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webrtc/call/rtp_video_sender.cc
Niels Möller 949f0fdc10 Move FrameCountObserver from RTPSender to RtpVideoSender 
Tbr: sprang@webrtc.org # Trivial change to rtp_video_stream_receiver.cc
Bug: webrtc:7135
Change-Id: Ic292fb02046ea800d7f0876900997d96ed0099d6
Reviewed-on: https://webrtc-review.googlesource.com/c/120161
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26441}
2019-01-29 09:31:11 +00:00

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/*
* Copyright (c) 2015 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 "call/rtp_video_sender.h"
#include <algorithm>
#include <memory>
#include <string>
#include <utility>
#include "absl/memory/memory.h"
#include "call/rtp_transport_controller_send_interface.h"
#include "modules/pacing/packet_router.h"
#include "modules/rtp_rtcp/include/rtp_rtcp.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/rtp_sender.h"
#include "modules/utility/include/process_thread.h"
#include "modules/video_coding/include/video_codec_interface.h"
#include "rtc_base/checks.h"
#include "rtc_base/location.h"
#include "rtc_base/logging.h"
#include "system_wrappers/include/field_trial.h"
namespace webrtc {
namespace {
static const int kMinSendSidePacketHistorySize = 600;
// Assume an average video stream has around 3 packets per frame (1 mbps / 30
// fps / 1400B) A sequence number set with size 5500 will be able to store
// packet sequence number for at least last 60 seconds.
static const int kSendSideSeqNumSetMaxSize = 5500;
// We don't do MTU discovery, so assume that we have the standard ethernet MTU.
static const size_t kPathMTU = 1500;
std::vector<std::unique_ptr<RtpRtcp>> CreateRtpRtcpModules(
const RtpConfig& rtp_config,
int rtcp_report_interval_ms,
Transport* send_transport,
RtcpIntraFrameObserver* intra_frame_callback,
RtcpBandwidthObserver* bandwidth_callback,
RtpTransportControllerSendInterface* transport,
RtcpRttStats* rtt_stats,
FlexfecSender* flexfec_sender,
BitrateStatisticsObserver* bitrate_observer,
RtcpPacketTypeCounterObserver* rtcp_type_observer,
SendSideDelayObserver* send_delay_observer,
SendPacketObserver* send_packet_observer,
RtcEventLog* event_log,
RateLimiter* retransmission_rate_limiter,
OverheadObserver* overhead_observer,
RtpKeepAliveConfig keepalive_config,
FrameEncryptorInterface* frame_encryptor,
const CryptoOptions& crypto_options) {
RTC_DCHECK_GT(rtp_config.ssrcs.size(), 0);
RtpRtcp::Configuration configuration;
configuration.audio = false;
configuration.receiver_only = false;
configuration.outgoing_transport = send_transport;
configuration.intra_frame_callback = intra_frame_callback;
configuration.bandwidth_callback = bandwidth_callback;
configuration.transport_feedback_callback =
transport->transport_feedback_observer();
configuration.rtt_stats = rtt_stats;
configuration.rtcp_packet_type_counter_observer = rtcp_type_observer;
configuration.paced_sender = transport->packet_sender();
configuration.transport_sequence_number_allocator =
transport->packet_router();
configuration.send_bitrate_observer = bitrate_observer;
configuration.send_side_delay_observer = send_delay_observer;
configuration.send_packet_observer = send_packet_observer;
configuration.event_log = event_log;
configuration.retransmission_rate_limiter = retransmission_rate_limiter;
configuration.overhead_observer = overhead_observer;
configuration.keepalive_config = keepalive_config;
configuration.frame_encryptor = frame_encryptor;
configuration.require_frame_encryption =
crypto_options.sframe.require_frame_encryption;
configuration.extmap_allow_mixed = rtp_config.extmap_allow_mixed;
configuration.rtcp_report_interval_ms = rtcp_report_interval_ms;
std::vector<std::unique_ptr<RtpRtcp>> modules;
const std::vector<uint32_t>& flexfec_protected_ssrcs =
rtp_config.flexfec.protected_media_ssrcs;
for (uint32_t ssrc : rtp_config.ssrcs) {
bool enable_flexfec = flexfec_sender != nullptr &&
std::find(flexfec_protected_ssrcs.begin(),
flexfec_protected_ssrcs.end(),
ssrc) != flexfec_protected_ssrcs.end();
configuration.flexfec_sender = enable_flexfec ? flexfec_sender : nullptr;
std::unique_ptr<RtpRtcp> rtp_rtcp =
std::unique_ptr<RtpRtcp>(RtpRtcp::CreateRtpRtcp(configuration));
rtp_rtcp->SetSendingStatus(false);
rtp_rtcp->SetSendingMediaStatus(false);
rtp_rtcp->SetRTCPStatus(RtcpMode::kCompound);
modules.push_back(std::move(rtp_rtcp));
}
return modules;
}
bool PayloadTypeSupportsSkippingFecPackets(const std::string& payload_name) {
const VideoCodecType codecType = PayloadStringToCodecType(payload_name);
if (codecType == kVideoCodecVP8 || codecType == kVideoCodecVP9) {
return true;
}
if (codecType == kVideoCodecGeneric &&
field_trial::IsEnabled("WebRTC-GenericPictureId")) {
return true;
}
return false;
}
// TODO(brandtr): Update this function when we support multistream protection.
std::unique_ptr<FlexfecSender> MaybeCreateFlexfecSender(
const RtpConfig& rtp,
const std::map<uint32_t, RtpState>& suspended_ssrcs) {
if (rtp.flexfec.payload_type < 0) {
return nullptr;
}
RTC_DCHECK_GE(rtp.flexfec.payload_type, 0);
RTC_DCHECK_LE(rtp.flexfec.payload_type, 127);
if (rtp.flexfec.ssrc == 0) {
RTC_LOG(LS_WARNING) << "FlexFEC is enabled, but no FlexFEC SSRC given. "
"Therefore disabling FlexFEC.";
return nullptr;
}
if (rtp.flexfec.protected_media_ssrcs.empty()) {
RTC_LOG(LS_WARNING)
<< "FlexFEC is enabled, but no protected media SSRC given. "
"Therefore disabling FlexFEC.";
return nullptr;
}
if (rtp.flexfec.protected_media_ssrcs.size() > 1) {
RTC_LOG(LS_WARNING)
<< "The supplied FlexfecConfig contained multiple protected "
"media streams, but our implementation currently only "
"supports protecting a single media stream. "
"To avoid confusion, disabling FlexFEC completely.";
return nullptr;
}
const RtpState* rtp_state = nullptr;
auto it = suspended_ssrcs.find(rtp.flexfec.ssrc);
if (it != suspended_ssrcs.end()) {
rtp_state = &it->second;
}
RTC_DCHECK_EQ(1U, rtp.flexfec.protected_media_ssrcs.size());
return absl::make_unique<FlexfecSender>(
rtp.flexfec.payload_type, rtp.flexfec.ssrc,
rtp.flexfec.protected_media_ssrcs[0], rtp.mid, rtp.extensions,
RTPSender::FecExtensionSizes(), rtp_state, Clock::GetRealTimeClock());
}
uint32_t CalculateOverheadRateBps(int packets_per_second,
size_t overhead_bytes_per_packet,
uint32_t max_overhead_bps) {
uint32_t overhead_bps =
static_cast<uint32_t>(8 * overhead_bytes_per_packet * packets_per_second);
return std::min(overhead_bps, max_overhead_bps);
}
int CalculatePacketRate(uint32_t bitrate_bps, size_t packet_size_bytes) {
size_t packet_size_bits = 8 * packet_size_bytes;
// Ceil for int value of bitrate_bps / packet_size_bits.
return static_cast<int>((bitrate_bps + packet_size_bits - 1) /
packet_size_bits);
}
} // namespace
RtpVideoSender::RtpVideoSender(
std::map<uint32_t, RtpState> suspended_ssrcs,
const std::map<uint32_t, RtpPayloadState>& states,
const RtpConfig& rtp_config,
int rtcp_report_interval_ms,
Transport* send_transport,
const RtpSenderObservers& observers,
RtpTransportControllerSendInterface* transport,
RtcEventLog* event_log,
RateLimiter* retransmission_limiter,
std::unique_ptr<FecController> fec_controller,
FrameEncryptorInterface* frame_encryptor,
const CryptoOptions& crypto_options)
: send_side_bwe_with_overhead_(
webrtc::field_trial::IsEnabled("WebRTC-SendSideBwe-WithOverhead")),
account_for_packetization_overhead_(!webrtc::field_trial::IsDisabled(
"WebRTC-SubtractPacketizationOverhead")),
active_(false),
module_process_thread_(nullptr),
suspended_ssrcs_(std::move(suspended_ssrcs)),
flexfec_sender_(MaybeCreateFlexfecSender(rtp_config, suspended_ssrcs_)),
fec_controller_(std::move(fec_controller)),
rtp_modules_(CreateRtpRtcpModules(rtp_config,
rtcp_report_interval_ms,
send_transport,
observers.intra_frame_callback,
transport->GetBandwidthObserver(),
transport,
observers.rtcp_rtt_stats,
flexfec_sender_.get(),
observers.bitrate_observer,
observers.rtcp_type_observer,
observers.send_delay_observer,
observers.send_packet_observer,
event_log,
retransmission_limiter,
this,
transport->keepalive_config(),
frame_encryptor,
crypto_options)),
rtp_config_(rtp_config),
transport_(transport),
transport_overhead_bytes_per_packet_(0),
overhead_bytes_per_packet_(0),
encoder_target_rate_bps_(0),
frame_counts_(rtp_config.ssrcs.size()),
frame_count_observer_(observers.frame_count_observer) {
RTC_DCHECK_EQ(rtp_config.ssrcs.size(), rtp_modules_.size());
module_process_thread_checker_.DetachFromThread();
// SSRCs are assumed to be sorted in the same order as |rtp_modules|.
for (uint32_t ssrc : rtp_config.ssrcs) {
// Restore state if it previously existed.
const RtpPayloadState* state = nullptr;
auto it = states.find(ssrc);
if (it != states.end()) {
state = &it->second;
shared_frame_id_ = std::max(shared_frame_id_, state->shared_frame_id);
}
params_.push_back(RtpPayloadParams(ssrc, state));
}
// RTP/RTCP initialization.
// We add the highest spatial layer first to ensure it'll be prioritized
// when sending padding, with the hope that the packet rate will be smaller,
// and that it's more important to protect than the lower layers.
// TODO(nisse): Consider moving registration with PacketRouter last, after the
// modules are fully configured.
for (auto& rtp_rtcp : rtp_modules_) {
constexpr bool remb_candidate = true;
transport->packet_router()->AddSendRtpModule(rtp_rtcp.get(),
remb_candidate);
}
for (size_t i = 0; i < rtp_config_.extensions.size(); ++i) {
const std::string& extension = rtp_config_.extensions[i].uri;
int id = rtp_config_.extensions[i].id;
RTC_DCHECK(RtpExtension::IsSupportedForVideo(extension));
for (auto& rtp_rtcp : rtp_modules_) {
RTC_CHECK(rtp_rtcp->RegisterRtpHeaderExtension(extension, id));
}
}
ConfigureProtection(rtp_config);
ConfigureSsrcs(rtp_config);
ConfigureRids(rtp_config);
if (!rtp_config.mid.empty()) {
for (auto& rtp_rtcp : rtp_modules_) {
rtp_rtcp->SetMid(rtp_config.mid);
}
}
// TODO(pbos): Should we set CNAME on all RTP modules?
rtp_modules_.front()->SetCNAME(rtp_config.c_name.c_str());
for (auto& rtp_rtcp : rtp_modules_) {
rtp_rtcp->RegisterRtcpStatisticsCallback(observers.rtcp_stats);
rtp_rtcp->RegisterSendChannelRtpStatisticsCallback(observers.rtp_stats);
rtp_rtcp->SetMaxRtpPacketSize(rtp_config.max_packet_size);
rtp_rtcp->RegisterVideoSendPayload(rtp_config.payload_type,
rtp_config.payload_name.c_str());
}
// Currently, both ULPFEC and FlexFEC use the same FEC rate calculation logic,
// so enable that logic if either of those FEC schemes are enabled.
fec_controller_->SetProtectionMethod(FecEnabled(), NackEnabled());
fec_controller_->SetProtectionCallback(this);
// Signal congestion controller this object is ready for OnPacket* callbacks.
if (fec_controller_->UseLossVectorMask()) {
transport_->RegisterPacketFeedbackObserver(this);
}
}
RtpVideoSender::~RtpVideoSender() {
for (auto& rtp_rtcp : rtp_modules_) {
transport_->packet_router()->RemoveSendRtpModule(rtp_rtcp.get());
}
if (fec_controller_->UseLossVectorMask()) {
transport_->DeRegisterPacketFeedbackObserver(this);
}
}
void RtpVideoSender::RegisterProcessThread(
ProcessThread* module_process_thread) {
RTC_DCHECK_RUN_ON(&module_process_thread_checker_);
RTC_DCHECK(!module_process_thread_);
module_process_thread_ = module_process_thread;
for (auto& rtp_rtcp : rtp_modules_)
module_process_thread_->RegisterModule(rtp_rtcp.get(), RTC_FROM_HERE);
}
void RtpVideoSender::DeRegisterProcessThread() {
RTC_DCHECK_RUN_ON(&module_process_thread_checker_);
for (auto& rtp_rtcp : rtp_modules_)
module_process_thread_->DeRegisterModule(rtp_rtcp.get());
}
void RtpVideoSender::SetActive(bool active) {
rtc::CritScope lock(&crit_);
if (active_ == active)
return;
const std::vector<bool> active_modules(rtp_modules_.size(), active);
SetActiveModules(active_modules);
}
void RtpVideoSender::SetActiveModules(const std::vector<bool> active_modules) {
rtc::CritScope lock(&crit_);
RTC_DCHECK_EQ(rtp_modules_.size(), active_modules.size());
active_ = false;
for (size_t i = 0; i < active_modules.size(); ++i) {
if (active_modules[i]) {
active_ = true;
}
// Sends a kRtcpByeCode when going from true to false.
rtp_modules_[i]->SetSendingStatus(active_modules[i]);
// If set to false this module won't send media.
rtp_modules_[i]->SetSendingMediaStatus(active_modules[i]);
}
}
bool RtpVideoSender::IsActive() {
rtc::CritScope lock(&crit_);
return active_ && !rtp_modules_.empty();
}
EncodedImageCallback::Result RtpVideoSender::OnEncodedImage(
const EncodedImage& encoded_image,
const CodecSpecificInfo* codec_specific_info,
const RTPFragmentationHeader* fragmentation) {
fec_controller_->UpdateWithEncodedData(encoded_image.size(),
encoded_image._frameType);
rtc::CritScope lock(&crit_);
RTC_DCHECK(!rtp_modules_.empty());
if (!active_)
return Result(Result::ERROR_SEND_FAILED);
shared_frame_id_++;
size_t stream_index = 0;
if (codec_specific_info &&
(codec_specific_info->codecType == kVideoCodecVP8 ||
codec_specific_info->codecType == kVideoCodecH264 ||
codec_specific_info->codecType == kVideoCodecGeneric)) {
// Map spatial index to simulcast.
stream_index = encoded_image.SpatialIndex().value_or(0);
}
RTC_DCHECK_LT(stream_index, rtp_modules_.size());
RTPVideoHeader rtp_video_header = params_[stream_index].GetRtpVideoHeader(
encoded_image, codec_specific_info, shared_frame_id_);
uint32_t frame_id;
if (!rtp_modules_[stream_index]->Sending()) {
// The payload router could be active but this module isn't sending.
return Result(Result::ERROR_SEND_FAILED);
}
bool send_result = rtp_modules_[stream_index]->SendOutgoingData(
encoded_image._frameType, rtp_config_.payload_type,
encoded_image.Timestamp(), encoded_image.capture_time_ms_,
encoded_image.data(), encoded_image.size(), fragmentation,
&rtp_video_header, &frame_id);
if (frame_count_observer_) {
FrameCounts& counts = frame_counts_[stream_index];
if (encoded_image._frameType == kVideoFrameKey) {
++counts.key_frames;
} else if (encoded_image._frameType == kVideoFrameDelta) {
++counts.delta_frames;
} else {
RTC_DCHECK_EQ(encoded_image._frameType, kEmptyFrame);
}
frame_count_observer_->FrameCountUpdated(counts,
rtp_config_.ssrcs[stream_index]);
}
if (!send_result)
return Result(Result::ERROR_SEND_FAILED);
return Result(Result::OK, frame_id);
}
void RtpVideoSender::OnBitrateAllocationUpdated(
const VideoBitrateAllocation& bitrate) {
rtc::CritScope lock(&crit_);
if (IsActive()) {
if (rtp_modules_.size() == 1) {
// If spatial scalability is enabled, it is covered by a single stream.
rtp_modules_[0]->SetVideoBitrateAllocation(bitrate);
} else {
std::vector<absl::optional<VideoBitrateAllocation>> layer_bitrates =
bitrate.GetSimulcastAllocations();
// Simulcast is in use, split the VideoBitrateAllocation into one struct
// per rtp stream, moving over the temporal layer allocation.
for (size_t i = 0; i < rtp_modules_.size(); ++i) {
// The next spatial layer could be used if the current one is
// inactive.
if (layer_bitrates[i]) {
rtp_modules_[i]->SetVideoBitrateAllocation(*layer_bitrates[i]);
} else {
// Signal a 0 bitrate on a simulcast stream.
rtp_modules_[i]->SetVideoBitrateAllocation(VideoBitrateAllocation());
}
}
}
}
}
void RtpVideoSender::ConfigureProtection(const RtpConfig& rtp_config) {
// Consistency of FlexFEC parameters is checked in MaybeCreateFlexfecSender.
const bool flexfec_enabled = (flexfec_sender_ != nullptr);
// Consistency of NACK and RED+ULPFEC parameters is checked in this function.
const bool nack_enabled = rtp_config.nack.rtp_history_ms > 0;
int red_payload_type = rtp_config.ulpfec.red_payload_type;
int ulpfec_payload_type = rtp_config.ulpfec.ulpfec_payload_type;
// Shorthands.
auto IsRedEnabled = [&]() { return red_payload_type >= 0; };
auto IsUlpfecEnabled = [&]() { return ulpfec_payload_type >= 0; };
auto DisableRedAndUlpfec = [&]() {
red_payload_type = -1;
ulpfec_payload_type = -1;
};
if (webrtc::field_trial::IsEnabled("WebRTC-DisableUlpFecExperiment")) {
RTC_LOG(LS_INFO) << "Experiment to disable sending ULPFEC is enabled.";
DisableRedAndUlpfec();
}
// If enabled, FlexFEC takes priority over RED+ULPFEC.
if (flexfec_enabled) {
if (IsUlpfecEnabled()) {
RTC_LOG(LS_INFO)
<< "Both FlexFEC and ULPFEC are configured. Disabling ULPFEC.";
}
DisableRedAndUlpfec();
}
// Payload types without picture ID cannot determine that a stream is complete
// without retransmitting FEC, so using ULPFEC + NACK for H.264 (for instance)
// is a waste of bandwidth since FEC packets still have to be transmitted.
// Note that this is not the case with FlexFEC.
if (nack_enabled && IsUlpfecEnabled() &&
!PayloadTypeSupportsSkippingFecPackets(rtp_config.payload_name)) {
RTC_LOG(LS_WARNING)
<< "Transmitting payload type without picture ID using "
"NACK+ULPFEC is a waste of bandwidth since ULPFEC packets "
"also have to be retransmitted. Disabling ULPFEC.";
DisableRedAndUlpfec();
}
// Verify payload types.
if (IsUlpfecEnabled() ^ IsRedEnabled()) {
RTC_LOG(LS_WARNING)
<< "Only RED or only ULPFEC enabled, but not both. Disabling both.";
DisableRedAndUlpfec();
}
for (auto& rtp_rtcp : rtp_modules_) {
// Set NACK.
rtp_rtcp->SetStorePacketsStatus(true, kMinSendSidePacketHistorySize);
// Set RED/ULPFEC information.
rtp_rtcp->SetUlpfecConfig(red_payload_type, ulpfec_payload_type);
}
}
bool RtpVideoSender::FecEnabled() const {
const bool flexfec_enabled = (flexfec_sender_ != nullptr);
const bool ulpfec_enabled =
!webrtc::field_trial::IsEnabled("WebRTC-DisableUlpFecExperiment") &&
(rtp_config_.ulpfec.ulpfec_payload_type >= 0);
return flexfec_enabled || ulpfec_enabled;
}
bool RtpVideoSender::NackEnabled() const {
const bool nack_enabled = rtp_config_.nack.rtp_history_ms > 0;
return nack_enabled;
}
uint32_t RtpVideoSender::GetPacketizationOverheadRate() const {
uint32_t packetization_overhead_bps = 0;
for (auto& rtp_rtcp : rtp_modules_) {
if (rtp_rtcp->SendingMedia()) {
packetization_overhead_bps += rtp_rtcp->PacketizationOverheadBps();
}
}
return packetization_overhead_bps;
}
void RtpVideoSender::DeliverRtcp(const uint8_t* packet, size_t length) {
// Runs on a network thread.
for (auto& rtp_rtcp : rtp_modules_)
rtp_rtcp->IncomingRtcpPacket(packet, length);
}
void RtpVideoSender::ConfigureSsrcs(const RtpConfig& rtp_config) {
// Configure regular SSRCs.
for (size_t i = 0; i < rtp_config.ssrcs.size(); ++i) {
uint32_t ssrc = rtp_config.ssrcs[i];
RtpRtcp* const rtp_rtcp = rtp_modules_[i].get();
rtp_rtcp->SetSSRC(ssrc);
// Restore RTP state if previous existed.
auto it = suspended_ssrcs_.find(ssrc);
if (it != suspended_ssrcs_.end())
rtp_rtcp->SetRtpState(it->second);
}
// Set up RTX if available.
if (rtp_config.rtx.ssrcs.empty())
return;
// Configure RTX SSRCs.
RTC_DCHECK_EQ(rtp_config.rtx.ssrcs.size(), rtp_config.ssrcs.size());
for (size_t i = 0; i < rtp_config.rtx.ssrcs.size(); ++i) {
uint32_t ssrc = rtp_config.rtx.ssrcs[i];
RtpRtcp* const rtp_rtcp = rtp_modules_[i].get();
rtp_rtcp->SetRtxSsrc(ssrc);
auto it = suspended_ssrcs_.find(ssrc);
if (it != suspended_ssrcs_.end())
rtp_rtcp->SetRtxState(it->second);
}
// Configure RTX payload types.
RTC_DCHECK_GE(rtp_config.rtx.payload_type, 0);
for (auto& rtp_rtcp : rtp_modules_) {
rtp_rtcp->SetRtxSendPayloadType(rtp_config.rtx.payload_type,
rtp_config.payload_type);
rtp_rtcp->SetRtxSendStatus(kRtxRetransmitted | kRtxRedundantPayloads);
}
if (rtp_config.ulpfec.red_payload_type != -1 &&
rtp_config.ulpfec.red_rtx_payload_type != -1) {
for (auto& rtp_rtcp : rtp_modules_) {
rtp_rtcp->SetRtxSendPayloadType(rtp_config.ulpfec.red_rtx_payload_type,
rtp_config.ulpfec.red_payload_type);
}
}
}
void RtpVideoSender::ConfigureRids(const RtpConfig& rtp_config) {
RTC_DCHECK(rtp_config.rids.empty() ||
rtp_config.rids.size() == rtp_config.ssrcs.size());
RTC_DCHECK(rtp_config.rids.empty() ||
rtp_config.rids.size() == rtp_modules_.size());
for (size_t i = 0; i < rtp_config.rids.size(); ++i) {
const std::string& rid = rtp_config.rids[i];
RtpRtcp* const rtp_rtcp = rtp_modules_[i].get();
rtp_rtcp->SetRid(rid);
}
}
void RtpVideoSender::OnNetworkAvailability(bool network_available) {
for (auto& rtp_rtcp : rtp_modules_) {
rtp_rtcp->SetRTCPStatus(network_available ? rtp_config_.rtcp_mode
: RtcpMode::kOff);
}
}
std::map<uint32_t, RtpState> RtpVideoSender::GetRtpStates() const {
std::map<uint32_t, RtpState> rtp_states;
for (size_t i = 0; i < rtp_config_.ssrcs.size(); ++i) {
uint32_t ssrc = rtp_config_.ssrcs[i];
RTC_DCHECK_EQ(ssrc, rtp_modules_[i]->SSRC());
rtp_states[ssrc] = rtp_modules_[i]->GetRtpState();
}
for (size_t i = 0; i < rtp_config_.rtx.ssrcs.size(); ++i) {
uint32_t ssrc = rtp_config_.rtx.ssrcs[i];
rtp_states[ssrc] = rtp_modules_[i]->GetRtxState();
}
if (flexfec_sender_) {
uint32_t ssrc = rtp_config_.flexfec.ssrc;
rtp_states[ssrc] = flexfec_sender_->GetRtpState();
}
return rtp_states;
}
std::map<uint32_t, RtpPayloadState> RtpVideoSender::GetRtpPayloadStates()
const {
rtc::CritScope lock(&crit_);
std::map<uint32_t, RtpPayloadState> payload_states;
for (const auto& param : params_) {
payload_states[param.ssrc()] = param.state();
payload_states[param.ssrc()].shared_frame_id = shared_frame_id_;
}
return payload_states;
}
void RtpVideoSender::OnTransportOverheadChanged(
size_t transport_overhead_bytes_per_packet) {
rtc::CritScope lock(&crit_);
transport_overhead_bytes_per_packet_ = transport_overhead_bytes_per_packet;
size_t max_rtp_packet_size =
std::min(rtp_config_.max_packet_size,
kPathMTU - transport_overhead_bytes_per_packet_);
for (auto& rtp_rtcp : rtp_modules_) {
rtp_rtcp->SetMaxRtpPacketSize(max_rtp_packet_size);
}
}
void RtpVideoSender::OnOverheadChanged(size_t overhead_bytes_per_packet) {
rtc::CritScope lock(&crit_);
overhead_bytes_per_packet_ = overhead_bytes_per_packet;
}
void RtpVideoSender::OnBitrateUpdated(uint32_t bitrate_bps,
uint8_t fraction_loss,
int64_t rtt,
int framerate) {
// Substract overhead from bitrate.
rtc::CritScope lock(&crit_);
uint32_t payload_bitrate_bps = bitrate_bps;
if (send_side_bwe_with_overhead_) {
uint32_t overhead_bps = CalculateOverheadRateBps(
CalculatePacketRate(
bitrate_bps,
rtp_config_.max_packet_size + transport_overhead_bytes_per_packet_),
overhead_bytes_per_packet_ + transport_overhead_bytes_per_packet_,
bitrate_bps);
RTC_DCHECK_LE(overhead_bps, bitrate_bps);
payload_bitrate_bps = bitrate_bps - overhead_bps;
}
// Get the encoder target rate. It is the estimated network rate -
// protection overhead.
encoder_target_rate_bps_ = fec_controller_->UpdateFecRates(
payload_bitrate_bps, framerate, fraction_loss, loss_mask_vector_, rtt);
uint32_t packetization_rate_bps = 0;
if (account_for_packetization_overhead_) {
// Subtract packetization overhead from the encoder target. If target rate
// is really low, cap the overhead at 50%. This also avoids the case where
// |encoder_target_rate_bps_| is 0 due to encoder pause event while the
// packetization rate is positive since packets are still flowing.
packetization_rate_bps =
std::min(GetPacketizationOverheadRate(), encoder_target_rate_bps_ / 2);
encoder_target_rate_bps_ -= packetization_rate_bps;
}
loss_mask_vector_.clear();
uint32_t encoder_overhead_rate_bps =
send_side_bwe_with_overhead_
? CalculateOverheadRateBps(
CalculatePacketRate(encoder_target_rate_bps_,
rtp_config_.max_packet_size +
transport_overhead_bytes_per_packet_ -
overhead_bytes_per_packet_),
overhead_bytes_per_packet_ +
transport_overhead_bytes_per_packet_,
bitrate_bps - encoder_target_rate_bps_)
: 0;
// When the field trial "WebRTC-SendSideBwe-WithOverhead" is enabled
// protection_bitrate includes overhead.
const uint32_t media_rate = encoder_target_rate_bps_ +
encoder_overhead_rate_bps +
packetization_rate_bps;
RTC_DCHECK_GE(bitrate_bps, media_rate);
protection_bitrate_bps_ = bitrate_bps - media_rate;
}
uint32_t RtpVideoSender::GetPayloadBitrateBps() const {
return encoder_target_rate_bps_;
}
uint32_t RtpVideoSender::GetProtectionBitrateBps() const {
return protection_bitrate_bps_;
}
int RtpVideoSender::ProtectionRequest(const FecProtectionParams* delta_params,
const FecProtectionParams* key_params,
uint32_t* sent_video_rate_bps,
uint32_t* sent_nack_rate_bps,
uint32_t* sent_fec_rate_bps) {
*sent_video_rate_bps = 0;
*sent_nack_rate_bps = 0;
*sent_fec_rate_bps = 0;
for (auto& rtp_rtcp : rtp_modules_) {
uint32_t not_used = 0;
uint32_t module_video_rate = 0;
uint32_t module_fec_rate = 0;
uint32_t module_nack_rate = 0;
rtp_rtcp->SetFecParameters(*delta_params, *key_params);
rtp_rtcp->BitrateSent(&not_used, &module_video_rate, &module_fec_rate,
&module_nack_rate);
*sent_video_rate_bps += module_video_rate;
*sent_nack_rate_bps += module_nack_rate;
*sent_fec_rate_bps += module_fec_rate;
}
return 0;
}
void RtpVideoSender::OnPacketAdded(uint32_t ssrc, uint16_t seq_num) {
const auto ssrcs = rtp_config_.ssrcs;
if (std::find(ssrcs.begin(), ssrcs.end(), ssrc) != ssrcs.end()) {
feedback_packet_seq_num_set_.insert(seq_num);
if (feedback_packet_seq_num_set_.size() > kSendSideSeqNumSetMaxSize) {
RTC_LOG(LS_WARNING) << "Feedback packet sequence number set exceed it's "
"max size', will get reset.";
feedback_packet_seq_num_set_.clear();
}
}
}
void RtpVideoSender::OnPacketFeedbackVector(
const std::vector<PacketFeedback>& packet_feedback_vector) {
rtc::CritScope lock(&crit_);
// Lost feedbacks are not considered to be lost packets.
for (const PacketFeedback& packet : packet_feedback_vector) {
auto it = feedback_packet_seq_num_set_.find(packet.sequence_number);
if (it != feedback_packet_seq_num_set_.end()) {
const bool lost = packet.arrival_time_ms == PacketFeedback::kNotReceived;
loss_mask_vector_.push_back(lost);
feedback_packet_seq_num_set_.erase(it);
}
}
}
void RtpVideoSender::SetEncodingData(size_t width,
size_t height,
size_t num_temporal_layers) {
fec_controller_->SetEncodingData(width, height, num_temporal_layers,
rtp_config_.max_packet_size);
}
} // namespace webrtc
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