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webrtc/call/rtp_video_sender.cc
Per K 02af84064c PacketRouter directly notify RtpTransportControllerSender when sending 
With this cl
RtpTransportControllerSend::OnAddPacket is instead directly invoked from PacketRouter::SendPacket instead of going via RTP module.

Transport sequence numbers are instead of directly written to header
extension, added to RtpPacketToSendMetaData and written to the extenion
by RTP module.
This is to allow transport sequence numbers without actually sending
them in an extension.

Bug: webrtc:15368
Change-Id: Idd03e02a4257dfc4d0f1898b2803345975d7dad2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/344720
Reviewed-by: Erik Språng <sprang@google.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41974}
2024-03-28 09:27:43 +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/strings/match.h"
#include "absl/strings/string_view.h"
#include "api/array_view.h"
#include "api/task_queue/task_queue_factory.h"
#include "api/transport/field_trial_based_config.h"
#include "api/units/time_delta.h"
#include "api/video_codecs/video_codec.h"
#include "call/rtp_transport_controller_send_interface.h"
#include "modules/pacing/packet_router.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
#include "modules/rtp_rtcp/source/rtp_sender.h"
#include "modules/video_coding/include/video_codec_interface.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/trace_event.h"
namespace webrtc {
namespace webrtc_internal_rtp_video_sender {
RtpStreamSender::RtpStreamSender(
std::unique_ptr<ModuleRtpRtcpImpl2> rtp_rtcp,
std::unique_ptr<RTPSenderVideo> sender_video,
std::unique_ptr<VideoFecGenerator> fec_generator)
: rtp_rtcp(std::move(rtp_rtcp)),
sender_video(std::move(sender_video)),
fec_generator(std::move(fec_generator)) {}
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;
bool PayloadTypeSupportsSkippingFecPackets(absl::string_view payload_name,
const FieldTrialsView& trials) {
const VideoCodecType codecType =
PayloadStringToCodecType(std::string(payload_name));
if (codecType == kVideoCodecVP8 || codecType == kVideoCodecVP9) {
return true;
}
if (codecType == kVideoCodecGeneric &&
absl::StartsWith(trials.Lookup("WebRTC-GenericPictureId"), "Enabled")) {
return true;
}
return false;
}
bool ShouldDisableRedAndUlpfec(bool flexfec_enabled,
const RtpConfig& rtp_config,
const FieldTrialsView& trials) {
// Consistency of NACK and RED+ULPFEC parameters is checked in this function.
const bool nack_enabled = rtp_config.nack.rtp_history_ms > 0;
// Shorthands.
auto IsRedEnabled = [&]() { return rtp_config.ulpfec.red_payload_type >= 0; };
auto IsUlpfecEnabled = [&]() {
return rtp_config.ulpfec.ulpfec_payload_type >= 0;
};
bool should_disable_red_and_ulpfec = false;
if (absl::StartsWith(trials.Lookup("WebRTC-DisableUlpFecExperiment"),
"Enabled")) {
RTC_LOG(LS_INFO) << "Experiment to disable sending ULPFEC is enabled.";
should_disable_red_and_ulpfec = true;
}
// 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.";
}
should_disable_red_and_ulpfec = true;
}
// 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, trials)) {
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.";
should_disable_red_and_ulpfec = true;
}
// Verify payload types.
if (IsUlpfecEnabled() ^ IsRedEnabled()) {
RTC_LOG(LS_WARNING)
<< "Only RED or only ULPFEC enabled, but not both. Disabling both.";
should_disable_red_and_ulpfec = true;
}
return should_disable_red_and_ulpfec;
}
// TODO(brandtr): Update this function when we support multistream protection.
std::unique_ptr<VideoFecGenerator> MaybeCreateFecGenerator(
Clock* clock,
const RtpConfig& rtp,
const std::map<uint32_t, RtpState>& suspended_ssrcs,
int simulcast_index,
const FieldTrialsView& trials) {
// If flexfec is configured that takes priority.
if (rtp.flexfec.payload_type >= 0) {
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;
}
if (absl::c_find(rtp.flexfec.protected_media_ssrcs,
rtp.ssrcs[simulcast_index]) ==
rtp.flexfec.protected_media_ssrcs.end()) {
// Media SSRC not among flexfec protected SSRCs.
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 std::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);
} else if (rtp.ulpfec.red_payload_type >= 0 &&
rtp.ulpfec.ulpfec_payload_type >= 0 &&
!ShouldDisableRedAndUlpfec(/*flexfec_enabled=*/false, rtp,
trials)) {
// Flexfec not configured, but ulpfec is and is not disabled.
return std::make_unique<UlpfecGenerator>(
rtp.ulpfec.red_payload_type, rtp.ulpfec.ulpfec_payload_type, clock);
}
// Not a single FEC is given.
return nullptr;
}
std::vector<RtpStreamSender> CreateRtpStreamSenders(
Clock* clock,
const RtpConfig& rtp_config,
const RtpSenderObservers& observers,
int rtcp_report_interval_ms,
Transport* send_transport,
RtpTransportControllerSendInterface* transport,
const std::map<uint32_t, RtpState>& suspended_ssrcs,
RtcEventLog* event_log,
RateLimiter* retransmission_rate_limiter,
FrameEncryptorInterface* frame_encryptor,
const CryptoOptions& crypto_options,
rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,
const FieldTrialsView& trials,
TaskQueueFactory* task_queue_factory) {
RTC_DCHECK_GT(rtp_config.ssrcs.size(), 0);
RTC_DCHECK(task_queue_factory);
RtpRtcpInterface::Configuration configuration;
configuration.clock = clock;
configuration.audio = false;
configuration.receiver_only = false;
configuration.outgoing_transport = send_transport;
configuration.intra_frame_callback = observers.intra_frame_callback;
configuration.rtcp_loss_notification_observer =
observers.rtcp_loss_notification_observer;
configuration.network_link_rtcp_observer = transport->GetRtcpObserver();
configuration.network_state_estimate_observer =
transport->network_state_estimate_observer();
configuration.rtt_stats = observers.rtcp_rtt_stats;
configuration.rtcp_packet_type_counter_observer =
observers.rtcp_type_observer;
configuration.report_block_data_observer =
observers.report_block_data_observer;
configuration.paced_sender = transport->packet_sender();
configuration.send_bitrate_observer = observers.bitrate_observer;
configuration.send_packet_observer = observers.send_packet_observer;
configuration.event_log = event_log;
if (trials.IsDisabled("WebRTC-DisableRtxRateLimiter")) {
configuration.retransmission_rate_limiter = retransmission_rate_limiter;
}
configuration.rtp_stats_callback = observers.rtp_stats;
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;
configuration.field_trials = &trials;
configuration.enable_send_packet_batching =
rtp_config.enable_send_packet_batching;
std::vector<RtpStreamSender> rtp_streams;
RTC_DCHECK(rtp_config.rtx.ssrcs.empty() ||
rtp_config.rtx.ssrcs.size() == rtp_config.ssrcs.size());
// Some streams could have been disabled, but the rids are still there.
// This will occur when simulcast has been disabled for a codec (e.g. VP9)
RTC_DCHECK(rtp_config.rids.empty() ||
rtp_config.rids.size() >= rtp_config.ssrcs.size());
for (size_t i = 0; i < rtp_config.ssrcs.size(); ++i) {
RTPSenderVideo::Config video_config;
configuration.local_media_ssrc = rtp_config.ssrcs[i];
std::unique_ptr<VideoFecGenerator> fec_generator =
MaybeCreateFecGenerator(clock, rtp_config, suspended_ssrcs, i, trials);
configuration.fec_generator = fec_generator.get();
configuration.rtx_send_ssrc =
rtp_config.GetRtxSsrcAssociatedWithMediaSsrc(rtp_config.ssrcs[i]);
RTC_DCHECK_EQ(configuration.rtx_send_ssrc.has_value(),
!rtp_config.rtx.ssrcs.empty());
configuration.rid = (i < rtp_config.rids.size()) ? rtp_config.rids[i] : "";
configuration.need_rtp_packet_infos = rtp_config.lntf.enabled;
std::unique_ptr<ModuleRtpRtcpImpl2> rtp_rtcp(
ModuleRtpRtcpImpl2::Create(configuration));
rtp_rtcp->SetSendingStatus(false);
rtp_rtcp->SetSendingMediaStatus(false);
rtp_rtcp->SetRTCPStatus(RtcpMode::kCompound);
// Set NACK.
rtp_rtcp->SetStorePacketsStatus(true, kMinSendSidePacketHistorySize);
video_config.clock = configuration.clock;
video_config.rtp_sender = rtp_rtcp->RtpSender();
video_config.frame_encryptor = frame_encryptor;
video_config.require_frame_encryption =
crypto_options.sframe.require_frame_encryption;
video_config.field_trials = &trials;
video_config.enable_retransmit_all_layers =
!video_config.field_trials->IsDisabled(
"WebRTC-Video-EnableRetransmitAllLayers");
const bool using_flexfec =
fec_generator &&
fec_generator->GetFecType() == VideoFecGenerator::FecType::kFlexFec;
const bool should_disable_red_and_ulpfec =
ShouldDisableRedAndUlpfec(using_flexfec, rtp_config, trials);
if (!should_disable_red_and_ulpfec &&
rtp_config.ulpfec.red_payload_type != -1) {
video_config.red_payload_type = rtp_config.ulpfec.red_payload_type;
}
if (fec_generator) {
video_config.fec_type = fec_generator->GetFecType();
video_config.fec_overhead_bytes = fec_generator->MaxPacketOverhead();
}
video_config.frame_transformer = frame_transformer;
video_config.task_queue_factory = task_queue_factory;
auto sender_video = std::make_unique<RTPSenderVideo>(video_config);
rtp_streams.emplace_back(std::move(rtp_rtcp), std::move(sender_video),
std::move(fec_generator));
}
return rtp_streams;
}
absl::optional<VideoCodecType> GetVideoCodecType(const RtpConfig& config) {
if (config.raw_payload) {
return absl::nullopt;
}
return PayloadStringToCodecType(config.payload_name);
}
bool TransportSeqNumExtensionConfigured(const RtpConfig& config) {
return absl::c_any_of(config.extensions, [](const RtpExtension& ext) {
return ext.uri == RtpExtension::kTransportSequenceNumberUri;
});
}
// Returns true when some coded video sequence can be decoded starting with
// this frame without requiring any previous frames.
// e.g. it is the same as a key frame when spatial scalability is not used.
// When spatial scalability is used, then it is true for layer frames of
// a key frame without inter-layer dependencies.
bool IsFirstFrameOfACodedVideoSequence(
const EncodedImage& encoded_image,
const CodecSpecificInfo* codec_specific_info) {
if (encoded_image._frameType != VideoFrameType::kVideoFrameKey) {
return false;
}
if (codec_specific_info != nullptr) {
if (codec_specific_info->generic_frame_info.has_value()) {
// This function is used before
// `codec_specific_info->generic_frame_info->frame_diffs` are calculated,
// so need to use a more complicated way to check for presence of the
// dependencies.
return absl::c_none_of(
codec_specific_info->generic_frame_info->encoder_buffers,
[](const CodecBufferUsage& buffer) { return buffer.referenced; });
}
if (codec_specific_info->codecType == VideoCodecType::kVideoCodecVP8 ||
codec_specific_info->codecType == VideoCodecType::kVideoCodecH264 ||
codec_specific_info->codecType == VideoCodecType::kVideoCodecGeneric) {
// These codecs do not support intra picture dependencies, so a frame
// marked as a key frame should be a key frame.
return true;
}
}
// Without depenedencies described in generic format do an educated guess.
// It might be wrong for VP9 with spatial layer 0 skipped or higher spatial
// layer not depending on the spatial layer 0. This corner case is unimportant
// for current usage of this helper function.
// Use <= to accept both 0 (i.e. the first) and nullopt (i.e. the only).
return encoded_image.SpatialIndex() <= 0;
}
} // namespace
RtpVideoSender::RtpVideoSender(
Clock* clock,
const 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,
rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,
const FieldTrialsView& field_trials,
TaskQueueFactory* task_queue_factory)
: field_trials_(field_trials),
use_frame_rate_for_overhead_(absl::StartsWith(
field_trials_.Lookup("WebRTC-Video-UseFrameRateForOverhead"),
"Enabled")),
has_packet_feedback_(TransportSeqNumExtensionConfigured(rtp_config)),
active_(false),
fec_controller_(std::move(fec_controller)),
fec_allowed_(true),
rtp_streams_(CreateRtpStreamSenders(clock,
rtp_config,
observers,
rtcp_report_interval_ms,
send_transport,
transport,
suspended_ssrcs,
event_log,
retransmission_limiter,
frame_encryptor,
crypto_options,
std::move(frame_transformer),
field_trials_,
task_queue_factory)),
rtp_config_(rtp_config),
codec_type_(GetVideoCodecType(rtp_config)),
transport_(transport),
independent_frame_ids_(
field_trials_.IsEnabled(
"WebRTC-Video-SimulcastIndependentFrameIds") &&
field_trials_.IsDisabled("WebRTC-GenericDescriptorAuth")),
transport_overhead_bytes_per_packet_(0),
encoder_target_rate_bps_(0),
frame_counts_(rtp_config.ssrcs.size()),
frame_count_observer_(observers.frame_count_observer) {
transport_checker_.Detach();
RTC_DCHECK_EQ(rtp_config_.ssrcs.size(), rtp_streams_.size());
if (has_packet_feedback_)
transport_->IncludeOverheadInPacedSender();
// 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, field_trials_));
}
// RTP/RTCP initialization.
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_) {
stream.rtp_rtcp->RegisterRtpHeaderExtension(extension, id);
}
}
ConfigureSsrcs(suspended_ssrcs);
if (!rtp_config_.mid.empty()) {
for (const RtpStreamSender& stream : rtp_streams_) {
stream.rtp_rtcp->SetMid(rtp_config_.mid);
}
}
bool fec_enabled = false;
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->SetMaxRtpPacketSize(rtp_config_.max_packet_size);
stream.rtp_rtcp->RegisterSendPayloadFrequency(rtp_config_.payload_type,
kVideoPayloadTypeFrequency);
if (stream.fec_generator != nullptr) {
fec_enabled = true;
}
}
// 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(fec_enabled, NackEnabled());
fec_controller_->SetProtectionCallback(this);
// Construction happens on the worker thread (see Call::CreateVideoSendStream)
// but subseqeuent calls to the RTP state will happen on one of two threads:
// * The pacer thread for actually sending packets.
// * The transport thread when tearing down and quering GetRtpState().
// Detach thread checkers.
for (const RtpStreamSender& stream : rtp_streams_) {
stream.rtp_rtcp->OnPacketSendingThreadSwitched();
}
}
RtpVideoSender::~RtpVideoSender() {
RTC_DCHECK_RUN_ON(&transport_checker_);
SetActiveModulesLocked(
/*sending=*/false);
}
void RtpVideoSender::SetSending(bool enabled) {
RTC_DCHECK_RUN_ON(&transport_checker_);
MutexLock lock(&mutex_);
if (enabled == active_) {
return;
}
SetActiveModulesLocked(/*sending=*/enabled);
}
void RtpVideoSender::SetActiveModulesLocked(bool sending) {
RTC_DCHECK_RUN_ON(&transport_checker_);
if (active_ == sending) {
return;
}
active_ = sending;
for (size_t i = 0; i < rtp_streams_.size(); ++i) {
RtpRtcpInterface& rtp_module = *rtp_streams_[i].rtp_rtcp;
rtp_module.SetSendingStatus(sending);
rtp_module.SetSendingMediaStatus(sending);
if (sending) {
transport_->RegisterSendingRtpStream(rtp_module);
} else {
transport_->DeRegisterSendingRtpStream(rtp_module);
}
}
auto* feedback_provider = transport_->GetStreamFeedbackProvider();
if (!sending) {
feedback_provider->DeRegisterStreamFeedbackObserver(this);
} else {
feedback_provider->RegisterStreamFeedbackObserver(rtp_config_.ssrcs, this);
}
}
bool RtpVideoSender::IsActive() {
RTC_DCHECK_RUN_ON(&transport_checker_);
MutexLock lock(&mutex_);
return IsActiveLocked();
}
bool RtpVideoSender::IsActiveLocked() {
return active_ && !rtp_streams_.empty();
}
EncodedImageCallback::Result RtpVideoSender::OnEncodedImage(
const EncodedImage& encoded_image,
const CodecSpecificInfo* codec_specific_info) {
fec_controller_->UpdateWithEncodedData(encoded_image.size(),
encoded_image._frameType);
MutexLock lock(&mutex_);
RTC_DCHECK(!rtp_streams_.empty());
if (!active_)
return Result(Result::ERROR_SEND_FAILED);
shared_frame_id_++;
size_t simulcast_index = encoded_image.SimulcastIndex().value_or(0);
RTC_DCHECK_LT(simulcast_index, rtp_streams_.size());
uint32_t rtp_timestamp =
encoded_image.RtpTimestamp() +
rtp_streams_[simulcast_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_[simulcast_index].rtp_rtcp->OnSendingRtpFrame(
encoded_image.RtpTimestamp(), 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);
}
TimeDelta expected_retransmission_time = TimeDelta::PlusInfinity();
if (encoded_image.RetransmissionAllowed()) {
expected_retransmission_time =
rtp_streams_[simulcast_index].rtp_rtcp->ExpectedRetransmissionTime();
}
if (IsFirstFrameOfACodedVideoSequence(encoded_image, codec_specific_info)) {
// In order to use the dependency descriptor RTP header extension:
// - Pass along any `FrameDependencyStructure` templates produced by the
// encoder adapter.
// - If none were produced the `RtpPayloadParams::*ToGeneric` for the
// particular codec have simulated a dependency structure, so provide a
// minimal set of templates.
// - Otherwise, don't pass along any templates at all which will disable
// the generation of a dependency descriptor.
RTPSenderVideo& sender_video = *rtp_streams_[simulcast_index].sender_video;
if (codec_specific_info && codec_specific_info->template_structure) {
sender_video.SetVideoStructure(&*codec_specific_info->template_structure);
} else if (absl::optional<FrameDependencyStructure> structure =
params_[simulcast_index].GenericStructure(
codec_specific_info)) {
sender_video.SetVideoStructure(&*structure);
} else {
sender_video.SetVideoStructure(nullptr);
}
}
absl::optional<int64_t> frame_id;
if (!independent_frame_ids_) {
frame_id = shared_frame_id_;
}
bool send_result =
rtp_streams_[simulcast_index].sender_video->SendEncodedImage(
rtp_config_.payload_type, codec_type_, rtp_timestamp, encoded_image,
params_[simulcast_index].GetRtpVideoHeader(
encoded_image, codec_specific_info, frame_id),
expected_retransmission_time);
if (frame_count_observer_) {
FrameCounts& counts = frame_counts_[simulcast_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[simulcast_index]);
}
if (!send_result)
return Result(Result::ERROR_SEND_FAILED);
return Result(Result::OK, rtp_timestamp);
}
void RtpVideoSender::OnBitrateAllocationUpdated(
const VideoBitrateAllocation& bitrate) {
RTC_DCHECK_RUN_ON(&transport_checker_);
MutexLock lock(&mutex_);
if (IsActiveLocked()) {
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::OnVideoLayersAllocationUpdated(
const VideoLayersAllocation& allocation) {
MutexLock lock(&mutex_);
if (IsActiveLocked()) {
for (size_t i = 0; i < rtp_streams_.size(); ++i) {
VideoLayersAllocation stream_allocation = allocation;
stream_allocation.rtp_stream_index = i;
rtp_streams_[i].sender_video->SetVideoLayersAllocation(
std::move(stream_allocation));
// Only send video frames on the rtp module if the encoder is configured
// to send. This is to prevent stray frames to be sent after an encoder
// has been reconfigured.
rtp_streams_[i].rtp_rtcp->SetSendingMediaStatus(
absl::c_any_of(allocation.active_spatial_layers,
[&i](const VideoLayersAllocation::SpatialLayer layer) {
return layer.rtp_stream_index == static_cast<int>(i);
}));
}
}
}
bool RtpVideoSender::NackEnabled() const {
const bool nack_enabled = rtp_config_.nack.rtp_history_ms > 0;
return nack_enabled;
}
DataRate RtpVideoSender::GetPostEncodeOverhead() const {
DataRate post_encode_overhead = DataRate::Zero();
for (size_t i = 0; i < rtp_streams_.size(); ++i) {
if (rtp_streams_[i].rtp_rtcp->SendingMedia()) {
post_encode_overhead +=
rtp_streams_[i].sender_video->PostEncodeOverhead();
}
}
return post_encode_overhead;
}
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(rtc::MakeArrayView(packet, length));
}
void RtpVideoSender::ConfigureSsrcs(
const std::map<uint32_t, RtpState>& suspended_ssrcs) {
// Configure regular SSRCs.
RTC_CHECK(ssrc_to_rtp_module_.empty());
for (size_t i = 0; i < rtp_config_.ssrcs.size(); ++i) {
uint32_t ssrc = rtp_config_.ssrcs[i];
RtpRtcpInterface* const rtp_rtcp = rtp_streams_[i].rtp_rtcp.get();
// Restore RTP state if previous existed.
auto it = suspended_ssrcs.find(ssrc);
if (it != suspended_ssrcs.end())
rtp_rtcp->SetRtpState(it->second);
ssrc_to_rtp_module_[ssrc] = rtp_rtcp;
}
// Set up RTX if available.
if (rtp_config_.rtx.ssrcs.empty())
return;
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];
RtpRtcpInterface* const rtp_rtcp = rtp_streams_[i].rtp_rtcp.get();
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::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();
// Only happens during shutdown, when RTP module is already inactive,
// so OK to call fec generator here.
if (rtp_streams_[i].fec_generator) {
absl::optional<RtpState> fec_state =
rtp_streams_[i].fec_generator->GetRtpState();
if (fec_state) {
uint32_t ssrc = rtp_config_.flexfec.ssrc;
rtp_states[ssrc] = *fec_state;
}
}
}
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();
}
return rtp_states;
}
std::map<uint32_t, RtpPayloadState> RtpVideoSender::GetRtpPayloadStates()
const {
MutexLock lock(&mutex_);
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) {
MutexLock lock(&mutex_);
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::OnBitrateUpdated(BitrateAllocationUpdate update,
int framerate) {
// Substract overhead from bitrate.
MutexLock lock(&mutex_);
size_t num_active_streams = 0;
size_t overhead_bytes_per_packet = 0;
for (const auto& stream : rtp_streams_) {
if (stream.rtp_rtcp->SendingMedia()) {
overhead_bytes_per_packet += stream.rtp_rtcp->ExpectedPerPacketOverhead();
++num_active_streams;
}
}
if (num_active_streams > 1) {
overhead_bytes_per_packet /= num_active_streams;
}
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 = update.target_bitrate.bps();
if (has_packet_feedback_) {
DataRate overhead_rate =
CalculateOverheadRate(update.target_bitrate, max_total_packet_size,
packet_overhead, Frequency::Hertz(framerate));
// TODO(srte): We probably should not accept 0 payload bitrate here.
payload_bitrate_bps = rtc::saturated_cast<uint32_t>(payload_bitrate_bps -
overhead_rate.bps());
}
// Get the encoder target rate. It is the estimated network rate -
// protection overhead.
// TODO(srte): We should multiply with 255 here.
encoder_target_rate_bps_ = fec_controller_->UpdateFecRates(
payload_bitrate_bps, framerate,
rtc::saturated_cast<uint8_t>(update.packet_loss_ratio * 256),
loss_mask_vector_, update.round_trip_time.ms());
if (!fec_allowed_) {
encoder_target_rate_bps_ = payload_bitrate_bps;
// fec_controller_->UpdateFecRates() was still called so as to allow
// `fec_controller_` to update whatever internal state it might have,
// since `fec_allowed_` may be toggled back on at any moment.
}
// Subtract post encode 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.
uint32_t post_encode_overhead_bps = std::min(
GetPostEncodeOverhead().bps<uint32_t>(), encoder_target_rate_bps_ / 2);
encoder_target_rate_bps_ -= post_encode_overhead_bps;
loss_mask_vector_.clear();
uint32_t encoder_overhead_rate_bps = 0;
if (has_packet_feedback_) {
// 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::BitsPerSec(encoder_target_rate_bps_),
max_total_packet_size - DataSize::Bytes(overhead_bytes_per_packet),
packet_overhead, Frequency::Hertz(framerate));
encoder_overhead_rate_bps = std::min(
encoder_overhead_rate.bps<uint32_t>(),
update.target_bitrate.bps<uint32_t>() - encoder_target_rate_bps_);
}
const uint32_t media_rate = encoder_target_rate_bps_ +
encoder_overhead_rate_bps +
post_encode_overhead_bps;
RTC_DCHECK_GE(update.target_bitrate, DataRate::BitsPerSec(media_rate));
// `protection_bitrate_bps_` includes overhead.
protection_bitrate_bps_ = update.target_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.rtp_rtcp->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_) {
stream.rtp_rtcp->SetFecProtectionParams(*delta_params, *key_params);
auto send_bitrate = stream.rtp_rtcp->GetSendRates();
*sent_video_rate_bps += send_bitrate[RtpPacketMediaType::kVideo].bps();
*sent_fec_rate_bps +=
send_bitrate[RtpPacketMediaType::kForwardErrorCorrection].bps();
*sent_nack_rate_bps +=
send_bitrate[RtpPacketMediaType::kRetransmission].bps();
}
return 0;
}
void RtpVideoSender::SetRetransmissionMode(int retransmission_mode) {
MutexLock lock(&mutex_);
for (const RtpStreamSender& stream : rtp_streams_) {
stream.sender_video->SetRetransmissionSetting(retransmission_mode);
}
}
void RtpVideoSender::SetFecAllowed(bool fec_allowed) {
MutexLock lock(&mutex_);
fec_allowed_ = fec_allowed;
}
void RtpVideoSender::OnPacketFeedbackVector(
std::vector<StreamPacketInfo> packet_feedback_vector) {
if (fec_controller_->UseLossVectorMask()) {
MutexLock lock(&mutex_);
for (const StreamPacketInfo& packet : packet_feedback_vector) {
loss_mask_vector_.push_back(!packet.received);
}
}
// 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 StreamPacketInfo& packet : packet_feedback_vector) {
if (packet.received && packet.ssrc) {
acked_packets_per_ssrc[*packet.ssrc].push_back(
packet.rtp_sequence_number);
}
}
// 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 StreamPacketInfo& packet : packet_feedback_vector) {
// Only include new media packets, not retransmissions/padding/fec.
if (!packet.received && packet.ssrc && !packet.is_retransmission) {
// 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_module_.find(ssrc);
RTC_CHECK(it != ssrc_to_rtp_module_.end());
RTPSender* rtp_sender = it->second->RtpSender();
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_module_.find(ssrc);
if (it == ssrc_to_rtp_module_.end()) {
// No media, likely FEC or padding. 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);
}
DataRate RtpVideoSender::CalculateOverheadRate(DataRate data_rate,
DataSize packet_size,
DataSize overhead_per_packet,
Frequency framerate) const {
Frequency packet_rate = data_rate / packet_size;
if (use_frame_rate_for_overhead_) {
framerate = std::max(framerate, Frequency::Hertz(1));
DataSize frame_size = data_rate / framerate;
int packets_per_frame = ceil(frame_size / packet_size);
packet_rate = packets_per_frame * framerate;
}
return packet_rate.RoundUpTo(Frequency::Hertz(1)) * overhead_per_packet;
}
} // namespace webrtc
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