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webrtc/call/rtp_video_sender.cc
Elad Alon 8f01c4e1b6 Define FecControllerOverride and plumb it down to VideoEncoder 
The purpose of this interface is to allow VideoEncoder to override
the bandwidth allocation set by FecController in RtpVideoSender.

This CL defines the interface and sends it down to VideoSender.
Two upcoming CLs will:
1. Make LibvpxVp8Encoder pass it on to the (injectable)
   FrameBufferController, where it might be put to good use.
2. Modify RtpVideoSender to respond to the message sent to it
   via this API.

TBR=kwiberg@webrtc.org

Bug: webrtc:10769
Change-Id: I2ef82f0ddcde7fd078e32d8aabf6efe43e0f7f8a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143962
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28416}
2019-06-28 15:57:22 +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/algorithm/container.h"
#include "absl/memory/memory.h"
#include "api/array_view.h"
#include "api/transport/field_trial_based_config.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/playout_delay_oracle.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 webrtc_internal_rtp_video_sender {
RtpStreamSender::RtpStreamSender(
std::unique_ptr<PlayoutDelayOracle> playout_delay_oracle,
std::unique_ptr<RtpRtcp> rtp_rtcp,
std::unique_ptr<RTPSenderVideo> sender_video)
: playout_delay_oracle(std::move(playout_delay_oracle)),
rtp_rtcp(std::move(rtp_rtcp)),
sender_video(std::move(sender_video)) {}
RtpStreamSender::~RtpStreamSender() = default;
} // namespace webrtc_internal_rtp_video_sender
namespace {
static const int kMinSendSidePacketHistorySize = 600;
// We don't do MTU discovery, so assume that we have the standard ethernet MTU.
static const size_t kPathMTU = 1500;
using webrtc_internal_rtp_video_sender::RtpStreamSender;
std::vector<RtpStreamSender> CreateRtpStreamSenders(
Clock* clock,
const RtpConfig& rtp_config,
int rtcp_report_interval_ms,
Transport* send_transport,
RtcpIntraFrameObserver* intra_frame_callback,
RtcpLossNotificationObserver* rtcp_loss_notification_observer,
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,
FrameEncryptorInterface* frame_encryptor,
const CryptoOptions& crypto_options) {
RTC_DCHECK_GT(rtp_config.ssrcs.size(), 0);
RtpRtcp::Configuration configuration;
configuration.clock = clock;
configuration.audio = false;
configuration.receiver_only = false;
configuration.outgoing_transport = send_transport;
configuration.intra_frame_callback = intra_frame_callback;
configuration.rtcp_loss_notification_observer =
rtcp_loss_notification_observer;
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.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<RtpStreamSender> rtp_streams;
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;
auto playout_delay_oracle = absl::make_unique<PlayoutDelayOracle>();
configuration.ack_observer = playout_delay_oracle.get();
auto rtp_rtcp = RtpRtcp::Create(configuration);
rtp_rtcp->SetSendingStatus(false);
rtp_rtcp->SetSendingMediaStatus(false);
rtp_rtcp->SetRTCPStatus(RtcpMode::kCompound);
auto sender_video = absl::make_unique<RTPSenderVideo>(
configuration.clock, rtp_rtcp->RtpSender(),
configuration.flexfec_sender, playout_delay_oracle.get(),
frame_encryptor, crypto_options.sframe.require_frame_encryption,
rtp_config.lntf.enabled, FieldTrialBasedConfig());
rtp_streams.emplace_back(std::move(playout_delay_oracle),
std::move(rtp_rtcp), std::move(sender_video));
}
return rtp_streams;
}
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(
Clock* clock,
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);
}
DataRate CalculateOverheadRate(DataRate data_rate,
DataSize packet_size,
DataSize overhead_per_packet) {
Frequency packet_rate = data_rate / packet_size;
// TOSO(srte): We should not need to round to nearest whole packet per second
// rate here.
return packet_rate.RoundUpTo(Frequency::hertz(1)) * overhead_per_packet;
}
} // namespace
RtpVideoSender::RtpVideoSender(
Clock* clock,
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")),
use_early_loss_detection_(
!webrtc::field_trial::IsDisabled("WebRTC-UseEarlyLossDetection")),
active_(false),
module_process_thread_(nullptr),
suspended_ssrcs_(std::move(suspended_ssrcs)),
flexfec_sender_(
MaybeCreateFlexfecSender(clock, rtp_config, suspended_ssrcs_)),
fec_controller_(std::move(fec_controller)),
rtp_streams_(
CreateRtpStreamSenders(clock,
rtp_config,
rtcp_report_interval_ms,
send_transport,
observers.intra_frame_callback,
observers.rtcp_loss_notification_observer,
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,
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_streams_.size());
module_process_thread_checker_.Detach();
// 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 (const RtpStreamSender& stream : rtp_streams_) {
constexpr bool remb_candidate = true;
transport->packet_router()->AddSendRtpModule(stream.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 (const RtpStreamSender& stream : rtp_streams_) {
RTC_CHECK(stream.rtp_rtcp->RegisterRtpHeaderExtension(extension, id));
}
}
ConfigureProtection();
ConfigureSsrcs();
ConfigureRids();
if (!rtp_config_.mid.empty()) {
for (const RtpStreamSender& stream : rtp_streams_) {
stream.rtp_rtcp->SetMid(rtp_config_.mid);
}
}
for (const RtpStreamSender& stream : rtp_streams_) {
// Simulcast has one module for each layer. Set the CNAME on all modules.
stream.rtp_rtcp->SetCNAME(rtp_config_.c_name.c_str());
stream.rtp_rtcp->RegisterRtcpStatisticsCallback(observers.rtcp_stats);
stream.rtp_rtcp->SetReportBlockDataObserver(
observers.report_block_data_observer);
stream.rtp_rtcp->RegisterSendChannelRtpStatisticsCallback(
observers.rtp_stats);
stream.rtp_rtcp->SetMaxRtpPacketSize(rtp_config_.max_packet_size);
stream.rtp_rtcp->RegisterSendPayloadFrequency(rtp_config_.payload_type,
kVideoPayloadTypeFrequency);
stream.sender_video->RegisterPayloadType(rtp_config_.payload_type,
rtp_config_.payload_name,
rtp_config_.raw_payload);
}
// 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.
transport_->RegisterPacketFeedbackObserver(this);
}
RtpVideoSender::~RtpVideoSender() {
for (const RtpStreamSender& stream : rtp_streams_) {
transport_->packet_router()->RemoveSendRtpModule(stream.rtp_rtcp.get());
}
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 (const RtpStreamSender& stream : rtp_streams_) {
module_process_thread_->RegisterModule(stream.rtp_rtcp.get(),
RTC_FROM_HERE);
}
}
void RtpVideoSender::DeRegisterProcessThread() {
RTC_DCHECK_RUN_ON(&module_process_thread_checker_);
for (const RtpStreamSender& stream : rtp_streams_)
module_process_thread_->DeRegisterModule(stream.rtp_rtcp.get());
}
void RtpVideoSender::SetActive(bool active) {
rtc::CritScope lock(&crit_);
if (active_ == active)
return;
const std::vector<bool> active_modules(rtp_streams_.size(), active);
SetActiveModules(active_modules);
}
void RtpVideoSender::SetActiveModules(const std::vector<bool> active_modules) {
rtc::CritScope lock(&crit_);
RTC_DCHECK_EQ(rtp_streams_.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_streams_[i].rtp_rtcp->SetSendingStatus(active_modules[i]);
// If set to false this module won't send media.
rtp_streams_[i].rtp_rtcp->SetSendingMediaStatus(active_modules[i]);
}
}
bool RtpVideoSender::IsActive() {
rtc::CritScope lock(&crit_);
return active_ && !rtp_streams_.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_streams_.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_streams_.size());
RTPVideoHeader rtp_video_header = params_[stream_index].GetRtpVideoHeader(
encoded_image, codec_specific_info, shared_frame_id_);
uint32_t rtp_timestamp =
encoded_image.Timestamp() +
rtp_streams_[stream_index].rtp_rtcp->StartTimestamp();
// RTCPSender has it's own copy of the timestamp offset, added in
// RTCPSender::BuildSR, hence we must not add the in the offset for this call.
// TODO(nisse): Delete RTCPSender:timestamp_offset_, and see if we can confine
// knowledge of the offset to a single place.
if (!rtp_streams_[stream_index].rtp_rtcp->OnSendingRtpFrame(
encoded_image.Timestamp(), encoded_image.capture_time_ms_,
rtp_config_.payload_type,
encoded_image._frameType == VideoFrameType::kVideoFrameKey)) {
// The payload router could be active but this module isn't sending.
return Result(Result::ERROR_SEND_FAILED);
}
absl::optional<int64_t> expected_retransmission_time_ms;
if (encoded_image.RetransmissionAllowed()) {
expected_retransmission_time_ms =
rtp_streams_[stream_index].rtp_rtcp->ExpectedRetransmissionTimeMs();
}
bool send_result = rtp_streams_[stream_index].sender_video->SendVideo(
encoded_image._frameType, rtp_config_.payload_type, rtp_timestamp,
encoded_image.capture_time_ms_, encoded_image.data(),
encoded_image.size(), fragmentation, &rtp_video_header,
expected_retransmission_time_ms);
if (frame_count_observer_) {
FrameCounts& counts = frame_counts_[stream_index];
if (encoded_image._frameType == VideoFrameType::kVideoFrameKey) {
++counts.key_frames;
} else if (encoded_image._frameType == VideoFrameType::kVideoFrameDelta) {
++counts.delta_frames;
} else {
RTC_DCHECK(encoded_image._frameType == VideoFrameType::kEmptyFrame);
}
frame_count_observer_->FrameCountUpdated(counts,
rtp_config_.ssrcs[stream_index]);
}
if (!send_result)
return Result(Result::ERROR_SEND_FAILED);
return Result(Result::OK, rtp_timestamp);
}
void RtpVideoSender::OnBitrateAllocationUpdated(
const VideoBitrateAllocation& bitrate) {
rtc::CritScope lock(&crit_);
if (IsActive()) {
if (rtp_streams_.size() == 1) {
// If spatial scalability is enabled, it is covered by a single stream.
rtp_streams_[0].rtp_rtcp->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_streams_.size(); ++i) {
// The next spatial layer could be used if the current one is
// inactive.
if (layer_bitrates[i]) {
rtp_streams_[i].rtp_rtcp->SetVideoBitrateAllocation(
*layer_bitrates[i]);
} else {
// Signal a 0 bitrate on a simulcast stream.
rtp_streams_[i].rtp_rtcp->SetVideoBitrateAllocation(
VideoBitrateAllocation());
}
}
}
}
}
void RtpVideoSender::ConfigureProtection() {
// 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 (const RtpStreamSender& stream : rtp_streams_) {
// Set NACK.
stream.rtp_rtcp->SetStorePacketsStatus(true, kMinSendSidePacketHistorySize);
// Set RED/ULPFEC information.
stream.sender_video->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 (size_t i = 0; i < rtp_streams_.size(); ++i) {
if (rtp_streams_[i].rtp_rtcp->SendingMedia()) {
packetization_overhead_bps +=
rtp_streams_[i].sender_video->PacketizationOverheadBps();
}
}
return packetization_overhead_bps;
}
void RtpVideoSender::DeliverRtcp(const uint8_t* packet, size_t length) {
// Runs on a network thread.
for (const RtpStreamSender& stream : rtp_streams_)
stream.rtp_rtcp->IncomingRtcpPacket(packet, length);
}
void RtpVideoSender::ConfigureSsrcs() {
// Configure regular SSRCs.
RTC_CHECK(ssrc_to_rtp_sender_.empty());
for (size_t i = 0; i < rtp_config_.ssrcs.size(); ++i) {
uint32_t ssrc = rtp_config_.ssrcs[i];
RtpRtcp* const rtp_rtcp = rtp_streams_[i].rtp_rtcp.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);
RTPSender* rtp_sender = rtp_rtcp->RtpSender();
RTC_DCHECK(rtp_sender != nullptr);
ssrc_to_rtp_sender_[ssrc] = rtp_sender;
}
// 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_streams_[i].rtp_rtcp.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 (const RtpStreamSender& stream : rtp_streams_) {
stream.rtp_rtcp->SetRtxSendPayloadType(rtp_config_.rtx.payload_type,
rtp_config_.payload_type);
stream.rtp_rtcp->SetRtxSendStatus(kRtxRetransmitted |
kRtxRedundantPayloads);
}
if (rtp_config_.ulpfec.red_payload_type != -1 &&
rtp_config_.ulpfec.red_rtx_payload_type != -1) {
for (const RtpStreamSender& stream : rtp_streams_) {
stream.rtp_rtcp->SetRtxSendPayloadType(
rtp_config_.ulpfec.red_rtx_payload_type,
rtp_config_.ulpfec.red_payload_type);
}
}
}
void RtpVideoSender::ConfigureRids() {
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_streams_.size());
for (size_t i = 0; i < rtp_config_.rids.size(); ++i) {
const std::string& rid = rtp_config_.rids[i];
rtp_streams_[i].rtp_rtcp->SetRid(rid);
}
}
void RtpVideoSender::OnNetworkAvailability(bool network_available) {
for (const RtpStreamSender& stream : rtp_streams_) {
stream.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_streams_[i].rtp_rtcp->SSRC());
rtp_states[ssrc] = rtp_streams_[i].rtp_rtcp->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_streams_[i].rtp_rtcp->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 (const RtpStreamSender& stream : rtp_streams_) {
stream.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_);
DataSize packet_overhead = DataSize::bytes(
overhead_bytes_per_packet_ + transport_overhead_bytes_per_packet_);
DataSize max_total_packet_size = DataSize::bytes(
rtp_config_.max_packet_size + transport_overhead_bytes_per_packet_);
uint32_t payload_bitrate_bps = bitrate_bps;
if (send_side_bwe_with_overhead_) {
DataRate overhead_rate = CalculateOverheadRate(
DataRate::bps(bitrate_bps), max_total_packet_size, packet_overhead);
// TODO(srte): We probably should not accept 0 payload bitrate here.
payload_bitrate_bps =
rtc::saturated_cast<uint32_t>(bitrate_bps - overhead_rate.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 = 0;
if (send_side_bwe_with_overhead_) {
// TODO(srte): The packet size should probably be the same as in the
// CalculateOverheadRate call above (just max_total_packet_size), it doesn't
// make sense to use different packet rates for different overhead
// calculations.
DataRate encoder_overhead_rate = CalculateOverheadRate(
DataRate::bps(encoder_target_rate_bps_),
max_total_packet_size - DataSize::bytes(overhead_bytes_per_packet_),
packet_overhead);
encoder_overhead_rate_bps =
std::min(encoder_overhead_rate.bps<uint32_t>(),
bitrate_bps - encoder_target_rate_bps_);
}
// 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_;
}
std::vector<RtpSequenceNumberMap::Info> RtpVideoSender::GetSentRtpPacketInfos(
uint32_t ssrc,
rtc::ArrayView<const uint16_t> sequence_numbers) const {
for (const auto& rtp_stream : rtp_streams_) {
if (ssrc == rtp_stream.rtp_rtcp->SSRC()) {
return rtp_stream.sender_video->GetSentRtpPacketInfos(sequence_numbers);
}
}
return std::vector<RtpSequenceNumberMap::Info>();
}
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 (const RtpStreamSender& stream : rtp_streams_) {
uint32_t not_used = 0;
uint32_t module_nack_rate = 0;
stream.sender_video->SetFecParameters(*delta_params, *key_params);
*sent_video_rate_bps += stream.sender_video->VideoBitrateSent();
*sent_fec_rate_bps += stream.sender_video->FecOverheadRate();
stream.rtp_rtcp->BitrateSent(&not_used, /*video_rate=*/nullptr,
/*fec_rate=*/nullptr, &module_nack_rate);
*sent_nack_rate_bps += module_nack_rate;
}
return 0;
}
void RtpVideoSender::SetFecAllowed(bool fec_allowed) {
// TODO(bugs.webrtc.og/10769): Handle this message.
}
void RtpVideoSender::OnPacketFeedbackVector(
const std::vector<PacketFeedback>& packet_feedback_vector) {
if (fec_controller_->UseLossVectorMask()) {
rtc::CritScope cs(&crit_);
for (const PacketFeedback& packet : packet_feedback_vector) {
if (packet.send_time_ms == PacketFeedback::kNoSendTime || !packet.ssrc ||
absl::c_find(rtp_config_.ssrcs, *packet.ssrc) ==
rtp_config_.ssrcs.end()) {
// If packet send time is missing, the feedback for this packet has
// probably already been processed, so ignore it.
// If packet does not belong to a registered media ssrc, we are also
// not interested in it.
continue;
}
loss_mask_vector_.push_back(packet.arrival_time_ms ==
PacketFeedback::kNotReceived);
}
}
// Map from SSRC to all acked packets for that RTP module.
std::map<uint32_t, std::vector<uint16_t>> acked_packets_per_ssrc;
for (const PacketFeedback& packet : packet_feedback_vector) {
if (packet.ssrc && packet.arrival_time_ms != PacketFeedback::kNotReceived) {
acked_packets_per_ssrc[*packet.ssrc].push_back(
packet.rtp_sequence_number);
}
}
if (use_early_loss_detection_) {
// Map from SSRC to vector of RTP sequence numbers that are indicated as
// lost by feedback, without being trailed by any received packets.
std::map<uint32_t, std::vector<uint16_t>> early_loss_detected_per_ssrc;
for (const PacketFeedback& packet : packet_feedback_vector) {
if (packet.send_time_ms == PacketFeedback::kNoSendTime || !packet.ssrc ||
absl::c_find(rtp_config_.ssrcs, *packet.ssrc) ==
rtp_config_.ssrcs.end()) {
// If packet send time is missing, the feedback for this packet has
// probably already been processed, so ignore it.
// If packet does not belong to a registered media ssrc, we are also
// not interested in it.
continue;
}
if (packet.arrival_time_ms == PacketFeedback::kNotReceived) {
// Last known lost packet, might not be detectable as lost by remote
// jitter buffer.
early_loss_detected_per_ssrc[*packet.ssrc].push_back(
packet.rtp_sequence_number);
} else {
// Packet received, so any loss prior to this is already detectable.
early_loss_detected_per_ssrc.erase(*packet.ssrc);
}
}
for (const auto& kv : early_loss_detected_per_ssrc) {
const uint32_t ssrc = kv.first;
auto it = ssrc_to_rtp_sender_.find(ssrc);
RTC_DCHECK(it != ssrc_to_rtp_sender_.end());
RTPSender* rtp_sender = it->second;
for (uint16_t sequence_number : kv.second) {
rtp_sender->ReSendPacket(sequence_number);
}
}
}
for (const auto& kv : acked_packets_per_ssrc) {
const uint32_t ssrc = kv.first;
auto it = ssrc_to_rtp_sender_.find(ssrc);
if (it == ssrc_to_rtp_sender_.end()) {
// Packets not for a media SSRC, so likely RTX or FEC. If so, ignore
// since there's no RTP history to clean up anyway.
continue;
}
rtc::ArrayView<const uint16_t> rtp_sequence_numbers(kv.second);
it->second->OnPacketsAcknowledged(rtp_sequence_numbers);
}
}
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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