1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31 32 33 34 35 36 37 38 39 40 41 42 43 44 45 46 47 48 49 50 51 52 53 54 55 56 57 58 59 60 61 62 63 64 65 66 67 68 69 70 71 72 73 74 75 76 77 78 79 80 81 82 83 84 85 86 87 88 89 90 91 92 93 94 95 96 97 98 99 100 101 102 103 104 105 106 107 108 109 110 111 112 113 114 115 116 117 118 119 120 121 122 123 124 125 126 127 128 129 130 131 132 133 134 135 136 137 138 139 140 141 142 143 144 145 146 147 148 149 150 151 152 153 154 155 156 157 158 159 160 161 162 163 164 165 166 167 168 169 170 171 172 173 174 175 176 177 178 179 180 181 182 183 184 185 186 187 188 189 190 191 192 193 194 195 196 197 198 199 200 201 202 203 204 205 206 207 208 209 210 211 212 213 214 215 216 217 218 219 220 221 222 223 224 225 226 227 228 229 230 231 232 233 234 235 236 237 238 239 240 241 242 243 244 245 246 247 248 249 250 251 252 253 254 255 256 257 258 259 260 261 262 263 264 265 266 267 268 269 270 271 272 273 274 275 276 277 278 279 280 281 282 283 284 285 286 287 288 289 290 291 292 293 294 295 296 297 298 299 300 301 302 303 304 305 306 307 308 309 310 311 312 313 314 315 316 317 318 319 320 321 322 323 324 325 326 327 328 329 330 331 332 333 334 335 336 337 338 339 340 341 342 343 344 345 346 347 348 349 350 351 352 353 354 355 356 357 358 359 360 361 362 363 364 365 366 367 368 369 370 371 372 373 374 375 376 377 378 379 380 381 382 383 384 385 386 387 388 389 390 391 392 393 394 395 396 397 398 399 400 401 402 403 404 405 406 407 408 409 410 411 412 413 414 415 416 417 418 419 420 421 422 423 424 425 426 427 428 429 430 431 432 433 434 435 436 437 438 439 440 441 442 443 444 445 446 447 448 449 450 451 452 453 454 455 456 457 458 459 460 461 462 463 464 465 466 467 468 469 470 471 472 473 474 475 476 477 478 479 480 481 482 483 484 485 486 487 488 489 490 491 492 493 494 495 496 497 498 499 500 501 502 503 504 505 506 507 508 509 510 511 512 513 514 515 516 517 518 519 520 521 522 523 524 525 526 527 528 529 530 531 532 533 534 535 536 537 538 539 540 541 542 543 544 545 546 547 548 549 550 551 552 553 554 555 556 557 558 559 560 561 562 563 564 565 566 567 568 569 570 571 572 573 574 575 576 577 578 579 580 581 582 583 584 585 586 587 588 589 590 591 592 593 594 595 596 597 598 599 600 601 602 603 604 605 606 607 608 609 610 611 612 613 614 615 616 617 618 619 620 621 622 623 624 625 626 627 628 629 630 631 632 633 634 635 636 637 638 639 640 641 642 643 644 645 646 647 648 649 650 651 652 653 654 655 656 657 658 659 660 661 662 663 664 665 666 667 668 669 670 671 672 673 674 675 676 677 678 679 680 681 682 683 684 685 686 687 688 689 690 691 692 693 694 695 696 697 698 699 700 701 702 703 704 705 706 707 708 709 710 711 712 713 714 715 716 717 718 719 720 721 722 723 724 725 726 727 728 729 730 731 732 733 734 735 736 737 738 739 740 741 742 743 744 745 746 747 748 749 750 751 752 753 754 755 756 757 758 759 760 761 762 763 764 765 766 767 768 769 770 771 772 773 774 775 776 777 778 779 780 781 782 783 784 785 786 787 788 789 790 791 792 793 794 795 796 797 798 799 800 801 802 803 804 805 806 807 808 809 810 811 812 813 814 815 816 817 818 819 820 821 822 823 824 825 826 827 828 829 830 831 832 833 834 835 836 837 838 839 840 841 842 843 844 845 846 847 848 849 850 851 852 853 854 855 856 857 858 859 860 861 862 863 864 865 866 867 868 869 870 871 872 873 874 875 876 877 878 879 880 881 882 883 884 885 886 887 888 889 890 891 892 893 894 895 896 897 898 899 900 901 902 903 904 905 906 907 908 909 910 911 912 913 914 915 916 917 918 919 920 921 922 923 924 925 926 927 928 929 930 931 932 933 934 935 936 937 938 939 940 941 942 943 944 945 946 947 948 949 950 951 952 953 954 955 956 957 958 959 960 961 962 963 964 965 966 967 968 969 970 971 972 973 974 975 976 977 978 979 980 981 982 983 984 985 986 987 988 989 990 991 992 993 994 995 996 997 998 999 1000 1001 1002 1003 1004 1005 1006 1007 1008 1009 1010 1011 1012 1013 1014 1015 1016 1017 1018 1019 1020 1021 1022 1023 1024 1025 1026 1027 1028 1029 1030 1031 1032 1033 1034 1035 1036 1037 1038 1039 1040 1041 1042 1043 1044 1045 1046 1047 1048
|
/*
* 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 <cmath>
#include <cstddef>
#include <cstdint>
#include <map>
#include <memory>
#include <optional>
#include <string>
#include <utility>
#include <vector>
#include "absl/algorithm/container.h"
#include "absl/base/nullability.h"
#include "absl/strings/match.h"
#include "absl/strings/string_view.h"
#include "api/array_view.h"
#include "api/call/bitrate_allocation.h"
#include "api/crypto/crypto_options.h"
#include "api/environment/environment.h"
#include "api/fec_controller.h"
#include "api/field_trials_view.h"
#include "api/frame_transformer_interface.h"
#include "api/rtp_headers.h"
#include "api/rtp_parameters.h"
#include "api/scoped_refptr.h"
#include "api/sequence_checker.h"
#include "api/task_queue/pending_task_safety_flag.h"
#include "api/task_queue/task_queue_base.h"
#include "api/transport/rtp/dependency_descriptor.h"
#include "api/units/data_rate.h"
#include "api/units/data_size.h"
#include "api/units/frequency.h"
#include "api/units/time_delta.h"
#include "api/video/encoded_image.h"
#include "api/video/video_bitrate_allocation.h"
#include "api/video/video_codec_type.h"
#include "api/video/video_frame_type.h"
#include "api/video/video_layers_allocation.h"
#include "api/video_codecs/video_codec.h"
#include "api/video_codecs/video_encoder.h"
#include "call/rtp_config.h"
#include "call/rtp_payload_params.h"
#include "call/rtp_transport_controller_send_interface.h"
#include "common_video/frame_counts.h"
#include "common_video/generic_frame_descriptor/generic_frame_info.h"
#include "modules/include/module_fec_types.h"
#include "modules/pacing/packet_router.h"
#include "modules/rtp_rtcp/include/flexfec_sender.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/rtp_rtcp/source/rtp_sender_video.h"
#include "modules/rtp_rtcp/source/rtp_sequence_number_map.h"
#include "modules/rtp_rtcp/source/ulpfec_generator.h"
#include "modules/rtp_rtcp/source/video_fec_generator.h"
#include "modules/video_coding/include/video_codec_interface.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "rtc_base/synchronization/mutex.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(
const Environment& env,
const RtpConfig& rtp,
const std::map<uint32_t, RtpState>& suspended_ssrcs,
int simulcast_index) {
// 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>(
env, rtp.flexfec.payload_type, rtp.flexfec.ssrc,
rtp.flexfec.protected_media_ssrcs[0], rtp.mid, rtp.extensions,
RTPSender::FecExtensionSizes(), rtp_state);
} else if (rtp.ulpfec.red_payload_type >= 0 &&
rtp.ulpfec.ulpfec_payload_type >= 0 &&
!ShouldDisableRedAndUlpfec(/*flexfec_enabled=*/false, rtp,
env.field_trials())) {
// Flexfec not configured, but ulpfec is and is not disabled.
return std::make_unique<UlpfecGenerator>(env, rtp.ulpfec.red_payload_type,
rtp.ulpfec.ulpfec_payload_type);
}
// Not a single FEC is given.
return nullptr;
}
std::vector<RtpStreamSender> CreateRtpStreamSenders(
const Environment& env,
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,
RateLimiter* retransmission_rate_limiter,
FrameEncryptorInterface* frame_encryptor,
const CryptoOptions& crypto_options,
scoped_refptr<FrameTransformerInterface> frame_transformer) {
RTC_DCHECK_GT(rtp_config.ssrcs.size(), 0);
RtpRtcpInterface::Configuration configuration;
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;
if (env.field_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.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(env, rtp_config, suspended_ssrcs, i);
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;
auto rtp_rtcp = std::make_unique<ModuleRtpRtcpImpl2>(env, configuration);
rtp_rtcp->SetSendingStatus(false);
rtp_rtcp->SetSendingMediaStatus(false);
rtp_rtcp->SetRTCPStatus(RtcpMode::kCompound);
// Set NACK.
rtp_rtcp->SetStorePacketsStatus(true, kMinSendSidePacketHistorySize);
video_config.clock = &env.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 = &env.field_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, env.field_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 = &env.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;
}
std::optional<VideoCodecType> GetVideoCodecType(const RtpConfig& config,
size_t simulcast_index) {
auto stream_config = config.GetStreamConfig(simulcast_index);
if (stream_config.raw_payload) {
return std::nullopt;
}
return PayloadStringToCodecType(stream_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(
const Environment& env,
TaskQueueBase* absl_nonnull transport_queue,
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,
RateLimiter* retransmission_limiter,
std::unique_ptr<FecController> fec_controller,
FrameEncryptorInterface* frame_encryptor,
const CryptoOptions& crypto_options,
scoped_refptr<FrameTransformerInterface> frame_transformer)
: env_(env),
use_frame_rate_for_overhead_(absl::StartsWith(
env.field_trials().Lookup("WebRTC-Video-UseFrameRateForOverhead"),
"Enabled")),
has_packet_feedback_(TransportSeqNumExtensionConfigured(rtp_config)),
transport_queue_(*transport_queue),
active_(false),
fec_controller_(std::move(fec_controller)),
fec_allowed_(true),
rtp_streams_(CreateRtpStreamSenders(env,
rtp_config,
observers,
rtcp_report_interval_ms,
send_transport,
transport,
suspended_ssrcs,
retransmission_limiter,
frame_encryptor,
crypto_options,
std::move(frame_transformer))),
rtp_config_(rtp_config),
transport_(transport),
independent_frame_ids_(
env.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),
safety_(PendingTaskSafetyFlag::CreateAttachedToTaskQueue(
/*alive=*/true,
transport_queue)) {
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, env.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 (size_t i = 0; i < rtp_streams_.size(); i++) {
const RtpStreamSender& stream = rtp_streams_[i];
// 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_.GetStreamConfig(i).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 (const RtpStreamSender& stream : rtp_streams_) {
SetModuleIsActive(sending, *stream.rtp_rtcp);
}
auto* feedback_provider = transport_->GetStreamFeedbackProvider();
if (!sending) {
feedback_provider->DeRegisterStreamFeedbackObserver(this);
} else {
feedback_provider->RegisterStreamFeedbackObserver(rtp_config_.ssrcs, this);
}
}
void RtpVideoSender::SetModuleIsActive(bool sending,
RtpRtcpInterface& rtp_module) {
if (rtp_module.SendingMedia() == sending) {
return;
}
rtp_module.SetSendingStatus(sending);
rtp_module.SetSendingMediaStatus(sending);
if (sending) {
transport_->RegisterSendingRtpStream(rtp_module);
} else {
transport_->DeRegisterSendingRtpStream(rtp_module);
}
}
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_.GetStreamConfig(simulcast_index).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 (std::optional<FrameDependencyStructure> structure =
params_[simulcast_index].GenericStructure(
codec_specific_info)) {
sender_video.SetVideoStructure(&*structure);
} else {
sender_video.SetVideoStructure(nullptr);
}
}
std::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_.GetStreamConfig(simulcast_index).payload_type,
GetVideoCodecType(rtp_config_, simulcast_index), 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<std::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.
// Reconfiguration of the RtpRtcp modules must happen on the transport queue
// to avoid races with batch sending of packets.
std::vector<bool> sending(rtp_streams_.size(), false);
for (const VideoLayersAllocation::SpatialLayer& layer :
allocation.active_spatial_layers) {
if (layer.rtp_stream_index < static_cast<int>(sending.size())) {
sending[layer.rtp_stream_index] = true;
}
}
transport_queue_.PostTask(
SafeTask(safety_.flag(), [this, sending = std::move(sending)] {
RTC_DCHECK_RUN_ON(&transport_checker_);
RTC_CHECK_EQ(sending.size(), rtp_streams_.size());
for (size_t i = 0; i < sending.size(); ++i) {
SetModuleIsActive(sending[i], *rtp_streams_[i].rtp_rtcp);
}
}));
}
}
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(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 (size_t i = 0; i < rtp_streams_.size(); ++i) {
const RtpStreamSender& stream = rtp_streams_[i];
RtpStreamConfig stream_config = rtp_config_.GetStreamConfig(i);
RTC_DCHECK(stream_config.rtx);
stream.rtp_rtcp->SetRtxSendPayloadType(stream_config.rtx->payload_type,
stream_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) {
std::optional<RtpState> fec_state =
rtp_streams_[i].fec_generator->GetRtpState();
if (fec_state) {
uint32_t fec_ssrc = rtp_config_.flexfec.ssrc;
rtp_states[fec_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 =
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,
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,
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;
}
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
|