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Nagram/TMessagesProj/jni/voip/webrtc/call/rtp_video_sender.cc
xaxtix d30f796d8c Update to 8.6.0
2022-03-11 19:49:54 +03:00

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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 "api/array_view.h"
#include "api/transport/field_trial_based_config.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/location.h"
#include "rtc_base/logging.h"
#include "rtc_base/task_queue.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(const std::string& payload_name,
const WebRtcKeyValueConfig& trials) {
const VideoCodecType codecType = PayloadStringToCodecType(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 WebRtcKeyValueConfig& 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 WebRtcKeyValueConfig& 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,
RtcpBandwidthObserver* bandwidth_callback,
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 WebRtcKeyValueConfig& trials) {
RTC_DCHECK_GT(rtp_config.ssrcs.size(), 0);
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.bandwidth_callback = bandwidth_callback;
configuration.network_state_estimate_observer =
transport->network_state_estimate_observer();
configuration.transport_feedback_callback =
transport->transport_feedback_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_side_delay_observer = observers.send_delay_observer;
configuration.send_packet_observer = observers.send_packet_observer;
configuration.event_log = event_log;
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;
std::vector<RtpStreamSender> rtp_streams;
RTC_DCHECK(rtp_config.rtx.ssrcs.empty() ||
rtp_config.rtx.ssrcs.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.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.enable_retransmit_all_layers = false;
video_config.field_trials = &trials;
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.send_transport_queue = transport->GetWorkerQueue()->Get();
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)
: send_side_bwe_with_overhead_(!absl::StartsWith(
field_trials_.Lookup("WebRTC-SendSideBwe-WithOverhead"),
"Disabled")),
use_frame_rate_for_overhead_(absl::StartsWith(
field_trials_.Lookup("WebRTC-Video-UseFrameRateForOverhead"),
"Enabled")),
has_packet_feedback_(TransportSeqNumExtensionConfigured(rtp_config)),
simulate_generic_structure_(absl::StartsWith(
field_trials_.Lookup("WebRTC-GenericCodecDependencyDescriptor"),
"Enabled")),
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->GetBandwidthObserver(),
transport,
suspended_ssrcs,
event_log,
retransmission_limiter,
frame_encryptor,
crypto_options,
std::move(frame_transformer),
field_trials_)),
rtp_config_(rtp_config),
codec_type_(GetVideoCodecType(rtp_config)),
transport_(transport),
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 (send_side_bwe_with_overhead_ && 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);
ConfigureRids();
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() {
// TODO(bugs.webrtc.org/13517): Remove once RtpVideoSender gets deleted on the
// transport task queue.
transport_checker_.Detach();
SetActiveModulesLocked(
std::vector<bool>(rtp_streams_.size(), /*active=*/false));
RTC_DCHECK(!registered_for_feedback_);
}
void RtpVideoSender::SetActive(bool active) {
RTC_DCHECK_RUN_ON(&transport_checker_);
MutexLock lock(&mutex_);
if (active_ == active)
return;
const std::vector<bool> active_modules(rtp_streams_.size(), active);
SetActiveModulesLocked(active_modules);
auto* feedback_provider = transport_->GetStreamFeedbackProvider();
if (active && !registered_for_feedback_) {
feedback_provider->RegisterStreamFeedbackObserver(rtp_config_.ssrcs, this);
registered_for_feedback_ = true;
} else if (!active && registered_for_feedback_) {
feedback_provider->DeRegisterStreamFeedbackObserver(this);
registered_for_feedback_ = false;
}
}
void RtpVideoSender::SetActiveModules(const std::vector<bool> active_modules) {
RTC_DCHECK_RUN_ON(&transport_checker_);
MutexLock lock(&mutex_);
return SetActiveModulesLocked(active_modules);
}
void RtpVideoSender::SetActiveModulesLocked(
const std::vector<bool> active_modules) {
RTC_DCHECK_RUN_ON(&transport_checker_);
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;
}
RtpRtcpInterface& rtp_module = *rtp_streams_[i].rtp_rtcp;
const bool was_active = rtp_module.SendingMedia();
const bool should_be_active = active_modules[i];
// Sends a kRtcpByeCode when going from true to false.
rtp_module.SetSendingStatus(active_modules[i]);
if (was_active && !should_be_active) {
// Disabling media, remove from packet router map to reduce size and
// prevent any stray packets in the pacer from asynchronously arriving
// to a disabled module.
transport_->packet_router()->RemoveSendRtpModule(&rtp_module);
}
// If set to false this module won't send media.
rtp_module.SetSendingMediaStatus(active_modules[i]);
if (!was_active && should_be_active) {
// Turning on media, register with packet router.
transport_->packet_router()->AddSendRtpModule(&rtp_module,
/*remb_candidate=*/true);
}
}
}
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 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());
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();
}
if (IsFirstFrameOfACodedVideoSequence(encoded_image, codec_specific_info)) {
// If encoder adapter produce FrameDependencyStructure, pass it so that
// dependency descriptor rtp header extension can be used.
// If not supported, disable using dependency descriptor by passing nullptr.
RTPSenderVideo& sender_video = *rtp_streams_[stream_index].sender_video;
if (codec_specific_info && codec_specific_info->template_structure) {
sender_video.SetVideoStructure(&*codec_specific_info->template_structure);
} else if (codec_specific_info &&
codec_specific_info->codecType == kVideoCodecVP9) {
const CodecSpecificInfoVP9& vp9 = codec_specific_info->codecSpecific.VP9;
FrameDependencyStructure structure =
RtpPayloadParams::MinimalisticStructure(vp9.num_spatial_layers,
kMaxTemporalStreams);
if (vp9.ss_data_available && vp9.spatial_layer_resolution_present) {
for (size_t i = 0; i < vp9.num_spatial_layers; ++i) {
structure.resolutions.emplace_back(vp9.width[i], vp9.height[i]);
}
}
sender_video.SetVideoStructure(&structure);
} else if (simulate_generic_structure_ && codec_specific_info &&
codec_specific_info->codecType == kVideoCodecGeneric) {
FrameDependencyStructure structure =
RtpPayloadParams::MinimalisticStructure(
/*num_spatial_layers=*/1,
/*num_temporal_layers=*/1);
sender_video.SetVideoStructure(&structure);
} else {
sender_video.SetVideoStructure(nullptr);
}
}
bool send_result = rtp_streams_[stream_index].sender_video->SendEncodedImage(
rtp_config_.payload_type, codec_type_, rtp_timestamp, encoded_image,
params_[stream_index].GetRtpVideoHeader(
encoded_image, codec_specific_info, shared_frame_id_),
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_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));
}
}
}
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(
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::ConfigureRids() {
if (rtp_config_.rids.empty())
return;
// 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.size() >= rtp_streams_.size());
for (size_t i = 0; i < rtp_streams_.size(); ++i) {
rtp_streams_[i].rtp_rtcp->SetRid(rtp_config_.rids[i]);
}
}
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 (send_side_bwe_with_overhead_ && 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 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.
uint32_t 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_ && 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_);
}
// 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(update.target_bitrate, DataRate::BitsPerSec(media_rate));
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::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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