[i]);
ptr += 4;
}
header[0] = (header[0] & 0xf0) | num_csrcs;
// Update length of header.
rtp_header_length += sizeof(uint32_t) * num_csrcs;
}
uint16_t len = BuildRTPHeaderExtension(header + rtp_header_length);
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,
const int8_t payload_type,
const bool marker_bit,
const uint32_t capture_timestamp,
int64_t capture_time_ms,
const bool timestamp_provided,
const bool inc_sequence_number) {
assert(payload_type >= 0);
CriticalSectionScoped cs(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;
int csrcs_length = 0;
if (include_csrcs_)
csrcs_length = num_csrcs_;
return CreateRTPHeader(data_buffer, payload_type, ssrc_, marker_bit,
timestamp_, sequence_number, csrcs_, csrcs_length);
}
uint16_t RTPSender::BuildRTPHeaderExtension(uint8_t* data_buffer) 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).
RtpUtility::AssignUWord16ToBuffer(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;
switch (type) {
case kRtpExtensionTransmissionTimeOffset:
block_length = BuildTransmissionTimeOffsetExtension(
data_buffer + kHeaderLength + total_block_length);
break;
case kRtpExtensionAudioLevel:
block_length = BuildAudioLevelExtension(
data_buffer + kHeaderLength + total_block_length);
break;
case kRtpExtensionAbsoluteSendTime:
block_length = BuildAbsoluteSendTimeExtension(
data_buffer + kHeaderLength + total_block_length);
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;
}
// Set header length (in number of Word32, header excluded).
assert(total_block_length % 4 == 0);
RtpUtility::AssignUWord16ToBuffer(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;
RtpUtility::AssignUWord24ToBuffer(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 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=0 |V| level | 0x00 | 0x00 |
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
//
// Note that we always include 2 pad bytes, which will result in legal and
// correctly parsed RTP, but may be a bit wasteful if more short extensions
// are implemented. Right now the pad bytes would anyway be required at end
// of the extension block, so it makes no difference.
// 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.
data_buffer[pos++] = 0; // Padding.
data_buffer[pos++] = 0; // Padding.
// kAudioLevelLength is including pad bytes.
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;
RtpUtility::AssignUWord24ToBuffer(data_buffer + pos, absolute_send_time_);
pos += 3;
assert(pos == kAbsoluteSendTimeLength);
return kAbsoluteSendTimeLength;
}
void RTPSender::UpdateTransmissionTimeOffset(
uint8_t *rtp_packet, const size_t rtp_packet_length,
const RTPHeader &rtp_header, const int64_t time_diff_ms) const {
CriticalSectionScoped cs(send_critsect_);
// Get id.
uint8_t id = 0;
if (rtp_header_extension_map_.GetId(kRtpExtensionTransmissionTimeOffset,
&id) != 0) {
// Not registered.
return;
}
// Get length until start of header extension block.
int extension_block_pos =
rtp_header_extension_map_.GetLengthUntilBlockStartInBytes(
kRtpExtensionTransmissionTimeOffset);
if (extension_block_pos < 0) {
LOG(LS_WARNING)
<< "Failed to update transmission time offset, not registered.";
return;
}
size_t block_pos = 12 + rtp_header.numCSRCs + extension_block_pos;
if (rtp_packet_length < block_pos + kTransmissionTimeOffsetLength ||
rtp_header.headerLength <
block_pos + kTransmissionTimeOffsetLength) {
LOG(LS_WARNING)
<< "Failed to update transmission time offset, invalid length.";
return;
}
// Verify that header contains extension.
if (!((rtp_packet[12 + rtp_header.numCSRCs] == 0xBE) &&
(rtp_packet[12 + rtp_header.numCSRCs + 1] == 0xDE))) {
LOG(LS_WARNING) << "Failed to update transmission time offset, hdr "
"extension not found.";
return;
}
// Verify first byte in block.
const uint8_t first_block_byte = (id << 4) + 2;
if (rtp_packet[block_pos] != first_block_byte) {
LOG(LS_WARNING) << "Failed to update transmission time offset.";
return;
}
// Update transmission offset field (converting to a 90 kHz timestamp).
RtpUtility::AssignUWord24ToBuffer(rtp_packet + block_pos + 1,
time_diff_ms * 90); // RTP timestamp.
}
bool RTPSender::UpdateAudioLevel(uint8_t *rtp_packet,
const size_t rtp_packet_length,
const RTPHeader &rtp_header,
const bool is_voiced,
const uint8_t dBov) const {
CriticalSectionScoped cs(send_critsect_);
// Get id.
uint8_t id = 0;
if (rtp_header_extension_map_.GetId(kRtpExtensionAudioLevel, &id) != 0) {
// Not registered.
return false;
}
// Get length until start of header extension block.
int extension_block_pos =
rtp_header_extension_map_.GetLengthUntilBlockStartInBytes(
kRtpExtensionAudioLevel);
if (extension_block_pos < 0) {
// The feature is not enabled.
return false;
}
size_t block_pos = 12 + rtp_header.numCSRCs + extension_block_pos;
if (rtp_packet_length < block_pos + kAudioLevelLength ||
rtp_header.headerLength < block_pos + kAudioLevelLength) {
LOG(LS_WARNING) << "Failed to update audio level, invalid length.";
return false;
}
// Verify that header contains extension.
if (!((rtp_packet[12 + rtp_header.numCSRCs] == 0xBE) &&
(rtp_packet[12 + rtp_header.numCSRCs + 1] == 0xDE))) {
LOG(LS_WARNING) << "Failed to update audio level, hdr extension not found.";
return false;
}
// Verify first byte in block.
const uint8_t first_block_byte = (id << 4) + 0;
if (rtp_packet[block_pos] != first_block_byte) {
LOG(LS_WARNING) << "Failed to update audio level.";
return false;
}
rtp_packet[block_pos + 1] = (is_voiced ? 0x80 : 0x00) + (dBov & 0x7f);
return true;
}
void RTPSender::UpdateAbsoluteSendTime(
uint8_t *rtp_packet, const size_t rtp_packet_length,
const RTPHeader &rtp_header, const int64_t now_ms) const {
CriticalSectionScoped cs(send_critsect_);
// Get id.
uint8_t id = 0;
if (rtp_header_extension_map_.GetId(kRtpExtensionAbsoluteSendTime,
&id) != 0) {
// Not registered.
return;
}
// Get length until start of header extension block.
int extension_block_pos =
rtp_header_extension_map_.GetLengthUntilBlockStartInBytes(
kRtpExtensionAbsoluteSendTime);
if (extension_block_pos < 0) {
// The feature is not enabled.
return;
}
size_t block_pos = 12 + rtp_header.numCSRCs + extension_block_pos;
if (rtp_packet_length < block_pos + kAbsoluteSendTimeLength ||
rtp_header.headerLength < block_pos + kAbsoluteSendTimeLength) {
LOG(LS_WARNING) << "Failed to update absolute send time, invalid length.";
return;
}
// Verify that header contains extension.
if (!((rtp_packet[12 + rtp_header.numCSRCs] == 0xBE) &&
(rtp_packet[12 + rtp_header.numCSRCs + 1] == 0xDE))) {
LOG(LS_WARNING)
<< "Failed to update absolute send time, hdr extension not found.";
return;
}
// Verify first byte in block.
const uint8_t first_block_byte = (id << 4) + 2;
if (rtp_packet[block_pos] != first_block_byte) {
LOG(LS_WARNING) << "Failed to update absolute send time.";
return;
}
// Update absolute send time field (convert ms to 24-bit unsigned with 18 bit
// fractional part).
RtpUtility::AssignUWord24ToBuffer(rtp_packet + block_pos + 1,
((now_ms << 18) / 1000) & 0x00ffffff);
}
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_);
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(const bool enabled) {
CriticalSectionScoped cs(send_critsect_);
sending_media_ = enabled;
}
bool RTPSender::SendingMedia() const {
CriticalSectionScoped cs(send_critsect_);
return sending_media_;
}
uint32_t RTPSender::Timestamp() const {
CriticalSectionScoped cs(send_critsect_);
return timestamp_;
}
void RTPSender::SetStartTimestamp(uint32_t timestamp, bool force) {
CriticalSectionScoped cs(send_critsect_);
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_);
return start_timestamp_;
}
uint32_t RTPSender::GenerateNewSSRC() {
// If configured via API, return 0.
CriticalSectionScoped cs(send_critsect_);
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_);
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_);
return ssrc_;
}
void RTPSender::SetCSRCStatus(const bool include) {
CriticalSectionScoped lock(send_critsect_);
include_csrcs_ = include;
}
void RTPSender::SetCSRCs(const uint32_t arr_of_csrc[kRtpCsrcSize],
const uint8_t arr_length) {
assert(arr_length <= kRtpCsrcSize);
CriticalSectionScoped cs(send_critsect_);
for (int i = 0; i < arr_length; i++) {
csrcs_[i] = arr_of_csrc[i];
}
num_csrcs_ = arr_length;
}
int32_t RTPSender::CSRCs(uint32_t arr_of_csrc[kRtpCsrcSize]) const {
assert(arr_of_csrc);
CriticalSectionScoped cs(send_critsect_);
for (int i = 0; i < num_csrcs_ && i < kRtpCsrcSize; i++) {
arr_of_csrc[i] = csrcs_[i];
}
return num_csrcs_;
}
void RTPSender::SetSequenceNumber(uint16_t seq) {
CriticalSectionScoped cs(send_critsect_);
sequence_number_forced_ = true;
sequence_number_ = seq;
}
uint16_t RTPSender::SequenceNumber() const {
CriticalSectionScoped cs(send_critsect_);
return sequence_number_;
}
// Audio.
int32_t RTPSender::SendTelephoneEvent(const uint8_t key,
const uint16_t time_ms,
const uint8_t level) {
if (!audio_configured_) {
return -1;
}
return audio_->SendTelephoneEvent(key, time_ms, level);
}
bool RTPSender::SendTelephoneEventActive(int8_t *telephone_event) const {
if (!audio_configured_) {
return false;
}
return audio_->SendTelephoneEventActive(*telephone_event);
}
int32_t RTPSender::SetAudioPacketSize(
const uint16_t packet_size_samples) {
if (!audio_configured_) {
return -1;
}
return audio_->SetAudioPacketSize(packet_size_samples);
}
int32_t RTPSender::SetAudioLevel(const uint8_t level_d_bov) {
return audio_->SetAudioLevel(level_d_bov);
}
int32_t RTPSender::SetRED(const 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);
}
// Video
VideoCodecInformation *RTPSender::CodecInformationVideo() {
if (audio_configured_) {
return NULL;
}
return video_->CodecInformationVideo();
}
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();
}
int32_t RTPSender::SetGenericFECStatus(
const bool enable, const uint8_t payload_type_red,
const uint8_t payload_type_fec) {
if (audio_configured_) {
return -1;
}
return video_->SetGenericFECStatus(enable, payload_type_red,
payload_type_fec);
}
int32_t RTPSender::GenericFECStatus(
bool *enable, uint8_t *payload_type_red,
uint8_t *payload_type_fec) const {
if (audio_configured_) {
return -1;
}
return 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;
}
return video_->SetFecParameters(delta_params, key_params);
}
void RTPSender::BuildRtxPacket(uint8_t* buffer, size_t* length,
uint8_t* buffer_rtx) {
CriticalSectionScoped cs(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.
if (payload_type_rtx_ != -1) {
data_buffer_rtx[1] = static_cast<uint8_t>(payload_type_rtx_);
if (rtp_header.markerBit)
data_buffer_rtx[1] |= kRtpMarkerBitMask;
}
// Replace sequence number.
uint8_t *ptr = data_buffer_rtx + 2;
RtpUtility::AssignUWord16ToBuffer(ptr, sequence_number_rtx_++);
// Replace SSRC.
ptr += 6;
RtpUtility::AssignUWord32ToBuffer(ptr, ssrc_rtx_);
// Add OSN (original sequence number).
ptr = data_buffer_rtx + rtp_header.headerLength;
RtpUtility::AssignUWord16ToBuffer(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_);
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_);
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_);
sequence_number_rtx_ = rtp_state.sequence_number;
}
RtpState RTPSender::GetRtxRtpState() const {
CriticalSectionScoped lock(send_critsect_);
RtpState state;
state.sequence_number = sequence_number_rtx_;
state.start_timestamp = start_timestamp_;
return state;
}
} // namespace webrtc